ETERNITY V11 System Manual PDF
ETERNITY V11 System Manual PDF
ETERNITY V11 System Manual PDF
System Manual
ETERNITY
The IP-PBX with Seamless Mobility
and Universal Connectivity
System Manual
Documentation Disclaimer
Matrix Comsec reserves the right to make changes in the design or components of the product as engineering and
manufacturing may warrant. Specifications are subject to change without notice.
This is a general documentation for all models of the product. The product may not support all the features and
facilities described in the documentation.
Information in this documentation may change from time to time. Matrix Comsec reserves the right to revise
information in this publication for any reason without prior notice. Matrix Comsec makes no warranties with respect
to this documentation and disclaims any implied warranties. While every precaution has been taken in the
preparation of this system manual, Matrix Comsec assumes no responsibility for errors or omissions. Neither is any
liability assumed for damages resulting from the use of the information contained herein.
Neither Matrix Comsec nor its affiliates shall be liable to the purchaser of this product or third parties for damages,
losses, costs or expenses incurred by the purchaser or third parties as a result of: accident, misuse or abuse of this
product or unauthorized modifications, repairs or alterations to this product or failure to strictly comply with Matrix
Comsec's operating and maintenance instructions.
Copyright
All rights reserved. No part of this system manual may be copied or reproduced in any form or by any means
without the prior written consent of Matrix Comsec.
Version 11
Release date: October 11, 2013
Contents
Introduction..................................................................................................................................................... 1
Welcome ............................................................................................................................................................. 1
About this System Manual .................................................................................................................................. 1
Know Your ETERNITY .................................................................................................................................... 5
Applications of ETERNITY .................................................................................................................................. 8
Models of ETERNITY ........................................................................................................................................ 15
Hardware Overview ........................................................................................................................................... 18
The Interfaces ................................................................................................................................................... 26
Installing ETERNITY ..................................................................................................................................... 35
Before You Start ................................................................................................................................................ 35
Protecting ETERNITY and Yourself .................................................................................................................. 42
Installing ETERNITY LE................................................................................................................................ 47
The Power Supply Card .................................................................................................................................... 51
The CPU Card ................................................................................................................................................... 53
The Single Line Telephone Card ...................................................................................................................... 57
The Digital Key Phone Card .............................................................................................................................. 67
The BRI Card .................................................................................................................................................... 84
The CO Card ................................................................................................................................................... 100
The T1E1PRI Card .......................................................................................................................................... 106
The E&M Card ................................................................................................................................................ 112
The Magneto Card .......................................................................................................................................... 122
The Mobile Card .............................................................................................................................................. 127
The VoIP Card ................................................................................................................................................ 131
SIP Extensions ................................................................................................................................................ 138
The Voice Mail System Card ........................................................................................................................... 157
Installing ETERNITY ME ............................................................................................................................. 163
The Power Supply Card .................................................................................................................................. 167
The Master Card ............................................................................................................................................. 171
The Switch Card-V2R2 ................................................................................................................................... 179
The Switch Card - V2R3 and Later ................................................................................................................. 184
The Single Line Telephone Card .................................................................................................................... 189
The Intercom Line Card .................................................................................................................................. 196
The Digital Key Phone Card ............................................................................................................................ 199
The BRI Card .................................................................................................................................................. 216
The CO Card ................................................................................................................................................... 232
Table of Contents
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Table of Contents
Table of Contents
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Table of Contents
Table of Contents
Table of Contents
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viii
Table of Contents
CHAPTER 1
Introduction
Welcome
Thank you for choosing the Matrix ETERNITY! We hope you will make optimum use of this intelligent, integrated
Voice Switch. Please read this document carefully to get acquainted with the product before installing and operating
it.
Intended Audience
This System Manual is aimed at:
System Engineers, who will install, maintain and support the ETERNITY. System Engineers are persons
who customize the system configuration to meet the requirements of the organization/users. It is assumed
that they are experienced in installing PBX, are familiar with telecom wiring technology, how it works, and
the various technical terms and functions associated with it. The SE must have undergone training in the
configuring ETERNITY.
No one, other than the System Engineer is permitted to make any alterations to the configuration of the
ETERNITY.
System Administrators, who are persons who will administer the ETERNITY. Generally an operator/
receptionist in an organization, or the staff manning the reception or front desk area of the establishment
are selected as System Administrators.
It is assumed that the System Administrators have some previous experience in administering a PBX and
its Terminals and Consoles. The System Administrators are not expected to install and program the PBX,
but only the routine jobs and features that are specific to them like generating SMDR reports, Setting
report filters, configuring the features, Setting Alarms, reminders, etc.
Users, persons/organizations who will use the resources of the ETERNITY. They may be executives,
include personnel of small and medium businesses, large enterprises, front desk and service staff of
Hotels/Motels, hospitals, and other commercial and public organizations/institutions.
Chapter 1: Introduction - gives an overview of this document, its purpose, intended audience,
organization, terms and conventions used to present information and instructions.
Chapter 2: Know Your ETERNITY - describes the system and its design, application scenarios, the
available models, the interfaces, and the hardware.
Chapter 3: Installing ETERNITY - gives step-by-step instructions for preparing for and installing the
ETERNITY in general, like setting up the main distribution frames for the wiring, the safety measures for
protecting the system and persons handling the installation and maintenance.
Chapter 4: Installing ETERNITY ME - provides step-by-step instructions for installing the ETERNITY ME
and its variants, inserting the cards, connecting the cables and powering the system.
Chapter 5: Installing ETERNITY GE - contains instructions for installing the ETERNITY GE and its
variants, inserting the cards, connecting the cables lines and powering the system.
Chapter 6: Installing ETERNITY PE - describes the installation of the ETERNITY PE and its variants,
inserting the cards, connecting the cables lines and powering the system.
Chapter 7: Configuring ETERNITY - contains description of the different tools and options available to
configure ETERNITY. It provides detailed description of how to configure the various extension and trunk
port types - SLT, DKP, ISDN Terminal, SIP Extensions, CO, Mobile, VoIP-SIP, T1E1PRI, BRI, E&M and
Magneto, Virtual Extensions - supported by ETERNITY.
Chapter 8: Features and Facilities - describes in detail, each feature and facility offered by the
ETERNITY. This includes a description of the feature/facility, how it works, and how to program the feature/
facility.
The feature description is arranged alphabetically by Feature Name to make it easy for you to locate the
description you want to look up.
Instructions
The instructions in this document are written in a numbered, step-by-step format, as follows. Each step, its outcome
and indication/notification, wherever they occur, have been described.
Access Codes
Access codes are strings of digits dialed by an extension to
The Access Codes provided in the instructions throughout this document, are default access codes. It is possible to
change the Access Codes according to user requirement and preferences. Verify with the Installer/System
Engineer, if the default Access Codes have been changed, and use the codes programmed by the System
Engineer. For more information, read the topic Access Codes in this document.
Notices
The following symbols have been used for notices to draw your attention to important items.
Important: to indicate something that requires your special attention or to remind you of
something you might need to do when you are using the system.
Caution: to indicate an action or condition that is likely to result in malfunction or damage to the
system or your property.
Warning: to indicate a hazard or an action that will cause damage to the system and or cause
bodily harm to the user.
Tip: to indicate a helpful hint giving you an alternative way to operate the system or carry out a
procedure, or use a feature more efficiently.
CO Lines: The lines subscribed from the CO Network. These may be Two-wire Trunk Lines, ISDN BRI,
ISDN PRI, etc.
Digital Key Phone (DKP): refers to EON, the proprietary digital key phone of Matrix supplied with the
ETERNITY. The term 'Digital Key Phone' refers to all models of EON.
Enterprise Application/Features: pertaining to the general and special telephone and call management
features required by business establishments, public and private organizations.
Extension: it is the port of the PBX to which a telephone instrument (DKP/SLT/ISDN) is connected.
External Calls: calls made by users of ETERNITY to subscribers of PSTN, PLMN, ITSPs, etc.
External Numbers: numbers of parties/individuals outside the PBX or PBX network. The unique number
string given to subscribers of PSTN, PLMN, ITSP, etc.
Hospitality Application/ Features: pertaining to the special telephone and guest/patient management
features required by accommodation establishments like hotels and hospitals.
Internal Calls: calls made from and received by one extension to another extension of the ETERNITY.
Mobile Extension: A mobile/landline phone used as a remote extension of ETERNITY. You can access all
the features of an extension of ETERNITY from the mobile/landline phone.
Port: the physical interfaces on the cards for trunk lines and extension lines.
Service Provider: the providers of telecom network lines/Internet - POTS, PSTN, GSM, ISDN PRI, ISDN
BRI, and Internet Telephony Service Providers (ITSP).
Single Line Telephone (SLT): any standard two-wire telephone attached as extensions of the ETERNITY.
System Administrator Commands/SA Commands: number strings dialed from the System
Administrator access/mode to operate features or set/cancel features for other extensions.
System Commands/SE Commands: number strings dialed from the System Engineer access/mode to
program the system features/functions.
CO trunks: Two-wire trunks, that is, analog trunk lines from the POTS network.
Using this Manual, we hope, you will be able to set up, operate and make optimum use of this feature packed PBX.
If you encounter any technical problems, please contact your Dealer/reseller or the Matrix Support team.
CHAPTER 2
Introduction
The Matrix ETERNITY is an Integrated Enterprise Voice Switch expandable up to 516 user ports. It is a unique
convergence of innovative switching technology and intelligent software features.
The system is built on PCM/TDM, 100 percent non-blocking, digital technology, providing high density switching. It
is powered by a 32-bit RISC processor for distributed processing. Thus the system offers reliable, efficient, and
unrestricted simultaneous communication (incoming and outgoing) by all users.
Universal Connectivity
ETERNITY offers Universal Connectivity, working with all major telecom interfaces: POTS, ISDN BRI/PRI, T1/E1,
GSM/3G, VoIP, E&M, and Magneto. So, you have access to multiple telecom networks on a single platform. The
system's intelligent Least Cost Routing logic diverts your calls through the appropriate network, ensuring least
possible call cost.
Besides Operator consoles (Digital Key phones, Direct Station Selection consoles) and standard telephones, you
can interface ETERNITY with different types of external devices such as a Fax machine, an external music source,
a public address system.
You can operate security devices with the ETERNITY. Any sensor device such as a smoke detector, an object
sensor, a glass break detector, can be connected to the ETERNITY to instigate a hooter, siren connected to the
ETERNITY.
You can also operate several automated control applications such as a door lock, lights glow signboards, bells,
water pump, and the like.
A door phone can also be connected to screen visitors.
ETERNITY supports video conferencing and data connectivity. You can interface any standard video phone
conference unit with suitable ISDN Interface with ETERNITY ISDN T1E1PRI and BRI cards.
ETERNITY supports Q-Sig. allowing you to network ETERNITY with another PBX/ETERNITY. So, you have feature
transparency and a network of PBXs working as a single unit.
The system is also designed to provide very high level of flexibility and scalability to meet your future
communication needs. The Universal Slot platform and the modular design of the cards allow you to start with the
minimum required configuration and expand the system capacity later, by adding more cards to the universal slots.
So, you can invest progressively in scaling up the system as the communication needs of your organization grow.
Redundancy
To reduce down time and provide uninterrupted communication, the ETERNITY ME supports redundancy option in
the ETERNITY ME10SR variant for the three cards that are critical to its functioning: The Power Supply Card1, the
Master Card and the Switch Card. There are two cards of each on the ETERNITY ME10SR. When the active card
fails, the standby card takes over.
Hot Swap
With the Hot Swap feature2 you can remove a card and insert it back without switching off the system. So, you can
replace a faulty card with a functional one without affecting the functioning of the system.
Q-Sig
With Q-Sig. you can network ETERNITY with another ETERNITY or any other ISDN-PBX to expand the PBX
resources. You can enjoy feature transparency between the PBXs.
Key Features
1.
2.
Account Codes
Auto Attendant
Automatic Call Distribution
CAS Interface
Class of Service
CLI Based Routing
Closed User Group
Redundancy is supported only for the PS48V Power Supply card! It is not supported for the PS UNI Power Supply Card. Refer the
topic The Power Supply Card to know more. ETERNITY GE and PE do not support Redundancy.
The Hot Swap feature is supported for all cards except Power Supply card in both ETERNITY ME10S and ME16S. The ETERNITY
GE and PE variants do not support Hot Swap.
Also refer Appendix for a complete list of Hardware and Software features and technical specifications.
Applications of ETERNITY
The Matrix ETERNITY can be deployed in small to large enterprises and institutions: manufacturing units,
corporate offices, banking and financial institutions, software firms, shopping malls, hospitals, hotels-motels, in
power line carrier communication of electric utilities, call centers, in institutions and, power line carrier
communication PLCC networks, as group PBX (GPAX).
The ETERNITY can work as a Gateway with the existing telephony infrastructure - TDM PBX, IP PBX - as
Universal Gateway for Calling Card Operators, Internet Telephony Service Providers (ITSP).
Illustrated in the following are various scenarios where the ETERNITY finds application.
Enterprise Application
Hotel Application
10
11
12
13
IP User Application
14
Models of ETERNITY
The Matrix offers following models of ETERNITY: ETERNITY LE, ETERNITY ME, ETERNITY GE, ETERNITY PE,
ETERNITY MEX12S (The Military Exchange).
Each model has variants with different configurations.
ETERNITY LE
ETERNITY ME
The ETERNITY ME is designed for medium and large organizations and it available in the following variants:
ETERNITY ME16S
16 universal slots
516 user ports
Available with AC and DC power supply.
ETERNITY ME10S
10 universal slots
324 user ports
Redundancy option for Power Supply,
Master Card and Switch Card
15
ETERNITY GE
ETERNITY GE is designed for small and medium organizations. It is available in the following variants:
ETERNITY GE12S
12 universal slots
240 user ports
ETERNITY GE6S
6 universal slots
120 user ports
ETERNITY GE 3S
3 universal slots
60 user ports
ETERNITY PE
The ETERNITY PE is designed for small and growing organizations/small organizations with growth potential. It
available in the following variants:
ETERNITY PE3SSa
3 universal slots
24 user ports
a.
16
ETERNITY PE3SP
3 universal slots
24 user ports
ETERNITY PE6SP
6 universal slots
48 user ports
ETERNITY PE3SS has been discontinued. Software support is still provided through Jeeves for users who
have already installed it.
ETERNITY MEX12S
The ETERNITY MEX12S is a rugged, all weather, voice switch designed as per military standards to provide
reliable and secure telecommunication to the defence services.
ETERNITY MEX12S
12 universal slots
240 user ports
17
Hardware Overview
ETERNITY LE
The Enclosure
The enclosure of ETERNITY LE consists of 'fixed' and universal slots. The fixed slots are occupied by specific
cards - Power Supply Card, Master Card, Switch Card - and cannot be changed, whereas in the universal slots,
you can install any of the various card.
The slot connectors are located on the motherboard on the backplane of the enclosure. Each slot has guide rails for
inserting the cards.
ETERNITY LE can be Wall Mounted or placed on a table. You can also affix wheels to the system; to move it like a
trolley. ETERNITY LE has two racks. The first rack has a total of 16 slots. The first three slots from the left are fixed
slots and the remaining 13 are universal slots. The second rack has a total of 16 slots. The first slot from the left is
a fixed slot and the remaining 15 are universal slots.
The Cards
ETERNITY LE houses the following Cards:
1. CPU Card
2. Power Supply Card - PS48V
3. SLT Card
4. CO Card
5. DKP
6. E&M Card
7. VMS Card
8. BRI Card
9. T1E1PRI Card
10. GSM/3G Card
11. VoIP Card
12. Magneto Card
ETERNITY ME
The Enclosure
The enclosure of ETERNITY ME consists of 'fixed' and universal slots. The fixed slots are occupied by specific
cards - Power Supply Card, Master Card, Switch Card - and cannot be changed, whereas in the universal slots,
you can install any of the various card.
The slot connectors are located on the motherboard on the backplane of the enclosure. Each slot has guide rails for
inserting the cards.
Illustrated below are design of the enclosure and the position of the slots on each model of ETERNITY ME.
18
ETERNITY ME16S
In the ETERNITY ME16S, the extreme left slot is reserved for the Power Supply card, the extreme right is reserved
for the Master card, and the second last slot is reserved for the Switch Card. The slots between these fixed slots
are the 16 universal slots to fit the other cards.
ETERNITY ME10S
In the ETERNITY ME10S, the first three slots from extreme left slot are reserved for the Power Supply card, the
Master Card and the Switch Card respectively. The remaining slots are the 10 universal slots.
19
The ETERNITY ME10SR, which offers the redundancy option, has the same organization of the fixed and universal
slots as the ME10S variant, starting with the Power Supply Card on the extreme left. Only the number of slots
exceeds because of the presence of the second Power Supply Card, Master Card and Switch Card, provided in this
variant to support the Redundancy feature.
The Cards
ETERNITY ME houses the following Cards:
13. Master Card
14. Switch Card
15. Power Supply Cards: PSUNI or PS48V
16. SLT Card
17. CO+SLT Card
18. DKP
19. Intercom Line Card
20. E&M Card
21. VMS Card
22. BRI Card
23. T1E1PRI Card
24. GSM/3G Card
25. VoIP Card
26. Magneto Card
27. SLT8+MAG2+CO2+LD2+ENM2 Card
28. SLT8-Magneto8 Card
29. CO8-Magneto8 Card
ETERNITY GE
The Enclosure
The enclosure of ETERNITY GE has 'fixed' and universal slots. The fixed slots are occupied by specific cards Power Card and the CPU Card - and cannot be changed, whereas in the universal slots you can install any of the
various cards.
Inside the enclosure of ETERNITY GE are slot connectors located on the motherboard on the backplane of the
enclosure. Each slot has guide rails for inserting the cards.
Illustrated below are the design of the enclosures and the position of the slots in each model of ETERNITY GE.
ETERNITY GE12S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
20
ETERNITY GE6S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
ETERNITY GE3S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
The Cards
The ETERNITY GE houses the following types of cards:
1. Power Supply Card - PSUNI Card (90-265V, 47-63Hz Mains as Input AC voltage power supply).
2. CPU Card
3. SLT Card
4. CO+SLT Card
5. DKP Card
6. DKP+SLT Card
7. Intercom Line Card
8. E&M Card
9. VMS Card
10. BRI Card
11. T1E1PRI Card
12. GSM/3G Card
13. VoIP Card
ETERNITY PE
The Enclosure
The ETERNITY PE has a different design. The enclosure of ETERNITY PE consists of a top plate, which functions
as the cover and can be removed.
The Power Supply unit and the CPU are in-built, and fixed on the bottom plane of the ETERNITY PE.
The CPU card is fixed on the bottom plate of the enclosure, with the 2-row connectors for the card slots facing up.
21
Illustrated in the following are the design of the enclosures and the position of the slots in each model of ETERNITY
PE.
ETERNITY PE6SP
The Power Supply unit and the CPU are in-built, and fixed on the bottom plane of the ETERNITY PE.
Universal slots are located on the CPU. The connectors of the slots are located on the CPU.
Cards are mounted on the CPU, and secured on the three studs labeled as H1, H2, H3 on the CPU, with the
screws are provided for this purpose.
ETERNITY PE3SP
ETERNITY PE3SP is similar to PE6S, except has it has only 3 universal slots.
22
ETERNITY PE3SS3
The Cards
The ETERNITY PE houses the following Cards:
1. SLT Card
2. DKP Card
3. CO Card
4. DKP+ SLT Card
5. CO + SLT Card
6. DKP + CO Card
7. DKP + CO + SLT Card
8. BRI Card (only ETERNITY PE6SP and PE3SP)
9. T1E1PRI Card (only ETERNITY PE6SP and PE3SP)
10. GSM/3G Card
11. VoIP Card
12. VMS Card
13. Door Phone Card
ETERNITY MEX12S
The Enclosure
The enclosure of ETERNITY MEX12S consists of fixed and universal slots. The fixed slots are occupied by specific
cards - Power Supply Card, CPU Card. No other card can be installed in these slots. In the universal slots, you can
install any of the other cards.
The slot connectors are located on the motherboard on the backplane of the enclosure. Each slot has guide rails for
inserting the cards.
Illustrated below are design of the enclosure and the position of the slots of ETERNITY MEX12S.
3.
ETERNITY PE3SS has been discontinued. Software support is still provided through Jeeves for users who have already installed
it.
23
ETERNITY MEX12S
Power Supply
Cards
CPU
Cards
12 Universal Slots
The ETERNITY MEX12S which offers the redundancy for Power Supply and the CPU Card. The first four slots from
the left are reserved for the two Power Supply cards and the two CPU Cards. The remaining slots are the 12
universal slots.
The Cards
ETERNITY MEX12S houses the following Cards:
1. Power Supply Cards
2. Central Processing Units
3. SLT Card
4. CO Card
5. DKP Card
6. CO + SLT Card
7. DKP + SLT Card
8. CO + DKP + SLT Card
9. BRU Card
10. E1FO Card
11. Combo Card (CO + LD + E&M + Magento + SLT)
12. Data Card
13. GSM/3G Card
14. VoIP Card
15. VMS Card
16. Radio Card
17. E&M Card
LE
ME16S
ME10S
GE12S
GE6S
GE3S
PE6SP
PE3SP
PE3SS
MEX12S
--
Master Card
--
--
--
--
--
--
--
Switch Card
--
--
--
--
--
--
--
PSUNI Power
Supply Card
48 VDC
Power Supply
Card
24
Card Name
LE
ME16S
ME10S
GE12S
GE6S
GE3S
PE6SP
PE3SP
PE3SS
MEX12S
CPU Card
--
--
##
##
##
SLT Card
CO Card
--
--
--
CO+SLT Card
DKP Card
DKP+SLT
Card
--
--
--
DKP+CO
Card
--
--
--
--
--
--
--
DKP+CO+SL
T Card
--
--
--
--
--
--
Intercom Line
Card
--
--
--
--
--
E&M Card
--
--
--
VMS Card
BRI/BRU
Card
--
TIE1PRI Card
--
GSM/3G Card
VoIP Card
Magneto Card
--
--
--
--
--
--
SLT-Magneto
Card
--
--
--
--
--
--
--
--
CO-Magneto
Card
--
--
--
--
--
--
--
--
SLT+MAG+CO
2+LD2+ENM2
Card
--
--
--
--
--
--
--
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The Interfaces
The ETERNITY supports the following interfaces for connecting to different telecom networks, digital key phones,
standard telephones and other external devices.
The CO Interface
The CO Interface enables the ETERNITY to be connected to the POTS Network. The POTS Networks across the
world support various standards and differ in features. For example, some networks support Caller ID Presentation
using DTMF signaling, while some support Caller ID Presentation using FSK signaling; some networks offer 600
Ohms Impedance, while others offer complex impedance.
ETERNITY's versatile architecture allows it to be connected to such networks differing in their characteristics. The
CO Interface supports following features:
PRI
Robbed Bit Signaling (RBS)
Q-Signaling (QSIG)
E&M
PRI
Channel Associated Signaling (CAS)
Q-Signaling (QSIG)
E&M
4.
5.
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T1 PRI (T-Carrier) offers 23 Bearer Channels and one Signaling Channel (23B+D). It is used in North America, Japan and Korea.
E1 PRI (E-Carrier) offers 30 Bearer Channels and two Signaling Channels (30B+D). It is used in all countries, except North
America, Japan and Korea.
ISDN BRI
The ISDN BRI Interface enables ETERNITY to be connected to ISDN BRI Lines and connect ISDN BRI compatible
devices with the ETERNITY.
The ISDN BRI Interface has the following features:
Depending on the requirement, each BRI Port can be configured in the TE/NT mode.
It is possible to feed power from the ETERNITY to the terminal equipment connected to the ETERNITY (on its BRI
port configured as NT).
27
The VoIP Card has an in-built Registrar Server that allows any SIP enabled device like a Wi-Fi mobile handset or
an IP-Phone to be registered with it and function as the 'SIP Extension' of the ETERNITY. The SIP Extension users
can make and receive calls to any extension user of the ETERNITY as well as any external numbers over PSTN,
GSM, VoIP and E&M. With SIP Extensions, organizations can communicate and stay connected at the lowest cost
without any geographical restrictions.
The VoIP Interface supports adaptive jitter buffer for reducing delay and improving speech quality.
The key features of the VoIP Interface are:
6.
7.
8.
28
STUN.
VLAN.
Broad Voice Codec Selection: G.723, G.729ab, GSM FR, iLBC - 30 ms, iLBC - 20 ms, G. 711 -Law, and
G. 711 A-Law.
ETERNITY GE and ETERNITY MEX12S supports 16 SIP Trunks and ETERNITY PE supports 4 SIP Trunks.
ETERNITY ME supports 999 SIP Extensions, ETERNITY GE and ETERNITY MEX12S supports 500 SIP Extensions, and ETERNITY PE supports a maximum of 50 SIP Extensions.
ETERNITY PE supports a maximum of 16 Simultaneous Calls.
Registration of SIP Extensions from 3 different locations and Shared Call Appearance.
Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer PLCC-An Introduction to know more.
Closed User Groups, where several PBXs are connected with each other through E&M tie lines9.
PBX expansion, where two PBXs are connected with each other with E&M tie lines.
Also, refer the topics E&M Connectivity and E&M Feature Template to know more.
The E&M Interface can be programmed to provide Trunk Interface, a Subscriber (Station) Interface or both, as a
Tie Line with the dual personality of a Trunk and a Subscriber.
The E&M Interface of the ETERNITY has the following features:
Selectable E&M Trunk Seizure Type11 - Immediate, Immediate + Wink, Seizure Pulse, Seizure Pulse +
Wink, Express, and Compander Control Signal.
Selectable Address Signaling - Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
9.
10.
11.
12.
The PBXs in a Closed User Group can be connected over ISDN T1/E1 Lines as well. Refer the topic Closed User Group (CUG)
to know more.
The number of wires used to transmit audio signals.
This is the line protocol that defines how the equipment seizes the E&M trunk. Also, referred to as Start Dial Supervision Signaling
Protocol.
A magneto telephone is a local battery telephone set, in which signaling current is provided by a magneto hand generator. The
hand generator, commonly referred to as 'crank', is located on the right hand side of the telephone set and is turned to produce
energy to ring other phones or to signal the CO. The magneto, also called the generator, is used to convert the mechanical motion
via the crank to produce sufficient energy to ring other phones or to signal the CO.
29
companies (signaling emergencies, crossings, etc.), electric utilities, pipeline companies, who need to have their
networks at places that are too remote to be serviced by public telephone networks.
ETERNITY can land calls from magneto field telephones on the extensions (SLT, DKP, ISDN Terminal) of the
ETERNITY and place calls from the extensions of the ETERNITY on magneto telephones.
To know more about how the Magneto Card works, refer the topic Configuring Magneto Interface.
30
ETERNITY's Voice Mail System also forms the basis of other features like:
Conversation Recording
Call Taping
Voice-guided Wake-up Calls and Reminders
Message Wait Notification
Call Transfer to Mailbox
Call Forward to Voice Mail
Department Calls - Mailbox for Department Groups
31
A Loop Dial port is used as a tie-line to connect two PBXs over Two-wire interface, as illustrated below.
Computer
You can connect a standalone computer with the ETERNITY over the RS232 serial Communication Port of the
system.
You can also connect ETERNITY to a standalone computer or to a LAN Switch over the Ethernet Port of the
ETERNITY.
32
Security Devices
Any type of sensor device like glass break sensor, smoke detector, object sensor, etc. can be connected to the
Digital Input Port of the ETERNITY.
You can connect a Siren or a Hooter to the Digital Output Port of the ETERNITY which can be activated to indicate
emergencies.
The sensor device connected to the Digital Input Port can be used to instigate the hooter or siren connected to the
Digital Output Port.
33
Door Phone
You can connect any standard 4-wire door phones to the Door Phone ports of the ETERNITY. The door phones
can be operated in conjunction with a Door Lock connected to the Digital Output Port of the ETERNITY.
The Door Phone interface is supported only on ETERNITY PE.
For an at-a-glance view of the maximum trunks and ports available for each of the aforementioned Interface
options on the various models of ETERNITY, refer System Capacity and Resources in Appendix Technical
Specifications.
34
CHAPTER 3
Installing ETERNITY
35
The cables or trunk lines to/from the Public Telephone Exchange terminate on the line side and cross connections
(jumpers) run to the opposite (PBX) side of the MDF. From those terminals, a multi-core cable runs from a second
set of terminals into the PBX.
A multi-core cable runs from the PBX into the MDF. From the distribution frame, the smaller cables run into each
individual extension telephone outlet or socket (RJ11 or RJ45).
In a multi-storied building or on a widely spread out premises, it is common to have more than one distribution
frame, called the Intermediate Distribution Frame (IDF) on each floor, to provide the connection between the MDF
and the individual telephone wiring. IDFs function as wiring points to gather and distribute wiring. IDFs are used
when a large number of extensions are to be connected and the wire runs extend over hundreds of feet; hence the
distance is too great to economically terminate every extension individually to the MDF.
Select a suitable MDF (and IDF, if required) with the standard lead-in cable termination KRONE modules.
Ensure that the MDF complies with the local building telecom wiring Guidelines, Rules and Regulations.
The MDF is normally installed inside the building in a location and position which is free from the ingress of
dust and moisture, and which is not subject to damp or humid conditions.
This also applies to MDF installed outside the building. It must be protected from exposure to weather
conditions, dust, dampness and humidity
36
In washing or toilet facilities, boiler/plant/machine rooms or any area subject to corrosive fumes and
fluids;
In fire escape stairways;
Within a cupboard containing a fire hose reel;
Within any refrigeration room or sauna heater room;
Near any water feature or water body like fountains, sprinklers, a bath, shower or other fixed water
container, a swimming pool, paddling pool, spa pool or tub; or any area where hosing down operations
are carried out.
In a high voltage electrical switch room or near a heavy voltage transformer.
The MDF should be robust and securely attached to a permanent building element such as a wall, floor or
column. Do not mount the MDF on movable elements such as hinged panels or wheeled trolleys.
Provide adequate space around the MDF where any person is required to pass to enable safe and
convenient access to the MDF and ready escape from the vicinity under emergency conditions.
Any room containing the MDF must not require the use of a tool, key, card, number pad or the like to exit
the room. Ensure a quick hurdle-free exit from such a room.
The MDF or the enclosure in which it is located should have the provision for securing with a key, lock or
tool. External MDF should be adequately secured against vandalism and access by children or
unauthorized persons.
The MDF enclosure should be designed so as to prevent access to live parts by unqualified persons and
should be free of exposed sharp edges.
The site of installation should be well-ventilated, moisture and dust free, and not exposed to direct sunlight,
heat, excessive cold or humidity.
The site should be equidistant from all the extensions to simplify cabling network and reduce cabling costs.
The system should be installed at a height of at least 3.5 feet from the ground. Installation at this height
makes preventive or corrective maintenance tasks easy.
The system should be installed away from any source of electromagnetic noise such as any radio
equipment, heavy transformers, faulty electric chokes of tube-lights, any device having faulty coil, etc.
Selecting Cables
Select standard good quality telephone cables with 0.5 mm conductor diameter for the internal as well as
over-head cabling.
The length of the cables must not be too long. They must have minimum number of joints. This will help
you detect cable faults easily.
Maintain cable records so that cables and cross-connections on the MDF can be correctly identified and
connected. The records should be in a clear, legible and updateable format.
any of the models of the proprietary Digital Key Phone (DKP) of the EON series.
37
Any standard telephone instrument like rotary phone, Pulse/tone switchable push-button phone, Feature
phone or Cordless phone. So, you can also use your existing telephone instruments.
You are recommended to connect DKPs with DSS of the EON series for Operator/Receptionist/ Front
Desk/Senior management extensions.
The ETERNITY ME/ GE/ PE work with input voltages ranging between 100-240VAC.
Arrange for a separate power point and switch, close to the system.
Power supply for the system must be separate from other heavy electrical loads like Air-conditioners,
heaters, welding machines, electrical motors, etc.
Terminate the CO Trunk Line cables from the CO (public telephone exchange) and E&M cables into the
'Trunk Lines' side of the MDF using the punch tool for Krone modules.
Terminate all the extension cables (connected to the wall sockets/outlets) into the 'Station Lines' side of
the MDF using the punch tool for Krone modules.
Label the trunk and extension line cables for easy identification and keep a record of the trunk and
extension lines in an updatable format.
Where multiple wiring cabinets/distribution frames are used, label each frame and reference its number on
the corresponding outlet.
Install Primary Protection modules with Gas Discharge Tubes (GDT) and fuses on entry points for all trunk
lines. This is to protect the system from heavy voltages from trunk lines and overhead stations.
The product warranty does not cover damages resulting from lack of primary protection on trunk lines.
It is recommended that you also install Primary Protection modules with GDT and fuses on all Extension
lines, particularly off-premise extensions, and E&M ports.
For this, you are recommended to use the Primary Protection Module (PPM4) supplied by Matrix.
38
A typical connection between a PBX and the MDF is illustrated in the figure below.
If you are using a smaller configuration of the ETERNITY, like ETERNITY PE3SS, PE3SP or 6SP, you may
refer the following diagram to connect the PBX and the Distribution Frame.
You are recommended to use the Primary Protection Module - PPM4 supplied by Matrix.
39
Installing PPM4
Refer the block diagram above for the location of the PPM4.
the PPM4. Also, take into consideration the length of the cables of the PPM4
.
40
3. Use the Mounting Template supplied with the PPM4 to drill holes on the wall to fix the PPM in the selected
location. Fix the screws supplied with the PPM4 into the drilled holes, with their heads protruding from the
wall.
4. You may mount the PPM4 first and connect the cables OR you may connect the cables first and then
and P4.
8. To do so, strip off about half a centimeter of the insulation of the wire ends of the first pair of CO Trunk you
in each opening.
11. Release pressure on the levers. Both wires will be held in place by spring clamp action.
Color
P1
P2
P3
P4
14. Replace the cover of the PPM4 by pressing back the snap fits on both sides.
15. Mount the PPM, if not done already.
41
Location
near a water source like a wash bowl, kitchen sink, laundry tub, near a swimming pool, or in a wet
basement.
In places where dust, oil, corrosive fumes may come in contact with the system.
Any area where it is exposed to direct sunlight, heat, excessive cold or humidity.
On moveable or unstable surfaces, which may cause the product to fall and get damaged.
Any area where shocks or vibration are frequent or strong.
Near High-Frequency generating devices such as Electric Welder, Sewing Machine or and Microwave
Oven.
Do not leave cables exposed on the ground where they may be trampled upon, or get damaged by
entangling with feet or pressure from other heavy objects.
Power Supply
This product should be operated with proper supply voltage. If you are not sure about supply voltage,
contact authorized dealer.
The ETERNITY does not work in isolation from the environment. Power is fed to the system for functioning
of the system. Being a PBX system, it has several interfaces like trunk lines and extensions, external
music, Public Address System, Printer interface, PC interface, etc. So there are chances of heavy voltages
entering the system through these interfaces. Also, static charges could find their way through the system
components.
42
The ETERNITY is designed to work with input voltages ranging between 100-240VAC. The Power Card of
ETERNITY have a 'switch mode' design to support such a wide range of operating voltage.
Protect the system from heavy voltages on the trunk lines and the
overhead stations
The ETERNITY may be damaged by heavy voltages entering the system from trunk lines or from
overhead stations. These heavy voltages may enter the trunk lines and from overhead stations due to:
Heavy voltage line falling on the CO line or on the overhead stations cable. A dangerous surge can
occur if a telephone line comes in contact with a power line.
Lightning/Thunderbolts.
To protect ETERNITY from these voltages, use Primary Protection/Surge Protectors on the trunk and long
distance extension lines to protect the system from lightning and electrical surges.
Install any standard Input Protection (punch down protection) on the Krone Modules of the MDF or the
Primary Protection Module - PPM4 supplied by Matrix at entry points for all CO trunks lines and all
overhead stations. The product Warranty does not cover damages resulting from heavy voltage on CO
lines and overhead stations!
It is recommended that you install the PPM on the MDF, as MDF cables from the CO are terminated on the
System MDF.
To protect ETERNITY from extremely high voltage currents associated with lightning strikes, install a
lightning protector on an outside (CO) line.
Every person carries some static charge in his/her body depending upon body composition and the
environment around them. Most of the times, this charge finds its way to the earth when the person
touches any object which is grounded, or when the person is barefoot.
Generally, persons installing or handling electronic and electrical equipment take precaution to wear
appropriate footwear to get protection from electric shocks. Doing so, the static charge accumulates in his/
her body and does not find its way to the ground. But when such a person touches any of the electronic
cards, the static charge finds its way through the electronic components thereby causing damage to the
cards.
So, the person installing or servicing the system must provide a path to the static charges, by wearing an
antistatic belt, which is properly earthed.
The Communication Port (COM Port) is provided on the ETERNITY for connecting a PC.
If an electrical wire carrying heavy voltage accidentally shorts with this cable, heavy voltages can damage
the communication port.
It is recommended that the communication cable (connecting ETERNITY and the PC) be run through the
conduit carrying telephone cables or through a separate conduit.
43
Protecting the system from heavy voltage on the Analog Input Port
The Analog Input Port of the ETERNITY should be protected from:
Heavy voltages on the cable connecting the system and the external music source due to shorting with any
electrical wire.
An audio signal, not complying with the specifications of this port, is fed to this port.
Refer the technical specifications of the AIP before connecting any external music device to it.
Protecting the system from heavy voltage on the Analog Output Port
The Analog Output Port of the ETERNITY should be protected from:
Heavy voltages on the cable connecting the system and the amplifier/speaker due to shorting with any
electrical wire.
Protecting the system from heavy voltage on the Digital Output Port
The Digital Output Port (DOP) of the system should be protected from:
Heavy voltages on the cable connecting the system and the device connected to the DOP.
Protecting the system from heavy voltage on the Digital Input Port
The Digital Input Port (DIP) of the system should be protected from:
Heavy voltages on the cable connecting the system and the sensor device or panic switch connected to
the DIP.
44
The Earth (Ground) is the most important safety procedure to prevent electrical shocks and fires. It
protects from lightning strikes, electrical transients, static discharges, electromagnetic interference and
electrical hazards.
Ensure that a proper electrical earth and a telecom earth are in place for the safety of people and the
system. Telecom earth is a dedicated earth for the PBX/any other telecom equipment.
Provide a separate Telecom Earth (Ground) to the system installation. Providing a separate earth to the
telecom equipment eliminates the possibility of any back-voltage on the earth.
Refer How to Make the Telecom Earth for instructions on making the perfect earth (ground).
Always wear a properly earthed, electrostatic discharge preventive wrist strap/belt while handling the
system and its cards to prevent damage to the system and harm to yourself.
Slots and openings in the cabinet and the back or bottom are provided for ventilation, to protect the system
from overheating. These openings must not be blocked or covered.
Never insert or push objects of any kind into this product through the cabinet slots as they may touch
dangerous voltage points or short out parts which may result in fire or electric shock.
Do not allow anything to rest on the power cord. Do not locate this product where the cord will be trampled
upon or get entangled.
This product is equipped with a plug having a third (ground) pin, which fits only into a grounding-type
outlet. This is a safety feature. So, if the existing outlet is not a three-pin and or if you are unable to insert
the plug into the outlet, have the outlet replaced by the electrician.
Do not overload wall outlets and extension cords as this can result in the risk of fire or electric shock.
Do not disassemble this product. Opening or removing covers may expose you to dangerous voltages or
other risks. Incorrect reassembly may cause electric shock when the appliance is used. Take the product
to a qualified technician when service or repair work is required.
Avoid using a telephone (other than a cordless type) during a storm, to prevent electric shock from
lightning.
Do not use the telephone to report a gas leak in the vicinity of the leak so as to prevent the risk of fire.
External Devices
When you connect external devices like headset, external music source (PC, Cassette Player, CD Player),
relay devices (door lock, door lock release), sensors, public address or paging devices, telephone
instruments, cables, connectors, etc., ensure that they are of standard make and good quality, so that the
functioning of the system is not affected.
Matrix does not guarantee the performance of external devices that are not supplied by it.
45
Do not use liquid cleaners or aerosol cleaners. Never spill liquid of any kind on the product.
Unplug this product from the wall outlet and refer servicing to a qualified service person under the following
conditions:
Battery
ETERNITY contains a 3VDC/15mAh Manganese Lithium Coin Battery (ML 1220 - Rechargeable) of diameter
12.5mm and height 2.0mm. The Battery is located on the CPU Card. The Battery should be replaced only by
authorised dealers of Matrix. End Users must not attempt to replace it.
Caution: There is risk of explosion if the Battery is replaced in an incorrect manner. Please dispose-off
used Batteries.
Disposal
This product must be disposed according to the national laws and regulations prevailing in the country
where it is installed.
46
Make sure that the RF Antenna is installed at least 20 cm away from other electronic and radio
transmission devices.
Make sure that the RF Antenna is installed at a place at 20 cm away from people's vicinity.
People carrying medical implants like cardiac pacemakers are advised to maintain appropriate distance
from the system. They are also advised to avoid being in the vicinity of the product for a long time.
CHAPTER 4
13.
The Matrix ETERNITY is to be installed by persons trained and experienced in telecom wiring.
The person installing the ETERNITY must be familiar with trunks, physical wiring of the MDF on both
the exchange (PBX) side and the line side (CO).
When installing any equipment, make sure that you take all the necessary precautions for handling
electronic and electrical appliances. Follow proper procedures for static electricity, while handling the
system and its cards to prevent damage to the system and harm to yourself.
Use a grounding mat and wear an anti-static strap/belt. Read the dos and don'ts listed in Protecting
ETERNITY and Yourself.
If you have complied with the requirements and instructions described in Before You Start, you may
now begin the installation of your ETERNITY LE.
ETERNITY LE can be Wall Mounted or placed on a table. You can also affix wheels to the system; to move it like a
trolley.
ETERNITY LE has two racks. The first rack has a total of 16 slots. The first three slots from the left are fixed slots
and the remaining 13 are universal slots.
The second rack has a total of 16 slots. The first slot from the left is a fixed slot and the remaining 15 are universal
slots.
The Matrix ETERNITY LE is shipped factory fitted with the Power supply card, the CPU Card in their respective
fixed slots (refer the section Know Your ETERNITY).
The cards - BRI, T1E1PRI, GSM, VoIP, DKP, CO, SLT, VMS, E&M, Magneto - are shipped separately as per the
order placed by individual customers. These cards can be installed in any of the 28 Universal slots.
The cards - BRI, T1E1PRI, GSM, VoIP, DKP, CO, SLT, VMS, E&M, Magneto - are shipped separately as per the
order placed by individual customers. These cards can be installed in any of the 28 Universal slots.
Illustrated below is the position of the fixed and universal slots in ETERNITY LE.
47
ETERNITY LE
Rack 1
CPU Card
CPU Card
13 Universal Slots
Rack 2
Power Supply Card
15 Universal Slots
Follow the installation instructions for cards described here, also when you expand the system (add more cards) or
remove or swap cards for maintenance and repair.
1. Unpack the box. Check the package contents (see Packing List). Contact your Dealer/Distributor if any of
the items is missing, faulty or damaged. Do not discard the packaging material.
dimensions and weight of the model you have. If mounting the system on a wall, you may refer the
mechanical dimensions and the Mounting Template for drilling holes at appropriate places on the wall.
3. When installing the system in a rack, allow adequate space between the system and other units for air
circulation.
4. Mount the system at the selected site. Make sure that the system is placed in such a way that you have full
access to the front and back panels. The holes in the panels are provided for ventilation; Make sure that
these are not blocked, to prevent overheating.
48
6. Check the voltage at the power point from where the supply is to be given to the system. It should be as
per the specifications. Earth the system properly. (Refer How to Make the Telecom Earth)
Inserting Cards
7. Make sure that the ETERNITY power is off and the power cord is unplugged.
8. Open the enclosure slot covers by pressing down the snap lugs.
9. Select a free slot from the universal slots.
10. Unscrew and remove the filler bracket that covers the card-slot opening of the slot you intend to use.
11. Hold the card with the connectors facing you. Do not grab the card from both ends.
12. Slide the card into the slot, along the guide rails provided for each slot at the top and bottom planes.
13. Ensure that the cards are inserted deep enough for all the connector pins on the cards make complete
14. When the card is firmly seated in the connector, push down the levers on the card mounting bracket and
Detailed installing instructions are provided for each card - Power Supply Card, CPU, DKP, SLT, CO, ISDN
BRI, ISDN T1E1PRI, GSM, VoIP, E&M, etc. - later in this section. Refer to them when installing each card
type.
17. To remove a card:
49
If you are removing the card permanently or for a certain period of time, install a filler bracket over the
empty card opening in the chassis.
Installing filler brackets over empty card-slot openings is necessary to protect the system from dust,
dirt, insects and damage.
18. Connect the cables supplied with the cards and lead the cables through the cable guides provided below
the slots in the enclosure. This will ensure neat and tangle-free cabling.
19. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.
20. Close the enclosure cover, pressing down the snap lug as you push each part of the cover in its place.
50
If already installed, switch OFF power supply, unplug the power cord. Remove the screws securing the
card. Lift the levers on the mounting bracket to release the card. As the card emerges from the slot, ease it
out of the slot.
2. Insert the Power Supply card into the guide rails into the slot designated for the Power Supply Card. Make
sure that the card is inserted deep enough to make perfect contact with the connectors on the
motherboard at the backplane.
3. Now, press down the levers on the card mounting bracket to secure the card in its slot.
4. Secure the card in the slot by screwing the bracket on both ends.
5. If installing the PS48V card, connect the Float cum Boost Charger (FCBC). Terminate the power cord from
the FCBC output into the 3-way termination block on the PS48V card.
Polarity is critical. Ensure that the wires are connected with the correct polarity. Follow the standard color
codes used by FCBC manufacturers:
Color
Signal
Red
+48VDC
Black
GND
Green
Earth
It is recommended that you measure the voltage before connecting the power cable to the power supply
card. Ensure that the earth is connected.
51
FCBC
10A
41 to 56V
ETERNITY ME
Card PS48VDC
48V Battery
If two PS48V cards are installed for redundancy , each must be connected to a separate FCBC and each
FCBC must be connected to a separate source of power supply.
6. Connect Battery back up to the FCBC14.
Battery backup time depends upon the total load. The total load is the sum of system's load and load of
active extensions.
The Battery back up time depends on the 'Ah' rating of the battery connected to the FCBC. If 48V/26Ah
batteries are connected to the FCBC for the ETERNITY LE system then backup time of 2.5 to 3 hrs can be
ensured. The FCBC uses the constant voltage charging method. So, the batteries get charged faster if
less power is consumed by the system when in mains mode.
7. Switch on power supply, after completing all other installation.
14.
When the batteries are drained, the FCBC goes into the boost mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery voltage goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to boost mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the batteries are charged with higher current initially.
52
Communication Port
This is a serial full duplex RS232C communication port. The COM Port has a DB-9 connector. The COM port is
meant for connecting a PC to the ETERNITY LE. With a PC connected to the ETERNITY LE you can install and
operate from the COM Port the following features:
The CPU card is designed for the ETERNITY LE models as a combination of the Master Card and Switch Card of
ETERNITY ME. The CPU card has the functions of both cards; it manages the entire system, controls all other
slave cards (SLT, DKP, E&M, BRI, VMS, T1E1, GSM, VoIP, etc.). All configuration and programming information is
stored on this card.
53
The CPU card occupies a fixed slot, second from the left, with a unique arrangement of connectors. So no other
card can be inserted in the slot of the CPU card.
The CPU card has an Ethernet Port, Communication Port, an Analog Input Port (AIP), an Analog Output Port
(AOP), and a USB Port on the front panel.
Ports and Connectors on the CPU Card at a Glance
Port Name
Connector
Location
Function
RJ45
Facia
USB
Fascia
Communication (COM
Port)
DB-9 female
Fascia
Audio jack,
3.5mm
Fascia
Audio jack,
3.5mm
Fascia
DKP Port
RJ45
Fascia
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the CPU Card into the guide rails of the slot designated for the card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
54
Capture Station Message Detail Recording-Report, Station Message Detail Recording-Online and
Station Message Detail Recording-Posting.
Capture System Activity Log and System Fault Log, Hotel Motel Activity Log
When you connect the ETERNITY LE to a standalone PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY LE and the Ethernet Port of the PC do
not conflict.
The Master Ethernet Port of ETERNITY LE and the Ethernet Port of the PC are in the same Subnet.
4. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
access the web-based programming tool Jeeves from any PC on the LAN.
set up and run software applications such as PMS and CAS on any PC on the LAN.
generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
When you connect the ETERNITY LE to a LAN PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY LE and the Ethernet Port of the PC do
not conflict.
The Master Ethernet Port of ETERNITY LE and the Ethernet Port of the PC are in the same Subnet.
15.
16.
55
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
0.707
Vrms across 600
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
DC Bias
Isolation
Internal Transformer
600
Termination provided
600
on the CPU card, locate the Jumper for External Music Source, Jumper J5. Change its position from
BC (default) to AB.
Plug in the audio jack of the device into the AIP connector.
Also refer the topics Music on Hold (MOH), Background Music (BGM), External Music.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages resulting from improper use.
56
ETERNITY ME Card
CO8+SLT24
Combination card, with 8-ports to connect to 8 Two-wire Analog trunk lines and
24 Single Line Telephones
Choose an SLT Card with the configuration that meets your requirement for SLT ports. Also consider the maximum
SLT Port capacity of the system you are installing. The maximum number of SLT ports supported by the ETERNITY
LE are 999.
Connectors
The SLT Cards have RJ45 connectors, with each connector having 4 SLT ports. A multi-pair, MDF cable is supplied
for each connector.
Only the SLT48 card has a 50-pin Centronics connector for the ports.
LEDs
The SLT cards for ETERNITY LE have a single, tri-color LED to indicate:
the status of any one extension during normal functioning of the system.
You may monitor any of the SLT Extension ports by assigning the LED to that port17.
17.
To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED
Number is the number of the LED on the card, which will monitor the port.
57
1. Decide the number of SLT extensions required and arrange for as many telephone instruments.
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require
you to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of
the phone to dial Flash.
2. Unpack the SLT card and check the package contents. Ensure that the power supply is switched off,
before you begin the installation of the card. Always wear an electrostatic discharge prevention wrist strap/
belt and use a grounding mat.
The SLT Card supports Hot Swap. So, you can insert the SLT Card while the system is switched on.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
For each connector on the SLT Card, there is a separate 4-pair cable with an RJ45 jack on one end and
free at the other end.
Port Number
Pin Number
Signalling
Wire Colour
SLT
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Port 2
Port 3
Port 4
58
Port Type
Port Number
Pin Number
Signalling
Wire Colour
Port 5
Tip
White
30
Ring
Slate
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
10
Tip
Red
35
Ring
Slate
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
15
Tip
Black
40
Ring
Slate
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
21
Tip
Yellow
46
Ring
Slate
22
Tip
Violet
47
Ring
Blue
Port 6
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
Port 13
Port 14
Port 15
Port 16
Port 17
Port 18
Port 19
Port 20
SLT
Port 21
59
Port Type
Port Number
Pin Number
Signalling
Wire Colour
Port 22
23
Tip
Violet
48
Ring
Orange
24
Tip
Violet
49
Ring
Green
25
Tip
Violet
50
Ring
Brown
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
White
30
Ring
Slate
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
10
Tip
Red
35
Ring
Slate
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
Port 23
Port 24
Connector 2
Port 25
Port 26
Port 27
Port 28
Port 29
Port 30
Port 31
Port 32
Port 33
Port 34
Port 35
Port 36
Port 37
60
Port 38
Port 39
Port 40
Port 41
Port 42
Port 43
Port 44
Port 45
Port 46
Port 47
Port 48
14
Tip
Black
39
Ring
Brown
15
Tip
Black
40
Ring
Slate
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
21
Tip
Yellow
46
Ring
Slate
22
Tip
Violet
47
Ring
Blue
23
Tip
Violet
48
Ring
Orange
24
Tip
Violet
49
Ring
Green
25
Tip
Violet
50
Ring
Brown
61
L1
Connector
Color
RJ45-1
01
02
RJ45-3
RJ45-4
SLT
SLT
SLT
SLT
13
14
RJ45-5
SLT
SLT
SLT
SLT
17
18
RJ45-6
SLT
SLT
SLT
SLT
21
22
RJ45-7
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
25
26
RJ45-2
7
RJ45-8
62
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
Connection
04
05
06
08
09
10
12
16
20
24
28
29
30
32
L1
Connector
Color
Connection
RJ45-1
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
01
02
RJ45-3
SLT
SLT
SLT
SLT
09
10
RJ45-4
SLT
SLT
SLT
SLT
13
14
1
RJ45-2
04
05
06
08
12
16
63
L1
Connector
1
RJ45-2
64
Color
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
Connection
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
01
02
04
05
06
08
L1
Connector
Color
RJ45-1
Connection
RJ45-3
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
RJ45-4
SLT
SLT
SLT
SLT
13
14
RJ45-5
SLT
SLT
SLT
SLT
17
18
RJ45-6
SLT
SLT
SLT
SLT
21
22
RJ45-7
TWT
TWT
TWT
TWT
TWT
TWT
TWT
TWT
01
02
1
RJ45-2
7
RJ45-8
04
05
06
08
09
10
12
16
20
24
04
05
06
08
7. Plug in the RJ45 end of the MDF cables supplied with the card into the respective connectors.
8. Terminate the open end of the cables into the punch down blocks of the Krone modules designated for
step. If you have completed all installation tasks, power ON the system, observe the Reset Cycle and the
LED Pattern of the SLT Card.
LED Pattern of the SLT Card
Stage
LED Color
Cadence
RED
ON-200ms-OFF 200ms
GREEN
ON-200ms-OFF 200ms
Auto Upgradationa
65
Stage
LED Color
Cadence
RED
ON 500ms-OFF 500ms
GREEN
ORANGE
ORANGE
Flash Failure
None
None
RAM Failure
None
None
Initialization
Stand-by task
Errors
LED Color
LED Cadence
RED
Togglea
RED
a. The current LED state will remain the same until the next command is received from the application on the SLT Port. For example, if the current LED
state is Green/Red ON, on the next command received, the LED will be
turned OFF. It will remain OFF until the next command is received. When the
next command is received it will be turned Green/Red ON again. This process continues.
For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging in
the phone cables into the RJ45 connectors on the card.
66
When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the first port on the
connector will be assigned to the SLT.
ETERNITY LE DKP32
ETERNITY LE DKP16
ETERNITY LE DKP8
16-port card to connect 16 DKP/DSS Consoles and the proprietary Digital Turret,
EON74a.
8-port card to connect 8 DKP/DSS Consoles
a. EON74, the proprietary Digital Turret of EON74 is currently supported only on ETERNITY LE DKP16
Card and ETERNITY LE DKP8 Card with firmware version-revision V04R01 and onwards.
Select a DKP Card with the configuration that meets your requirement for DKP Ports. Also consider the maximum
DKP Port capacity of the system you are installing.
ETERNITY LE supports a maximum of 128 DKP Ports.
Connectors
The DKP Cards have RJ45 connectors, with each connector having 4 DKP ports. A multi-pair MDF cable is
supplied for each connector on the card.
LEDs
The DKP cards for ETERNITY LE have a single, tri-color LED to indicate:
the status of any one of the ports during normal functioning of the system. By default it is assigned to DKP
Port 1.
You may monitor any of the DKP ports by assigning the LED to that port18.
18.
You can do this from the SE mode, by dialing the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor the port.
67
1. Unpack the DKP card and check the package contents. Before handling the card, make sure that power
supply is switched off and you are wearing an antistatic-wrist strap/belt and have a grounding mat.
2. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket, keep for future use to cover empty slots.
3. Insert the DKP card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
4. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
Refer the connector pin details for each DKP Card type given in the following.
68
L1
Connector
Color
Connection
RJ45-1
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
01
02
RJ45-3
DKP
DKP
DKP
DKP
RJ45-4
13
14
RJ45-5
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
RJ45-6
21
22
RJ45-7
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
1
RJ45-2
7
RJ45-8
DKP
DKP
DKP
DKP
04
05
06
08
09
10
12
16
17
18
20
24
25
26
28
29
30
32
69
L1
Connector
Color
Connection
RJ45-1
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
01
02
RJ45-3
DKP
DKP
DKP
DKP
09
10
RJ45-4
DKP
DKP
DKP
DKP
13
14
1
RJ45-2
70
04
05
06
08
12
16
L1
Connector
Color
RJ45-1
1
RJ45-2
Connection
DKP
DKP
DKP
DKP
01
02
DKP
DKP
DKP
DKP
04
05
06
08
Plug in the RJ45 end of the MDF cables provided with the DKP card into the respective connectors.
Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the ETERNITY LE DKP Port must be terminated to the bottom of the Krone
Connector, while the wire-pair of the extension line to be connected to this port must be terminated on
the top of the Krone connector. Refer the topic The Main Distribution Frame (MDF) for illustration.
6. Connect the Digital Key Phones to the wall jacks at their respective locations. Detailed installations
instructions for EON, EONSOFT are provided separately. Installation instructions for EON74 are provided
in the EON74 User Guide.
If you have completed all installation tasks, power on the system and observe the Reset Cycle and the LED Pattern
of the DKP Card.
71
LED Color
Cadence
RED
ON-200ms-OFF 200ms
GREEN
ON-200ms-OFF 200ms
RED
ON 500ms-OFF 500ms
GREEN
ON 500ms-OFF 500ms
ORANGE
ON 500ms-OFF 500ms
ORANGE,
GREEN
Flash Failure
None
None
RAM Failure
None
None
Auto Upgradation
Initialization
Stand-by task
Errors
LED Color
LED Cadence
RED
RED
a. The current LED state will remain the same until the next event is received from the application on the DKP Port.
For example, if the current LED state is Green/Red ON, on the next event, the LED will be turned OFF. It will
remain OFF until the next event occurs. When the next event is received it will be turned Green/Red ON again.
This process continues.
b. Same as the above note.
72
Installing EON48
1. Unpack the box and verify the package contents19.
2. Mount the phone on a desk or on the wall at a convenient location.
3. To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
Foot Stand
DND
Redial Release
Hold
abc
3 def
4 ghi
jkl
6 mno
tuv
9 wxyz
7 pqrs 8
CA 3
Keyhole
Slot 2
Line
4P4C Spring
Cord
Press
down
to detach
Foot Stand
Press down
to detach
Foot Stand
Names
CA 4
Keyhole
Slot 1
Headset
Port
CA 2
CA 1
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
5. Connect the handset of the EON48 to the phone body using the spring cord.
19.
73
6. To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
Headset
You may also plug in a stereo headset with an RJ12 connector into the headset port at the bottom of the
phone, marked with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
7. Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
74
8. When the ETERNITY is powered ON, the EON will get reset. The EON communicates with the ETERNITY.
The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the
message 'Please wait' appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PL EASE WAI T .. .
9. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
202 Reception
M on 2 4 A U G 1 2 : 0 0
10. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-
Operation.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
Installing EON31020
1. Unpack the box and verify the package contents21.
2. Mount the phone on a desk or on the wall at a convenient location.
20.
21.
75
3. To mount EON310 on a wall, detach the Foot Stand on the bottom of the phone. Refer to the illustrations in
EON48.
CA 1
CA 2
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON310. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON310 on a desk, you can attach the Foot Stand in two ways - 30 and 50 degree
You may also plug in a stereo headset with an RJ12 connector into the headset port marked with the
symbol
, on the left side panel of the phone as illustrated in the figure below.
Headset
Casio Jack
Headset
(R J12 Connector)
7. Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
76
8. When the ETERNITY is powered ON, the EON will get reset and the message 'Welcome to Matrix.
9. The EON communicates with the ETERNITY. The handshaking lasts for 5-6 seconds. The EON model,
version and revision number, along with the message Please Waitappears on the LCD display.
10. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
11. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-
Operation.
22.
77
You can install two DSS consoles to a DKP. Refer Direct Station Selection Console for possible
combinations for installing the models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated on the bottom of
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
Windows 98
Windows XP
Windows NT
Windows 2003
Windows Vista
Windows 2007
23.
78
2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
have of the operating systems mentioned above, install any compatible Windows Operating System.
6. Now insert the EONSOFT CD-ROM supplied with this PC-based DKP into the CD drive of your Computer.
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
9. Click the Matrix EONSOFT Icon to run the program.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
79
11. Click the shortcut to open the program. The EONSOFT window will open:
12. Click Options at the top left of the window. A drop down menu will appear.
80
14. Select the COM Port to which the communication cable is connected.
81
15. EONSOFT is now connected. If you have already configured the DKP parameters like Access Code and
Name for the port to which EONSOFT is connected, these will appear.
If this window does not appear after you have selected the COM Port Option, test the COM Port for
data transfer.
If the wrong COM port has been selected, a window will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
82
From the drop down menu of Options, select the COM Port to which you have connected the
communication cable.
Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
Click the button labeled Start Test in the COM Port Settings dialog box.
After clicking this button, observe the Test Result section on the dialog box.
The Error Count would show zero as value, if both the communication cable and the COM port were
working.
The above figure shows that the COM Port/communication cable is working.
If the Error Count shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
Remove the communication cable from the COM Port of the PC.
Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
Now, if the error count is zero, please check the Communication Cable.
If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
83
Connectors
The BRI cards have RJ45 connectors. The ETERNITY LE BRI8 card has 8 RJ45 connectors for 8 BRI ports.
The ETERNITY LE BRI4 card has 4 RJ45 connectors for 4 BRI ports. A separate cable is supplied for each
connector.
LEDs
The ETERNITY LE BRI8 has 4 LEDs and BRI4 has 4 LEDs.
ISDN
Network
NT 1
BRI Port
ETERNITY
Power
U-Interface
(2-wire)
S/T
Interface
Customer Premises
Where,
U Interface = between the NT1 equipment and the ISDN central office.
S/T Interface = between the ISDN user equipment, that is, ETERNITY and the Network Interface
Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the
ETERNITY.
84
TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY via
the NT1 device. Here, the ETERNITY is the Terminal Equipment, so the BRI Port must be programmed to work in
the TE mode.
When an ISDN Phone is to be connected to the BRI port of ETERNITY, the BRI port must be programmed to work
in NT mode.
When a BRI port of another ISDN PBX is to be connected to the BRI port of the ETERNITY, in such a configuration,
you may configure
the BRI port of the other ISDN PBX in the TE mode and the BRI Port of the ETERNITY in the NT mode.
OR
the BRI port of the other ISDN PBX in the NT mode and the BRI Port of the ETERNITY in the TE mode
Point-to-Point Configuration
ISDN
Network
NT
BRI Line
BRI Port
(TE Mode)
(UP to 1 Km.)
ETERNITY
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY, can be upto 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
Further, two configurations are possible in a Point-to-Multipoint configuration:
1. Short Passive Bus Configuration
2. Extended Passive Bus Configuration
85
NT
BRI Port
(TE Mode)
Terminal
Resistance 100
ETERNITY
ISDN Phone
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
Terminal 8
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
In a Short Passive Bus Configuration,
A maximum of 8 TEs or ISDN devices can be connected to a single NT on a bus up to 200 meters from the
NT.
100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access a BRI line and can make outgoing
calls. At the same time, another subscriber from ETERNITY or any ISDN phone shown in the figure can
make outgoing call from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.
d1 < 30 meters
NT
86
Terminal
Resistance 100
BRI Port
(TE Mode)
ETERNITY
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
Where,
TE = Terminal equipment of any ISDN Equipment
NT = Network Termination provided by Service Provider
TR Terminal Resistance 100
d = distance from NT to the last TE Equipment
d1 = the total distance from first TE equipment and the last TE equipment.
In an Extended Passive Bus Configuration,
You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access the BRI line and make outgoing calls.
At the same time, another subscriber from the ETERNITY or any ISDN phone shown in the figure can
make outgoing calls from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
power supply, always wear an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
2. Unpack the BRI card and check the package contents.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
the Orientation Type, change the position of the jumpers located on the Main Board of the card. Refer the
following tables for Jumper Positions for each BRI port.
Jumper Position
Mode
BRI Port 1
BRI Port 2
J20
J21
J22
J23
J24
J25
J26
J27
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
87
Jumper Position
Mode
BRI Port 3
BRI Port 4
J28
J29
J30
J31
J32
J33
J34
J35
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
Jumper Position
Mode
BRI Port 5
BRI Port 6
J36
J37
J38
J39
J40
J41
J42
J43
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
Jumper Position
Mode
BRI Port 7
BRI Port 8
J44
J45
J46
J47
J48
J49
J50
J51
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
By default, Orientation Type is TE. So, you may skip to the next step.
88
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
Last TE equipment.
Last point of the bus bar where the last TE equipment is connected.
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Function
Jumper Position
J3
J4
AB
AB
BC
BC
By default, Termination Resistance of 100 is set on the BRI port (Jumpers J3 and J4 are in AB position).
89
Tx 3
Rx 4
Rx 5
Tx 6
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
100
100
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
90
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Function
Jumper Position
J1
J2
AB
AB
BC
BC
Open
Open
To feed the power on Tx and Rx wires, set the jumpers J1 and J2 of BRI module in AB position.
To feed the power on separate pairs of wires, set the jumpers J1 and J2 of BRI module in BC position.
To power the ISDN terminal from external power source, keep the jumpers J1 and J2 open.
The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
9. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
bracket in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots. Remember to set the Orientation Type, Termination
Resistance and Power Feed, as required.
91
10. Use the straight cables supplied for each connector on the BRI card to connect the BRI Ports to the NT1
device supplied by your ISDN service provider. Refer the configuration and pinout details given below for
guidance.
Configuration details of the U interface (RJ-45) at NT1
Pin Number
Pin Details
Tx
Rx
Pin Details
Rx1
Tx1
Tx2
Rx2
Signal
Color
--
Orange-White
--
Orange
TX_A
Green-White
RX_A
Blue
RX_B
Blue-White
TX_B
Green
VOUT-
Brown-White
VOUT+
Brown
92
Signal
Color
--
Orange-White
--
Orange
RX_A
Green-White
TX_A
Blue
TX_B
Blue-White
RX_B
Green
VOUT-
Brown-White
VOUT+
Brown
The following diagram shows how to connect a BRI Line to the ETERNITY LE BRI port in the TE mode.
11. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
The BRI8 Card has 4 LEDs: L124, L2, L3, L4. These display the status of the first four ports, that is port 1
to 4. To view the status of the ports 5 to 8, from the SE mode dial the SE Command 5323-Software Slot17.
The BRI4 Card has 4 LEDs: L1, L2, L3 and L4. These display the status of each port.
The LEDs show the Status of the Ports as summarized in the table below:
Port Status
LED Color
LED Cadence
RED
Continuously ON
Port is active
GREEN
Continuously ON
24.
This LED keeps blinking. It displays the system heart bits, at the rate of one second. It will remain OFF for one second and will
show the status of port 1 for the next one second.
93
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
94
Last TE equipment.
Last point of the bus bar where the last TE equipment is connected.
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Module 3 (M3)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J6
J8
J7
J9
J6
J8
J7
J9
To insert 100
termination
AB
AB
AB
AB
AB
AB
AB
AB
To remove 100
termination
BC
BC
BC
BC
BC
BC
BC
BC
Module 4 (M4)
Function
Module 5 (M5)
BRI Port 5
BRI Port 6
BRI Port 7
BRI Port 8
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J6
J8
J7
J9
J6
J8
J7
J9
To insert 100
termination
AB
AB
AB
AB
AB
AB
AB
AB
To remove 100
termination
BC
BC
BC
BC
BC
BC
BC
BC
By default, Termination Resistance of 100 is set on the BRI port (the Jumpers are in AB position).
Tx 3
Rx 4
Rx 5
Tx 6
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
100
100
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Matrix ETERNITY System Manual
95
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Enable Feed Power on the BRI Port. For instructions see Power Feed under Configuring BRI Trunks.
By default, the Jumpers are set in AB position to feed power through Tx and Rx wires (Phantom
Power).
If you want to feed power through a separate pair of wires, you may change the position of the Jumpers
on the BRI module as mentioned in the table below.
Module 2 (M2)
Function
96
Module 3 (M3)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J4
J5
J2
J3
J4
J5
J2
J3
AB
AB
AB
AB
AB
AB
AB
AB
Module 2 (M2)
Function
Module 3 (M3)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J4
J5
J2
J3
J4
J5
J2
J3
BC
BC
BC
BC
BC
BC
BC
BC
Module 4 (M4)
Function
Module 5 (M5)
BRI Port 5
BRI Port 6
BRI Port 7
BRI Port 8
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J4
J5
J2
J3
J4
J5
J2
J3
AB
AB
AB
AB
AB
AB
AB
AB
BC
BC
BC
BC
BC
BC
BC
BC
The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
10. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
bracket in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots. Remember to set the Orientation Type, Termination
Resistance and Power Feed, as required.
11. Use the straight cables supplied for each connector on the BRI card to connect the BRI Ports to the NT1
device supplied by your ISDN service provider. Refer the configuration and pinout details given below for
guidance.
Configuration details of the U interface (RJ-45) at NT1
Pin Number
Pin Details
Tx
Rx
97
Pin Details
Rx1
Tx1
Tx2
Rx2
Signal
Color
--
Orange-White
--
Orange
TX_A
Green-White
RX_A
Blue
RX_B
Blue-White
TX_B
Green
VOUT-
Brown-White
VOUT+
Brown
98
Signal
Color
--
Orange-White
--
Orange
RX_A
Green-White
TX_A
Blue
TX_B
Blue-White
RX_B
Green
VOUT-
Brown-White
VOUT+
Brown
The following diagram shows how to connect a BRI Line to the ETERNITY LE BRI port in the TE mode.
12. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle, and the LED pattern of the BRI Card.
The BRI8 Card has 4 LEDs: L125, L2, L3, L4. These display the status of the first four ports, that is port 1
to 4. To view the status of the ports 5 to 8, from the SE mode dial the SE Command 5323-Software Slot17.
The BRI4 Card has 4 LEDs: L1, L2, L3 and L4. These display the status of each port.
The LEDs show the Status of the Ports as summarized in the table below:
Port Status
LED Color
LED Cadence
RED
Continuously ON
Port is active
GREEN
Continuously ON
25.
This LED keeps blinking. It displays the system heart bits, at the rate of one second. It will remain OFF for one second and will
show the status of port 1 for the next one second.
99
The CO Card
The CO Card provides the interface to connect the ETERNITY with the Two-Wire Analog Trunk lines from the CO
Network. The CO Card supports the different standards and features of CO Networks across the world.
The CO Card is available in the following configurations for ETERNITY LE. CO interface is also available in
combination with SLT ports on a single card.
Choose a CO Card with the configuration that meets your requirement for CO trunk ports, keeping in mind the
maximum CO Trunk Port capacity of the system you are installing.
ETERNITY LE supports a maximum of 128 CO Ports.
Connectors
The CO Card has RJ45 connectors, with 4 CO ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
LED
The CO Cards have a single tri-color LED to indicate:
the status of a selected Trunk port during normal functioning of the system.
You can assign the LED to any CO port on the card which you want to monitor26.
that power supply is turned off before you begin the installation of the card. Put on an electrostaticdischarge preventive wrist strap/belt and use a grounding mat.
2. Unpack the CO card and check the package contents.
26.
100
To assign the LED to a selected port for monitoring its functioning, you must enter SE mode and dial the SE Command 7902-SlotLED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor
the port.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
should make perfect contact with those of the slot on the backplane motherboard.
5. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
Distribution Frame.
You may refer the illustrations below for pinout details of the connectors on the card.
101
L1
Connector
Color
Connection
RJ45-1
CO
CO
CO
CO
CO
CO
CO
CO
01
02
RJ45-3
RJ45-4
CO
CO
CO
CO
CO
CO
CO
CO
1
RJ45-2
102
04
05
06
08
09
10
12
13
14
16
L1
Connector
RJ45-1
1
RJ45-2
Color
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
RJ45-3
Unused
RJ45-4
Unused
Connection
CO
CO
CO
CO
CO
CO
CO
CO
103
L1
Connector
Color
RJ45-1
13
14
RJ45-3
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
RJ45-4
SLT
SLT
SLT
SLT
RJ45-5
SLT
SLT
SLT
SLT
RJ45-6
SLT
SLT
SLT
SLT
21
22
RJ45-7
CO
CO
CO
CO
CO
CO
CO
CO
01
02
1
RJ45-2
7
RJ45-8
Connection
04
05
06
08
09
10
12
16
17
18
20
24
04
05
06
08
7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors.
8. Terminate the free end of the CO Card cable into the punch down blocks of the Krone modules designated
104
LED Color
Cadence
RED
ON-200ms-OFF 200ms
GREEN
ON-200ms-OFF 200ms
RED
ON 500ms-OFF 500ms
GREEN
ORANGE
ORANGE
Flash Failure
None
None
RAM Failure
None
None
Auto Upgradationa
Initialization
Stand-by task
Errors
LED Color
LED Cadence
RED
RED
a. The current LED state will remain the same until the next event is received from the application on
the CO Port. For example, if the current LED state is Green/Red ON, on the next event, the LED
will be turned OFF. It will remain OFF until the next event occurs. When the next event is received
it will be turned Green/Red ON again. This process continues.
b. Same as above note.
105
PRI
Robbed Bit Signaling
Q-Signaling (QSIG)
E&M
When connected to E1 carrier lines, the card supports the following signaling types:
PRI
Channel Associated Signaling (CAS)
Q-Signaling (QSIG)
E&M
The T1E1PRI Card is available in the following configurations for ETERNITY LE:
2-Port card with QSIG support to connect 2 ISDN T1/E1 PRI Lines or ISDN
Compatible Devices
1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Compatible Device
Connectors
The T1E1PRI card has an RJ45 Connector for each port. The ETERNITY LE T1E1PRI Dual card has 2 RJ45
Connectors for the two ports, while the ETERNITY LE T1E1PRI Single card has a single RJ45 Connector.
A cable with RJ45 plugs on both ends is supplied for each connector.
LEDs
The ETERNITY LE T1E1PRI Dual Card has four LEDs: L1, L2, L3 and L4.
The ETERNITY LE T1E1PRI Single Card has two LEDs L1 and L2.
an electrostatic-discharge preventive wrist strap and use a grounding mat. Make sure the power supply is
turned off.
106
Pin1
Pin2
Pin3
Pin4
Termination Resistance ()
OFF
OFF
OFF
ON
0 (Default)
5. By default, termination resistance of PRI port is set as 120, which is for E1 connectivity.
To use the PRI Port for T1 connectivity, termination resistance must be changed to 100.
Use DIP Switch SW5 to change the Termination Resistance of PRI Port 1. Set the Pins of SW5 as
shown below:
Pin-1
Pin-2
Pin-3
Pin-4
Resistance
OFF
OFF
ON
OFF
OFF
ON
OFF
OFF
If using the ETERNITY LE T1E1PRI Dual Card, use DIP Switch SW2 to change the Termination
Resistance of PRI Port 2. Set the Pins of SW2 as shown below:
Pin-1
Pin-2
Pin-3
Pin-4
Resistance
OFF
OFF
ON
OFF
OFF
ON
OFF
OFF
6. Insert the T1E1PRI Card into the guide rails of the free slot you selected for the card. Make sure that the
connectors on the card make perfect contact with those of the slot on the backplane motherboard.
7. Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
interface equipment (modem), which is generally supplied by your ISDN Service Provider along with the
PRI line.
107
ISDN
Network
ETERNITY
G.703
Modem
4-wire
HDSL
(RJ-45 Connector)
DTE
(RJ-45 Connector)
4-wire
PRI Port
G.703
Modem
Power
Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI line, that is,
one at the Local Exchange and other at the Customer's Premises.
At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
The DTE interface of the modem is be connected to the PRI port (RJ-45 connector on the Matrix
ETERNITY LE T1E1PRI Dual/Single Card).
9. Refer the following pin details for connecting the Network Termination Unit with the ETERNITY.
Pin details of HDSL Interface of the G.703 Modem. (HDSL Network Termination Unit)
Pin Number
Pin Details
Line A
Line A
Not used
Line B
Line B
Not used
Not used
Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
108
Pin Number
Pin Details
TX1 (Tip)
TX2 (Ring)
Not used
RX1 (Ring)
RX2 (Tip)
Not used
Not used
Not used
Most of the HDSL Network Termination Unit manufacturers use these connectors. But you are advised to
read the instrument guide of the HDSL Network Termination Unit being used by you.
Tx1 (Ring)
Tx2 (Tip)
NC
NC
Rx2 (Tip)
NC
Rx1 (Ring)
NC
5
6
7
8
The cable wires may have to be crossed depending on the pinout of the DTE Interface of the modem.
10. If you have completed all other installation tasks. Power the system. After the Reset Cycle is completed,
The ETERNITY LE T1E1PRI Single Card has two LEDs: L1 and L2.
Given below are the LED Patterns defined for indicating port states in the signaling types supported by the
ETERNITY LE.
Color
Cadence
GREEN
Continuous ON
CRC4 Alarm
GREEN
109
Port Status
Color
Cadence
BFA Alarm
RED
LOS Alarm
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
RAI Alarm
RED
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
CRC4 Alarm
GREEN
MFA Alarm
RED
BFA Alarm
RED
LOS Alarm
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
Y-Bit Alarm
GREEN
AIS16 Alarm
RED
RAI Alarm
RED
RED
Continuous ON
LED2/LED4 Pattern:
LED2/LED4 Pattern:
110
Port Status
Color
Cadence
No Alarm
GREEN
Continuous ON
RED
AIS Alarm
RED
LOS Alarm
RED
Continuous ON
LED2/LED4 Pattern:
Port Status
Color
Cadence
GREEN
Continuous ON
RED
Continuous ON
Color
Cadence
Maintenance Mode
RED -GREEN
LED2/LED4 Pattern:
Port Status
Color
Cadence
RED
RED
Continuous ON
RED
GREEN
GREEN
Continuous ON
GREEN
Color
Cadence
Port Disable
RED
Continuous ON
LED2/LED4 Pattern:
Port Status
Color
Cadence
Port Disabled
OFF
OFF
111
Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer PLCC-An Introduction to know more.
Closed User Group (CUG), where several PBXs are connected with each other through E&M tie lines27.
PBX expansion, where two PBXs are connected with each other with E&M tie lines.
E&M Trunk Seizure Type28: Immediate, Immediate + Wink, Seizure Pulse, Seizure Pulse + Wink, Express,
and Compander Control Signal.
Address Signaling: Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The ETERNITY E&M Card is available in the following configuration for the ETERNITY LE:
Connectors
The E&M8 card is supplied with RJ45 or Amphenol connectors. On the ETERNITY LE E&M8 card with Amphenol
connectors, the first 4 E&M ports (E&M1 to E&M4) are located on the lower connector, and the remaining four E&M
ports (E&M5 to E&M8) are located on the upper connector.
The ETERNITY LE E&M4 card has a single Amphenol connector with 4 ports.
A separate MDF cable is supplied for each connector.
27.
28.
112
The PBXs in a Closed User Group (CUG) can be connected over ISDN T1E1PRI Lines as well. Refer the topic Closed User
Groups to know more.
This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
LEDs
The ETERNITY LE Card E&M8 has eight tri-color LEDs. The ETERNITY LE Card E&M4 has 4 LEDs, to indicate
the functioning of the ports.
a Trunk - works like a trunk interface when any of the extensions of the PBX makes an outgoing call
through it.
OR
a Tie Line - takes on a dual personality: functioning as both an extension and a trunk. The E&M port works
like an extension interface for incoming calls. It works like a trunk interface when any extension makes an
outgoing call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
PLCC-An Introduction to know more.
1. Have the necessary wiring for the E&M Analog trunk in place. Take the necessary safety precautions
before you begin handling the card; switch off power supply and always wear an antistatic wrist strap and
use a grounding mat.
2. Unpack the E&M card and check the package contents.
3. The E&M Card supports E&M Interface Type IV and Type V connection. To select the appropriate
Interface Type out of the two, you need to change the Jumper Settings.
Refer the table below to select the desired Interface Type and Speech Interface.
Jumper Number
Position
Function
J1 and J2
AB
BC
113
To select the Type-V connection for the E&M Port, set Jumpers J1 and J2 (given on E&M module) in
BC Position.
4. Select the speech interface - 2-wire speech or 4-wire speech - as required, by changing the jumper
Position
Function
J3 and J4
AB
BC
By default all the E&M Ports are set to support 2-wire Speech Interface.
To select 2-wire speech interface for the E&M Port, set Jumpers J3 and J4 (given on E&M module) to
BC Position.
To select 4-wire speech interface for the E&M Port, set Jumpers J3 and J4 on E&M module to AB
Position.
contact with those on the backplane motherboard. Secure the card by pressing down the levers and fix the
bracket with the screws provided with the card.
7. Connect the cables supplied with the E&M card into the RJ45/Amphenol connectors on the E&M Card.
8. Connect the other end of the cable into the E&M Ports of the other PBX/Router/Tie Line equipment by
114
L1 L5
L2 L6
L3 L7
L4 L8
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
RJ45-1
02
RJ45-2
RJ45-3
04
RJ45-4
RJ45-5
06
RJ45-6
08
RJ45-8
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Colour
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Connection
01
02
03
04
05
06
07
08
09
10
Pin No.
RJ45-7
115
116
L1 L3
L2 L4
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
RJ45-1
02
RJ45-2
04
RJ45-4
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
RJ45-3
117
118
119
If you are connecting two PLCC EPAX in a Power Line Carrier Communication Network Compander
Control Signal (CCS) Connection should be made as illustrated in the block diagram below for any of the
four combinations of E&M and Speech Interfaces illustrated in the previous step.
Compander Control Signal (CCS) is a special type of signal used by Power Line Carrier Communication
Networks to improve quality of speech transmission. The PLCC network expects this signal from the PBX
when speech is established. The E&M Card supports this facility. The ETERNITY sends CCS signal to the
PLCC panel.
When the E&M port is used as an Endpoint; the system sends a CCS to the PLCC panel while making
an outgoing call through the E&M port or when a call is received at the E&M port.
When the E&M port is used for Transit Exchange; the system sends a CCS to the PLCC panel while
there is a Transit call through the E&M port.
9. If you have finished all installation tasks, power ON the system, observe the Reset Cycle and the LED
LED Color
LED Cadence
RED
GREEN
GREEN, ORANGE
Initialization
At Power ON
Stand-Bya
Normal (Port Event)
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Stage
LED Color
LED Cadence
M-Wire High
GREEN
M-Wire Low
E-Wire High
M-Wire Low
E-Wire and M-Wire High
ORANGE
ORANGE
ORANGE
Eprom failure
ORANGE
ORANGE
Errors
121
Connectors
The Magneto Card may have an amphenol connector or RJ45 connectors, with 4 ports on each connector. A multipair cable is provided for each connector.
LED
The ETERNITY Magneto8 has 8 LEDs for each magneto port supported by the card.
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports during the normal
functioning of the system.
You may install an MDF to connect the Magneto Ports with the Field Telephone wires.
OR
You may connect the wires from the Magneto Field Telephones directly to the Magneto Port.
You are advised to use a separate set of Krone Modules for connecting the Magneto phones to the
Magneto ports of ETERNITY.
2. Prepare for the card installation by switching off power supply and wearing an electrostatic discharge
29.
122
A magneto telephone is a local battery telephone set, in which signaling current is provided by a magneto hand generator, usually
a magneto. The hand generator, commonly referred to as 'crank', is located on the right hand side of the telephone set and is
turned to produce energy to ring other phones or to signal the CO. The magneto, also called the generator, is used to convert the
mechanical motion via the crank to produce sufficient energy to ring other phones or to signal the CO.
contact with the connectors on the backplane motherboard. Press down the levers of the mounting bracket
to secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
6. Now, plug in the cables supplied with the Magneto Card into the connectors on the card. Terminate the
123
7. Connect the pairs of wires from the Magneto Field Phones to the appropriate pairs emerging from the
Magneto Card of the ETERNITY on the MDF. Refer the cable diagram for the Magneto card with amphenol
connector below.
124
8. If the your Magneto Card has RJ45 connectors, refer to the diagram below.
power cord, switch ON power supply from the mains, switch on the Power supply of the ETERNITY and
observe the Reset Cycle and the LED Pattern of the Magneto Card.
125
LED Color
LED Cadence
RED
GREEN
RED
GREEN
ORANGE
ORANGE,
GREEN,
RED
Initialization
Stand-Bya
Normal (Port Event)
Ring (incoming/outgoing call)
Port Disabled
OFF-Hook (in Speech)
Errors
Invalid Card Configuration Jumper
ORANGE
ORANGE
126
8-port card to connect to 8 GSM networks with 3G support (8 SIM Cards can be
installed)
4-port card to connect to 4 GSM networks with 3G support (4 SIM Cards can be
installed)
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
ETERNITY LE supports up to 128 Mobile ports.
SIM cards from different service providers can be used.
Antenna
There is a single rooftop (RT) antenna for four GSM ports. A splitter connects all the four ports on the card into a
single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to another position
(in case of weak signal).
LEDs
There is a tri-color LED for each mobile port on the card to indicate the functioning of the card and the status of the
ports.
127
If using a GSM/3G card, get the get the SIM Card from the GSM/3G service provider of your choice
ready. Use SIM PIN protection, if required.
power supply should be turned off, and you must be wearing an electrostatic discharge preventive wrist
strap and a have a grounding mat, before you begin handling the card.
3. Unpack the Mobile Card and verify the package contents.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
5. Insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the Mobile
card. You can insert multiple SIM cards of the same GSM service provider or of different service providers.
128
6. Insert the Mobile card into the guide rails of the Universal Slot you have selected for this card. Make sure
that the card is inserted deep enough to make perfect contact with the connectors in the backplane. Now,
press down the levers on the card mount bracket to secure the card in its slot.
7. Connect the antenna provided with the card on the splitter connector on the front panel of the card. You
may also use the antenna cable to place the antenna at another position.
129
register with the Mobile networks automatically. Now, observe the LED Patterns of the Mobile Ports.
At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
Each time the Mobile Port sends a request, such as a Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the Mobile network. It is fixed for 150 seconds for all Mobile ports.
Color
Port disabled
LED OFF
Port idle
LED OFF
Red
Continuous ON
Ring Event
Green
Speech
Green
Continuous ON
GSM initialization
Orange
PUK required
Orange
Orange
SIM Absent
Orange
Orange
130
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The VoIP card for ETERNITY LE is available in the following configuration.
ETERNITY LE VoIP32
ETERNITY LE VoIP16
Voice Channels
There are 32 Voice Channels on the VoIP32 Card and 16 Voice Channels on the VoIP16 Card, allowing as many
simultaneous calls to be made (using SIP Trunks and/or Extensions) as the number of Voice Channels supported
by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
131
SIP Trunks
The ETERNITY LE VoIP Card supports up to 32 SIP Trunks. You can subscribe to as many as 32 different Internet
Telephony Service Providers (ITSP).
It is possible to program all 32 SIP trunks on a single VoIP Card or in a distributed manner, where more than one
VoIP card is installed in the system.
SIP Extensions
ETERNITY LE supports 999 SIP Extensions. Upto 250 SIP Extensions can be registered with a single VoIP Card.
To register more than 250 SIP Extensions, you need at least two VoIP Cards.
You can register any SIP-enabled device like an IP-phone, a Softphone, analog phone adapter, with the VoIP Card
as the 'SIP Extension' of the ETERNITY LE.
The SIP Extensions function in the same ways as other extensions of the ETERNITY. SIP Extension users can
make and receive calls from and to other extensions of ETERNITY and external numbers over PSTN, GSM, VoIP
and E&M lines30. You can also connect the Standard and Extended IP Phones offered by Matrix as SIP
Extensions.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic License Management.
A SIP Extension can be registered with the VoIP Card of ETERNITY from three different locations. This helps
organizations overcome geographical distances and reduce call costs.
To know more about connecting SIP Extensions, see SIP Extensions.
30.
132
Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your country prohibit call traffic between the public telephony networks and IP networks, you must configure Logical Partition in your system.
To know more, see Logical Partition.
LAN Switch/Hub
LAN
Master Card
Router
Switch Card
WAN
IP
IP
A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
SIP ID/User ID
Authentication User ID
Authentication Password
SIP Registrar Server Address
SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
Network Information:
31.
Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
133
Master Card
Switch Card
WAN
LAN
LAN Switch/Hub
IP
IP
Router
The card is located behind the NAT Router and Private IP is assigned to the WAN port.
When connecting the card in a Private Network, you would require the following information:
134
IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
Gateway Address
DNS Address
DNS Domain Name (if applicable)
VoIP Card connected to the Public Network for Matrix Extended IP Phones
Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
To install the VoIP Card, do the following:
1. Get the items/information listed ready before you install the VoIP card and connect it to the IP network.
2. Observe all prescribed safety precautions when inserting or removing cards. Make sure the Power Supply
is switched off, and you are wearing an antistatic wrist strap/belt and have a grounding mat.
3. Unpack the VoIP card and verify the package contents.
4. Select any of the free Universal Slots of ETERNITY to insert the VoIP Card. Unscrew and remove the filler
bracket of the slot. Preserve the filler bracket for future use.
5. Insert the card into the guide rails of the slot. The card should be inserted deep enough to make perfect
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end into the Router/Modem.
135
Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end of the cable into the Router/Modem.
Connect the LAN Port of the VoIP Card to the LAN Switch/Hub.
8. To insert and connect another VoIP card, repeat the same steps as described above.
9. If you have completed all other installation tasks, you may switch on power supply and observe the Reset
LED Indication
There are two LEDs on the VoIP Card: LED 1 and LED 2.
LED Color
Cadence
Green
Continuous ON
Red
Continuous ON
Red
ON 1 sec-OFF 1 sec
ON 1 sec-OFF 1 sec
Red
Green
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 2500msec
136
LED Color
Cadence
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 500msec
ON 500msec-OFF 1500 msec
Green
LED Color
Cadence
Red
Continuous ON
Red
ON 500msec-OFF 3500msec
Red
ON 500msec-OFF 500msec
ON 500msec-OFF 2500 msec
Registration in Progress
Green
Registration Successful
Green
Continuous ON
SIP Trunk Status will be indicated by LED2 only after you have programmed LED Indication in VoIP Port
Parameters.
137
SIP Extensions
ETERNITY LE supports up to 999 SIP Extensions. The SIP Extensions function like DKP/SLT extensions of the
ETERNITY LE. SIP Extension users can make and receive calls to any extension user of the ETERNITY and to
external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP32.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY LE.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY LE, and you must have the IP8
License. For more information on Licensing, see License Management.
You can register upto 250 SIP Extensions with a single VoIP Card of ETERNITY LE. However, at a time, only as
many extensions as the number of Voice Channels supported by the The VoIP Card can make calls.
You can register the same SIP Extension from three different locations.
You may also connect the Standard and Extended IP Phones of Matrix.
The Matrix Extended IP Phone, SPARSH VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Screening
To connect SPARSH VP248 with ETERNITY, see Connecting SPARSH VP248 as Extended SIP Extension.
SPARSH VP330 is proprietary Extended IP Phones with graphical touch-screen user interface. This feature-rich
SIP based phone support most features and functions of the proprietary digital key phones of ETERNITY except
the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Supervision
Login Station from DISA
You cannot program SIP Extension from Enterprise or Hotel Wizard.
To connect SPARSH VP330 with ETERNITY, see Connecting SPARSH VP330 as Extended SIP Extension.
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
32.
138
Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
Logical Partition.
The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
If the ETERNITY LE CPU Card and VoIP Card are connected to a Public Network,
Connect SPARSH VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
ETERNITY ME16S
LAN Switch/Hub
LAN
Master Card
Router
Switch Card
WAN
IP
IP
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
the Registrar Server Address of ETERNITY LE
the Registrar Server Port
the SIP ID
Authentication ID and Password.
139
If the ETERNITY LE Master Card and VoIP Card are connected to a Private Network (Behind the NAT),
ETERNITY ME16S
Master Card
Switch Card
WAN
LAN
LAN Switch/Hub
IP
IP
Router
Connect SPARSH VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
140
use DHCP option 224 and Data Type as String to provide Server Address to the Extended IP
Phones.
Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY LE in the DHCP option.
Log in to Jeeves. For instructions, read the topic Using Jeeves under Configuring ETERNITY.
Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see Configuring SIP Extensions.
For the SIP extension number you assigned to the Extended IP Phone, go to the Location settings of the
extension, and do the following:
For instructions, see the topic Configuring SIP Extension Settings as per the Extended Phone Type
under Configuring SIP Extensions.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SPARSH VP248. For the purpose of illustration, the premium model, SPARSH VP248P, has been used.
1. Unpack the SPARSH VP248 box and verify package contents.
2. Mount the phone on a desk at a location convenient to you.
When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
Foot Stand attached at 30 Angle
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your
desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
141
Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack headset jack with the symbol
Headset
OR
142
You may plug a headset with an RJ12 connector into the headset port at the bottom of the phone, marked
with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
5. Connect the LAN Port of SPARSH VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone33. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
8. Plug the Power Adapter into a power outlet.
9. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
143
The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
As soon as the Loading... message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
When the cursor is placed under the Extended IP Phone, press Enter key.
144
The phone will start loading the Extended IP Phone Firmware. It will display current firmware being loaded.
We l c o me to M a t ri x
L oa d in g V 0 5R 0 1 Ex t S I P
After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See Network Settingsat the end of
this topic.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY LE.
145
As the phone downloads the configuration files, the file names will appear one by one.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
On successful download of all configuration files, the phone attempts to register with ETERNITY LE.
On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The Up key
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
146
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to Local Menu.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
DN D
Names
Local Menu
CA04
CA03
Redial Release
abc
Hold
3 def
4 ghi
jkl
6 mno
7 pqrs
tuv
9 wxyz
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
147
You can configure Network Parameters and view Network status from the Local Menu.
In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
You can configure all network parameters described below, except the MAC Address.
Connection Type
Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
Enter the desired Static IP Address by pressing the digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Subnet Mask
If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Gateway Address
148
If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
DNS Server
If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key 1 twice.
If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
ETERNITY LE Master Card works as the Auto Configuration Server for the phone. Enter the IP Address or
the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default: blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as String to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
Valid range of the port is: 80 or 102465535. Default: 80.
149
VLAN Setting
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic34.
The meaning of CoS bits with respect to traffic type is as follows:
CoS
Traffic Type
Best Effort
Background
Spare
Excellent Effort
Controlled Load
Video
Voice
Network Control
Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
connected. Valid range: 0-4094. Default: 1.
Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
34. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), that is, better handling of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred
to as Class of Service (CoS) which is defined by IEEE 802.1P.
150
PCAP
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see Using PCAP Trace for Matrix Extended IP Phone, under PCAP Trace.
When you change the Network Settings, the phone will restart.
In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
MAC:
IP:
MASK:
G W:
DNS:
N E T W O R K S TAT U S
0 0:1 b:0 9:0 0:9 a:a 7
1 9 2 . 1 6 8 . 2 0 1 .2 0 5
2 5 5 . 25 5 . 2 5 5 .0
1 9 2 . 16 8 . 2 0 1 .3
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY LE.
S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY LE.
Decide the location where you want to place SPARSH VP330 within your LAN.
To use the DHCP Server on your LAN for assigning IP Address to SPARSH VP330, make sure you do the
following:
151
Use DHCP option 224 and Data Type as String to provide Master Ethernet Port Address to SPARSH
VP330.
Program the IP Address or the Domain Name of the Master Ethernet Port of ETERNITY LE in the
DHCP option 224.
Log in to Jeeves. For instructions, read the topic Using Jeeves under Configuring ETERNITY.
You must configure the necessary parameters in ETERNITY so that SPARSH VP330 can register as a SIP
Extension. For instructions, see Configuring Matrix SPARSH VP330.
When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
3. When you mount the phone on a desk, you can attach the Foot Stand in two ways at 30 Angle or at 50
Angle.
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
Plug the long straightened end of the phone cord into the handset jack on the left side panel of the
phone marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
5. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 3.5 mm single
connector into the headset jack headset jack with the symbol
OR
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You may plug a headset with an RJ12 connector into the headset port on the side panel of the phone,
marked with the symbol
Headset
Casio Jack
Headset
(R J12 Connector)
6. Connect the LAN Port of SPARSH VP330 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
7. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port at the bottom of the phone to the LAN Port of the computer.
8. Plug the connector of the Power Adapter in to the power jack at the back of the phone35. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
9. Plug the Power Adapter into a power outlet.
10. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
The LCD display will light up and the following message will appear on it, as the phone boots:
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The Starting SPARSH VP330 ... message appears on the phone display, while loading the
application.
The Applying Network Parameters... message appears on the phone display, while the Static
Network parameters are being applied.
If you want to change the Network Settings or want to use Wi-Fi for connectivity, press Settings
154
To change the Network Settings of the phone and configure the network parameters.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY LE. The Configuring the phone... message appears on the phone.
On successful download of all configuration files, the phone attempts to register with ETERNITY LE. The
Registering the phone... message appears on the phone display.
155
The Updating firmware... message appears on the phone display, when the firmware is being updated.
After the firmware is updated, the phone will reboot. The Rebooting the phone... message appears on
the phone display.
The phone will register successfully, only if the SIP Extension parameters in ETERNITY have been
correctly configured as per your installation scenario.
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The card provides mailbox facility to all extensions of ETERNITY LE. Each Mailbox has the capacity of storing
15,000 messages. The maximum size of each Mailbox is 60,000 minutes. By default, the size of each Mailbox is
set to 5 minutes. The maximum Message Length for each Mailbox is 9999 seconds. By default, the Maximum
Message Length for each Mailbox is set to 15 seconds36.
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 4GB Pen Drive with the VMS
card. The Pen Drive supports 72 hours of recording. However, you may use a Pen Drive of upto 32GB.
The VMS Card has an Ethernet Port, a communication port (COM1), a USB port, and four LEDs.
Ethernet Port
The Ethernet Port is used to connect the VMS card to a computer (standalone or connected in a LAN) to access
and use the embedded FTP server for Software Upgrades, Backup of configuration files and Mailbox messages.
The Ethernet Port can also be used for VMS Debug.
USB Port
The USB port is an internal port, located on the main board of the card. The Pen Drive provided by Matrix with the
VMS Card is connected to this port. All the voice messages, mailbox messages, greetings and other messages and
prompts are stored in the Pen Drive.
The 4GB Pen Drive is factory fitted and shipped with the card. However, you may use a Pen Drive of upto 32GB.
For instructions see Replacing the Pen Drive at the end of this topic.
LEDs
The ETERNITY LE VMS16 has four LEDs: L1, L2, L3 and L4.
36.
When the ETERNITY is installed in the Hospitality Application (Hotel Mode), the default Mailbox size would be 300 minutes and
the default length of messages is 999 seconds.
157
The L1 shows the Status the Card and L2 shows the Status of the USB.
5. Insert the VMS card into the guide rails of the slot. Make sure its connectors fit perfectly into those on the
backplane.
6. Secure the card in its slot by pushing down the levers of the mounting bracket and fixing the card with the
LED
Color
Cadence
L1 to L4
OFF
L1
GREEN
L2 to L4
Continuously ON
OFF
Configuration is being
transferred from Master to
VMS
L1 and L2
GREEN
Blinking
Generating Directories/
Reading Messages
L1 to L3
GREEN
Continuously ON
Initialization
L1 to L4
RED,
GREEN,
ORANGE
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In normal condition
LED L1 will behave in the following manner:
Condition
Color
Cadence
Normal
GREEN
RED
RED
Color
Normal
Cadence
OFF
RED
RED
8. Open Jeeves, and configure the VMS Card. Refer the topic Configuring Voice Mail System for further
configuration instructions.
9. If you need to generate debug reports, connect the COM Port of the VMS Card with that of a PC using the
download Mailbox messages, connect the Ethernet port of the VMS card to a standalone PC or a PC on
LAN.
Plug in one end of the Ethernet cable supplied with the card into the Ethernet Port of the VMS Card.
Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a to a LAN PC, you need to make sure that:
The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the LAN PC are not the
same.
The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
To format the Pen Drive with FAT32, follow the steps given below:
159
160
Click My Computer.
Right-click the removable disk to which you have connected your Pen Drive, in this example Removable
Disk (F:).
The Format Removable Disk (F:) options appear on your screen. In File Format select FAT32.
You will get an alert: WARNING: Formatting will erase ALL data on this disk. To format the disk, click OK.
To quit, click CANCEL.
Click OK to format.
When the formatting process is complete, the message Format Complete will appear on your screen.
Now, copy the contents of the factory fitted Pen Drive onto the new Pen Drive.
161
162
CHAPTER 5
Installing ETERNITY ME
The Matrix ETERNITY is to be installed by persons trained and experienced in telecom wiring.
The person installing the ETERNITY must be familiar with trunks, physical wiring of the MDF on both
the exchange (PBX) side and the line side (CO).
When installing any equipment, make sure that you take all the necessary precautions for handling
electronic and electrical appliances. Follow proper procedures for static electricity, while handling the
system and its cards to prevent damage to the system and harm to yourself.
Use a grounding mat and wear an anti-static strap/belt. Read the dos and don'ts listed in 'Protecting
ETERNITY and Yourself.
If you have complied with the requirements and instructions described in Before You Start, you may
now begin the installation of your ETERNITY ME.
The Matrix ETERNITY ME is shipped factory fitted with the Power supply card, the Master and Switch Card in their
respective fixed slots (refer the section Know Your ETERNITY).
The cards - BRI, T1E1PRI, GSM, VoIP, DKP, CO, SLT, VMS, E&M, Magneto - are shipped separately as per the
order placed by individual customers. These cards can be installed in any of the Universal slots.
Illustrated below is the position of the fixed and universal slots in each variant of ETERNITY ME.
ETERNITY ME16S
In the ETERNITY ME16S, the extreme left slot is reserved for the Power Supply card, the extreme right is reserved
for the Master card, and the second last slot is reserved for the Switch Card. The slots between these fixed slots
are the 16 universal slots to fit the other cards.
163
ETERNITY ME10S
In the ETERNITY ME10S, the first three slots from extreme left slot are reserved for the Power Supply card, the
Master Card and the Switch Card respectively. The remaining slots are the 10 universal slots.
The ETERNITY ME10SR, which offers the redundancy option, has the same organization of the fixed and universal
slots as the ME10S variant, starting with the Power Supply Card on the extreme left. Only, the number of slots
exceeds on account of the second Power Supply Card, Master Card, and Switch Card, provided in this variant to
support the Redundancy feature.
Follow the installation instructions for cards described here, also when you expand the system (add more cards) or
remove or swap cards for maintenance and repair.
1. Unpack the box. Check the package contents (see Packing List). Contact your Dealer/Distributor if any of
the items is missing, faulty or damaged. Do not discard the packaging material.
164
dimensions and weight of the model you have. If mounting the system on a wall, you may refer the
mechanical dimensions and the Mounting Template for drilling holes at appropriate places on the wall.
3. When installing the system in a rack, allow adequate space between the system and other units for air
circulation.
4. Mount the system at the selected site. Make sure that the system is placed in such a way that you have full
access to the front and back panels. The holes in the panels are provided for ventilation; Make sure that
these are not blocked, to prevent overheating.
per the specifications. Earth the system properly. (Refer How to Make the Telecom Earth)
Inserting Cards
7. Make sure that the ETERNITY power is off and the power cord is unplugged.
8. Open the enclosure slot covers by pressing down the snap lugs.
165
Do not force the card into the slot. Doing so can damage the card or the slot connector.
14. When the card is firmly seated in the connector, push down the levers on the card mounting bracket and
Detailed installing instructions are provided for each card - Power Supply Card, Switch Card, DKP, SLT,
CO, ISDN BRI, ISDN T1E1PRI, GSM, VoIP, E&M, etc. - later in this section. Refer to them when installing
each card type.
17. To remove a card:
If you are removing the card permanently or for a certain period of time, install a filler bracket over the
empty card opening in the chassis.
Installing filler brackets over empty card-slot openings is necessary to protect the system from dust,
dirt, insects and damage.
18. Connect the cables supplied with the cards and lead the cables through the cable guides provided below
the slots in the enclosure. This will ensure neat and tangle-free cabling.
19. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.
20. Close the enclosure cover, pressing down the snap lug as you push each part of the cover in its place.
166
PS UNI Card with 100-240VAC, 47-63Hz Mains as Input AC Voltage Power Supply.
This card is designed on the SMPS scheme. As this card does not have any provision for battery backup,
it is recommended that a UPS be connected to keep the system powered during outages.
This card has four LEDs, a Mains Switch, and a Socket assembly for connecting the mains cord.
PS48V Card with 48VDC as Input DC Power Supply Voltage. A Float cum Boost Charger (FCBC) is
required to feed 48VDC power to the card. The FCBC works on input AC mains.
This card is available in two variants - DC to DC 400 W and DC to DC 200W.
The card has four LEDs, an MCB Switch, a power ON/OFF Switch, and a 3-way termination block for
connecting the power cord.
Both, the PS UNI card and the PS48V Card provide DC output voltages as: +3.5V, +5.0V, -27V and -85V.
These are indicated by LEDs.
The ETERNITY provides Redundancy option for the Power Supply card only in the ETERNITY ME10S
variant and for the PS48V card only.
The ETERNITY ME10S model supports two PS48V power supply cards. Whenever there is a fault in
one, the other takes over the control, providing uninterrupted communication.
The maximum number of ports supported by the GSM, SLT and DKP Cards may vary according to the
type of Power Supply used. Refer the following table for maximum ports supported with Universal
Power (PSUNI) and DC Power Supply.
GSM
SLT
DKP
96 ports maximum
Analog SLT ports supported for Short Loop with Loop Current programmed
Loop Current Programmed
20mA
25mA
30mA
35mA
40mA
250
200
172
150
128
167
GSM
SLT
DKP
128 ports
GSM
SLT
DKP
96 ports maximum
If already installed, switch OFF power supply, unplug the power cord. Remove the screws securing the
card. Lift the levers on the mounting bracket to release the card. As the card emerges from the slot, ease it
out of the slot.
2. Insert the Power Supply card into the guide rails of the first slot on the extreme left, designated for the
Power Supply Card. Make sure that the card is inserted deep enough to make perfect contact with the
connectors on the motherboard at the backplane.
3. Now, press down the levers on the card mounting bracket to secure the card in its slot.
4. Secure the card in the slot by screwing the bracket on both ends.
To install a second PS48V card on the ETERNITY ME10S for redundancy, insert the second card on the
next slot. Also refer the topic Hardware Overview in Know Your ETERNITY.
5. If installing the PSUNI card, connect the three-pin power cord into the socket of the PS UNI card and plug
168
Select a UPS considering the typical power consumption of ETERNITY presented in the table below:
Model
Power Consumption
(Typical)
ETERNITY ME10S
70 watts
ETERNITY ME16S
100 watts
6. If installing the PS48V card, connect the Float cum Boost Charger (FCBC). Terminate the power cord from
the FCBC output into the 3-way termination block on the PS48V card.
Polarity is critical. Ensure that the wires are connected with the correct polarity. Follow the standard color
codes used by FCBC manufacturers:
Color
Signal
Red
+48VDC
Black
GND
Green
Earth
It is recommended that you measure the voltage before connecting the power cable to the power supply
card. Ensure that the earth is connected.
FCBC
10A
41 to 56V
ETERNITY ME
Card PS48VDC
48V Battery
169
If two PS48V cards are installed for redundancy (possible in ETERNITY ME10S only), each must be
connected to a separate FCBC and each FCBC must be connected to a separate source of power supply.
7. Connect Battery back up to the FCBC37.
Battery backup time depends upon the total load. The total load is the sum of system's load and load of
active extensions. The power consumed by the variants of ETERNITY ME is given in the table below:
Model
ETERNITY ME10S
70 watts
ETERNITY ME16S
100 watts
The Battery back up time depends on the 'Ah' rating of the battery connected to the FCBC. If 48V/26Ah
batteries are connected to the FCBC for the ETERNITY ME 10S system then backup time of 2.5 to 3 hrs
can be ensured. The FCBC uses the constant voltage charging method. So, the batteries get charged
faster if less power is consumed by the system when in mains mode.
8. Switch on power supply, after completing all other installation.
37.
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When the batteries are drained, the FCBC goes into the boost mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery voltage goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to boost mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the batteries are charged with higher current initially.
Communication Ports
There are two asynchronous, serial, full-duplex RS-232C Communication (COM) Ports, labeled as COM1 and
COM2. The COM Ports have two identical DB-9 connectors.
The COM port allows you to connect a PC to the ETERNITY, so that you can install and operate the following
features:
38
The Ethernet Port on the fascia of the Master Card is provided to connect ETERNITY to a PC or a LAN to operate
the web-based configuration software Jeeves and the Property Management Software (PMS) for Hotel Application.
A cable with a standard RJ45 connector is provided for the Master Ethernet Port.
Printer Port
The Printer port, labeled PRN, on the Master Card is an industry standard Centronics port with a DB-25 female
connector. You can connect any standard printer. The system sends data in the pure ASCII format. No special
characters or control sequences are sent.
38.
The Ethernet port is supported on Master Cards with PCB version V3R0 onwards.
171
Connector
Location
Function
Printer
DB-25 female
Fascia
Communication
(COM Port)
DB-9 female
Fascia
Communication
(COM Port)
DB-9 female
Fascia
Ethernet Port
RJ45
Fascia
Fascia
USB Port
Digital Input
Port
Loop Sensing-Open/
Close, 5mA, Push-type
Connector.
Fascia
Digital Output
Port
Fascia
LEDs
There are three LEDs and a Reset LED located on the Master Card.
The three LEDs indicate the health of the card during the reset cycle and the health of the system during its normal
functioning. The LED pattern of the Master card is summarized in the table below.
Stage
L1
L2
L3
ORANGE ON
ORANGE ON
ORANGE ON
OFF
OFF
OFF
In normal condition
GREEN Toggle
OFF
OFF
In stand-by
RED Toggle
OFF
OFF
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The Reset LED indicates the status of the COMM Manager. The LED pattern is summarized in the table below.
Stage
Reset LED
ON-1sec-OFF-1sec(Red)
ON-1sec-OFF-1sec(Green)
ON-1sec-OFF-1sec(Orange)
In stand-by
OFF
GREEN Toggle
Jumpers
Jumper J9 on the Master Card is used to Reset the SE Password. Refer the table below:
Jumper Number for
PCB Version V3R3
Position
Function
J8
J10
AB (default)
External Boot
BC
Internal Boot
AB
Reset SE Password
BC (default)
Normal
AB (default)
Embedded ICE
BC
JTAG Mode
AB (default)
BC
Disable: Tx debug
AB (default)
J9
J10
J8
J11
J12
J12
J13
Disable: Rx debug
Do not change the position of Jumpers number J8, J10, J12 and J13.
Redundancy for the Master Card is supported only in the ETERNITY ME 10S model. Two Master Cards
can be installed in ETERNITY ME10S. When the active card fails, the standby card takes over control. As
the system restarts during the take over, all existing calls get disconnected. When the standby card
becomes the active card, the system re-boots automatically, restoring communication, within a few minutes
(2-3 minutes).
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If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the Master Card into the guide rails of the slot designated for the card.
On the ETERNITY ME10S model, the third and fourth slots from the left are fixed for the Master Card.
(Refer the slot illustrations at the beginning of this topic)
On the ETERNITY ME16S model, the last slot on the right side is designated to the Master Card. (Refer
the slot illustrations at the beginning of this topic)
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
3. If installing a second Master Card on the ETERNITY ME10S for redundancy, insert the second card on the
fourth slot from the left, next to the first Master Card.
4. You can connect the following external devices to the appropriate ports on the ETERNITY Master Card:
Connecting a Printer
5. You can connect any standard printer to the Printer Port (25-pin connector).
1 mA
7 mA
Use 0.5mm, non-stranded cables to connect the sensor device to the DIP.
174
strip off about half a centimeter of the insulation off the wire ends of the sensor device.
using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
ensure that both wires fit neatly into the opening.
release pressure on the levers. Both wires will be held in place by spring clamp action.
If you are using ETERNITY ME10S with Redundancy Option, connect the DIP on the Active and the
Standby Master Cards as illustrated below.
A DC contactor (60VDC max.) can be connected to the DOP. Any external relay based device can be
interfaced with the DOP via this DC contactor.
The DOP has a two-wire, push-in (spring clamp action) connector to attach the relay device.
Matrix ETERNITY System Manual
175
Contact Arrangement
Operation Time
5 ms
strip off about half a centimeter of the insulation off the wire ends of the gadget.
using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
ensure that both wires fit snugly into the openings.
release pressure on the levers. Both wires will be held in place by spring clamp action.
If you are using ETERNITY ME10S with Redundancy Option, connect the DOP on the Active and the
Standby Master Cards as illustrated below.
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The Ethernet Port is located on the Master Card on ETERNITY ME. With the ETERNITY connected to a
LAN, you can:
access the web-based configuration tool Jeeves from any PC on the LAN.
set up and run software applications such as PMS and CAS on any PC on the LAN.
generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
When you connect the ETERNITY ME to a LAN PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY ME and the Ethernet Port of the PC do
not conflict.
The Master Ethernet Port of ETERNITY ME and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port at the end of this topic.
When you connect the ETERNITY ME to a standalone PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY ME and the Ethernet Port of the PC do
not conflict, are not the same.
The Master Ethernet Port of ETERNITY ME and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port at the end of this topic.
10. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
39.
177
Capture Station Message Detail Recording-Report, Station Message Detail Recording-Online and
Station Message Detail Recording-Posting.
Capture System Activity Log and System Fault Log, Hotel Motel Activity Log
If the system is connected to a LAN PC, ask the LAN Administrator to assign an IP Address and a Subnet
Mask to the ETERNITY ME41.
12. Switch ON the system.
13. Change the IP Address and the Subnet Mask of the Master Ethernet Port by dialing the following
Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
To change IP Address
Switch off power supply and continue with other installation tasks. If you have completed all installation
tasks, start the system and observe the Reset Cycle.
If you have Redundancy option in the Master Card on your ETERNITY ME10S, all configuration settings
must be updated on both the Master Cards by the System Engineer via Jeeves, so that when the standby
card takes over, the system will function with the same configuration settings as the first card. Thus
ensuring smooth take over by the redundant (second) card.
40.
41.
178
Connector
Location
Description
Fascia
Analog Output
Port
Fascia
Fascia
Switch Card Redundancy option is supported only in the ETERNITY ME 10S model. An additional
Switch Card can be installed for redundancy.
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If the main Switch Card fails, the other card takes over. During this period, all existing calls get
disconnected and the standby card becomes active. The system reboots automatically, restoring
communication within 2-3 minutes.
You can 'Hot Swap' the Switch Card in ETERNITY ME 10S and 16S.
If your ME10S has Switch Card Redundancy and you have connected Digital Key Phones to the DKP
ports on the main card Switch Card, you must have the requisite wiring in place to ensure that the
digital key phones continue to work when the redundant Switch card takes control.
Refer the wiring diagram below to install the digital key phones with the main Switch Card and the
Redundant card.
If you are using a new Switch Card (1000 ports) of firmware version V5Rx, use a Master Card with
software version higher than or equal to 'V6R10' only.
If you are using old Switch card (512 ports) of firmware V4Rx, you can use a Master Card with any
software version. (However, in 7th slot, use the cards with configuration of less than 16 ports).
If you are using a DSP based Switch card it is better to use the firmware version of 'V5R1' to avoid
compatibility issues with software version of Master Card in use. (If you are using firmware version of
Switch card less than 'V4Rx', the cards in the 7th and 8th slot will not work. However, in 7th slot, you
can use the cards with configuration, less than 16 ports).
For example, if you are using Switch card with software version 'V4R2', change it to 'V5R1'.
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the Switch Card into the guide rails of the slot designated for the card. On the ETERNITY ME10S
model, the fifth and the sixth slots from the left are reserved for the Switch Card. On the ETERNITY
ME16S model, the second last slot on the right side is designated to the Switch Card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
3. If installing a second Switch Card on the ETERNITY ME10S for redundancy, insert the second card on the
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4. Use the 24 Way-10 Pair (Amphenol connector) MDF cable supplied with the Switch Card to connect the
The cable pinout details for the DKP Ports on the Switch Card are shown in the above figure.
Terminate the free end of the wires of the DKP Ports into the Main Distribution Frame. Crimping each wire
into the punch down block of the Krone module. Also, refer The Main Distribution Frame (MDF),
Installing the Digital Key Phone Card.
Value
Interface Type
Audio Signal
181
Specification
Value
Frequency
300Hz to 3400Hz
Maximum Voltage
strip off about half a centimeter of insulation of the wire-pair of the amplifier.
join the stripped end of the amplifier wires with the free end of the wires of the Analog Output Port - the
Blue-Red wire pair - in the Switch Card cable.
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
DC Bias
Isolation
Internal Transformer
600
Termination provided
600
strip off about half a centimeter of insulation of the wire-pair of the external music device.
join the stripped ends of the device wires with the free end of the wires of the Analog Input Port - the GreyWhite wire pair - emerging from the Switch Card cable.
Also refer the topics Background Music (BGM), Music on Hold (MOH), External Music.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages resulting from improper use.
182
8. If you have finished all installation tasks, switch on power supply and observe the Reset Cycle and the
Color
Cadence
RED
GREEN
Togglea
RED
a. The current LED state will remain the same until the next command is received
from the application on the DKP Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command is
received it will be turned Green/Red ON again. This process continues.
Color
Cadence
RED
GREEN
Continuous ON
GREEN
Togglea
RED
a. The current LED state will remain the same until the next command is received from
the application on the DKP Port. For example, if the current LED state is Green/Red
ON, on the next command received, the LED will be turned OFF. It will remain OFF
until the next command is received. When the next command is received it will be
turned Green/Red ON again. This process continues.
183
Connector
Location
Description
Fascia
Analog Output
Port
Fascia
RJ45
Fascia
184
Switch Card Redundancy option is supported only in the ETERNITY ME 10S model. An additional
Switch Card can be installed for redundancy.
If the main Switch Card fails, the other card takes over. During this period, all existing calls get
disconnected and the standby card becomes active. The system reboots automatically, restoring
communication within 2-3 minutes.
You can 'Hot Swap' the Switch Card in ETERNITY ME 10S and 16S.
If your ME10S has Switch Card Redundancy and you have connected Digital Key Phones to the DKP
ports on the main card Switch Card, you must have the requisite wiring in place to ensure that the
digital key phones continue to work when the redundant Switch card takes control.
Refer the wiring diagram below to install the digital key phones with the main Switch Card and the
Redundant card.
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the Switch Card into the guide rails of the slot designated for the card. On the ETERNITY ME10S
model, the fifth and the sixth slots from the left are reserved for the Switch Card. On the ETERNITY
ME16S model, the second last slot on the right side is designated to the Switch Card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
3. If installing a second Switch Card on the ETERNITY ME10S for redundancy, insert the second card on the
185
Refer the illustration below for the pinout details to help you identify the ports.
L1
L2
AIP
AOP
Connector
RJ45
Color
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
Connection
DKP
DKP
DKP
DKP
The cable pinout details for the DKP Ports on the Switch Card are shown in the above figure.
Terminate the free end of the wires of the DKP Ports into the Main Distribution Frame. Crimping each wire
into the punch down block of the Krone module. Also, refer The Main Distribution Frame (MDF),
Installing the Digital Key Phone Card.
186
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
Physical Connector
Specification
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
DC Bias
Isolation
Internal Transformer
600
Termination provided
600
Physical Connector
Plug in the audio jack of the device into the AIP connector.
Also refer the topics Music on Hold (MOH), Background Music (BGM), External Music.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages resulting from improper use.
8. If you have finished all installation tasks, switch on power supply and observe the Reset Cycle and the
187
Color
Cadence
RED
GREEN
Togglea
RED
a. The current LED state will remain the same until the next command is received
from the application on the DKP Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command is
received it will be turned Green/Red ON again. This process continues.
Color
Cadence
RED
GREEN
Continuous ON
GREEN
Togglea
RED
a. The current LED state will remain the same until the next command is received from
the application on the DKP Port. For example, if the current LED state is Green/Red
ON, on the next command received, the LED will be turned OFF. It will remain OFF
until the next command is received. When the next command is received it will be
turned Green/Red ON again. This process continues.
188
ETERNITY ME Card
CO8+SLT24
Combination card, with 8-ports to connect to 8 Two-wire Analog trunk lines and
24 Single Line Telephones
Choose an SLT Card with the configuration that meets your requirement for SLT ports. Also consider the maximum
SLT Port capacity of the system you are installing. The maximum number of SLT ports supported by the variants of
ETERNITY ME are:
Connectors
The SLT Cards have RJ45 connectors, with each connector having 4 SLT ports. A multi-pair, MDF cable is supplied
for each connector.
LEDs
The SLT cards for ETERNITY ME models have a single, tri-color LED to indicate:
the status of any one extension during normal functioning of the system.
You may monitor any of the SLT Extension ports by assigning the LED to that port42.
42.
To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED
Number is the number of the LED on the card, which will monitor the port.
189
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require
you to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of
the phone to dial Flash.
2. Unpack the SLT card and check the package contents. Ensure that the power supply is switched off,
before you begin the installation of the card. Always wear an electrostatic discharge prevention wrist strap/
belt and use a grounding mat.
The SLT Card supports Hot Swap. So, you can insert the SLT Card while the system is switched on.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
For each connector on the SLT Card, there is a separate 4-pair cable with an RJ45 jack on one end and
free at the other end.
190
L1
Connector
Color
RJ45-1
RJ45-3
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
RJ45-4
SLT
SLT
SLT
SLT
RJ45-5
SLT
SLT
SLT
SLT
RJ45-6
SLT
SLT
SLT
SLT
RJ45-7
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
1
RJ45-2
7
RJ45-8
Connection
04
05
06
08
09
10
12
13
14
16
17
18
20
21
22
24
25
26
28
29
30
32
191
L1
Connector
Color
Connection
RJ45-1
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
01
02
RJ45-3
SLT
SLT
SLT
SLT
RJ45-4
SLT
SLT
SLT
SLT
1
RJ45-2
192
04
05
06
08
09
10
12
13
14
16
L1
Connector
1
RJ45-2
Color
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
Connection
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
01
02
04
05
06
08
193
L1
Connector
Color
RJ45-1
Connection
RJ45-3
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
RJ45-4
SLT
SLT
SLT
SLT
RJ45-5
SLT
SLT
SLT
SLT
RJ45-6
SLT
SLT
SLT
SLT
RJ45-7
TWT
TWT
TWT
TWT
TWT
TWT
TWT
TWT
1
RJ45-2
7
RJ45-8
04
05
06
08
09
10
12
13
14
16
17
18
20
21
22
24
01
02
04
05
06
08
7. Plug in the RJ45 end of the MDF cables supplied with the card into the respective connectors.
8. Terminate the open end of the cables into the punch down blocks of the Krone modules designated for
step. If you have completed all installation tasks, power ON the system, observe the Reset Cycle and the
LED Pattern of the SLT Card.
LED Pattern of the SLT Card
Stage
LED Color
Cadence
RED
ON-200ms-OFF 200ms
GREEN
ON-200ms-OFF 200ms
Auto Upgradationa
194
Stage
LED Color
Cadence
RED
ON 500ms-OFF 500ms
GREEN
ORANGE
ORANGE
Flash Failure
None
None
RAM Failure
None
None
Initialization
Stand-by task
Errors
LED Color
LED Cadence
RED
Togglea
RED
a. The current LED state will remain the same until the next command is received from the application on the SLT Port. For example, if the current LED
state is Green/Red ON, on the next command received, the LED will be
turned OFF. It will remain OFF until the next command is received. When the
next command is received it will be turned Green/Red ON again. This process continues.
For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging in
the phone cables into the RJ45 connectors on the card.
When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the first port on the
connector will be assigned to the SLT.
195
Choose an ILC Card with the configuration that meets your requirement for intercom ports. Also, consider the
maximum Port capacity of the system you are installing. The maximum number of intercom ports supported by the
variants of ETERNITY ME are:
Connectors
The ILC Cards have RJ45 connectors with four ports on each connector. A multi-pair, MDF cable is supplied for
each connector.
LEDs
The ILC cards for ETERNITY ME have a single, tri-color LED to indicate the health of the card during the Reset
Cycle.
the wall jack are terminated in the Main Distribution Frame and the telephones are connected to the wall
jacks.
3. Always wear an electrostatic discharge prevention wrist strap/belt and use a grounding mat to prevent
the card. Since, ETERNITY ME supports Hot Swap, you can install the card in power on condition.
43.
196
Check Availability. This card is supported by Firmware V10R06 and later only.
5. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Keep the filler
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
7. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
L1
Connector
Color
RJ45-1
RJ45-3
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
RJ45-4
SLT
SLT
SLT
SLT
RJ45-5
SLT
SLT
SLT
SLT
RJ45-6
SLT
SLT
SLT
SLT
RJ45-7
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
1
RJ45-2
7
RJ45-8
Connection
04
05
06
08
09
10
12
13
14
16
17
18
20
21
22
24
25
26
28
29
30
32
197
10. If you have completed all other installation tasks, power ON the system, observe the Reset Cycle and the
LED Color
Cadence
Auto Upgradationa
Card waiting for application
Card is up, loaded with new application
RED
ON-200ms-OFF 200ms
GREEN
ON-200ms-OFF 200ms
RED
ON 500ms-OFF 500ms
GREEN
ORANGE
ORANGE,
GREEN
Initialization
Stand-by task
Errors
Flash Failure
None
None
RAM Failure
None
None
198
ETERNITY ME DKP32
ETERNITY ME DKP16
ETERNITY ME DKP8
a. EON74, the proprietary Digital Turret of EON74 is currently supported only on ETERNITY ME DKP16 Card and ETERNITY ME DKP8
Card with firmware version-revision V04R01 and onwards.
Select a DKP Card with the configuration that meets your requirement for DKP Ports. Also consider the maximum
DKP Port capacity of the system you are installing.
Both ETERNITY ME 10S and ME16S support a maximum of 128 DKP Ports.
If you have used up the four in-built DKP ports on the Switch Card, you can connect a maximum of 124
DKPs.
Connectors
The DKP Cards have RJ45 connectors, with each connector having 4 DKP ports. A multi-pair MDF cable is
supplied for each connector on the card.
LEDs
The DKP cards for ETERNITY ME have a single, tri-color LED to indicate:
the status of any one of the ports during normal functioning of the system. By default it is assigned to DKP
Port 1.
You may monitor any of the DKP ports by assigning the LED to that port44.
44.
You can do this from the SE mode, by dialing the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor the port.
199
supply is switched off and you are wearing an antistatic-wrist strap/belt and have a grounding mat.
2. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket, keep for future use to cover empty slots.
3. Insert the DKP card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
4. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
Refer the connector pin details for each DKP Card type given in the following.
45.
200
L1
Connector
Color
Connection
RJ45-1
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
01
02
RJ45-3
DKP
DKP
DKP
DKP
RJ45-4
RJ45-5
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
RJ45-6
RJ45-7
1
RJ45-2
7
RJ45-8
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
04
05
06
08
09
10
12
13
14
16
17
18
20
21
22
24
25
26
28
29
30
32
201
L1
Connector
Color
Connection
RJ45-1
DKP
DKP
DKP
DKP
DKP
DKP
DKP
DKP
01
02
RJ45-3
DKP
DKP
DKP
DKP
RJ45-4
DKP
DKP
DKP
DKP
1
RJ45-2
202
04
05
06
08
09
10
12
13
14
16
L1
Connector
Color
RJ45-1
1
RJ45-2
Connection
DKP
DKP
DKP
DKP
01
02
DKP
DKP
DKP
DKP
04
05
06
08
Plug in the RJ45 end of the MDF cables provided with the DKP card into the respective connectors.
Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the ETERNITY ME DKP Port must be terminated to the bottom of the Krone
Connector, while the wire-pair of the extension line to be connected to this port must be terminated on
the top of the Krone connector. Refer the topic The Main Distribution Frame (MDF) for illustration.
6. Connect the Digital Key Phones to the wall jacks at their respective locations. Detailed installations
instructions for EON, EONSOFT are provided separately. Installation instructions for EON74 are provided
in the EON74 User Guide.
If you have completed all installation tasks, power on the system and observe the Reset Cycle and the LED Pattern
of the DKP Card.
203
LED Color
Cadence
RED
ON-200ms-OFF 200ms
GREEN
ON-200ms-OFF 200ms
RED
ON 500ms-OFF 500ms
GREEN
ON 500ms-OFF 500ms
ORANGE
ON 500ms-OFF 500ms
ORANGE,
GREEN
Flash Failure
None
None
RAM Failure
None
None
Auto Upgradation
Initialization
Stand-by task
Errors
LED Color
LED Cadence
RED
RED
a. The current LED state will remain the same until the next event is received from the application on the DKP Port.
For example, if the current LED state is Green/Red ON, on the next event, the LED will be turned OFF. It will
remain OFF until the next event occurs. When the next event is received it will be turned Green/Red ON again.
This process continues.
b. Same as the above note.
204
Installing EON48
1. Unpack the box and verify the package contents46.
2. Mount the phone on a desk or on the wall at a convenient location.
3. To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
Foot Stand
DND
Redial Release
Hold
abc
3 def
4 ghi
jkl
6 mno
tuv
9 wxyz
7 pqrs 8
CA 3
Keyhole
Slot 2
Line
4P4C Spring
Cord
Press
down
to detach
Foot Stand
Press down
to detach
Foot Stand
Names
CA 4
Keyhole
Slot 1
Headset
Port
CA 2
CA 1
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
5. Connect the handset of the EON48 to the phone body using the spring cord.
46.
205
6. To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
Headset
You may also plug in a stereo headset with an RJ12 connector into the headset port at the bottom of the
phone, marked with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
7. Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
206
8. When the ETERNITY is powered ON, the EON will get reset. The EON communicates with the ETERNITY.
The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the
message 'Please wait' appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PL EASE WAI T .. .
9. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
202 Reception
M on 2 4 A U G 1 2 : 0 0
10. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-
Operation.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
Installing EON31047
1. Unpack the box and verify the package contents48.
2. Mount the phone on a desk or on the wall at a convenient location.
47.
48.
207
3. To mount EON310 on a wall, detach the Foot Stand on the bottom of the phone. Refer to the illustrations in
EON48.
CA 1
CA 2
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON310. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON310 on a desk, you can attach the Foot Stand in two ways - 30 and 50 degree
You may also plug in a stereo headset with an RJ12 connector into the headset port marked with the
symbol
, on the left side panel of the phone as illustrated in the figure below.
Headset
Casio Jack
Headset
(R J12 Connector)
7. Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
208
8. When the ETERNITY is powered ON, the EON will get reset and the message 'Welcome to Matrix.
9. The EON communicates with the ETERNITY. The handshaking lasts for 5-6 seconds. The EON model,
version and revision number, along with the message Please Waitappears on the LCD display.
10. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
11. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-
Operation.
49.
209
You can install two DSS consoles to a DKP. Refer Direct Station Selection Console for possible
combinations for installing the models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated on the bottom of
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
Windows 98
Windows XP
Windows NT
Windows 2003
Windows Vista
Windows 2007
50.
210
2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
have of the operating systems mentioned above, install any compatible Windows Operating System.
6. Now insert the EONSOFT CD-ROM supplied with this PC-based DKP into the CD drive of your Computer.
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
9. Click the Matrix EONSOFT Icon to run the program.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
211
11. Click the shortcut to open the program. The EONSOFT window will open:
12. Click Options at the top left of the window. A drop down menu will appear.
212
14. Select the COM Port to which the communication cable is connected.
213
15. EONSOFT is now connected. If you have already configured the DKP parameters like Access Code and
Name for the port to which EONSOFT is connected, these will appear.
If this window does not appear after you have selected the COM Port Option, test the COM Port for
data transfer.
If the wrong COM port has been selected, a window will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
214
From the drop down menu of Options, select the COM Port to which you have connected the
communication cable.
Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
Click the button labeled Start Test in the COM Port Settings dialog box.
After clicking this button, observe the Test Result section on the dialog box.
The Error Count would show zero as value, if both the communication cable and the COM port were
working.
The above figure shows that the COM Port/communication cable is working.
If the Error Count shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
Remove the communication cable from the COM Port of the PC.
Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
Now, if the error count is zero, please check the Communication Cable.
If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
215
Connectors
The BRI cards have RJ45 Connectors. The ETERNITY ME BRI8 card has 8 RJ45 connectors for 8 BRI ports.
The ETERNITY ME BRI4 card has 4 RJ45 connectors for 4 BRI ports. A separate cable is supplied for each
connector.
LEDs
The ETERNITY ME BRI8 has 4 LEDs and BRI4 has 4 LEDs.
ISDN
Network
NT 1
BRI Port
ETERNITY
Power
U-Interface
(2-wire)
S/T
Interface
Customer Premises
Where,
U Interface = between the NT1 equipment and the ISDN central office.
S/T Interface = between the ISDN user equipment, that is, ETERNITY and the Network Interface
Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the
ETERNITY.
216
TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY via
the NT1 device. Here, the ETERNITY is the Terminal Equipment, so the BRI Port must be programmed to work in
the TE mode.
When an ISDN Phone is to be connected to the BRI port of ETERNITY, the BRI port must be programmed to work
in NT mode.
When a BRI port of another ISDN PBX is to be connected to the BRI port of the ETERNITY, in such a configuration,
you may configure
the BRI port of the other ISDN PBX in the TE mode and the BRI Port of the ETERNITY in the NT mode.
OR
the BRI port of the other ISDN PBX in the NT mode and the BRI Port of the ETERNITY in the TE mode
Point-to-Point Configuration
ISDN
Network
NT
BRI Line
BRI Port
(TE Mode)
(UP to 1 Km.)
ETERNITY
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY, can be upto 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
Further, two configurations are possible in a Point-to-Multipoint configuration:
1. Short Passive Bus Configuration
2. Extended Passive Bus Configuration
217
NT
BRI Port
(TE Mode)
Terminal
Resistance 100
ETERNITY
ISDN Phone
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
Terminal 8
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
In a Short Passive Bus Configuration,
A maximum of 8 TEs or ISDN devices can be connected to a single NT on a bus up to 200 meters from the
NT.
100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access a BRI line and can make outgoing
calls. At the same time, another subscriber from ETERNITY or any ISDN phone shown in the figure can
make outgoing call from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.
d1 < 30 meters
NT
218
Terminal
Resistance 100
BRI Port
(TE Mode)
ETERNITY
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
Where,
TE = Terminal equipment of any ISDN Equipment
NT = Network Termination provided by Service Provider
TR Terminal Resistance 100
d = distance from NT to the last TE Equipment
d1 = the total distance from first TE equipment and the last TE equipment.
In an Extended Passive Bus Configuration,
You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access the BRI line and make outgoing calls.
At the same time, another subscriber from the ETERNITY or any ISDN phone shown in the figure can
make outgoing calls from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
power supply, always wear an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
2. Unpack the BRI card and check the package contents.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
the Orientation Type, change the position of the jumpers located on the Main Board of the card. Refer the
following tables for Jumper Positions for each BRI port.
Jumper Position
Mode
BRI Port 1
BRI Port 2
J20
J21
J22
J23
J24
J25
J26
J27
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
219
Jumper Position
Mode
BRI Port 3
BRI Port 4
J28
J29
J30
J31
J32
J33
J34
J35
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
Jumper Position
Mode
BRI Port 5
BRI Port 6
J36
J37
J38
J39
J40
J41
J42
J43
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
Jumper Position
Mode
BRI Port 7
BRI Port 8
J44
J45
J46
J47
J48
J49
J50
J51
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
By default, Orientation Type is TE. So, you may skip to the next step.
220
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
Last TE equipment.
Last point of the bus bar where the last TE equipment is connected.
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Function
Jumper Position
J3
J4
AB
AB
BC
BC
By default, Termination Resistance of 100 is set on the BRI port (Jumpers J3 and J4 are in AB position).
221
Tx 3
Rx 4
Rx 5
Tx 6
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
100
100
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
222
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Function
Jumper Position
J1
J2
AB
AB
BC
BC
Open
Open
To feed the power on Tx and Rx wires, set the jumpers J1 and J2 of BRI module in AB position.
To feed the power on separate pairs of wires, set the jumpers J1 and J2 of BRI module in BC position.
To power the ISDN terminal from external power source, keep the jumpers J1 and J2 open.
The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
9. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
bracket in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots. Remember to set the Orientation Type, Termination
Resistance and Power Feed, as required.
223
10. Use the straight cables supplied for each connector on the BRI card to connect the BRI Ports to the NT1
device supplied by your ISDN service provider. Refer the configuration and pinout details given below for
guidance.
Configuration details of the U interface (RJ-45) at NT1
Pin Number
Pin Details
Tx
Rx
Pin Details
Rx1
Tx1
Tx2
Rx2
Signal
Color
--
Orange-White
--
Orange
TX_A
Green-White
RX_A
Blue
RX_B
Blue-White
TX_B
Green
VOUT-
Brown-White
VOUT+
Brown
224
Signal
Color
--
Orange-White
--
Orange
RX_A
Green-White
TX_A
Blue
TX_B
Blue-White
RX_B
Green
VOUT-
Brown-White
VOUT+
Brown
The following diagram shows how to connect a BRI Line to the ETERNITY ME BRI port in the TE mode.
11. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
The BRI8 Card has 4 LEDs: L151, L2, L3, L4. These display the status of the first four ports, that is port 1
to 4. To view the status of the ports 5 to 8, from the SE mode dial the SE Command 5323-Software Slot17.
The BRI4 Card has 4 LEDs: L1, L2, L3 and L4. These display the status of each port.
The LEDs show the Status of the Ports as summarized in the table below:
Port Status
LED Color
LED Cadence
RED
Continuously ON
Port is active
GREEN
Continuously ON
51.
This LED keeps blinking. It displays the system heart bits, at the rate of one second. It will remain OFF for one second and will
show the status of port 1 for the next one second.
225
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
226
Last TE equipment.
Last point of the bus bar where the last TE equipment is connected.
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Module 3 (M3)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J6
J8
J7
J9
J6
J8
J7
J9
To insert 100
termination
AB
AB
AB
AB
AB
AB
AB
AB
To remove 100
termination
BC
BC
BC
BC
BC
BC
BC
BC
Module 4 (M4)
Function
Module 5 (M5)
BRI Port 5
BRI Port 6
BRI Port 7
BRI Port 8
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J6
J8
J7
J9
J6
J8
J7
J9
To insert 100
termination
AB
AB
AB
AB
AB
AB
AB
AB
To remove 100
termination
BC
BC
BC
BC
BC
BC
BC
BC
By default, Termination Resistance of 100 is set on the BRI port (the Jumpers are in AB position).
Tx 3
Rx 4
Rx 5
Tx 6
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
100
100
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Matrix ETERNITY System Manual
227
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Enable Feed Power on the BRI Port. For instructions see Power Feed under Configuring BRI Trunks.
By default, the Jumpers are set in AB position to feed power through Tx and Rx wires (Phantom
Power).
If you want to feed power through a separate pair of wires, you may change the position of the Jumpers
on the BRI module as mentioned in the table below.
Module 2 (M2)
Function
228
Module 3 (M3)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J4
J5
J2
J3
J4
J5
J2
J3
AB
AB
AB
AB
AB
AB
AB
AB
Module 2 (M2)
Function
Module 3 (M3)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J4
J5
J2
J3
J4
J5
J2
J3
BC
BC
BC
BC
BC
BC
BC
BC
Module 4 (M4)
Function
Module 5 (M5)
BRI Port 5
BRI Port 6
BRI Port 7
BRI Port 8
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J4
J5
J2
J3
J4
J5
J2
J3
AB
AB
AB
AB
AB
AB
AB
AB
BC
BC
BC
BC
BC
BC
BC
BC
The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
10. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
bracket in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots. Remember to set the Orientation Type, Termination
Resistance and Power Feed, as required.
11. Use the straight cables supplied for each connector on the BRI card to connect the BRI Ports to the NT1
device supplied by your ISDN service provider. Refer the configuration and pinout details given below for
guidance.
Configuration details of the U interface (RJ-45) at NT1
Pin Number
Pin Details
Tx
Rx
229
Pin Details
Rx1
Tx1
Tx2
Rx2
Signal
Color
--
Orange-White
--
Orange
TX_A
Green-White
RX_A
Blue
RX_B
Blue-White
TX_B
Green
VOUT-
Brown-White
VOUT+
Brown
230
Signal
Color
--
Orange-White
--
Orange
RX_A
Green-White
TX_A
Blue
TX_B
Blue-White
RX_B
Green
VOUT-
Brown-White
VOUT+
Brown
The following diagram shows how to connect a BRI Line to the ETERNITY ME BRI port in the TE mode.
12. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle, and the LED pattern of the BRI Card.
The BRI8 Card has 4 LEDs: L152, L2, L3, L4. These display the status of the first four ports, that is port 1
to 4. To view the status of the ports 5 to 8, from the SE mode dial the SE Command 5323-Software Slot17.
The BRI4 Card has 4 LEDs: L1, L2, L3 and L4. These display the status of each port.
The LEDs show the Status of the Ports as summarized in the table below:
Port Status
LED Color
LED Cadence
RED
Continuously ON
Port is active
GREEN
Continuously ON
52.
This LED keeps blinking. It displays the system heart bits, at the rate of one second. It will remain OFF for one second and will
show the status of port 1 for the next one second.
231
The CO Card
The CO Card provides the interface to connect the ETERNITY with the Two-Wire Analog Trunk lines from the CO
Network. The CO Card supports the different standards and features of CO Networks across the world.
The CO Card is available in the following configurations for the variants of ETERNITY ME. CO interface is also
available in combination with SLT ports on a single card.
Choose a CO Card with the configuration that meets your requirement for CO trunk ports, keeping in mind the
maximum CO Trunk Port capacity of the system you are installing.
ETERNITY ME 10S and ME16S both support a maximum of 128 CO Ports.
Connectors
The CO Card has RJ45 connectors, with 4 CO ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
LED
The CO Cards have a single tri-color LED to indicate:
the status of a selected Trunk port during normal functioning of the system.
You can assign the LED to any CO port on the card which you want to monitor53.
that power supply is turned off before you begin the installation of the card. Put on an electrostaticdischarge preventive wrist strap/belt and use a grounding mat.
2. Unpack the CO card and check the package contents.
53.
232
To assign the LED to a selected port for monitoring its functioning, you must enter SE mode and dial the SE Command 7902-SlotLED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor
the port.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
should make perfect contact with those of the slot on the backplane motherboard.
5. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
Distribution Frame.
You may refer the illustrations below for pinout details of the connectors on the card.
233
L1
Connector
Color
Connection
RJ45-1
CO
CO
CO
CO
CO
CO
CO
CO
01
02
RJ45-3
RJ45-4
CO
CO
CO
CO
CO
CO
CO
CO
1
RJ45-2
234
04
05
06
08
09
10
12
13
14
16
L1
Connector
RJ45-1
1
RJ45-2
Color
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
RJ45-3
Unused
RJ45-4
Unused
Connection
CO
CO
CO
CO
CO
CO
CO
CO
235
L1
Connector
Color
RJ45-1
RJ45-3
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
RJ45-4
SLT
SLT
SLT
SLT
RJ45-5
SLT
SLT
SLT
SLT
RJ45-6
SLT
SLT
SLT
SLT
RJ45-7
CO
CO
CO
CO
CO
CO
CO
CO
1
RJ45-2
7
RJ45-8
Connection
04
05
06
08
09
10
12
13
14
16
17
18
20
21
22
24
01
02
04
05
06
08
7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors.
8. Terminate the free end of the CO Card cable into the punch down blocks of the Krone modules designated
236
LED Color
Cadence
RED
ON-200ms-OFF 200ms
GREEN
ON-200ms-OFF 200ms
RED
ON 500ms-OFF 500ms
GREEN
ORANGE
ORANGE
Flash Failure
None
None
RAM Failure
None
None
Auto Upgradationa
Initialization
Stand-by task
Errors
LED Color
LED Cadence
RED
RED
a. The current LED state will remain the same until the next event is received from the application on
the CO Port. For example, if the current LED state is Green/Red ON, on the next event, the LED
will be turned OFF. It will remain OFF until the next event occurs. When the next event is received
it will be turned Green/Red ON again. This process continues.
b. Same as above note.
237
PRI
Robbed Bit Signaling
Q-Signaling (QSIG)
E&M
When connected to E1 carrier lines, the card supports the following signaling types:
PRI
Channel Associated Signaling (CAS)
Q-Signaling (QSIG)
E&M
The T1E1PRI Card is available in the following configurations for ETERNITY ME:
2-Port card with QSIG support to connect 2 ISDN T1/E1 PRI Lines or ISDN
Compatible Devices
1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Compatible Device
Connectors
The T1E1PRI card has an RJ45 Connector for each port. The ETERNITY ME T1E1PRI Dual card has 2 RJ45
Connectors for the two ports, while the ETERNITY ME T1E1PRI Single card has a single RJ45 Connector.
A cable with RJ45 plugs on both ends is supplied for each connector.
LEDs
The ETERNITY ME T1E1PRI Dual Card has four LEDs: L1, L2, L3 and L4.
The ETERNITY ME T1E1PRI Single Card has two LEDs L1 and L2.
an electrostatic-discharge preventive wrist strap and use a grounding mat. Make sure the power supply is
turned off.
238
Pin1
Pin2
Pin3
Pin4
Termination Resistance ()
OFF
OFF
OFF
ON
0 (Default)
5. By default, termination resistance of PRI port is set as 120, which is for E1 connectivity.
To use the PRI Port for T1 connectivity, termination resistance must be changed to 100.
Use DIP Switch SW5 to change the Termination Resistance of PRI Port 1. Set the Pins of SW5 as
shown below:
Pin-1
Pin-2
Pin-3
Pin-4
Resistance
OFF
OFF
ON
OFF
OFF
ON
OFF
OFF
If using the ETERNITY ME T1E1PRI Dual Card, use DIP Switch SW2 to change the Termination
Resistance of PRI Port 2. Set the Pins of SW2 as shown below:
Pin-1
Pin-2
Pin-3
Pin-4
Resistance
OFF
OFF
ON
OFF
OFF
ON
OFF
OFF
6. Insert the T1E1PRI Card into the guide rails of the free slot you selected for the card. Make sure that the
connectors on the card make perfect contact with those of the slot on the backplane motherboard.
7. Press down the levers on the card mounting brackets to secure the card in its slot. Fix the mounting
interface equipment (modem), which is generally supplied by your ISDN Service Provider along with the
PRI line.
239
ISDN
Network
ETERNITY
G.703
Modem
4-wire
HDSL
(RJ-45 Connector)
DTE
(RJ-45 Connector)
4-wire
PRI Port
G.703
Modem
Power
Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI line, that is,
one at the Local Exchange and other at the Customer's Premises.
At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
The DTE interface of the modem is be connected to the PRI port (RJ-45 connector on the Matrix
ETERNITY ME T1E1PRI Dual/Single Card).
9. Refer the following pin details for connecting the Network Termination Unit with the ETERNITY.
Pin details of HDSL Interface of the G.703 Modem. (HDSL Network Termination Unit)
Pin Number
Pin Details
Line A
Line A
Not used
Line B
Line B
Not used
Not used
Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
240
Pin Number
Pin Details
TX1 (Tip)
TX2 (Ring)
Not used
RX1 (Ring)
RX2 (Tip)
Not used
Not used
Not used
Most of the HDSL Network Termination Unit manufacturers use these connectors. But you are advised to
read the instrument guide of the HDSL Network Termination Unit being used by you.
Tx1 (Ring)
Tx2 (Tip)
NC
NC
Rx2 (Tip)
NC
Rx1 (Ring)
NC
5
6
7
8
The cable wires may have to be crossed depending on the pinout of the DTE Interface of the modem.
10. If you have completed all other installation tasks. Power the system. After the Reset Cycle is completed,
The ETERNITY ME T1E1PRI Single Card has two LEDs: L1 and L2.
Given below are the LED Patterns defined for indicating port states in the signaling types supported by the
ETERNITY ME.
Color
Cadence
GREEN
Continuous ON
CRC4 Alarm
GREEN
241
Port Status
Color
Cadence
BFA Alarm
RED
LOS Alarm
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
RAI Alarm
RED
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
CRC4 Alarm
GREEN
MFA Alarm
RED
BFA Alarm
RED
LOS Alarm
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
Y-Bit Alarm
GREEN
AIS16 Alarm
RED
RAI Alarm
RED
RED
Continuous ON
LED2/LED4 Pattern:
LED2/LED4 Pattern:
242
Port Status
Color
Cadence
No Alarm
GREEN
Continuous ON
RED
AIS Alarm
RED
LOS Alarm
RED
Continuous ON
LED2/LED4 Pattern:
Port Status
Color
Cadence
GREEN
Continuous ON
RED
Continuous ON
Color
Cadence
Maintenance Mode
RED -GREEN
LED2/LED4 Pattern:
Port Status
Color
Cadence
RED
RED
Continuous ON
RED
GREEN
GREEN
Continuous ON
GREEN
Color
Cadence
Port Disable
RED
Continuous ON
LED2/LED4 Pattern:
Port Status
Color
Cadence
Port Disabled
OFF
OFF
243
8-port card to connect to 8 GSM networks with 3G support (8 SIM Cards can be
installed)
4-port card to connect to 4 GSM networks with 3G support (4 SIM Cards can be
installed)
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
Both ETERNITY ME10S and ME16S support up to 32 Mobile ports.
SIM cards from different service providers can be used.
Antenna
There is a single rooftop (RT) antenna for four GSM ports. A splitter connects all the four ports on the card into a
single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to another position
(in case of weak signal).
LEDs
There is a tri-color LED for each mobile port on the card to indicate the functioning of the card and the status of the
ports.
244
If using a GSM/3G card, get the get the SIM Card from the GSM/3G service provider of your choice
ready. Use SIM PIN protection, if required.
power supply should be turned off, and you must be wearing an electrostatic discharge preventive wrist
strap and a have a grounding mat, before you begin handling the card.
3. Unpack the Mobile Card and verify the package contents.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
5. Insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the Mobile
card. You can insert multiple SIM cards of the same GSM service provider or of different service providers.
245
6. Insert the Mobile card into the guide rails of the Universal Slot you have selected for this card. Make sure
that the card is inserted deep enough to make perfect contact with the connectors in the backplane. Now,
press down the levers on the card mount bracket to secure the card in its slot.
7. Connect the antenna provided with the card on the splitter connector on the front panel of the card. You
may also use the antenna cable to place the antenna at another position.
246
register with the Mobile networks automatically. Now, observe the LED Patterns of the Mobile Ports.
At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
Each time the Mobile Port sends a request, such as a Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the Mobile network. It is fixed for 150 seconds for all Mobile ports.
Color
Port disabled
LED OFF
Port idle
LED OFF
Red
Continuous ON
Ring Event
Green
Speech
Green
Continuous ON
GSM initialization
Orange
PUK required
Orange
Orange
SIM Absent
Orange
Orange
247
Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer PLCC-An Introduction to know more.
Closed User Group (CUG), where several PBXs are connected with each other through E&M tie lines54.
PBX expansion, where two PBXs are connected with each other with E&M tie lines.
E&M Trunk Seizure Type55: Immediate, Immediate + Wink, Seizure Pulse, Seizure Pulse + Wink, Express,
and Compander Control Signal.
Address Signaling: Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The ETERNITY E&M Card is available in the following configuration for the ETERNITY ME:
Connectors
The E&M8 card is supplied with RJ45 or Amphenol connectors. On the ETERNITY ME E&M8 card with Amphenol
connectors, the first 4 E&M ports (E&M1 to E&M4) are located on the lower connector, and the remaining four E&M
ports (E&M5 to E&M8) are located on the upper connector.
The ETERNITY ME E&M4 card has a single Amphenol connector with 4 ports.
A separate MDF cable is supplied for each connector.
54.
55.
248
The PBXs in a Closed User Group (CUG) can be connected over ISDN T1E1PRI Lines as well. Refer the topic Closed User
Groups to know more.
This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
LEDs
The ETERNITY ME Card E&M8 has eight tri-color LEDs. The ETERNITY ME Card E&M4 has 4 LEDs, to indicate
the functioning of the ports.
ETERNITY ME10S: 80
ETERNITY ME16S: 128
a Trunk - works like a trunk interface when any of the extensions of the PBX makes an outgoing call
through it.
OR
a Tie Line - takes on a dual personality: functioning as both an extension and a trunk. The E&M port works
like an extension interface for incoming calls. It works like a trunk interface when any extension makes an
outgoing call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
PLCC-An Introduction to know more.
1. Have the necessary wiring for the E&M Analog trunk in place. Take the necessary safety precautions
before you begin handling the card; switch off power supply and always wear an antistatic wrist strap and
use a grounding mat.
2. Unpack the E&M card and check the package contents.
3. The E&M Card supports E&M Interface Type IV and Type V connection. To select the appropriate
Interface Type out of the two, you need to change the Jumper Settings.
Refer the table below to select the desired Interface Type and Speech Interface.
Jumper Number
Position
Function
J1 and J2
AB
BC
249
4. Select the speech interface - 2-wire speech or 4-wire speech - as required, by changing the jumper
Position
Function
J3 and J4
AB
BC
By default all the E&M Ports are set to support 2-wire Speech Interface.
To select 2-wire speech interface for the E&M Port, set Jumpers J3 and J4 (given on E&M module) to
BC Position.
To select 4-wire speech interface for the E&M Port, set Jumpers J3 and J4 on E&M module to AB
Position.
contact with those on the backplane motherboard. Secure the card by pressing down the levers and fix the
bracket with the screws provided with the card.
7. Connect the cables supplied with the E&M card into the RJ45/Amphenol connectors on the E&M Card.
8. Connect the other end of the cable into the E&M Ports of the other PBX/Router/Tie Line equipment by
250
L1 L5
L2 L6
L3 L7
L4 L8
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
RJ45-1
02
RJ45-2
RJ45-3
04
RJ45-4
RJ45-5
06
RJ45-6
08
RJ45-8
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Colour
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Connection
01
02
03
04
05
06
07
08
09
10
Pin No.
RJ45-7
251
252
L1 L3
L2 L4
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
RJ45-1
02
RJ45-2
04
RJ45-4
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
Connection
Colour
01
02
03
04
05
06
07
08
09
10
Open
SB
M OUT
RX SPCH A
SPCH A
SPCH B
RX SPCH B
E IN
BGND
CCC
Gray
Green-White
Green
Orange-White
Blue
Blue-White
Orange
Brown-White
Brown
Gray-White
Pin No.
RJ45-3
253
254
255
If you are connecting two PLCC EPAX in a Power Line Carrier Communication Network Compander
Control Signal (CCS) Connection should be made as illustrated in the block diagram below for any of the
four combinations of E&M and Speech Interfaces illustrated in the previous step.
Compander Control Signal (CCS) is a special type of signal used by Power Line Carrier Communication
Networks to improve quality of speech transmission. The PLCC network expects this signal from the PBX
when speech is established. The E&M Card supports this facility. The ETERNITY sends CCS signal to the
PLCC panel.
When the E&M port is used as an Endpoint; the system sends a CCS to the PLCC panel while making
an outgoing call through the E&M port or when a call is received at the E&M port.
When the E&M port is used for Transit Exchange; the system sends a CCS to the PLCC panel while
there is a Transit call through the E&M port.
9. If you have finished all installation tasks, power ON the system, observe the Reset Cycle and the LED
LED Color
LED Cadence
RED
GREEN
GREEN, ORANGE
Initialization
At Power ON
Stand-Bya
Normal (Port Event)
256
Stage
LED Color
LED Cadence
M-Wire High
GREEN
M-Wire Low
E-Wire High
M-Wire Low
E-Wire and M-Wire High
ORANGE
ORANGE
ORANGE
Eprom failure
ORANGE
ORANGE
Errors
257
The maximum number of Magneto Ports supported by each variant of ETERNITY ME are:
Connectors
The Magneto Card may have an amphenol connector or RJ45 connectors, with 4 ports on each connector. A multipair cable is provided for each connector.
LED
The ETERNITY Magneto8 has 8 LEDs for each magneto port supported by the card.
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports during the normal
functioning of the system.
You may install an MDF to connect the Magneto Ports with the Field Telephone wires.
OR
You may connect the wires from the Magneto Field Telephones directly to the Magneto Port.
You are advised to use a separate set of Krone Modules for connecting the Magneto phones to the
Magneto ports of ETERNITY.
56.
258
A magneto telephone is a local battery telephone set, in which signaling current is provided by a magneto hand generator, usually
a magneto. The hand generator, commonly referred to as 'crank', is located on the right hand side of the telephone set and is
turned to produce energy to ring other phones or to signal the CO. The magneto, also called the generator, is used to convert the
mechanical motion via the crank to produce sufficient energy to ring other phones or to signal the CO.
2. Prepare for the card installation by switching off power supply and wearing an electrostatic discharge
contact with the connectors on the backplane motherboard. Press down the levers of the mounting bracket
to secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
6. Now, plug in the cables supplied with the Magneto Card into the connectors on the card. Terminate the
259
7. Connect the pairs of wires from the Magneto Field Phones to the appropriate pairs emerging from the
Magneto Card of the ETERNITY on the MDF. Refer the cable diagram for the Magneto card with amphenol
connector below.
260
8. If the your Magneto Card has RJ45 connectors, refer to the diagram below.
power cord, switch ON power supply from the mains, switch on the Power supply of the ETERNITY and
observe the Reset Cycle and the LED Pattern of the Magneto Card.
261
LED Color
LED Cadence
RED
GREEN
RED
GREEN
ORANGE
ORANGE,
GREEN,
RED
Initialization
Stand-Bya
Normal (Port Event)
Ring (incoming/outgoing call)
Port Disabled
OFF-Hook (in Speech)
Errors
Invalid Card Configuration Jumper
ORANGE
ORANGE
262
Connectors
The Radio Card has RJ45 connectors. A multi-pair cable is provided for each connector.
LED
The Radio Card has one dual color LED.
You may install an MDF to connect the Radio Ports with the Radio device wires.
OR
You may connect the wires from the Radio device directly to the Radio Port.
You are advised to use a separate set of Krone Modules for connecting the Radio devices to the Radio
ports of ETERNITY.
2. Prepare for the card installation by switching off power supply and wearing an electrostatic discharge
with the connectors on the backplane motherboard. Press down the levers of the mounting bracket to
secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
263
6. Now, plug in the cables supplied with the Radio Card into the connectors on the card. Terminate the free
of the ETERNITY on the MDF. For more details, see The Main Distribution Frame (MDF)
Refer to the Pin-out details given below..
Color
Pin Number
Signaling
PTT
Orange
PTT_RTN
Rx-
Blue
Tx+
Tx-
Green
Rx+
Unused
Brown
Unused
Color
Pin Number
Signaling
PTT
Orange
PTT_RTN
Rx-
Blue
Tx+
Tx-
Green
Rx+
Unused
Brown
Unused
H/w Port
Offset
01 to 08
H/w Port
Offset
01 to 04
power cord, switch ON power supply from the mains, switch on the Power supply of the ETERNITY and
observe the Reset Cycle and the LED Pattern of the Radio Card.
264
LED Pattern
Stage
LED Color
LED Cadence
RED
GREEN
RED
GREEN
ORANGE
ORANGE,
GREEN
RED
GREEN
Initialization
Stand-By
Port Status
Position
Function
J3
AB
Normal
BC
Program
AB
Normal
BC
Program
J4 and J5
In PCB-P-200-80-01-01, for 4-wire speech the Jumpers J1 and J2 on the modules must always be set in
AB position.
In PCB-P-200-80-01-02 onwards only 4-wire speech is possible therefore there are no Jumpers on the
modules.
265
LEDs
The Data Card has one dual color LED.
Status
Color
Cadence
Red
Up-gradation in progress
Green
Red
Continuous ON
Green
Continuous ON
Red
Jumpers
Jumper Number
Position
Function
J9
AB (default)
External Boot
BC
Internal Boot
supply, before you begin the installation of the card. Always wear an electrostatic discharge prevention
wrist strap/belt and use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
3. Insert the Data Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
266
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one Data Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
4. Use the cable supplied with the card to connect the Data Ports to the Ethernet Network (Switch/PC).
5. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
267
ETERNITY ME Card
SLT8+MAG2+CO2+LD2+ENM2 Card
The number of LD ports supported by ETERNITY ME depends on the model and the number of Loop Dial
combination cards installed in the system.
The maximum number of LD ports supported by each model of ETERNITY ME are:
Connectors
The SLT8+MAG2+CO2+LD2+ENM2 Card has 7 RJ45 connectors. A separate cable is provided for each
connector.
LED
There are 8 dual color LEDs on the SLT8+MAG2+CO2+LD2+ENM2 Card
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports after the Reset Cycle.
The LEDs are programmable. You can assign an LED to any port to monitor its functioning57.
57.
268
To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is number of the port on the card to which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor the port.
Installation Scenario
The LD Trunk port is used for connecting two PBX systems with each other, as illustrated below.
EPBX A
EPBX B
LD Port
CO
LD Port
CO
The LD port works as a trunk port for extension users, when they make outgoing calls. When the extension
user of PBX A grabs its LD port by dialing a feature access code, it will receive the dial tone of PBX B. The
extension user of PBX A can dial the desired extension number or external number just like any other
extension of PBX B.
The LD port works as an extension for incoming calls. When the extension user of PBX B grabs the LD
port of PBX B, PBX A feeds dial tone to PBX B over its LD port.
1. Have the necessary wiring in place for LD Ports ready between the desired number of PBXs you want to
interconnect. Prepare for the card installation. Switch off power supply; always wear an electrostatic
discharge preventive wrist strap and use a grounding mat.
2. Unpack the SLT8+MAG2+CO2+LD2+ENM2 Card and check the package contents.
3. Select any universal slot to insert the card. Unscrew the filler bracket and remove it by pushing up the
with the connectors on the backplane motherboard. Press down the levers of the mounting bracket to
secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
5. Now, plug in the cables supplied with the Card into the connectors on the card. Terminate the free ends of
269
Connector Color
RJ45-1
RJ45-3
SLT
SLT
SLT
SLT
SLT
SLT
SLT
SLT
LD
LD
-
RJ45-4
CO
CO
-
MAG
MAG
-
01
02
E&M
01
E&M
02
1
RJ45-2
4
RJ45-5
01
02
04
05
06
08
01
02
02
RJ45-7
For detailed instructions, refer the topics The Main Distribution Frame (MDF) and Terminating Trunk and
Extension Cables on the MDF.
Also refer the topics The Single Line Telephone Card, The CO Card, The E&M Card, The Magneto
Card.
6. Repeat the same steps to install another card.
7. If you have finished installing the desired number of cards, plug in the power cord, switch ON power supply
from the mains, switch on the Power supply of the ETERNITY and observe the Reset Cycle and the LED
Pattern of the card.
270
LED Pattern
Reset Cycle
Stage
LED Color
LED Cadence
RED
GREEN
ORANGE
GREEN, RED
ORANGE
Initialization
At Power ON
Stand-Bya
Errors
Invalid Slot detection
Color
Cadence
RED
Port Idle
GREEN
Off
Port Disable
Off
GREEN
RED
GREEN
GREEN
E-wire low
M-wire and E-wire high
ORANGE
271
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The VoIP card for ETERNITY ME is available in the following configuration.
ETERNITY ME VoIP32
ETERNITY ME VoIP16
Voice Channels
There are 32 Voice Channels on the VoIP32 Card and 16 Voice Channels on the VoIP16 Card, allowing as many
simultaneous calls to be made (using SIP Trunks and/or Extensions) as the number of Voice Channels supported
by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
272
SIP Trunks
The ETERNITY ME VoIP Card supports up to 32 SIP Trunks. You can subscribe to as many as 32 different Internet
Telephony Service Providers (ITSP).
It is possible to program all 32 SIP trunks on a single VoIP Card or in a distributed manner, where more than one
VoIP card is installed in the system.
SIP Extensions
ETERNITY ME supports 999 SIP Extensions. Upto 250 SIP Extensions can be registered with a single VoIP Card.
To register more than 250 SIP Extensions, you need at least two VoIP Cards.
You can register any SIP-enabled device like an IP-phone, a Softphone, analog phone adapter, with the VoIP Card
as the 'SIP Extension' of the ETERNITY ME.
The SIP Extensions function in the same ways as other extensions of the ETERNITY. SIP Extension users can
make and receive calls from and to other extensions of ETERNITY and external numbers over PSTN, GSM, VoIP
and E&M lines58. You can also connect the Standard and Extended IP Phones offered by Matrix as SIP
Extensions.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic License Management.
A SIP Extension can be registered with the VoIP Card of ETERNITY from three different locations. This helps
organizations overcome geographical distances and reduce call costs.
To know more about connecting SIP Extensions, see SIP Extensions.
58.
Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your country prohibit call traffic between the public telephony networks and IP networks, you must configure Logical Partition in your system.
To know more, see Logical Partition.
273
LAN Switch/Hub
LAN
Master Card
Router
Switch Card
WAN
IP
IP
A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
SIP ID/User ID
Authentication User ID
Authentication Password
SIP Registrar Server Address
SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
Network Information:
59.
274
Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
Master Card
Switch Card
WAN
LAN
LAN Switch/Hub
IP
IP
Router
The card is located behind the NAT Router and Private IP is assigned to the WAN port.
When connecting the card in a Private Network, you would require the following information:
IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
Gateway Address
DNS Address
DNS Domain Name (if applicable)
275
VoIP Card connected to the Public Network for Matrix Extended IP Phones
Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
To install the VoIP Card, do the following:
1. Get the items/information listed ready before you install the VoIP card and connect it to the IP network.
2. Observe all prescribed safety precautions when inserting or removing cards. Make sure the Power Supply
is switched off, and you are wearing an antistatic wrist strap/belt and have a grounding mat.
3. Unpack the VoIP card and verify the package contents.
4. Select any of the free Universal Slots of ETERNITY to insert the VoIP Card. Unscrew and remove the filler
bracket of the slot. Preserve the filler bracket for future use.
5. Insert the card into the guide rails of the slot. The card should be inserted deep enough to make perfect
276
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end into the Router/Modem.
Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end of the cable into the Router/Modem.
Connect the LAN Port of the VoIP Card to the LAN Switch/Hub.
8. To insert and connect another VoIP card, repeat the same steps as described above.
9. If you have completed all other installation tasks, you may switch on power supply and observe the Reset
LED Indication
There are two LEDs on the VoIP Card: LED 1 and LED 2.
LED Color
Cadence
Green
Continuous ON
Red
Continuous ON
Red
ON 1 sec-OFF 1 sec
ON 1 sec-OFF 1 sec
Red
Green
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 2500msec
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LED Color
Cadence
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 500msec
ON 500msec-OFF 1500 msec
Green
LED Color
Cadence
Red
Continuous ON
Red
ON 500msec-OFF 3500msec
Red
ON 500msec-OFF 500msec
ON 500msec-OFF 2500 msec
Registration in Progress
Green
Registration Successful
Green
Continuous ON
SIP Trunk Status will be indicated by LED2 only after you have programmed LED Indication in VoIP Port
Parameters.
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SIP Extensions
ETERNITY ME supports up to 999 SIP Extensions. The SIP Extensions function like DKP/SLT extensions of the
ETERNITY ME. SIP Extension users can make and receive calls to any extension user of the ETERNITY and to
external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP60.
ETERNITY supports Video calling on SIP Extensions and SIP Trunks. You can make/receive video calls using SIP
Extension/SIP Trunk. However, video calling is possible only for a SIP to SIP call and the call must be routed
through the same VoIP Card. An Audio call can be converted to video call and vice versa.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY ME.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY ME, and you must have the IP8
License. For more information on Licensing, see License Management.
You can register upto 250 SIP Extensions with a single VoIP Card of ETERNITY ME. However, at a time, only as
many extensions as the number of Voice Channels supported by the The VoIP Card can make calls.
You can register the same SIP Extension from three different locations.
You may also connect the Standard and Extended IP Phones of Matrix.
The Matrix Extended IP Phone, SPARSH VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Screening
To connect SPARSH VP248 with ETERNITY, see Connecting SPARSH VP248 as Extended SIP Extension.
SPARSH VP330 is proprietary Extended IP Phones with graphical touch-screen user interface. This feature-rich
SIP based phone support most features and functions of the proprietary digital key phones of ETERNITY except
the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Supervision
Login Station from DISA
You cannot program SIP Extension from Enterprise or Hotel Wizard.
To connect SPARSH VP330 with ETERNITY, see Connecting SPARSH VP330 as Extended SIP Extension.
60.
Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
Logical Partition.
279
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
If the ETERNITY ME Master Card and VoIP Card are connected to a Public Network,
Connect SPARSH VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
ETERNITY ME16S
LAN Switch/Hub
LAN
Master Card
Router
Switch Card
WAN
IP
IP
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
the Registrar Server Address of ETERNITY ME
the Registrar Server Port
the SIP ID
Authentication ID and Password.
If the ETERNITY ME Master Card and VoIP Card are connected to a Private Network (Behind the NAT),
280
ETERNITY ME16S
Master Card
Switch Card
WAN
LAN
LAN Switch/Hub
IP
IP
Router
Connect SPARSH VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
use DHCP option 224 and Data Type as String to provide Server Address to the Extended IP
Phones.
Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY ME in the DHCP option.
Log in to Jeeves. For instructions, read the topic Using Jeeves under Configuring ETERNITY.
Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see Configuring SIP Extensions.
281
For the SIP extension number you assigned to the Extended IP Phone, go to the Location settings of the
extension, and do the following:
For instructions, see the topic Configuring Matrix SPARSH VP248 - Extended IP Phone.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SPARSH VP248. For the purpose of illustration, the premium model, SPARSH VP248P, has been used.
1. Unpack the SPARSH VP248 box and verify package contents.
2. Mount the phone on a desk at a location convenient to you.
When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
Foot Stand attached at 30 Angle
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your
desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
282
Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack headset jack with the symbol
Headset
OR
283
You may plug a headset with an RJ12 connector into the headset port at the bottom of the phone, marked
with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
5. Connect the LAN Port of SPARSH VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone61. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
8. Plug the Power Adapter into a power outlet.
9. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
284
The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
As soon as the Loading... message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
When the cursor is placed under the Extended IP Phone, press Enter key.
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The phone will start loading the Extended IP Phone Firmware. It will display current firmware being loaded.
We l c o me to M a t ri x
L oa d in g V 0 5R 0 1 Ex t S I P
After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See Network Settingsat the end of
this topic.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY ME.
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As the phone downloads the configuration files, the file names will appear one by one.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
On successful download of all configuration files, the phone attempts to register with ETERNITY ME.
On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The Up key
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
287
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to Local Menu.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
DN D
Names
Local Menu
CA04
CA03
Redial Release
abc
Hold
3 def
4 ghi
jkl
6 mno
7 pqrs
tuv
9 wxyz
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
288
You can configure Network Parameters and view Network status from the Local Menu.
In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
You can configure all network parameters described below, except the MAC Address.
Connection Type
Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
Enter the desired Static IP Address by pressing the digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Subnet Mask
If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Gateway Address
If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
289
DNS Server
If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key 1 twice.
If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
ETERNITY ME Master Card works as the Auto Configuration Server for the phone. Enter the IP Address
or the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default: blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as String to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
Valid range of the port is: 80 or 102465535. Default: 80.
290
VLAN Setting
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic62.
The meaning of CoS bits with respect to traffic type is as follows:
CoS
Traffic Type
Best Effort
Background
Spare
Excellent Effort
Controlled Load
Video
Voice
Network Control
Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
connected. Valid range: 0-4094. Default: 1.
Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
62. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), that is, better handling of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred
to as Class of Service (CoS) which is defined by IEEE 802.1P.
291
PCAP
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see Using PCAP Trace for Matrix Extended IP Phone, under PCAP Trace.
When you change the Network Settings, the phone will restart.
In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
MAC:
IP:
MASK:
G W:
DNS:
N E T W O R K S TAT U S
0 0:1 b:0 9:0 0:9 a:a 7
1 9 2 . 1 6 8 . 2 0 1 .2 0 5
2 5 5 . 25 5 . 2 5 5 .0
1 9 2 . 16 8 . 2 0 1 .3
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY
ME.
S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY ME.
292
Decide the location where you want to place SPARSH VP330 within your LAN.
To use the DHCP Server on your LAN for assigning IP Address to SPARSH VP330, make sure you do the
following:
Use DHCP option 224 and Data Type as String to provide Master Ethernet Port Address to SPARSH
VP330.
Program the IP Address or the Domain Name of the Master Ethernet Port of ETERNITY ME in the
DHCP option 224.
Log in to Jeeves. For instructions, read the topic Using Jeeves under Configuring ETERNITY.
You must configure the necessary parameters in ETERNITY so that SPARSH VP330 can register as a SIP
Extension. For instructions, see Configuring Matrix SPARSH VP330.
When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
3. When you mount the phone on a desk, you can attach the Foot Stand in two ways at 30 Angle or at 50
Angle.
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
Plug the long straightened end of the phone cord into the handset jack on the left side panel of the
phone marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
5. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 3.5 mm single
connector into the headset jack headset jack with the symbol
OR
293
You may plug a headset with an RJ12 connector into the headset port on the side panel of the phone,
marked with the symbol
Headset
Casio Jack
Headset
(R J12 Connector)
6. Connect the LAN Port of SPARSH VP330 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
7. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port at the bottom of the phone to the LAN Port of the computer.
8. Plug the connector of the Power Adapter in to the power jack at the back of the phone63. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
9. Plug the Power Adapter into a power outlet.
10. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
The LCD display will light up and the following message will appear on it, as the phone boots:
294
The Starting SPARSH VP330 ... message appears on the phone display, while loading the
application.
The Applying Network Parameters... message appears on the phone display, while the Static
Network parameters are being applied.
If you want to change the Network Settings or want to use Wi-Fi for connectivity, press Settings
To change the Network Settings of the phone and configure the network parameters.
295
296
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY ME. The Configuring the phone... message appears on the phone.
On successful download of all configuration files, the phone attempts to register with ETERNITY ME. The
Registering the phone... message appears on the phone display.
The Updating firmware... message appears on the phone display, when the firmware is being updated.
After the firmware is updated, the phone will reboot. The Rebooting the phone... message appears on
the phone display.
The phone will register successfully, only if the SIP Extension parameters in ETERNITY have been
correctly configured as per your installation scenario.
297
The card provides mailbox facility to all extensions of ETERNITY ME. Each Mailbox has the capacity of storing
15,000 messages. The maximum size of each Mailbox is 60,000 minutes. By default, the size of each Mailbox is
set to 5 minutes. The maximum Message Length for each Mailbox is 9999 seconds. By default, the Maximum
Message Length for each Mailbox is set to 15 seconds64.
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 4GB Pen Drive with the VMS
card. The Pen Drive supports 72 hours of recording. However, you may use a Pen Drive of upto 32GB.
The VMS Card has an Ethernet Port, a communication port (COM1), a USB port, and four LEDs.
Ethernet Port
The Ethernet Port is used to connect the VMS card to a computer (standalone or connected in a LAN) to access
and use the embedded FTP server for Software Upgrades, Backup of configuration files and Mailbox messages.
The Ethernet Port can also be used for VMS Debug.
USB Port
The USB port is an internal port, located on the main board of the card. The Pen Drive provided by Matrix with the
VMS Card is connected to this port. All the voice messages, mailbox messages, greetings and other messages and
prompts are stored in the Pen Drive.
The 4GB Pen Drive is factory fitted and shipped with the card. However, you may use a Pen Drive of upto 32GB.
For instructions see Replacing the Pen Drive at the end of this topic.
LEDs
The ETERNITY ME VMS16 has four LEDs: L1, L2, L3 and L4.
64.
298
When the ETERNITY is installed in the Hospitality Application (Hotel Mode), the default Mailbox size would be 300 minutes and
the default length of messages is 999 seconds.
The L1 shows the Status the Card and L2 shows the Status of the USB.
5. Insert the VMS card into the guide rails of the slot. Make sure its connectors fit perfectly into those on the
backplane.
6. Secure the card in its slot by pushing down the levers of the mounting bracket and fixing the card with the
LED
Color
Cadence
L1 to L4
OFF
L1
GREEN
L2 to L4
Continuously ON
OFF
Configuration is being
transferred from Master to
VMS
L1 and L2
GREEN
Blinking
Generating Directories/
Reading Messages
L1 to L3
GREEN
Continuously ON
Initialization
L1 to L4
RED,
GREEN,
ORANGE
299
In normal condition
LED L1 will behave in the following manner:
Condition
Color
Cadence
Normal
GREEN
RED
RED
Color
Normal
Cadence
OFF
RED
RED
8. Open Jeeves, and configure the VMS Card. Refer the topic Configuring Voice Mail System for further
configuration instructions.
9. If you need to generate debug reports, connect the COM Port of the VMS Card with that of a PC using the
download Mailbox messages, connect the Ethernet port of the VMS card to a standalone PC or a PC on
LAN.
Plug in one end of the Ethernet cable supplied with the card into the Ethernet Port of the VMS Card.
Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a to a LAN PC, you need to make sure that:
The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the LAN PC are not the
same.
The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
To format the Pen Drive with FAT32, follow the steps given below:
300
Click My Computer.
Right-click the removable disk to which you have connected your Pen Drive, in this example Removable
Disk (F:).
301
302
The Format Removable Disk (F:) options appear on your screen. In File Format select FAT32.
You will get an alert: WARNING: Formatting will erase ALL data on this disk. To format the disk, click OK.
To quit, click CANCEL.
Click OK to format.
When the formatting process is complete, the message Format Complete will appear on your screen.
Now, copy the contents of the factory fitted Pen Drive onto the new Pen Drive.
Starting Up ETERNITY ME
Power ON
1. If you have completed all the installation tasks, switch on power supply.
For PSUNI card installed in the system, connect the three-prong plug of the power cord from the
ETERNITY into the AC outlet, and switch on power supply.
For PS48V card installed in the system, keep the MCB Switch ON and power the FCBC.
Reset Cycle
All the LEDs of the system, the cards and the keys of the DKP attached to the System are turned on.
Interpreting LEDs
The functioning of the LEDs of the system and the various cards and their meaning are summarized in the
tables below. This will help you to verify if the system is operating properly and locate faults, where they
occur.
Reset cycle and the LED pattern of each card, where applicable.
4. Now, close the enclosure cover, pressing down the snap lug as you push each part of the cover in its
place.
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304
CHAPTER 6
Installing ETERNITY GE
The Matrix ETERNITY is to be installed by persons trained and experienced in telecom wiring.
The person installing the ETERNITY must be familiar with trunks, physical wiring of the MDF on both
the exchange (PBX) side and the line side (CO).
When installing any equipment, make sure that you take all the necessary precautions for handling
electronic and electrical appliances. Follow proper procedures for static electricity, while handling the
system and its cards to prevent damage to the system and harm to yourself.
Use a grounding mat and wear an anti-static strap/belt. Read the dos and don'ts listed in Protecting
ETERNITY and Yourself.
If you have complied with the requirements and instructions described in 'Before You Start, you may
now begin the installation of your ETERNITY GE.
The Matrix ETERNITY GE is shipped factory fitted with the Power supply card, the CPU Card in their respective
fixed slots (refer the section Know Your ETERNITY).
The cards - BRI, T1E1PRI, GSM, VoIP, DKP, CO, SLT, VMS, E&M - are shipped separately as per the order placed
by individual customers. These cards are installed in any of the Universal slots.
Illustrated below is the position of the fixed and universal slots in each variant of ETERNITY GE.
ETERNITY GE12S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
305
ETERNITY GE6S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
ETERNITY GE3S
The first two slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
Follow the installation instructions for cards described here also when you expand the system (add more cards) or
remove or swap cards for maintenance and repair.
1. Unpack the box. Check the package contents (see Packing List). Contact your Dealer/Distributor if any of
the items is missing, faulty or damaged. Do not discard the packaging material.
dimensions and weight of the model you have. If mounting the system on a wall, you may refer the
mechanical dimensions and the Mounting Template for drilling holes at appropriate places on the wall.
If you are mounting ETERNITY GE6S, make sure the system orientation is horizontal.
306
3. When installing the system in a rack, allow adequate space between the system and other units for air
circulation.
If you are installing ETERNITY GE6S in a rack, make sure the system orientation is horizontal.
4. Mount the system at the selected site. Make sure that the system is place such that you have full access to
the front and back panels. The holes in the panels are provided for ventilation; Make sure that these are
not blocked, to prevent overheating.
per the specifications. Earth the system properly. (Refer How to Make the Telecom Earth)
Inserting Cards
7. Make sure that the ETERNITY power is off and the power cord is unplugged.
8. Select a free slot from the universal slots.
9. Unscrew and remove the filler bracket that covers the card-slot opening of the slot you intend to use.
10. Hold the card with the connectors facing you. Do not grab the card from both ends.
11. Slide the card into the slot, along the guide rails provided for each slot at the top and bottom planes.
12. Ensure that the cards are inserted deep enough for all the connector pins on the cards make complete
307
13. When the card is firmly seated in the connector, push down the levers on the card mounting bracket and
Detailed installing instructions are provided for each card - DKP, SLT, CO, ISDN BRI, ISDN T1E1PRI,
GSM, VoIP, E&M - later in this section. Refer to them when installing each card type.
16. To remove a card:
If you are removing the card permanently or for a certain period of time, install a filler bracket over the
empty card opening in the chassis.
Installing filler brackets over empty card-slot openings is necessary to protect the system from dust,
dirt, insects and damage.
17. Using the cables supplied with each card, and terminate the cables in the Main Distribution Frame (SLT,
DKP, CO, and E&M lines), the NT1 device (ISDN BRI lines), ISDN Modem (ISDN PRI Lines), as
applicable.
Lead the cables neatly and tangle-free into the MDF.
18. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.
308
PS UNI Card with 100-240VAC, 47-63Hz Mains as Input AC Voltage Power Supply.
This card is designed on the SMPS scheme. As this card does not have any provision for battery backup,
it is recommended that a UPS be connected to keep the system powered during outages.
This card has four LEDs, a Mains Switch, and a Socket assembly for connecting the mains cord.
PS48V Card with 48VDC as Input DC Power Supply Voltage. A Float cum Boost Charger (FCBC) is
required to feed 48VDC power to the card. The FCBC works on input AC mains.
The card has four LEDs, an MCB Switch, a power ON/OFF Switch, and a 3-way termination block for
connecting the power cord.
Both, the PS UNI card and the PS48V Card provide DC output voltages as: +3.5V, +5.0V, -27V and -85V.
These are indicated by LEDs.
If already installed, switch OFF power supply, unplug the power cord. Remove the screws securing the
card. Lift the levers on the mounting bracket to release the card. As the card emerges from the slot, ease it
out of the slot.
2. Insert the Power Supply card into the guide rails of the first slot on the extreme left, designated for the
Power Supply Card. Make sure that the card is inserted deep enough to make perfect contact with the
connectors on the motherboard at the backplane.
3. Now, press down the levers on the card mounting bracket to secure the card in its slot.
4. Secure the card in the slot by screwing the bracket on both ends.
5. If installing the PSUNI card, connect the three-pin power cord into the socket of the PS UNI card and plug
ETERNITY GE3S
25 watts
ETERNITY GE6S
30 watts
309
Model
ETERNITY GE12S
50 watts
6. If installing the PS48V card, connect the Float cum Boost Charger (FCBC). Terminate the power cord from
the FCBC output into the 3-way termination block on the PS48V card.
Polarity is critical. Ensure that the wires are connected with the correct polarity. Follow the standard color
codes used by FCBC manufacturers:
Color
Signal
Red
+48VDC
Black
GND
Green
Earth
It is recommended that you measure the voltage before connecting the power cable to the power supply
card. Ensure that the earth is connected.
FCBC
10A
41 to 56V
ETERNITY GE
Card PS48VDC
48V Battery
310
When the batteries are drained, the FCBC goes into the charge mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery voltage goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to charge mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the batteries are charged with higher current initially.
Battery backup time depends upon the total load. The total load is the sum of system's load and load of
active extensions. The power consumed by the variants of ETERNITY GE is given in the table below:
Model
ETERNITY GE3S
25 watts
ETERNITY GE6S
30 watts
ETERNITY GE12S
50 watts
The Battery back up time depends on the 'Ah' rating of the battery connected to the FCBC. If 48V/26Ah
batteries are connected to the FCBC for the ETERNITY GE system you can ensure a backup time of 2.5 to
3 hours. The FCBC uses the constant voltage charging method. So, the batteries get charged faster if less
power is consumed by the system when in mains mode.
8. Switch on power supply, after completing all other installation tasks.
311
Communication Port
This is a single asynchronous, serial, full duplex RS232C communication port, labeled as COM1. The COM Port
has a DB-9 connector. The COM port is meant for connecting a PC to the ETERNITY GE. With a PC connected to
the ETERNITY GE you can install and operate from the COM Port the following features:
312
Connector
Location
Function
RJ45
Facia
USB
Fascia
Communication (COM
Port)
DB-9 female
Fascia
Push-type
Fascia
Push-type
Fascia
Push-type
Fascia
Push-type
Fascia
LED
The CPU card has two dual color (Green and Red) LEDs.
LED 1 - L1 works as a Heart Bit of CPU Card. In Normal Condition, L1 will be turned ON Green for 1 sec
and OFF for 1 sec.
LED 2 - L2 indicates the Layer Application status. In Normal condition, L2 will be turned on Orange and will
blink very fast.
313
Jumpers
The position and function of the Jumpers on the CPU Card are:
Jumper Number
Position
Function
J12
AB
Default SE Password.
BC (default)
Normal.
AB
External Music.
BC (default)
Internal Music.
AB (default)
External Boot.
BC
Internal Boot.
J5
J7
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the CPU Card into the guide rails of the slot designated for the card. On all variants of ETERNITY
GE, the second slot from the left (next to the Power Supply Card) is designated for the CPU Card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
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Capture System Activity Log and System Fault Log, Hotel Motel Activity Log
When you connect the ETERNITY GE to a standalone PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY GE and the Ethernet Port of the PC do
not conflict.
The Master Ethernet Port of ETERNITY GE and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port at the end of this topic.
4. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
access the web-based programming tool Jeeves from any PC on the LAN.
set up and run software applications such as PMS and CAS on any PC on the LAN.
generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
When you connect the ETERNITY GE to a LAN PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY GE and the Ethernet Port of the PC do
not conflict.
The Master Ethernet Port of ETERNITY GE and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port below.
315
If the system is connected to a LAN PC, ask the LAN Administrator to assign an IP Address and a Subnet
Mask to the ETERNITY GE68.
7. Switch ON the system.
8. Change the IP Address and the Subnet Mask of the Ethernet Port69 by dialing the following commands
Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
You get programming tone.
To change IP Address
9. Switch off power supply and continue with other installation tasks.
68.
69.
316
Type
1 mA
7 mA
This will not be necessary, if there is a Dynamic Host Configuration Protocol (DHCP) server on the LAN.
If the ETERNITY is connected to a LAN (without a DHCP server), use the IP Address and Subnet Mask given by your LAN Administrator as the new IP Address, Subnet Mask.
Use 0.5mm, non-stranded cables to connect the sensor device to the DIP.
strip off about half a centimeter of the insulation off the wire ends of the sensor device.
using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
ensure that both wires fit neatly into the opening.
release pressure on the levers. Both wires will be held in place by spring clamp action.
A DC contactor (60VDC max.) can be connected to the DOP. Any external relay based device can be
interfaced with the DOP via this DC contactor.
The DOP has a two-wire, push-in (spring clamp action) connector to attach the relay device.
Contact Arrangement
Operation Time
5 ms
317
strip off about half a centimeter of the insulation off the wire ends of the gadget.
using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
insert the stripped ends of the two wires into the two (green-color) openings of the connector, with
one wire in each opening.
ensure that both wires fit snugly into the openings.
release pressure on the levers. Both wires will be held in place by spring clamp action.
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
Use shielded cable for connecting the amplifier with the speakers.
Also refer the topic Paging.
To connect the amplifier and speakers,
318
insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
ensure that both wires fit snugly into the openings.
release pressure on the levers. Both wires will be held in place by spring clamp action.
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
DC Bias
Isolation
Internal Transformer
600
Termination provided
600
on the CPU card, locate the Jumper for External Music Source, Jumper J5. Change its position from
BC (default) to AB.
strip off about half a centimeter of insulation of the wire-pair of the external music device.
using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the AIP connector.
insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
ensure that both wires fit snugly into the openings.
release pressure on the levers. Both wires will be held in place by spring clamp action.
Also refer the topics Music on Hold (MOH), Background Music (BGM), External Music.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
319
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages resulting from improper use.
320
Communication Port
This two serial, full duplex RS232C communication port, labeled as COM1 and COM2. The COM Port has a DB-9
connector. The COM port is meant for connecting a PC to the ETERNITY GE. With a PC connected to the
ETERNITY GE you can install and operate from the COM Port the following features:
321
Connector
Location
Function
RJ45
Facia
USB
Fascia
Communication (COM
Port)
DB-9 female
Fascia
Push-type
Fascia
Push-type
Fascia
Audio jack,
3.5mm
Fascia
Audio jack,
3.5mm
Fascia
LED
The CPU card has two dual color (Green and Red) LEDs.
LED 1 - L1 works as a Heart Bit of CPU Card. In Normal Condition, L1 will be turned ON Green for 1 sec
and OFF for 1 sec.
LED 2 - L2 indicates the Layer Application status. In Normal condition, L2 will be turned on Orange and will
blink very fast.
322
Jumpers
The position and function of the Jumpers on the CPU Card are:
Jumper Number
Position
Function
J12
AB
Default SE Password.
BC (default)
Normal.
AB
External Music.
BC (default)
Internal Music.
AB (default)
External Boot.
BC
Internal Boot.
J5
J7
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the CPU Card into the guide rails of the slot designated for the card. On all variants of ETERNITY
GE, the second slot from the left (next to the Power Supply Card) is designated for the CPU Card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
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Capture System Activity Log and System Fault Log, Hotel Motel Activity Log
When you connect the ETERNITY GE to a standalone PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY GE and the Ethernet Port of the PC do
not conflict.
The Master Ethernet Port of ETERNITY GE and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port at the end of this topic.
4. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
access the web-based programming tool Jeeves from any PC on the LAN.
set up and run software applications such as PMS and CAS on any PC on the LAN.
generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
When you connect the ETERNITY GE to a LAN PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY GE and the Ethernet Port of the PC do
not conflict.
The Master Ethernet Port of ETERNITY GE and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port below.
324
If the system is connected to a LAN PC, ask the LAN Administrator to assign an IP Address and a Subnet
Mask to the ETERNITY GE72.
7. Switch ON the system.
8. Change the IP Address and the Subnet Mask of the Ethernet Port73 by dialing the following commands
Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
You get programming tone.
To change IP Address
9. Switch off power supply and continue with other installation tasks.
1 mA
7 mA
Use 0.5mm, non-stranded cables to connect the sensor device to the DIP.
To connect the sensor device to the DIP,
72.
73.
strip off about half a centimeter of the insulation off the wire ends of the sensor device.
using a blunt pin or a small flat screw driver, push back the (orange-color) levers of the connector.
insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one
wire in each opening.
ensure that both wires fit neatly into the opening.
release pressure on the levers. Both wires will be held in place by spring clamp action.
This will not be necessary, if there is a Dynamic Host Configuration Protocol (DHCP) server on the LAN.
If the ETERNITY is connected to a LAN (without a DHCP server), use the IP Address and Subnet Mask given by your LAN Administrator as the new IP Address, Subnet Mask.
325
A DC contactor (60VDC max.) can be connected to the DOP. Any external relay based device can be
interfaced with the DOP via this DC contactor.
The DOP has a two-wire, push-in (spring clamp action) connector to attach the relay device.
Contact Arrangement
Operation Time
5 ms
326
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
0.707
Vrms across 600
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
DC Bias
Isolation
Internal Transformer
600
Termination provided
600
on the CPU card, locate the Jumper for External Music Source, Jumper J5. Change its position from
BC (default) to AB.
Plug in the audio jack of the device into the AIP connector.
Also refer the topics Music on Hold (MOH), Background Music (BGM), External Music.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages resulting from improper use.
327
ETERNITY GE Card
CO2+DKP2+SLT16
ETERNITY GE Card
CO4+DKP2+SLT12
The maximum number of SLT ports supported by the variants of ETERNITY GE are:
Connectors
The SLT Cards have RJ45 connectors, with each connector having 4 SLT ports. A multi-pair, MDF cable is supplied
for each connector.
LEDs
The Card SLT8 has 2 LEDs, while SLT20 has no LED:
The LEDs indicate the health of the card during the Reset Cycle.
the status of any one of the extension ports during normal functioning of the system.
You can monitor any of the SLT Extension ports by assigning the LED to that port74.
328
LED Color
LED Cadence
GREEN
RED
a. The current LED state will remain the same until the next command is received from the application on the SLT Port. For example, if the current LED state is Green/Red ON, on the
next command received, the LED will be turned OFF. It will remain OFF until the next command is received. When the next command is received it will be turned Green/Red ON
again. This process continues.
b. Same as above note.
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require
you to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of
the phone to dial Flash.
2. Unpack the SLT card and check the package contents. Ensure that the power supply is switched off,
before you begin the installation of the card. Always wear an electrostatic discharge prevention wrist strap/
belt and use a grounding mat.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
74.
To do this, enter SE mode, enter the programming mode from any extension connected to the ETERNITY, by dialing 1#91-1234.
Dial the command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and
Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on
the card, which will monitor the port. Exit programming mode by dialing '00'.
329
For each connector on the SLT Card, there is a separate 4-pair cable with an RJ45 jack on one end and
free at the other end.
Refer the illustrations below for pin out details of each connector.
Connector Color
330
Connection
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
SLT
SLT
SLT
SLT
13
14
15
16
RJ45-5
SLT
SLT
-
17
18
-
RJ45-6
SLT
SLT
-
19
20
-
Connector Color
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
SLT
SLT
SLT
SLT
13
14
15
16
Connection
331
332
333
334
L1 L2
Connector Color
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
TWT
TWT
TWT
TWT
01
02
03
04
RJ45-5
DKP
DKP
01
02
5
RJ45-6
Connection
Unused
335
336
7. Plug in the RJ45 end of the MDF cables supplied with the card into the respective connectors. Refer to the
pinout details of the connectors of each SLT Card type illustrated above.
8. Terminate the open end of the cables into the punch down blocks of the Krone modules designated for
For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging
in the phone cables into the RJ45 connectors on the card.
When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the SLT will be
connected on the first port on the connector.
337
Choose an ILC Card with the configuration that meets your requirement for intercom ports. Also, consider the
maximum Port capacity of the system you are installing. The maximum number of intercom ports supported by the
variants of ETERNITY GE are:
Connectors
The ILC Cards have RJ45 connectors, with each connector having 4 ports. A multi-pair, MDF cable is supplied for
each connector.
LEDs
The Card ILC8 has 2 LEDs, while ILC 20 has no LED:
The LEDs indicate the health of the card during the Reset Cycle.
the status of any one of the extension ports during normal functioning of the system.
the wall jack are terminated in the Main Distribution Frame and the telephones are connected to the wall
jacks.
3. Always wear an electrostatic discharge prevention wrist strap/belt and use a grounding mat to prevent
75.
338
Check Availability. This card is supported by Firmware V10R06 and later only.
4. Unpack the ILC card and check the package contents. Switch off power supply before you install the card.
5. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Keep the filler
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
7. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
Connector Color
Connection
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
SLT
SLT
SLT
SLT
13
14
15
16
RJ45-5
SLT
SLT
-
17
18
-
RJ45-6
SLT
SLT
-
19
20
-
10. If you have completed all other installation tasks, power ON the system, observe the Reset Cycle.
339
ETERNITY GE DKP16
ETERNITY GE DKP8
ETERNITY GE Card
DKP4+SLT16
Combination card, with 4-ports to connect to 4 Digital Key Phones and 16 ports to
connect 16 Single Line Telephones
ETERNITY GE Card
CO2+DKP2+SLT16
Combination card, with 2 ports to connect 2 Two-wire Trunk lines, 2 ports to connect 2
Digital Key Phones, and 16 ports to connect 16 Single Line Telephones
ETERNITY GE Card
CO4+DKP2+SLT12
Combination card, with 4 ports to connect 4 Two-wire Trunk lines, 2 ports to connect 2
Digital Key Phones and 12 ports to connect 12 Single Line Telephones
To connect the proprietary digital key phones with ETERNITY, you must have at least one of the above mentioned
DKP Cards installed in the system.
The maximum number of DKP Ports supported by each variant of ETERNITY GE is:
Connectors
The DKP Cards have RJ45 connectors, with each connector having 4 DKP ports. A multi-pair MDF cable is
supplied for each connector on the card.
LEDs
The DKP16 and DKP8 cards have two dual color LEDs:
LED1 indicates the health of the card during the Reset Cycle.
LED2 monitors the status of any one of the extension ports during normal functioning of the system.
LED 2 can be assigned to any DKP port to monitor the status of that port76.
76.
340
You can do this from the SE mode, by dialing the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor the port.
supply is switched off and you are wearing an antistatic-wrist strap/belt and have a grounding mat.
2. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket, keep for future use to cover empty slots.
3. Insert the DKP card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
4. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
Refer the connector pin details for each DKP Card type given in the following.
77.
341
342
343
L1 L2
Connector Color
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
TWT
TWT
TWT
TWT
01
02
03
04
RJ45-5
DKP
DKP
01
02
RJ45-6
Unused
01
02
Connection
6. Plug in the RJ45 end of the MDF cables provided with the DKP card into the respective connectors.
7. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
LED Color
LED Cadence
GREEN
Togglea
RED
344
a. The current LED state will remain the same until the next command is received
from the application on the DKP Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command
is received it will be turned Green/Red ON again. This process continues.
Installing EON48
1. Unpack the box and verify the package contents78.
2. Mount the phone on a desk or wall at a convenient location.
3. To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
Foot Stand
DND
Redial Release
Hold
abc
3 def
4 ghi
jkl
6 mno
tuv
9 wxyz
7 pqrs 8
CA 3
Keyhole
Slot 2
Line
4P4C Spring
Cord
Press
down
to detach
Foot Stand
Press down
to detach
Foot Stand
Names
CA 4
Keyhole
Slot 1
Headset
Port
CA 2
CA 1
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
78.
345
5. Connect the handset of the EON48 to the phone body using the spring cord.
6. To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
Headset
You may also plug in a stereo headset with an RJ12 connector into the headset port at the bottom of the
phone, marked with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
7. Plug one end of the RJ45 cable supplied with the phone into the RJ45 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
346
8. When the ETERNITY is powered ON, the EON will get reset. The EON communicates with the ETERNITY.
The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the
message 'Please wait' appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
9. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters, these will appear, as illustrated below.
202 Reception
M on 2 4 A U G 1 2 : 0 0
10. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-
Operation.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
Installing EON31079
1. Unpack the box and verify the package contents80.
2. Mount the phone on a desk or on the wall at a convenient location.
79.
80.
347
3. To mount EON310 on a wall, detach the Foot Stand on the bottom of the phone.Refer to the illustrations in
EON48.
CA 1
CA 2
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON310. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON310 on a desk, you can attach the Foot Stand in two ways - 35 and 55 degree
You may also plug in a stereo headset with an RJ12 connector into the headset port marked with the
symbol
, on the left side panel of the phone as illustrated in the figure below.
Headset
Casio Jack
Headset
(R J12 Connector)
7. Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
348
8. When the ETERNITY is powered ON, the EON will get reset and the message 'Welcome to Matrix.
9. The EON communicates with the ETERNITY. The handshaking lasts for 5-6 seconds. The EON model,
version and revision number, along with the message Please Waitappears on the LCD display.
10. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-Operation.
81.
349
You can install two DSS consoles to a DKP. Refer Direct Station Selection Console for possible
combinations for installing the models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated on the bottom of
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
Windows 98
Windows XP
Windows NT
Windows 2003
Windows Vista
Windows 2007
82.
350
2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
9. Click the Matrix EONSOFT Icon to execute installation of the program.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
351
11. Click the shortcut to open the program. The EONSOFT window will open:
12. Click Options at the top left of the window. A drop down menu will appear.
352
14. Select the COM Port to which the communication cable is connected.
353
15. EONSOFT is now connected. If you have already configured the DKP parameters like Access Code and
Name for the port to which EONSOFT is connected, these will appear.
If this window does not appear after you have selected the COM Port Option, test the COM Port for
data transfer.
If the wrong COM port has been selected, a dialog box will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
354
From the drop down menu of Options, select the COM Port to which you have connected the
communication cable.
Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
Click the button labeled Start Test in the COM Port Settings dialog box.
After clicking this button, observe the Test Result section on the dialog box.
The Error Count value shows zero, if both the communication cable and the COM port are working.
The above screen shows that the COM Port/communication cable is working.
If the Error Count shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
Remove the communication cable from the COM Port of the PC.
Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
Now, if the error count is zero, please check the Communication Cable.
If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
355
The CO Card
The CO Card provides the interface to connect the ETERNITY with the Two-Wire Analog Trunk lines from the CO
Network. The CO Card supports the different standards and features of CO Networks across the world.
The CO Card is available in the following configurations for the variants of ETERNITY GE. CO interface is also
available in combination with SLT and DKP ports on a single card.
ETERNITY GE Card
CO8+SLT8
Combination card, with 8 CO ports to connect 8 CO analog trunk lines and 8 SLT
ports to connect 8 Single Line Telephones
ETERNITY GE Card
CO4+SLT16
ETERNITY GE Card
CO2+DKP2+SLT16
ETERNITY GE Card
CO4+DKP2+SLT12
ETERNITY GE 3S: 48
ETERNITY GE6S: 96
ETERNITY GE12S: 128
Connectors
The CO Card has RJ45 connectors, with 4 CO ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
LED
The CO16 and CO8 Cards have two LEDs to indicate:
You can assign the LED to any CO port on the card which you want to monitor83.
83.
356
To assign the LED to a selected port for monitoring its functioning, you must enter SE mode and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor
the port.
Among the combination cards CO2+DKP2+SLT16, CO2+DKP2+SLT8, CO4+CO16 have no LEDs. The
combination card CO8+SLT8 has two LEDs.
LED Color
LED Cadence
GREEN
Togglea
RED
a. The current LED state will remain the same until the next command is received
from the application on the CO Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command
is received it will be turned Green/Red ON again. This process continues.
that power supply is turned off before you begin the installation of the card. Put on an electrostaticdischarge preventive wrist strap/belt and use a grounding mat.
2. Unpack the CO card and check the package contents.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
should make perfect contact with those of the slot on the backplane motherboard.
5. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
Distribution Frame.
Refer the illustrations below for the pinout details of the connectors on each card.
357
L1 L2
Connector Color
RJ45-1
CO
CO
CO
CO
01
02
03
04
RJ45-2
CO
CO
CO
CO
05
06
07
08
RJ45-3
CO
CO
CO
CO
09
10
11
12
RJ45-4
CO
CO
CO
CO
13
14
15
16
L1 L2
Connector Color
RJ45-1
CO
CO
CO
CO
01
02
03
04
RJ45-2
CO
CO
CO
CO
05
06
07
08
RJ45-3
Unused
RJ45-4
Unused
358
L1 L2
Connector Color
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
CO
CO
CO
CO
01
02
03
04
RJ45-4
CO
CO
CO
CO
05
06
07
08
L1 L2
Connector Color
Blue - (Blue & White)
Orange - (Orange & White)
Green - (Green & White)
Brown - (Brown & White)
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
SLT
SLT
SLT
SLT
13
14
15
16
RJ45-5
CO
CO
01
02
RJ45-6
CO
CO
03
04
RJ45-1
359
L1 L2
Connector Color
Connection
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
SLT
SLT
SLT
SLT
13
14
15
16
RJ45-5
DKP
DKP
01
02
RJ45-6
CO
CO
01
02
L1 L2
Connector Color
Connection
RJ45-1
SLT
SLT
SLT
SLT
01
02
03
04
RJ45-2
SLT
SLT
SLT
SLT
05
06
07
08
RJ45-3
SLT
SLT
SLT
SLT
09
10
11
12
RJ45-4
CO
CO
CO
CO
01
02
03
04
RJ45-5
DKP
DKP
01
02
RJ45-6
Unused
360
7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors. Refer to the connector
361
The maximum number of BRI lines supported by each variant of ETERNITY GE are:
Connectors
The BRI card has 4 RJ45 Connectors. A separate cable is supplied for each connector.
ISDN
Network
NT 1
BRI Port
ETERNITY
Power
U-Interface
(2-wire)
S/T
Interface
Customer Premises
Where,
U Interface = between the NT1 equipment and the ISDN central office.
S/T Interface = between the ISDN user equipment, in this case, ETERNITY and the Network Interface
Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the ETERNITY.
362
TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY via
the NT1 device. Here, the ETERNITY is the Terminal Equipment, so the BRI Port must be programmed to work in
the TE mode.
When an ISDN Phone is to be connected to the BRI port of ETERNITY, the BRI port must be programmed to work
in NT mode.
When a BRI port of another ISDN PBX is to be connected to the BRI port of the ETERNITY, in such a configuration,
you may configure
the BRI port of the other ISDN PBX in the TE mode and the BRI Port of the ETERNITY in the NT mode.
OR
the BRI port of the other ISDN PBX in the NT mode and the BRI Port of the ETERNITY in the TE mode
Point-to-Point Configuration
ISDN
Network
NT
BRI Line
BRI Port
(TE Mode)
(UP to 1 Km.)
ETERNITY
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY, can be up to 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
Further, two configurations are possible in a Point-to-Multipoint configuration:
a. Short Passive Bus Configuration
b. Extended Passive Bus Configuration
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NT
BRI Port
(TE Mode)
Terminal
Resistance 100
ETERNITY
ISDN Phone
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
Terminal 8
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
In a Short Passive Bus Configuration,
A maximum of 8 TEs or ISDN devices can be connected to a single NT on a bus up to 200 meters from the
NT.
100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access a BRI line and can make outgoing
calls. At the same time, another subscriber from ETERNITY or any ISDN phone shown in the figure can
make outgoing call from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.
d1 < 30 meters
NT
Terminal
Resistance 100
364
Terminal
Resistance 100
BRI Port
(TE Mode)
ETERNITY
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
Where,
TE = Terminal equipment of any ISDN Equipment
NT = Network Termination provided by Service Provider
TR Terminal Resistance 100
d = distance from NT to the last TE Equipment
d1 = the total distance from first TE equipment and the last TE equipment.
In an Extended Passive Bus Configuration,
You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access the BRI line and make outgoing calls.
At the same time, another subscriber from the ETERNITY or any ISDN phone shown in the figure can
make outgoing calls from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
power supply, always wear an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
2. Unpack the BRI card and check the package contents.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
365
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
Last TE equipment
Last point of the bus bar where the last TE equipment is connected.
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
366
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Module 2 (M2)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J6
J8
J7
J9
J6
J8
J7
J9
To insert 100
termination
AB
AB
AB
AB
AB
AB
AB
AB
To remove 100
termination
BC
BC
BC
BC
BC
BC
BC
BC
By default, Termination Resistance of 100 is set on the BRI port (the Jumpers are in AB position)
Tx 3
Rx 4
Rx 5
Tx 6
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
100
100
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
1
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
367
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Enable Feed Power on the BRI Port. For instructions see Power Feed under Configuring BRI Trunks.
By default, the Jumpers are set in AB position to feed power through Tx and Rx wires (Phantom
Power). If you want to feed power through a separate pair of wires, you may change the position of the
Jumpers on the BRI module as mentioned in the table below.
Module 1 (M1)
Function
Module 2 (M2)
BRI Port 1
BRI Port 2
BRI Port 3
BRI Port 4
Jumper Position
Jumper Position
Jumper Position
Jumper Position
J4
J5
J2
J3
J4
J5
J2
J3
AB
AB
AB
AB
AB
AB
AB
AB
BC
BC
BC
BC
BC
BC
BC
BC
The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
9. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
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If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots. Remember to set the Orientation Type, Termination
Resistance and Power Feed, as required.
10. Use the straight cables supplied for each connector on the BRI card to connect the BRI Ports to the NT1
device supplied by your ISDN service provider. Refer the configuration and pinout details given below for
guidance.
Configuration details of the U interface (RJ-45) at NT1
Pin Number
Pin Details
Tx
Rx
Pin Details
Rx1
Tx1
Tx2
Rx2
Color
Connection
Orange-White
Not connected
Orange
Not connected
Green-White
TxA
Blue
RxA
Blue-White
RxB
Green
TxB
Brown-White
V-
Brown
V+
Color
Connection
Orange-White
Not connected
Orange
Not connected
Green-White
RxA
Blue
TxA
Blue-White
TxB
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Pin
Color
Connection
Green
RxB
Brown-White
V-
Brown
V+
The following diagram shows how to connect a BRI Line to the ETERNITY ME BRI port in the TE mode.
11. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
LED Color
LED Cadence
RED
Continuously ON
Port is active
GREEN
Continuously ON
370
Mode
J1
J2
J3
J4
J6
J7
J8
J9
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
Mode
J10
J16
J17
J18
J14
J15
J19
J20
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
By default all the BRI ports are configured in the TE mode so, all Jumpers are set in AB position.
If the BRI Port is to be configured in the NT mode, all the related Jumpers of each port should be set in BC
position.
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
371
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
Last TE equipment
Last point of the bus bar where the last TE equipment is connected.
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Function
J3
J4
J3
J4
AB
AB
AB
AB
BC
BC
BC
BC
Function
372
J3
J4
J3
J4
AB
AB
AB
AB
BC
BC
BC
BC
By default, Termination Resistance of 100 is set on the BRI port (Jumpers J3 and J4 are in AB position)
8. To remove the 100 termination from the BRI port set the Jumpers J3 and J4 (provided on the BRI
module) in BC position.
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
Tx 3
100
Rx 4
Rx 5
100
Tx 6
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
1
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
373
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Jumper Position
Function
BRI Port1
BRI Port2
BRI Port3
BRI Port4
J1
J2
J1
J2
J1
J2
J1
J2
AB
AB
AB
AB
AB
AB
AB
AB
BC
BC
BC
BC
BC
BC
BC
BC
Open
Open
Open
Open
Open
Open
Open
Open
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The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
11. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
12. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
NT-1
4
ISDN
Network
Rx1
Tx1
Rx
Tx2 5
6
Rx2
Power
U-Interface
(2-wire)
TxA
Tx
RxA
RxB
TxB
NC
V-
NC
V+
S/T Interface
(4-wire)
A BRI port of
ETERNITY
The Tx and Rx of the BRI port should be connected with the Rx and Tx of the NT1 respectively.
Ascertain the pin details of the NT1 from your Service Provider. If required, cross connect the cables.
V- and V+ are used when a TE is connected to BRI port (in this case the port functions as network or
NT).
13. To connect the BRI Lines to the BRI ports, refer the configuration and pinout details given below for
guidance.
Configuration details of the U interface (RJ-45) at NT1
Pin Number
Pin Details
Tx
Rx
Pin Details
Rx1
Tx1
Tx2
Rx2
375
Color
Connection
Orange-White
Not connected
Orange
Not connected
Green-White
TxA
Blue
RxA
Blue-White
RxB
Green
TxB
Brown-White
V-
Brown
V+
Color
Connection
Orange-White
Not connected
Orange
Not connected
Green-White
RxA
Blue
TxA
Blue-White
TxB
Green
RxB
Brown-White
V-
Brown
V+
14. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
LED Color
LED Cadence
RED
Continuously ON
Port is active
GREEN
Continuously ON
376
PRI
Robbed Bit Signaling
Q-Signaling (QSIG)
E&M
When connected to E1 carrier lines, the card supports the following signaling types:
PRI
Channel Associated Signaling (CAS)
Q-Signaling (QSIG)
E&M
The T1E1PRI Card is available in the following configuration for ETERNITY GE:
1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Compatible Device
The maximum number of PRI Lines supported by each variant of ETERNITY GE is:
Connectors
The T1E1PRI card has an RJ45 Connector. A cable with RJ45 plugs on both ends is supplied for the connector.
LEDs
The ETERNITY GE T1E1PRI Card has 2 LEDs - L1 and L2 - for indicating the port states.
an electrostatic-discharge preventive wrist strap and use a grounding mat. Make sure the power supply is
turned off.
2. Unpack the T1E1PRI card and check the package contents.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
377
Position
Meaning
J5
BC
J5
AB
5. Insert the T1E1PRI Card into the guide rails of the free slot you selected for the card. Make sure that the
connectors on the card make perfect contact with those of the slot on the backplane motherboard.
6. Now, press down the levers on the card mounting brackets to secure the card in its slot. Fix the card in
interface equipment (modem), which is usually supplied by your ISDN Service Provider along with the PRI
line.
The diagram below illustrates this.
Customer Premises
ISDN
Network
ETERNITY
G.703
Modem
4-wire
HDSL
(RJ-45 Connector)
DTE
(RJ-45 Connector)
4-wire
PRI Port
G.703
Modem
Power
Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI lineone at
the Local Exchange and other at the Customer's Premises.
At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
The DTE interface of the modem is to be connected to the PRI port (RJ-45 connector on the Matrix
ETERNITY GE T1E1PRI Card).
8. Plug in one end of the RJ45 cable supplied with the card into the card's connector. Plug the other end of
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Pin details of HDSL Interface of the G.703 Modem. (HDSL Network Termination Unit)
Pin Number
Pin Details
Line A
Line A
Not used
Line B
Line B
Not used
Not used
Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
Pin Number
Pin Details
TX1 (Tip)
TX2 (Ring)
Not used
RX1 (Ring)
RX2 (Tip)
Not used
Not used
Not used
Most of the HDSL Network Termination Unit manufacturers use these connectors. But you are advised to
read the installation guide of the HDSL Network Termination Unit being used by you.
Tx1 (Ring)
Tx2 (Tip)
NC
NC
Rx2 (Tip)
NC
Rx1 (Ring)
NC
5
6
7
8
379
The cable wires may have to be crossed depending on the pinout of the DTE Interface of the modem.
10. Repeat the same steps to install another card. It is not necessary to install the other T1E1PRI cards in a
LED Patterns
The ETERNITY GE T1E1PRI Card has 2 LEDs: L1 and L2. Given below are the LED Patterns defined for each port
state in the different signaling types supported by the ETERNITY GE.
Color
Cadence
GREEN
Continuous ON
CRC4 Alarm
GREEN
BFA Alarm
RED
LOS Alarm
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
RAI Alarm
RED
RED
Continuous ON
Port Status
Color
Cadence
GREEN
Continuous ON
CRC4 Alarm
GREEN
MFA Alarm
RED
BFA Alarm
RED
LOS Alarm
RED
Continuous ON
LED2 Pattern:
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LED2 Pattern:
Port Status
Color
Cadence
GREEN
Continuous ON
Y-Bit Alarm
GREEN
AIS16 Alarm
RED
RAI Alarm
RED
RED
Continuous ON
Color
Cadence
No Alarm
GREEN
Continuous ON
RED
AIS Alarm
RED
LOS Alarm
RED
Continuous ON
LED2 Pattern:
Port Status
Color
Cadence
GREEN
Continuous ON
RED
Continuous ON
Color
Cadence
Maintenance Mode
RED -GREEN
LED2 Pattern:
Port Status
Color
Cadence
RED
RED
Continuous ON
RED
GREEN
GREEN
Continuous ON
GREEN
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Color
Cadence
Port Disable
RED
Continuous ON
LED2 Pattern:
382
Port Status
Color
Cadence
Port Disabled
OFF
OFF
Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer PLCC-An Introduction to know more.
Closed User Group (CUG), where several PBXs are connected with each other through E&M tie lines84.
PBX expansion, where two PBXs are connected with each other with E&M tie lines.
E&M Trunk Seizure Type85: Immediate, Immediate + Wink, Immediate with Ack, Immediate with
Ack+Wink, Seizure Pulse, Seizure Pulse + Wink, Express, and Compander Control Signal.
Address Signaling: Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The ETERNITY E&M Card is available in the following configuration for the ETERNITY GE
The number of E&M lines that you can interface with the ETERNITY using the E&M Card varies according to
number of E&M ports supported by the each variant of ETERNITY GE.
The maximum number of E&M ports supported by each variant of ETERNITY are:
84.
85.
ETERNITY GE3S: 12
ETERNITY GE6S: 24
ETERNITY GE12S: 48
The PBXs in a Closed User Group (CUG) can be connected over ISDN T1E1PRI Lines as well. Refer the topic Closed User
Groups to know more.
This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
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Connectors
The E&M card has RJ45 Connectors. A separate MDF cable is supplied for each connector.
LEDs
The ETERNITY GE E&M4 Card has 4 LEDs to indicate the functioning of the ports.
a Trunk - works like a trunk interface when any of the extensions of the PBX makes an outgoing call
through it.
OR
a Tie Line - takes on a dual personality: functioning as both as an extension and a trunk. The E&M port
works like an extension interface for incoming calls. It works like a trunk interface when any extension
makes an outgoing call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
PLCC-An Introduction to know more.
You cannot connect a trunk line or an SLT or a DKP to an E&M port.
1. Have the necessary wiring for the E&M Analog trunk in place. Take the necessary safety precautions
before you begin handling the card; switch off power supply and always wear an antistatic wrist strap and
use a grounding mat.
2. Unpack the E&M card and check the package contents.
3. The E&M Card supports E&M Interface Type IV and Type V connection. To select the appropriate
Interface Type out of the two, you need to change the Jumper Settings.
Refer the table below to select the desired Interface Type and Speech Interface.
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Jumper Number
Position
Function
J1 and J2
AB
BC
To select the Type-V connection for the E&M Port, set Jumpers J1 and J2 (located on the E&M
module) in BC Position.
4. Select the speech interface - 2-wire speech or 4-wire speech - as required, by changing the jumper
Position
Function
J3 and J4
AB
BC
By default all the E&M Ports are set to support 2-wire Speech Interface.
To select 2-wire speech interface for the E&M Port, set Jumpers J3 and J4 (given on E&M module) to
BC Position.
To select 4-wire speech interface for the E&M Port, set Jumpers J3 and J4 on E&M module to AB
Position.
contact with those on the backplane motherboard. Secure the card by pressing down the levers and fix the
bracket with the screws provided with the card.
7. Connect the cables supplied with the E&M card into the RJ45 connectors on the E&M Card.
8. Connect the free ends of the cables into the E&M Ports of the other PBX/Router/Tie Line equipment by
385
386
387
9. If you are connecting two PLCC EPAX in a Power Line Carrier Communication Network Compander
Control Signal (CCS) Connection should be made as illustrated in the block diagram below for any of the
four combinations of E&M and Speech Interfaces illustrated in the previous step.
Compander Control Signal (CCS) is a special type of signal used by Power Line Carrier Communication
Networks to improve quality of speech transmission. The PLCC network expects this signal from the PBX
when speech is established. The E&M Card supports this facility. The ETERNITY sends CCS signal to the
PLCC panel.
When the E&M port is used as an Endpoint; the system sends a CCS to the PLCC panel while making
an outgoing call through the E&M port or when a call is received at the E&M port.
When the E&M port is used for Transit Exchange; the system sends a CCS to the PLCC panel while
there is a Transit call through the E&M port.
10. If you have completed all installation tasks, power ON the system, observe the Reset Cycle and the LED
LED Color
LED Cadence
At Power ON
LED OFF
LED OFF
RED
GREEN
RED
GREEN
Green
E-Wire Low
E-Wire and M-Wire High
388
Orange
4-port card to connect to 4 GSM networks with 3G support (4 SIM Cards can
be installed)
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
The maximum Mobile ports supported by the variants of ETERNITY GE are:
ETERNITY GE3S: 12
ETERNITY GE6S: 24
ETERNITY GE12S: 32
Antenna
ETERNITY GE Card GSM4 has a single antenna for the four ports. A splitter connects all the four ports on the card
into a single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to another
position (in case of weak signal).
389
If using a GSM/3G card, get the SIM Card from the GSM/3G service provider of your choice ready. Use
SIM PIN protection, if required.
2. Make sure that the ETERNITY is installed at a location where sufficient network coverage is available. The
power supply should be turned off, and you must be wearing an electrostatic discharge preventive wrist
strap and a have a grounding mat, before you begin handling the card.
3. Unpack the Mobile card and verify the package contents.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
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5. Insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the Mobile
card. You can insert multiple SIM cards of the same GSM service provider or of different service providers.
6. Insert the Mobile card into the guide rails of the Universal Slot you have selected for this card. Make sure
that the card is inserted deep enough to make perfect contact with the connectors in the backplane. Now,
press down the levers on the card mount bracket to secure the card in its slot.
7. Connect the antenna provided with the card on the splitter connector on the front panel of the card. You
may also use the antenna cable to place the antenna at another position.
8. Repeat Steps 1-7 to insert another Mobile card.
9. If you have completed all installations tasks, power the system.
10. Wait for the system to register with the Mobile network. By default, the Mobile ports are set to select and
register with the Mobile networks automatically. Now, observe the LED Patterns of the Mobile Ports.
At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
Each time the Mobile Port sends a request, such as Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the Mobile network. It is fixed for 150 seconds for all Mobile ports.
391
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The VoIP card is available in the following configuration for the ETERNITY GE.
ETERNITY GE VoIP32
ETERNITY GE VoIP16
Voice Channels
There are 32 Voice Channels on the VoIP32 Card and 16 Voice Channels on the VoIP16 Card, allowing as many
simultaneous calls to be made (using SIP Trunks and/or Extensions) as the number of Voice Channels supported
by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
392
SIP Trunks
The ETERNITY GE supports a maximum of 16 SIP Trunks, allowing you to subscribe to as many as 16 different
Internet Telephony Service Providers (ITSP).
It is possible to program all 16 SIP trunks on a single VoIP Card or program them in a distributed manner, where
more than one VoIP card is installed in the system.
SIP Extensions
ETERNITY GE supports 500 SIP Extensions. Upto 250 SIP Extensions can be registered with a single VoIP Card.
To register more than 250 SIP Extensions, you need at least two VoIP Cards.
Any SIP-enabled device like an IP-phone, a Softphone, analog phone adapter, can be registered with the VoIP
Card and function as the 'SIP Extension' of the ETERNITY GE.
The SIP Extensions function in the same way as other extensions of the ETERNITY. SIP Extension users can make
and receive calls from and to other extensions of ETERNITY and external numbers over PSTN, GSM, VoIP and
E&M lines86. You can also connect the Standard and Extended IP Phones offered by Matrix as SIP Extensions.
A SIP Extension can be registered with the ETERNITY GE from three different locations. This helps organizations
overcome geographical distances and reduce call costs.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic License Management.
LAN Switch/Hub
LAN
CPU Card
WAN
86.
IP
IP
Router
Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your country prohibit call traffic between the public telephony networks and IP networks, you must configure Logical Partition in your system.
To know more, see Logical Partition.
393
A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
SIP ID/User ID
Authentication User ID
Authentication Password
SIP Registrar Server Address
SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
Network Information:
LAN
87.
394
CPU Card
WAN
LAN Switch/Hub
IP
IP
Router
Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
The card is located behind the NAT Router and Private IP is assigned to the WAN port.
When connecting the card in a Private Network, you would require the following information:
IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
Gateway Address
DNS Address
DNS Domain Name (if applicable)
VoIP Card connected to the Public Network for Matrix Extended IP Phones
Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
To install the VoIP Card, do the following:
1. Get the items/information listed ready before you install the VoIP card and connect it to the IP network.
2. Observe all prescribed safety precautions when inserting or removing cards. Make sure the Power Supply
is switched off, and you are wearing an antistatic wrist strap/belt and have a grounding mat.
3. Unpack the VoIP card and verify the package contents.
4. Select any of the free Universal Slots of ETERNITY to insert the VoIP Card. Unscrew and remove the filler
bracket of the slot. Preserve the filler bracket for future use.
5. Insert the card into the guide rails of the slot. The card should be inserted deep enough to make perfect
395
7. Using the Ethernet cable supplied with the VoIP card, connect the LAN and the WAN Port to the IP
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP card and
the other end into the Router/Modem.
Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP card and
the other end of the cable into the Router/Modem.
Connect the LAN Port of the VoIP card to the LAN Switch/Hub.
8. To insert and connect another VoIP card, repeat the same steps as described above.
9. If you have completed all other installation tasks, you may switch on power supply and observe the Reset
LED Pattern
There are two LEDs on the VoIP Card: LED 1 and LED 2.
LED 2 indicates the status of any of the SIP Trunks to which this LED is assigned.
LED Color
Cadence
Green
Continuous ON
Red
Continuous ON
Red
ON 1 sec-OFF 1 sec
ON 1 sec-OFF 1 sec
396
LED Color
Cadence
Red
ON 500msec-OFF 3500msec
Green
ON 500msec-OFF 3500msec
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 2500msec
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 500msec
ON 500msec-OFF 1500msec
Green
LED Color
Cadence
Red
Continuous ON
Red
ON 500msec-OFF 3500msec
Red
ON 500msec-OFF 500msec
ON 500msec-OFF 2500msec
Registration in Progress
Green
ON 500msec-OFF 3500msec
Registration Successful
Green
Continuous ON
SIP Trunk Status will be indicated by LED2 only after you have programmed the LED Indication in the VoIP
Port Parameters.
397
SIP Extensions
ETERNITY GE supports up to 500 SIP Extensions. The SIP Extensions function in the same way as DKP/SLT
extensions of the ETERNITY GE. SIP Extension users can make and receive calls to any extension user of the
ETERNITY and to external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP88.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY GE.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY GE, and must have the IP8 License.
For more information on Licensing, see License Management.
You can register upto 250 SIP Extensions with a single VoIP Card of ETERNITY GE. However, at a time, only as
many extensions as the number of Voice Channels supported by the VoIP Card can make calls.
You can register the same SIP Extension from three different locations.
You may connect the Standard and Extended IP Phones of Matrix as SIP Extensions.
The Matrix Extended IP Phone, SPARSH VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Screening
To connect SPARSH VP248 with ETERNITY, see Connecting SPARSH VP248 as Extended SIP Extension.
SPARSH VP330 is proprietary Extended IP Phones with graphical touch-screen user interface. This feature-rich
SIP based phone support most features and functions of the proprietary digital key phones of ETERNITY except
the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Supervision
Login Station from DISA
You cannot program SIP Extension from Enterprise or Hotel Wizard.
To connect SPARSH VP330 with ETERNITY, see Connecting SPARSH VP330 as Extended SIP Extension.
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
88.
398
Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
Logical Partition.
If the ETERNITY GE Master Ethernet Port and the VoIP Card are connected to a Public Network,
Connect SPARSH VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
ETERNITY GE12S
LAN Switch/Hub
LAN
CPU Card
WAN
IP
IP
Router
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
the Registrar Server Address of ETERNITY GE
the Registrar Server Port
the SIP ID
Authentication ID and Password.
If the ETERNITY GE Master Ethernet Port and VoIP Card are connected to a Private Network (Behind the NAT),
ETERNITY GE12S
LAN
CPU Card
WAN
LAN Switch/Hub
IP
IP
Router
399
Connect SPARSH VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
use DHCP option 224 and Data Type as String to provide Server Address to the Extended IP
Phones.
Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY GE in the DHCP option.
Log in to Jeeves. For instructions, read the topic Using Jeeves under Configuring ETERNITY.
Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see Configuring Matrix SPARSH VP248 - Extended IP Phone.
For the SIP extension number you assigned to the Extended IP Phone, go to the Location settings of the
extension, and do the following:
For instructions, see the topic Configuring Matrix SPARSH VP248 - Extended IP Phone.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SPARSH VP248. For the purpose of illustration, the premium model, SPARSH VP248P, has been used.
1. Unpack the SPARSH VP248 box and verify package contents.
2. Mount the phone on a desk at a location convenient to you.
400
When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your
desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
401
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack headset jack with the symbol
Headset
OR
402
You may plug a headset with an RJ12 connector into the headset port at the bottom of the phone, marked
with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
5. Connect the LAN Port of SPARSH VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone89. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
8. Plug the Power Adapter into a power outlet.
9. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
403
The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
As soon as the Loading... message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
When the cursor is placed under the Extended IP Phone, press Enter key.
404
The phone will start loading the Extended IP Phone Firmware. It will display current firmware being loaded.
We l c o me to M a t ri x
L oading V 05R01 Ext SI P
After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See Network Settingsat the end of
this topic.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY GE.
405
As the phone downloads the configuration files, the file names will appear one by one.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
On successful download of all configuration files, the phone attempts to register with ETERNITY GE.
On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The Up key
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
406
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to Local Menu.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
DN D
Names
Local Menu
CA04
CA03
Redial Release
abc
Hold
3 def
4 ghi
jkl
6 mno
7 pqrs
tuv
9 wxyz
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
407
You can configure Network Parameters and view Network status from the Local Menu.
In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
You can configure all network parameters described below, except the MAC Address.
Connection Type
Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
Enter the desired Static IP Address by pressing the digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Subnet Mask
If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Gateway Address
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If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
DNS Server
If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key 1 twice.
If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
ETERNITY GE CPU Card works as the Auto Configuration Server for the phone. Enter the IP Address or
the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default: blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as String to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
Valid range of the port is: 80 or 102465535. Default: 80.
409
VLAN Setting
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic90.
The meaning of CoS bits with respect to traffic type is as follows:
CoS
Traffic Type
Best Effort
Background
Spare
Excellent Effort
Controlled Load
Video
Voice
Network Control
Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
connected. Valid range: 0-4094. Default: 1.
Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
90. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), that is, better handling of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred
to as Class of Service (CoS) which is defined by IEEE 802.1P.
410
PCAP
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see Using PCAP Trace for Matrix Extended IP Phone, under PCAP Trace.
When you change the Network Settings, the phone will restart.
In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
MAC:
IP :
MASK:
G W:
DNS:
N E T W O R K S TAT U S
0 0:1 b:0 9:0 0:9 a:a 7
1 92. 16 8. 2 0 1 .2 0 5
2 5 5 . 25 5 . 2 5 5 .0
1 9 2 . 16 8 . 2 0 1 .3
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY
GE.
S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY GE.
Decide the location where you want to place SPARSH VP330 within your LAN.
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To use the DHCP Server on your LAN for assigning IP Address to SPARSH VP330, make sure you do the
following:
Use DHCP option 224 and Data Type as String to provide Master Ethernet Port Address to SPARSH
VP330.
Program the IP Address or the Domain Name of the Master Ethernet Port of ETERNITY ME in the
DHCP option 224.
Log in to Jeeves. For instructions, read the topic Configuring using Web-based GUI: Jeeves under
Configuring ETERNITY.
You must configure the necessary parameters in ETERNITY so that SPARSH VP330 can register as a SIP
Extension. For instructions, see Configuring Matrix SPARSH VP330.
When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
3. When you mount the phone on a desk, you can attach the Foot Stand in two ways at 30 Angle or at 50
Angle.
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
Plug the long straightened end of the phone cord into the handset jack on the left side panel of the
phone marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
5. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 3.5 mm single
connector into the headset jack headset jack with the symbol
OR
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You may plug a headset with an RJ12 connector into the headset port on the side panel of the phone,
marked with the symbol
Headset
Casio Jack
Headset
(R J12 Connector)
6. Connect the LAN Port of SPARSH VP330 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
7. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port at the bottom of the phone to the LAN Port of the computer.
8. Plug the connector of the Power Adapter in to the power jack at the back of the phone91. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
9. Plug the Power Adapter into a power outlet.
10. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
The LCD display will light up and the following message will appear on it, as the phone boots:
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The Starting SPARSH VP330 ... message appears on the phone display, while loading the
application.
The Applying Network Parameters... message appears on the phone display, while the Static
Network parameters are being applied.
If you want to change the Network Settings or want to use Wi-Fi for connectivity, press Settings
414
To change the Network Settings of the phone and configure the network parameters.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY ME. The Configuring the phone... message appears on the phone.
On successful download of all configuration files, the phone attempts to register with ETERNITY ME. The
Registering the phone... message appears on the phone display.
415
The Updating firmware... message appears on the phone display, when the firmware is being updated.
After the firmware is updated, the phone will reboot. The Rebooting the phone... message appears on
the phone display.
The phone will register successfully, only if the SIP Extension parameters in ETERNITY have been
correctly configured as per your installation scenario.
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Voice Mail System providing voice mail facility to all extensions of ETERNITY.
The card provides mailbox facility to all extensions of ETERNITY GE. Each Mailbox has the capacity of storing
15,000 messages. The maximum size of each Mailbox is 60,000 minutes. By default, the size of each Mailbox is
set to 5 minutes. The maximum Message Length for each Mailbox is 9999 seconds. By default, the Maximum
Message Length for each Mailbox is set to 15 seconds92
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 4GB Pen Drive with the VMS
card. The Pen Drive supports 72 hours of recording. However, you may use a Pen Drive of upto 32GB.
The VMS Card has an Ethernet Port, a communication port, a USB port, and two LEDs.
Ethernet Port
The Ethernet Port is used to connect the VMS card to a computer (standalone or connected in a LAN) to access
and use the embedded FTP server for Software Upgrades, Backup of configuration files and Mailbox messages.
The Ethernet Port can also be used for VMS Debug.
USB Port
The USB port is an internal port, located on the main board of the card. The Pen Drive provided by Matrix with the
VMS Card is connected to this port. All the voice messages, mailbox messages, greetings and other messages and
prompts are stored in the Pen Drive.
The 4GB Pen Drive is factory fitted and shipped with the card. However, you may use a Pen Drive of upto 32GB.
For instructions see Replacing the Pen Drive at the end of this topic.
LEDs
The ETERNITY GE VMS16 has two LEDs: L1 and L2.
The L1 shows the Status the Card and L2 shows the Status of the USB.
92.
When the ETERNITY is installed in the Hospitality Application (Hotel Mode), the default Mailbox size would be 300 minutes and
the default length of messages is 999 seconds.
417
backplane.
6. Secure the card in its slot by pushing down the levers of the mounting bracket and fixing the card with the
Connecting to a Computer
7. Now, connect the card to a standalone PC/LAN.
Plug in one end of the Ethernet cable supplied with the card into the Ethernet Port of the VMS Card.
Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a standalone/LAN PC, you need to make sure that:
The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the PC do not conflict.
The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
When your system configuration has been done using the maximum capacity of system resources,
configuration transfer from ETERNITY CPU to the VMS Card will take longer to complete. At first power
on, it will take 7 to 10 minutes for the VMS to initialize. At subsequent power on, the VMS card will take
around 3 minutes to initialize.
LED
Color
Cadence
L1 and L2
OFF
L1
GREEN
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L2
L1 and L2
Continuously ON
OFF
GREEN
Blinking
Condition
LED
Color
Cadence
Generating Directories/
Reading Messages
L1 and L2
GREEN
Continuously ON
L1 and L2
RED,
GREEN,
ORANGE
Initialization
In normal condition
LED L1 will behave in the following manner:
Condition
Color
Cadence
Normal
GREEN
RED
RED
Color
Normal
Cadence
OFF
RED
RED
To format the Pen Drive with FAT32, follow the steps given below:
419
420
Click My Computer.
Right-click the removable disk to which you have connected your Pen Drive, in this example Removable
Disk (F:).
The Format Removable Disk (F:) options appear on your screen. In File Format select FAT32.
You will get an alert: WARNING: Formatting will erase ALL data on this disk. To format the disk, click OK.
To quit, click CANCEL.
Click OK to format.
When the formatting process is complete, the message Format Complete will appear on your screen.
Now, copy the contents of the factory fitted Pen Drive onto the new Pen Drive.
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Starting Up ETERNITY GE
Power ON
1. If you have completed all the installation tasks, switch on power supply.
For PSUNI card installed in the system, connect the three-prong plug of the power cord from the
ETERNITY into the AC outlet, and switch on power supply.
For PS48V card installed in the system, keep the MCB Switch ON and power the FCBC.
Reset Cycle
All the LEDs of the system, the cards and the keys of the DKP attached to the System are turned on.
Interpreting LEDs
The functioning of the LEDs of the system and the various cards and their meaning are summarized at the
end of the installation instructions for each Card Type.
Refer to the LED Patterns described for each Card Type to verify if the system is operating properly and locate
faults, where they occur.
When the reset cycle is successful, the default Extension Access Codes loaded by the system and the date
and time of the Real Time Clock (RTC) of the system will appear on the LCD display of the DKPs you have
connected with the system.
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CHAPTER 7
Installing ETERNITY PE
The Matrix ETERNITY is to be installed by persons who are trained and experienced in telecom wiring;
familiar with trunks, physical wiring of the MDF on both the PBX side and the line side (CO).
Take all the necessary precautions for handling electronic and electrical appliances. Follow prescribed
procedures for preventing electrostatic discharges, to prevent damage to the cards and harm to
yourself.
Wear an anti-static wrist strap/belt and use a grounding mat. Read the dos and don'ts listed in
Protecting ETERNITY and Yourself.
If you have complied with the requirements and instructions described in Before You Start, you may
now begin the installation of your ETERNITY PE.
The Matrix ETERNITY PE is shipped factory fitted with the Power supply unit and the CPU Card (refer the section
Know Your ETERNITY).
The cards - SLT, DKP, CO, BRI, T1E1PRI, GSM, VoIP, VMS and the Door Phone Card - are shipped separately as
per the order placed by individual customers.
Illustrated below are the positions of the slots in each variant of ETERNITY PE.
ETERNITY PE6SP
The Power Supply unit and the CPU are in-built, and fixed on the bottom plane of the ETERNITY PE.
423
The Power Supply unit and the CPU are in-built, and fixed on the bottom plane of the ETERNITY PE.
Universal slots are located on the CPU. The connectors of the slots are located on the CPU.
Cards are mounted on the CPU and secured on the three studs on the CPU, with the screws provided.
The ETERNITY PE 6S has an Ethernet port, a Communication Port, an Analog Input Port and an Analog Output
Port.
ETERNITY PE3SP
ETERNITY PE3SP is similar to PE6S, except it has only 3 universal slots.
ETERNITY PE3SS93
ETERNITY PE3SS has 3 universal slots and an Ethernet Port.
93.
424
ETERNITY PE3SS has been discontinued. Software support is still provided through Jeeves for users who have already installed
it.
There is no Communication (COM) Port, no USB port, no Analog Input Port and no Analog Output port.
Instructions are provided in the following for installation of the cards. These instructions are to be followed also
when you expand the system (add more cards) or remove cards for maintenance and repair.
1. Have all the necessary wiring ready. Read the topic Main Distribution Frame for guidance on how to set up
the MDF and connect the system with the MDF, and install Primary Protection against heavy voltages.
2. Unpack the box. Check the package contents (see Packaging List). Contact your Dealer/Distributor if any
of the items is missing, faulty or damaged. Do not discard the packaging material.
If you have decided to mount the ETERNITY PE on a wall, use the Mounting Template for drilling the holes
at appropriate distances on the wall.
4. Mount the system at the selected site. Make sure that the system is placed such that you have full access
to the front and back panels. The holes in the side panels are provided for ventilation; Make sure that these
are not blocked, to prevent overheating.
When installing the system in a rack94 allow adequate space between the system and other units for air
circulation.
Inserting Cards
7. Make sure the power supply is turned off and the power cord is unplugged, before you begin inserting the
cards.
8. Unpack the card and check the package contents.
94.
425
9. Unscrew the top cover of the ETERNITY PE and remove it by sliding it out. Keep the cover and the screws
aside.
card is seated perfectly for all the connector pins on the cards make complete contact with those on the
CPU (motherboard) on the bottom plane.
426
12. When the card is firmly seated in the connector, use the three screws provided with the card to secure it on
Detailed installing instructions are provided for each card - DKP, SLT, CO, ISDN BRI, ISDN T1/E1/PRI,
GSM, VoIP, VMS, Door Phone - later in this section. Refer to them when installing each card type.
14. Using the cables supplied with the cards, connect the card interfaces with the MDF (for SLT, DKP and CO),
the NT1 termination device (for BRI lines), the ISDN modem (for T1/E1 PRI lines), the IP network, a
Computer as applicable for each card.
15. Lead the cables out of the enclosure through any of the two cable outlets on either side of the enclosure.
it in place. Secure the cover with the two screws you removed.
Since the connectors of the cards will not be visible after the cover has been replaced, you are advised to
label the cables appropriately to facilitate identification.
ETERNITY PE3SS
75W
ETERNITY PE3SP
75W
ETERNITY PE6SP
75W
The Power Supply Unit is factory-fitted. It must be removed and refitted by trained technicians only, and
only for the purpose of fault repair or replacement.
427
The CPU
The CPU of ETERNITY PE manages the entire system, controls all other cards (SLT, DKP, CO, DKP+SLT,
CO+SLT, DKP+CO, BRI, T1E1PRI Single, GSM, VoIP, etc.). All configuration and programming information is
stored on this card.
The CPU is factory-fitted on the bottom plane of the ETERNITY PE.
The CPU may be removed and reinstalled solely for the purpose of fault repair or replacement, and by
trained technicians only.
Connectors
The connectors of the universal slots of ETERNITY PE are located on the CPU. So, the cards of ETERNITY PE
must be mounted on the connectors on the CPU.
The CPU card has an Ethernet Port, a USB Port, a Communication Port, an Analog Input Port (AIP) and an Analog
Output Port (AOP), and a USB Port.
USB Port
The USB port functions as the Device port for downloading data on to the ETERNITY PE using a PC.
The USB port is not available on the ETERNITY PE3SS.
Communication Port
This is a single asynchronous, serial, full duplex RS232C communication port, labeled as COM 10101. The COM
Port has a DB-9 connector. The COM port is meant for connecting a PC to the ETERNITY PE to install and operate
Property Management Software (PMS), Call Accounting Software (CAS), and download Station Message
Recording (SMDR) reports and System reports, and Hotel Reports.
428
Connector
Function
RJ45
DB-9 female
USB
LED
The CPU has two dual color - Green and Red - LEDs labeled L1 and L2.
L1 - indicates the health of the card during the normal functioning of the system.
L2 - indicates the health of the card during the reset cycle. After power ON, when the system becomes
stable LED blinks Green for 1 second ON and OFF.
Jumpers
Jumper J9 on the CPU card of ETERNITY PE3SS/PE3SP and J13/J7 on the CPU card of ETERNITY PE6SP are
used to Reset the SE Password. Refer the table below:
Jumper Number
Position
Function
J11
AB (default)
BC
External Boot.
Internal Boot.
J13*
J7 #
AB
BC (default)
Reset SE Password.
Normal.
ETERNITY PE3SS
ETERNITY PE3SP
ETERNITY PE6SP
J5
J5
J9
J9
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Do not change the position of Jumper number J5 on ETERNITY PE3SS and PE3SP.
Do not change the position of the Jumper J11 on ETERNITY PE6SP.
* For PCB V1R3 (P-138-001-01-03).
# For PCB V1R5 (P-138-001-01-05).
ETERNITY PE does not support Redundancy and Hot Swap.
access the web-based programming tool Jeeves from any PC on the LAN.
set up and run software applications such as PMS and CAS on any PC on the LAN.
generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
When you connect the ETERNITY PE to a LAN PC, you need to make sure that
The IP Address of the Master Ethernet Port of the ETERNITY PE and the Ethernet Port of the PC are
not the same.
The Ethernet Port of ETERNITY PE and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port at the end of this topic.
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2. Connect the Communication Port of ETERNITY with the Communication Port of the stand-alone PC using
When you connect the ETERNITY PE to a standalone PC, you need to make sure that
The IP Address of the Master Ethernet Port of the ETERNITY PE and the Ethernet Port of the PC are
not the same.
The Ethernet Port of ETERNITY PE and the Ethernet Port of the PC are in the same Subnet.
For instructions to change the IP address and Subnet Mask, refer Changing IP Address and Subnet
Mask of the Master Ethernet Port at the end of this topic.
Change the IP Address and the Subnet Mask of the Ethernet Port by dialing the following commands from
an extension of the ETERNITY PE.
Dial 1#91-1234 (to enter programming mode. 1234 is the default SE Password)
You get programming tone.
To change IP Address:
95.
96.
97.
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If there is a DHCP server on the LAN to which the Ethernet Port of the ETERNITY PE is connected,
there is no need to change the IP Address or Subnet Mask, as these will be provided automatically by
the DHCP server.
You must only enable the DHCP flag of the Ethernet Port of ETERNITY PE.
Refer the Technical Specifications of the Analog Output Port (AOP) and select a compatible Public
Address System device.
Connect a good quality external amplifier and matching speakers to the port. You may use a combination
of 10W amplifier and a 4W speaker.
Technical Specifications of the AOP
Specification
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Maximum Voltage
Physical Connector
Refer the Technical Specifications of the Analog Input Port and select a compatible external music
source - Cassette Player, FM Radio, CD Player, etc.
Analog Input Port Technical Specifications
432
Specification
Value
Interface Type
Audio Signal
Frequency
300Hz to 3400Hz
Specification
Value
Maximum Voltage
DC Bias
Isolation
Internal Transformer
600
Termination provided
600
Physical Connector
Plug in the audio jack of the device into the AIP connector.
Also refer the topics Music on Hold (MOH), Background Music (BGM), External Music.
The volume of the external music source must be set to a level such that the music on the trunks is neither
very low nor very high. The volume of the signal coming from this device must never increase beyond the
specified limits - 0.707Vrms across 600.
Do not apply electrical signal of higher volume than the specified limit to this port, as it may cause
permanent damage to the system. Matrix Warranty does not cover damages as a result of improper use.
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ETERNITY PE Card
DKP2+SLT6
Combination card, with 2 ports to connect to 2 Digital Key Phones and 6 Single
Line Telephones
ETERNITY PE Card
CO4+SLT4
Combination card, with 4 ports to connect to 4 Two-wire Analog trunk lines and 4
Single Line Telephones
ETERNITY PE Card
CO2+DKP2+SLT4
Combination card, with 2 ports to connect to 2 Two-wire Analog trunk lines, 2 Digital
Key Phones, and 4 Single Line Telephones
ETERNITY PE Card
CO2+SLT6
Combination card with 2 ports to connect 2 Two-wire Analog trunk lines, and 6
ports to connect 6 Single Line Telephones
Choose an SLT Card with the configuration that meets your requirement for SLT ports. Also consider the maximum
SLT Port capacity of the system you are installing.
The maximum number of SLT ports supported by the variants of ETERNITY PE are:
Connectors
The SLT Cards have RJ45 connectors. A multi-pair, cable is supplied for each connector on the card.
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
434
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require you
to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of the
phone to dial Flash.
2. Unpack the SLT card and check the package contents.
3. Make sure that the power supply is switched off, before you begin the installation of the card. Always wear
seated such that its connector pins make perfect contact with those on the CPU (motherboard) on the
bottom plane.
7. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
8. Repeat the same steps to install another SLT card. It is not necessary to install the other SLT cards in
subsequent slots. You may install the other SLT cards in any of the universal slots.
9. Now, use the cables supplied with the SLT card to connect the SLT wires with the Main Distribution Frame.
For each connector on the SLT Card, there is a separate cable with an RJ45 plug on one end and free at
the other end.
Refer to the pinout details of the connectors on each card type to connect the wire-pairs.
Color
RJ45-1
01
02
03
04
RJ45-2
RJ45-3
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10. Plug in the RJ45 end of the SLT cables into the respective connectors.
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11. Lead the cables out of the enclosure through any of the two cable outlets on either side of the enclosure.
12. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging
in the phone cables into the RJ45 connectors on the card.
When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the SLT will be
connected to the first port on the connector.
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ETERNITY PE Card
DKP2+SLT6
Combination card, with 2 ports to connect to 2 Digital Key Phones and 6 Single
Line Telephones
ETERNITY PE Card
CO2+DKP2+SLT4
To connect the proprietary digital key phones with ETERNITY, you must have at least one of the above mentioned
DKP Cards installed in the system.
Select a DKP Card with the configuration that meets your requirement for DKP Ports. Also consider the maximum
DKP Port capacity of the system you are installing.
The maximum number of DKP ports supported by each variant of ETERNITY PE is:
supply is switched off and you are wearing an antistatic-wrist strap/belt and have a grounding mat.
2. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket, keep for future use to cover empty slots.
3. Insert the DKP card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
98.
438
4. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
Refer the connector pin details for each DKP Card type given in the following.
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Plug in the RJ45 end of the MDF cables provided with the DKP card into the respective connectors.
Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the ETERNITY ME DKP Port must be terminated to the bottom of the Krone
Connector, while the wire-pair of the extension line to be connected to this port must be terminated on
the top of the Krone connector. Refer the topic The Main Distribution Frame (MDF) for illustration.
6. Connect the Digital Key Phones to the wall jacks at their respective locations. Detailed installations
Installing EON48
1. Unpack the box and verify the package contents99.
2. Mount the phone on a desk or wall at a convenient location.
99.
440
3. To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
Foot Stand
DND
Redial Release
Hold
abc
3 def
4 ghi
jkl
6 mno
tuv
9 wxyz
7 pqrs 8
CA 3
Keyhole
Slot 2
Line
4P4C Spring
Cord
Press
down
to detach
Foot Stand
Press down
to detach
Foot Stand
Names
CA 4
Keyhole
Slot 1
Headset
Port
CA 2
CA 1
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
5. Connect the handset of the EON48 to the phone body using the spring cord.
441
6. To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
Headset
You may also plug in a stereo headset with an RJ12 connector into the headset port at the bottom of the
phone, marked with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
7. Plug one end of the RJ45 cable supplied with the phone into the RJ45 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
442
8. When the ETERNITY is powered ON, the EON will get reset. The EON communicates with the ETERNITY.
The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the
message 'Please wait' appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
9. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
202 Reception
M on 2 4 A U G 1 2 : 0 0
10. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-
Operation.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
Installing EON310100
1. Unpack the box and verify the package contents101.
2. Mount the phone on a desk or on the wall at a convenient location.
443
3. To mount EON310 on a wall, detach the Foot Stand on the bottom of the phone. Refer to the illustrations in
EON48.
CA 1
CA 2
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON310. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON310 on a desk, you can attach the Foot Stand in two ways - 35 and 55 degree
You may also plug in a stereo headset with an RJ12 connector into the headset port marked with the
symbol
, on the left side panel of the phone as illustrated in the figure below.
Headset
Casio Jack
Headset
(R J12 Connector)
7. Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
444
8. When the ETERNITY is powered ON, the EON will get reset and the message 'Welcome to Matrix.
9. The EON communicates with the ETERNITY. The handshaking lasts for 5-6 seconds. The EON model,
version and revision number, along with the message Please Waitappears on the LCD display.
10. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-Operation.
445
You can install two DSS consoles to a DKP. Refer Direct Station Selection Console for possible
combinations for installing the models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated on the bottom of
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
Windows 98
Windows XP
Windows NT
Windows 2003
Windows Vista
Windows 2007
446
2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
9. Click the Matrix EONSOFT Icon to execute installation of the program.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
447
11. Click the shortcut to open the program. The EONSOFT window will open:
12. Click Options at the top left of the window. A drop down menu will appear.
448
14. Select the COM Port to which the communication cable is connected.
449
15. EONSOFT is now connected. If you have already configured the DKP parameters like Access Code and
Name for the port to which EONSOFT is connected, these will appear.
If this window does not appear after you have selected the COM Port Option, test the COM Port for
data transfer.
If the wrong COM port has been selected, a dialog box will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
450
From the drop down menu of 'Options', select the 'COM Port' to which you have connected the
communication cable.
Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
Click the button labeled 'Start Test' in the COM Port Settings dialog box.
After clicking this button, observe the Test Result section in the dialog box.
The Error Count value shows zero as value, if both the communication cable and the COM port are
working.
The above screen shows that the COM Port/communication cable is working.
If the Error Count shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
Remove the communication cable from the COM Port of the PC.
Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
Now, if the error count is zero, please check the Communication Cable.
If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
451
The CO Card
The CO Card provides the interface to connect the ETERNITY with the Two-Wire Analog Trunk lines from the CO
Network. The CO Card supports the different standards and features of CO Networks across the world.
The CO Card is available in the following configurations for the variants of ETERNITY PE. CO interface is also
available in combination with SLT ports on a single card.
8-port card to connect 8 Two-wire analog Trunk lines from the CO network
ETERNITY PE Card
CO4+SLT4
Combination card, with 4 CO ports to connect 4 Two-wire analog Trunk lines from
the CO network, and 4 SLT ports to connect 4 Single Line Telephones
ETERNITY PE Card
CO2+DKP2+SLT4
Combination card, with 2 ports to connect to 2 Two-wire Analog trunk lines, 2 ports
to connect 2 DKP/DSS Consoles, and 4 ports to connect 4 Single Line Telephones
ETERNITY PE Card
CO2+SLT6
Combination card with 2 ports to connect 2 Two-wire Analog trunk lines, and 6
ports to connect 6 Single Line Telephones
Choose a CO Card with the configuration that meets your requirement for CO trunk ports, keeping in mind the
maximum CO Trunk Port capacity of the system you are installing.
Connectors
The CO Card has RJ45 connectors, with 4 CO ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
that power supply is turned off, and the power cord is unplugged before you begin the installation of the
card. Put on an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
2. Unpack the CO card and check the package contents.
3. Unscrew and remove the top cover of the ETERNITY PE, and keep it aside with the screws.
4. Select any of the free slots from the universal slots.
5. Seat the card onto the connectors of the selected slot. The connector pins of the card should make perfect
452
7. Repeat the same steps to install another CO card. You may install the other CO cards in any of the
For each connector on the CO Card, there is a separate cable with an RJ45 plug on one end and free at
the other end.
9. Plug in the RJ45 end of the CO cables into the respective connectors. Refer to the pinouts of the
453
10. Lead the cables out of the enclosure through any of the two cable outlets on either side of the enclosure.
11. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
454
The maximum number of BRI lines supported by each variant of ETERNITY PE are:
Connectors
The BRI card has 2 RJ45 Connectors. A separate cable is supplied for each connector.
ISDN
Network
NT 1
BRI Port
ETERNITY
Power
U-Interface
(2-wire)
S/T
Interface
Customer Premises
Where,
U Interface = between the NT1 equipment and the ISDN central office.
S/T Interface = between the ISDN user equipment, in this case, ETERNITY and the Network Interface
Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the ETERNITY.
455
TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY via
the NT1 device. Here, the ETERNITY is the Terminal Equipment, so the BRI Port must be programmed to work in
the TE mode.
When an ISDN Phone is to be connected to the BRI port of ETERNITY, the BRI port must be programmed to work
in NT mode.
When a BRI port of another ISDN PBX is to be connected to the BRI port of the ETERNITY, in such a configuration,
you may configure
the BRI port of the other ISDN PBX in the TE mode and the BRI Port of the ETERNITY in the NT mode.
OR
the BRI port of the other ISDN PBX in the NT mode and the BRI Port of the ETERNITY in the TE mode
Point-to-Point Configuration
ISDN
Network
NT
BRI Line
BRI Port
(TE Mode)
(UP to 1 Km.)
ETERNITY
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY, can be up to 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
Further, two configurations are possible in a Point-to-Multipoint configuration:
a. Short Passive Bus Configuration
b. Extended Passive Bus Configuration
456
NT
BRI Port
(TE Mode)
Terminal
Resistance 100
ETERNITY
ISDN Phone
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
Terminal 8
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
In a Short Passive Bus Configuration,
A maximum of 8 Terminal equipment or ISDN devices can be connected to a single NT on a bus up to 200
meters from the NT.
100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access a BRI line and can make outgoing
calls. At the same time, another subscriber from ETERNITY or any ISDN phone shown in the figure can
make outgoing call from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.
d1 < 30 meters
NT
Terminal
Resistance 100
Terminal
Resistance 100
BRI Port
(TE Mode)
ETERNITY
ISDN Phone
ISDN Phone
Terminal 1
Terminal 2
Terminal 3
457
Where,
TE = Terminal equipment of any ISDN Equipment
NT = Network Termination provided by Service Provider
TR = Terminal Resistance 100
d = distance from NT to the last TE Equipment
d1 = the total distance from first TE equipment and the last TE equipment.
In an Extended Passive Bus Configuration,
You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
shown in the figure.
Using this configuration, any subscriber from ETERNITY can access the BRI line and make outgoing calls.
At the same time, another subscriber from the ETERNITY or any ISDN phone shown in the figure can
make outgoing calls from the same BRI. In the same way, incoming calls are possible on the same BRI.
Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
power supply, always wear an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
2. Unpack the BRI card and check the package contents.
3. Unscrew and remove the top cover of the ETERNITY PE, and keep it aside with the screws.
4. Select any of the free slots from the universal slots.
5. Seat the card onto the connectors of the selected slot. The connector pins of the card should make perfect
To connect ISDN phones, an ISDN PBX or any ISDN equipment, the BRI Port must be configured in the
NT mode.
By default, BRI Ports are configured in the TE mode.
To set Orientation Type of the BRI Port, under Configuration, open the BRI Configuration link, and
under BRI Parameters, set the Orientation Type.
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
Last TE equipment
Last point of the bus bar where the last TE equipment is connected.
When BRI port is configured in the NT mode.
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when:
459
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
BRI Port 2
Jumper Position
Jumper Position
J5
J6
J4
J12
AB
AB
AB
AB
BC
BC
BC
BC
Function
By default, Termination Resistance of 100 is set on the BRI port (the Jumpers are in AB position).
Tx 3
Rx 4
Rx 5
Tx 6
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
100
100
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
460
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Enable Feed Power on the BRI Port. For instructions see Power Feed under Configuring BRI Trunks.
By default, the Jumpers are set in AB position to feed power through Tx and Rx wires (Phantom
Power). If you want to feed power through a separate pair of wires, you may change the position of the
Jumpers on the BRI module as mentioned in the table below.
BRI Port 1
BRI Port 2
Jumper Position
Jumper Position
J3
J2
J11
J10
AB
AB
AB
AB
BC
BC
BC
BC
Function
The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
13. Use the straight cables supplied for each connector on the BRI card to connect the BRI Ports to the NT1
device supplied by your ISDN service provider. Refer the configuration and pinout details given below for
guidance.
Configuration details of the U interface (RJ-45) at NT1
Pin Number
Pin Details
Tx
Rx
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Pin Details
Rx1
Tx1
Tx2
Rx2
Connection
Color
NC
Orange-White
NC
Orange
Tx0A
Green-White
Rx0A
Blue
Rx0B
Blue-White
Tx0B
Green
VOut 0-
Brown-White
VOut 0+
Brown
Connection
Color
NC
Orange-White
NC
Orange
Tx1A
Green-White
Rx1A
Blue
Rx1B
Blue-White
Tx1B
Green
VOut 1-
Brown-White
VOut 1+
Brown
Port 2:
Pin
462
Connection
Color
NC
Orange-White
Pin
Connection
Color
NC
Orange
Rx0A
Green-White
Tx0A
Blue
Tx0B
Blue-White
Rx0B
Green
VOut 0-
Brown-White
VOut 0+
Brown
Connection
Color
NC
Orange-White
NC
Orange
Rx1A
Green-White
Tx1A
Blue
Tx1B
Blue-White
Rx1B
Green
VOut 1-
Brown-White
VOut 1+
Brown
Port 2:
Pin
The following diagram shows how to connect a BRI Line to the ETERNITY PE BRI port in the TE mode.
ETERNITY
NT-1
4
ISDN
Network
Rx1
TxA
Tx
Tx1
Rx
Tx2 5
6
Rx2
RxA
RxB
TxB
Power
U-Interface
(2-wire)
NC
V-
NC
V+
S/T Interface
(4-wire)
A BRI port of
ETERNITY
14. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle.
By default the Orientation Type of the BRI ports of ETERNITY are set as 'Terminals' (TE mode). So, you
may skip to the next step.
463
If the BRI Port is to be configured in the NT mode, all the related Jumpers should be set in BC position.
Refer the table below.
Jumper Position for BRI Port1
J1
J2
J4
J5
J7
J8
J10
J11
NT
BC
BC
BC
BC
BC
BC
BC
BC
TE
AB
AB
AB
AB
AB
AB
AB
AB
Mode
When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.
ISDN
Network
BRI Line
NT
BRI TE
BRI TE
BRI TE
Other ISDN
Equipment
Other ISDN
Equipment
ETERNITY
Last TE equipment
Last point of the bus bar where the last TE equipment is connected.
When BRI port is configured in the NT mode.
If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when:
464
Termination need not be inserted if the BRI port of ETERNITY (configured in TE mode) is connected as
any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Function
J3
J4
J3
J4
AB
AB
AB
AB
BC
BC
BC
BC
By default, Termination Resistance of 100 is set on the BRI port (Jumpers J3 and J4 are in AB position)
Tx 3
Rx 4
Rx 5
Tx 6
RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
100
100
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
465
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
1
RJ45 Connector
ports on BRI Bus
Bar to which the
ISDN TE
Equipment is
connected
The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Function
Jumper Position
J1
J2
AB
AB
BC
BC
Open
Open
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The maximum power that can be fed to a single BRI port is 50mA.
From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
20. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle.
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PRI
Robbed Bit Signaling
Q-Signaling (QSIG)
E&M
When connected to E1 carrier lines, the card supports the following signaling types:
PRI
Channel Associated Signaling (CAS)
Q-Signaling (QSIG)
E&M
The T1E1PRI Card is available in the following configuration for ETERNITY PE:
1-port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Compatible Device
The maximum number of ISDN PRI lines supported by the variants of ETERNITY PE are:
Connectors
The T1E1PRI card has an RJ45 Connector. A cable with RJ45 plugs on both ends is supplied with the card.
an electrostatic-discharge preventive wrist strap and use a grounding mat. Make sure the power supply is
turned off.
2. Unpack the T1E1PRI card and check the package contents.
3. Unscrew and remove the top cover of the ETERNITY PE (if not opened already). Keep the cover and the
screws aside.
4. Select any free slot from the Universal Slots.
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5. Grasp the card by its sides or corners. Fit the card's connectors into the connectors of the selected slot.
Ensure that the card's connector pins make perfect contact with those on the CPU on the bottom plane.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Repeat the same steps to install another T1E1PRI card. You may install the other T1E1PRI cards in any of
the universal slots, but not necessarily in a sequence. Any card can be inserted in any of the universal
slots.
interface equipment (modem), which is usually supplied by your ISDN Service Provider along with the PRI
line.
The block diagram illustrates this.
Customer Premises
ISDN
Network
G.703
Modem
4-wire
HDSL
(RJ-45 Connector)
DTE
(RJ-45 Connector)
4-wire
PRI
Port
G.703
Modem
Power
Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI line, one at
the Local Exchange and other at the Customer's Premises.
At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
The DTE interface of the modem is to be connected to the PRI port (RJ45 connector on the Matrix
ETERNITY PE T1E1PRI Single Card).
9. Plug in one end of the RJ45 cable supplied with the card into the card's connector. Lead the cable out of
the enclosure through any of the two cable outlets on either side of the enclosure.
10. Plug the other end of the RJ45 cable into the Network Termination Unit.
11. Refer the following pin details for connecting the Network Termination Unit with the ETERNITY.
Pin details of HDSL Interface of the G.703 Modem. (HDSL Network Termination Unit)
Pin Number
Pin Details
Line A
Line A
Not used
Line B
469
Pin Number
Pin Details
Line B
Not used
Not used
Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
Pin Number
Pin Details
TX1 (Tip)
TX2 (Ring)
Not used
RX1 (Ring)
RX2 (Tip)
Not used
Not used
Not used
Most of the HDSL Network Termination Unit manufacturers use these connectors. But you are advised to
read the installation guide of the HDSL Network Termination Unit being used by you.
Pin details of ETERNITY T1E1PR1 Port
The T1E1PRI Port of the ETERNITY PE terminates in an 8-pin RJ45, female connector and is wired
according to the figure below.
4
3
2
1
Tx1 (Ring)
Tx2 (Tip)
NC
NC
Rx2 (Tip)
NC
Rx1 (Ring)
NC
5
6
7
8
The cable wires may have to be crossed depending on the pinout of the DTE Interface of the modem.
470
Position
Meaning
J2
BC
J2
AB
To use the PRI Port for T1 connectivity, termination resistance must be changed to 100, by changing
the position of jumper J2 to AB position.
13. Repeat the same steps to connect another PRI card, if installed.
14. If you have no other card to install, replace the top cover, by sliding it in place. Secure the cover with the
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Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
The maximum number of Mobile ports trunks supported by each variant of ETERNITY PE are:
Antenna
For all four mobile ports, there is a single antenna with a male connector on the card. A splitter connects all the four
ports on the card into a single antenna. An antenna cable is provided.
472
you are wearing an electrostatic discharge preventive wrist strap and have a grounding mat, before you
begin handling the card.
2. Get the SIM Card from the GSM service provider of your choice ready. Use SIM PIN protection, if required.
change the SIM PIN to 1234 (this is the default PIN for all SIM cards used in the system). Changing the
SIM PIN to '1234' enables you to change the SIM PIN from the ETERNITY later.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
5. Now, insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the
Mobile card. You can insert multiple SIM cards of the same GSM service provider or of different service
providers.
473
6. Remove the top cover of the ETERNITY PE, if not opened already. Keep the cover and the screws aside.
7. Select any of the free universal slots. Grasping the card by its sides or corners fit it onto the connectors of
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At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
Each time the Mobile Port sends a request, such as a Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the GSM network. It is fixed for 150 seconds for all Mobile ports.
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The VoIP card is available in the following configuration for the ETERNITY PE.
ETERNITY PE VoIP16
ETERNITY PE VoIP8
Voice Channels
There are 16 Voice Channels on the VoIP16 Card and 8 Voice Channels on the VoIP8 Card, allowing as many
simultaneous calls to be made (using SIP Trunks and/or Extensions) as the number of Voice Channels supported
by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
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SIP Trunks
The ETERNITY PE supports a maximum of 4 SIP Trunks. It is possible to program all 4 SIP trunks on a single VoIP
Card or program them in a distributed manner, where more than one VoIP card is installed in the system.
SIP Extensions
ETERNITY PE supports 50 SIP Extensions. All the SIP Extensions can be registered with a single VoIP Card.
Any SIP-enabled device like an IP-phone, a Softphone, Analog Phone Adapter, can be registered with the VoIP
Card and function as the 'SIP Extension' of the ETERNITY PE.
The SIP Extensions function like extensions of the ETERNITY PE. SIP Extension users can make and receive calls
from and to other extensions of ETERNITYPE and external numbers over PSTN, GSM, VoIP and E&M lines104.
You can also connect the Standard and Extended IP Phones offered by Matrix as SIP Extensions.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic License Management.
A SIP Extension can be registered with the ETERNITY from three different locations. This helps organizations
overcome geographical distances and reduce call costs.
104. Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your country prohibit call traffic between the public telephony networks and IP networks, you must configure Logical Partition in your system.
To know more, see Logical Partition.
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A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
SIP ID/User ID
Authentication User ID
Authentication Password
SIP Registrar Server Address
SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
Network Information:
105. Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
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The card is located behind the NAT Router and Private IP is assigned to the WAN port.
When connecting the card in a Private Network, you would require the following information:
IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
Gateway Address
DNS Address
DNS Domain Name (if applicable)
VoIP Card connected to the Public Network for Matrix Extended IP Phones
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Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
Get these items/information ready before you install the VoIP card and connect it to the IP network.
1. Observe all prescribed safety precautions when inserting or removing cards. Make sure the Power Supply
is switched off, and you are wearing an antistatic wrist strap/belt and have a grounding mat.
2. Unpack the VoIP card and verify the package contents.
3. Unscrew the top cover of the ETERNITY PE and slide it out. Keep the cover and the screws aside.
4. Select any of the free slots from the universal slots.
5. Grasping the card by its sides or corners fit it onto the connectors of the selected slot. The card should be
seated such that its connector pins make perfect contact with those on the CPU (motherboard) on the
bottom plane.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Using the Ethernet cable supplied with the VoIP card, connect the LAN and the WAN Port to the IP
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP card and
the other end into the Router/Modem.
Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP Card and
the other end of the cable into the Router/Modem.
Connect the LAN Port of the VoIP Card to the LAN Switch/Hub.
8. To insert and connect another VoIP card, repeat the same steps as described above.
9. If you have completed all installation tasks, replace the top cover by sliding it in place. Secure the cover
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SIP Extensions
ETERNITY PE supports up to 50 SIP Extensions. The SIP Extensions function like DKP/SLT extensions of the
ETERNITY PE. SIP Extension users can make and receive calls to any extension user of the ETERNITY and to
external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP106.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY PE.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY PE and you must have the IP8
License. For more information on Licensing, see License Management.
You can register upto 50 SIP Extensions with a single VoIP Card of ETERNITY PE. However, at a time, only as
many extensions as the number of Voice Channels supported by the VoIP Card can make calls. For more
information, see Voice Channels under the description for theThe VoIP Card for ETERNITY PE.
You can register the same SIP Extension from three different locations.
You may also connect the Standard and Extended IP Phones of Matrix.
The Matrix Extended IP Phone, SPARSH VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Screening
To connect SPARSH VP248 with ETERNITY, see Connecting SPARSH VP248 as Extended SIP Extension.
SPARSH VP330 is proprietary Extended IP Phones with graphical touch-screen user interface. This feature-rich
SIP based phone support most features and functions of the proprietary digital key phones of ETERNITY except
the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Supervision
Login Station from DISA
You cannot program SIP Extension from Enterprise or Hotel Wizard.
To connect SPARSH VP330 with ETERNITY, see Connecting SPARSH VP330 as Extended SIP Extension.
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
106. Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
Logical Partition.
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The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
If the ETERNITY PE Master Ethernet Port and the VoIP Card are connected to a Public Network,
Connect SPARSH VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
the Registrar Server Address of ETERNITY PE
the Registrar Server Port
the SIP ID
Authentication ID and Password.
481
If the ETERNITY PE Master Ethernet Port and VoIP Card are connected to a Private Network (Behind the NAT),
Connect SPARSH VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
use DHCP option 224 and Data Type as String to provide Server Address to the Extended IP
Phones.
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Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY PE in the DHCP option.
Log in to Jeeves. For instructions, read the topic Configuring using Web-based GUI: Jeeves under
Configuring ETERNITY.
Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see Configuring Matrix SPARSH VP248 - Extended IP Phone.
For the SIP extension number you assigned to the Extended IP Phone, go to the Location settings of the
extension, and do the following:
For instructions, see the topic Configuring Matrix SPARSH VP248 - Extended IP Phone.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SPARSH VP248. For the purpose of illustration, the premium model, SPARSH VP248P, has been used.
1. Unpack the SPARSH VP248 box and verify package contents.
2. Mount the phone on a desk at a location convenient to you.
When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
Foot Stand attached at 30 Angle
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your
desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
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Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack headset jack with the symbol
Headset
OR
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You may plug a headset with an RJ12 connector into the headset port at the bottom of the phone, marked
with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
5. Connect the LAN Port of SPARSH VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone107. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
8. Plug the Power Adapter into a power outlet.
9. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
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The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
As soon as the Loading... message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
When the cursor is placed under the Extended IP Phone, press Enter key.
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The phone will start loading the Extended IP Phone Firmware. It will display current firmware being loaded.
We l c o me to Ma t ri x
L oa din g V 0 5 R0 1 Ex t SI P
After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See Network Settingsat the end of
this topic.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY PE.
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As the phone downloads the configuration files, the file names will appear one by one.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
On successful download of all configuration files, the phone attempts to register with ETERNITY PE.
On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The Up key
488
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to Local Menu.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
DN D
Names
Local Menu
CA04
CA03
Redial Release
abc
Hold
3 def
4 ghi
jkl
6 mno
7 pqrs
tuv
9 wxyz
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
You can configure Network Parameters and view Network status from the Local Menu.
489
In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
You can configure all network parameters described below, except the MAC Address.
Connection Type
Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
Enter the desired Static IP Address by pressing the digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Subnet Mask
If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Gateway Address
If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
DNS Server
490
If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key 1 twice.
If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
ETERNITY PE CPU Card works as the Auto Configuration Server for the phone. Enter the IP Address or
the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default: blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as String to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
Valid range of the port is: 80 or 102465535. Default: 80.
VLAN Setting
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
491
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic108.
The meaning of CoS bits with respect to traffic type is as follows:
CoS
Traffic Type
Best Effort
Background
Spare
Excellent Effort
Controlled Load
Video
Voice
Network Control
Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
PCAP
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see Using PCAP Trace for Matrix Extended IP Phone, under PCAP Trace.
When you change the Network Settings, the phone will restart.
108.The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), that is, better handling of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred
to as Class of Service (CoS) which is defined by IEEE 802.1P.
492
In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
MAC:
IP :
MASK:
G W:
DNS:
N E T W O R K S TAT U S
0 0:1 b:0 9:0 0:9 a:a 7
1 92. 16 8. 2 0 1 .2 0 5
2 5 5 . 25 5 . 2 5 5 .0
1 9 2 . 16 8 . 2 0 1 .3
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY
PE.
S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY PE.
Decide the location where you want to place SPARSH VP330 within your LAN.
To use the DHCP Server on your LAN for assigning IP Address to SPARSH VP330, make sure you do the
following:
Use DHCP option 224 and Data Type as String to provide Master Ethernet Port Address to SPARSH
VP330.
Program the IP Address or the Domain Name of the Master Ethernet Port of ETERNITY ME in the
DHCP option 224.
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Log in to Jeeves. For instructions, read the topic Configuring using Web-based GUI: Jeeves under
Configuring ETERNITY.
You must configure the necessary parameters in ETERNITY so that SPARSH VP330 can register as a SIP
Extension. For instructions, see Configuring Matrix SPARSH VP330.
When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
3. When you mount the phone on a desk, you can attach the Foot Stand in two ways at 30 Angle or at 50
Angle.
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
Plug the long straightened end of the phone cord into the handset jack on the left side panel of the
phone marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
5. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 3.5 mm single
connector into the headset jack headset jack with the symbol
OR
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You may plug a headset with an RJ12 connector into the headset port on the side panel of the phone,
marked with the symbol
Headset
Casio Jack
Headset
(R J12 Connector)
6. Connect the LAN Port of SPARSH VP330 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
7. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port at the bottom of the phone to the LAN Port of the computer.
8. Plug the connector of the Power Adapter in to the power jack at the back of the phone109. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
9. Plug the Power Adapter into a power outlet.
10. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
The LCD display will light up and the following message will appear on it, as the phone boots:
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The Starting SPARSH VP330 ... message appears on the phone display, while loading the
application.
The Applying Network Parameters... message appears on the phone display, while the Static
Network parameters are being applied.
If you want to change the Network Settings or want to use Wi-Fi for connectivity, press Settings
496
To change the Network Settings of the phone and configure the network parameters.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY ME. The Configuring the phone... message appears on the phone.
On successful download of all configuration files, the phone attempts to register with ETERNITY ME. The
Registering the phone... message appears on the phone display.
497
The Updating firmware... message appears on the phone display, when the firmware is being updated.
After the firmware is updated, the phone will reboot. The Rebooting the phone... message appears on
the phone display.
The phone will register successfully, only if the SIP Extension parameters in ETERNITY have been
correctly configured as per your installation scenario.
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The card provides mailbox facility to all extensions of ETERNITY PE. Each Mailbox has the capacity of storing
15,000 messages. The maximum size of each Mailbox is 60,000 minutes. By default, the size of each Mailbox is
set to 5 minutes. The maximum Message Length for each Mailbox is 9999 seconds. By default, the Maximum
Message Length for each Mailbox is set to 15 seconds111.
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 4GB Pen Drive with the VMS
card. The Pen Drive supports 72 hours of recording. However, you may use a Pen Drive of upto 32GB.
The VMS Card has an Ethernet Port and a USB port.
Ethernet Port
The Ethernet Port is used to connect the VMS card to a computer (standalone or connected in a LAN) to access
and use the embedded FTP server for Software Upgrades, Backup of configuration files and Mailbox messages.
The Ethernet Port can also be used for VMS Debug.
USB Port
The USB port is an internal port, located on the main board of the card. The USB port is used for connecting the
USB Pen Drive to the VMS Card. The pen drive is supplied by Matrix and contains all the voice messages, mailbox
messages, greetings and other messages and prompts.
The 4GB Pen Drive is factory fitted and shipped with the card. However, you may use a Pen Drive of upto 32GB.
For instructions see Replacing the Pen Drive at the end of this topic.
power supply to the ETERNITY is turned off, and the power cord is unplugged.
2. Unpack the VMS card and check the package contents with the packing list.
3. Remove the top cover of the ETERNITY, if not opened already. Keep the cover and the screws aside.
110. The VMS Card with firmware V06R03 is supported in ETERNITY V10R10.
111. When the ETERNITY is installed in the Hospitality Application (Hotel Mode), the default Mailbox size would be 300 minutes and
the default length of messages is 999 seconds.
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4. Select from any of the free universal slots to install the VMS Card.
5. Hold the card by its sides and seat the card on the connectors of the slot on the CPU. Make sure that the
Connecting to a Computer
8. Now, connect the card to a standalone PC/LAN.
Plug in one end of the Ethernet cable supplied with the card into the Ethernet Port of the VMS Card.
Lead the Ethernet cable out of the enclosure through any of the two cable outlets on either side of the
enclosure.
Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a to a standalone/LAN PC, you need to make sure that
The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the PC do not conflict,
are not the same.
The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
To format the Pen Drive with FAT32, follow the steps given below:
500
Click My Computer.
Right-click the removable disk to which you have connected your Pen Drive, in this example Removable
Disk (F:).
501
502
The Format Removable Disk (F:) options appear on your screen. In File Format select FAT32.
You will get an alert: WARNING: Formatting will erase ALL data on this disk. To format the disk, click OK.
To quit, click CANCEL.
Click OK to format.
When the formatting process is complete, the message Format Complete will appear on your screen.
Now, copy the contents of the factory fitted Pen Drive onto the new Pen Drive.
Value
Speaker Output
Mic Input
Value
Relay Type
Power Relay
Contact Arrangement
1.5Amp; 120VAC/150VDC
Operation Time:
8ms (max.)
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The DIP can be used for certain Automated Control Applications and Security Alarm and Reporting. Read
these topics to know more.
Also refer the topic Digital Input Port (DIP).
Technical Specifications of the DIP
Specification
Value
13mA
Type
Open/Close Sensing
1000
seated such that its connector pins make perfect contact with those on the CPU (motherboard) on the
bottom plane.
6. Secure the card on the studs labeled H1, H2 and H3 with the three screws provided.
7. Now, plug in the cables supplied with the card into the connectors. Lead the cables out of the cable outlet
and terminate the free end of the cables into the Distribution Frame.
For each connector on the card, there is a separate cable with an RJ45 jack on one end and free at the
other end.
Refer to the pinout details of the ports on the Door Phone card to connect the wires appropriately.
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Pin
1
2
Port
DOP-1
3
RJ45-1
4
5
Door Phone-1
6
7
8
1
2
DIP-1
DOP-2
3
RJ45-2
4
5
Door Phone-2
Color
Relay A
Orange-White
Relay B
Orange
Status
Green-White
MIC Input
Blue
Speaker Output
Blue-White
Analog Ground
Green
DIP_B
Brown-White
DIP_A
Brown
Relay A
Orange-White
Relay B
Orange
Status
Green-White
MIC Input
Blue
Speaker Output
Blue-White
Analog Ground
Green
Not Connected
Not Connected
Brown-White
Not Connected
Not Connected
Brown
Relay A
Orange-White
Relay B
Orange
Status
Green-White
MIC Input
Blue
Speaker Output
Blue-White
Analog Ground
Green
1
2
DOP-3
3
RJ45-3
Connection
4
5
Door Phone-3
6
7
Not Connected
Not Connected
Brown-White
Not Connected
Not Connected
Brown
8. The wires of the devices - Door Phone, Door Lock, Sensor - you want to connect to the ETERNITY PE
should be terminated in to the Distribution Frame. Refer the pinout details of the ports of the Door Phone to
connect the cables.
Connecting Door Phones
There is a door phone port on each connector of the Door Phone Card. You can connect any standard
4-wire door phone. Refer to the pin out details given above to make the connections.
Make sure that the door phone you connect conforms with the Technical Specifications of the door
phone port.
505
If a Door Lock is to be used in conjunction with the Door Phone, connect the Door Lock to the Digital
Output Port (DOP) of the ETERNITY PE.
There is a Digital Output Port on each of the three connectors of the Door Phone Card. You can
connect a Door Lock or any other gadget you want to operate as an automated control application to
this port. Refer to the pin out details given above to make the connections.
Make sure that the gadget you connect conforms with the Technical Specifications of the DOP.
The Digital Input Port is located on the first connector of the Door Phone Card. Make sure that the
sensor of panic switch that you connect to the DIP conforms with the technical specifications of the DIP.
9. If you have completed all installation tasks, replace the top cover by sliding it in place. Secure the cover
506
Starting Up ETERNITY PE
Power ON
1. If you have completed all the installation tasks, connect the three-prong plug of the power cord from the
Reset Cycle
The LEDs L1 and L2 are turned on, and glow Orange for a minute.
The LEDs of the keys of the Digital Key Phones attached to the system are turned on sequentially.
When the Reset Cycle is successful, the default Extension Access Codes loaded by the system and the
date and time of the Real Time Clock of the system will appear on the LCD display of the Digital Key
Phones you have connected with the system.
507
508
CHAPTER 8
Installing ETERNITY
MEX12S113
The Matrix ETERNITY MEX12S is to be installed by persons trained and experienced in telecom
wiring.
The person installing the ETERNITY MEX12S must be familiar with trunks, physical wiring of the MDF
on both the exchange (PBX) side and the line side (CO).
When installing any equipment, make sure that you take all the necessary precautions for handling
electronic and electrical appliances. Follow proper procedures for static electricity, while handling the
system and its cards to prevent damage to the system and harm to yourself.
Use a grounding mat and wear an anti-static strap/belt. Read the dos and don'ts listed in 'Protecting
ETERNITY and Yourself.
If you have complied with the requirements and instructions described in Before You Start, you may
now begin the installation of your ETERNITY MEX12S.
The Matrix ETERNITY MEX12S is shipped factory fitted with the Power Supply card and the CPU card in their
respective fixed slots (refer the section Know Your ETERNITY).
The cards - BRU, E1FO, GSM, VoIP, DKP, CO, SLT, VMS, E&M, Radio and Data - are shipped separately as per
the order placed by individual customers. These cards can be installed in any of the Universal slots.
All the Cards of ETERNITY MEX12S support Hot Swapping. The Power Supply Card and the CPU Card also
support Redundancy.
Illustrated below is the position of the fixed and universal slots in ETERNITY MEX12S.
509
SLOT#12
SLOT # 01
CPU Card
CPU Card
12 Universal Slots
The first four slots from the extreme left are reserved for the Power Supply Card and the CPU card respectively.
Follow the installation instructions for cards described here, when you expand the system (add more cards) or
remove or swap cards for maintenance and repair.
1. Unpack the box. Check the package contents (see Packing List). Contact your Dealer/Distributor if any of
the items is missing, faulty or damaged. Do not discard the packaging material.
dimensions and weight of the model you have. If mounting the system on a wall, you may refer the
mechanical dimensions and the Mounting Template for drilling holes at appropriate places on the wall.
3. When installing the system in a rack, allow adequate space between the system and other units for air
circulation.
4. Mount the system at the selected site. Make sure that the system is placed in such a way that you have full
access to the front and back panels. The holes in the panels are provided for ventilation; Make sure that
these are not blocked, to prevent overheating.
per the specifications. Earth the system properly. (Refer How to Make the Telecom Earth)
Inserting Cards
7. Make sure that the ETERNITY MEX12S power is off and the power cord is unplugged.
8. Select a free slot from the universal slots.
9. Unscrew and remove the filler bracket that covers the card-slot opening of the slot you intend to use.
10. Hold the card with the connectors facing you. Do not grab the card from both ends.
510
11. Slide the card into the slot, along the guide rails provided for each slot at the top and bottom planes.
12. Ensure that the cards are inserted deep enough for all the connector pins on the cards make complete
13. When the card is firmly seated in the connector, push down the levers on the card mounting bracket and
Secure with
Screw
If you are removing the card permanently or for a certain period of time, install a filler bracket over the
empty card opening in the chassis.
Installing filler brackets over empty card-slot openings is necessary to protect the system from dust,
dirt, insects and damage.
17. Using the cables supplied with each card, terminate the cables in the Main Distribution Frame (SLT, DKP,
CO, E&M lines), the NT1 device (ISDN BRI lines), ISDN Modem (ISDN PRI Lines), as applicable.
Lead the cables neatly and tangle-free into the MDF, through the cable guides provided below the slots in
the enclosure.
511
18. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.
19. Close the enclosure cover, pressing down the snap lug as you push each part of the cover in its place.
512
If already installed, switch OFF power supply, unplug the power cord. Remove the screws securing the
card. Lift the levers on the mounting bracket to release the card. As the card emerges from the slot, ease it
out of the slot.
2. Insert the Power Supply card into the guide rails of the first slot on the extreme left, designated for the
Power Supply Card. Make sure that the card is inserted deep enough to make perfect contact with the
connectors on the motherboard at the backplane.
3. Now, press down the levers on the card mounting bracket to secure the card in its slot.
4. Secure the card in the slot by screwing the bracket on both ends.
To install a second PS48V card on the ETERNITY MEX12S for redundancy, insert the second card on the
next slot.
5. To the PS48V card, connect the Float cum Boost Charger (FCBC). Terminate the power cord from the
FCBC output into the 3-way termination block on the PS48V card.
Polarity is critical. Ensure that the wires are connected with the correct polarity. Follow the standard color
codes used by FCBC manufacturers:
Color
Signal
Red
+48VDC
Black
GND
Green
Earth
It is recommended that you measure the voltage before connecting the power cable to the power supply card.
Ensure that the earth is connected.
513
FCBC
Battery
Load
48VDC
48VDC Battery
FCBC
10A
41 to 56V
ETERNITY
MEX12S
Card PS48VDC
48V Battery
114. When the batteries are drained, the FCBC goes into the charge mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery voltage goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to charge mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the batteries are charged with higher current initially.
514
LEDs
There are eight Single colour (Red) LEDs on the Power Supply Card. The LED indication for each LED is
summarized in the following.
LED
Event
Color
Cadence
L1
+3.5V Indication
Red
Will glow
L2
Red
Will glow
L3
Power
Red
Will glow
L4
Red
Will glow
L5
+5V Indication
Red
Will glow
L6
Red
Will glow
L7
Red
Will glow
L8
Red
Will glow
Features
Redundancy
Hot-Swappable
Over-voltage and Under-voltage Protection
Reverse Input Protection
515
Connector
Function
Communication
(COM Port)
DB-9 female
Ethernet Port
RJ45
USB
USB
Push-type
Push-type
Operator Console
RJ45
Unused
LEDs
There are four dual color (Red and Green) LEDs on the CPU Card. The LED indication for each is summarized
below.
516
LED
Event
L1
ARM
L2
DSP
L3
ARM
L4
CM
LED Color
LED Cadence
Red
Green
Green
Green
Red
ON
Red
Initialization
Jumpers
Jumper J16 on the CPU Card is used to Reset the System Engineer Password.
Jumper Number
Position
Function
J12
AB (default)
External Boot
BC
Internal Boot
AB
BC (default)
Normal.
AB (default)
BC
Disable: Tx debug
COM Port transmit connected on ARM9 UART Port 0 transmit
AB (default)
BC
Disable: Rx debug
COM Port receive connected on ARM9 UART Port 0 receive
AB (default)
BC
UART Boot
J16
J10
J14
J1
Do not change the position of any of the Jumpers. Change the position of Jumper J16 only if you need to
reset the System Engineer password.
517
If the card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
2. Insert the CPU Card into the guide rails of the slot designated for the card.
Ensure that the card makes perfect contact with the connectors on the backplane of the motherboard.
Press down the levers on the mounting bracket to secure the card in its slot.
1 mA
7 mA
Use 0.5mm, non-stranded cables to connect the sensor device to the DIP.
Sensor
A DC contactor (60VDC max.) can be connected to the DOP. Any external relay based device can be
interfaced with the DOP via this DC contactor.
The DOP has a two-wire, push-in (spring clamp action) connector to attach the relay device.
Contact Arrangement
Operation Time
5 ms
Contactor 1-Phase/
3-Phase depending on load
AC Mains
R B Y N
Battery
24VDC
R B Y N
Load
519
When you connect the ETERNITY MEX12S to a standalone PC, you need to make sure that:
The IP Address of the Master Ethernet Port of the ETERNITY MEX12S and the Ethernet Port of the PC
do not conflict.
The Master Ethernet Port of ETERNITY MEX12S and the Ethernet Port of the PC are in the same
Subnet.
6. Connect the Communication Port of ETERNITY MEX12S with the Communication Port of the stand-alone
access the web-based programming tool Jeeves from any PC on the LAN.
set up and run software applications such as CAS on any PC on the LAN.
generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
When you connect the ETERNITY MEX12S to a LAN PC, you need to make sure that:
520
The IP Address of the Master Ethernet Port of the ETERNITY MEX12S and the Ethernet Port of the PC
do not conflict.
The Master Ethernet Port of ETERNITY MEX12S and the Ethernet Port of the PC are in the same
Subnet.
Features
521
LEDs
All SLT Cards have one dual colour LED. The LED indicates the Status of the Card.
LED Pattern
Stage
LED Color
LED Cadence
Red
Green
Red
Initialization
522
Stage
LED Color
LED Cadence
Green
Orange
Orange,
Green
Red
Green
Stand-By
Port Status
Position
Function
J2 and J3
AB
Normal Operation
BC
Debug
AB
Normal Operation
BC
Boot
BC
Normal
BC
Debug
J5
J8
Position
Function
J2 and J3
AB
Normal Operation
BC
Debug
AB
Normal Operation
BC
Boot
AB
Normal
BC
Debug
J5
J8
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
523
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require
you to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of
the phone to dial Flash.
2. Make sure that the necessary wiring for the SLT Extensions, from the wall jack to the MDF, is done.
3. Unpack the SLT card and check the package contents. It is recommended that you switch off the power
supply, before you begin the installation of the card. However, the SLT Card supports Hot Swap so you can
insert the SLT Card while the system is switched on. Always wear an electrostatic discharge prevention
wrist strap/belt and use a grounding mat.
4. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
6. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
Refer the illustrations below for pin out details of each connector.
524
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
21
Tip
Violet
46
Ring
Blue
Port 2
Port 3
Port 4
Port 5
Port 6
Port 7
Port 8
SLT
Port 9
Port 10
Port 11
Port 12
Port 13
Port 14
Port 15
Port 16
Port 17
525
Port Type
Port Number
Pin Number
Signalling
Wire Colour
Port 18
22
Tip
Violet
47
Ring
Orange
24
Tip
Violet
49
Ring
Brown
25
Tip
Violet
50
Ring
Slate
Port 19
SLT
Port 20
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
Port 2
Port 3
Port 4
Port 5
Port 6
SLT
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
Port 13
526
Port Type
SLT
Port Number
Pin Number
Signalling
Wire Colour
Port 14
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 15
Port 16
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
Port 2
Port 3
Port 4
SLT
Port 5
Port 6
Port 7
Port 8
CO
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Port 2
Port 3
527
Port Type
Port Number
Pin Number
Signalling
Wire Colour
Port 4
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 5
CO
Port 6
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
SLT
Port 13
Port 14
Port 15
Port 16
528
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
Port 3
Port 4
DKP
Port 5
Port 6
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
SLT
Port 13
Port 14
Port 15
Port 16
529
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
DKP
Port 3
Port 4
Port 5
Port 6
CO
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
SLT
Port 13
Port 14
Port 15
Port 16
8. Plug in the Centronic Male Connector of the cable supplied with the card into the connector on the SLT
Card. Refer to the pinout details of the connectors of each SLT Card type illustrated above.
530
9. Terminate the open end of the cables into the punch down blocks of the Krone modules designated for
For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging
in the phone cables into the RJ45 connectors on the card.
When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT card, the SLT will be
connected on the first port on the connector.
If you have completed all installation tasks, power on the system and observe the Reset Cycle.
531
ETERNITY MEX12S
DKP16
ETERNITY MEX12S
DKP8
Combination card, with 8-ports to connect to 8 Digital Key Phones and 8 ports to
connect 8 Single Line Telephones
LEDs
All DKP Cards have one dual colour LED. The LED indicates the Status of the Card.
LED Pattern
Stage
LED Color
LED Cadence
Red
Green
Red
Green
Orange
Initialization
532
Stage
LED Color
LED Cadence
Stand-By
Orange,
Green
Red
Green
Port Status
Jumpers
Jumper Number
Position
Function
J2 and J3
AB
Normal Operation
BC
Debug
AB
Normal Operation
BC
Boot
AB
Normal
BC
Debug
J5
J8
supply, before you begin the installation of the card. However, the DKP Card supports Hot Swap so you
can insert the DKP Card while the system is switched on. Always wear an electrostatic discharge
prevention wrist strap/belt and use a grounding mat.
4. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket, keep for future use to cover empty slots.
5. Insert the DKP card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
6. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
533
If you are installing more than one DKP card, it is not necessary to install the next card in the subsequent
slot.
7. Use the cable supplied with the DKP card to connect the DKP Ports with the Main Distribution Frame.
Refer the connector pin details for each DKP Card type given in the following.
534
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
Port 3
Port 4
Port 5
Port 6
Port 7
Port 8
DKP
Port 9
Port 10
Port 11
Port 12
Port 13
Port 14
Port 15
Port 16
535
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
Port 2
Port 3
Port 4
DKP
Port 5
Port 6
Port 7
Port 8
536
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
Port 3
Port 4
DKP
Port 5
Port 6
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
SLT
Port 13
Port 14
Port 15
Port 16
537
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
DKP
Port 3
Port 4
Port 5
Port 6
CO
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
SLT
Port 13
Port 14
Port 15
Port 16
538
8. Plug in the Centronic Male Connector of the cable supplied with the card into the connector on the DKP
Card. Refer to the pinout details of the connectors of each DKP Card type illustrated above.
9. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
Installing EON48
1. Unpack the box and verify the package contents115.
2. Mount the phone on a desk or wall at a convenient location.
3. To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
Foot Stand
DND
Redial Release
Hold
abc
3 def
4 ghi
jkl
6 mno
tuv
9 wxyz
7 pqrs 8
CA 3
Keyhole
Slot 2
Line
4P4C Spring
Cord
Press
down
to detach
Foot Stand
Press down
to detach
Foot Stand
Names
CA 4
Keyhole
Slot 1
Headset
Port
CA 2
CA 1
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
539
4. When you mount EON48 on a desk, you can attach the Foot Stand in two ways as illustrated below.
5. Connect the handset of the EON48 to the phone body using the spring cord.
6. To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
Headset
540
You may also plug in a stereo headset with an RJ12 connector into the headset port at the bottom of the
phone, marked with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
7. Plug one end of the RJ45 cable supplied with the phone into the RJ45 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
8. When the ETERNITY is powered ON, the EON will get reset. The EON communicates with the ETERNITY.
The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the
message 'Please wait' appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
9. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters, these will appear, as illustrated below.
202 Reception
M on 2 4 A U G 1 2 : 0 0
541
10. You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-
Operation.
For the purpose of testing, you may connect one or two DKPs directly to the connectors of the ETERNITY
DKP card.
Installing EON310116
1. Unpack the box and verify the package contents117.
2. Mount the phone on a desk or on the wall at a convenient location.
3. To mount EON310 on a wall, detach the Foot Stand on the bottom of the phone.Refer to the illustrations in
EON48.
CA 1
CA 2
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole Slots 1
and 2 of EON310. The screws should protrude from the wall to fit into the Keyhole Slots.
Now, mount the phone with the screws fitting into the keyhole slots.
4. When you mount EON310 on a desk, you can attach the Foot Stand in two ways - 35 and 55 degree
542
You may also plug in a stereo headset with an RJ12 connector into the headset port marked with the
symbol
, on the left side panel of the phone as illustrated in the figure below.
Headset
Casio Jack
Headset
(R J12 Connector)
7. Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector and the other end into the
wall jack. The cable in the wall jack originates from the DKP card through the MDF.
8. When the ETERNITY is powered ON, the EON will get reset and the message 'Welcome to Matrix.
9. The EON communicates with the ETERNITY. The handshaking lasts for 5-6 seconds. The EON model,
version and revision number, along with the message Please Waitappears on the LCD display.
543
10. After successful handshaking and reset cycle, the default extension number, day, date and time will appear
on the LCD of the phone. If you have already assigned extension number and name, in the DKP
Parameters. These will appear, as illustrated below.
You may adjust the LCD for brightness, contrast and backlight. Refer the topic, Digital Key Phone-Operation.
You can install two DSS consoles to a DKP. Refer Direct Station Selection Console for possible
combinations for installing the models of DSS Consoles.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated on the bottom of
544
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
Windows 98
Windows XP
Windows NT
Windows 2003
Windows Vista
Windows 2007
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
The EONSOFT has a self-executing program and will automatically install itself on your PC.
7. If the software does not perform auto install on your PC, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
9. Click the Matrix EONSOFT Icon to execute installation of the program.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
545
11. Click the shortcut to open the program. The EONSOFT window will open:
12. Click Options at the top left of the window. A drop down menu will appear.
546
14. Select the COM Port to which the communication cable is connected.
547
15. EONSOFT is now connected. If you have already configured the DKP parameters like Access Code and
Name for the port to which EONSOFT is connected, these will appear.
If this window does not appear after you have selected the COM Port Option, test the COM Port for
data transfer.
If the wrong COM port has been selected, a dialog box will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
548
From the drop down menu of Options, select the COM Port to which you have connected the
communication cable.
Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
Click the button labeled Start Test in the COM Port Settings dialog box.
After clicking this button, observe the Test Result section on the dialog box.
The Error Count value shows zero, if both the communication cable and the COM port are working.
The above screen shows that the COM Port/communication cable is working.
If the Error Count shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
Remove the communication cable from the COM Port of the PC.
Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
Now, if the error count is zero, please check the Communication Cable.
If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.
549
The CO Card
The CO Card provides the interface to connect the ETERNITY with the Two-Wire Analog Trunk lines from the CO
Network. The CO Card supports the different standards and features of CO Networks across the world.
The ETERNITY MEX12S supports upto 128 CO ports.The CO Card can be installed in any of the 12 Universal
Slots of ETERNITY MEX12S
The CO Card is available in different configurations. CO interface is also available in combination with SLT and
DKP ports on a single card.
Combination card, with 8 CO ports to connect 8 CO analog trunk lines and 8 SLT
ports to connect 8 Single Line Telephones
LEDs
All CO Cards have one dual colour LED. The LED indicates the Status of the Card.
LED Pattern
Stage
LED Color
LED Cadence
Red
Green
Red
Green
Orange
Initialization
550
Stage
LED Color
LED Cadence
Stand-By
Orange,
Green
Red
Green
Port Status
Jumpers
Jumper Number
Position
Function
J2 and J3
AB
Normal Operation
BC
Debug
AB
Normal Operation
BC
Boot
AB
Normal
BC
Debug
J5
J8
supply, before you begin the installation of the card. However, the CO Card supports Hot Swap so you can
insert the CO Card while the system is switched on. Always wear an electrostatic discharge prevention
wrist strap/belt and use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
should make perfect contact with those of the slot on the backplane motherboard.
4. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
Refer the illustrations below for the pinout details of the connectors on each card.
551
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
Port 3
Port 4
CO
Port 5
Port 6
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
CO
Port 13
Port 14
Port 15
Port 16
552
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
Port 2
Port 3
Port 4
CO
Port 5
Port 6
Port 7
Port 8
553
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
Port 3
Port 4
CO
Port 5
Port 6
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
SLT
Port 13
Port 14
Port 15
Port 16
554
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 2
DKP
Port 3
Port 4
Port 5
Port 6
CO
Port 7
Port 8
Port 9
Port 10
Port 11
Port 12
SLT
Port 13
Port 14
Port 15
Port 16
6. Plug in the Centronic Male Connector of the cable supplied with the card into the connector on the CO
Card. Refer to the pinout details of the connectors of each CO Card type illustrated above.
555
7. Terminate the free end of the CO Card cable into the punch down blocks of the Krone modules designated
556
ETERNITY
ULSB MK
MEX12S
III
BRI 1
NT 1
ISDN Compatible
Deivce 1
ISDN Compatible
Deivce 8
ISDN Compatible
Deivce 1
ISDN Compatible
Deivce 8
BRI 2
BRI 3
PWR
BRI 8
NT 1
PWR
U-Interface
(2-wire)
S/T
Interface
Where,
U Interface = between the ETERNITY MEX12S and the NT1 equipment.
S/T Interface = between the ISDN user equipment and the Network Interface Equipment (NT1).
The BRI port of the ETERNITY MEX12S is terminated on the NT1 device.
The ETERNITY MEX12S supports upto 32 BRI Ports. The BRU Card can be installed in any of the 12 Universal
Slots of ETERNITY MEX12S.
557
LEDs
The BRI-U Interface Card has 8 dual colour LEDs, one for each port on the card.
The LEDs show the Status of the Ports as summarized in the table below:
Port Status
LED Color
LED Cadence
Layer 1 - Up
Green
Continuously ON
Layer 2 - Up
Red
Continuously ON
HeartBeatStatus (LED1)
Green/Red according to
Port Status
Jumpers
Jumper Number
Position
Function
J1
AB (default)
External Boot
BC
Internal Boot
supply, before you begin the installation of the card. However, the BRI Card supports Hot Swap so you can
insert the BRI Card while the system is switched on. Always wear an electrostatic discharge prevention
wrist strap/belt and use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
3. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
4. Use the cable supplied with the card to connect the BRI Ports to the NT1 device supplied by your ISDN
service provider.
Refer the connector pin details for each DKP Card type given in the following.
Port Number
Port 1
Port 2
558
Pin Number
Signaling
Wire Colour
Tip
White
26
Ring
Blue
Tip
White
28
Ring
Green
Port Number
Port 3
Port 4
Port 5
Port 6
Port 7
Port 8
Pin Number
Signaling
Wire Colour
Tip
White
30
Ring
Slate
Tip
Red
32
Ring
Orange
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
13
Tip
Black
38
Ring
Green
15
Tip
Black
40
Ring
Slate
Pin Details
Tx
Rx
Pin Details
Rx1
Tx1
Tx2
Rx2
5. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
559
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
Antenna
ETERNITY MEX12S Mobile Card8 has 2 antennas and Mobile Card4 has a single antenna. A splitter connects four
ports on the card into a single antenna. An antenna cable is also provided, giving you the flexibility to move the
antenna to another position (in case of weak signal).
LEDs
The number of tri-color LEDs on the GSM card corresponds with the number of mobile ports on the card.
LED Pattern
At Power On: All LEDs will blink 1 second ON and 1 second OFF in the color sequence: Red-Green-Orange until
the Reset cycle is complete.
In the Stand-by state: All LEDs will glow Orange for a second and turn Green for a second, repeatedly.
560
During normal functioning: The LEDs will various events on the Mobile port in the color and cadence described
in the table below:
Event
Color
Port disabled
LED OFF
Port idle
LED OFF
Red
Continuous ON
Ring Event
Green
Speech
Green
Continuous ON
GSM initialization
Orange
PUK required
Orange
Orange
SIM Absent
Orange
Orange
If using a GSM/3G card, get the SIM Card from the GSM/3G service provider of your choice ready. Use
SIM PIN protection, if required.
2. Make sure that the ETERNITY MEX12S is installed at a location where sufficient network coverage is
available.
3. Unpack the Mobile Card and check the package contents. It is recommended that you switch off the power
supply, before you begin the installation of the card. However, the Mobile Card supports Hot Swap so you
can insert the Mobile Card while the system is switched on. Always wear an electrostatic discharge
prevention wrist strap/belt and use a grounding mat.
561
change the SIM PIN to 1234 (this is the default PIN for all SIM cards used in the system). Changing the
SIM PIN to '1234' enables you to change the SIM PIN from the ETERNITY later (Refer SIM PIN for
instructions).
remove the SIM from the mobile handset.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
5. Insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the Mobile
card. You can insert multiple SIM cards of the same service provider or of different service providers.
Port Type
1
2
3
4
5
6
7
8
Module Number
M4
M3
M2
M1
6. Insert the Mobile card into the guide rails of the Universal Slot you have selected for this card. Make sure
that the card is inserted deep enough to make perfect contact with the connectors in the backplane. Now,
press down the levers on the card mount bracket to secure the card in its slot.
562
7. Connect the antenna provided with the card on the splitter connector on the front panel of the card. You
may also use the antenna cable to place the antenna at another position.
8. Repeat Steps 1-7 to insert another Mobile card.
9. If you have completed all installations tasks, power the system.
10. Wait for the system to register with the Mobile network. By default, the Mobile ports are set to select and
register with the Mobile networks automatically. Now, observe the LED Patterns of the Mobile Ports.
At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
Each time the Mobile Port sends a request, such as Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the Mobile network. It is fixed for 150 seconds for all Mobile ports.
563
LEDs
The E1FO Card has two Dual color LEDs: L1, L2.
L1 is assigned to FO1 (E1 Port 1)
L2 is assigned to FO2 (E1 Port 2)
The Indication of both LEDs is summarized in the following.
Port Status
Color
Cadence
Port Disabled
Green
Green
Continuous ON
Green
Green
Loss of Signal
Red
Continuous ON
AIS Alarm
Red
MFA Error
Red
CRC 4 (Disabled)
Orange
Continuous ON
RAI
Orange
BFA
Orange
Jumpers
Jumper Number
Position
Function
J2
AB (default)
External Boot
BC
Internal Boot
564
Jumper Number
Position
Function
J5
AB
Backplane communication
BC (default)
Backplane communication
AB (default)
Backplane communication
BC
Backplane communication
AB
BC (default)
AB
BC (default)
J6
J11
J19
supply, before you begin the installation of the card. However, the E1FO Card supports Hot Swap so you
can insert the E1FO Card while the system is switched on. Always wear an electrostatic discharge
prevention wrist strap/belt and use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
3. Insert the E1FO Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one E1FO Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
4. Use the cable supplied with the card to connect the E1FO Ports to another PBX/Modem (Copper
interface) or at the desired location where Fibre Optic termination is required (FibreOptic interface).
5. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Features
LEDs
The Data Card has one dual color LED.
Status
Color
Cadence
Red
Up-gradation in progress
Green
Red
Continuous ON
Green
Continuous ON
Red
Jumpers
Jumper Number
Position
Function
J9
AB (default)
External Boot
BC
Internal Boot
supply, before you begin the installation of the card. However, the Data Card supports Hot Swap so you
can insert the Data Card while the system is switched on. Always wear an electrostatic discharge
prevention wrist strap/belt and use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
566
3. Insert the Data Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one Data Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
4. Use the cable supplied with the card to connect the Data Ports to the Ethernet Network (Switch/PC).
5. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Features
567
Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer PLCC-An Introduction to know more.
Closed User Group (CUG), where several PBXs are connected with each other through E&M tie lines120.
PBX expansion, where two PBXs are connected with each other with E&M tie lines.
E&M Trunk Seizure Type121: Immediate, Immediate + Wink, Immediate with Ack, Immediate with
Ack+Wink, Seizure Pulse, Seizure Pulse + Wink, Express, and Compander Control Signal.
Address Signaling: Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The ETERNITY E&M Card is available in the following configuration for the ETERNITY GE
The number of E&M lines that you can interface with the ETERNITY using the E&M Card varies according to
number of E&M ports supported by the each variant of ETERNITY GE.
The maximum number of E&M ports supported by each variant of ETERNITY are:
ETERNITY GE3S: 12
ETERNITY GE6S: 24
ETERNITY GE12S: 48
120. The PBXs in a Closed User Group (CUG) can be connected over ISDN T1E1PRI Lines as well. Refer the topic Closed User
Groups to know more.
121. This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
568
Connectors
The E&M card has RJ45 Connectors. A separate MDF cable is supplied for each connector.
LEDs
The ETERNITY GE E&M4 Card has 4 LEDs to indicate the functioning of the ports.
a Trunk - works like a trunk interface when any of the extensions of the PBX makes an outgoing call
through it.
OR
a Tie Line - takes on a dual personality: functioning as both as an extension and a trunk. The E&M port
works like an extension interface for incoming calls. It works like a trunk interface when any extension
makes an outgoing call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
PLCC-An Introduction to know more.
You cannot connect a trunk line or an SLT or a DKP to an E&M port.
1. Have the necessary wiring for the E&M Analog trunk in place. Take the necessary safety precautions
before you begin handling the card; switch off power supply and always wear an antistatic wrist strap and
use a grounding mat.
2. Unpack the E&M card and check the package contents.
3. The E&M Card supports E&M Interface Type IV and Type V connection. To select the appropriate
Interface Type out of the two, you need to change the Jumper Settings.
Refer the table below to select the desired Interface Type and Speech Interface.
Jumper Number
Position
Function
J1 and J2
AB
BC
To select the Type-V connection for the E&M Port, set Jumpers J1 and J2 (located on the E&M
module) in BC Position.
569
4. Select the speech interface - 2-wire speech or 4-wire speech - as required, by changing the jumper
Position
Function
J3 and J4
AB
BC
By default all the E&M Ports are set to support 2-wire Speech Interface.
To select 2-wire speech interface for the E&M Port, set Jumpers J3 and J4 (given on E&M module) to
BC Position.
To select 4-wire speech interface for the E&M Port, set Jumpers J3 and J4 on E&M module to AB
Position.
contact with those on the backplane motherboard. Secure the card by pressing down the levers and fix the
bracket with the screws provided with the card.
7. Connect the cables supplied with the E&M card into the RJ45 connectors on the E&M Card.
8. Connect the free ends of the cables into the E&M Ports of the other PBX/Router/Tie Line equipment by
570
571
572
9. If you are connecting two PLCC EPAX in a Power Line Carrier Communication Network Compander
Control Signal (CCS) Connection should be made as illustrated in the block diagram below for any of the
four combinations of E&M and Speech Interfaces illustrated in the previous step.
Compander Control Signal (CCS) is a special type of signal used by Power Line Carrier Communication
Networks to improve quality of speech transmission. The PLCC network expects this signal from the PBX
when speech is established. The E&M Card supports this facility. The ETERNITY sends CCS signal to the
PLCC panel.
When the E&M port is used as an Endpoint; the system sends a CCS to the PLCC panel while making
an outgoing call through the E&M port or when a call is received at the E&M port.
When the E&M port is used for Transit Exchange; the system sends a CCS to the PLCC panel while
there is a Transit call through the E&M port.
10. If you have completed all installation tasks, power ON the system, observe the Reset Cycle and the LED
LED Color
LED Cadence
At Power ON
LED OFF
LED OFF
RED
GREEN
RED
GREEN
Green
E-Wire Low
E-Wire and M-Wire High
Orange
573
LEDs
The Radio Card has one dual color LED.
LED Pattern
Stage
LED Color
LED Cadence
Red
Green
Red
Green
Orange
Orange,
Green
Red
Green
Initialization
Stand-By
Port Status
574
Position
Function
J2 and J4
AB
Normal
BC
Debug
AB
Normal
BC
Boot
J5
Position
Function
J1 and J2
AB
4-wire speech
BC
2-wire speech
supply, before you begin the installation of the card. However, the Radio Card supports Hot Swap so you
can insert the Radio Card while the system is switched on. Always wear an electrostatic discharge
prevention wrist strap/belt and use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
3. Insert the Radio Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one Radio Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
4. Use the cable supplied with the Radio Card to connect the Radio devices.
575
Port 1
Port 2
Port 3
Port 4
Port 5
576
Pin Number
Signalling
Wire Colour
SPCH1_A
White
26
SPCH1_B
Blue
RX1_SPCH_A
White
27
RX1_SPCH_B
Orange
CCC1
White
28
CCC1_RTN
Green
SPCH1_A
White
29
SPCH1_B
Brown
RX1_SPCH_A
White
30
RX1_SPCH_B
Slate
CCC1
Red
31
CCC1_RTN
Blue
SPCH1_A
Red
32
SPCH1_B
Orange
RX1_SPCH_A
Red
33
RX1_SPCH_B
Green
CCC1
Red
34
CCC1_RTN
Brown
10
SPCH1_A
Red
35
SPCH1_B
Slate
11
RX1_SPCH_A
Black
36
RX1_SPCH_B
Blue
12
CCC1
Black
37
CCC1_RTN
Orange
13
SPCH1_A
Black
38
SPCH1_B
Green
14
RX1_SPCH_A
Black
39
RX1_SPCH_B
Brown
15
CCC1
Black
40
CCC1_RTN
Slate
Port Number
Pin Number
Signalling
Wire Colour
Port 6
16
SPCH1_A
Yellow
41
SPCH1_B
Blue
17
RX1_SPCH_A
Yellow
42
RX1_SPCH_B
Orange
18
CCC1
Yellow
43
CCC1_RTN
Green
19
SPCH1_A
Yellow
44
SPCH1_B
Brown
20
RX1_SPCH_A
Yellow
25
RX1_SPCH_B
Slate
21
CCC1
Violet
46
CCC1_RTN
Blue
22
SPCH1_A
Violet
47
SPCH1_B
Orange
23
RX1_SPCH_A
Violet
48
RX1_SPCH_B
Green
24
CCC1
Violet
49
CCC1_RTN
Brown
Port 7
Port 8
Port 1
Port 2
Pin Number
Signalling
Wire Colour
SPCH1_A
White
26
SPCH1_B
Blue
RX1_SPCH_A
White
27
RX1_SPCH_B
Orange
CCC1
White
28
CCC1_RTN
Green
SPCH1_A
White
29
SPCH1_B
Brown
RX1_SPCH_A
White
30
RX1_SPCH_B
Slate
CCC1
Red
31
CCC1_RTN
Blue
577
Port Number
Port 3
Port 4
Pin Number
Signalling
Wire Colour
SPCH1_A
Red
32
SPCH1_B
Orange
RX1_SPCH_A
Red
33
RX1_SPCH_B
Green
CCC1
Red
34
CCC1_RTN
Brown
10
SPCH1_A
Red
35
SPCH1_B
Slate
11
RX1_SPCH_A
Black
36
RX1_SPCH_B
Blue
12
CCC1
Black
37
CCC1_RTN
Orange
5. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
578
extension phones, any standard, two-wire, analog single line telephone instrument - rotary, pulse-tone,
cordless, feature phones with or without Calling Line Identification.
the ETERNITY MEX12S with the Two-Wire Analog Trunk lines from the CO Network.
the proprietary digital key phones of the EON series, the proprietary PC-based phone EONSOFT, the
Direct Station Selection (DSS) Consoles, with ETERNITY MEX12S.
The CO4+DKP4+SLT8 Card can be installed in any of the 12 Universal Slots of ETERNITY MEX12S.
LEDs
The CO4+DKP4+SLT8 Card one dual colour LED. The LED indicates the Status of the Card.
LED Pattern
Stage
LED Color
LED Cadence
Red
Green
Red
Green
Orange
Orange,
Green
Red
Green
Initialization
Stand-By
Port Status
579
Jumpers
Jumper Number
Position
Function
J2 and J3
AB
Normal Operation
BC
Debug
AB
Normal Operation
BC
Boot
AB
Normal
BC
Debug
J5
J8
before you begin the installation of the card. However, this card supports Hot Swap so you can insert it
while the system is switched on. Always wear an electrostatic discharge prevention wrist strap/belt and
use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
should make perfect contact with those of the slot on the backplane motherboard.
4. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
Refer the illustrations below for the pinout details of the connectors on each card.
Port Number
Pin Number
Signalling
Wire Colour
Port 1
Tip
White
26
Ring
Blue
Tip
White
27
Ring
Orange
Tip
White
28
Ring
Green
Tip
White
29
Ring
Brown
Port 2
DKP
Port 3
Port 4
580
Port Type
Port Number
Pin Number
Signalling
Wire Colour
Port 5
Tip
Red
31
Ring
Blue
Tip
Red
32
Ring
Orange
Tip
Red
33
Ring
Green
Tip
Red
34
Ring
Brown
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
19
Tip
Yellow
44
Ring
Brown
Port 6
CO
Port 7
Port 8
Port 9
Port 10
Port 11
SLT
Port 12
Port 13
Port 14
Port 15
SLT
Port 16
6. Plug in the Centronic Male Connector of the cable supplied with the card into the connector on the card.
Refer to the pinout details of the connectors of the card illustrated above.
7. Terminate the free end of the card cable into the punch down blocks of the Krone modules designated for
Trunk Lines for CO Ports, Station Lines for SLT and DKP ports.
Refer the topics The Main Distribution Frame (MDF) and Terminating Trunk and Extension Cables on
the MDF.
8. Repeat these steps to install other CO4+DKP4+SLT8 cards.
581
Function
CO1 and CO 2
Two Loop Dial (LD) ports to connect two EPABXs on each port.
SLT1 to SLT6
The Combo Card has a 50-pin Centronics Connector for the ports.
LEDs
There are 2 dual color LEDs on the Combo Card.
Stage
LED Color
LED Cadence
Red
Green
Red
Green
Orange
Orange,
Green
Red
Green
Initialization
Stand-By
Port Status
582
Jumpers
Jumper Number
Position
Function
J1
AB (default)
Normal
BC
Boot/Programming
AB
BC (default)
Backplane communication
AB
Backplane communication
BC (default)
Backplane communication
AB
BC (default)
AB (default)
Four-wire Speech
BC
Two-wire Speech
AB (default)
Four-wire Speech
BC
Two-wire Speech
AB
Type IV
BC (default)
Type V
AB
Type IV
BC (default)
Type V
AB
Do Not Change!
BC (default)
Do Not Change!
J2
J3
Module Jumpers
Magneto
J1
E&M
J3
J4
J1
J2
J5
J6
Open
BC
supply, before you begin the installation of the card. However, the Combo Card supports Hot Swap so you
can insert the Combo Card while the system is switched on. Always wear an electrostatic discharge
prevention wrist strap/belt and use a grounding mat.
2. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Do not discard the filler bracket! Preserve it for future use!
3. Insert the Combo Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
583
If installing more than one Combo Card, it is not necessary to insert the other cards in subsequent slots.
Any card can be inserted in any of the Universal Slots.
4. Use the cable supplied with the card to connect the Combo Ports to the CO, LD, Magneto, E&M and SLT
E & M Port 1
Pin Number
Signalling
Wire Colour
SPCH1_A
White
26
SPCH1_B
Blue
RX1_SPCH_A
White
27
RX1_SPCH_B
Orange
E1_In_M
White
28
DGND_EM
Green
M1_Out_M
White
29
SB1
Brown
CCC1
White
30
E & M Port 2
Slate
SPCH2_A_M
Red
31
SPCH2_B_M
Blue
RX2_SPCH_A
Red
32
RX2_SPCH_B
Orange
E2_In_M
Red
33
DGND_EM
Green
M2_Out_M
Red
34
SB2
Brown
10
CCC2
Red
35
Magneto Port 1
Magneto Port 1
SLT Port 1
SLT Port 2
SLT Port 3
584
Slate
11
Tip
Black
36
Ring
Blue
12
Tip
Black
37
Ring
Orange
13
Tip
Black
38
Ring
Green
14
Tip
Black
39
Ring
Brown
15
Tip
Black
40
Ring
Slate
Port Number
Pin Number
Signalling
Wire Colour
SLT Port 4
16
Tip
Yellow
41
Ring
Blue
17
Tip
Yellow
42
Ring
Orange
18
Tip
Yellow
43
Ring
Green
21
Tip
Violet
46
Ring
Blue
22
Tip
Violet
47
Ring
Orange
24
Tip
Violet
49
Ring
Brown
25
Tip
Violet
50
Ring
Slate
52
Earth
SLT Port 5
SLT Port 6
19
Not Used
44
20
45
LD Port 1
LD Port 2
Not Used
CO Port 1
CO Port 2
23
48
5. Plug in the Centronic Male Connector of the cable supplied with the card into the connector on the card.
Refer to the pinout details of the connectors of the card illustrated above.
6. Terminate the free end of the card cable into the punch down blocks of the Krone modules designated for
Trunk Lines for CO Ports, Station Lines for SLT and DKP ports. Refer the topics The Main Distribution
Frame (MDF) and Terminating Trunk and Extension Cables on the MDF.
7. Repeat these steps to install other Combo Cards.
8. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Features
585
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
The ETERNITY MEX12S supports upto 12 VoIP Ports. The VoIP Card can be installed in any of the 12 Universal
Slots of ETERNITY MEX12S.
Voice Channels
There are 32 Voice Channels on the VoIP32 Card, allowing as many simultaneous calls to be made (using SIP
Trunks and/or Extensions) as the number of Voice Channels supported by these cards.
A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
voice channels, whereas a call made from an SLT or DKP extension to a SIP Extension or SIP Trunk will
consume one voice channel. Thus, the number of speech paths available to make simultaneous calls will
depend not only on the number of voice channels, but also be the number of channels consumed by such
SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
586
SIP Trunks
The ETERNITY MEX12S supports a maximum of 16 SIP Trunks, allowing you to subscribe to as many as 16
different Internet Telephony Service Providers (ITSP).
It is possible to program all 16 SIP trunks on a single VoIP Card or program them in a distributed manner, where
more than one VoIP card is installed in the system.
SIP Extensions
ETERNITY MEX12S supports 500 SIP Extensions. Upto 250 SIP Extensions can be registered with a single VoIP
Card. To register more than 250 SIP Extensions, you need at least two VoIP Cards.
Any SIP-enabled device like an IP-phone, a Softphone, analog phone adapter, can be registered with the VoIP
Card and function as the 'SIP Extension' of the ETERNITY MEX12S.
The SIP Extensions function in the same way as other extensions of the ETERNITY. SIP Extension users can make
and receive calls from and to other extensions of ETERNITY and external numbers over PSTN, GSM, VoIP and
E&M lines122. You can also connect the Standard and Extended IP Phones offered by Matrix as SIP Extensions.
A SIP Extension can be registered with the ETERNITY MEX12S from three different locations. This helps
organizations overcome geographical distances and reduce call costs.
SIP Extensions require a license. To know more about Licensing requirements and how to acquire and
activate a license key, see the topic License Management.
LED
There are two Dual color LEDs on the VoIP Card: LED 1 and LED 2.
LED 2 indicates the status of any of the SIP Trunks to which this LED is assigned.
LED Color
Cadence
Green
Continuous ON
Red
Continuous ON
Red
ON 1 sec-OFF 1 sec
ON 1 sec-OFF 1 sec
122. Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your country prohibit call traffic between the public telephony networks and IP networks, you must configure Logical Partition in your system.
To know more, see Logical Partition.
587
LED Color
Cadence
Red
ON 500msec-OFF 3500msec
Green
ON 500msec-OFF 3500msec
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 2500msec
Green
ON 500msec-OFF 500msec
ON 500msec-OFF 500msec
ON 500msec-OFF 1500msec
Green
LED Color
Cadence
Red
Continuous ON
Red
ON 500msec-OFF 3500msec
Red
ON 500msec-OFF 500msec
ON 500msec-OFF 2500msec
Registration in Progress
Green
ON 500msec-OFF 3500msec
Registration Successful
Green
Continuous ON
SIP Trunk Status will be indicated by LED2 only after you have programmed the LED Indication in the VoIP
Port Parameters.
Jumpers
Jumper Number
Position
Function
J7
AB (default)
External Boot
BC
Internal Boot
588
LAN Switch/Hub
LAN
CPU Card
WAN
IP
IP
Router
A Broadband Internet Connection to make/receive calls through the Public Internet. If you wish to make
calls within your network (LAN), you do not need an Internet connection.
SIP ID/User ID
Authentication User ID
Authentication Password
SIP Registrar Server Address
SIP Registrar Server Port
You may ask your Internet Service Provider / LAN administrator for the above information.
Network Information:
123. Peer-to-Peer calls are calls made without the intervention of a SIP Server or Proxy Server.
589
Gateway Address
DNS Address
DNS Domain Name (if applicable)
IP Address of your LAN
IP Address of the LAN Port of the VoIP Card (Default: 192.168.002.031)
LAN
CPU Card
WAN
LAN Switch/Hub
IP
IP
Router
The card is located behind the NAT Router and Private IP is assigned to the WAN port.
When connecting the card in a Private Network, you would require the following information:
IP Addressing Scheme of your network; whether the Connection Type is DHCP, Static, PPPoE
IP Address of the WAN Port of the VoIP Card (Default: 192.168.001.116)
Subnet Mask of the Network to which the WAN Port is connected. (Default: 255.255.255.000)
Gateway Address
DNS Address
DNS Domain Name (if applicable)
VoIP Card connected to the Public Network for Matrix Extended IP Phones
590
Public IP is assigned to the WAN Port of the VoIP card and the Ethernet Port of the Master Card.
Here, the LAN port of the VoIP Card is connected to the LAN Switch/Hub. The WAN Port of the Card is connected
to the Public Network and the Master Ethernet Port of ETERNITY is also connected to the Public Network.
This installation is required when you want to register the Matrix Extended IP Phone with ETERNITY from the
Public Network. The Master Ethernet Port is used for Auto Configuration of the Matrix Extended IP Phones.
To install the VoIP Card, do the following:
1. Get the items/information listed ready before you install the VoIP card and connect it to the IP network.
2. Unpack the VoIP card and verify the package contents. It is recommended that you switch off the power
supply, before you begin the installation of the card. However, the VoIP Card supports Hot Swap so you
can insert the VoIP Card while the system is switched on. Always wear an electrostatic discharge
prevention wrist strap/belt and use a grounding mat.
3. Select any of the free Universal Slots of ETERNITY to insert the VoIP Card. Unscrew and remove the filler
bracket of the slot. Preserve the filler bracket for future use.
4. Insert the card into the guide rails of the slot. The card should be inserted deep enough to make perfect
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP card and
the other end into the Router/Modem.
Plug one end of the Ethernet cable supplied with the card into the WAN Port of the card and the other
end into the LAN Switch/Hub.
Plug one end of the Ethernet cable supplied with the VoIP card into the WAN Port of the VoIP card and
the other end of the cable into the Router/Modem.
Connect the LAN Port of the VoIP card to the LAN Switch/Hub.
7. To insert and connect another VoIP card, repeat the same steps as described above.
8. If you have completed all other installation tasks, you may switch on power supply and observe the Reset
591
SIP Extensions
ETERNITY MEX12S supports up to 500 SIP Extensions. The SIP Extensions function in the same way as DKP/
SLT extensions of the ETERNITY MEX12S. SIP Extension users can make and receive calls to any extension user
of the ETERNITY and to external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and
VoIP124.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the SIP
Extension of the ETERNITY MEX12S.
To register SIP Extensions, a VoIP Card must be installed in the ETERNITY MEX12S, and must have the IP8
License. For more information on Licensing, see License Management.
You can register upto 250 SIP Extensions with a single VoIP Card of ETERNITY MEX12S. However, at a time, only
as many extensions as the number of Voice Channels supported by the VoIP Card can make calls.
You can register the same SIP Extension from three different locations.
You may connect the Standard and Extended IP Phones of Matrix as SIP Extensions.
The Matrix Extended IP Phone, SPARSH VP248, takes on all the functions of EON48, the proprietary digital key
phone of Matrix, except the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Screening
To connect SPARSH VP248 with ETERNITY, see Connecting SPARSH VP248 as Extended SIP Extension.
SPARSH VP330 is proprietary Extended IP Phones with graphical touch-screen user interface. This feature-rich
SIP based phone support most features and functions of the proprietary digital key phones of ETERNITY except
the following features:
Background Music
Trunk Call Waiting
Hot Desking
Live Call Supervision
Login Station from DISA
You cannot program SIP Extension from Enterprise or Hotel Wizard.
To connect SPARSH VP330 with ETERNITY, see Connecting SPARSH VP330 as Extended SIP Extension.
If you register the Extended IP Phone outside the Region/Country selected for ETERNITY, the time and
Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of ETERNITY. Also, Access Codes and Emergency
Numbers will work according to the Region/Country selected for ETERNITY.
124. Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
Logical Partition.
592
The SIP Extensions may be registered over WAN or over LAN according to your preference and your IP network
installation scenario.
If the ETERNITY MEX12S Master Ethernet Port and the VoIP Card are connected to a Public Network,
Connect SPARSH VP248, the Extended IP Phone, or any Open SIP device to the LAN Switch.
Register any SIP device (Extended IP phone or Open SIP phone) on the public network as SIP extension.
ETERNITY GE12S
LAN Switch/Hub
LAN
CPU Card
WAN
IP
IP
Router
When you register the Matrix Extended IP Phone with ETERNITY, make sure the Master Ethernet Port and
the WAN port of the VoIP Card are connected to the public network. The Master Ethernet Port is used for
Auto Configuration of the Matrix Extended IP Phones.
When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension of ETERNITY, in this SIP device, you must configure the following:
the Registrar Server Address of ETERNITY MEX12S
the Registrar Server Port
the SIP ID
Authentication ID and Password.
593
If the ETERNITY MEX12S Master Ethernet Port and VoIP Card are connected to a Private Network (Behind the NAT),
ETERNITY GE12S
LAN
CPU Card
WAN
LAN Switch/Hub
IP
IP
Router
Connect SPARSH VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension.
When you register the Matrix Extended IP Phone with ETERNITY, configure Port Forwarding for Master
Ethernet Port and the WAN port of the VoIP Card on the Router. The Master Ethernet Port is used for
Auto Configuration of the Extended IP Phones.
Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server on your LAN for assigning IP Address to the Extended IP Phone, do
the following:
use DHCP option 224 and Data Type as String to provide Server Address to the Extended IP
Phones.
594
Program the IP Address or the Dynamic DNS Domain Name of the Master Ethernet Port of
ETERNITY MEX12S in the DHCP option.
Log in to Jeeves. For instructions, read the topic Configuring using Web-based GUI: Jeeves under
Configuring ETERNITY.
Assign an extension number (SIP ID or Access Code) to the Extended IP Phone. For instructions on
assigning SIP ID, see Configuring Matrix SPARSH VP248 - Extended IP Phone.
For the SIP extension number you assigned to the Extended IP Phone, go to the Location settings of the
extension, and do the following:
For instructions, see the topic Configuring Matrix SPARSH VP248 - Extended IP Phone under
Configuring SIP Extensions.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SPARSH VP248. For the purpose of illustration, the premium model, SPARSH VP248P, has been used.
1. Unpack the SPARSH VP248 box and verify package contents.
2. Mount the phone on a desk at a location convenient to you.
When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated in the
following.
Foot Stand attached at 30 Angle
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your
desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
595
Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack headset jack with the symbol
Headset
OR
596
You may plug a headset with an RJ12 connector into the headset port at the bottom of the phone, marked
with the symbol
Foot Stand
Keyhole
Slot 1
Keyhole
Slot 2
Headset
Handset
5. Connect the LAN Port of SPARSH VP248 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone125. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
8. Plug the Power Adapter into a power outlet.
9. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
597
The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
As soon as the Loading... message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
When the cursor is placed under the Extended IP Phone, press Enter key.
598
The phone will start loading the Extended IP Phone Firmware. It will display current firmware being loaded.
We l c o me to M a t ri x
L oading V 05R01 Ext SI P
After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, press the Enter key. Detailed instructions for changing the
Network Settings of the phone are provided at the end of this topic. See Network Settingsat the end of
this topic.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
D H C P d i s c o v e r y. . . !
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY MEX12S.
599
As the phone downloads the configuration files, the file names will appear one by one.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
On successful download of all configuration files, the phone attempts to register with ETERNITY MEX12S.
On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 10 M AY 1 5: 4 0
2 00 1 Re ce pt i on
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
The Up key
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
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You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
3. When the phone is in idle state. You must press the DSS key assigned to Local Menu.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt io n
DN D
Names
Local Menu
CA04
CA03
Redial Release
abc
Hold
3 def
4 ghi
jkl
6 mno
7 pqrs
tuv
9 wxyz
CA02
CA01
When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any process, the
Local Menu appears on your phone display.
LO C AL ME N U
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
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You can configure Network Parameters and view Network status from the Local Menu.
In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
You can configure all network parameters described below, except the MAC Address.
Connection Type
Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.
IP Address
If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
Enter the desired Static IP Address by pressing the digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Subnet Mask
If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
To enter the dot/period in the IP Address, press the digit key 1 twice.
Gateway Address
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If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
DNS Server
If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the digit key 1 twice.
If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
Internet Service Provider.
PPPoE Password
This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.
Server Address
ETERNITY MEX12S CPU Card works as the Auto Configuration Server for the phone. Enter the IP
Address or the Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY here. Default:
blank.
The phone sends the request for configuration files to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address automatically from
the DHCP Server. For this, use DHCP option 224 and Data Type as String to provide Server Address
from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
Enter the Web Server Port of the Master Ethernet Port of ETERNITY here.
The phone sends the request for configuration files to this port.
Valid range of the port is: 80 or 102465535. Default: 80.
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VLAN Setting
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic126.
The meaning of CoS bits with respect to traffic type is as follows:
CoS
Traffic Type
Best Effort
Background
Spare
Excellent Effort
Controlled Load
Video
Voice
Network Control
Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
connected. Valid range: 0-4094. Default: 1.
Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
126.The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), that is, better handling of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred
to as Class of Service (CoS) which is defined by IEEE 802.1P.
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PCAP
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 2 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see Using PCAP Trace for Matrix Extended IP Phone, under PCAP Trace.
When you change the Network Settings, the phone will restart.
In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.
MAC:
IP :
MASK:
G W:
DNS:
N E T W O R K S TAT U S
0 0:1 b:0 9:0 0:9 a:a 7
1 92. 16 8. 2 0 1 .2 0 5
2 5 5 . 25 5 . 2 5 5 .0
1 9 2 . 16 8 . 2 0 1 .3
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
S. ADD: The IP Address or Dynamic DNS Domain Name of the Master Ethernet Port of ETERNITY
MEX12S.
S. PORT: The Web Server Port of the Master Ethernet Port of ETERNITY MEX12S.
Decide the location where you want to place SPARSH VP330 within your LAN.
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To use the DHCP Server on your LAN for assigning IP Address to SPARSH VP330, make sure you do the
following:
Use DHCP option 224 and Data Type as String to provide Master Ethernet Port Address to SPARSH
VP330.
Program the IP Address or the Domain Name of the Master Ethernet Port of ETERNITY ME in the
DHCP option 224.
Log in to Jeeves. For instructions, read the topic Configuring using Web-based GUI: Jeeves under
Configuring ETERNITY.
You must configure the necessary parameters in ETERNITY so that SPARSH VP330 can register as a SIP
Extension. For instructions, see Configuring Matrix SPARSH VP330.
When mounting the phone on the wall, detach the Foot Stand from the bottom of the phone.
Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
3. When you mount the phone on a desk, you can attach the Foot Stand in two ways at 30 Angle or at 50
Angle.
If you attach the Foot Stand at 50, the phone will be placed in an almost upright position on your desk.
Decide which of these positions would work for you best and accordingly attach the Foot Stand.
Plug the long straightened end of the phone cord into the handset jack on the left side panel of the
phone marked with the handset symbol.
Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
5. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 3.5 mm single
connector into the headset jack headset jack with the symbol
OR
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You may plug a headset with an RJ12 connector into the headset port on the side panel of the phone,
marked with the symbol
Headset
Casio Jack
Headset
(R J12 Connector)
6. Connect the LAN Port of SPARSH VP330 to the LAN Switch/Hub or a Router/Modem, according to your
installation scenario.
7. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port at the bottom of the phone to the LAN Port of the computer.
8. Plug the connector of the Power Adapter in to the power jack at the back of the phone127. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
9. Plug the Power Adapter into a power outlet.
10. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
All keys with LED, including the Speaker key, and the Ringer LED, will glow.
The LCD display will light up and the following message will appear on it, as the phone boots:
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The Starting SPARSH VP330 ... message appears on the phone display, while loading the
application.
The Applying Network Parameters... message appears on the phone display, while the Static
Network parameters are being applied.
If you want to change the Network Settings or want to use Wi-Fi for connectivity, press Settings
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To change the Network Settings of the phone and configure the network parameters.
The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from ETERNITY ME. The Configuring the phone... message appears on the phone.
On successful download of all configuration files, the phone attempts to register with ETERNITY ME. The
Registering the phone... message appears on the phone display.
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The Updating firmware... message appears on the phone display, when the firmware is being updated.
After the firmware is updated, the phone will reboot. The Rebooting the phone... message appears on
the phone display.
The phone will register successfully, only if the SIP Extension parameters in ETERNITY have been
correctly configured as per your installation scenario.
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Voice Mail System providing voice mail facility to all extensions of ETERNITY.
The card provides mailbox facility to all extensions of ETERNITY MEX12S. Each Mailbox has the capacity of
storing 15,000 messages. The maximum size of each Mailbox is 60,000 minutes. By default, the size of each
Mailbox is set to 5 minutes. The maximum Message Length for each Mailbox is 9999 seconds. By default, the
Maximum Message Length for each Mailbox is set to 15 seconds.
The VMS card utilizes a USB memory stick as its storage medium. Matrix provides a 4GB Pen Drive with the VMS
card. The Pen Drive supports 72 hours of recording. However, you may use a Pen Drive of upto 32GB.
The VMS Card has an Ethernet Port, a communication port, a USB port, and four LEDs.
USB Port
The USB port is an internal port, located on the main board of the card. The Pen Drive provided by Matrix with the
VMS Card is connected to this port. All the voice messages, mailbox messages, greetings and other messages and
prompts are stored in the Pen Drive.
The 4GB Pen Drive is factory fitted and shipped with the card. However, you may use a Pen Drive of upto 32GB.
For instructions see Replacing the Pen Drive at the end of this topic. Make sure the new Pen Drive is inserted in
the same USB Port as the factory fitted one.
LEDs
The ETERNITY MEX12S VMS16 Card has four Dual color LEDs.
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Condition
LED
Color
Cadence
OFF
L1
Green
Continuously ON
L2, L3, L4
OFF
Initialization
L1, L2
L3, L4
OFF
Configuration is being
transferred from
Master to VMS on
Power On
L1, L2
Green
Toggling
L3, L4
OFF
Generating
Directories/Reading
Messages on Power
On [After configuration
transfer]
L1, L2, L3
Green
Continuously ON
L4
OFF
Configuration transfer
completes & VMS
Card is ready to
access features
L1
Green
L2, L3, L4
OFF
L1
Red
L2, L3, L4
OFF
Jumpers
Jumper Number
Position
Function
J5
AB
Default SE Password
BC
Normal
AB (default)
External Boot
BC
Internal Boot
J9
backplane.
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6. Secure the card in its slot by pushing down the levers of the mounting bracket and fixing the card with the
Connecting to a Computer
7. Now, connect the card to a standalone PC/LAN.
Plug in one end of the Ethernet cable supplied with the card into the Ethernet Port of the VMS Card.
Plug the other end of the cable into the Ethernet port of a standalone PC or into a LAN Switch.
When you connect the VMS Card to a standalone/LAN PC, you need to make sure that:
The IP Address of the Ethernet Port of the VMS Card and the Ethernet Port of the PC do not conflict.
The Ethernet Port of the VMS Card and the Ethernet Port of the PC are in the same Subnet.
When your system configuration has been done using the maximum capacity of system resources,
configuration transfer from ETERNITY CPU to the VMS Card will take longer to complete. At first power
on, it will take 7 to 10 minutes for the VMS to initialize. At subsequent power on, the VMS card will take
around 3 minutes to initialize.
To format the Pen Drive with FAT32, follow the steps given below:
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Click My Computer.
Right-click the removable disk to which you have connected your Pen Drive, in this example Removable
Disk (F:).
The Format Removable Disk (F:) options appear on your screen. In File Format select FAT32.
You will get an alert: WARNING: Formatting will erase ALL data on this disk. To format the disk, click OK.
To quit, click CANCEL.
Click OK to format.
When the formatting process is complete, the message Format Complete will appear on your screen.
Now, copy the contents of the factory fitted Pen Drive onto the new Pen Drive.
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CHAPTER 9
Configuring ETERNITY
A telephone
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SE Password
For Firmware Version V10R11 and later
The SE password is a code used to prevent unauthorized access and alterations or misuse of the features and
facilities.The password can be a minimum of 4 characters to a maximum of 12 alphanumeric characters. All ASCII
characters (except '%', '#', '=', '+', '&', '/', '<', '>', 'Double Quote (")' and 'Space' ) and digits 0 to 9 are allowed. As this
password is meant for restricting access to the SE mode, it must be kept strictly confidential with the System
Engineer. The default, SE Password is 1234.
The SE password can be changed. Refer the topic System Security - V10R11 and later for instructions on how to
change the SE password.
If you forget the SE password, you can restore the default SE password. Read the topic System Security - V10R11
and later for instructions on restoring the default SE password.
Quick Installation Wizard - Standard PBX: This wizard helps the Installer/System Engineer to quickly
set-up the ETERNITY for the standard PBX (Enterprise) Application.
Using this Wizard, the Installer/System Engineer can configure as much as 80 percent of the system
configuration, covering all the parameters necessary for the functioning of the system. For advanced
configuration of features and facilities, the Installer/System Engineer must use the Using Configuration
mode.
Detailed information on this Wizard and instructions for using it for system configuration are given later in
this chapter. Refer the topic Using Quick Installation Wizard - Standard PBX.
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Hotel Installation Wizard - Hospitality PBX: This is a special wizard for the Hospitality Application of the
ETERNITY. The Hotel Installation Wizard simplifies the configuration process and helps the Installer/
System Engineer to configure the system for the special telephone and guest/patient management
facilities and features for hotels and hospitals.
You may use this Wizard if you have deployed the ETERNITY as a Hospitality Application. The Hotel
Installation Wizard is recommended to be used when configuring the ETERNITY for the first time.
Refer the separate ETERNITY Hospitality System Manual to know more.
Configuration
While the Quick Installation Wizard provides a fast-track way for system configuration, detailed and advanced
configuration of the system can be done only from the links under Configuration of Jeeves.
The Configuration, as the title itself suggests, enables the Installer to change the values of all configurable
parameters of the system, including those not covered by the Wizard.
Refer the topic Using Configuration for more information and instructions.
As many as four System Engineers can simultaneously login into SE Mode and configure the system via
the Configuration and the Quick Installation Wizard-Standard PBX. However, it is recommended that
multiple login be avoided when using the Quick Installation Wizard and the use of the Wizard be restricted
to a single person only.
SA Password
For Firmware Version V10R11 and later
The access to SA mode may be protected by means of an SA password.
The SA password is a code for preventing unauthorized access to the SA mode. The password can be a minimum
of 4 characters to a maximum of 12 characters. All ASCII characters (except '%', '#', '=', '+', '&', '/', '<', '>', 'Double
Quote (")' and 'Space' ) and digits 0 to 9 are allowed. The default, SA Password is 1111.
Refer the topic System Security - V10R11 and later for instructions on how to change and reset the SA Password.
If you forget the SA password, you can restore the default SA password. Read the topic System Security - V10R11
and later for instructions on restoring the default SA password.
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If you forget the SA password, you can restore the default SA password. Read the topic System Security - V10R10
and earlier for instructions on restoring the default SA password.
Refer the topic System Security - V10R10 and earlier for instructions on how to change and reset the SA
Password.
Four persons can simultaneously login and change system settings from the SA mode.
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you can view the command strings that you have keyed in on the LCD display of EON;
you will get prompts and confirmatory messages on the phone's LCD display, in addition to confirmation
tone played to you.
SE Commands
These are number strings to be dialed by the System Engineer on entering the SE mode via a telephone. SE
Commands are unique to the feature/facility being programmed. Hence SE Commands for configuring a particular
feature/facility are presented in description of that feature/facility in this manual.
For a complete List of all SE Commands, refer System Commands.
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If the SE Password you entered is incorrect, you will be played an Error Tone and an Error Message on
EON.
The system accepts and executes the command immediately, but it takes approximately 2 minutes to
save a command. So, it is advisable that you do not turn OFF the system for 2 to 3 minutes after
entering the last command.
128. You may use the default SE password, 1234, if the password has not been changed already. If you have changed the password
and the new password is less than 12 digits, you must dial #* after the new password to indicate end of dialing.
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SA Commands
SA commands consist of a prefix string 1072, followed by the Command string. For example: the SA command for
setting Do Not Disturb for an extension is 1072-001-extension number-1, where 1072 is the prefix string and 001
is the command string.
The Prefix string in the SA Command (1072) can be changed by the installer/System Engineer. However, the
command strings of the SA Command (001 in the above example) cannot be changed.
The command for entering SA mode (1#92) is also non-programmable. Refer Access Codes in the chapter
Features and Facilities.
To know how to use change feature settings with SA Commands, please refer the description of individual features
under Features and Facilities.
For a complete list of SA Commands, refer SA Commands.
When the ETERNITY is used in the Standard PBX Application, you can enter SA mode only from
extensions which have the features 'SA mode' and/or 'SA Extension enabled in their Class of Service.
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When the feature SA Extension is enabled in the Class of Service of an extension, the extension will
always be in SA mode. You do not need to enter SA mode by dialing the SA password. You can enter
the SA mode by dialing the SA command prefix string.
When the feature 'SA Mode' is enabled in the Class of Service of an extension, dialing of the SA
Password is required to enter the SA mode. SA Commands can be dialed only after successfully
entering the SA Mode.
There is no restriction on the number of persons who can simultaneously enter and operate from the
SA mode using a telephone.
You can also exit the SA mode before the SA Mode Timer expires by dialing this command. If the SA
Mode Timer is set to 000 minutes, you can exit the SA mode only by dialing the command 1#92.
129. You may use the default SA password, 1111, if the password has not been changed already. If you have changed the password
and the new password is less than 12 digits, you must dial #* after the new password to indicate end of dialing.
When you dial this command, the system will check if the facility 'SA Mode' is enabled in the Class of Service allowed to the extension from which you have dialed this command. If the SA mode is not allowed, an Error Tone will be played. The Error Tone will be
played also when the SA password is entered incorrectly.
If the facility 'SA Extension' is enabled in the Class of Service of the extension you are using, you can skip this step and directly dial
the SA Command (1072-<Command String>).
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To do this:
Dial 1#91-SE Password to enter SE mode from a DKP/SLT.
Dial 2118-Time
Where,
Time is from 001 to 255 minutes.
For example, to set log out time to 45 minutes, dial 2118-045.
Press 'Enter' key to save setting, if using EON.
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Using Jeeves
To be able to access Jeeves,
the Master Ethernet Port of ETERNITY must be connected with a stand-alone PC or in a LAN.
a web-browser, either Internet Explorer 7 or later or Mozilla Firefox 3.5.1 or later, must be installed on the
PC.
Refer the installation instructions for connecting a standalone/LAN as applicable to your model of ETERNITY
(Installing ETERNITY ME, Installing ETERNITY GE, and Installing ETERNITY PE).
When the system is connected to a standalone/LAN, change the IP Address and Subnet Mask so that the IP
Address of the PC and the Master Ethernet Port of ETERNITY do not clash and the PC and the Master
Ethernet Port of ETERNITY are in the same Subnet.
Refer the instructions provided under the sub-topic 'Changing IP Address and Subnet Mask of the Master
Ethernet Port' under the topics Installing ETERNITY ME, Installing ETERNITY GE, and Installing
ETERNITY PE as applicable to your model.
1. Open the browser (Internet Explorer/Mozilla Firefox) on the PC (Standalone or LAN PC) to which the
ETERNITY is connected.
2. Enter the current IP address of Master Ethernet Port of ETERNITY on the address bar or the browser.
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English
Italian
Spanish
French
German
Portuguese
The language that you select on the 'Welcome' page or on any of the Login pages is valid for the
session only. The default language will be applied on next login.
When you select the Configuring Region for the country in which ETERNITY is being installed, the
system will load the country-specific default settings and automatically select the local language of the
country, if the local language is among the above listed languages. The default local language set on
selecting the Region will be applied for every login session, unless you select another language as the
default local language.
The default local language set on selecting the Region can also be changed from the System
Parameters page of Jeeves.
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Click the desired link to configure its parameters or click the Wizard icon
Wizard-Standard PBX. The Wizard will open, you may navigate further.
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To exit the SE programming mode in Jeeves, click the Logout button on top right of the browser window.
Each login session into the SE Mode is set to 60 minutes by default. So, the login session will expire at
the end of 60 minutes. The duration of the login session can be extended or shortened according to
your preference by changing the settings for the 'Web Configuration Timer'. Refer the topic Changing
Login Session Time Out of Jeeves.
It is possible for four users to simultaneously log into the System Engineer Mode of Jeeves. It is also
possible to log out all of these users at once or log out any of these users selectively. Refer the topic
Logging Out Users from Jeeves.
If you are using the model ETERNITY ME10S, with the Redundancy Option for the Master Card, you
will have two Master Cards; each card is loaded with the GUI Jeeves. Now, the Jeeves of the second
card can be accessed and used in the same way as described above. Make sure that the configuration/
settings of the active card are updated in the second card.
In Password enter the SA user name (if assigned, not mandatory) and the SA password.
You may use the default SA password, 1111, if the password has not been changed already. If the
password has been changed, use the new password.
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On successful login, the Feature Menu for the SA mode is displayed as links on the left side panel as
navigation links on the page and on every SA programming page.
Click the link of the desired Feature to open the page. Program the feature and click Submit at the bottom
of the page to save changes. Click Default to reset the default values.
To exit the SA programming mode in Jeeves, click the Logout button on the bottom right of the browser
window.
Configuration of ETERNITY using FTP is meant for Installers who want to complete system configuration at their
end or are unable to configure the system on-site at the customer's end.
The advantage of using FTP for configuring the system is that Installers can complete the entire system
configuration as per their customer's requirement at their end, copy these configuration files, and then upload these
configuration files on to the customer's system.
Further, once the configuration file has been created, Installers only need to make the desired changes in the
relevant files and upload the updated files again.
Before you configure the system using the FTP server, make sure you have completed the following tasks at your
end:
Now, copy the system configuration you have just completed for the users system on a CD or a Pen Drive
using the embedded FTP server. To do this, you may use their Windows FTP or FireFTP. See
Configuration Upload for step-by-step instructions.
If you have multiple customers and you want to configure your customers' systems using FTP, you are
recommended to tag the names of the configuration folders you create for your customers with some
identification, like name and date.
Open the web browser (Internet Explorer/Mozilla Firefox) on the computer connected to the customers
ETERNITY.
Now, copy the system configuration from the CD/ Pen Drive onto the users system using the embedded
FTP server. To do this, you may use their Windows FTP or FireFTP. See Configuration Upload for stepby-step instructions.
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The FTP can be used to store back-up of configuration files, SMDR, and System Software files. Refer the
topics Backup-System Configuration, Backup-SMDR and Backup-System Software to know more.
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standard computer with Window and NT operating system) using a crossed communication cable with 9pin D-type female connector on both ends.
2. Enable the option Communication Port fro Programming in the ETERNITY for serial communication, select
the COM Port for programming, and program the COM Port Attributes of ETERNITY and that of the PC.
You can do this in two ways:
a. Using Jeeves
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Go to the option Enable Programming through Comm. Port and select the checkbox.
In the Communication Port for Programming box, select the port you want to use for
programming: Comm. 1, Comm. 2. By default no COM Port is selected.
Change the parameters of the selected COM Port - Speed, Data Bits, Parity, Stop Bits, Flow
Control, DSR Sensing - to match the values of the same attributes of the COM Port of the PC. Refer
the topic Communication Ports to know more.
b. Using SE Commands
Open Notepad (point cursor on Start Programs Accessories; click NotePad, a new, untitled file will
open).130
Each command string should be prefixed with the character '^' (press shift+6)
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After you have entered the SE commands, save the text file containing the commands with an
appropriate file name with.txt extension. Name the file such that you will know what commands it
contains. For example: If you have entered SE commands to configure SLT ports, save the file as SLTport-parameters.txt.
You may save the command file for future reference and use. You may also print the command files for
verification and future reference.
2. Open HyperTerminal (point cursor on Start Programs Accessories Communication; click
HyperTerminal icon.
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3. You may enter your location information, or choose cancel to ignore and proceed further.
5. Enter the name you desire to give to the new connection and select an icon. Click OK.
If you have ignored the location information in Step 3, you may be prompted again. You may ignore the
prompt.
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6. The Connect To window will pop up. Select the COM Port you have assigned for configuring ETERNITY
7. Configure the Port Settings - Bits per second, Parity, Data Bits, Stop Bits, Flow Control. Remember, these
values must match with the values you entered for the COM Port of the ETERNITY to which this COM Port
is connected. Click Apply and click OK.
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9. In the ETERNITY Properties window, change the option Emulation to VT100 (default: Autodetect).
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10. In the same window, click ASCII Setup. Under ASCII Sending options, click to enable Echo typed
11. Click Transfer on the task bar of HyperTerminal window. Select the option Send Text File from the drop
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12. Browse (from Look in: ) to the location where you saved the text file and select the desired text file by
13. The computer is now sends the command file character by character. The ETERNITY receives these
If the PBX accepts the programming, it responds by sending the character Y. If the PBX does not accept it,
it responds by sending character N.
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feature/facility you programmed using HyperTerminal. For example, click SLT Parameters page and check
if the command strings you uploaded in the text file SLT-port-parameters.txt are reflected here.
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Help Text
There is a help text to explain the parameters on each screen. You may use this help text to guide you in entering
the information.
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The Buttons
The Wizard has the following buttons:
Next: Clicking this button will cause the existing values of the parameters in a page to be submitted and
takes you to the next page of the Wizard.
Submit: Clicking this button will cause the parameters configured in the page to be submitted, but will not
take you to the next page. The 'Submit' button is to be used when there are multiple pages for configuring
a facility, for example, configuring extensions, CO trunks. Instead of navigating all the pages, you can
access the desired page, make changes and submit them. Similarly, if you want to make any changes
post-installation, instead of navigating through all pages of the Wizard and clicking Done to effect the
changes, you can reach the desired page, make the necessary changes and submit the changes.
To navigate further, you must click the Next button.
Skip: Clicking this button will take you to the next page, without making any changes to the current page.
The Skip button is to be used when the Wizard is used post-installation to reach a page which is to be
modified, without changing the existing configuration settings on other pages.
Undo: Clicking this button refreshes the page with the parameters configured in the system. You may use
this button if you are not sure about the values you entered or have entered incorrect values and wish to
start all over again. This button is available only on select pages of the Wizard.
Help: Clicking this button on a page opens a new window, containing Help Text for that page. The window
can be maximized.
Default: Clicking this button will populate all the fields of the page with the default values. Use this button
when you want to assign default values to parameters.
Clear: Clicking this button will cause the values of all the fields on the page to be cleared. Use this button
to make corrections or start entering the values all over again.
Exit: Clicking this button will exit the Jeeves amidst of an activity.
The changes you make in the system configuration under Configuration will not be reflected in the Quick
Installation Wizard.
You are advised to either use the Wizard only during installation and to make modifications post
installation OR use the Wizard only for basic configuration (first time installation) and for making
subsequent changes click the desired link under Configuration.
Open Jeeves (see Using Jeeves earlier in this chapter for instructions). The Login page will open.
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To configure the Basic Settings, select the Use Quick Installation Wizard-Standard PBX.
Jeeves allows 4 persons to simultaneously log in as System Engineer. When you are logged into the
Wizard, it is recommended that no other person logs into Jeeves as System Engineer.
Configuring Parameters
The Wizard gives you two options for configuring the system:
Selective Configuration - This option allows you to choose what you want to configure and click the
related links on the left navigation bar of the Wizard page. You are recommended to use this option for
making the desired changes post-installation.
Navigate Wizard - In this option, the Wizard will lead you logically, step-by-step through the configuration
of the desired parameters. Click the Next button to navigate through the Wizard. You are recommended to
use this option when installing the system for the first time.
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Region
In the Region list, select the country where the system is installed, and then click Next.
Selecting the Region will instruct the system to load the default values of features/facilities according to
the country specific requirements.
The default Region is India.
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This will load the default settings in the system. You will get an alert, asking you, if you want to continue.
Click OK to continue.
The system will restart and load the default values according to the Region you selected on the previous
screen.
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If you are using the Wizard post-installation, do not default the system. This will roll back the entire system
configuration to the default values. Use the Skip button, to move to the next page, without effecting
changes on the parameters of the current page.
System Pre-requisites
Enter the Customer Name. You may enter the name (and address, if desired) of the organization/
enterprise in this field. For example: Prudent Investment, 701 Sunshine Boulevard, Bannerghatta,
Bangalore. The Customer Name can be a maximum of 80 characters.
If you enable the On Site configuration flag, the configuration GUI of ETERNITY, Jeeves, will show the
pages for only those trunks and extension port types that are on board in the system, that is, detected by
the system at Power-On.
In Model Type, select the model of ETERNITY you are configuring. For example, ETERNITY ME16S.
When you select the Model Type, the Wizard will display the different port types according to the system
capacity (the maximum number of ports) supported by the model and variant of ETERNITY you have
selected as the Model Type. For example, if you selected ETERNITY GE6S as Model Type, the maximum
number of CO Ports will be 96 and the maximum number of DKP extensions would be 96 ports. If you
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selected ETERNITY PE6SP, the maximum number of CO Ports will be 16, and the maximum number of
DKP ports will be 48.
Select the number of ports to be used for each Port Type (CO, DKP, SLT, Mobile, T1E1 PRI, BRI, VoIP,
SIP Trunks, SIP Extensions) from the respective boxes. For example, if you want 8 CO Trunks, 24 DKP
extensions, and 128 SLT extensions to be used, select the same numbers from the box.
If you want to use voice mail, or connect any device to the Digital Input Port or the Digital Output port,
select Yes.
If you have selected the System Pre-requisites, navigate to the next page of the Wizard by clicking the
Next button.
You can set the Wizard to display only those trunk and extension port types which are present in the
system. This can be done by enabling the 'On Site Configuration' flag.
When the On Site Configuration flag is enabled, the Wizard will display the only those ports that are
present in the system (detected by the system at Power-ON) on this page.
For example, the system detects ETERNITY ME SLT32 card at power-on, so the maximum number of
SLT ports in the box will be 32. If you want only 20 SLTs to be used, select 20 in the box.
The Wizard will now consider that there are only 20 SLTs in the system and will modify the relevant
pages accordingly.
To cite another example, if your system is ETERNITY PE3SS, which does not support ISDN BRI or
PRI, the fields of these port types and ISDN Terminals on this page are displayed as non-editable.
Similarly, the field of the Door Phone ports will be editable only if the system has detected a Door
Phone card.
The fields for port types which are not available on-board (detected at Power-On) are displayed as noneditable.
To enable the On site Configuration flag you must login as System Engineer.
The Wizard does not provide for the port types Magneto and E&M as these are less commonly used. If
your system has Magneto card and/or E&M card installed, you must configure the related trunk
parameters under Configuration.
It is recommended that you enable the On Site Configuration flag when you are configuring the system at
the installation site.
Assign extension numbers and names for the Door Phone, SLT, DKP and ISDN Terminal ports as desired
on this page. You can also program Access Codes from this page.
The desired Extension Number can be 6 digits long. The digits 0 to 9, # and * are allowed.
The desired Extension Name may be the name of the person who will use the extension. The name can be
a maximum of 18 characters.
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When you change the extension numbers, make sure that they do not clash with any other Feature Access
Code in the dialing phase. To know more, refer the topic Access Codes.
The Wizard pops up an alert about clashing extension numbers and prompts for you to resolve the
conflicting numbers.
The Wizard will display only those ports available in the system and the number of ports you have defined
earlier in the System Pre-requisites page. The Wizard automatically detects the Hardware Slot and Port
Offset of the ports and assigns them to Software Ports. The Wizard also assigns the default extension
numbers to the ports, but leaves the extension names blank.
The default extension numbers for the above port types are:
Blank for Door Phone 1 to 3.
2001 to 2512 for SLTs 001 to 512.
3001 to 3128 for DKPs 001 to 128.
Blank for ISDN Terminals 01 to 64.
SIP Extensions are a licensed feature. To configure the SIP Extensions, under Configuration click VoIP
Configuration and then click SIP Extension Settings.
To assign the extension numbers and names all over again, click the Clear button on this page.
To default all extension numbers, click the Default button on this page.
Access Codes
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Change the Feature Access Codes, if required. Feature Access Codes may consist of single digits or a
sequence of a maximum of 6 digits. Digits 0 to 9,# and * are allowed. The default Access Codes that
appear on the screen are country-dependant. Also, refer Access Codes to know more.
If you assign the same Access Code to more than one feature, the Wizard will pop up a Total Conflict
message and ask you to resolve the conflicting codes. It will not allow you to submit until you have
resolved the conflict. If you assign an Access Code which has the same first digit as another Access Code,
the Wizard will pop up an alert about the clashing numbers. You must resolve the clashing numbers by
clicking OK.
You may choose to resolve the clashing number by clicking Cancel or you may ignore the alert and
continue programming by clicking OK.
Program names for Trunk ports for easy identification of the trunks. The name may consist of a maximum
of 18 alphanumeric characters.
The Wizard automatically detects the Hardware Slot and Port Offset of the trunk ports and assigns them to
Software Ports. The Wizard also assigns the default trunk port names along with their respective software
port numbers. For example, if a Two-wire Trunk is assigned software port number 001; the name will be
displayed as CO -001. Thus CO trunks are named as CO-001 to CO-128, Mobile Ports as MOB-001 to
064, BRI Ports as BRI-001 to BRI-032, and so on.
The Wizard will display only those Trunk port types available in the system and the number of Trunk ports
you have defined for each trunk port type earlier on the System Pre-requisites page.
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Day-Night Time
Ask your customer about their working days and working hours (24 hours format) and program the
parameters accordingly. The Wizard considers the working hours you have selected as Daytime and the
remaining hours as Night Time (non-working hours). The default working hours are from 09:00 to 18:00.
The default Working Days are Monday to Saturday.
Day-Night Time is assigned to Trunks and Extensions, so that they behave differently according to the
Time of the day. For example, the customer may want the system to route trunk calls to the security
personnel when the office is closed, or deny certain extensions access to outgoing long distance calls
during non-working hours and days, or to play a different greeting message to callers on holidays.
This parameter is based on Time Table 1131, which is assigned to Trunks and Extensions by default.
The Wizard simplifies the assignment of Time Tables to trunks and extensions, requiring you to
program only the working hours and working days, instead of prompting you to define the non-working
hours and break hours. The Wizard automatically applies the working hours and days you have
programmed to time table-dependent facilities and features such as Class of Service, Toll Control,
Outgoing Trunk Access, etc.
Skip this page if you feel that the requirements of your customer are not served by this parameter132.
Configure the Time Tables and related features like Class of Service, Toll Control, Outgoing Trunk
Access, etc. under Configuration.
131. A Time Table is a schedule of the three time zones (working hours, break hours, non-working hours) for a week. There are 8 different Time Table templates to select from. Different Time Tables can be assigned to different trunks and extensions. Refer the section Time Tables to know more.
132. For instance, the working hours are not the same throughout the week.
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If you want to change the working hours and days at a later stage while navigating the Wizard, you may
use the Back button of your browser to return to this page and make the changes.
Number Patterns
This parameter is related to the Toll Control Feature, which allows you to define a particular calling permission for
each extension, referred to as Call Privilege. A Call Privilege allows the extension to call certain areas and restricts
it from calling others. The extension can also be restricted from the dialing of specific telephone numbers. The
ETERNITY supports different types of Call Privileges, these are: No Calls, All Calls, Local Calls, Regional Calls,
National Calls and International Calls.
On this page, you are required to define the number strings which the system should consider as Local
Numbers, Regional Numbers, National Numbers and International Numbers.
In the field Numbers starting with, you may enter only the first digit of the number string, or a part of the
string, or the complete number string.
In the field except, enter the number strings which you want to restrict from being dialed out.
Each number string you enter must not exceed 16 characters. Separate number strings with comma.
Do not provide <space> between commas and numbers.
The Call Privilege Type No Calls and All Calls does not require any number pattern programming.
The Call Privilege Type Limited Calls allows the dialing of only specific telephone numbers. It can be
programmed only under Configuration. To know more, refer the feature description for Toll Control.
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'Operator'
It is possible to change the sequence of extensions on the right side box using the Up/Down Arrow.
To remove/delete any extension number from the right box, select the extension number and press the
delete key from the keyboard.
A maximum of 16 extensions can be selected as Operator extension.
You will be shown an alert if you program more than 16 extensions. You may select and delete the excess
extensions or choose to ignore the alert by clicking 'OK' or closing the alert dialog box. Regardless of this,
the system will consider only 16 extensions. You can change the sequence of the extensions on the right
hand side using the Up and Down arrows.
When you click OK, all the extensions you selected will appear sequentially, separated by comma in the
order you selected the extensions.
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Define the Class of Service for the Operator extension. Select the features to be allowed to the Operator
extension during the day and at night by selecting the checkboxes of the features listed in the table.
Configure the Toll Control for the Operator extension for day time and night time. Select the type of calls to
be allowed during the day, and the type of calls to be allowed during the night. By default, all types of calls
are allowed during the day and at night.
Select the Outgoing Trunks for the Day Time (the trunks through which calls are to be routed during the
day). If you want the system to use Least Cost Routing, select the check box.
When you double click this field, the list box opens. Select the outgoing trunks from the left side box.
As you can see, the same trunk types are arranged sequentially, regardless of their hardware port location.
If you select trunks of the same type in sequential order, for example: CO-001, CO-002, CO-003, BRI-001,
BRI-002 and MOB-002, the same trunk type will be grouped in one OG Trunk Bundle: CO-001, CO-002,
CO-003 will be OGTB #1, BRI-001 and BRI-002 will be grouped as OGTB#2, and MOB-002 as OGTB#3.
These are default trunk names. If you have changed the trunk names (see Naming of Trunks), the name
will appear here, instead of the default trunk names.
If you select the trunks of the same type in a non-sequential order, for example, CO-001,
MOB-001, BRI-001 and CO-002, four OGTBs will be formed with CO-001 as member of OGTB#1, MOB001 as member of OGTB#2, BRI-001 in OGTB#3 and CO-002 in OGTB#4. So, despite two same trunk
types being selected (CO-001 and CO-002) they are grouped in separate OGTBs.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
It is possible to change the sequence of the trunks on the right side box using the Up/Down Arrow.
To remove/delete any trunk number from the right box, select the trunk number and press the delete key
from the keyboard.
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Select the Outgoing Trunks for the Night Time (the trunks through which calls are to be routed during the
night). If you want the system to use Least Cost Routing, select the check box.
Follow the same instructions for selecting the trunks from the list box as described in the previous step.
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Set the Priority133 for the Operator extension, by selecting the desired Priority Level from 1-9 from the
box. By default the priority level for the Operator extension is set to level 5.
Extensions
Create a User Profile for extensions. A User Profile consists of Class of Service (COS), Toll Control and
Trunk Access to be assigned to an extension during day time and night time.
133. Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3, 4...to 9. With 1 being the lowest priority and 9
being the highest priority. The calls from an extension with higher priority have preference in call landing. When an extension with
higher priority calls another with lower priority, a triple ring is placed on the called extension, and the call will land first on the extension when there are multiple incoming calls on the extension with lower priority.
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The Wizard makes configuration of the extensions easy with User Profiles. Instead of configuring each
extension individually, you can group together extensions that are to be allowed the same COS group, Toll
Control, Trunk Access (Outgoing Trunk Bundle and Outgoing Trunk Bundle Group) and Priority in a single
User Profile.
User Profiles meet the requirements of organizations that desire to assign a different set of features to their
personnel according to their position in the organization, like senior managers, field executives,
administrative assistants etc. In such cases, each of these groups of users can be assigned a different
User Profile.
As many as 8 different User Profiles can be programmed using the Wizard.
You can name each User Profile such that it reflects the extension user group to which it is assigned. For
example, you may rename User Profile-1 created for managers as 'Manager', User Profile-2 created for
field executives as 'Executive', User Profile-3 created for administrative assistants as 'Admin'.
To change the label of the User Profile, click the 'Pencil' icon.
A prompt will pop up, asking you if you want to rename the User Profile Name. Enter the desired name in
the field. You can enter a maximum of 18 characters as name. Click 'OK'.
The name you entered will appear in place of 'User Profile' number.
To assign the User Profile to the desired extensions, double click the box.
The Wizard will display the configured extension numbers on the left box. Place your cursor on the desired
extension numbers and click Select >>. The selected extension numbers will appear on the right box.
You can select the extensions for the User Profile after configuring the other parameters of the User
Profile.
Define the Class of Service for the User Profile for day time and night time by selecting the check boxes of
the features you want to allow in the Class of Service.
Define the Toll Control for the User Profile for the day time and night time by selecting the desired Toll
Control level in the box.
The Toll Control levels on this page are based on the number lists you have programmed earlier on the
Number Patterns page of the Wizard.
Select the Outgoing Trunks for the Day Time (the trunks through which calls are to be routed during the
day). If you want to the system to use Least Cost Routing, select the check box.
When you double click the field, the list box opens. Select the outgoing trunks from the left box.
As you can see, the same trunk types are arranged sequentially, regardless of their hardware port location.
If you select trunks of the same type in sequential order, for example, CO-001, CO-002, CO-003, BRI-001,
BRI-002 and MOB-002, the same trunk type will be grouped in one OG Trunk Bundle: CO-001, CO-002,
CO-003 will be OGTB #1, BRI-001 and BRI-002 will be grouped as OGTB#2, and MOB-002 as OGTB#3.
If you select the trunks of the same type in a non-sequential order, such as: CO-001, MOB-001, BRI-001
and CO-002, four OGTBs will be formed with CO-001 as member of OGTB#1, MOB-001 as member of
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OGTB#2, BRI-001 in OGTB#3 and CO-002 in OGTB#4. So, despite two same trunk types being selected
(CO001 and CO002) they are grouped in separate OGTBs.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
It is possible to change the sequence of trunks on the right side box using the Up/Down Arrow. You can
also delete a particular trunk from the right box.
Select the Outgoing Trunks for the Night Time (the trunks through which calls are to be routed during the
night). If you want to the system to use Least Cost Routing, select the check box.
Follow the same instructions for selecting the trunks from the box as described in the previous step.
Set the Priority134 for the extension, by selecting the desired Priority Level from 1-9 in the box. By default
the priority level for the extension is set to level 5.
You may set a different Priority Level in each User Profile, depending on the requirement of the extension
users, whose extension the User Profile is to be assigned. For example, you may set a higher Priority level
to the User Profile to be assigned to Managers.
Rename the label, if required, by clicking the 'Pencil' icon. Repeat all the steps described above to define
Class of Service, Toll Control, Trunk Access and Priority.
Click Submit to save changes. Repeat the same steps to program another User Profile.
When you have finished programming the desired number of User Profiles, click Next button to navigate
the Wizard further.
The User Profiles 1-8 use the following resources in the system:
Station Basic Feature Template number 2 to 9
Class of Service Groups 2 to 17.
Outgoing Trunk Bundle Groups 2 to 17
Outgoing Trunk Bundles 5 to 68.
134. Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3, 4...to 9. With 1 being the lowest priority and 9
being the highest priority. The calls from an extension with higher priority has preference in call landing. When an extension with
higher priority calls another with lower priority, a triple ring is placed on the called extension, and the call will land first on the extension when there are multiple incoming calls on the extension with lower priority. Refer the feature description for Priority to know
more.
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Door Phones
Configure the following door phone parameters for each Door Phone you have connected to the Door
Phone Port of the ETERNITY PE.
Route Door Phone Calls: Select the mode of routing for calls landing on the Door Phone.
If you want the flexibility to have calls routed to a group of extensions or to an external number as you
desire, select Manual.
If you want the system to route the Door Phone Calls automatically to landing destination phone
(extension or external number) according to the time of the day., select As per Schedule,
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Route Door Phone Calls during Day to: Select the radio button of the desired destination External Number or Extensions - on which calls should be landed during the day time.
Route Door Phone Calls during Night to: Select the radio button of the desired landing
destination - External Number or Extensions - for calls during the night time.
Extensions for Day Time: Select the Extension numbers where calls should land during the day.
When you double click this field, the multiple selection box will open. The Wizard will display the
extensions you have configured on the left box. Place your cursor on the desired extension and
click select. The selected extensions will appear on the right box. You can select a maximum of 16
extensions as the landing destination for the door phone calls.
Extensions for Night Time: Select the Extension numbers where calls should land during the
night. To select the extensions, double click the field, the multiple selection box will open. Place
your cursor on the desired extensions appearing on the left side box and click select. You can
select a maximum of 16 extensions. The selected extensions will appear on the right side box.
External Number: Enter the External Number. This number is common for Day time and Night
Time, which means, you cannot have calls routed to different external numbers for the Day Time
and Night Time.
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If you have selected Manually, enter the External Number and select the Extensions from the box.
For the External number you have programmed (manually or scheduled), define the Outgoing
Trunk Bundle Group.
When you click the field To Dial External Number, Use Trunk:, the box will open, displaying the
trunk types present in the system in the left side box. Click the desired trunk type and click Select.
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If you select trunks of the same type in sequential order, for example: CO-001, CO-002, CO-003,
BRI-001, BRI-002 and MOB-002, the same trunk type will be grouped in one OG Trunk Bundle, for
example: CO-001, CO-002, CO-003 will be OGTB #1, BRI-001 and BRI-002 will be grouped as
OGTB#2, and MOB-002 as OGTB#3.
If you select the trunks of the same type in a non-sequential order, for example: CO-001,
MOB-001, BRI-001 and CO-002, four OGTBs will be formed with CO-001 as member of OGTB#1,
MOB-001 as member of OGTB#2, BRI-001 in OGTB#3 and CO-002 in OGTB#4. So, despite two
same trunk types being selected (CO-001 and CO-002) they are grouped in separate OGTBs.
It is possible to change the sequence of trunks on the right side box using the drag and drop action.
You can also delete a particular trunk from the right box.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert
indicating that the system is short of resources.
Select the Use LCR check box to enable Least Cost Routing.
Change the Door Phone Ring Timer, if required. This is the time for which the Door Phone will ring
on the extensions programmed as the landing destination. This Timer has a range of 001 to 255
seconds. By default the Timer is set to 30 seconds.
If you are using a Door Lock in conjunction with the Door Phone connected to this Door Phone,
select the Use Door Lock checkbox to enable.
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Repeat the same steps to program another door phone. If you have no other door phone to program, click
Next to navigate further.
The Wizard uses the following resources for this feature:
Outgoing Trunk Bundle Groups (OGTBG) 20 to 22.
Outgoing Trunk Bundles (OGTB) 77 to 88.
Routing Group 58-66.
Do not change their settings when configuring the system from Configuration.
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Assign Cost Factor to the trunks. This parameter is of relevance only if 'Least Cost Routing' feature is
applied on the CO Trunk port.
On this page, the Wizard displays the number of trunk ports you selected in the System Pre-Requisites
page.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a preference
number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same preference must be
assigned the same Cost Factor. Different trunk types can also be assigned the same Cost Factor. These
trunks are used for routing calls.
Now configure the number of or part of the number and the preferred trunk to route that number. The
number can be a maximum of 16 digits.
Select the cost factor trunk applicable for each number in the order of preference. Select the foremost
preferred Cost Factor trunk in Preference 1, the second most preferred Cost Factor trunk in Preference 2,
the third preferred Cost Factor trunk in Preference 3 and the least preferred Cost Factor trunk in
Preference 4.
While you can create as many as 01 to 99 CPU Groups, and assign extensions to these groups, the
Wizard allows you to create and assign extensions to 01 to 16 CPU Groups. By default, all extensions are
assigned to CPU Group number 99.
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To create a CPU group, click the CPU group number, for example CPU 01, and double click to select the
extensions to be assigned to this group from the box.
Click Submit to save the settings of the CPU group you created. Now, repeat the same steps to create
another group.
If you have finished programming CPU groups, click Next to navigate to the next page of the Wizard.
To know more about this feature, refer the topic Call Pick Up.
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CO Trunks
Configure features of the Two-wire Trunks (CO) connected to the ETERNITY on this page.
The Wizard makes configuration of the CO Trunks easy with CO Profiles. Instead of configuring each CO
Trunk individually, you can group together trunks that are to be assigned the same features - Calling Line
Identification, Trunk Landing Group - in a single CO Profile.
You can program as many as 8 different CO Profiles using the Wizard.
It is also possible to name each CO Profile by the service provider. For example, you may rename CO
Profile-1 created as 'BSNL', CO Profile-2 as 'Reliance' and so on.
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A prompt will pop up, asking you if you want to rename the CO Profile Name. Enter the desired name in
the field. You can enter a maximum of 18 characters as name. Click OK.
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To assign the CO Profile to the desired trunks, double click the field Select Trunks/Apply to Trunks.
The left side box will display the number of CO trunks (along their names) you configured in the System
Pre-requisites page. Place your cursor on the desired trunks and click Select >>. The selected CO trunks
will appear on the right side box.
You can select the trunks for the CO Profile after configuring the other parameters of the CO Profile.
Define the Calling Line Identification Format for the CO Profile by selecting the desired option in the box.
You are advised to consult the service provider in this regard.
In the field Route Calls during Day to, select the landing destination for calls on trunks in the CO Profile
during day time. You may select any option: Operator extension, other extensions, Built-In Auto Attendant
or the Voice Mail Auto Attendant (if available) from the box.
If you select Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the default Voice
Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting and Guidance
Messages) will be applied. Refer Voice Message Applications to know more.
If you select Extensions as the landing destination, select the extensions in the corresponding field.
Double click the field, the list box will open. Select the extensions from the left box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using the Up/Down Arrow.
The selected extension numbers will appear in the field, separated by comma.
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In the field Route Calls during Night to, select the landing destination for calls on trunks in the CO Profile
during night time. Follow the same instruction as in the previous step.
Click Submit to save your settings in the CO Profile. Repeat these steps to create another CO Profile.
If you have finished creating CO profiles and assigning them to trunks, click Next to navigate further.
If there is any trunk you have not assigned a CO Profile, the Wizard will pop-up an alert informing you
about the trunk you have not programmed.
You must complete configuring the parameters for this CO trunk and only then will the wizard allow you to
configure other parameters.
The Wizard uses the following resources for this page:
Trunk Feature Templates 34 to 41.
Routing Groups 34 to 49.
CO Hardware Templates 02 to 09.
Do not make any modifications to them when configuring the system from Configuration.
BRI Trunks
You will reach this page only if your system has detected the presence of BRI Trunks or if you have
specified the number of BRI Trunks to be used in the System Pre-requisites page.
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The ETERNITY supports a maximum of 32 BRI Trunks, depending on the system capacity of your
model.
Using the Wizard, you can configure only the first four BRI Trunks in your system. You can program the
remaining BRI Trunks from Configuration.
Select the mode for routing incoming calls. Incoming calls may be routed according to MSN Numbers, DDI
Numbers or Port-wise.
Route Incoming Calls MSN Number wise: Select this option if you want to route incoming calls
according to MSN numbers135. You can program maximum 6 MSN numbers.
Route Incoming Calls DDI Number wise: Select this option if you want to route incoming calls
according to DDI Numbers136. Select this option only if your extensions are arranged sequentially. If
the extensions are not arranged sequentially (for example, DDI numbers 2630551 to 2630559 are to be
routed to extensions 3001 to 3009) you are recommended to program this parameter from the Full
Programming Access mode. You can program maximum 12 DDI numbers.
Route Incoming Calls Port wise: Select this option if you want to route all incoming calls on the BRI
trunk port to extensions, irrespective of dialed MSN/DDI number.
MSN Number: Enter the MSN Number provided by your Service Provider. This number is used to
route the incoming calls and sent to the ISDN Exchange when making an outgoing call. You can enter
up to 6 MSN numbers using the Wizard.
Route calls during day to: Select where you want to the route the calls during day time. You can route
the calls to the Operator or to a group of extensions or to Built-In Auto Attendant or to the Voice Mail
Card's Auto Attendant137.
If you click the option Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting
and Guidance Messages) will be applied by the Wizard. Refer Voice Message Applications to know
more.
135. MSN numbers is a set of numbers with no logical connection between the numbers themselves. For example, MSN numbers for a
BRI connection could be 2630555, 2634872, 2635098, etc. Up to 8 MSN numbers are provided on a single BRI connection.
136. DDI numbers are a set of numbers arranged sequentially, for example DDI numbers for a BRI connection could be 2630551 to
2630559.
137. The Built-In Attendant offers 5 simultaneous calls, whereas Voice Mail's Auto Attendant offers 16 simultaneous calls.
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If you select 'Extensions', double click the empty field, the multiple selection box will open. Select the
extensions from the left side box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using drag and drop option.
The selected extension numbers will appear in the field, separated by comma.
When NR?: Select an option for what action the system should take when there is no response (NR),
that is, the incoming call is not answered within the DDI Ring Timer (default: 45 sec.) by the destination.
By default, it is set to 'Disconnect'.
When Busy?: Select an option for what action the system should take when the landing destination is
busy. By default, it is set to 'Disconnect'.
Route calls during night to: Select where you want to the route the calls during night time. If you
selected group of extensions as the destination, select the extensions from the list box, as described
above. This may be a different group of extensions than the one selected for calls during day time.
If you select Extensions', you must now select the extensions by clicking the field provided for it. A
maximum of 16 extensions can be selected. Follow the same procedure described above for selection
of extensions for calls during day time.
When NR?: Select an option for what action the system should take when there is no response (NR),
that is, the incoming call is not answered within the DDI Ring Timer (default: 45 sec.) by the destination.
By default, it is set to 'Disconnect'.
When Busy?: Select an option for what action the system should take when the landing destination is
busy. By default, it is set to 'Disconnect'.
The Wizard allows you to select the desired landing extension (Operator/ extensions/ Built-In Auto
Attendant/Voice Mail's Auto Attendant) only for the first 3 MSN numbers. The remaining 3 MSN
numbers can be routed to a single extension only.
So, if you want to route incoming calls to a particular extension during the day time and night time,
enter the MSN numbers in MSN number 4 to MSN number 6.
If you selected DDI Number-wise routing, configure the following parameters for each DDI#:
Root DDI #: Enter the Root DDI number. This number is used to send the DDI number to the ISDN
Exchange. This number is also sent to the Exchange when an outgoing call is made by an extension
which is not assigned DDI number. This is the same as the Pilot Number.
Start DDI #: Enter the first DDI number provided by your Service Provider. More often than not, the
Start DDI# is the same as the Root DDI#.
Total DDIs: Enter the total DDI numbers provided by your Service Provider.
Start Extension#: Enter the number of the first extension from which the DDI number assignment is to
be done.
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For example, your Service Provider has given you DDI numbers 2630550 to 2630560. These are to be
assigned to extension numbers 2001 to 2010. In this case 2630550 will be the Root DDI# as well as
Start DDI#. As 10 DDI numbers are provided to you, the Total DDI# would be 10, and the Start
Extension# would be 2001.
When NR?: Select an option for what action the system should take when there is no response (NR),
that is, the incoming call is not answered within the DDI Ring Timer (default: 45 sec.) by the destination.
By default, it is set to 'Disconnect'.
When Busy?: Select an option for what action the system should take when the landing destination is
busy, during DDI Ring Timer (default: 45 sec.). By default, it is set to 'Disconnect'.
Route Calls during Day to: Select the landing destination for calls on BRI trunk during day time. You
may select any option: Operator extension, other extensions, Built-In Auto Attendant or the Auto
Attendant of the Voice Mail System (if available).
If you click the option Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting
and Guidance Messages) will be applied by the Wizard. Refer Voice Message Applications to know
more.
If you select Extensions as landing destination, you must select the extensions. Double click the field
and select the extensions from the left box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using the Up/Down Arrow.
The selected extension numbers will appear in the field, separated by comma.
Route Calls during Night to: Select the landing destination for calls on BRI trunk during night time.
Follow the same procedure for selecting extensions as described above.
Click Submit to save changes. Repeat the same steps to program the other BRI Trunks.
If you have finished configuring the BRI Trunks, click Next to navigate further.
If there is a BRI trunk you have not configured, the Wizard will pop-up an alert informing you about the
trunk you have not programmed.
You must complete configuring the parameters for this trunk and only then will the wizard allow you to
configure other parameters.
For the BRI parameters, the Wizard uses the following resources:
Entries 01 to 48 of the Incoming Call (IC) Reference Table.
Entries 01 to 48 of the Outgoing Reference Table.
Routing Groups 01 to 25
Trunk Feature Template 01 to 25.
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When programming the system from Configuration, do not modify the settings of these resources. It will
affect the settings made by the Wizard.
The ETERNITY supports a maximum of 8 T1E1PRI Trunks, depending on the system capacity of your
model.
However, using the Wizard you can configure only the first 2 PRI Trunks in the system. You may
configure the remaining Trunks, if applicable, from Configuration.
Change the label of the T1E1PRI Trunk, if required. Click the Pencil icon. A prompt will pop-up. Enter the
desired name in the field of the prompt and click OK. The name you programmed will appear.
You may enter the name of the Service Provider to make the identification of the Trunk easy.
The Wizard will display the Hardware Slot-Port Number of the first T1E1 PRI Trunk the system has
detected in this field.
Select the Signaling Type as applicable: from PRI, RBS, QSIG, and E&M. These are the signaling types
supported by the ETERNITY. By default PRI selected.
Select the Framing Mode, as applicable. By default CEPT1 MF (Auto CRC) is selected as Framing Mode.
Select the ISDN Switch variant. By default ETSI NET5 is selected as the variant.
Select the mode for routing incoming calls. Incoming calls may be routed according to DDI Numbers or
Port-wise.
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Route Incoming Calls Port wise: Select this option if you want to route all incoming calls on the
T1E1PRI trunk port to groups of extensions, without identifying the DDI number.
Route Incoming Calls DDI Number wise: Select this option if you want to route incoming calls
according to DDI numbers. Select this option only if your extension numbers are arranged sequentially.
If the extensions are not arranged sequentially you are recommended to program this parameter from
the Full Programming Access mode.
If your installation uses multiple DDI blocks on the same T1E1PRI connection, you can configure only 2
such DDI blocks using the Wizard. You can configure more DDI blocks on a single T1E1PRI
connection from Configuration.
Root DDI #1: enter the Root DDI # provided by your Service Provider. The Root DDI # is the main
number assigned to the T1E1PRI trunk. It is also known as MSN Number, Pilot Number or Main
Number138.
If your exchange requires area code to be sent with the DDI number, program the root number with
area code. For example: Root DDI # is 2630555 and area code where the PBX is installed is 265, enter
2652630555 in this field.
Route calls during day to: Select where you want to the route the calls during day time. You can route
the calls to the Operator or to a group of extensions or to Built-In Auto Attendant or to the Voice Mail
Card's Auto Attendant139.
If you click the option Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting
and Guidance Messages) will be applied by the Wizard. Refer Voice Message Applications to know
more.
If you selected Extensions, then select the extensions by clicking the empty field. A multiple selection
box will open. Select the extensions from the left box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right box using drag and drop option.
The selected extension numbers will appear in the field, separated by comma.
When NR?: Select an option for what action the system should take when there is no response (the
incoming call is not answered within the DDI Ring Timer, set by default to 45 seconds) by the
destination. By default, it is set to 'Disconnect'.
When Busy?: Select an option for what action the system should take when the landing destination
is busy (for the duration of the DDI Ring Timer; default: 45 seconds). By default, it is set to
'Disconnect'.
138. This number will be used to prepare the DDI number to the Exchange (Reverse DDI) when an Outgoing call is made from the
ETERNITY by the Extension assigned DDI Number. Also, this number will be sent to the Exchange without any modification when
an Outgoing call will be made by an extension which is not assigned DDI number.
139. The Built-In Attendant offers 5 simultaneous calls, whereas Voice Mail's Auto Attendant offers 16 simultaneous calls.
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Route calls during night to: Select where you want to the route the calls during night time from the
options: Operator or to a group of extensions or to Built-In Auto Attendant or to the Voice Mail Card's
Auto Attendant.
If you selected group of extensions as the destination, select the extensions from the box, as described
above. This may be a different group of extensions than the one selected for calls during day time.
If you selected Extensions, you must now select the extensions by clicking the field provided for it. A
maximum of 16 extensions can be selected. Follow the same procedure described above for selection
of extensions for calls during day time.
When NR?: same as described above for calls during day time.
When Busy?: same as described above for calls during day time.
Start DDI#: Enter the first DDI number given by the Service Provider. The Start DDI Number may be
the Root DDI number, as seen in most of the cases.
Total DDIs: Enter the Total DDI numbers. Ask your Service Provider.
Start Extension#: Enter the number of the first extension from which the DDI assignment is to be
done.
For example, your Service Provider has given you DDI numbers 2630551 to 2630560. These numbers
are to be routed to extensions 2001 to 2010. In this case, the Root DDI number as well as the Start
DDI# will be 2630550. As 10 DDI numbers are used, enter Total DDI numbers =10 and Start Extension
# = 2001.
When NR?: same as described above for calls during day time.
When Busy?: same as described above for calls during day time.
Route Calls during Day to: Select the landing destination for calls on T1E1PRI trunks during day
time. You may select any option: Operator extension, other extensions, Built-In Auto Attendant or the
Auto Attendant of the Voice Mail System (if available).
If you click the option Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting
and Guidance Messages) will be applied by the Wizard. Refer Voice Message Applications to know
more.
If you selected Extensions as landing destination, you must select the extensions. Double click the
field and select the extensions from the left box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using the Up/Down Arrow.
The selected extension numbers will appear in the field, separated by comma.
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Route Calls during Night to: Select the landing destination for calls on T1E1PRI trunks during night
time. Follow the same procedure for selecting extensions as described above.
Click Submit to save changes. Repeat the same steps to program the second T1E1PRI Trunk.
If you have finished configuring the T1E1 PRI Trunks, click Next to navigate further.
Mobile Trunks
You will reach this page only if your system has detected the presence of the Mobile Card in the system
(as the On-Site Configuration flag is enabled) or if you have specified the number of Mobile Ports
Used earlier in the System Pre-requisites page of the Wizard.
ETERNITY supports up to 64 Mobile Ports, depending upon the system capacity of your model.
The Wizard makes configuration of the Mobile Trunks easy with Mobile Profiles. Instead of configuring
each Mobile Trunk individually, you can group together trunks that are to be assigned the same
features in a single Mobile Profile.
You can program as many as 4 different Mobile Profiles using the Wizard.
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Change the label of the Mobile Profile, if required. Select the desired Mobile Trunk tab and click the Pencil
icon. A prompt will pop-up. Enter the desired name in the field of the prompt and click OK. The name you
programmed will appear. For example rename Mobile Profile 1 as 'Vodafone'. Naming the profile with the
Service Provider's name makes identification of the Trunk on which this profile is applied easy.
Select the mobile trunks on which this Mobile Profile is to be applied. It is also possible to configure all the
other parameters and then select the mobile trunks.
To select the desired mobile trunks, double click the box. The Wizard will display the number of Mobile
Trunks you specified in the System Pre-requisites page on the left side box. All the Mobile Trunks appear
in this box and are arranged sequentially in the increasing order of their software port number.
Place your cursor on the desired trunks and click Select>> button. The selected Mobile Trunks will appear
on the right side box. It is also possible to select a range of trunks at a time pressing the 'SHIFT and 'Down'
arrow keys.
You must click Submit after you enter the new PIN. Wait for 5 seconds, and then refresh this page to
view the new SIM PIN.
Select the extensions to which the incoming calls on the Mobile Trunks should be routed.
In the field Route Calls during Day to, select the landing destination for calls on Mobile trunks during day
time. You may select any option: Operator extension, other extensions, Built-In Auto Attendant or the Auto
Attendant of the Voice Mail System (if available).
If you select the option Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the default
Voice Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting and
Guidance Messages) will be applied by the Wizard. Refer Voice Message Applications to know more.
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If you click the option Extensions as the landing destination, select the extensions in the corresponding
field. Double click this field, the multiple selection box will open. Select the extensions from the left side
box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using the Up/Down Arrow.
The selected extension numbers will appear in the field, separated by a comma.
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In the field Route Calls during Night to, select the landing destination for calls on Mobile trunks during
night time. Follow the same instruction as in the previous step to select extensions as the landing
destination.
Click Submit to save your settings in the Mobile Profile. Repeat these steps to configure another Mobile
Profile.
If you have finished configuring the Mobile Profiles, click Next to navigate further.
If there is any mobile trunk you have not assigned a Mobile Profile, the Wizard will pop-up an alert
informing you about the trunk you have not programmed. You must complete configuring the parameters
for this trunk and only then will the wizard allow you to configure other parameters.
VoIP Network
You will reach this page only if your system has detected the presence of the VoIP Card in the system
(as the On-Site Configuration flag is enabled).
In case the On-site Configuration flag is disabled, you must specify Number of VoIP Ports Used in
the System Pre-requisites page of the Wizard to be able to reach this page.
Each VoIP card has a WAN Port and a LAN Port. The Wizard covers only the WAN Port parameters.
You can program LAN Port parameters from Configuration.
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Regardless of the number of VoIP Ports supported by your model of ETERNITY and the number of
VoIP ports you have defined in the System Pre-requisites page, the Wizard allows you to configure
only the first four VoIP ports in your system.
Configure the Network Parameters of each VoIP (WAN) Port. The system will automatically detect the
Hardware Slot and Port Offset and display the same for each VoIP Port.
Change the label of the VoIP (WAN) Port, if required. This will make identification of the port easy.
Click the Pencil icon and enter the name of your choice in the blank field of the prompt that will pop up on
your screen. Click OK.
In the Network Connection Type box, select the IP addressing scheme of the network to which the VoIP
(WAN) Port is connected: Static, DHCP, PPPoE. Ask your LAN administrator for this information. Select
the appropriate radio button.
If your network connection is DHCP, skip the next two steps and select the DNS connection type.
The DHCP server on your network will automatically assign the IP Address and Subnet Mask, Gateway
Address and other parameters to the VoIP Port.
If your network connection is Static, enter the following information from your LAN Administrator
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If your network connection is PPPoE, enter the following information from your Service Provider:
Enter the Router's Public IP Address. This address will be used in SIP messages if the parameter
Source Port IP Address on the page Configuring SIP trunks is configured as Router's Public IP
Address.
Click the Use SIP Port fetched using STUN check box, if you want to use the port number fetched using
STUN in the SIP message.
This parameter should be disabled if you are using Port-Forwarding in the Router for SIP messages.
Enter SIP UDP Port. This is port on which the ETERNITY (VoIP Port) listens for SIP messages. The VoIP
Port also uses this port to send SIP messages to the remote peer. The default value of SIP UDP Port is
5060.
Enter RTP Listen Port. This port defines the port on which ETERNITY (VoIP Port) listens for RTP
packets. The system also uses the port as source port for sending RTP packets to the remote peer.
Click Submit to save the configuration of the VoIP (WAN) Port. Now, repeat the same steps to configure
another VoIP (WAN) Port.
If you have finished configuring all VoIP ports, click 'Next' to navigate to further.
SIP Trunks
You will reach this page only if you have specified Number of SIP Trunks Used on the System Prerequisites page.
SIP Trunks are to be configured only if you are using Internet Telephony Service Providers for VoIP
calls.
The number of SIP Trunks supported by ETERNITY varies according to its model. ETERNITY ME
supports 32 SIP Trunks. ETERNITY GE supports 16 SIP Trunks and ETERNITY PE supports 4 SIP
Trunks.
You can configure all SIP Trunks on a single VoIP port, or if you have installed more than one VoIP
Card, you may program the IP Trunks on the VoIP ports in a distributed manner. For example, if you
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have installed four VoIP Cards in ETERNITY ME, you can program either 8 SIP Trunks each on the 4
VoIP ports OR program 12 SIP Trunks on the VoIP Port of the first card, 8 on the VoIP Port of the
second card, 6 SIP Trunks each on the VoIP Ports of the third and fourth card.
The Wizard however, allows you to configure only 4 SIP Trunks (even if you have specified more than
4 SIP Trunks being used on the System Pre-Requisites page). You can program the remaining SIP
Trunks from Configuration.
Assign a VoIP Software Port to this SIP trunk in the field VoIP S/W Port #.
You may change the label of the SIP Trunk (rename 'SIP Trunk 1'), if required. You may use the name of
the ITSP to make identification of the SIP Trunk easy.
Click the Pencil icon. A prompt will pop up, asking you if you want to change the name of this SIP Trunk.
Enter the desired name in the blank field. Click OK. The new name will appear instead of the SIP Trunk
number.
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Select the Enable SIP Trunk check box to enable the SIP Trunk.
Enter the SIP ID provided by the ITSP. This is the ID which callers will use to call this SIP Trunk. The SIP
ID may be a number or text.
Enter the Proxy/Registrar Server Address and the Registrar Server Port provided by the ITSP. The
Registrar Server Address can be an IP Address or domain. The Registrar Server Listening Port ranges
from 1024 to 65535. The default Registrar Server Port is 5060.
Enter the Authentication ID (User ID) and Password provided by the ITSP.
Select the Enable Outbound Proxy checkbox, if the ITSP who provided this SIP Trunk has a SIP
outbound server to handle voice calls. By default Outbound Proxy is disabled.
If you have enabled Outbound Proxy, enter the Outbound Proxy Server Address and the Server Port
provided by the ITSP. This can be an IP Address or Domain name.
Set the desired Vocoder Preference for this SIP Trunk. Vocoders are the various Voice Codecs used to
compress the data in RTP packets for optimum use of bandwidth and for ensuring voice quality. You can
set 7 Vocoders options in the order of preference for this SIP account.
Select the DTMF Option. The DTMF option you select will determine how the DTMF digits will be sent
over the IP Network, when a DTMF digit is pressed. ETERNITY VoIP Card supports three DTMF options:
RTP (RFC 2833), SIP Info, and InBand. Select the appropriate option. By default RTP (RFC 2833) is
selected.
Select the appropriate Fax Type. The ETERNITY VoIP card supports T.38 (UDPTL), T.39 (RTP) and
Pass-Through.
Select Use Ethernet Port IP Address, if the VoIP port (of the ETERNITY VoIP Card) is connected
directly to the public internet.
Select Use IP Address Fetched Using STUN, if the VoIP port (of the ETERNITY VoIP Card) is
located behind a NAT router other than Symmetric.
Select Use Router's Public IP Address, if the VoIP port (of the ETERNITY VoIP Card) is located
behind a NAT Router (any type).
In No. of Simultaneous Calls, select the number of simultaneous calls you want to allow on this SIP
Trunk.
The number of simultaneous SIP calls depends on the number of simultaneous calls allowed by the ITSP
with whom you have subscribed this SIP Trunk and the number of simultaneous calls supported by your
model of ETERNITY.
The ETERNITY ME and GE support 32 simultaneous calls on a single VoIP card, while ETERNITY PE
supports 16 simultaneous calls on a single VoIP card. Ask the ITSP about the number of simultaneous SIP
calls supported on a SIP Trunk. Program this parameter only if the ITSP supports less than the number of
simultaneous calls supported by your model of ETERNITY.
In Route Calls during Day to, select the landing destination for calls on SIP trunks during day time. You
may select any option: Operator extension, other extensions, Built-In Auto Attendant or the Auto Attendant
of the Voice Mail System (if available).
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If you select Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the default Voice
Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting and Guidance
Messages) will be applied by the Wizard. Refer Voice Message Applications to know more.
If you select Extensions as the landing destination, select the extensions in the corresponding field.
Double click the field, the list box will open. Select the extensions from the left box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using Up/Down Arrow.
The selected extension numbers will appear in the field, separated by comma.
In Route Calls during Night to, select the landing destination for calls on SIP trunks during night time.
Follow the same instruction as in the previous step to select extensions as the landing destination.
Click Submit to save your settings in the SIP Trunk. Repeat these steps to configure another SIP Trunk.
If you have finished configuring the SIP Trunks, click Next to navigate further.
If there is any SIP Trunk you have not configured, the Wizard will pop-up an alert informing you about the
trunk you have not programmed.
You must complete configuring the parameters for this SIP Trunk and only then will the wizard allow you to
configure other parameters.
For the SIP Trunks, the Wizard uses the following resources:
Routing Groups 67-74.
Trunk Feature Template 46-49.
SIP Hardware Template 2-5.
Do not change the these resources when using Configuration.
Emergency Numbers
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This page displays the default Emergency Numbers according to the Region selected for the system.
In other words, it displays the Emergency Numbers specific to your country where the ETERNITY is
installed.
If there are no Emergency Numbers defined as per the selected region, the system displays the
message No Emergency Number programmed.
The Emergency Numbers on this page are non-editable. All you need to do is to select the Outgoing
Trunk Bundle Group (OGTB) for each Emergency Number. For example, '112' is the default
Emergency Number for the mobile network. So, you may select the Mobile Trunk for dialing this
number.
Make sure that the trunks configured by default for each Emergency number route the Emergency call
to the correct network.
For each default Emergency number, select the trunks to be used to route this call. Double click the
Through field. The box will open, displaying the trunk types present in the system in the left box.
All the Trunk types present in the system are arranged by their type, in the sequence of their software port,
irrespective of their hardware location.
Double click the desired trunk type and click Select>>. The selected trunk type will appear on the box on
the right.
If you select trunks of the same type in sequential order, like: CO-001, CO-002, CO-003, BRI-001, BRI-002
and MOB-002, the same trunk type will be grouped in one OG Trunk Bundle; CO-001, CO-002, CO-003
will be OGTB #1, BRI-001 and BRI-002 will be grouped as OGTB#2, and MOB-002 as OGTB#3.
If you select the trunks of the same type in a non-sequential order, for example: CO-001, MOB-001, BRI001 and CO-002, four OGTBs will be formed with CO-001 as member of OGTB#1, MOB-001 as member
of OGTB#2, BRI-001 in OGTB#3 and CO-002 in OGTB#4. So, despite two same trunk types being
selected (CO-001 and CO-002) they are grouped in separate OGTBs.
It is possible to change the sequence of trunks on the right side box using the Up/Down Arrow. You can
also delete a particular trunk from the right box.
A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
Click OK. All the trunks you selected for a particular Emergency Number (in the right-hand side box) will
appear on the 'Through' field, sequentially separated by comma in the order of selection made.
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You may program the Emergency Numbers of your country from Configuration.
Disclaimer:
Matrix Comsec will not be responsible for incorrect programming of Emergency Numbers.
Done
You have now reached the last screen of the Wizard. Click Done to submit the configuration you have done using
the Wizard.
After completing the validation of the configuration, the Wizard will inform you about successful configuration of the
system with the message: Congratulations! System is configured as per your wish".
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The Standard PBX Wizard has been designed keeping a very broad user base in mind and is limited to
the configuration of the parameters most commonly required by a broad user group. So, it does not
cover the configuration of all the features and facilities of the system.
The Wizard does not cover the configuration of the following features and facilities. These are to be
configured from the Configuration only.
Global Directory
Personal Directory
Account Name List
Automatic Number Translation
Automated Control Applications
Call Cost Calculation
Call Duration Control
CLI-based Routing
Closed User Groups
Communication Port
Department Groups
Digital Input Port
Digital Output Port
DISA CLI Authentication
Digital Key Phone Key Template
DND Text messages
DST Parameters
E&M Parameters
E&M Feature Templates
ISDN Terminal Parameters
Keyboard Macros
Logical Partition
Master Ethernet Parameters
Page Zones
Peer-to-Peer table
Ring parameters
RTC parameters
Security Alarm Parameters
Station Message Detail Records
System Activity Log
System Fault Log
System General Parameters
System Timers and Counts
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Using Configuration
The Configuration allows configuration of all programmable parameters of the system.
You can program all the parameters using Jeeves or an extension Telephone (DKP or SLT) of the ETERNITY.
You are recommended to use Jeeves. If you choose to use a telephone, you are recommended to use a DKP for
ease of operation.
To configure the Configuration parameters, you must log into Jeeves via System Engineer Login, and click the
Configuration link.
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Also, refer the topic System Engineer Login under Using Jeeves.
To program the parameters using a Telephone, enter into SE mode from a DKP/SLT by dialing 1#91 followed by
the SE password (default: 1234).
For detailed instructions, refer the topic Entering the SE mode using a Telephone.
Avoid modifications or use of the following system resources, when configuring the system from the
Configuration.
Station Basic Feature Template number 2 to 10.
Class of Service Groups 2 to 19.
Outgoing Trunk Bundle Groups 2 to 27
Outgoing Trunk Bundles 5 to 108.
The system uses these resources for the Quick Installation Wizard-Standard PBX. Changes made in them will not
be updated in the Wizard.
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On-site Configuration
ETERNITY makes configuration even more focused by making it possible to configure only those trunk and
extension ports which are actually present in the system.
This can be done by enabling the On Site Configuration flag.
When On Site Configuration flag is enabled, ETERNITY will detect all the different trunk and extension port types
present in the system (at Power-ON). Accordingly, all the relevant pages of Jeeves will show only those ports
detected by the system for configuration.
The system will detect the presence of ports at each Power ON/Reset. Whenever a new card is found, the range of
ports is updated and displayed on Jeeves. You can then define the number of ports to be used.
By default, the On-site Configuration flag is disabled.
When the On Site Configuration flag is enabled, the Quick Installation Wizard - Standard PBX will also
display only the ports and port types that are on-board. Enable this flag if you want the Wizard to display
only the port types present in the system.
It is recommended that you enable the On-Site Configuration flag when you are configuring the system at
the installation site.
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Customer Name: You can assign the name of the enterprise/organization that is using ETERNITY as
the Customer Name. The Customer Name may contain up to 80 characters. You may enter the
address of organization/enterprise along with the name.
The Customer Name you assign will appear on the various System Reports generated and printed by
the ETERNITY.
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To cite another example, if your system is ETERNITY PE3SS, which does not support ISDN BRI or
PRI, the fields of these port types and ISDN Terminals on this page are displayed as non-editable.
Similarly, the field of the Door Phone ports will be editable only if the system has detected a Door
Phone card.
If the system has detected BRI ports, the fields BRI Trunks and ISDN Terminals will be editable.
By default, the number of BRI Trunks will be equal to the Number of BRI ports used.
The number of ISDN terminals used by default will be zero. Only when the System Engineer changes
the value of the Number of BRI Trunks to be used will the number of ISDN Terminals change to the
number of available BRI ports x 8. In other words, the number of ISDN terminals will be: Number of BRI
Ports used minus the Number of BRI Trunks used multiplied by 8.
Model Type: Select the model of ETERNITY you are configuring from this list.
When you select the Model Type, Jeeves will display the different port types according to the system
capacity (the maximum number of ports) supported by the model and variant of ETERNITY you have
selected as the Model Type.
For example, if you selected ETERNITY GE6S as Model Type, the maximum number of CO Ports will
be 96 and the maximum number of DKP extensions would be 96 ports. If you selected ETERNITY
PE6SP, the maximum number of CO Ports will be 16, and the maximum number of DKP ports will be
48.
Number of Ports Used: Define the number of ports to be used for each Port Type (CO, DKP, SLT,
Mobile, T1E1 PRI, BRI, VoIP, SIP) in the respective boxes.
For example, if you want 8 CO Trunks, 24 DKP extensions, and 128 SLT extensions to be used, select
the same numbers in the respective boxes.
If you want to use Voice Mail, or connect any device to the Digital Input Port or the Digital Output port,
select Yes.
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Customer Name: Customer Name is the name of the organization/enterprise using ETERNITY LE.
Default: Blank.
The Customer Name you assign will appear on the various System Reports generated and printed by the
ETERNITY.
The Customer Name may consist of a maximum of 80 characters, including punctuation marks. So, you
can enter also the organization's address along with the Customer Name.
Model Type: ETERNITY detects the model type which is on-board (connected). In this case, it wil display
ETERNITY LE28S. Accordingly, all pages of Jeeves display only those ports for configuration as
supported by ETERNITY LE.
You can then define the number of ports you want to configure out of the existing port types.
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For example, ETERNITY LE is connected, this page automatically detects the Model Type as ETERNITY
LE28S as well as the page will show maximum CO Ports as 128 , maximum DKP extensions ports as 128,
maximum SLT ports as 1344.
Number of Ports Used: Define the number of ports used for each port type: CO, DKP, SLT, Mobile, BRI,
T1E1, SIP from the respective combo boxes.
For example, if you want 10 CO Trunks, 50 DKP extension, and 120 SLT extensions to be used, select the
same numbers from the combo box.
If you want to use voice mail, select Yes.
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Configuring Region
The ETERNITY is a versatile system that can operate anywhere in the world, meeting the diverse customer
requirements worldwide.
To speed up the process of system configuration, ETERNITY is supplied with factory-set values for the system and
feature settings, referred to as Default Settings. These factory-set values are loaded when the system is installed
and are sufficient for getting the system into operation. However, users may alter or customize the Default Settings
to match their exact requirement.
ETERNITY provides Default Settings to match country/region-specific requirements of users around the world. The
system is designed to work efficiently in any country with these default settings.
To load the country-specific Default Settings, users must select the Region that is the country in which the system
is installed.
India is selected as the default Region. So, if you are installing ETERNITY in a country other than India, change the
Region.
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In the Region list, select the country where the system is installed.
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You will get an alert that default values will be assigned and ask you if you want to continue.
Click OK.
You will get the prompt: "Enter reverse SE password". Enter the password and click OK.
Afghanistan
002
Algeria
003
004
Argentina
005
Australia (Perth)
006
Australia (Adelaide)
007
008
Austria
009
Bahamas
010
Bahrain
011
Bangladesh
012
Belarus
013
Belgium
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704
014
Bhutan
015
Bolivia
016
017
Botswana
018
Brunei
019
020
021
Brazil (Manaus)
022
Brazil (Acre)
023
Bulgaria
024
Cambodia
025
Cameroon
026
027
Canada (Halifax)
028
029
Canada (Winnipeg)
030
Canada (Calgary)
031
Canada (Vancouver)
032
Chile
033
China
034
Colombia
035
Costa Rica
036
Croatia
037
Cuba
038
Cyprus
039
Czech Republic
040
Denmark
041
Egypt
042
Fiji
043
Finland
044
France
045
Germany
046
Greece
047
Guyana
048
Hong Kong
049
Hungary
050
India
051
Indonesia
052
Iran
053
Iraq
054
Ireland
055
Israel
056
Italy
057
Japan
058
Jordan
059
Kazakhstan
060
Kenya
Korea - North
062
Korea - South
063
Kuwait
064
Kyrgyzstan
065
Lebanon
066
Libya
067
Malaysia
068
Maldives
069
Mauritius
070
071
Mexico (Chihuahua)
072
Mexico (Tijuana)
073
Mongolia
074
Mozambique
075
Myanmar
076
Namibia
077
Nepal
078
Netherlands
079
New Zealand
080
Nigeria
081
Norway
082
Oman
083
Pakistan
084
Paraguay
085
Peru
086
Philippines
087
Poland
088
Portugal
089
Qatar
090
Romania
091
092
Russia (Novosibirsk)
093
Russia (Vladivostok)
094
Singapore
095
Slovakia
096
South Africa
097
Spain
098
Sri Lanka
099
Sudan
100
Sweden
101
Switzerland
102
Syria
103
Taiwan
104
Tajikistan
105
Thailand
106
Turkey
107
Uganda
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Ukraine
109
110
United Kingdom
111
112
United States (Chicago, Dallas, Des Moines, Memphis, Minneapolis, New Orleans,
Oklahoma, Omaha, St. Louis)
113
114
United States (Las Vegas, Los Angeles, Phoenix, San Francisco, Seattle)
115
116
117
Uzbekistan
118
Venezuela
119
Vietnam
120
Yemen
121
Yugoslavia
122
Zambia
123
Zimbabwe
124
Saudi Arabia
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Exit SE mode.
Open the browser, Internet Explorer/Mozilla Firefox, on the standalone PC/LAN PC to which the Master
Ethernet Port of ETERNITY is connected.
Enter the IP address of Master Ethernet Port of ETERNITY on the address bar or the browser.
You may enter the default IP Address of the Master Ethernet Port: 192.168.1.101
OR
If the IP has been changed, enter the current IP Address of the Master Ethernet Port.
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IP Address: Enter the IP Address you obtained from your network Administrator for the Master
Ethernet Port of ETERNITY in this field. Make sure that the IP Address does not conflict with that of
any other device on the LAN.
Network Mask: Enter the Network Mask you obtained from your network Administrator for the
Master Ethernet Port in this field. When connected on a LAN, ETERNITY should be in the same
Subnet as the LAN PC from which it is to be accessed.
Gateway Address: Enter the Routers LAN Interface IP Address as the Gateway IP Address. Now
assign these to the Master Ethernet Port.
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User Name: Enter the User Name provided by the Internet Service Provider. The User Name may be a
maximum of 64 characters.
Password: Enter the User Password provided by the Internet Service Provider. The password may be
a maximum of 64 characters.
Service Name: Enter the Service Name, if provided by your Internet Service Provider. The Service
Name may consist of a maximum of 64 characters. If Service Name is not provided, leave this field
blank.
If you configure the PPPoE User Name, Password and Service Name using commands, you can
configure a maximum of 16 characters only.
DNS IP Address: If you selected DHCP or PPPoE as Connection Type and the DHCP/PPPoE Server
provides DNS Address, select Auto as the DNS Connection Type.
If you selected DHCP or PPPoE as Connection Type, but the DHCP/PPPoE server does not assign
DNS IP Address, then you must select Static as DNS Connection Type, and manually program the
DNS IP Address.
If you selected Static as Connection Type, you can select only Static as the DNS Connection Type and
manually program the DNS IP Address.
Dynamic DNS makes it easy for users to access and use the GUI Jeeves from the public IP network by
using a Host Name instead of the IP Address of the Master Ethernet Port.
ETERNITY supports Dynamic DNS client of the Service Provider DynDNS.org. If you wish to use
DynDNS.org, program the following parameters:
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Use Dynamic DNS: Select this check box if you are using the services of DynDNS.org. By default, this
flag is disabled.
User ID: Enter the User ID you created on DynDNS.org here. The User ID may be 24-characters long.
Password: Enter the password you created on the DynDNS.org here. A maximum of 40 characters,
including all ASCII characters are allowed.
Matrix ETERNITY System Manual
Host Name: Enter the Host Name you created on DynDNS.org here. The Host Name may consist of a
maximum of 40 characters.
Retry Trials: This count defines the number of attempts that the Master Ethernet Port of ETERNITY
should make to send the Public IP Address Update Request to the Dynamic DNS Server. The Retry
Count may be set from 1 to 9. By default the count is set to 5.
Update IP Address now?: When the Master Ethernet Port is registered with the Dynamic DNS server,
the server stores the mapping between the host name and the public IP address and updates this
whenever the Public IP address changes. However, it is possible to update the IP address at will, by
clicking this button.
This parameter is to be configured when Dynamic DNS is to be used in the following scenario:
If this is your installation scenario, enter the Routers IP Address in this field. By default this field is blank.
This parameter is to be configured when Dynamic DNS is to be used in the following scenario:
STUN Server Address:Port: Enter the STUN Server Address; a maximum of 40 characters are
allowed.
Enter the Listening Port of the STUN Server. The valid range for this field is from 1024-65535. The
default STUN Port is 03478.
STUN Query Interval (min): This parameter defines the interval between each STUN query for the
Public IP Address of the NAT Router. The range of this interval is from 0001 to 9999 minutes. By
default, it is set to 120 minutes.
The Master Ethernet Port uses its own unique MAC Address as the source MAC Address on all Ethernet
frames.
MAC Address Selection: If you want the Ethernet port to use a MAC Address other than its own
unique MAC Address, select the MAC Address Selection check box.
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Clone MAC Address: If you have enabled MAC Address Selection, enter the MAC Address to be
cloned in this field.
The MAC address must be in hexadecimal format, e.g. 00:50:c2:55:b0:10.
Define the Listening Port for the Web Server of Jeeves in this field. The valid port range is 1025 to 65535.
By default, 80 is assigned as Web Server Port for Jeeves.
If you have finished programming the Ethernet Port Parameters, log out of Jeeves or continue
programming.
Click Status.
To view the status of the Ethernet Port from the Status link, click Status, then click the Network link.
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Default: Static
To program Web server Port, dial:
2121-Port
Where,
Port range is: 80, 1025 to 65535.
Default: 80
To enable/disable Dynamic DNS, dial:
2125-Code
Where,
Code is
0 for Disable
1 for Enable
Default: Disable.
To program Dynamic DNS User ID, dial:
2126-DDNS User ID
Where,
DDNS User ID may be a maximum of 40 ASCII characters.
To program Dynamic DNS User Password, dial:
2127-DDNS User Password
Where,
DynDNS User Password may consist of a maximum of 24 characters.
To program Dynamic DNS Host Name, dial:
2128 - DDNS Host Name
Where,
Host Name may consist of a maximum of 40 ASCII characters.
To program Dynamic DNS - Retry Trial Count, dial:
2129 - DDNS Retry Count
Where,
Retry Trial Count is from 1 to 9.
Default: 5.
To Update Dynamic DNS IP Address binding (Update IP Address Now? flag), dial
2130
To program Router's Public IP Address, dial:
2132-IP Address
Default: Blank.
To program STUN Server Address, dial:
2133-STUN Server Address.
Where,
STUN Server Address may be a maximum of 40 ASCII Characters.
Default: Blank.
To program STUN Server Port, dial:
2134- Port
Where,
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Exit SE mode.
Exit SE mode.
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Configuring Extensions
The ETERNITY supports the following types of extension ports:
SLT Extension Ports: Single Line Telephones (SLT) is connected to these ports. ETERNITY supports up
to 512140 SLT extensions. The number of SLT ports available to you depends on the model and variant of
ETERNITY, and number and configuration of the SLT Cards installed in the system.
DKP Extension Ports: The proprietary Digital Key Phone of Matrix is connected to these ports.
ETERNITY supports up to 128141 DKP extensions. The number of DKP ports available to you depends on
the model and variant of ETERNITY, and number and configuration of the DKP Cards installed in the
system.
ISDN Terminals: These are ISDN phones connected to the BRI Ports of the ETERNITY. ISDN Terminals
can be connected only in a Point-to-Multipoint BRI configuration (Short or Extended Passive Bus
Configuration). Refer the topic Installing BRI Card under Installation instructions for your model of
ETERNITY to know more.
A maximum of 8 ISDN Terminals (phones) can be connected on a single BRI Bus line in a Point-toMultipoint configuration. In a Short Passive Bus Configuration, you can connect up to 8 ISDN Terminals
while in the Extended Passive Bus Configuration you can connect up to 3 ISDN Terminals.
Depending on the number of BRI ports available to you and the type of Point-to-Multipoint Configuration
(Short or Extended Passive Bus), a maximum of 64 ISDN Terminals can be connected to the ETERNITY.
SIP Extensions: Any SIP-enabled device like an IP-phone, a Softphone or a Wi-Fi mobile handset can be
registered with the VoIP Card of ETERNITY and function as the 'SIP Extension' of the ETERNITY.
SIP Extensions function like any normal DKP/SLT extension of the ETERNITY, allowing you to make and
receive calls to any extension user of the ETERNITY as well as any external numbers over PSTN, GSM,
VoIP and E&M lines, depending on the Logical Partition configured in the System.
ETERNITY ME supports 999 SIP Extensions. ETERNITY GE supports 500 SIP extensions and
ETERNITY PE supports 50 SIP Extensions. SIP Extensions are a licensed feature. To know more, refer
the topic License Management.
Radio Extensions/Ports: Radio devices are connected to these ports and function as extensions of the
ETERNITY ME. A maximum of 16 Radio Ports are supported.
Magneto Ports: Magneto telephones are connected to these ports and function as extensions of the
ETERNITY ME. A maximum of 128 Magneto Ports are supported.
E&M Ports functioning as Stations: An E&M port of ETERNITY can be programmed to take on the
function of a Subscriber (Station), to work like an extension interface, receiving incoming calls.
Presuming that you have connected the extensions successfully, you may now configure the extensions
using Jeeves or a Telephone.
140. The maximum number of SLT ports supported in ETERNITY ME may vary according to the type of Power Supply (whether PSUNI
or DC) being used. See Technical Specifications provided in the Appendix.
141. The maximum number of DKP ports supported in ETERNITY ME may vary according to the type of Power Supply (whether PSUNI
or DC) being used. See Technical Specifications provided in the Appendix.
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Station Basic Feature Template - for DKP, SLT, and SIP Extensions, ISDN Terminals, and E&M Ports
functioning as Stations, Magneto ports, Radio ports and Virtual Extensions .
Station Advanced Feature Template - for DKP, SLT, and SIP Extensions, ISDN Terminals, and E&M Ports
functioning as Stations, Radio ports, Magneto ports, and Virtual Extensions.
SIP Hardware Template - for SIP Extensions (and SIP Trunks) only. See SIP Hardware Template under
Configuring Trunks.
You can use these templates to program extensions which are to be assigned the same set of features at one go,
saving you the effort for painstaking configuration of each extension.
The features in these templates are loaded with default values that fulfill the requirements of a very broad user
base. The Templates may be customized as per user requirements and applied to the extensions.
Before you start the configuration of the extensions, please read the description of the templates and how to
customize the templates according to user requirements.
is known as Calling Line Identification and Presentation (CLIP). For this feature to work, the telephone
instrument connected to the SLT port must support CLIP.
The ETERNITY supports 3 signaling protocols for CLI on the SLT port: DTMF, FSK-V.23, and FSKBellCore. Select the appropriate signaling protocol.
If you want to disable CLI on the SLT port, select 'None'.
By default, DTMF protocol is set as CLIP Type.
2. Digit Pad Count: Certain SLT instruments that support CLI require a minimum number of digits in the
calling party's number to be able identify and display it. The Digit Pad Count signifies the number of zeroes
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to be added with the Calling party's number before displaying it on the called party's instrument. This count
entirely depends on the instrument connected to it. By default, Digit Pad Count is set as 0.
3. Ring Type: The SLIC used with SLT port allows you to change the Ring type: Sinusoidal, Trapezoidal,
Low Sinusoidal, Low Trapezoidal. This is helpful in cases when telephone instruments, which expect
sinusoidal type of ringing current, are connected to the SLT port. By default the Ring Type for all SLT ports
is Trapezoidal.
4. SLT Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive Gain)
and Outgoing Speech (Transmit Gain) on the SLT port by changing the Rx Gain and Tx Gain to the
desired level. Different levels can be set for each port type in the SLT Gain Settings Template. By default,
SLT Gain Template 1 is assigned to all the SLT Hardware Templates. If you want to assign a different
Template, you must customize the SLT Gain Settings Template first and then assign the number of the
SLT Gain Settings Template in this Template. To customise the SLT Gain Settings, see Gain Settings.
5. AC Impedance: The SLIC used with each SLT port provides a facility to adjust the AC impedance of the
code. Flash is breaking the loop current for 70ms to 900ms. Flash Timer defines the time period which
should be considered as Flash, if the loop current breaks.
The range of the timer is from 70 to 900 msecs. By default, the Flash Timer is set to 101-600 msec.
Program the Flash Timer as per user requirement.
If the Flash Timer range is configured as 70-100 msec, Pulse dialing for that particular SLT Phone will not
work.
7. Answer Signaling: An Answer Signal is a signal generated on the SLT port to indicate that the called
party (remote party) has answered the call and the call is now mature.
Answer Signaling on the SLT port is particularly useful when there is a PCO machine or any Billing
equipment connected to the SLT port. With Answer Signaling enabled on an SLT port, during an outgoing
call is made from that SLT port to any other port - CO/Mobile/SIP/T1E1/BRI - when the called party (remote
party) answers, the Public Network provides an Answer Signal to the trunk port to indicate call maturity.
This information can be passed on to the PCO machine billing equipment in the form of Answer Signaling.
On detecting Answer Signaling the PCO machine billing equipment can start billing.
Answer Signaling is generated in the form of Polarity Reversal or Battery Reversal, whereby the Battery
polarity of the SLT port gets reversed. For example, if the battery polarity of the SLT port is +ve for TIP and
-ve for RING in speech condition, then on call maturity, TIP becomes -ve and Ring becomes +ve.
To generate Answer Signaling on the SLT Port, select Polarity Reversal. Select None if Answer Signaling
is not be generated on the SLT port.
By default None is selected.
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8. Disconnect Signaling: A Disconnect Signal is the signal generated on the SLT port to indicate that the
Polarity Reversal: Call Disconnection is signaled in the form of Polarity Reversal. The Battery polarity
of the SLT port will be reversed. For example, if the battery polarity of the SLT port is '+ve' for TIP and 've' for RING in speech condition then on disconnection on other port, TIP will become '-ve' and Ring
'+ve'. When call is disconnected, user will get Error tone.
Open Loop: Call Disconnection is signaled in the form of Open Loop Disconnect Pulse, whereby the
Battery voltage on the SLT port is removed for the duration of the Open Loop Disconnect Timer
programmed for that SLT port and will be restored on the expiry of this Timer. However, the Polarity of
Battery Voltage on the SLT port is not changed. When call is disconnected, the SLT extension user
gets an Error tone.
To generate Disconnect Signaling on the SLT Port, select Polarity Reversal or Open Loop as
appropriate. Select None if Disconnect Signaling is not be generated on the SLT port.
By default None is selected.
9. Open Loop Disconnect Timer (msec): This parameter is applicable only if the option Open Loop
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11. Loop Length: The Loop Length is the distance between the Central Office and the telephone instrument
connected to the SLT port. Select the Loop Length as Upto 5 Km or Above 5 Km according to your
installation scenario.
12. Minimum Current for OFF-Hook Detection: ETERNITY detects OFF-Hook state of an SLT instrument
and gives dial tone on the basis of the current drawn by it from the SLT port. However, all types and brands
of SLT instruments may not uniformly draw the same minimum current; some may draw lesser and some
may draw more, making OFF-Hook detection difficult for ETERNITY. To resolve this, ETERNITY provides
for programmable values for threshold current for OFF-Hook detection: 10mA, 12mA, 14mA, 16mA and
18mA.
By default, the value of the Minimum Current for OFF-Hook detection is set to 12 mA. Change this value
according to the current drawn by your SLT instrument.
When an SLT instrument draws current equal to or greater than the programmed threshold value of current
for off-hook detection, ETERNITY will consider the SLT instrument as OFF-Hook and will offer dial tone to
the SLT.
13. ON-Hook Detection Current (mA): ETERNITY detects ON-Hook state of an SLT instrument to route calls
on the basis of the current drawn by it from the SLT port. However, as all types and brands of SLT
instruments may not uniformly draw the same current, ON-Hook detection becomes difficult for the system.
To resolve this, ETERNITY provides for programmable values for threshold current for ON-Hook
Detection: 10mA, 12mA, 14mA, 16mA and 18mA.
By default, the value of the ON-Hook Detection Current is set to 10 mA.
When an SLT instrument draws current equal to or lower than the programmed threshold value of current
for ON-Hook detection, ETERNITY will consider the SLT instrument as ON-Hook.
SLT instruments also vary in the level of current drawn during the normal 'idle' state and when Flash is
dialed142 (the simulated idle state). So, when the Flash key of an SLT instrument is pressed, and if the
instrument draws a higher current than the threshold defined for the 'idle' state, the system will not be able
to detect Flash (ON-Hook state).
Consider this when changing the value of ON-Hook Detection Current. Define the value considering the
current drawn by your SLT instrument in idle state, as well as when Flash key is pressed.
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By default SLT Hardware Template Number 01 is assigned to all the SLT ports. This template has default values
fulfilling the common requirements of a very broad user base.
If the default SLT Hardware Template 01 fulfills the user and country requirements, retain Template 01.
If you want to change the values of certain SLT Hardware Parameters, but apply the same parameter values to all
SLT ports, simply customize the desired parameters in Template 01.
However, if different hardware parameters are to be applied to different SLTs, then you can customize different the
SLT Hardware Templates using Jeeves or a Telephone.
Select a Template number you wish to customize, for example Template 02.
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Now, apply this SLT Hardware Template 02 on the SLT ports. To do this,
Go to the SLT software ports to which this Template is to be assigned, for example SLT-001, 002, and 003.
Enter the number of the Template you customized, 02, in the field SLT Hardware Template of each of
these SLT ports.
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For example, to change the CLIP Type in Template 02 from default the DTMF to FSK-Bell, dial 5702-102-01-3
Where,
02 is the template number
01 is the parameter number for CLIP Type
3 is the code for FSK-Bell
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The default values of the SLT Hardware Templates are for the default Region India. The default values
will differ according to the Region you have selected for the system.
Exit SE mode.
725
Time Table: A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours,
Non-Working hours for a week.
Certain features of the ETERNITY like Operator, Class of Service, Toll Control, Outgoing Trunk Access,
among others, require the extension to behave differently in each Time Zone143.
So, a Time Table is assigned to extensions defining the Time Zones for the entire week, so that the system
can execute the Time Zone-dependent features and facilities according to the Time Table.
There are 8 different Time Table templates to select from. By default, the Time Table 1 is assigned to all
Station Basic Feature Templates. All seven days of the week are 'working hours 9:00 to 18:00' with break
hours from '13:00 to14:00'.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the
Station Basic Feature Template. Please refer the topic Time Tables for more details.
Operator: Define the Operator for the extensions on which the template is applied.
The system supports multiple Operators. In each Time Zone one of the four Operators can be
programmed.
Operator can be a single extension or a group of extensions, so that call management is more efficient. For
instance certain extensions may be assigned Operator 1, certain others Operator 2 and the rest may be
assigned Operator 3.
Operator 1 is the default in the Station Basic Feature Template. If you want to assign different extensions
to different Operators, you must program a separate Station Basic Feature Template with a different
Operator for each extension group.
Refer the topic Configuring 'Operator' to know more.
143. For example, incoming calls are to be routed to the security personnel extension, instead of the Operator when the office is closed
(non-working hours), or certain features in the Class of Service are to be allowed only during working hours, or access to outgoing
long distance calls are to be denied during non-working hours, or the extension must play a different greeting message to the callers during break hours and holidays (non-working hours).
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Class of Service: Class of Service (COS) defines the set features of the PBX that the extension is to be
allowed access to.
Not all extensions may require the same set of features. Some extensions may require voice mail, while
another group of extensions may need the ability to forward calls to a cell phone, and still others may have
no need to make calls outside the office.
Similarly, certain features may be required during working hours, but not during break or non-working
hours.
It is possible to assign a different Class of Service to different extensions according to their feature
requirements as well as according to the Time Zones.
By default COS group 01 is assigned to the Station Basic Feature Templates for all Time Zones. If you
want to assign a different COS for each Time Zone, you must customize the COS group first and then
assign the number of the COS group in the Template.
Refer the topic Class of Service (COS) to know more and for instructions on how to enable or disable a
feature in a COS group.
Call Budget: This flag is for enabling the Call Budget feature. The Call Budget feature will allot a 'budget'
limit for outgoing calls made by extensions on which the template is applied. Refer Call Budget for more
details.
Toll Control: This Toll Control Level allows you to define the Call Privilege (calling permission) to be
allowed to extensions according to the time of the day, during working hours (WH), break hours (BH) and
non-working hours (NH). For each Time Zone, you may define the calling permission to be allowed to
extensions by selecting the Type of Call Privilege.
Toll Control Level 0 (WH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to an extension during Working Hours.
Call Privilege: Define the type of calling permission to be allowed to the extension during the Working
Hours. The call privilege options are: No Calls, Local Calls, Regional Calls, National Calls, International
Calls, All Calls, Limited Calls.
Allowed List: For the type of Call Privilege you select, define the list of numbers to be allowed during
Working Hours.
Denied List: For the type of Call Privilege you select, define the list of numbers to be denied during the
Working Hours.
Toll Control Level 0 (BH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to an extension during Break Hours.
Call Privilege: Define the type of calling permission to be allowed to the extension during the Break
Hours. The call privilege options are: No Calls, Local Calls, Regional Calls, National Calls, International
Calls, All Calls, Limited Calls.
Allowed List: For the type of Call Privilege you select, define the list of numbers to be allowed during
Break Hours.
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Denied List: For the type of Call Privilege you select, define the list of numbers to be denied during the
Break Hours.
Toll Control Level 0 (NH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to an extension during Non-Working Hours.
Call Privilege: Define the type of calling permission to be allowed to the extension during the NonWorking Hours. The call privilege options are: No Calls, Local Calls, Regional Calls, National Calls,
International Calls, All Calls, Limited Calls.
Allowed List: For the type of Call Privilege you select, define the list of numbers to be allowed during
Non-Working Hours.
Denied List: For the type of Call Privilege you select, define the list of numbers to be denied during the
Non-Working Hours.
Toll Control Level 1: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed to an extension, regardless of Time Zone. By default Toll Control Level 1 is set to Local Calls.
Toll Control Level 2: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed to an extension, regardless of Time Zone. By default Toll Control Level 2 is set to National
Calls.
Toll Control Level 3: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed to an extension, regardless of Time Zone. By default Toll Control Level 3 is set to No Calls.
Toll Control - Call Budget Consumed: This Toll Control Level allows you to define the Call Privilege
(calling permission) to be allowed to an extension, when the Call Budget allotted to the extension is
consumed. By default the Toll Control is set to No Calls.
OG-Trunk Bundle Group (WH): This is the Outgoing Trunk Bundle Group to be allotted to the extension
for Working Hours. The extension will be allowed to make outgoing calls through the trunks in this group.
OG-Trunk Bundle Group (BH): This is the Outgoing Trunk Bundle Group to be allotted to the extension
for Break Hours.
OG-Trunk Bundle Group (NH): This is the Outgoing Trunk Bundle Group to be allotted to the extension
for Non-Working Hours.
Refer the topic OG Trunk Bundle Group for more details.
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Store Outgoing Calls: You can enable or disable the storage of call details - Station Message Detail
Records - of Outgoing Calls landing on the extensions on which the template is applied. Select the check
box to enable and clear it to disable. Refer the topic Station Message Detail Recording-Storage for more
details.
Store Incoming Calls: You can enable or disable the storage of call details - Station Message Detail
Records - of Incoming Calls landing on the extensions on which the template is applied. Select the check
box to enable and clear it to disable. Refer the topic Station Message Detail Recording-Storage for more
details.
Under Configuration, click Station Basic Feature Template to open the page.
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Select a Template number you wish to customize, for example Template 10.
Change the values of the Station Basic Feature Template parameters as desired.
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Go to the SLT software ports to which this Template is to be assigned, for example SLT-003 and 004.
Enter the number of the Template you customized, 10, in the field Station Basic Feature Template of
each of these SLT ports.
Go to the DKP software ports to which this Template is to be assigned, for example DKP-005 to 008.
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Enter the number of the Template you customized, 10, in the field Station Basic Feature Template of
each of these DKP software ports.
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Under Configuration, click the link ISDN Terminal Parameters to open the page.
Go to the ISDN Terminal software ports to which this Template is to be assigned, for example ISDN-01.
Enter the number of the Template you customized, 10, in the field Station Basic Feature Template of
each of the ISDN Terminal software port.
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Go to the SIP Extensions, for example SIP Extension 1, to which this Template is to be assigned, and
enter the Template number.
Repeat the same steps to customize another template and apply it on the extension ports.
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For example, you want to customize Template number 10, by enabling Call Budget feature and select
'Local Calls' as the Toll Control when Call Budget Consumed.
Dial 5502-1-10-06-1 to enable Call Budget feature (number 06) on Template 10.
Dial 5502-1-10-19-1 to select Local Calls (1) as Toll Control when Call Budget Consumed (19) in
Template 10.
Refer the following Table for the Feature Number and Codes for the Station Basic Feature Templates.
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For example, to assign Station Basic Feature Template number 10 to the SLT software ports 004 to
010, dial 5503-2-004-010-10.
To assign a Station Basic Feature Template to a DKP, dial:
5504-1-DKP-Template Number to assign a template to a single DKP.
5504-2-DKP-DKP-Template Number to assign the same template to a range of DKPs.
5504-*-Template Number to assign the same template to all DKPs.
Where,
DKP is the number of the Software port of the DKP, from 001 to 128.
Template Number is the number of the Station Basic Feature Template, from 01 to 50.
Default: Template 01 is assigned to all DKP ports.
For example, to assign Station Basic Feature Template 10 to DKP software ports DKP-005 to 010, dial
5504-2-005-010-10
To assign a Station Basic Feature Template to an ISDN Terminal port, dial:
5507-1-ISDN-Template Number to assign a template to a single ISDN Terminal port.
5507-2-ISDN-ISDN-Template Number to assign the same template to a range of ISDN terminal ports.
5507-*-Template Number to assign the same template to all ISDN terminal ports.
Where,
ISDN Terminal is the number of the ISDN Terminal Software port, from 01 to 64.
Template Number is the number of the Station Basic Feature Template, from 01 to 50.
Default: Template 01 is assigned to all ISDN Terminals.
For example, to assign Station Basic Feature Template number 10 to the ISDN Terminal Software
ports 01 to 04, dial 5507-2-01-04-10
To assign a Template to an E&M (Station) port, dial:
5505-1-E&M-Template Number to assign a template to a single E&M port.
5505-2-E&M-E&M-Template Number to assign the same template to a range of E&M ports.
5505-*-Template Number to assign the same template to all E&M ports.
Where,
E&M is the number of the E&M Software port, from 001 to 128.
Template Number is the number of the Station Basic Feature Template, from 01 to 50.
Default: Template 01 is assigned to all E&M (Station) ports
To assign a Station Basic Feature Template to a T1E1PRI port, dial:
5506-1-T1E1PRI-Template Number to assign a template to a single T1E1 port.
5506-2-T1E1PRI-T1E1PRI-Template Number to assign the same template to a range of T1E1 ports.
5506-*-Template Number to assign the same template to all T1E1 ports.
Where,
T1E1PRI is the number of the T1E1PRI Software port, from 1 to 8.
Template Number is the number of the Station Basic Feature Template, from 01 to 50.
Default: Template 01 is assigned to all T1E1 ports.
To assign a Station Basic Feature Template to a BRI port, dial:
5509-1-BRI-Template Number to assign a template to a single BRI port.
5509-2-BRI-BRI-Template Number to assign the same template to a range of BRI ports.
5509-*-Template Number to assign the same template to all BRI ports
Where,
BRI is the number of the BRI Software port, from 01 to 32.
Template is the number of the Station Basic Feature Template, from 01 to 50.
Default: Template 01 is assigned to all BRI ports.
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Exit SE mode.
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Caller ID Presentation while Transfer: This parameter is related to the CLIP feature. It allows you to
select whether you want the CLI of the Held Party or the CLI of the Transferring Party to be displayed to
the transfer destination extension while the call is being transferred. Refer the feature description for
Calling Line Identification and Presentation (CLIP) to know more.
Call Forward No-Reply Timer: This parameter is related to the Call-Forward-No Reply feature. The Timer
is the duration for which the system will wait for an extension to answer an incoming call, before forwarding
the call it to the programmed destination phone number as Call Forward-No Reply. By default the Timer is
set to 30 seconds. Refer the feature description for Call Forward to know more.
DDI IC Routing: This flag is relevant only if you are using DDI IC Routing. As this flag is enabled by
default, hence all the extensions configured in the IC Reference Table will behave as DDI Extensions. If
you do not want any of these extensions to function as a DDI extensions, disable this flag in the feature
template applied on that extension. By default, the flag is enabled.
Refer the topics Direct Dialing-In (DDI), DDI Routing Table and IC Reference Table to know more.
Send DDI Number as CLI?: This flag is relevant only if the extension is programmed as a DDI Extensions
(the DDI IC Routing is enabled). You can choose whether to the Calling Line Identification (CLI) of the DDI
extension should be sent for outgoing calls made from that extension. By default the flag is enabled.
Internal Calls Storage: This parameter is related to the storage of Station Message Detail Records of
internal calls, made between extensions of the ETERNITY. You can select the type of internal calls to be
stored by the system: i) Calls made from the extension, ii) Calls made to the extension, iii) Calls made from
as well as calls made to the extension, and iv) all types of calls, and v) no internal calls. By default, all
types of internal calls on the extensions are stored. Select desired type of internal call storage.
Refer Station Message Detail Recording-Storage to know more.
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Walk-Out Mode: This parameter is related to the feature Walk-In Class of Service. ETERNITY offers two
types of Walk-In: i) One-Call per Walk-In, whereby the user is automatically logged out after a call. ii) WalkIn until Logout, whereby the user remains logged on until s/he manually walks out or a second user walks
into the same extension.
You must select the Walk-Out mode for the extension. If One-Call per Walk-In is to be supported on the
extension, select 'One Call' as Walk-Out mode.
If Walk-In until Logout is to be supported on the extension, select 'Multiple Calls' as the Walk-Out mode.
To know more about this feature, refer Walk-In Class of Service.
CDC Table: This parameter is to be programmed if you have enabled the Call Duration Control (CDC)
feature on the extension. The system will check the Call Duration Control (CDC) Table applied to the
extension to implement this feature on the extension. So, you must first program the CDC Table and enter
the number of the CDC Table you have programmed in this field.
You can program 8 different CDC tables. By default, CDC Table No. 1 is assigned to all extensions. If CDC
is to be applied on extensions of the ETERNITY, simply program the default CDC Table No. 1.
To do this, click the link CDC Table to open the page. Program the CDC Table parameters and 'Submit' to
save your settings. Now return to the Station Advanced Feature Template and enter the number of the
CDC Table you programmed in the CDC Table field of the template.
Refer the feature description Call Duration Control (CDC) to know more and for instructions on creating
CDC Tables.
Forced Account Code: This flag used to enable or disable the feature Forced Account Code on the SLT/
DKP/ISDN Terminal extensions. When this flag is enabled, the system will allow the extension user to dial
an external number only after entering the Account Code. Refer the feature description for Account
Codes to know more.
Department Billing Group: This parameter enables you to know the total cost of the calls made by a
particular group of extensions. This parameter is used as a one of the filters for printing SMDR Reports
namely, Print outgoing calls department group wise. To be able to use this filter, you must assign the
extensions to a Department Bill Group. You can create as many as 99 different Department Bill Groups.
Enter the number of the Bill Group you want to assign the extensions to in this field.
Floor Service Group: This parameter is related to the Floor Service feature. Floor Service can be floorwise or centralized. Floor Service requires you to program Routing Groups as landing destinations for
extension calls.
Program the Floor Service (Routing) Group first and enter this Floor Service (Routing) Group number in
this field. There are 96 different Routing Groups to be programmed as Floor Service Groups. By default,
no Routing Group is assigned to Floor Service in the Template ('00').
To know more about this feature, refer the feature description for Floor Service.
Calls from the extension will land on the Floor Service (Routing) Group you have assigned in this field.
Alarm Notification Type: This parameter is related to the Alarm feature of the ETERNITY. You can select
any from the following options for Alarm Notification to extension users for Alarm calls:
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Voice Message: Extension users will be played a message recorded in the Voice Module, when they
answer the alarm call. By default, this option is selected.
Music-on-Hold: Extension users will be played music-on-hold when they answer the alarm call.
Voice Mail: The Extension users will be played the message recorded in the VMS, when they answer
the alarm call. You must have the VMS Card installed in your system if you want to select this option.
External Music: Extension users will be connected to live music when they answer the alarm call. If
you select this option, make sure you have connected a live music source to the Analog Input Port of
the ETERNITY.
Routing Group: Extension users will be connected to the extensions programmed in the 'Alarm
Notification Group'. For this you must have programmed a Routing Group.
If you select this option, you can also connect external Messaging Devices to play real-time updated
information like date, time, greetings, weather information, specific event announcements, etc., when
then extension users answer the alarm calls. The messaging devices must be connected to any of the
SLT ports and included in the Routing Group for alarm notification.
Alarm Notification Routing Group: Program this parameter if you have selected 'Routing Group' as the
Alarm Notification Type. Enter the number of the Routing Group you have programmed for Alarm Calls.
By default Routing Group 31 is assigned as the Alarm Notification Routing Group. If the same Routing
Group is to be assigned to all extensions, click the link 'Alarm Notification Routing Group' to open the
Routing Groups page. Select members (extensions) in this routing group. Save your changes by clicking
'Submit' button.
You can program a different Routing Group repeating these steps. Make sure to enter the number of the
Routing Group you programmed in this field.
Refer the feature description for Alarms to know more.
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Help Desk: Enable this flag if you want to define the extension as a Help Desk. When this flag is
enabled, Auto Call Back will be automatically set whenever this extension is found busy.
GPAX - Charge Internal Calls: This parameter is related to the GPAX application. If the extension is
programmed as a GPAX user, enable this flag for billing internal calls made by the extension. When this
flag is enabled, the system will record all calls made from the extension in the Station Message Detail
Record-Outgoing buffer for GPAX Billing. If the flag is disabled the calls will not be billed and will be
recorded in the Station Message Detail Record - Internal buffer as an internal call.
Call Taping: If you want to use the Call Taping feature on the extension, you must program the related
parameters described below.
Tape Calls coming without CLI: Enable this flag if you want incoming calls without CLI to be taped.
By default, the flag is disabled. The system will not tape incoming calls without CLI.
Number List-Incoming Calls: Assign a Number List containing numbers of Incoming Calls that must
be taped. You must first program the Number List. By default, Number List 09 is assigned.
Number List-Outgoing Calls: Assign a Number List containing numbers of Outgoing Calls that must
be taped. You must first program the Number List. By default, Number List 10 is assigned also for
outgoing calls.
If Number list 10 is already used for another application, prepare a different number list and assign it to
the template.
Call Taping for Internal Calls: Enable this flag if you want to allow Call Taping of internal calls made
and received by the extension.
Allow External Call Forward for: This parameter defines the types of calls for which the External Call
Forwarding is to be applied. This parameter is relevant for the features Call Forward and Mobility
Extension. You may select from the following options:
Internal Calls Only
Trunk Calls Only
Internal + Trunk Calls.
Under Configuration, click Station Advanced Feature Template to open the page.
Select a Template number you wish to customize, for example Template 02.
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Change the values of the Station Advanced Feature Template parameters as desired.
Now, apply this Template 02 on the SLT/DKP/ISDN Terminal ports, and Virtual Extensions.
742
Enter the number of the Template you customized, 02, in the field Station Advanced Feature Template
of each of these SLT ports.
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Go to the DKP software ports to which this Template is to be assigned, for example DKP-005 to 008.Enter
the number of the Template you customized, 02, in the field Station Advanced Feature Template of each
of these DKP software ports.
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Go to the ISDN Terminal software ports to which this Template is to be assigned, for example ISDN-01.
Enter the number of the Template you customized, 02, in the field Station Basic Feature Template of
each of this ISDN Terminal software port.
Repeat the same steps to customize another template and apply it on other extension ports.
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Forced Account Code Flag to be enabled, 5602-1-02-08-1 (Template 02, Feature Number 08 for
Forced Account Code, Code 1 for enable flag).
Refer the following Table for the Feature Number and Codes for the Station Advanced Feature Templates.
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For example, to assign Station Advanced Feature Template number 02 to the SLT software ports 004
to 010, dial 5603-2-004-010-02
To assign a Station Advanced Feature Template to a DKP, dial:
5604-1-DKP-Template Number to assign a template to a single DKP.
5604-2-DKP-DKP-Template Number to assign the same template to a range of DKPs.
5604-*-Template Number to assign the same template to all DKPs.
Where,
DKP is the number of the Software port of the DKP, from 001 to 128.
Template Number is the number of the Station Advanced Feature Template, from 01 to 50.
Default: Template 01 is assigned to all DKP ports.
For example, to assign Station Advanced Feature Template 02 to DKP software ports DKP-005 to 010,
dial 5604-2-005-010-02
To assign a Station Advanced Feature Template to an ISDN Terminal port, dial:
5607-1-ISDN-Template Number to assign a template to a single ISDN Terminal port.
5607-2-ISDN-ISDN-Template Number to assign the same template to a range of ISDN terminal ports.
5607-*-Template Number to assign the same template to all ISDN terminal ports.
Where,
ISDN is the number of the ISDN Terminal Software port, from 01 to 64.
Template Number is the number of ~Template number is the number of the Station Advanced Feature
Template, from 01 to 50.
Default: Template 01 is assigned to all SIP extensions.
To assign a Station Advanced Feature Template to a Virtual Extension, dial:
5613 - 1 - Virtual Extension - Station Advance Feature Template to assign the template to a single
extension.
5613-2-Virtual Extension-Virtual Extension-Station Advance Feature Template to assign the same
template to a range of extensions.
5613-*- Station Advance Feature Template to assign the same template to all extensions.
Where,
Virtual Extension is from 01 to 64.
Station Advance Feature Template is the number of the Station Advanced Feature Template, from 01
to 50.
Default: Template 01 is assigned to all Virtual Extensions.
To assign a Station Advance Feature Template to a Magneto Port, dial:
5611-1-Magneto-Station Advanced Feature Template Number to assign a template to a single
Magneto port.
5611-2-Magneto-Magneto- Station Advanced Feature Template Number to assign the same
template to a range of Magneto ports.
5611-*- Station Advanced Feature Template Number to assign the same template to all Magneto
ports.
Where,
Magneto is the Software Port number of the Magneto port from 001 to 128.
Station Advanced Feature Template is from 01 to 50.
Default: Station Advanced Feature Template Number 01.
Exit SE mode.
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Configure the following parameters for each SLT port on this page:
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SLT Port No.: This non-editable field is the number of the software port of the SLT port.
Hardware - Slot - Port: 'Slot' is the number of the Universal Slot in which the SLT Card is inserted.
'Port' is the number of the SLT hardware port on which the telephone instrument is connected.
The ETERNITY can automatically detect and assign the hardware slot and port numbers automatically
to the SLT software ports.
For example: if you have inserted the SLT8 Card in Slot number 02 and SLT16 Card in Slot number 03
of ETERNITY ME16S, the system will assign the hardware slot 02 and port numbers 01-08 to the SLT
Software Ports from 001 to 008 respectively. The system will assign Slot 03 and port numbers 09-24 to
the SLT Software Ports 009 to 024. Refer the topic Software Port and Hardware ID to know more.
However, if required, you may change the Hardware Slot and Port assigned to the SLT software port. In
which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
Access Code: Assign Station Access Codes to the SLT Port. Station Access Codes are commonly
referred to as Extension Numbers. These may be a maximum of 6 digits, which are dialed to call the
SLT port to which they are assigned.
All SLT ports are assigned the following Station Access Codes as default.
SLT Software Port
Access Codes
001
2001
002
2002
003
2003
512
2512
You may either apply the default Station Access Codes to the SLT ports or assign them according to
your requirement and preferences.
To assign Station Access Codes according to your preference and requirment to a range of SLT Ports,
see Assigning Access Codes to a Range of Extensions.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics Access Codes and Conflict Dialing to know
more.
Name: Assign a 'Name' to the SLT port. The name may be of the person who will use the SLT or the
name of a department. This name will be displayed on the LCD of the remote user's phone, if it is
equipped with Caller ID.
You can program a name of a maximum of 18 alphanumeric characters.
Station Basic Feature Template: Assign a Station Basic Feature Template to the SLT. A Station
Basic Feature Template includes a set of features that completely define the behaviour of the extension
port, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call Budget,
and Station Message Detail Records (storage of Incoming and Outgoing Call details).
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By default, Station Basic Feature Template 01 is assigned to all extensions of the system that includes
DKP ports, ISDN Terminals, SIP extensions, and E&M Lines with Station as Orientation Type.
Check if the default template fulfills the feature requirements (like Class of Service (COS), Toll
Control, OG Trunk Bundle Group, etc.) of the SLT extension user.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all SLTs, retain Template 01.
If different sets of features are to be allowed to different SLTs, then prepare separate Station Basic
Feature Templates and apply them on the ports. To do this,
Under Configuration, click Station Basic Feature Template to open the page.
Customize Template number 10 and click Submit at the bottom of the page.
Enter the number of the Template you customized, Template 10 in the Station Basic Feature
Template field of the SLT Port, for instance SLT-003, on which you want to apply this template.
Repeat the same steps to customize and assign a different Template to another SLT port.
Also, refer the topic Station Basic Feature Template to know more about customizing the templates
and applying on the ports.
Station Advanced Feature Template: Assign a Station Advanced Feature Template to the SLT. The
Advanced Feature Template consists of a set of less commonly used features like Message Wait
Notification Type, Alarm Notification Type, Caller ID Presentation for Transferred Calls, DDI Incoming
Call Routing, Storage of Internal Calls, Call Duration Control, Floor Service, Call Taping, etc.
By default Station Advanced Feature Template 01 is assigned to all extensions of the ETERNITY,
which includes DKP Ports, ISDN Terminals, SIP Extensions, and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the SLT extension users by clicking the
link Station Advanced Feature Template.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to
all SLT ports, retain Template 01.
If different sets of features are to be allowed to different SLT Ports, then prepare separate Station
Advanced Feature Templates and apply them on the ports.
To do this,
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Under Configuration, click Station Advanced Feature Template to open the page.
Enter the number of the Template you customized, template 02, in the Station Advanced Feature
Template field of the SLT Ports on which you want to apply this template.
Repeat the same steps to customize and assign a different Template to another SLT port.
Also refer the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the ports.
SLT Hardware Template: Assign an SLT Hardware Template to the SLT port. An SLT Hardware
Template is a set of features that define the behavior of the SLT hardware port. The SLT Hardware
Template allows you to configure according to user requirements a common set of features for all SLT
Hardware Ports, like Caller ID Presentation (DTMF, FSK), Digit Pad Count, Ring Type, AC Impedance,
Answer Signaling type, Speech Transmit and Receive Gains, Open Loop Disconnect, Loop Current,
and Fax connectivity.
There are 50 SLT Hardware Templates that can be customized and assigned to the SLT ports. By
default SLT Hardware Template Number 01 is assigned to all the SLTs. This template has default
values fulfilling the common requirements of a very broad user base.
Check if the values in this template fulfill your requirements.
If the default SLT Hardware Template 01 fulfills the feature requirements and if the same features are
to be allowed to all SLTs, retain Template 01.
If different sets of hardware features are to be allowed to different SLTs, then prepare separate SLT
Hardware Templates and apply them on the ports. To do this,
Customize Template number 02 and click Submit at the bottom of the page.
Enter the number of the Template you customized, template 02, in the SLT Hardware Template
field of the SLT Ports on which you want to apply this template.
Repeat the same steps to customize and assign a different SLT Hardware Template to another SLT
port.
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Also, refer the topic SLT Hardware Template to know more about customizing the templates and
applying on the SLT ports.
Call Pick-Up Group: Configure this parameter if you want to assign the SLT to a particular Call Pick
Up group.
Call Pick Up allows the SLT extension user to 'pick up' (answer) calls ringing on any other extension, by
dialing a feature code, without physically going to the ringing extension.
For this to work, both extensions, the ringing extension and the extension picking up the call, must be in
the same Call Pick Up Group. Refer Call Pick Up for instructions on how to create groups. You can
create as many as 99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SLT in this field.
Station Type: This parameter is relevant only for the Hotel Application of the ETERNITY. Extensions
are identified as 'Administrator' or 'Guest' extensions according to the intended user of the SLT. When
the Station Type is selected, the system will automatically assign the Guest and Administrator (Hotel
Staff) features to the SLT. To know more, refer the ETERNITY Hospitality System Manual.
Advanced Configuration
For extension users who need to be provided features like Personal Directory or assigned a Priority, you may click
the Advance button at the bottom of the page and program the following parameters:
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Mobile Number: Enter the Mobile Number of the extension user you wish to store. The Number can be a
maximum of 16 digits.
Email ID: Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of
64 characters.
Group: You can assign the extension user to a Group. The system clubs together extension users
assigned the same Group. The Group can be a maximum of 16 characters. Default: Blank.
Personal Directory: Enter the number of the Personal Directory that you want to assign to this SLT. A
Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (OG Trunk Bundle Group). The Personal Directory is
necessary for using the features Abbreviated Dialing and Dial By Name.
When a Personal Directory is assigned to an SLT, it must also be configured.
The Personal Directory can be configured by the SLT users and by the System Engineer. Refer the topic
Abbreviated Dialing for instructions on configuring the Personal Directory.
If you have completed configuring SLT parameters, click Submit at the bottom of the page to save your
settings.
It is possible to default all the parameters by clicking the Default button. You can also restore default
values of the parameters of a single SLT by clicking the Default One button and specifying the SLT you
want to set to default.
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Configure the following voice mail settings for the SLT extensions you wish to assign voice mail.
SLT Port No.: This is the software port number of the SLT extension.
H/w Slot - Port: It is the hardware slot and port number assigned to the SLT software port (that is, the
hardware slot and port number to which the SLT extension is connected).
Access Code: This is the access code (that is, extension number) assigned to the SLT port in the SLT
Parameters. If you have changed the default Access Codes in the SLT Parameters page, the same will
appear here.
Name: This is the name assigned to the SLT port in the SLT parameters.
Personal Mailbox: Select the checkbox to assign Personal Mailbox to SLT extensions. By default,
Personal Mailbox is not assigned to SLT extensions in Enterprise mode.
In the Hotel mode, Personal mailbox is assigned to all extension phones, by default.
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Mailbox Size (min): You may increase or decrease the size of the personal mailbox assigned to the
SLT extension, by changing the default Mailbox size of 5 minutes. You may change the mailbox size to
any desired value from 00001 to 60000 minutes. Default: Enterprise mode: 5 minutes; Hotel mode: 999
minutes.
Maximum Message Length (sec): You can define the length of each message (in seconds) callers
are to be allowed to record in the mailbox. You may change the maximum message length to any
desired value from 0001 to 9999 seconds. Default: Enterprise mode: 15 seconds; Hotel mode: 120
seconds.
The VMS card will stop recording the message of the callers if it exceeds the maximum message
length, and will store only that part of the message recorded within the maximum message length limit.
New Message Delivery option in Mailbox Full condition: When the personal mailbox is full, you may
select one of the following options for delivery of new messages:
Do not offer to record a message: The VMS will not allow the caller to record a message by
declining delivery of the message.
Deliver new message to General mailbox: The VMS will record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their Class of Service (COS) are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the SLT extension. Refer Class of Service (COS) for instructions.
Overwrite old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS starts overwriting the oldest message first.
By default, Deliver to General mailbox is selected.
Play message details after delivery of message: After the extension user has finished listening to a
message in the mailbox, you can also have the VMS play message details such as Date and Time
when the message was recorded, the callers number146, and the extension number dialed by the
caller147 to the extension user.
You may select from one of the following options to Play message details:
Never: The VMS will not play message details to the mailbox owner after playing the message.
Always: The VMS will play message details to the mailbox owner after playing each message.
On Demand: The VMS will play message details to the mailbox owner only when the mailbox
owner requests it. On completion of each message, the VMS will prompt the extension user to
press a digit for date and time stamp. When the mailbox owner presses the digit, the VMS will play
the message details.
By default, On Demand is selected.
Ask Password to access Mailbox: By default, access to the mailbox is password protected. The
User Password is required to access the mailbox. Whenever the mailbox owner accesses the
mailbox, the VMS will ask for the (user) password.
If you want to remove password protection, clear this checkbox.
146. The number of person who left the message in the mailbox.
147. The number of the extension user for whom the message is intended.
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Since a Mailbox can be accessed using the default User Password, 1111, extension users who are assigned
a mailbox are recommended to change their User Password to a unique 4 digit number to prevent
unauthorised access to their mailbox.
Allow Mailbox Management: Mailbox Management allows the extension user to change mailbox
settingsrecord Extension Name for the mailbox, redirect messages from the mailbox, delete all old
messages from the mailbox, record greeting messages for the mailbox. By default, Mailbox
Management is enabled.
If you do not want to allow the extension user to change the mailbox settings, in Allow Mailbox
Management, select No.
To know more about this feature, see Mailbox Settings
Auto Delete Messages: Select the type of messages you want the VMS to automatically delete from
your mailbox. You can select All or Old. Default: None.
Days for Auto Delete Messages: Select the number of days after which you want the VMS to
automatically delete the messages in your mailbox. Default: 90 days.
Department Group Mailbox: You can assign the Mailbox of a Department Group to SLT extensions,
even to those SLT extensions that are not included in the Department Group. Refer the topic
Department Call.
To assign the Department Group mailbox to an SLT extension, select the number of the Department
Group (1 to 16) from the combo box.
If you do not want to assign Department Group mailbox to an SLT extension, select None.
By default None is selected.
Voice Mail Auto Attendant Features: This parameter is applicable only if you are using the VMS Auto
Attendant.
Voice Mail Auto Attendant Profile: Select a profile for the SLT extension. The profile determines
the welcome message to be played to mailbox owners when they reach the home node. It also
determines whether or not the user should be taken to the root node directly.
Abbreviated Name: When the VMS is used as Auto Attendant, the callers can be prompted to Dial
by Name the desired extension users instead of their extension numbers.
To allow callers to reach the SLT extension using Dial By Name, abbreviate the extension users
name to three letters and enter it in this field.
Announce Name: If you want the VMS to announce the extension users name to the caller when
transferring the call to the extension, select the checkbox to enable Announce Name. By default,
Announce Name is disabled.
If you enable Announce Name, make sure you record the extension users name on the VMS. Refer
Recording Extension Names for instructions.
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Call Transfer: Select the desired method for transferring the call answered by the VMS Auto
Attendant to the SLT extension. You may select any of the following methods of call transfer for
each time zone, Working Hours (WH), Break Hours (BH) and Non-working Hours (NH):
None: When the caller dials the extension number, the VMS Auto Attendant will check if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
Blind: When the caller dials the extension number, the VMS Auto Attendant will transfer the call
on the extension without checking whether it is busy or free.
Wait for Ring: When the caller dials the extension number, the VMS Auto Attendant will wait for
the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the
extension, if assigned, or take the caller back to the home node.
Wait for Answer: When the caller dials the extension number, the VMS Auto Attendant will
transfer the call when the extension answers (goes OFF-Hook).
If the extension does not answer148, the VMS Auto Attendant will transfer the call to the mailbox
of the extension, if assigned, or take the caller back to the home node.
Screen: The VMS Auto Attendant prompts the caller to record his/her name. It puts the caller on
hold and places the call on the desired extension. If the extension is free and answers the call,
the VMS announces the callers name to the extension user and prompts the extension user to
choose whether or not to speak to the caller. If the extension user chooses to talk, the VMS
transfers the call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the
extension user, if assigned, and asks the caller to leave a message.
By default, Wait for Answer is selected as Call Transfer method for all time zones.
Message Wait Indication: This parameter allows you to select the type of indication to be given to
the extension user for new messages in the mailbox and message wait set by another extension
user.
You can select from any of the four types of indicators described below for new messages:
Stuttered Dial Tone/Voice Message: When the extension user goes OFF-Hook, s/he will hear
a voice message, if a pre-recorded Voice Module has been assigned for Message Wait
Notification. If no voice module is recorded and assigned, the extension user will hear a
stuttered dial tone instead.
If you want voice message to be played as message wait notification, record and assign a Voice
Module. Refer Voice Message Applications for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait Notification
will not be played if there are already 4 being played simultaneously. In which case, Stuttered Dial Tone will
be played as Message Wait Indication, when the extension user goes OFF-Hook.
148. The VMS will wait for the duration of the Wait for Answer Timer (default: 30 seconds; the timer is configurable). If the call is not
answered before this timer expires, it is treated as No Reply.
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LED Lamp: If the SLT has a 'Message Wait' lamp, you may select this option as Message Wait
Indication. The lamp will blink continuously and will be turned off when the extension user has
retrieved all the waiting messages.
Ring: The extension will ring for the duration of the Message Wait Ring Timer (configurable;
default: 30 seconds), for as many times as the Message Wait Ring Count (configurable; default:
10 times), at the interval set as the Message Wait Ring Timer Interval (configurable; default: 30
minutes).
When the extension user answers the call, the VMS informs the user of the new message and
allows the extension user to access it.
Refer the feature description Message Wait to know more.
Message Wait Notification via Call: The message wait notification will be sent to a number
(destination number). This number can be an internal or an external number. To use this feature,
configure the following parameters:
Type: If you want the notifications to be sent as soon as a new message arrives in the mailbox
of the extension user, select Immediate.
If you want the notification to be sent at fixed time schedules, select Scheduled.
If you do not want to set message wait notification via call, select None. Default: None.
Profile: Assign the Profile according to which you want the system to send the notifications.
The Message Wait Notification Profile determines how notification calls are to be made to the
destination numbers.
Destination Number: Enter the number on which you want the system to send the notification
calls.
The destination number can be an internal or an external number. The destination number can
be a maximum of 16 digits. Valid digits are 0 to 9, # and *.
When the notification call is answered, the VMS informs the callee of the new message and
allows the callee to access it.
Refer the feature description Message Wait Notification via Call to know more.
Message Wait Notification via E-mail: The message wait notification will be sent to the e-mail
address of the extension user. To use this feature, configure the following parameters:
Notification: If you want the message wait notification to be mailed to the extension user along
with the new voice message as attachment, select the option Send With Attachment.
If you want only the notification to be mailed, select the option Send Without Attachment.
If do not want to set message wait notification via e-mail, select Do not send. Default: Do not
send.
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E-mail Address: Enter the e-mail ID of the extension user to which the notification is to be sent.
E-mail ID may consist of up to 64 characters. Default: blank.
Extensions users will receive notifications only for the mailbox memory utilization, if you configure the E-mail
Address and select Do not sent as the Notification option.
Refer the feature description Email Based Notification to know more.
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Exit SE mode.
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3 - Ring
Default: LED Lamp.
To enable/disable Message Notification through E-mail for a mailbox, dial:
5723-1-SLT-Code-#* to enable/disabled Message Notification through E-mail for a mailbox on a single
SLT extension.
5723-2-SLT-SLT-Code-#* to enable/disabled Message Notification through E-mail for a mailbox on a
range of SLT extension.
5723-*-Code-#* to enable/disabled Message Notification through E-mail for a mailbox on all the SLT
extension.
Where,
SLT is the Software Port number of the SLT port from 001 to 512.
Code is:
0 - Disabled
1 - With Attachment
2 - Without Attachment
Default - Disabled.
To program E-mail ID for a mailbox, dial:
5724-1-SLT-E-mail ID-#* to assign an E-mail ID for a mailbox to a single SLT extension.
5724-2-SLT-SLT-E-mail ID-#* to assign an E-mail ID for a mailbox to a range of SLT extensions.
5724-*-E-mail ID-#* to assign an E-mail ID for a mailbox to all the SLT extensions.
Where,
SLT is the Software Port number of the SLT port from 001 to 512.
E-mail ID is 64 ASCII Characters.
Default: Blank.
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Exit SE mode.
The DKP Ports appear in tabs, with eight DKP Ports in each tab, 001-008, 009-016, 017-024, and so forth.
DKP H/w Slot - Port: 'Slot' is the number of the Universal Slot in which the DKP Card is inserted. 'Port'
is the number of the DKP hardware port on which the proprietary DKP EON is connected.
The ETERNITY can automatically detect and assign the hardware slot and port numbers automatically
to the DKP software ports.
For example: if you have inserted the DKP8 Card in Slot number 03 and DKP16 Card in Slot number
04 of ETERNITY ME16S, the system will assign the hardware slot 03 and port numbers 01-08 to the
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DKP Software Ports from 001 to 008 respectively. The system will assign Slot 04 and port numbers 0916 to the DKP Software Ports 009 to 024. Refer the topic Software Port and Hardware ID to know
more.
However, if required, you may change the Hardware Slot and Port assigned to the DKP software port.
In which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
If your ETERNITY is ME10S or ME16S, the system will detect and assign the first four Software Ports to
the four DKP Ports located on the Switch Card.
Access Code: Assign Station Access Codes to the DKP Port. Station Access Codes are commonly
referred to as Extension Numbers. These may be a maximum of 6 digits, which are dialed to call the
DKP port to which they are assigned.
All DKP ports are assigned the following Station Access Codes as default.
DKP (Software) Port No.
Access Codes
001
3001
002
3002
003
3003
128
3128
You may either apply the default Station Access Codes to the DKP ports or assign them according to
your requirement and preferences.
To assign Station Access Codes according to your preference and requirment to a range of DKP Ports,
see Assigning Access Codes to a Range of Extensions.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics Access Codes and Conflict Dialing to know
more.
Name: Assign a 'Name' to the DKP port. The name may be of the person who will use the DKP or the
name of a department. This name will be displayed on the LCD of the DKP of the user and on other
extension user's phones, provided these are also a model or EON or are equipped with Caller ID.
You can program a name of maximum 18 alphanumeric characters.
Station Basic Feature Template: Assign a Station Basic Feature Template to the DKP.
A Station Basic Feature Template includes a set of features that completely define the behaviour of the
Extension, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call
Budget, and Station Message Detail Records (storage of Incoming and Outgoing Call details).
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By default, Station Basic Feature Template 01 is assigned to all extensions of the system that includes
DKP ports as well as DKP ports, ISDN Terminals, E&M Lines with Station as Orientation Type, and SIP
Extensions.
Check if the default template fulfills the feature requirements (like Class of Service (COS), Toll
Control, OG Trunk Bundle Group, etc.) of the DKP.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all DKPs, retain Template 01.
If different sets of features are to be allowed to different DKPs, then prepare separate Station Basic
Feature Templates and apply them on the ports. To do this,
Under Configuration, click the link Station Basic Feature Template to open the page.
Customize Template number 11 and click 'Submit' at the bottom of the page.
Enter the number of the Template you customized, Template 11, in the Station Basic Feature
Template field of the DKP Port, for instance DKP-005, on which you want to apply this template. If
you want to apply this template to other ports too, like DKP-006, 007, and 008, assign the Template
11 to all these ports.
Repeat the same steps to customize and assign a different Template to another DKP port.
Also, refer the topic Station Basic Feature Templates to know more about customizing the
templates and applying on the ports.
Station Advanced Feature Template: Assign a Station Advanced Feature Template to the DKP. The
Advanced Feature Template consists of a set of less commonly used features like Message Wait
Notification Type, Alarm Notification Type, Caller ID Presentation for Transferred Calls, DDI Incoming
Call Routing, Storage of Internal Calls, Call Duration Control, Floor Service, Call Taping, etc.
By default Station Advanced Feature Template 01 is assigned to all extensions of the ETERNITY,
which includes DKP Ports, ISDN Terminals, E&M Lines configured as Stations, and SIP Extensions.
Check if this default template fulfills the feature requirements of the DKP Ports.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to
all DKP ports, retain Template 01.
If different sets of features are to be allowed to different DKP Ports, then prepare separate Station
Advanced Feature Templates and apply them on the ports.
To do this,
Under Configuration, click Station Advanced Feature Template to open the page.
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Enter the number of the Template you customized, Template 02 in the Station Advanced Feature
Template field of the DKP Port, for instance DKP-001, on which you want to apply this template. If
you want to apply this template to other ports too, like DKP-002, 003, and 004, assign the Template
04 to all these ports.
Repeat the same steps to customize and assign a different Template to another DKP port.
Also refer the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the ports.
Call Capacity: Call Capacity is the number of Call Appearances (also referred to as 'call loops')
assigned to a (DKP) extension. It is the ability of an Extension to handle multiple calls simultaneously. A
Call Appearance allows an extension user to attend to more than one calling party at a time.
A minimum of two Call Appearances must be assigned to a DKP Extension - Operator extension or
Executive extension - so that the Extension user can put one party on hold while talking to another. A
third Call Appearance allows the extension user to put two calls on hold, make/attend a third call and
toggle between three calls.
The higher the call capacity (the more the number of Call Appearances assigned to an extension), the
more the number of calls the Extension user can handle.
The ETERNITY supports a maximum of 10 Call Appearances as Call Capacity of the DKP extensions.
DKP extensions for Executives are usually assigned 2 Call Appearances, while the Operator Extension
is assigned 6 Call Appearances to handle 6 calls simultaneously. The default call capacity of the DKP
ports is 02.
Now, select the number of Call Appearances you wish to assign to the DKP port in the column, 'Call
Capacity'.
Call Waiting Tone: During an on-going conversation, if there is a second incoming call, the system
plays beeps to indicate the second incoming call. You can set the frequency of the beeps as per your
requirement. You can select from the following options:
Off
Beep Once
Beep until Answered
Default: Beep Once
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Key Map: EON is designed to function as Operator, Executive, Hotel Attendant, and Hotel Guest
extensions, providing default key settings (key maps) for all these functions. All you need to do is
assign a Key Map Template according to the intended user of the DKP.
For example, if the DKP is to be used by the Operator, select Operator's Template. The DKP will be
assigned the key template with the special features required by Operators, such as more DSS keys for
Trunk Access and Call Appearances, a Call Release Key, etc.
Similarly, if the user of the DKP is a Hotel Attendant, select 'Hotel Attendant's Template'. The key map
with the specific Front Desk User features such as Check-In, Check-Out, Guest In/Out, Change Room
Clean Status, Room Shift, will be automatically assigned to the DKP.
You can also customize the key map of the DKP, by selecting the option Personalized and assign
functions to keys as per your requirement.
To know more about key maps, key templates and how to customize them, see DSS Keys
Programming.
Call Pick-Up Group: Program this parameter if you want to assign the DKP to a particular Call Pick
Up group.
Call Pick Up allows the DKP extension user to 'pick up' (answer) calls ringing on any other extension,
by dialing a feature code, without physically going to the ringing extension. For this to work, both the
ringing extension and the extension picking up the call must be in the same 'Call Pick Up Group'. Refer
Call Pick Up for instructions on how to create groups. You can create as many as 99 groups,
numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this DKP in this field.
Station Type: This parameter is relevant only for the Hotel Application of the ETERNITY. Extensions
are identified as 'Administrator' or 'Guest' extensions according to the intended user of the DKP. When
the Station Type is selected, the system will automatically assign the Guest and Administrator (Hotel
Staff) features to the DKP. To know more, refer the ETERNITY Hospitality System Manual.
For instructions on configuring DSS Keys, DSS1 Keys and DSS2 Keys, see the topic DSS Keys Programming
Advanced Configuration
The above listed parameters fulfill the basic DKP extension port configuration requirements of most users.
However, it is anticipated that some users may need to configure the more advanced features like Personal
Directory, Language Selection, DKP Settings - Backlight Brightness and Contrast, Headset/Handset/Speaker
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Volume levels, Select a Ringer Tune, attach Direct Station Selection Consoles, etc. For such users, you may click
the 'Advanced' button.
Mobile Number: Enter the Mobile Number of the extension user you wish to store. The Number can be a
maximum of 16 digits.
Email ID: Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of
64 characters.
Group: You can assign the extension user to a Group. The system clubs together extension users
assigned the same Group. The Group can be a maximum of 16 characters. Default: Blank.
Personal Directory: Enter the number of the Personal Directory that you want to assign to the DKP. A
Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (OG Trunk Bundle Group). The Personal Directory is
necessary for using the features Abbreviated Dialing and Dial By Name.
To be able to assign a Personal Directory to a DKP you must first program it. Refer the topic Abbreviated
Dialing for instructions on programming the Personal Directory.
Priority: Select a Priority Level for the DKP from 1, 2, 3... to 9, with '1' being lowest Priority and '9' being
highest Priority. Whenever an extension (DKP) with higher priority calls an extension with lower priority, a
triple ring is placed on the called extension. To know more, read the feature description Priority.
By default, the Priority of all DKP ports is set to '5-Normal'.
CO CLIP Pattern: This parameter allows you to select the type of Calling Line Presentation on the DKP for
incoming calls from trunks. You can select any of the below options:
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Number + Name: both the name and the number of the caller will be displayed.
By default, Number + Name is selected as the CO CLIP Pattern for all DKPs.
Language: ETERNITY provides language support for English, French, German, Spanish, Portuguese,
and Italian. When you select any of these languages, all the command strings and prompts will appear in
the selected language. By default English is selected.
DKP users can change the language by accessing and navigating through the phone menu.
The SA can change the Language by logging into the SA Jeeves.
Ringer Mode: You can select a Ringer mode for each DKP from the four options:
Ring immediately (it rings immediately as a fresh calls lands on the DKP).
Ring if idle (rings only if the DKP is idle).
Ring after a delay (if the call is still not answered).
Silent.
By default the Ringer Mode is set to 'Ring Immediately'. Change the Ringer Mode on the DKP as per the
requirement and preferences of the DKP users.
Ring Delay Timer: The Ring Delay Timer is the time in seconds the ETERNITY will wait to ring on
receiving a call. This Timer needs to be set only if you have selected 'Ring after a delay' as the Ringer
Mode for the DKP in the previous parameter.
By default, the Ringer Delay Timer is set to 10 seconds. You may change the Ring Delay Timer according
to the preferences and requirements of the DKP user.
Acknowledge Timer: This Timer is to be programmed to enable the Ringer Auto Acknowledge mode.
This mode determines when to stop the ring on the DKP. There are two options for ringer auto
acknowledge:
To stop the ring on the DKP after a delay, the Acknowledge Timer must be programmed. The range of the
timer is 01 to 99 seconds.
To stop the ring only when the Call is answered or manually acknowledged, the Acknowledge Timer must
be set to '00'. By default, Ring Auto Acknowledge is turned OFF.
Play Ring ON: With this parameter, you can assign the Ring Destination for the DKP; you can choose
whether the Ring should be played on the Speakerphone or Headset of the DKP. Default: Speakerphone is
selected.
When you select the Headset as the destination, ensure that you have enabled Headset Connectivity flag
and have connected a Headset to the DKP.
The speech path of both, the Headset and the Handset is common. If the Headset is not connected and
you have selected the Headset as the ring destination, the ring will be played on the speaker of the
Handset.
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Ring Tune: You can select from different ringer tunes for each DKP according to your preferences and
requirement. By default, Ringer Tune is set to 1.
Ring Volume: You can select from different ringer volumes for each DKP according to your preferences
and requirement. By default, Ringer Volume is set to 5.
Handset Transmit Volume Level: This parameter is used for increasing or decreasing the volume of
outgoing speech (Transmit Gain) on the Handset of the DKP. Select the desired Handset Tx Volume Level
from 0 to 9. By default, Handset Tx Volume Level is 4.
Handset Receive Volume Level: This parameter is used for increasing or decreasing the volume of
incoming speech (Receive Gain) on the Handset of the DKP. Select the desired Handset Rx Volume Level
from 0 to 9. By default, Handset Rx Volume Level is 4.
Headset Transmit Volume Level: This parameter is used for increasing or decreasing the volume of
outgoing speech (Transmit Gain) on the Headset port of the DKP. Select the desired Headset Tx Volume
Level from 0 to 9. By default, Headset Tx Volume Level is 4.
Headset Receive Volume Level: This parameter is used for increasing or decreasing the volume of
incoming speech (Receive Gain) on the Headset port of the DKP. Select the desired Headset Rx Volume
Level from 0 to 9. By default, Headset Rx Volume Level is 4.
Right Handset Transmit Volume Level (EON74): This parameter is applicable only when EON74 is
connected to the DKP Port. This parameter is used for increasing or decreasing the volume of outgoing
speech (Transmit Gain) on the Right Handset of EON74. Select the desired Tx Volume Level from 0 to 9.
By default, the Right Handset Tx Volume Level is 4.
Right Handset Receive Volume Level (EON74): This parameter is applicable only when EON74 is
connected to the DKP Port. This is This parameter is used for increasing or decreasing the volume of
incoming speech (Receive Gain) on the Right Handset of EON74. Select the desired Headset Rx Volume
Level from 0 to 9. By default, the Right Handset Rx Volume Level is 4.
Hands-free Transmit Volume Level: With this parameter you may change the Volume level of Transmit
Gain of the Speaker phone MIC volume from 0 to 9, as per your preference. This parameter is to be used
for increasing or decreasing the volume levels of outgoing speech on the Speaker of the DKP. By default,
Hands-free Tx volume level is 4.
Hands-free Receive Volume Level: With this parameter you may change the Volume level of Receive
Gain of the Speaker phone MIC volume from 0 to 9, as per your preference. This parameter is to be used
for increasing or decreasing the volume levels of incoming speech on the Speaker of the DKP. By default,
Hands-free Rx volume level is 4.
Key Click Volume Level: You may change the Key Click Volume (Key DTMF Side tone) of the DKP. Key
Click Volume is the tone you hear as you press the dial pad keys of EON. Select the desired volume level
from 0 to 9. By default, the volume level is set to 5.
DTMF Generation Flag: This flag is used to enable or disable DTMF dialing on the DKP. By default, the
flag is enabled. To disable the flag, click the checkbox.
DTMF Transmit Level: You can select the desired Transmit Level from 0 to 9 for DTMF generation from
the DKP. By default, the DTMF Transmit Level is set to 2.
Headset Connected?: Enable this parameter by selecting the checkbox if you want to use a Headset with
the DKP.
Make sure that you have also connected a Headset to the DKP.
Auto Answer: Enable this parameter by selecting the checkbox if you want to set the Auto Answer
feature on the DKP.
When this feature is set, the DKP goes OFF-Hook automatically after a preset period of time, without the
user having to pick up the handset or press the speaker or headset key.
Auto Answer Timer (sec): This parameter is to be programmed if you have enabled the Auto Answer
feature on the DKP.
When the Auto Answer feature is enabled, the Auto Answer Timer must be programmed. This timer
defines the time in seconds that the DKP should wait before going OFF-Hook. The range of this timer is 1
to 9 seconds. By default, the Auto Answer Timer is set to 1 second.
LCD Backlight Level: You can change the LCD Backlight Brightness of EON. The intensity of the
backlight brightness increases from 0 to 4, where '0' will cause the backlight to be turned OFF. '1' signifies
minimum intensity, '4' signifies maximum intensity. Select any of the levels from 1 to 4 from the list.
LCD Backlight OFF Timer: The backlight of the LCD display of EON can be kept switched ON
continuously, or can be set to switch OFF automatically after a predefined period of time, known as the
Backlight OFF Timer. The range of the Backlight OFF Timer is 000-999 seconds. By default it is set to
switch OFF after 010 seconds.
LCD Contrast Level: The EON offers 4-level contrast control for its LCD display. Level 1 signifies
minimum and level 4 is the maximum. The contrast increases in steps of 1 to 4. By default the contrast is
set to level 3. You may adjust it to the level comfortable to you. Select a level from 1-4.
Line Echo Cancellation (LEC): Enable this parameter to cancel the echo generated on the other end.
You must also configure the Line Echo Cancellation Start Timer. When DKP users go Off-Hook to make/
receive calls, the Line Echo Cancellation Timer starts. After the expiry of this timer the ETERNITY will start
Line Echo Cancellation. To configure the Echo Cancellation Start Timer, see System Timers and Counts.
CPLD Version: This parameter displays the CPLD Version of the Digital Key Phone Card.
You are recommended to enable Line Echo Cancellation only if you hear echo after you have set the
AC Impedance value in the CO Hardware Template according to the Accurate Test conducted by you.
For details see, AC Impedance Test and CO Hardware Template.
The Call Progress Tones heard by the DKP user may be affected, if you enable LEC.
LEC is supported only in EON48D, Version V3R15 or later and in EON310.
The CPLD Version will be displayed only if supported by your DKP Card.
Current Firmware: This parameter displays the current firmware of the Digital Key Phone connected to
the DKP Port.
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Upgrade Firmware: Enable this parameter by selecting the checkbox, if you want to upgrade the firmware
of the Digital Key Phone connected to the ETERNITY.
You can upgrade firmware of any two DKP phones at a time. Before upgrading, make sure both these
phones are in idle state. You cannot use these phones while their upgradation is in process.
You can view and Upgrade the current firmware of the DKP phone only from Jeeves.
DSS1 H/w Slot-Port: This parameter is to be configured only if you have attached a Direct Station
Selection Console with the DKP.
You can attach two Direct Station Selection Console to each DKP to increase the number of DSS keys
for Direct Station Calling. You may attach any two DSS Consoles of the same model or of two different
models to the DKP.
DSS Consoles are connected to the DKP Cards just like digital key phones (refer the topic Installing DSS
Consoles in the installation instructions for your model of ETERNITY.
If you have attached a single DSS Console to a DKP, you must enter the following information in the 'DSS1
H/w-Slot to Port' column:
the number of the Universal Slot in which the DKP card, to which the DSS Console is connected, is
located.
the number of the Port on the card, on which the DSS Console is connected.
DSS2 H/w Slot-Port: To configure a second DSS Console attached to the DKP, enter the same
information as above here, that is, the number of the Universal Slot in which the DKP card to which the
DSS Console is located, and the number of the Port on the card, on which the DSS Console is connected.
The DSS Console you want to attach to a particular DKP must not necessarily be located on the same card
as the DKP to which it is attached.
For example, you want to attach two DSS Consoles to DKP Port Number 001 in Slot number 18. The first
DSS Console is connected to DKP Port Number 01 on the card which is occupying Slot number 02. The
second DSS Console is connected to the DKP Port Number 02 on the same card. Now, to attach the DSS
Consoles to the DKP connected on Port Number 001, enter the hardware ID and port number as follows:
Enter 02 and 01 as the hardware ID and port number in the column DSS1. Similarly, enter 02 and 02 as
the hardware ID and port number in the column DSS2 of the DKP port.
DKP
H/w-Slot to Port
DSS1
H/w-Slot to Port
DSS2
H/w-Slot to Port
001
17
01
02
02
002
17
02
003
17
03
004
17
04
005
17
05
006
17
06
007
17
07
DKP No.
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01
02
DKP No.
DKP
H/w-Slot to Port
008
17
08
009
02
01
010
02
02
DSS1
H/w-Slot to Port
DSS2
H/w-Slot to Port
If you have completed configuration of all the above listed DKP Parameters, click Submit to save your
changes.
Configure the following voice mail settings for the DKP extensions you wish to assign voice mail.
DKP Port No.: This is the software port number of the DKP extension.
H/w Slot - Port: It is the hardware slot and port number assigned to the DKP software port (that is, the
hardware slot and port number to which the DKP extension is connected).
Access Code: This is the access code (that is, extension number) assigned to the DKP port in the
DKP Parameters. If you have changed the default Access Codes in the DKP Parameters page, the
same will appear here.
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Name: This is the name assigned to the DKP port in the DKP parameters.
Personal Mailbox: Select the checkbox to assign Personal Mailbox to the DKP extension. By default,
Personal Mailbox is not assigned to extensions in Enterprise mode.
In the Hotel mode, Personal mailbox is assigned to all extension phones, by default.
Mailbox Size (min.): You may increase or decrease the size of the personal mailbox assigned to the
DKP extension, by changing the default Mailbox size of 5 minutes. You may change the mailbox size to
any desired value from 00001 to 60000 minutes. Default: Enterprise mode: 5 minutes; Hotel mode: 999
minutes.
Maximum Message Length (sec): You can define the length of each message (in seconds) callers
are to be allowed to record in the mailbox. You may change the maximum message length to any
desired value from 0001 to 9999 seconds. Default: Enterprise mode: 15 seconds; Hotel mode: 120
seconds.
The VMS card will stop recording the message of the callers if it exceeds the maximum message
length, and will store only that part of the message recorded within the maximum message length limit.
New Message Delivery Option in Mailbox Full condition: When the personal mailbox is full, you
may select one of the following options for delivery of new messages:
Do not offer to record a message: The VMS will not allow the caller to record a message by
declining delivery of the message.
Deliver new message to General Mailbox: The VMS will record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their Class of Service (COS) are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the DKP extension. Refer Class of Service (COS) for instructions.
Overwrite old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS starts overwriting the oldest message first.
By default, Deliver to General mailbox is selected.
Play message details after delivery of message: After the extension user has finished listening to a
message in the mailbox, you can also have the VMS play message details such as Date and Time
when the message was recorded, the callers number149, and the extension number dialed by the
caller150 to the extension user.
You may select from one of the following options for Play message details:
Never: The VMS will not play message details to the mailbox owner after playing the message.
Always: The VMS will play message details to the mailbox owner after playing each message.
149. The number of person who left the message in the mailbox.
150. The number of the extension user for whom the message is intended.
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On Demand: The VMS will play message details to the mailbox owner only when the mailbox
owner requests it. On completion of each message, the VMS will prompt the extension user to
press a digit for date and time stamp. When the mailbox owner presses the digit, the VMS will play
the message details.
Ask Password to access Mailbox: By default, access to the mailbox is password protected. The
User Password is required to access the mailbox. Whenever the mailbox owner accesses the
mailbox, the VMS will ask for the (user) password.
If you want to remove password protection, clear this checkbox.
Since a Mailbox can be accessed using the default User Password, 1111, extension users who are
assigned a mailbox are recommended to change their User Password to a unique 4 digit number to
prevent unauthorised access to their mailbox.
Allow Mailbox Management: Mailbox Management allows the extension user to change mailbox
settingsrecord Extension Name for the mailbox, redirect messages from the mailbox, delete all old
messages from the mailbox, record greeting messages for the mailbox. By default, Mailbox
Management is enabled.
If you do not want to allow the extension user to change the mailbox settings, in Allow Mailbox
Management, select No.
To know more about this feature, see Mailbox Settings
Auto Delete Messages: Select the type of messages you want the VMS to automatically delete from
your mailbox. You can select All or Old. Default: None.
Days for Auto Delete Messages: Select the number of days after which you want the VMS to
automatically delete the messages in your mailbox. Default: 90 days.
Department Group Mailbox: You can assign the Mailbox of a Department Group to DKP extensions,
even to those DKP extensions that are not included in the Department Group. Refer the topic
Department Call.
To assign the Department Group mailbox to a DKP extension, select the number of the Department
Group (1 to 16) from the combo box.
If you do not want to assign Department Group mailbox to a DKP extension, select None.
Default: None.
Voice Mail Auto Attendant Features: This parameter is applicable only if you are using the VMS Auto
Attendant.
Voice Mail Auto Attendant Profile: Select a profile for the DKP extension. The profile determines
the welcome message to be played to mailbox owners when they reach the home node. It also
determines whether or not the user should be taken to the root node directly.
Abbreviated Name: When the VMS is used as Auto Attendant, the callers can be prompted to Dial
by Name of the desired extension users instead of their extension numbers.
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To allow callers to reach the DKP extension using Dial By Name, abbreviate the DKP extension
users name to three letters and enter it in this field.
Announce Name: If you want the VMS to announce the extension users name to the caller when
transferring the call to the extension, select the checkbox to enable Announce Name. By default,
Announce Name is disabled.
If you enable Announce Name, make sure you record the extension users name on the VMS. Refer
Recording Extension Names for instructions.
Call Transfer: Select the desired method for transferring the call answered by the VMS Auto
Attendant to the DKP extension. You may select any of the following methods of call transfer for
each time zone, Working Hours (WH), Break Hours (BH) and Non-working Hours (NH):
None: When the caller dials the extension number, the VMS Auto Attendant will check if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
Blind: When the caller dials the extension number, the VMS Auto Attendant will transfer the call
on the extension without checking whether it is busy or free.
Wait for Ring: When the caller dials the extension number, the VMS Auto Attendant will wait for
the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the
extension, if assigned, or take the caller back to the home node.
Wait for Answer: When the caller dials the extension number, the VMS Auto Attendant will
transfer the call when the extension answers (goes OFF-Hook).
If the extension does not answer151, the VMS Auto Attendant will transfer the call to the mailbox
of the extension, if assigned, or take the caller back to the home node.
Screen: The VMS Auto Attendant prompts the caller to record his/her name. It puts the caller on
hold and places the call on the desired extension. If the extension is free and answers the call,
the VMS announces the callers name to the extension user and prompts the extension user to
choose whether or not to speak to the caller. If the extension user chooses to talk, the VMS
transfers the call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the
extension user, if assigned, and asks the caller to leave a message.
By default, Wait for Answer is selected as Call Transfer method for all time zones.
Message Wait Indication: This parameter allows you to select the type of indication to be given to the
extension user for new messages in the mailbox and message wait set by another extension user.
You can select from any from the type of indicators described below for new messages:
Stuttered Dial Tone/Voice Message: When the extension user goes OFF-Hook, s/he will hear a
voice message, if a pre-recorded Voice Module has been assigned for Message Wait Notification. If
151. The VMS will wait for the duration of the Built-In Auto Attendant Ring Timer (default: 30 seconds; the timer is configurable). If the
call is not answered before this timer expires, it is treated as No Reply.
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no voice module is recorded and assigned, the extension user will hear a stuttered dial tone
instead.
If you want voice message to be played as message wait notification, record and assign a Voice
Module. Refer Voice Message Applications for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait Notification
will not be played if there are already 4 being played simultaneously. In which case, Stuttered Dial Tone will
be played for Message Wait Notification, when the extension user goes OFF-Hook.
Ring: The extension will ring for the duration of the Message Wait Ring Timer (configurable;
default: 30 seconds), for as many times as the Message Wait Ring Count (configurable; default: 10
times), at the interval set as the Message Wait Ring Timer Interval (configurable; default: 30
minutes).
Message Wait Notification via Call: The message wait notification will be sent to a number
(destination number). This number can be an internal or an external number. To use this feature,
configure the following parameters:
Type: If you want the notifications to be sent as soon as a new message arrives in the mailbox of
the extension user, select Immediate.
If you want the notification to be sent at fixed time schedules, select Scheduled.
If do not want to set message wait notification via call, select None. Default: None.
Profile: Assign the Profile according to which you want the system to send the notifications. The
Profile determines the time intervals during which the notifications must be sent to the destination
number.
Destination Number: Enter the number on which you want the system to send the notification
calls.
The destination number can be an internal or an external number. The destination number can be a
maximum of 16 digits. Valid digits are 0 to 9, # and *.
When the notification call is answered, the VMS informs the callee of the new message and allows
the callee to access it.
Refer the feature description Message Wait Notification via Call to know more.
Message Wait Notification via Email: The message wait notification will be sent to the email address
of the extension user. To use this feature, configure the following parameters:
Notification: If you want the message wait notification to be mailed to the extension user along with
the new voice message as attachment, select the option Send With Attachment.
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If you want only the notification to be mailed, select the option Send Without Attachment.
If do not want to set message wait notification via email, select Do not send. Default: Do not send.
Email Address: Enter the email ID of the extension user to which the notification is to be sent.
Email ID may consist of up to 64 characters. Default: blank.
Extensions users will receive notifications only for the mailbox memory utilization, if you configure the E-mail
Address and select Do not sent as the Notification option.
Refer the feature description Email Based Notification to know more.
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Name is a string of a maximum of 18 alphanumeric characters. Terminate the commands with #* if the
number string has fewer than 18 characters.
To clear the Name of the DKP Port, dial:
5403-1-DKP-#* to clear the Name of a single DKP port.
5403-2-DKP-DKP-#* to clear the Names of a range DKP ports.
5403-*-#* to clear the Names of all DKP ports.
To assign a Station Basic Feature Template to a DKP Port, dial:
5504-1-DKP-Template Number to assign a template to a single DKP port.
5504-2-DKP-DKP-Template Number to assign the same template to a range of DKP ports.
5504-*-Template Number to assign the same template to all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Template Number is the number of the Station Basic Feature Template, from 01 to 50. Default: 01
781
For Advanced Configuration of the DKP Ports, use the following commands:
To assign a Personal Directory to a DKP Port, dial:
1906-1-DKP-Personal Directory to assign directory to a single DKP port.
1906-2-DKP-DKP- Personal Directory to assign the same directory to a range of DKP ports.
1906-*-Personal Directory to assign the same directory to all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Personal Directory number is from 01 to 50.
To clear the Personal Directory assigned to a DKP Port, dial:
1906-1-DKP-00 to clear the directory of a single DKP port.
1906-2-DKP-DKP-00 to clear the directory of a range of DKP ports.
1906-*-00 to clear the directory from all DKP ports.
To define the Priority for a DKP Port, dial:
3912-1-DKP-Priority to define Priority for a single DKP port.
3912-2-DKP-DKP-Priority to define the same Priority for a range of DKP ports.
3912-*-Priority to define the same Priority for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Priority is from 1 to 9. Default: 5-Normal
To select CO CLIP Pattern for a DKP Port, dial:
1243-1-DKP-CO CLIP Pattern to select the CLIP Pattern for a single DKP port.
1243-2-DKP-DKP-CO CLIP Pattern to select the same CLIP Pattern for a range of DKP ports.
1243-*-CO CLIP Pattern to select the same CLIP Pattern for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
CO CLIP Pattern is
1 for Name only
2 for Number only
3 for Number+Name
Default: Number+Name
To select Language for a DKP port, dial:
1224-1-DKP-Language to select language for a single DKP port.
1224-2-DKP-DKP-Language to select the same language for a range of DKP ports.
1224-*-Language to select the same language for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Language is
1 for English
2 for Franais
3 for Deutsch
4 for Espaol
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5 for Portugus
6 for Italian
Default: English
To select Ringer Mode for a DKP Port, dial:
1204-1-DKP-Ringer Mode to select Ringer mode for a single DKP port.
1204-2-DKP-DKP-Ringer Mode to select Ringer mode for a range of DKP ports.
1204-*-Ringer Mode to select Ringer mode for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Ringer Mode is
1 for Ring Immediate
2 for Ring if Idle
3 for Ring after Delay
4 for Ring OFF
Default: Ring Immediate
To select Ring Delay Timer for a DKP Port, dial:
1205-1-DKP-Ring Delay Timer to select Delay Timer for a single DKP port.
1205-2-DKP-DKP- Ringer Delay Timer to select Delay Timer for a range of DKP ports.
1205-*- Ringer Delay Timer to select Delay Timer for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Ring Delay Timer is 01 to 99 seconds. Default: 10 seconds.
To set the Acknowledgement Mode, dial:
1206-1-DKP-Ringer Auto Acknowledge Mode to select mode for a single DKP port.
1206-2-DKP-DKP-Ringer Auto Acknowledge Mode to select the same mode for a range of DKP
ports.
1206-*-Ringer Auto Acknowledge Mode to select the same mode for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Ringer Auto Acknowledge Mode is
0 for OFF
1 for ON
To set the Ringer Auto Acknowledge Timer, dial:
1207-1-DKP-Ringer Auto Acknowledge Timer to set Timer for a single DKP port.
1207-2-DKP-DKP-Ringer Auto Acknowledge Timer to set the same Time for a range of DKP ports.
1207-*-Ringer Auto Acknowledge Timer to set the same Time for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Ringer Auto Acknowledge Timer is from 01 to 99 seconds. Default: 00.
To select Destination for 'Play Ring ON' for a DKP Port, dial:
1220-1-DKP-Ring Destination to select Play Ring ON destination for a single DKP port.
1220-2-DKP-DKP-Ring Destination to select the same Play Ring ON destination for a range of DKP
ports.
1220-*-Ring Destination to select the same Play Ring ON destination for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Ring Destination is
1 for Play Ring on Speaker Phone
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Setting volume level to '0' will cause the Handset Speaker volume to be turned OFF.
To set Right Handset Receive (Rx) Volume Level for a DKP, dial:
1226-1-DKP-Handset Speaker Volume Level to set receive volume for a single DKP port.
1226-2-DKP-DKP-Handset Speaker Volume Level to set the same receive volume level for a range
of DKP ports.
1226-*-Handset Speaker Volume Level to set the same receive volume level for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Handset Speaker Volume Level from 0 to 9. Default: 5.
Setting volume level to '0' will cause the Handset Speaker volume to be turned OFF.
To set Headset Transmit (Tx) Transmit Volume Level for a DKP, dial:
1222-1-DKP-Headset MIC Volume Level to set receive volume for a single DKP port.
1222-2-DKP-DKP- Headset MIC Volume Level to set the same receive volume level for a range of
DKP ports.
1222-*- Headset MIC Volume Level to set the same receive volume level for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Headset MIC Volume Level is from 0 to 9. Default: 5.
To set Headset Receive (Rx) Volume Level for a DKP, dial:
1223-1-DKP-Headset Speaker Volume Level to set receive volume for a single DKP port.
1223-2-DKP-DKP- Headset Speaker Volume Level to set the same receive volume level for a range
of DKP ports.
1223-*- Headset Speaker Volume Level to set the same receive volume level for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Headset Speaker Volume Level is from 0 to 9. Default: 5.
To set Hands-free Transmit (Tx) Volume Level for a DKP, dial:
1210-1-DKP-Speaker Phone MIC Volume Level to set volume level for a single DKP port.
1210-2-DKP-DKP-Speaker Phone MIC Volume Level to set the same volume level for a range of
DKP ports.
1210-*-Speaker Phone MIC Volume Level to set the same volume level for all DKP ports.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Speaker Phone MIC Volume Level from 0 to 9. Default: 5.
Setting volume level to '0' will cause the Speaker Phone MIC volume to be turned OFF.
785
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Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Auto Call Answer Timer is from 1 to 9 seconds. Default: 3 seconds.
To set LCD Back Light Level of a DKP, dial:
1216-1-DKP-LCD Backlight Level to set brightness of a single DKP.
1216-2-DKP-DKP-LCD Backlight Level to set the same brightness level for a range of DKPs.
1216-*-LCD Backlight Level to set the same brightness level for all DKPs.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Backlight Level is Brightness level from 1 to 4. Default: 3
To change LCD Backlight OFF Timer of a DKP, dial:
1219-1-DKP-LCD Backlight OFF Timer to change the timer for a single DKP.
1219-2-DKP-DKP-LCD Backlight OFF Timer to set the same timer duration for a range of DKPs.
1219-*-LCD Backlight OFF Timer to set the same timer duration for all DKPs.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Backlight OFF Timer is from 000-999 seconds. Default: 010 seconds.
To change LCD Contrast Level of a DKP, dial:
1217-1-DKP-LCD Contrast Level to set the Contrast level for a single DKP
1217-2-DKP-DKP-LCD Contrast Level to set the same Contrast level for a range of DKPs.
1217-*-LCD Contrast Level to set the same Contrast level for all DKPs.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
LCD Contrast Level is from 1 to 4. Default: 3.
To assign Hardware Slot-Port to DSS connected to a DKP, dial:
1103-DKP-DSS-Slot-Port offset on the card
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
DSS is 1 for the first DSS Console and 2 for the second DSS Console attached to the DKP
Slot is the number of the Universal Slot in which the DKP card is located, to which the DSS Console is
connected, from 01 to 16.
Port is the number of the hardware port on the card on which the DSS Console is connected.
To clear the hardware Slot-Port assigned to the DSS software port, dial:
1103-DKP-DSS-00-00
Exit SE mode.
DKP extension users can change several phone settings according their preferences and requirement.
These are referred to as DKP Personal Settings and include:
Ringer Volume
Ringer Tune
Ringer Mode
Ringer Acknowledge Mode
Speech Volume (Transmit/Receive)
User Status (Present/Absent)
Keypad Security (Lock/Open)
Call Answer Type - Manual/Auto
Headset/Handset Connectivity option
787
To be able change the DKP personal settings, the DKP users must access and navigate the phone menu.
Refer Digital Key Phone-Operation.
788
789
790
2 - LED Lamp
3 - Ring
Default: LED Lamp.
To enable/disable Message Notification through E-mail for a mailbox, dial:
1283-1-DKP-Code-#* to enable/disabled Message Notification through E-mail for a mailbox on a single
DKP extension.
1283-2-DKP-DKP-Code-#* to enable/disabled Message Notification through E-mail for a mailbox on a
range of DKP extension.
1283-*-Code-#* to enable/disabled Message Notification through E-mail for a mailbox on all the DKP
extension.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Code is:
0 - Disabled
1 - With Attachment
2 - Without Attachment
Default - Disabled.
To program E-mail ID for a mailbox, dial:
1284-1-DKP-E-mail ID-#* to assign an E-mail ID for a mailbox to a single DKP extension.
1284-2-DKP-DKP-E-mail ID-#* to assign an E-mail ID for a mailbox to a range of DKP extensions.
1284-*-E-mail ID-#* to assign an E-mail ID for a mailbox to all DKP extensions.
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Where E-mail ID is 64 ASCII Characters.
Default: Blank.
Exit SE mode.
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792
Configure the following parameters for each ISDN Terminal on this page:
ISDN Terminal: This non-editable field is the number of the software port of the ISDN Terminal.
BRI Port: This is the number of the BRI Software port to which the ISDN Terminal is connected on a
Short/Extended Passive Bus line.
Access Code: Assign Station Access codes to the ISDN Terminal. Station Access codes are
commonly referred to as Extension Numbers. These may be a combination of a maximum of 6 digits,
which are dialed to call the ISDN Terminal to which they are assigned.
A maximum of 6 digits are allowed in an Access Code. By default, the Station Access Codes are blank
for all ISDN Terminal ports.
To assign Station Access Codes according to your preference and requirment to a range of ISDN
Terminals, see Assigning Access Codes to a Range of Extensions.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics Access Codes and Conflict Dialing to know
more.
Name: Assign a 'Name' to the ISDN Terminal. The name may be of the person who will use the ISDN
Terminal or the name of a department. This name will be displayed to the called extension.
You can program a name of maximum 18 alphanumeric characters.
Station Basic Feature Template: Assign a Station Basic Feature Template to the ISDN Terminal.
A Station Basic Feature Template includes a set of features that completely define the behaviour of the
Extension, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call
Budget, and Station Message Detail Records (storage of Incoming and Outgoing Call details).
By default, Station Basic Feature Template 01 is assigned to all extensions of the system that also
includes SLT ports, ISDN ports and E&M Lines with Station as Orientation Type.
Check if the default template fulfills the feature requirements (like Class of Service (COS), Toll
Control, OG Trunk Bundle Group, etc.) of the ISDN Terminal.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all ISDN Terminals, retain Template 01.
If different sets of features are to be allowed to different ISDN Terminals, then prepare separate Station
Basic Feature Templates and apply them on the ports. To do this,
Under Configuration, click Station Basic Feature Template to open the page.
Customize Template number 12 and click 'Submit' at the bottom of the page.
Enter the number of the Template you customized, Template 12, in the Station Basic Feature
Template field of the ISDN Terminal, for instance, ISDN-01, on which you want to apply this
793
template. If you want to apply this template to other terminals too, like ISDN-02, 03, and 04, assign
the Template 12 to all them.
Also, refer the topic Station Basic Feature Template to know more about customizing the templates
and applying on extension ports.
Station Advanced Feature Template: Assign a Station Advanced Feature Template to the ISDN
Terminal. The Advanced Feature Template consists of a set of less commonly used features like Message
Wait Notification Type, Alarm Notification Type, Caller ID Presentation for Transferred Calls, DDI Incoming
Call Routing, Storage of Internal Calls, Call Duration Control, Floor Service, Call Taping, etc.
By default Station Advanced Feature Template 01 is assigned to all extensions of the ETERNITY, which
includes ISDN ports, SLT ports, and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the ISDN Terminal by clicking the 'Station
Advanced Feature Template' link.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to all
ISDN Terminals, retain Template 01.
If different sets of features are to be allowed to different ISDN Terminals, then prepare separate Station
Advanced Feature Templates and apply them on the ports.
To do this,
Under Configuration, click Station Advanced Feature Template to open the page.
Enter the number of the Template you customized, Template 04 in the Station Advanced Feature
Template field of the ISDN Terminal, for instance, ISDN-01, on which you want to apply this template.
If you want to apply this template to other terminals too, like ISDN-01, 02, and 03, assign the Template
04 to all these ports.
Repeat the same steps to customize and assign a different Template to another ISDN Terminal.
Also refer the topic Station Advanced Feature Template for instructions on customizing these templates
and applying them on the extension ports.
794
Station Type: This parameter is relevant only for the Hotel Application of the ETERNITY. Extensions are
identified as 'Administrator' or 'Guest' extensions according to the intended user of the ISDN Terminal.
When the Station Type is selected, the system will automatically assign the Guest and Administrator (Hotel
Staff) features to the ISDN Terminal. To know more, refer the ETERNITY Hospitality System Manual.
Advanced features
The above listed parameters fulfill the ISDN Terminal configuration requirements of most users. If you need to use
other features like Personal Directory, Call Pick-Up Groups, for some users, you may click the Advance button.
Mobile Number: Enter the Mobile Number of the extension user you wish to store. The Number can be a
maximum of 16 digits.
Email ID: Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of
64 characters.
Group: You can assign the extension user to a Group. The system clubs together extension users
assigned the same Group. The Group can be a maximum of 16 characters. Default: Blank.
Personal Directory: Enter the number of the Personal Directory that you want to assign to the ISDN
Terminal. A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index
number (location code), Name and Trunk Access Codes (OG Trunk Bundle Group). The Personal
Directory is necessary for using the features Abbreviated Dialing and Dial By Name.
The Personal Directory number that you assign to an ISDN Terminal must also be programmed either by
you, the System Engineer, or by the ISDN Terminal user. Refer the topic Abbreviated Dialing for
instructions on programming the Personal Directory.
Priority: Select a Priority Level for the ISDN Terminal from 1, 2, 3... to 9, with '1' being lowest Priority and
'9' being highest Priority. Whenever an extension (ISDN Terminal) with higher priority calls an extension
795
with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description Priority.
By default, the Priority of all ISDN Terminals is set to '5-Normal'.
Call Pick-Up Group: Configure this parameter if you want to assign the ISDN Terminal to a particular Call
Pick Up group.
Call Pick Up allows the ISDN Terminal user to 'pick up' (answer) calls ringing on any other extension, by
dialing a feature code, without physically going to the ringing extension. For this to work, both the ringing
extension and the extension picking up the call must be in the same 'Call Pick Up Group'. Refer Call Pick
Up for instructions on how to create groups. You can create as many as 99 groups numbered from 01 to
99.
Enter the number of the Call Pick-Up Group you created for this ISDN Terminal in this field.
If you have completed configuration of all the above listed ISDN Terminal Parameters, click 'Submit' at the
bottom of the page to save your changes.
796
Configure the following voice mail settings for the ISDN extensions you wish to assign voice mail.
ISDN Terminal No.: This is the software port number of the ISDN extension.
BRI Port: This is the number of the BRI Software port to which the ISDN Terminal is connected on a
Short/Extended Passive Bus line.
Access Code: This is the access code (that is, extension number) assigned to the ISDN port in the
ISDN Parameters. If you have changed the default Access Codes in the ISDN Terminal Parameters
page, the same will appear here.
Name: This is the name assigned to the ISDN port in the ISDN Terminal parameters.
Personal Mailbox: Select the checkbox to assign Personal Mailbox to the ISDN terminal extension.
By default, Personal Mailbox is not assigned to extensions in Enterprise mode. In the Hotel mode,
Personal mailbox is assigned to all extension phones, by default.
Mailbox Size (min.): You may increase or decrease the size of the personal mailbox assigned to the
ISDN extension, by changing the default Mailbox size of 5 minutes. You may change the mailbox size
to any desired value from 00001 to 60000 minutes. Default: Enterprise mode: 5 minutes; Hotel mode:
999 minutes.
797
Maximum Message Length (sec): You can define the length of each message (in seconds) callers
are to be allowed to record in the mailbox. You may change the maximum message length to any
desired value from 0001 to 9999 seconds. Default: Enterprise mode: 15 seconds; Hotel mode: 120
seconds.
The VMS card will stop recording the message of the callers if it exceeds the maximum message
length, and will store only that part of the message recorded within the maximum message length limit.
New Message Delivery Option in Mailbox Full condition: When the personal mailbox is full, you
may select one of the following options for delivery of new messages:
Do not offer to record a message: The VMS will not allow the caller to record a message by
declining delivery of the message.
Deliver new message to General Mailbox: The VMS will record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their Class of Service (COS) are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the ISDN terminal extension. Refer Class of Service (COS) for instructions.
Overwrite old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS starts overwriting the oldest message first.
By default, Deliver to General mailbox is selected.
Play message details after delivery of message: After the extension user has finished listening to a
message in the mailbox, you can also have the VMS play message details such as Date and Time
when the message was recorded, the callers number152, and the extension number dialed by the
caller153 to the extension user.
You may select from one of the following options for Play message details:
Never: The VMS will not play message details to the mailbox owner after playing the message.
Always: The VMS will play message details to the mailbox owner after playing each message.
On Demand: The VMS will play message details to the mailbox owner only when the mailbox
owner requests it. On completion of each message, the VMS will prompt the extension user to
press a digit for date and time stamp. When the mailbox owner presses the digit, the VMS will play
the message details.
Ask Password to Access Mailbox: By default, access to the mailbox is password protected. The
User Password is required to access the mailbox. Whenever the mailbox owner accesses the
mailbox, the VMS will ask for the (user) password.
152. The number of person who left the message in the mailbox.
153. The number of the extension user for whom the message is intended.
798
Allow Mailbox Management: Mailbox Management allows the extension user to change mailbox
settingsrecord Extension Name for the mailbox, redirect messages from the mailbox, delete all old
messages from the mailbox, record greeting messages for the mailbox. By default, Mailbox
Management is enabled.
If you do not want to allow the extension user to change the mailbox settings, in Allow Mailbox
Management, select No.
To know more about this feature, see Mailbox Settings
Auto Delete Messages: Select the type of messages you want the VMS to automatically delete from
your mailbox. You can select All or Old. Default: None.
Days for Auto Delete Messages: Select the number of days after which you want the VMS to
automatically delete the messages in your mailbox. Default: 90 days.
Department Group Mailbox: You can assign the Mailbox of a Department Group to ISDN extensions,
even to those ISDN extensions that are not included in the Department Group. Refer the topic
Department Call.
To assign the Department Group mailbox to an ISDN extension, select the number of the Department
Group (1 to 16) from the combo box.
If you do not want to assign Department Group mailbox to an ISDN extension, select None.
Default: None.
Voice Mail Auto Attendant Features: This parameter is applicable only if you are using the VMS Auto
Attendant.
Voice Mail Auto Attendant Profile: Select a profile for the ISDN extension. The profile determines
the welcome message to be played to mailbox owners when they reach the home node. It also
determines whether or not the user should be taken to the root node directly.
Abbreviated Name: When the VMS is used as Auto Attendant, the callers can be prompted to Dial
by Name of the desired extension users instead of their extension numbers.
To allow callers to reach the ISDN terminal extension using Dial By Name, abbreviate the extension
users name to three letters and enter it in this field.
Announce Name: If you want the VMS to announce the extension users name to the caller when
transferring the call to the extension, select the checkbox to enable Announce Name. By default,
Announce Name is disabled.
If you enable Announce Name, make sure you record the extension users name on the VMS. Refer
Recording Extension Names for instructions.
799
Call Transfer: Select the desired method for transferring the call answered by the VMS Auto
Attendant to the ISDN extension. You may select any of the following methods of call transfer for
each time zone, Working Hours (WH), Break Hours (BH) and Non-working Hours (NH):
None: When the caller dials the extension number, the VMS Auto Attendant will check if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
Blind: When the caller dials the extension number, the VMS Auto Attendant will transfer the call
on the extension without checking whether it is busy or free.
Wait for Ring: When the caller dials the extension number, the VMS Auto Attendant will wait for
the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the
extension, if assigned, or take the caller back to the home node.
Wait for Answer: When the caller dials the extension number, the VMS Auto Attendant will
transfer the call when the extension answers (goes OFF-Hook).
If the extension does not answer154, the VMS Auto Attendant will transfer the call to the mailbox
of the extension, if assigned, or take the caller back to the home node.
Screen: The VMS Auto Attendant prompts the caller to record his/her name. It puts the caller on
hold and places the call on the desired extension. If the extension is free and answers the call,
the VMS announces the callers name to the extension user and prompts the extension user to
choose whether or not to speak to the caller. If the extension user chooses to talk, the VMS
transfers the call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the
extension user, if assigned, and asks the caller to leave a message.
By default, Wait for Answer is selected as Call Transfer method for all time zones.
Message Wait Indication: This parameter allows you to select the type of indication to be given to the
extension user for new messages in the mailbox and message wait set by another extension user.
You can select from any from the type of indicators described below for new messages:
Stuttered Dial Tone/Voice Message: When the extension user goes OFF-Hook, s/he will hear a
voice message, if a pre-recorded Voice Module has been assigned for Message Wait Notification. If
no voice module is recorded and assigned, the extension user will hear a stuttered dial tone
instead.
If you want voice message to be played as message wait notification, record and assign a Voice
Module. Refer Voice Message Applications for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait Notification
will not be played if there are already 4 being played simultaneously. In which case, Stuttered Dial Tone will
be played for Message Wait Notification, when the extension user goes OFF-Hook.
154. The VMS will wait for the duration of the Built-In Auto Attendant Ring Timer (default: 30 seconds; the timer is configurable). If the
call is not answered before this timer expires, it is treated as No Reply.
800
Ring: The extension will ring for the duration of the Message Wait Ring Timer (configurable;
default: 30 seconds), for as many times as the Message Wait Ring Count (configurable; default: 10
times), at the interval set as the Message Wait Ring Timer Interval (configurable; default: 30
minutes).
Message Wait Notification via Call: The message wait notification will be sent to a number
(destination number). This number can be an internal or an external number. To use this feature,
configure the following parameters:
Type: If you want the notifications to be sent as soon as a new message arrives in the mailbox of
the extension user, select Immediate.
If you want the notification to be sent at fixed time schedules, select Scheduled.
If do not want to set message wait notification via call, select None. Default: None.
Profile: Assign the Profile according to which you want the system to send the notifications. The
Profile determines the time intervals during which the notifications must be sent to the destination
number.
Destination Number: Enter the number on which you want the system to send the notification
calls.
The destination number can be an internal or an external number. The destination number can be a
maximum of 16 digits. Valid digits are 0 to 9, # and *.
When the notification call is answered, the VMS informs the callee of the new message and allows
the callee to access it.
Refer the feature description Message Wait Notification via Call to know more.
Message Wait Notification via E-mail: The message wait notification will be sent to the e-mail
address of the extension user. To use this feature, configure the following parameters:
Notification: If you want the message wait notification to be mailed to the extension user along with
the new voice message as attachment, select the option Send With Attachment.
If you want only the notification to be mailed, select the option Send Without Attachment.
If do not want to set message wait notification via e-mail, select Do not send. Default: Do not send.
E-mail Address: Enter the e-mail ID of the extension user to which the notification is to be sent. Email ID may consist of up to 64 characters. Default: blank.
Extensions users will receive notifications only for the mailbox memory utilization, if you configure the E-mail
Address and select Do not sent as the Notification option.
801
803
Exit SE mode.
Default:General Mailbox.
To enable/ disable message details option for a mailbox, dial:
6265-1-ISDN Terminal-Message Detail Option-#* to enable/disable message details option for a single
ISDN Terminal.
6265-2-ISDN Terminal-ISDN Terminal-Message Detail Option-#* to enable/disable the same message
detail option for a range of ISDN Terminal.
6265-*-Message Detail Option-#* to enable/disable the same message detail option for all ISDN
Terminal.
Where,
ISDN Terminal is the Software Port number of the Terminal from 01 to 64.
Message detail option is:
0 for Never
1 for Always
2 for On demand
Default:On Demand.
To enable mailbox password prompt, dial:
6266-1-ISDN Terminal-Password Option-#* to enable mailbox password on a single ISDN Terminal.
6266-2-ISDN Terminal-ISDN Terminal-Password Option-#* to enable mailbox password on a range of
ISDN Terminals.
6266-*-Password Option-#* to enable mailbox password on all ISDN Terminals.
Where,
ISDN Terminal is the Software Port number of the Terminal from 01 to 64.
Password Option is:
1 for Ask for Password
0 for Dont ask for Password.
Default:Ask for Password.
To enable/disable Allow Mailbox Management option, dial:
6275-1-ISDN Terminal-Allow Mailbox Management Option-#* to enable/disable the allow mailbox
management option for a single ISDN Terminal.
6275-2-ISDN Terminal- ISDN Terminal-Allow Mailbox Management Option-#* to enable/disable the
allow mailbox management option for a range of ISDN Terminals.
6275-*-Allow Mailbox Management Option-#* to enable/disable the allow mailbox management option
for all ISDN Terminals.
Where,
ISDN Terminal is the Software Port number of the Terminal from 01 to 64.
Allow Mailbox Management Option is
1 for Enabled
0 for Disabled
Default: Disabled.
To assign Department Group Mailbox to an ISDN Terminal, dial:
6267-1-ISDN Terminal-Department mailbox Option-#* to assign department mailbox to a single ISDN
Terminal.
6267-2-ISDN Terminal-ISDN Terminal-Department mailbox Option-#* to assign department mailbox to
a range of ISDN Terminals.
6267-*-Department mailbox Option-#* to assign department mailbox to all ISDN Terminal.
Where,
ISDN Terminal is the Software Port number of the Terminal from 01 to 64.
Department Mailbox Option is: None, 01 to 16.
Default: None.
805
806
6272-1-ISDN Terminal-Notification code-#* to assign Message Wait Notification Type to a single ISDN
Terminal.
6272-2-ISDN Terminal-ISDN Terminal-Notification code-#*to assign Message Wait Notification Type to
a range of ISDN Terminals.
6272-*-Notification code-#* to assign Message Wait Notification Type to all ISDN Terminals.
Where,
ISDN Terminal is the Software Port number of the Terminal from 01 to 64.
Notification code is:
0 - None
1 - Stuttered Dial Tone/Voice Message
2 - LED Lamp
3 - Ring
Default: LED Lamp.
Exit SE mode.
807
The keys in the Key Templates are numbered only for the purpose of locating the keys when programming.
Key numbers do not appear on the key labels on the phone body.
Hotel Attendant
Mute
3PConf
Transfer
Guest
Mute
3PConf
Transfer
Mute
3PConf
Transfer
17
18
19
23
24
25
17
18
19
23
24
25
17
18
19
23
24
25
20
21
22
26
27
28
20
21
22
26
27
28
20
21
22
26
27
28
CallFwd
DND
Names
CallFwd
DND
Names
CallFwd
DND
Names
Redial Release
01
09
02
10
1
4
ghi
abc
jkl
tuv
9 wxyz
03
11
04
12
7 pqrs
05
CA 4
13
06
CA 3
14
07
CA 2
15
08
CA 1
16
808
Hold
def
mno
29
Redial Release
01
09
02
10
1
4
ghi
abc
jkl
6 mno
tuv
9 wxyz
03
11
04
12
7 pqrs
05
Retrv
Msg.
13
06
CA 3
14
07
CA 2
15
08
CA 1
16
Hold
def
29
Redial Release
01
09
02
10
11
03
1
4
ghi
abc
jkl
6 mno
tuv
9 wxyz
12
7 pqrs
05
DKP2
13
06
DKP1
14
07
CA 2
15
08
CA 1
16
04
Hold
def
29
Hotel Attendant
Guest
01
01
01
02
02
02
03
03
03
04
04
04
CA 1
05
CA 1
05
CA 1
05
CA 2
06
CA 2
06
CA 2
06
809
Hotel Attendant
Guest
Options Help
Options Help
Options Help
SLT1
01
DKP1
06
CO1
11
CO1
16
CA1
21
SLT 2
02
DKP2
07
CO2
12
CO1
17
CA2
22
SLT3
03
DKP3
08
CO3
13
CO1
18
CA3
23
DKP4
09
CO3
14
CO1
19
CA4
24
26
R-DND
03
DKP2
08
Print AS
13
Call Back
18
CA3
23
Operator
01
Voice Mail
06
DKP1
11
CO1
16
CA1
21
Voice Mail
09
Clean
14
AR-Set
19
CA4
24
Floor Ser
02
DND
07
DKP2
12
CO2
17
CA2
22
26
Alarm-VG
03
Voice Help
08
SLT1
13
Call Log
18
Recall
23
Mute
09
SLT2
14
BGM
19
Release
24
26
By using Key Templates you can prepare and assign common key maps to all or as many DKPs and Extended IP
Phones as you want, at one go.
ETERNITY also offers the flexibility to personalize the Key Maps of each DKP/Extended IP Phone, instead of using
the Key Templates. For example, if you have assigned a common Executive Key Template to 12 DKPs, but you
want to reassign some of the keys on two of these DKPs, ETERNITY allows you to selectively personalize the key
maps of these two DKPs.
810
List the features/facilities that you want to change in each of the existing (default) Key Templates of
Operator, Executive, Hotel Attendant, and Guest.
For each template, decide the keys that will be reassigned the features you listed.
You may use the key templates printed above to decide the position of keys.
For each template that you customize, list down the DKPs which will be assigned the template,
along with their Software port numbers and their corresponding Hardware Slot and Port Offset.
Similarly list Extended IP Phones along with their Software port numbers (SIP Extension numbers)
and VoIP Port numbers.
Similarly, list the DKPs and the Extended IP Phones which are to be assigned personalized Key
Maps, along with their Software port numbers and their corresponding Hardware Slot and Port
Offset.
Illustrated below is an example of a customized Operator's Template for the EON48 model:
Voice Mail ACB Set Cancel
17
18
19
20
21
22
CallFwd
DND
Names
Mute
3PConf Transfer
23
24
25
26
27
28
Redial Release
01
09
02
10
03
11
04
12
05
CA 4
13
06
CA 3
14
07
CA 2
15
08
CA 1
16
Hold
abc
def
jkl
mno
7 pqrs
tuv
9 wxyz
1
4
ghi
29
811
You can customize the key template and assign it to DKPs using Jeeves or a Telephone.
EON48 and SPARSH VP248 have 12 Touch-sense feature keys. While you can reassign the features on
these keys, you cannot re-label the keys. Avoid reassigning features on touch-sense keys.
EON310 has 9 feature keys, displaying feature icons as labels. While you can reassign the features on
these keys, you cannot re-label the keys. Avoid reassigning features on these keys.
812
The links to the default key templates for Operator, Executive, Hotel Attendant and Guests for EONSOFT,
each model of EON and the Matrix Extended IP Phone appear on your screen.
Select the Key Template you want to customize according to the model of EON/Extended IP Phone in use,
by clicking the respective link. In case you have more than one model of EON, you may customize the
desired Key Template for each model.
Let us attempt to program the sample Operator template we customized earlier for EON48.
Click the Operator template link of EON48.The Operator Key template for EON48 will appear on your
screen.
Refer the key template we customized on paper. As per the customized template the keys that need to be
reassigned features are as follows:
Existing function on the key
To be replaced by
DKP4
Hotline
DKP3
DKP2
DND-Override
SLT4
SLT3
SLT2
SLT1
813
All features that can be assigned to keys will appear in the Select Offset list.
814
815
816
As Remote DND is an SA Command, select the option SA Command in the Select Function Type list.
In the Select Offset list, click Set DND for remote station.
817
Click OK. The window closes. The Remote DND feature appears in abbreviated form as R-DND on the
key label.
Repeat these steps to reassign other keys, selecting the appropriate Function Type and the Offset for each
feature/function.
For example, DND-Override is a feature, so select 'Feature' as the function type.
To assign direct access to Mobile Trunk 1, select Mobile as function type and '01' as Offset.
To assign direct access to SIP Trunk 1, select SIP as function type and '01' as Offset.
To assign direct access to BRI Trunk 1, select BRI as function type and 01 as Offset. When you select
BRI as function type, you will be asked to Select Channel. Since you do not want to assign direct access
to any particular BRI channel of this trunk, retain the option All Channels for Select Channel.
To assign direct access to Channel Number 2 or T1E1 Trunk 1, select 'T1E1' as function type, '01' as
Offset.
818
When you select T1E1, as function type, you will be given the Select Channel option. Since you want to
assign direct access to Channel 2 of T1E1 Trunk 1, select the option Channel 2 in the Select Channel
list.
Always click OK when you select a Function Type and Offset.
When you have completed assigning functions to keys, click Submit at the bottom of the page to save
your settings.
Scroll with the horizontal bar to reach the parameter Key Map of this DKP 001.
819
Click Submit.
Also refer the topic Configuring DKP Parameters using Jeeves for instructions on assigning the
Key Map in the DKP Parameters.
When customizing key templates using a Telephone, two attributes of a key must be programmed, namely:
a. Location - the physical location of the key on the phone body.
b. Function - the function that the key should perform, that is, as direct access to a feature or an SLT or DKP
820
To assign the same feature to the same key number in all the templates, dial:
1261-*-Terminal Type-Key Number-Function Type-Function Number-Channel
Where,
Key Template is
1 for Operator
2 for Executive
3 for Hotel Attendant
4 for Guest
Terminal Type is
1 for EON45/EONSOFT
2 for EON42
3 for EON48
4 for SPARSH VP248
6 for EON310
7 for SPARSH VP330
Key Number is the number of the key which is to be reassigned the feature.
Key Numbers on EONSOFT are from 01 to 25
Key Numbers on EON42 are from 01 to 25
Key Numbers on EON48 are from 01 to 29.
Key Numbers on SPARSH VP248 are from 01 to 29.
Key Numbers on EON310 are from 01 to 15.
Key Numbers on SPARSH VP330 are from 01 to 15.
For numbering of the keys, refer the default Key Maps illustrated at the beginning of this topic.
Function Type is the function to be performed by the key, from 00 to 34.
Function Number defines the exact function under each Function Type selected for the key.
Channel is the number of the channel in the BRI or PRI Line to be accessed.
Channel number for BRI Line is
01 for Channel 1
02 for Channel 2
00 for both Channels
Channel number for PRI Line is
01 to 30 for Channels 1 to 30 respectively.
00 for all Channels
Channel number for all function types other than BRI and PRI is 00
Refer the tables below for the complete list of function types and function numbers, and channel
numbers.
Table: Function Types and Function Numbers
Function
Type
00
Meaning
Function
Number
Meaning
None
821
Function
Type
Meaning
Function
Number
Meaning
01
001 to 512
02
001 to 128
03
CO (Direct Trunk
Access)
001 to 128
04
001 to 032
05
001 to 008
822
06
001 to 128
11
01 to 16
12
Quick Dial
(Abbreviated
Dialing)
001 to 999
13
01 to 10
14
MACRO
01 to 25
15
1 to 6
Function
Type
Meaning
Function
Number
Meaning
18
FEATURE (Direct
Feature Access)
01 to 87
21
Voice Mail
22
SA Command
(Direct Dialing of
SA Command)
001 to 173
24
Special Keys
01 to 15
25
MOBILE (Direct
Trunk Access)
01 to 32
26
01 to 32
28
ISDN Terminal
(Direct Station Call)
01 to 64
29
Magneto (Direct
Station Call)
001 to 128
31
ROOM (Direct
Station Call)
001 to 512
33
1 to 3
34
SIP Extension
(Direct Access)
001 to 999
35
LD
01-32
36
Virtual Ext.
01-64
37
Radio Port
01-16
Feature Number
Yes
823
Feature
824
Feature Number
Redial
10
Abbreviated Dialing
11
Operator
12
Call Forward
13
Dynamic Lock
14
Hotline
15
Yes
Alarm
16
Yes
Do Not Disturb
17
Yes
Interrupt Request
18
Barge-In
19
Raid
20
Trunk Reservation
21
Call Toggle
22
Conference
23
24
Dial-In Conference
25
Call Park
26
27
Room Monitor
28
29
Voice Help
30
31
32
Paging
33
DISA Login
34
35
36
37
Flashing on Trunk
38
User Absent/Present
39
40
41
Yes
Yes
Yes
Yes
Feature
Feature Number
Background Music
42
Yes
Meet Me Paging
43
Hot Desk
44
45
Presence
46
Yes
47
Yes
Conversation Recording
48
Forced Release
49
Transfer
50
51
Forced Answer
52
53
54
Minibar Details
55
Mute
56
Emergency Conference
57
58
Call Chaining
59
SA Command Prefix
60
61
Floor Service
62
Keypad Lock
63
Yes
CLI Restriction
64
Yes
65
Reminder
66
67
68
69
70
Yes
71
Yes
72
73
74
Open a Door
75
76
Yes
Yes
Yes
825
Feature
Feature Number
77
Invoke RCOC
78
79
80
81
82
General Mailbox
83
Intercom
84
Terminate Conference
86
87
88
Yes
89
Yes
90
Yes
Voice Mail
Yes
Yes
826
Function Number
Check-In
001
Check-Out
002
Guest Name
003
Guest Group
004
Guest-In/Out
005
Guest Title
006
007
008
009
Room Shift
010
011
012
SA Command Name
Function Number
013
014
Set DND-Remote
015
016
017
018
019
020
021
022
023
Yes
024
Yes
025
026
027
028
Change SA password
029
030
031
032
033
034
035
036
037
038
039
040
041
042
043
044
045
046
047
Yes
827
SA Command Name
828
Function Number
048
049
050
051
052
053
054
055
056
057
058
059
060
061
062
063
064
065
066
067
068
069
Set filter to print internal calls with duration greater than that
given here
070
071
072
073
074
075
076
077
078
079
080
SA Command Name
Function Number
081
082
Set filter to print all calls with speech duration More than timer
083
Set filter to print all calls unanswered for duration More than
timer
084
Set filter to print all calls kept on hold for duration more than
timer
085
086
087
088
089
090
091
092
093
094
Set filter to print all IC calls recd. From nos. matching the
Number List
095
096
097
098
099
100
101
102
103
104
105
106
107
108
109
110
111
112
Reminder - Remote
113
Yes
829
SA Command Name
830
Function Number
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
Yes
SA Command Name
Function Number
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
To Broadcast Message
175
176
177
178
179
180
181
182
183
831
SA Command Name
Function Number
184
185
186
187
Function Number
Flash
001
No
Pause
002
No
Digit 'A
003
No
Digit 'B
004
No
Digit 'C
005
No
Digit 'D
006
No
Release
007
No
Acknowledge
008
No
Enter
009
No
Speed
010
No
Emergency Alarm
Log
011
Yes
Speaker
012
Yes
Headset
013
Yes
Cancel
014
No
Answer
015
No
Call Log
016
Local Menu
017
No
Examples:
Let us attempt to program the same sample Operator Template we customized earlier for EON48 using
a telephone. For this, we need to know the location of the key, that is, the key number.
Existing function on the key
832
To be replaced by
DKP4
Hotline
Key No. 01
DKP3
Key No. 02
DKP2
DND-Override
Key No. 03
SLT4
Key No. 09
SLT3
Key No. 10
To be replaced by
SLT2
Key No. 11
SLT1
Key No. 12
To assign Hotline to the key currently assigned to DKP4 key (key 01) in the Operator Template, dial:
1261-1-1-3-01-18-15-00
Where,
1 is Operator Template
3 is for EON48 Terminal
01 is the Key Number
18 is Function Type for Features
15 is Function Number for Hotline feature.
00 is Channel.
To assign Mobile Trunk 1 in place of SLT4 on DSS key 09 on the Operator Template, dial:
1261-1-1-3-09-25-01-00
Where,
1 is Operator Template
3 is for EON48 Terminal
09 is the Key Number
25 is Function Type for Mobile Trunks
01 is Function Number, that is, the port offset of the Mobile Port.
00 is Channel
To assign BRI Trunk 1 in place of SLT2 on DSS Key 11 on the Operator Template, dial:
1261-1-1-3-11-04-01-00
Where,
1 is Operator Template
3 is for EON48 Terminal
11 is the Key Number
04 is Function Type for BRI Trunks
01 is Function Number, that is, number of the BRI port.
00 is Channel number.
To assign Channel 2 of T1E1 Trunk 1 in place of SLT1 on DSS Key 12 on Operator Template, dial:
1261-1-1-3-12-05-01-02
Where,
1 is Operator Template
3 is for EON48 Terminal
12 is the Key Number
05 is Function Type for T1E1 Trunks.
01 is Function Number, that is, number of the T1E1 port.
02 is Channel number.
To assign a DKP Key template to a DKP, dial:
1221-1-DKP-DKP Key Template to assign the template to a single DKP.
1221-2-DKP-DKP-DKP Key Template to assign the same template to a range of DKPs
1221-*-DKP Key Template to assign the same template to all DKPs
Where,
DKP is number of the Software Port of the DKP to which the Template is to be assigned, from 001 to
128.
833
Exit SE Mode.
834
Go to the DKP you want to assign a personalized key map, for example, DKP002.
Select the option Personalized as the Key Map for the DKP.
835
836
To assign features to keys, follow the same steps as you did for customizing the key templates.
Click the key you want to assign the function. For example, you want Barge-In on the key assigned to
DKP4, click this key.
The options for the Functions to be Performed by the key will open in a new window.
837
Click OK. The window will close. The new label will appear on the key.
Repeat the same steps to program another key on this key map.
When you have completed personalizing the key map, close the window.
Follow the same steps to personalize the key map of another DKP.
For instructions on personalizing the Key Map of the Matrix Extended IP Phone, see, Configuring Matrix SPARSH
VP248 - Extended IP Phone.
Enter SE mode.
To assign a Personalized Key map to a DKP, dial:
1221-1-DKP-0 to assign personalized key map to a single DKP.
1221-2-DKP-DKP-0 to assign the personalized key map to a range of DKPs.
1221-*-0 to assign personalized key map to all DKPs.
Where,
DKP is number of the Software Port of the DKP to which the Template is to be assigned, from 001 to
128.
838
839
Location is 1 to 3.
Key Number is 01 to 29 (as located on the phone)
Function Type is 00 to 34 (00 is none).
Function Number (as per function type selected)
Channel Is the number of the channel in the BRI or PRI Line to be accessed.
Channel number for BRI Line is
01 for Channel 1
02 for Channel 2
00 for both Channels
Channel number for PRI Line is
01 to 30 for Channels 1 to 30 respectively.
00 for all Channels
Channel number for all function types other than BRI and PRI is 00
For the complete list of Function Types and Function Numbers see Customizing Key Templates using
a Telephone.
The Key Numbers of the MATRIX Extended IP Phone is the same as EON48 described under this
topic.
Also see Configuring SIP Extension Settings as per the Extended Phone Type.
Exit SE Mode.
After you have finished assigning key templates and key maps, test the functioning of the keys.
840
01
17
33
49
01
25
49
02
18
34
50
02
26
50
03
19
35
51
03
27
51
04
20
36
52
04
28
52
05
21
37
53
05
29
53
06
22
38
54
06
30
54
07
23
39
55
07
31
55
08
24
40
56
08
32
56
09
25
41
57
09
33
57
10
26
42
58
10
34
58
11
27
43
59
12
28
44
60
13
29
45
61
14
30
46
62
15
31
47
63
32
48
16
64
11
35
59
12
36
60
13
37
61
14
38
62
15
39
63
16
40
64
17
41
65
18
42
66
19
43
67
20
44
68
21
45
69
22
46
70
23
47
71
24
48
72
In the default DSS Key assignment, all DSS Keys are assigned SLT ports for Direct Station Calling.
Accordingly,
If a single DSS64 is attached to a DKP, the SLT ports 005 to 068 are assigned to the keys. Recall that by
default SLT001 to SLT004 are assigned to the DSS keys on the DKP.
If a single DSS72 is attached to a DKP, the SLT ports 005 to 076 are assigned to the keys. Recall that by
default SLT001 to SLT004 are assigned to the DSS keys on the DKP.
If two DSS72 are attached to a single DKP, the SLT Ports 005 to 076 are assigned to DSS1 (the first DSS
Console) and the SLT ports 077 to 148 are assigned to DSS2 (the second DSS Console).
If the first DSS is a DSS72 and the second DSS64, the SLT Ports 005 to 076 are assigned to DSS1 and
the SLT ports 077 to 140 are assigned to DSS2.
However,
If two DSS64 are attached to a single DKP, the SLT ports 005 to 068 are assigned to DSS1 (the first DSS
Console) and SLT ports from 077 to 140 are assigned to DSS2 (the second DSS Console). The SLT ports
069 to 076 and from 141 to 148 cannot be accessed from the DSS. So, the keys of both DSS1 and DSS2
must be programmed.
841
Similarly, if the first DSS is a DSS64 and the second DSS72, the SLT Ports 005 to 068 are assigned to
DSS1 and the SLT ports 077 to 148 are assigned to DSS2. The SLT ports 069 to 076 and from 141 to 148
cannot be accessed from the DSS. So, the keys of only DSS2 must be programmed.
The steps for programming the Keys of the proprietary DSS Consoles of Matrix, DSS64 and DSS72, are
quite similar the programming of the DKP keys. It can be done using Jeeves as well as a Telephone.
842
Using the horizontal scroll bar on the page, scroll to DSS1 Keys, the first Console attached to the DKP.
The default key map of the DSS Console appears on your screen. By default all keys are assigned to
SLTs.
843
844
The options for the Functions to be Performed by the key will open in a new window.
Now, configure the key just like you configured the DKP/Extended IP Phone keys.
Select the desired 'Function Type to be assigned to the key and the desired Offset for the Function Type.
Click OK.
After you have finished assigning functions to the DSS1 keys, close the window.
Follow the same steps as described above to program the keys of DSS2.
Enter SE mode.
To assign a function to a DSS key, dial:
1254-1-DKP-DSS-Key Number-Function Type-Function Number-Channel to assign a function to a
DSS Key on the DSS of a single DKP.
1254-2-DKP-DKP-DSS-Key Number-Function Type-Function Number-Channel to assign the same
function on the same key number on the DSS of a range of DKPs.
1254-*-DSS-Key Number-Function Type-Function Number-Channel to assign the same function on
the same key number on the DSS of all DKPs
Where,
DKP is the number of the Software Port of the DKP from 001 to 128.
DSS is the number of the DSS Console attached to the DKP.
1 for DSS1
2 for DSS2
Key Number is the number of the key which is to be assigned the function/feature. The Key numbers
vary according to the type of DSS Console being used.
Key Numbers on DSS64 are from 01 to 64.
Key Numbers on DSS72 are from 01 to 72.
For numbering of the keys, refer the default DSS Key Maps illustrated at the beginning of this topic.
Function Type is the function to be performed by the key, from 00 to 34.
Function Number defines the exact function under each Function Type selected for the key.
Channel is the number of the channel in the BRI or PRI Line to be accessed.
Channel number for BRI Line is
01 for Channel 1
02 for Channel 2
00 for both Channels
Channel number for PRI Line is
01 to 30 for Channels 1 to 30 respectively.
00 for all Channels
Channel number for all function types other than BRI and PRI is 00
Refer the tables provided under Customizing Key Templates using a Telephone for the complete list
of function types and function numbers.
845
846
Exit SE mode.
Configuring 'Operator'
In the context of a PBX, users understand the term 'Operator' as a person who handles multiple simultaneous calls
and functions as the link between callers and called parties.
For the PBX however, an 'Operator' is a Routing Group; a group of extensions to which calls made by extensions
by dialing '9' are to be landed. This also includes Auto Attendant calls on trunks during which the caller dials '9'.
Depending on the size of the Enterprise and the amount of call traffic to be managed, more than one Operator may
be employed. Also, it is not uncommon to have different Operator extensions according to the time of the day. For
instance, during working hours calls may be handled by the Receptionists or Front Desk Personnel, whereas during
non-working hours, calls may be handled by the Security Personnel.
To meet this requirement, ETERNITY offers configuration of up to 20 different Operators (Routing Group).
However, at a time, only one Operator can be assigned to the extensions and trunks.
Each 'Operator' is assigned a Routing Group for the Time Zones - Working Hours, Break Hours and Non-Working
Hours.
Each 'Operator' is assigned a Time Table, which defines the Working Hours, Break Hours and Non-working Hours
for a week. The system follows this Time Table to assign a Routing Group as 'Operator' according to the current
Time Zone.
Configuration of 'Operator' involves the following steps:
1. Configuring Routing Groups as 'Operator': A routing group may be made up of one or more than one
extensions, depending on user requirement. If the user requires only one extension as 'Operator', include
only one extension as member in the Routing Group for Operator. If the user requires five extensions as
'Operator', create a Routing Group of the five desired extensions to be used as 'Operator'.
If the user requires Time-Zone based 'Operator', then prepare a different routing group for each Time
Zone. If the user requires the same Operator for all Time Zones, use the same Routing Group number in
all Time Zones.
2. Configuring a Time Table for Operator: This is applicable only when Operator extensions are different
Extensions of the ETERNITY can be assigned to an 'Operator' in their Station Basic Feature Template.
All extensions may be assigned to the same Operator, or different groups of extensions may be assigned
to different Operators, so that call management is more efficient.
Operator 1 is the default in the Station Basic Feature Templates. If you want to assign different extensions
to different Operators, you must program separate Station Basic Feature Templates with a different
Operator for each extension group.
847
4. Assignment of 'Operator' to Trunks: Trunks are also assigned an 'Operator', so that when a caller dials
'9' using Built-In Auto Attendant, the call is routed to the Routing Group defined as Operator for the trunk
for a particular Time Zone. For example, the during working hours, a caller on trunk 001 dials '9', the call
lands on 3001; when a caller on trunk 001 dials '9' during non-working hours the call lands on 3003 and
when the caller dials '9' during break hours the call lands on 3002.
Similarly, it is also possible to assign different Operators to different trunks.
Decide the number of Operators to be configured on the basis of the user's requirement.
848
Select a Routing Group you want to program for Operator. By default, Routing Group 32 is assigned to
Operator. You may configure this group, or select another one.
Select the type of extension to be included in the group as Member Type. The extension may be a DKP,
an SLT, an ISDN terminal, an OTBG, a SIP Extension, a Virtual Extension or a Voice Mail Auto Attendant
Profile.
Enter the Port Number (software port number) to which the SLT/DKP/ISDN Terminal/SIP extension is
connected. If a Virtual Extension is selected as the Member Type, enter the Port number of the Landing
Destination here. You can program as many as 32 members in the Routing Group. If the user requires only
one extension as Operator, configure the first Member and disable all other 'members' of the routing group
by setting Member Type to None.
849
By default Time Table 1 is assigned to all Operators. If this time table meets your requirement, retain it. If
not, select another Time Table. Customize it by defining the Working Hours, Break Hours and NonWorking Hours for the week.
If you want to assign different Time Tables to different Operators, repeat the above steps to prepare the
other Time Tables.
If you have completed configuring the Time Table,
850
Select the Operator number you want to configure. By default Operator 1 is assigned to all extensions and
trunks.
Select the number of the Time Table you prepared for the selected Operator.
Enter the number of the Routing Group you prepared for the selected Operator for Working Hours, for
Break Hours and for Non-Working hours. If the same Routing Group is to be kept for all Time Zones, enter
the same number in fields of all three time zones.
851
852
Exit SE mode.
Configuring Trunks
The ETERNITY supports the following types of trunk ports:
E&M Feature Template (for E&M Lines and T1E1PRI Lines that use E&M signaling)
Using these templates, you can configure all Trunks that are to be assigned the same set of hardware and software
features at one go, instead of configuring each trunk individually.
CO Hardware Template
The CO Hardware Template contains a set of features, such as AC Termination Impedance, Pulse-Tone Dialing,
Answer Supervision, Disconnect Supervision, DTMF detection, etc. that define the behavior of the hardware of the
CO ports of the ETERNITY.
A CO Hardware template must be assigned to all the CO trunk ports. Using the CO templates, you can configure
CO ports which are to be assigned the same set of features at one go, instead of configuring port-by-port.
The ETERNITY offers 50 CO Hardware Templates. These templates have commonly used values, but can be
customized per the requirement and applied on the extensions.
ETERNITY supports only 'Loop Start' on CO Interface.
Trunk Type: Three types of Trunks can be interfaced to a CO port of the ETERNITY:
Normal Dial type: This is the conventional CO trunk available from the PSTN.
Hotline type: The CO trunk connecting two destinations immediately on grabbing the trunk.
853
Delayed Hotline: A special CO trunk available from the PSTN, which works as a normal dial type for
some time and works as a hotline thereafter.
By default all the CO ports are set as Normal Dial type. You may select the CO Dial Type you want to
assign to the CO port.
Dial Type: You can select the Dialing method as Pulse or Tone (with configurable Pulse Ratio and DTMF
ON-OFF period) according to the Dialing method supported by the CO network to which the CO port is
connected.
By default, Tone is selected as the Dial Type.
Pulse Dial Ratio: This parameter is to be configured if you have selected Pulse as the Dial Type in the
previous parameter. The system supports the six different Pulse Dialing Ratios on CO ports. Select the
appropriate Pulse Dial Ratio from the following according to the type of Pulse Dialing Ratio supported by
your CO Network:
10PPS, 1:2
10PPS, 2:3
10PPS, 1:1
20PPS, 1:2
20PPS, 2:3
20PPS, 1:1
By default, 10PPS, 1:2 is selected as the Pulse Dial Ratio.
Rx CLI Type: ETERNITY detects the CLI sent by the CO network and sends this information to the
landing extension/operator with the ringing signal. You must select the CLI Type supported by your CO
network from the following options:
Any ETSI DTMF format
Any FSK V.23 format
Any FSK Bellcore format
1st Ring, ETSI DTMF, 2nd Ring
Polarity Reversal, ETSI DTMF, 1st Ring
1st Ring, FSK, 2nd Ring
DT-AS, FSK, 1st Ring
RP-AS, FSK. 1st Ring
Polarity Reversal DT-AS, FSK, 1st Ring
Any DTMF Format (without Start/Stop Code)
By default, Any ETSI DTMF format is selected as the Rx CLI Type.
854
AC Impedance: The AC Termination Impedance of the CO port must match with the AC Termination
Impedance supported by the PSTN network. The system supports the following AC Termination
Impedance:
600
900
270 + (750 || 150 nF)
220 + (820 || 115 nF)
370+ (620 || 310 nF)
320 + (1050 || 230 nF)
370 + (820 || 110 nF)
275 + (780 || 115 nF)
120 + (820 || 110 nF)
By default, the AC Termination Impedance is set as per the Region you have selected.
CO Termination: This parameter allows you to increase near-end echo cancellation on the CO trunk.
Near-end echo is primarily caused by the mismatch between AC Termination Impedance (presented by
CO port of ETERNITY to the line) and CO Termination (Impedance presented by the Central Office to the
line), and to some extent by the transmit and receive signal path.
By correcting the line impedance mismatch, you can increase near-end echo cancellation. This is done by
selecting the AC Termination Impedance and CO Termination, and selecting a Line Type that most closely
models the line that connects the CO port of ETERNITY to the Central Office.
In the CO Termination list, select the appropriate line impedance match. This would depend on the region
where ETERNITY is deployed. For example, if AC Termination Impedance in your location is 600and the
CO Termination impedance is 900 in series with 2.16F, select AC Impedance as 600and CO
Termination as 900 + 2.16F. By default, None is selected.
Now, select the line model to be used from the CO Line Type list.
You are recommended to conduct the AC Impedance Test for the line connected to the CO Trunk port on
which you will apply this template. The AC Impedance Test will help you determine the most appropriate
values for the AC impedance, CO Termination and the CO Line Type. For more information see the topic
AC Impedance Test
CO Line Type: This parameter allows you to select the Line model for the CO Termination you have
selected. You need to select a line type that most closely models the line connecting ETERNITY to the
Central Office. In the CO Line Type list, you may select a specific EIA line model from the eight options
(EIA-0 to EIA-7) or a specific wire gauge and length (2000 ft. 22/24/26awg). By default, None is selected.
CO Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive Gain)
and Outgoing Speech (Transmit Gain) on the Trunk by changing the Rx Gain and Tx Gain to the desired
level. Different levels can be set for each port type in the CO Gain Settings Template. By default, CO Gain
Template 1 is assigned to all the CO Hardware Templates. If you want to assign a different Template, you
must customize the CO Gain Settings Template first and then assign the number of the CO Gain Settings
Template in this Template. To customise the CO Gain Settings, see Gain Settings.
Answer Supervision: It is a signal from the CO network to indicate to call maturity. Whenever you make
an outgoing call from CO trunk, the CO network will give answer signaling when the called party answers
the call.
This feature is particularly useful if you want to use Call Cost Calculation (CCC) to enable accurate
billing. When the signal is received, the billing will start and in the absence of this signal, the call will not be
billed, ensuring that unanswered and unsuccessful call attempts are not billed.
855
Pseudo Answer: It is used when no signaling is available from the PSTN. If this option is selected, the
call will be considered as matured on the expiry of the 'Pseudo Answer Supervision Timer'
(configurable; default 10 seconds), irrespective of whether or not the call actually gets matured. After
this, the Call Duration Timer starts. Finally, the system starts detecting the Disconnect Supervision
signal configured for the CO port.
Polarity Reversal: It is used as maturity signal when the answer signaling is given in the form of
Battery Reversal. If the battery polarity of the line is -ve for TIP and +ve for RING, when the called party
has answered the call, the CO network will reverse the battery polarity, TIP becomes +ve and Ring -ve.
After this, the Call Duration Timer is started. Finally system starts detecting the Disconnect Supervision
signal configured for the CO port.
By default, Answer Supervision is set as Pseudo Answer for each CO port.
Select the same Answer Supervision signal as provided by your CO Network. If the type of Answer
Supervision signal selected in the system does not match with that of the CO network, the call will not be
stored in the Station Message Detail Record (SMDR) buffer. For example, if the CO network does not
support Answer Supervision, but you have set Polarity Reversal as Answer Supervision Type, the call will
be considered as matured and will not be stored in the Station Message Detail Record (SMDR) buffer.
Pseudo Answer Supervision Timer: Configure this timer if you have selected 'Pseudo Answer' as
Answer Supervision Signal option.
This is the time period for which the system will wait before treating a call as matured (regardless of
whether or not it was answered). The range of this Timer is from 001 to 255 seconds. By default the
Pseudo Answer Supervision Timer is set to 10 seconds.
When Pseudo Answer is selected as Answer Supervision signal, the call duration measured by the system
will not accurately reflect the actual call duration because the Pseudo Answer Supervision Timer is not
related to the actual call maturity. For example, if the Pseudo Answer Supervision Timer is set to 015
seconds, the call will be considered as matured after 015 seconds, even if it is answered after 20 seconds.
Similarly, if this Timer is set to 080 seconds, but the call was answered after 020 seconds and
disconnected after 040 seconds, this call will never be considered as matured as it ends before 080
seconds.
Disconnect Supervision: It is a signal from the CO network to indicate call disconnection. Whenever a
call (incoming or outgoing) made from the CO trunk is disconnected by the remote party, the CO network
will send Disconnect signal to the CO port. ETERNITY will detect this signal and release the CO port.
Disconnect Supervision signal is important when a PCO machine is connected to the (SLT Port)
ETERNITY and or when ETERNITY is deployed in a Gateway application.
In such application scenarios, it is desirable that calls that are disconnected by either end - calling party or
called party - is terminated by the system and the port is released. If the called (remote) party has
disconnected the call but the calling party (extension that made the outgoing call from ETERNITY) has not
disconnected the call, the call remains live within the system.
So, Disconnect Supervision signal is important, particularly when calls are routed from CO-to-CO ports, to
indicate to the system that it needs to disconnect the call and release the port.
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None: When there this no signaling supported. Select this option only if there is no Disconnect
Supervision signal supported.
Polarity Reversal: Call disconnection is signaled as Polarity Reversal when the call is disconnected by
the remote user. For example, if the battery polarity of the CO port is '+ve' for TIP and '-ve' for RING in
speech condition then on disconnection by the remote user, TIP will become '-ve' and Ring '+ve'. The
user gets an Error tone and the CO port is released.
Open Loop Disconnect: Call Disconnection is signaled in the form of Open Loop, whereby the Battery
voltage on the CO port is removed for a short duration. Voltage is restored after this short duration.
However, the Polarity of Battery Voltage on the CO port is not changed.
This option is to be selected when call disconnection is signaled in the form of Open Loop Disconnect
pulse by the CO network. System will check Open Loop Disconnect signal for the time configured for
Open Loop Disconnect Timer for each CO port. If the time of the Open Loop signal detected is less
than the Open Loop Disconnect Timer configured, it will not be considered as valid Open Loop signal
for releasing the CO port. But if the Open loop is detected continuously for at least for the time set as
the Open Loop Disconnect timer, it is considered as a valid Disconnect Supervision signal. The call will
be released and caller will get error tone.
By default, Disconnect Supervision is set to None for each CO port.
Select the same Disconnect Supervision signal as provided by your CO Network.
Select the same Answer Supervision and Disconnect Supervision signal type supported by your CO
network for the CO ports. Consider the following case:
The CO network supports Polarity Reversal signal as Answer and Disconnect Supervision.
But you have configured 'Pseudo Answer' as Answer Supervision signal and 'Polarity Reversal' as
Disconnect Supervision signal for the CO ports in the system.
In this case, when a call is made through the CO port, the call will be considered as matured after
the Pseudo Answer Supervision Timer.
Now, when the called party answers the call, the CO generates 'Polarity Reversal' as answer
supervision signal on the CO port.
But as 'Polarity Reversal' is also configured as the Disconnect Supervision for the port, the system
will interpret this (Answer Signaling) signal as Disconnect Supervision signal and disconnect the
call.
Open Loop Disconnect Timer (msec): This parameter is applicable only if the option Open Loop
Disconnect is selected as Disconnect Supervision on the CO port.
The range of this timer is from 017 to 986 milliseconds. By default, the Timer is set to 204 msec.
Disconnect Tone Detection: This parameter is to be configured if Call Disconnection is signaled by the
CO network in the form of Disconnect Tone.
When there is an incoming/outgoing call on/from the CO port is answered, the system will check whether
the flag Disconnect Tone detection is enabled. Only if the flag is enabled, the system will detect the
Disconnect Tone.
If Disconnect Tone is detected, the system will consider the call as ended and will release the CO port.
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Disconnect Tone Cadence: To enable the system to detect the Disconnect Tone accurately, you must
set the Cadence (ON-OFF time) and Frequency of the Disconnect Tone, as supported by the CO network.
Configure the following Disconnect Tone Cadence parameters:
Operator: This parameter has 3 options: No operator, Modulation (*), Addition (+). Default: No.
If No operator is selected frequency 2 will not be applicable.
If Modulation is selected, frequency 1and frequency 2 will be used as modulation, F1* F2
If Addition is selected, frequency 1 and frequency 2 will be used as addition, F1 + F2.
Frequency 2 (Hz): Frequency 2 is from 20 to 1400 Hz. Select Frequency 2 if the Disconnect Tone
supported by the CO network is Dual Frequency. Default: 25Hz.
ON Time 1 (ms), OFF Time 1 (ms): Select Cadence for the first cycle ON Time1 and OFF Time 1. It
may be 0 to 9999 milliseconds. Default: 750 ms ON Time 1, 750 ms OFF Time 1
ON Time 2(ms), OFF Time 2 (ms): Select Cadence for the second cycle ON Time 2 and OFF Time 2.
It may be 0 to 9999 milliseconds. Default: 750 ms ON Time 2, 750 ms OFF Time 2.
ON Time 3(ms), OFF Time 3 (ms): Select Cadence for the third cycle ON Time 3 and OFF Time 3. It
may be 0 to 9999 milliseconds. Default: 0 ms ON Time 3, 0 ms OFF Time 3.
ON Time 4(ms), OFF Time 4 (ms): Select Cadence for the fourth cycle ON Time 4 and OFF Time 4. It
may be 0 to 9999 milliseconds. Default: 0 ms ON Time 4, 0 ms OFF Time 4.
When disconnect tone detected on the port matches the Frequency and Cadences you have set, the
call will be disconnected and the CO port will be released.
When Disconnect cadence is zero, ETERNITY will skip that cadence and match the next cadence.
ETERNITY will match the cadence for 3 cycles and then release the trunk.
Speech Delay Timer: It is the time after which the system gives dial tone to the extension, when the
extension user grabs the CO.
To understand the significance of this timer, let us consider a situation. Extension 2001 does not have
calling permission for long distance numbers. The user of extension 2001 grabs a CO trunk, and dials a
number 1022-6305555. The system dials out this number, as it starts with '1', but since the actual dial tone
from the CO comes after some time, the CO interprets this number as 022-6305555 and establishes
speech. This way an extension user who does not have permission for long distance calling, can dial out a
long distance number. This situation can be prevented by setting the Speech Delay Timer to an
appropriate value.
The range of this timer is from 000 to 255 seconds. By default it is set to '0' second.
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Pause Timer: This Timer is required for inserting delay while digits of a number string are out dialed from
the CO trunk. The Pause Timer is applied when the features Closed User Group (CUG), Multi-Stage
Dialing,Emergency Dialing, Last Number Redial, Auto Redial, Abbreviated Dialing, Call Back on
Trunk Ports, Quick Dial, RCOC (Return Call to Original Caller), Least Cost Routing (Configuring LCR)
are used to dial out the numbers from the CO port. The range of this timer is from 0500 to 2500
milliseconds. By default the timer is set to 1000 milliseconds.
Ring Cadence OFF Timer: Configure this timer to set OFF time for Ring cadence. During the incoming
call on CO port, if the CO gives ring in which the Ring OFF period is quite long, the system will consider
that the ring has been stopped, and will stop ringing the SLT port, even though the incoming call is still
present.
To get accurate indication, the system supports Ring Cadence OFF timer on CO port so that ring can
continue even for incoming calls with long Ring OFF period.
The range of the Ring Cadence OFF timer is from 1 to 9 seconds. By default the timer is set to 6 seconds.
DTMF Out Dial: While dialing out the DTMF digits from the CO port, the following attributes of DTMF
signal are critical.
DTMF Signal ON Time (ms): It is the width of DTMF digit to be dialed out by the CO port and is
configurable. By default the ON Time is set to 102 milliseconds.
DTMF Inter-Digit Pause Timer (ms): When the CO port dials out the DTMF digits on the CO, it waits
for the Inter Digit Pause Timer, while dialing the DTMF digits on CO trunk. This timer is configurable. By
default the timer is set to 102 milliseconds.
The 'level' of each DTMF digit is fixed, at -6.0 dB, but you may configure these parameters to match the
CO network requirement.
These DTMF Out Dial attributes are applied when the features Redial, Auto Redial and Abbreviated
Dialing are used to dial out the numbers from the CO port. These attributes are also applicable when
you make a call from a DKP that has DTMF generation disabled.
DTMF Detection: The default settings of DTMF Detection serve the requirements of most of the
applications. However, you may fine tune the following parameters if you face any problems in DTMF
detection.
Minimum Level (dB): This parameter signifies the minimum level (dB) of the DTMF digit to be
considered as valid. By default, Minimum levels set to -4.5dB.
Minimum ON Time (msec): This parameter signifies the minimum time period for which the DTMF
signal should be present in order to be detected. The valid range of this time is 17 to 204 milliseconds.
By default, Minimum ON Time is set to 34 milliseconds.
Minimum OFF Time (msec): This parameter signifies the minimum time period between successive
DTMF digits. The valid range of this time is 17 to 204 milliseconds. By default, Minimum OFF Time is
set to 68.
Flash Timer (msec): This parameter is relevant for dialing out Flash on the CO trunk to access some of
the features of the PSTN. Configure the desired time of Flash to be generated on the CO trunk.
The range of the timer is from 83 to 900 msecs. By default the Flash Timer is set to 600 msecs.
ON Hook Speed (msec): This parameter allows you to set the amount of time for the line-side device to
go on-hook.
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The ON-Hook speed specified is measured from the time the ON-Hook bit is cleared until loop current
equals zero. Select the desired ON-Hook Speed from the following options:
<0.5ms
3 ms
26 ms
By default, <0.5ms is selected as ON-Hook Speed.
OFF Hook Speed (ms): This parameter defines the time to settle the line transients after which
transmission or reception can occur. Select the desired OFF-Hook Speed from the following options:
512 ms
128 ms
64 ms
8 ms
By default, OFF-Hook Speed is set to 8 milliseconds.
Current Limiting: With this flag you can enable Loop Current Limiting mode. When this flag is enabled,
the Loop Current will be limited to a maximum of 60mA.
By default, the flag is disabled.
Minimum Loop Current (mA): This parameter sets the minimum loop current at which DAA module of the
CO port can operate. Select the minimum operational loop current from the following options as per your
requirement:
10
12
14
16
The minimum Operational Loop Current set by default is set to '10 mA'.
Tip Ring Voltage (Volts): This parameter allows you TIP/Ring Voltage Adjustment on the line side.
Countries where Low voltage is required should use lower TIP/RING voltage. Adjust the values of the Tip
Ring Voltage to match your country requirements from the following options:
3.1
3.2
3.35
3.5
The default Tip/Ring voltage is 3.5.
Ringer Impedance: Set the Ringer Impedance - High or Synthesized - for the CO port according your
country-specific requirement.
'High' signifies 20Mohm Ringer Impedance. This is the default Ringer Impedance provided on the line side
by the DAA module of the CO port. The DAA Module can provide higher impedance when 'Synthesized'
impedance is selected.
Some countries like Poland, South Africa and Slovenia require higher ring impedance which is achieved by
the DAA module, when Ringer Impedance is set to 'Synthesized' impedance.
By default 'High' (20Mohm) is selected.
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Ringer Threshold (Vrms): This parameter defines the level below which the CO port would not validate
the Ring signal and the level above which it validate the Ring signal. Set Ringer Threshold to the desired
value from the following options:
13.5 - 16.5
19.35 - 23.65
40.5 - 49.5
By default 13.5 - 16.5 Vrms is set as Ringer Threshold.
PPDC: 'Pre-PSTN Digit Count' or PPDC is parameter is to be configured, only if the CO Trunk ports on
which the template is applied are in a Behind the PBX Application.
PPDC is the number of digits (dialed by an extension) to be ignored by the system before toll control check
is begun. It is the same as the number.
In Behind the PBX Applications, another PBX may be connected to the ETERNITY, with some of its CO
Trunks terminating into the Stations of the other PBX and other trunks directly connected to the PSTN.
PPDC for CO Trunk ports directly connected to the PSTN must be set to '0'.
For Trunk ports connected to stations of another PBX, PPDC must be configured as per the number of
digits in the Trunk Access Code defined for that PBX.
If the TAC is a single digit, select '1'. If TAC is double or triple digit number, accordingly select '2' or '3' as
the PPDC.
To know more about this feature, refer Behind the PBX Application.
Gateway Application - Answer Signaling: This parameter is to be configured if the CO Port is being
used in a gateway application as a source port (from where calls originate).
The calls originating on the source port (CO port) are routed using another Trunk port, the terminating port,
which may be any trunk port, for example: T1E1. When call made from the terminating port gets matured,
this is signaled to the source port in the form of DTMF digits.
Use: Enable this flag if you want the CO port to be used in a Gateway Application.
DTMF String (max. 4 digits): Configure the DTMF digits to be sent to signal call maturity to the source
port.
Category (Logical Partitioning): This parameter assigns the CO Port to a trunk category for the purpose
of Logical Partitioning. By default all CO Ports are assigned to Category 1156.
If you have re-defined Category 1 or have assigned CO ports to a different category, say Category 3, enter
the same number here.
You may configure the call permission between the Category assigned to CO Ports and other Categories
(assigned to other Trunk ports). Refer the feature description Logical Partition to know more.
156. Trunk ports interfaced with PSTN /PLMN (Public Land Mobile Network) are assigned this category.
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Rx Gain at SIP Trunk (Pass Through Fax): This parameter allows you to improve the quality of Fax over
IP157. Configure this parameter if you have selected Pass Through Fax as the Type of Fax over IP on SIP
Trunks, and if Pass Through Fax is to be received on a CO Trunk.
Data Gain (dB): select the dB Level for Data Gain. By default, Data Gain is set to -11 dB.
Bypass Gain (dB): select the dB Level for Bypass Gain. By default, Bypass gain is set to -9 dB.
Idle Wait Timer: This is the time taken by the Central Office to detect and release the line after the CO
trunk has been released by ETERNITY. This time may vary from Central Office to Central Office. Set the
Idle Wait Timer as per the time taken by your Central Office.
Set this timer accurately. If the set time is more than the actual time taken by the CO to release the line, it
will result in delay to the caller. If the set time is less than the actual time taken by the Central Office, no
dial tone will be played to the caller.
Valid range 001 to 255. By default, it is set as 002 seconds.
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Select a CO Hardware Template Number you wish to customize. For example, Template 07.
863
864
Customize the CO Hardware Template number 07 by setting the parameters to the desired values.
Go to the CO software ports to which this Template you customized (07) is to be assigned, for example
CO-007 and 008.
Enter the number of the Template you customized, 07, in the field CO Hardware Template of each of
these CO ports.
Repeat the same steps to customize another template and apply it to the CO Port.
865
866
867
868
869
870
871
872
Table 1:
For example, to change the Rx CLI Type in Template 07 from 'Any ETSI DTMF format' to 'Any FSK V.23
format', dial 5902-1-07-01-3
Where,
07 is the template number
05 is the parameter number for Rx CLI Type
2 is the code for Any FSK V.23 format.
To default CO Hardware Template, dial:
5901-1- CO Hardware Template Number to default a single template.
5901-2- CO Hardware Template Number - CO Hardware Template Number to default a range of
templates.
5901-* to default all templates.
To assign a CO Hardware Template to a CO port, dial:
5903-1-CO-CO Hardware Template Number to assign a hardware template to a single CO port.
5903-2-CO-CO-CO Hardware Template Number to assign a hardware template to a range of CO
ports.
5903-*-CO Hardware Template Number to assign a hardware template to all CO ports.
Where,
CO is the Software Port number of the CO port from 001 to 128.
Template Number is the number of the customized CO Hardware Template, from 01 to 50. Default:
Template 01.
Exit SE mode.
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Operator: Define the Operator for the Trunks on which the template is applied. Operator is used to route
the call when the caller dials '9' during an Auto Attendant call. This parameter is of significance only if BuiltIn Auto Attendant/DISA is enabled on the trunk.
The system supports multiple Operators. In each Time Zone any one of the four Operators can be
selected.
Trunks may be assigned to a single Operator, or different groups of Trunks may be assigned to different
Operators, so that call management is more efficient. For instance certain Trunks may be assigned to
Operator 1, while some may be assigned to Operator 2 and the rest to Operator 3.
Operator 1 is the default in the Trunk Feature Template. If you want to assign different trunks to different
Operators, you must create a separate Trunk Feature Template with a different Operator for each trunk
group.
Refer the topic 'Operator' to know more.
CLI Based Routing: Select the checkbox to enable CLI Based Routing on the Trunk for each Time Zone:
Working Hours (WH), Break Hours (BH) and Non-working Hours (NH). Default: disabled.
If you enable CLI Based Routing on the trunk for a Time Zone, make sure you also configure the CLI
Based Routing Table. To know more, refer the feature description CLI Based Routing.
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Trunk Landing Group (Routing Group): This parameter allows you to configure the group of extensions
on which incoming calls on the trunks (to which this template is assigned) are to be landed. This group of
extensions is referred to as 'Trunk Landing Group' (TLG).
To configure the TLG, you must first configure Routing Groups. Refer Trunk Landing Group (TLG) for
instructions on configuring trunk landing groups. Also refer the topic Routing Group.
There are as many as 96 Routing Groups which can be assigned as TLG. By default, Routing Group 01 is
assigned as TLG for all Time Zones. If you have prepared a different TLG for each Time Zone, for
example, Routing Group 02 for Working Hours and Break Hours, Routing Group 3 for Non-Working Hours,
then enter the number of these Routing Groups in the TLG field.
Auto Attendant: This parameter is to be configured if you want to enable Auto Attendant on the trunk
ports on which you will apply the template.
Auto Attendant can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break
Hours (BH) and Non-Working Hours (NH).
For each Time Zone, you may select the desired Auto Attendant option from the following:
OFF: Select this option if you want to disable Auto Attendant for the Time Zone.
Built-In Auto Attendant: Select this option if you want the calls to be answered by the built-in Auto
Attendant of the ETERNITY. In Built-In Auto Attendant, ETERNITY answers the call using Voice
Modules, if assigned, or it answers the call and plays the appropriate call progress tone - Dial tone,
Ring Back tone, Busy tone - for each call state.
If you select this option, make sure you also configure the Built-In Auto Attendant related Timers and
Flags, record and assign the Built-In Auto Attendant related Voice Message and set the Start Time for
the Greeting Messages. Refer the topics Auto Attendant , Voice Message Applications and
Greeting Message Time in System Parameters for instructions.
Voice Mail Auto Attendant: Select this option if you want to the calls to be answered by the Auto
Attendant of the Voice Mail System. The Voice Mail System of ETERNITY answers calls and
processes them according to the Voice Mail Auto Attendant Profile assigned to the trunk.
If you select this option, make sure you also select and assign the desired Voice Mail Auto Attendant
Profile to the trunk.
By default, Auto Attendant is disabled (OFF) for all the Time Zones.
Auto Attendant Delayed Timer: Set this Timer, if you want to enable Delayed Auto Attendant on the
trunk.
When you enable Delayed Auto Attendant, ETERNITY routes the incoming call on the trunk to the Trunk
Landing Group assigned to this trunk. It waits for the duration of the Auto Attendant Delayed Timer for any
of the extensions in the Trunk Landing Group to answer the call.
If none of the extensions in the Trunk Landing Group answers the call before the expiry of the Auto
Attendant Delayed Timer, ETERNITY processes the call according to the type of Auto Attendant - Built-In
Auto Attendant or Voice Mail Auto Attendant, set for the trunk.
875
To enable Delayed Auto Attendant, set the timer to the desired value from the list. By default, it is set as
Never that is it is disabled.
Voice Mail Auto Attendant Profile: Select the desired Voice Mail Auto Attendant Profile from 1 to 16, if
you have enabled Voice Mail Auto Attendant for a time zone. By default, 1 is selected.
DISA: This parameter is to be configured if you want to enable Direct Inward System Access (DISA) on
the trunk ports on which you will apply the template.
DISA can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break Hours (BH) and
Non-Working Hours (NH).
For each Time Zone, you may select the desired DISA option from the following:
PIN Auth.-Multiple calls: Select this option if you want to enable DISA with PIN Authentication and
allow multiple calls during the DISA login session.
CLI Auth.-Multiple calls: Select this option if you want to enable DISA login with CLI Authentication
and allow multiple calls to be made during the DISA login session.
Caller numbers that do not match with the CLI Table will be routed as per the logic of the Trunk Feature
Template.
CLI Auth.-One call: Select this option if you want to enable DISA session with CLI Authentication, and
allow only a single call to be made during the DISA login session. This form of DISA is used when
ETERNITY is installed in a Gateway application. This form of DISA is applicable on CO Trunks only.
By default, DISA is disabled for all Time Zones.
Trunk Auto Answer: This parameter is relevant only if you want to enable the Trunk Auto Answer
feature on the Trunk ports on which this template is applied.
Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller
with a voice message before the call is actually handled.
You can set the following types of Trunk Auto Answer:
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OFF: Select this option if you do not want Trunk Auto Answer on the trunk.
For all Calls: Select this option if you want all incoming calls landing on the trunk line to be answered.
When Busy: Select this option if you want the system to answer incoming calls on the trunk to be
answered if the landing destination is busy.
Delayed: Select this option if you want ETERNITY to answer the incoming calls on the trunk if not
answered by the landing destination within a certain time period, set as the Delayed Trunk Auto
Answer Timer.
The system first routes the incoming calls to the Trunk Landing Group. It waits for the duration of the
Delayed Trunk Auto Answer timer for any of the extensions in the Trunk Landing Group to answer the
call.
If the call is not answered by the any of the extensions, before the expiry of this timer, the system
answers the call and routes it as per the Trunk Auto Answer logic.
If you select Delayed as the Trunk Auto Answer option, you must also configure the Delayed Trunk
Auto Answer timer. Valid Range of the timer is 01 to 99 seconds. By default, it is set as 10
seconds.
Trunk Auto Answer can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break
Hours (BH) and Non-Working Hours (NH).
By default, Trunk Auto Answer is Off.
If you have enabled Trunk Auto Answer for All Calls, When Busy or Delayed, you must also set the Trunk
Auto Answer Greeting Message.
Trunk Auto Answer Greeting Message: This parameter is to be configured only if you have enabled
Trunk Auto Answer For All Calls or When Busy for a Time Zone in the previous parameter.
Assign the number of the Trunk Auto Answer Greeting with which callers will be greeted. For this you must
first record a Voice Module with the desired Greeting Message and assign it to this parameter.
You can assign up to 4 Greetings Messages for Trunk Auto Answer and assign a different Greeting
message for each Time Zone. Refer the topic Voice Message Applications for instructions on configuring
the greetings.
Enter the number of the Greeting Message you want to be played for each Time Zone. By default, Trunk
Auto Answer Greeting number 1 is assigned to all Time Zones.
Trunk Auto Answer RBT Message Type: This parameter is to be configured only if you have enabled
Trunk Auto Answer 'For All Calls' or 'When Busy' for a Time Zone in the previous parameter.
When Trunk Auto Answer is enabled on a trunk, the system will answer the caller with a Greeting message
once, and play the Ring Back Ton (RBT) Message Type you have selected. You can select an RBT
Message Type from the following options:
None: The system will play Ring Back Tone to the caller after the Trunk Auto Answer Greeting
Message.
Internal MOH: The system will play internal music-on-hold to the caller after playing the Trunk Auto
Answer Greeting Message.
External Music: The system will play music to the caller from an external source connected to the
Analog Input Port.
If you select this option, you must also connect an External Music source to the Analog Input Port. You
may refer to the Installation Instructions for your model of ETERNITY.
RBT Message: The system will play a voice message continuously to the caller.
877
If you select this option, you must first record a Voice Module with the desired RBT Greeting Message.
You can set up to 4 RBT Messages. You can also assign a different RBT message for each Time Zone.
Refer the topic Voice Message Applications for instructions on configuring the RBT Message.
Assign the number of the RBT Message you want to be played to callers in each Time Zone.
By default, 'None' is selected as RBT Message Type for all Time Zones.
Trunk Auto Answer Busy Bye Message: This parameter is to be configured only if you have enabled
Trunk Auto Answer ('For All Calls' or 'When Busy') for a Time Zone.
When Trunk Auto Answer is enabled on a trunk, the system will answer the caller with a Greeting message
once, and play the Ring Back Tone (RBT) Message Type you have selected (see previous parameter)
continuously for the duration of the Built-In Auto Attendant Inactivity Timer. If the landing destination (called
extension) is busy on the expiry of this Timer, the system will inform the caller about the busy state in two
ways, which you can select from the following options:
Bye Message: The system will play a voice message to the caller.
If you select this option, you must first record a Voice Module with the desired Bye Message.
You can set up to 4 Busy Bye Message. You can also assign a different Busy Bye message for each
Time Zone. Refer the topic Voice Message Applications for instructions on configuring the Busy Bye
Message.
Assign the number of the Bye Message you want to be played to callers in each Time Zone.
By default 'None' is selected as Busy Bye Message for all Time Zones.
Priority: Select a Priority Level for the trunks on which the template will be applied.
Each trunk of the ETERNITY can be assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority.
Whenever there are incoming calls on multiple trunks, the call on the trunk with higher priority will be
answered by the Operator extension first. To know more, read the feature description Priority.
By default, the Priority of all trunks is set to '9-Highest'. Decide what Priority Level you will assign to the
trunks and set the desired level for the trunk.
878
SMDR-OG Storage: This flag is used to enable or disable the storage of details of outgoing calls from the
trunk. Please refer the topic Station Message Detail Recording-Storage for more details. By default,
storage of outgoing calls is enabled.
SMDR-IC Storage: This flag is used to enable or disable storage of details of incoming calls on the trunk.
Please refer the topic Station Message Detail Recording-Storage to know more. By default, storage of
incoming calls is enabled.
Hold on DSS Key Press: This flag defines the 'Hold' state of the external called party, when an extension
user presses a DSS key to dial another port.
For example, the DKP extension user (on DKP-001 port) is in the middle of speech with an external party
on a Trunk port CO-002.
If extension user of DKP-001 presses a DSS key to call another extension port DKP-003, two situations
are possible, depending on whether the Hold on DSS Key Press flag is enabled or disabled:
When the Hold Flag is enabled: CO-002 will be played music-on-hold. DKP-001 will hear Ring Back
Tone and the call will be placed on DKP-003.
When Hold Flag is disabled: CO-002 will be disconnected. DKP-001 will hear Ring Back Tone, and
call will be placed on DKP-003.
Forced Account Code: This parameter is related to the Account Codes feature of the ETERNITY. This
flag must be enabled, if the feature Forced Account Code is to be applied on the trunks.
When this flag is enabled, the system will prompt extension users to dial the Account Code whenever they
grab a trunk to dial out a number. The system will allow extension users to dial out numbers only when
after they have dialed the Account Code or Name.
By default, the flag is disabled. Refer the feature description for Account Codes to know more.
Account Codes feature must also be enabled in the Class of Service of extension users who are to be
allowed this feature.
Call Cost Calculation Pulse Rate Option: This parameter is to be configured only if you want to apply
the Call Cost Calculation (CCC) feature on the trunks on which the template is applied.
You have four options for Pulse Rate Types. Select from Pulse Rate Type for Pulse Rate Option 1 to 4
which you want to apply on the trunks.
Call Cost Calculation Time Schedule: This parameter is to be configured only if you want to apply the
Call Cost Calculation (CCC) feature on the trunks on which the template is applied.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
must first define the Time Zone (time of the day) for which a particular Pulse Rate should be applied and
the Time Schedule for each Time Zone.
You can configure up to four different Time Zones - T1, T2, T3 and T4 with different Pulse rates in the
CCC-Configuring Pulse Rate Types.
Now, configure the Call Cost Calculation Time Schedule, by specifying the Start Time and the End time (in
24hours: minutes format) for each Time Zone.
The default Time Schedule (starts and end time) for each Time Zone Index are as follows:
Time Zone Index
Start
Time
End Time
T1
00:00
23:59
T2
00:00
23:59
T3
00:00
23:59
879
Start
Time
End Time
T4
00:00
23:59
If your service provider offers the same Pulse Rate for the entire day,
configure only one Time Zone Index with the Pulse Rate, for instance, T1, in the CCC-Normal Pulse
Rate Table.
Now, set the Time Schedule for Time Zone, T1, with the start and end time in Hours: Minutes format;
set the start and end time of the other Time Zone Index, T2 to T4, to 00:00 (hours: minutes).
Similarly, if your service provider supports two different Pulse Rates in a day, set the Start and the End time
for two Time Zones and set the other two to 00:00.
880
Apply the Trunk Feature Template you customized to the CO Port type.
Go to the CO software ports to which this Template you customized is to be assigned, for instance CO001 to 003.
Enter the number of the Trunk Feature Template you customized, 02, in the Trunk Feature Template
field of this port.
881
Enter the number of the template you customized in the field Trunk Feature Template of the Mobile
Software ports to which you want to assign this template.
Repeat the same steps to create another template and apply it on the desired Trunk Port type.
To apply the template on E&M ports, open E&M Parameters page under E&M Configuration.
To apply the template on SIP Trunks, open SIP Parameters page under VoIP Configuration.
To apply the template to BRI Trunks, open BRI Parameters page under BRI Configuration.
To apply the template on T1 lines, open T1 Port Parameters page under T1E1 Configuration.
To apply the template on E1 lines, open E1 Port Parameters page under T1E1 Configuration.
882
Refer the following table for the parameter numbers and the values for the codes.
883
884
For example: Assign Trunk Landing Group 03 to Working Hours and Break Hours and Trunk Landing
Group 04 to Non-Working Hours in Template 02.
To assign TLG 03 to Working Hours, dial:
5802-1-02-03-03
Where,
02 is the template number
03 is the parameter number for TLG-Working Hours
03 is the code for the TLG to be assigned to Working Hours.
To assign TLG 03 to Break Hours, dial:
5802-1-02-04-03
Where,
02 is the template number
04 is the parameter number for TLG-Break Hours
03 is the code for the TLG to be assigned to Break Hours.
To assign TLG 04 to Non-Working Hours, dial:
5802-1-02-05-03
Where,
02 is the template number
05 is the parameter number for TLG-Non-Working Hours
03 is the code for the TLG to be assigned to Non-Working Hours.
To default Trunk Feature Templates, dial:
5801-1-Trunk Feature Template Number to default a single template.
5801-2-Trunk Feature Template Number-Trunk Feature Template Number to default a range of
templates.
5801-* to default all templates.
885
886
Exit SE mode.
Vocoder Preference: Vocoders are the various voice codecs used to compress the data in RTP packets
for optimum use of bandwidth and for ensuring voice quality. You can set 7 Vocoder options in the order of
preference for a SIP trunk.
The Vocoders supported by the VoIP card of the ETERNITY in the order of preference, from 1st to 7th, by
default are: G.723, G.729AB, GSM FR, iLBC - 30 ms, iLBC - 20 ms, G. 711-Law, and G. 711 A-Law.
If you do not want to select any Vocoder, you can select the option 'None' in the Template. However, if all
Vocoder Preferences from 1 to 7 are set to 'None', incoming and outgoing calls will be blocked.
G.723 Bit Rate (kbps): You can select the Bit Rate for G.723 codec as 5.3 kbps or 6.3 kbps. When G.723
is negotiated, the selected Bit Rate will be applied only when sending the RTP packets. When receiving
RTP packets from the remote end, both Bit Rates of G.723 will be accepted. The default G.723 Bit Rate is
6.3kpbs.
Silence Suppression for Vocoders: This parameter suppresses the 'Silence' packets, allowing only the
Voice packets through. ETERNITY supports Silence Suppression for all Vocoders except GSM FR.
Silence Suppression must be disabled if you have selected 'Pass Through' as the Fax Type.
SIP Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive Gain)
and Outgoing Speech (Transmit Gain) on the SIP Trunks/Extensions by changing the Rx Gain and Tx
Gain to the desired level. Different levels can be set for each port type in the SIP Gain Settings Template.
By default, SIP Gain Template 1 is assigned to all the SIP Hardware Templates. If you want to assign a
different Template, you must customize the SIP Gain Settings Template first and then assign the number
of the SIP Gain Settings Template in this Template. To customise the SIP Gain Settings, see Gain
Settings.
887
DTMF Type: This parameter will determine how the DTMF digits will be sent over the IP Network, when a
DTMF digit is pressed. The ETERNITY supports three DTMF options: i) RTP (RFC 2833), ii) SIP Info, and
iii) In-Band. You may select the appropriate option. By default RTP (RFC 2833) is selected.
RFC2833 Payload Type: If you have selected RTP (RFC 2833) as the DTMF Type, you must
configure the value of RFC2833 Payload Type. The RTP packets will be tagged as DTMF as per the
set value. The value of RFC2833 Payload Type can be set from 96 to 124.
Echo Cancellation: ETERNITY supports Echo Cancellation for SIP to CO trunk calls and SIP to Digital
Trunks (BRI, T1E1, Mobile, SIP) and Extensions (DKP, ISDN Terminals). If you want to apply Echo
Cancellation, you must enable configure the following parameters.
Enable: This flag is to be enabled to apply Echo Cancellation on SIP to CO and SIP to Digital Trunks/
Extensions. By default Echo Cancellation is enabled.
Tail Length (msec) for CO: Echo Cancellation Tail Length for SIP to CO trunks can be 32/64/128
milliseconds. By default, Echo Cancellation Tail Length for CO is set to 128 milliseconds.
Tail Length (msec) for Stations and Digital Trunks: Echo Cancellation Tail Length for SIP to Digital
Trunks/Extensions can be 32/64/128 milliseconds. By default, Echo Cancellation Tail Length for Digital
Trunks/Extensions is set to 32 milliseconds.
Jitter Buffer: The speed at which voice packets travel through a network depends on the condition of the
network. All voice packets may not come at the same speed. This variation in the delay in receiving
packets, known as Jitter, affects voice quality. Jitter Buffer helps overcome the delay in receiving voice
packets and improves voice quality. Jitter Buffer receives voice packets, stores them and sends it to the
DSP to process it at evenly spaced intervals, thus improving voice quality.
ETERNITY supports two types of Jitter Buffer: Static and Dynamic. Static Jitter Buffer's internal delay is
static, whereas, the Dynamic Jitter Buffer's internal delay adapts itself to the jitter in the network.
Type: Select the type of Jitter Buffer - Static or Dynamic - you want to use. If you selected Static Jitter
Buffer, you may set the 'Minimum Delay'. The value configured in the Minimum Delay determines the
size of the Static Jitter buffer.
If you selected Dynamic Jitter Buffer, configure the 'Optimization Factor' and 'Minimum Delay'. The
minimum size of the Dynamic Jitter buffer depends on the 'Minimum Delay' you configured. The
Optimization Factor determines the rate of adaptation of the Dynamic Jitter Buffer to the jitter in the
network.
Minimum Delay (msec): This parameter is to be configured for both Static and Dynamic Jitter Buffer.
The Minimum Delay determines the size of the Static Jitter Buffer and When Jitter Buffer type is
selected as Static, the Minimum Delay defines the size of the Static Jitter Buffer. The Static Jitter Buffer
will store each received voice packets for the time you set and then it will send it to DSP for voice
processing.
When Jitter Buffer type is Dynamic, the Minimum Delay specifies the minimum time for which the
Dynamic Jitter Buffer will store the received voice packet before sending it to the DSP for voice
processing. 'Minimum Delay' can be from 10 to 280 milliseconds. By default Minimum Delay is set to
150 milliseconds.
888
Optimization Factor: This parameter must be configured if you have selected Dynamic Jitter Buffer.
The Optimization Factor can be set from 01 to 13. By default, it is set to '10'.
In networks with higher jitter, a higher value should be set as Optimization Factor.
The actual size of the Dynamic Jitter Buffer will be determined by the DSP on the basis of the value of
the Optimization Factor and actual network condition. Dynamic Jitter buffer can go up to maximum 300
milliseconds.
Fax Type: This parameter allows you to select the protocol of Fax over IP (FoIP). You can send/receive
Fax from a Fax machine connected to the SLT port of the ETERNITY.
The ETERNITY VoIP Card supports the fax options: T.38 (UDPTL), T.38 (RTP), and Pass Through.
'Pass Through' and 'T.38' will work only if the peer devices also support the same option.
If you select 'Pass through' as Fax type, you must disable 'Silence Suppression'.
If the fax sent using T.38 is rejected, ETERNITY will use Pass Through for sending the Fax.
T.38 Fax Parameters: This parameter is relevant only if you have selected T.38 as the Fax Type for Fax
over IP. Receiving a good quality fax over SIP trunk depends on high 'Eye Quality Monitor' (EQM). The
higher the Eye Quality Monitor, the better the Fax quality. To improve the quality of T.38 fax reception, you
may configure the below parameters.
Version: Configure this parameter as supported by the Remote Peer, which may be a Proxy Server or
a SIP Device. While sending a fax, the Version will be sent in the re-INVITE Message to the Remote
Peer. While receiving a fax ETERNITY will accept a Version equal to or less than the configured
Version.
A different Version can be configured for each SIP Trunk. This is useful when you have proxy SIP
Trunks registered with different service providers supporting different versions.
The valid range for the Version is from 0 to 2. By default Version is set as 0.
Max Rate (Kbps): This parameter controls the Fax image transfer speed. As EQM is inversely
proportional to Fax Max Rate, if you receive poor quality fax, the Fax Max Rate should be reduced. The
default Max rate is 14.4 kbps.
Packet Period (msec): This parameter sets the sampling rate of TDM signal. If you cannot improve fax
quality by lowering Fax Max Rate, you may reduce the Fax Packet Period.
When you reduce the Fax Packet Period, the fax image will be sent at lower speed. EQM is inversely
proportional to Fax Packet Period.
The default packet period is 40msec. Do not change the default settings unless it is required.
Image Redundancy Level: The Fax Image Level is redundancy level for output Image, which can be
from 0 to 3.
Fax Image transfer speed is inversely proportional to this parameter. If this parameter is low then fax is
transferred faster. EQM is directly proportional to this parameter. If this parameter is high, good quality
fax can be achieved.
You may increase the Fax Image Level from 1 to 3 if the quality of fax does not improve with Fax Max
Rate and Fax Packet Period.
889
Data Redundancy Level: This is a redundancy level for T.30 control data. Fax Data Level can be set
from 0 to 7. Level 0 means no redundancy. Redundancy level increases from 1 towards 7. The higher
the level set, the slower would be the fax transmission.
EQM is directly proportional to this parameter. The higher the Fax Data Redundancy Level, the better
the EQM. By default, Data Redundancy Level is set to 3.
Pass Through FAX Codec: When the Fax option is selected as Pass Through, you must configure the
Pass Through FAX Codec as supported by the Remote Peer. The Remote Peer may be a Proxy Server or
a SIP Device.
You may select the Codec as G.711 A-law or G.711 LawWhile sending a fax this Codec is sent in the
re-INVITE message to the Remote Peer, but while receiving a fax ETERNITY will accept the fax with any
Codec.
Pass Through FAX Rx Gain (SIP-Digital Trunk Call): This parameter is of relevance if you have
selected 'Pass Through' as the Fax type. Pass Through Fax packets are transported using RTP protocol.
Normally, fax calls require less gain compared to voice calls. However, to improve fax reception,
ETERNITY allows the configuring of gain settings for fax. Fax gain settings consist of Data Gain and
Bypass Gain.
ETERNITY supports Fax Receive Gain for SIP to Digital Trunk calls as well as SIP to SLT Calls.
Configure Pass Through Fax Rx Gain (SIP-Digital Trunk Call), if Pass Through Fax is to be received on a
Digital Trunk (Mobile, BRI, T1E1PRI).
Data Gain (dB): select the dB Level for Data Gain. By default, Data Gain is set to -11 dB.
Bypass Gain (dB): select the dB Level for Bypass Gain. By default, Bypass gain is set to -9 dB.
Pass Through FAX Rx Gain (SIP-SLT Call): configure this parameter if Pass Through Fax is to be
received on a fax machine connected to an SLT port.
Data Gain (dB): select the dB Level for Data Gain. By default, Data Gain is set to -11 dB.
Bypass Gain (dB): select the dB Level for Bypass Gain. By default, Bypass gain is set to -9 dB.
890
Select a Template number you wish to customize, for example Template 02.
Now, apply this Template 02 on the SIP Trunks and SIP Extensions.
Go to the SIP Trunks to which this Template is to be assigned, for example SIP Trunk 02, 03 and 04.
Enter the number of the Template you customized, 02, in the field SIP Hardware Template of each of
these SIP Trunks.
Go to the SIP Extension software ports to which the Template is to be assigned, for example SIP
Extensions-005 to 008.
891
Enter the number of the Template you customized, 02, in the field SIP Hardware Template of each of
these SIP Extensions.
Repeat the same steps to customize another template and apply it on the SIP Trunks and Extensions.
892
893
For example, to select T.38 (RTP) as Fax Type in Template 02l, dial 7806-1-02-19-2
Where,
02 is the template number
19 is the parameter number for Fax Type
2 is the code for T.38 (RTP)
There may be other parameters that can be configured using Jeeves only. The above default template only
displays the parameters that can be configured using phone.
894
Exit SE mode.
Seizure Type: E&M Trunk Seizure Type is the line protocol that defines how the equipment seizes the
E&M Trunk. It is also referred to as Start Dial Supervision Signaling Protocol. ETERNITY supports the
following Seizure Types:
895
Immediate: The method of seizing the E&M Line using Immediate Start for Outgoing and Incoming
calls is illustrated below.
Outgoing Call:
While making an outgoing call, when the extension user of PBX A seizes the E&M Port of PBX A, the
status of the "M" wire of its E&M port will go high, indicating that it has seized the E&M line.
There will not be any signaling over the "E" wire of PBX A's E&M Port during seizure.
Incoming Call:
While receiving an incoming call over its E&M port, PBX-B will be ready to receive digits as soon as it
detects high state on its "E" wire.
There will not be any signaling over the "M" wire of E&M Port of PBX B while receiving an incoming
call.
Immediate with Ack: The method of seizing the E&M Line using Immediate with Acknowledgement for
Outgoing and Incoming calls is as follows:
Outgoing Call:
If this seizure type is selected, while an outgoing call is made by seizing the E&M Port, the 'M" wire will
go high immediately.
The remote end will acknowledge this by making its 'M' wire high, which in turn will activates (high) 'E'
wire of the E&M port of PBX-A.
On sensing high signal on 'E' wire, PBX-A will start dialing out the DTMF/Pulse digits.
Incoming Call:
On detecting high signal on "E" wire of the E&M port, the system will consider it to be an incoming call
seizure and hence it will immediately make its 'M" wire high, which will allow the remote end to dial out
the DTMF/Pulse digits.
Call Disconnection:
If the parameter 'Release Type' is configured as 'Status Change' and for this type of seizure, the "M"
wire at the remote end goes Low for some call condition, the call will be disconnected. For example,
"M" wire at the remote end will go 'Low' in the following conditions:
896
Remote end user dials invalid number and does not hang up on getting Error Tone.
Remote end user dials valid extension number and after conversation remote end hangs up first.
Remote end user dials valid number and extension does not reply, but the remote party does not
disconnect the call.
Remote end has made 'Orientation Type' of E&M port as 'Trunk' and for Incoming calls, all the
extensions in the trunk landing group are busy and also second call is not allowed to the extension
user, the system will disconnect the call after the expiry of the Ring Back Tone Timer, by making the
'M' wire Low.
Immediate + Wink: The method of seizing the E&M Line using Immediate + Wink Start for Outgoing
and Incoming calls is described below.
0V
0V
-48V
-48V
2001
PBX-A
2002
Wink
Pulse
Rx
Rx
Tx
Tx
SA
SA
SB
SB
3001
PBX-B
3002
Outgoing Call:
While making an outgoing call when the PBX A attempts a seizure (grab), the state of the "M" wire of
the E&M port of PBX A will go high.
To acknowledge this, the E&M port of PBX B will send Wink signal over its "M" wire, when PBX B is
ready to receive digits.
PBX A will wait for the duration of the 'Wait Wink Timer'. On receiving the acknowledgment in the form
of Wink signal on the "E" wire of its E&M port, before the expiry of the Wait Wink Timer, PBX A will
consider the trunk seizure as successful. Digits will be dialed out from E&M port.
If Wink is not received from PBX B within the Wait Wink Timer, PBX-A will drop the call.
Incoming Call:
While receiving an incoming call over the 'M' wire of its E&M port, PBX B will send the Wink signal to
the PBX A, which has initiated the seizure (grab).
The Wink signal will be sent by the PBX B when it is ready to receive the digits from the PBX A.
You can change the 'wink' pulse width by configuring the 'Wink Pulse Timer'.
The width of the Wink Pulse ranges from 0000 to 9999 milliseconds.
897
Immediate with Ack + Wink (MFC R2): The method of seizing the E&M Line using Immediate with
Ack + Wink (MFC R2) for outgoing and incoming calls is described below.
Outgoing Call:
When PBX-A attempts a seizure of (attempts to grab) the E&M Port, the 'M' Wire of the E&M Port of
PBX-A will go high and wait for the E&M Port of PBX B to turn its 'M' wire high.
PBX-B detects this on its 'E' wire. To acknowledge this, the E&M port of PBX-B will turn its 'M' wire high
and send a Wink signal over its 'M' Wire. Sending of the wink signal indicates the readiness of PBX-B
to receive the digits of the called party number.
PBX-A will wait for the duration of the 'Wait Wink Timer' to receive the acknowledgment in form of the
Wink Signal on the 'E' wire of its E&M port.
When PBX-A receives the acknowledgment from PBX-B, before the Wait Wink Timer expires, PBX-A
considers the trunk seizure as successful and starts dialing out DTMF digits as per the MFCR2
Signaling protocol.
However, if PBX-A does not receive the Wink Signal within the Wait Wink Timer', or if invalid Wink
Pulse is received (not according to Wink Pulse Timer), PBX-A will drop the call by turning its 'M' wire
low.
Incoming Call:
On detecting high signal on 'E' wire of the E&M port, PBX-B will consider it to be an incoming call
seizure and hence it will immediately make its 'M' wire high and send the Wink signal on the 'M' wire of
PBX-A to indicate its readiness to receive the called party number digits. The width of the Wink Pulse
(referred to as 'Proceed To Send' in MFCR2 Signaling) can be changed by setting the 'Wink Pulse
Timer'.
PBX-A dials out the digits as per the MFCR2 Signaling protocol.
When the called extension of PBX-B answers the call, PBX-B sends the Wink signal on the 'M' wire of
its E&M Port to indicate the call maturity.
Call Disconnection:
The call can be disconnected by the calling party, PBX-A, or the called party, PBX-B by changing the
status of the 'M' wire to Low.
Call Disconnection takes place when 'M' wire is low. So, it is recommended that the Call 'Release Type'
of the E&M Port for this Seizure Type (Immediate with Ack+Wink) be set to 'Status Change'.
898
If you select Immediate with Ack+Wink as the Seizure Type, you must configure the MFCR2 Signaling
parameters.
Seizure Pulse: The method of seizing the E&M Line using Seizure Pulse for Outgoing and Incoming
Calls is described below.
Outgoing Call:
While making an out going call from the E&M port of PBX-A, it will send Seizure Pulse over the "M" wire
of its E&M port to seize the line.
Incoming Call:
While receiving an incoming call over the E&M Port PBX B will detect valid Seizure Pulse over the "E"
wire of its E&M port.
Seizure Pulse can be set for various time periods T1, T2 and T3 as required.
The Seizure Pulse for T1, T2 and T3 ranges from 000 to 999 milliseconds.
Seizure Pulse + Wink: The method of seizing the E&M Line using Seizure Pulse + Wink for Outgoing
and Incoming Calls is as follows:
Outgoing Call:
While making an OG call, "Seizure Pulse" (as configured) will be sent on the "M" wire of E&M Port and
will start Wait Wink Timer and expect Wink from the remote device.
On receiving a valid Wink Pulse from the remote end within the Wait Wink Timer, digits will be dialed
out on the E&M Port.
If a valid Wink Pulse is not received from the remote end, digits will be dialed out on the expiry of Wait
Wink Timer.
Incoming Call:
On detecting valid Seizure pulse (matching with configured value of seizure pulse) on "E" wire of the
E&M Port, the E&M Port will send Wink pulse (of configured value) on "M" wire, and the call will be
considered to be present.
899
To make a call from PBX A to PBX B, the caller from PBX A presses the DSS Key of the desired E&M
port.
For as long as the DSS key is pressed by caller from PBXA, the signal on the "M" wire of E&M port of
PBX A will be high.
When the destination extension on PBX B answers, the caller from PBX A releases the DSS key, as
the line seizure is successful.
When the caller from PBX A releases the DSS key, the signal on the 'M' wire of the E&M port of PBX A,
and the "E" wire of the E&M port of PBX B will go low.
Radio A: This seizure type to be used for supporting Combat Net Radio (CNR) signaling on the E&M
port.
The E&M port of the ETERNITY will detect this pulse on the 'E' wire of its E&M port and recognize it as
a seizure signal (incoming call indication).
The length of this input signal (pulse) can be defined by setting the 'Minimum Pulse Width for Radio
Seizure' Timer. The ETERNITY will recognize the input signal on the 'E' wire of its E&M port as a
seizure signal only if the signal is present for the duration of this timer.
Once the incoming call is detected by the E&M port, the call is routed to the Routing Group number as
per the call routing logic configured for the E&M port.
When Routing Group member (extension) answers the call, speech is established with the CNR user.
Outgoing Call:
When any extension user of the PBX wants to contact the CNR user (Soldier), extension user must
grab the E&M port by dialing TAC/Selective Trunk Access / DSS key of E&M Port.
When E&M Port is grabbed by the extension user, the 'M' wire of the E&M port is made high, to indicate
the seizure signal to the radio equipment. The radio equipment then passes on the call to the CNR
user's wireless phone.
Incoming Call:
When status of "E" wire of the E&M port goes high for greater than or equal to the 'Minimum Pulse
Width for Radio Call' it is considered to be an incoming call, and the call is routed as per the current call
routing logic.
900
The Minimum Pulse Width for Radio Call' timer will be applicable only if the seizure type is configured
as 'Radio A' or 'Radio B'. The range of this timer is from 0000 to 9999 milliseconds. The default value of
this pulse is 150 milliseconds.
When an extension user of the PBX answers an incoming call, the 'M' wire of the E&M Port will be
made high.
The call between the extension user and the CNR user can be disconnected only if the extension user
disconnects the call. So, it is recommended that the call 'Release Type' of the E&M port be set to
'None'.
Radio B: Characteristics of M and E wire for seizure type 'Radio B' are as follows:
Outgoing Call:
When any extension user of the PBX wants to contact the CNR user (Soldier), extension user must
grab the E&M port by dialing TAC/Selective Trunk Access / DSS key of E&M Port.
When E&M Port is grabbed by the extension user, the 'M' wire of the E&M port is made high, to indicate
the seizure signal to the radio equipment. The radio equipment then passes on the call to the CNR
user's wireless phone.
Incoming Call:
When the status of "E" wire of the E&M port goes high for greater than or equal to 'Minimum Pulse
Width for Radio Call' ETERNITY considers it to be an incoming call, and routes the call as per the
current call routing logic.
When an extension user answers the call, there is no signaling on the 'M' wire of the E&M port by
ETERNITY.
The call between the extension user and the CNR user can be disconnected only if the extension user
disconnects the call. So, it is recommended that the call 'Release Type' of the E&M port be set to
'None'.
The following table shows the status of the "M" wire of an E&M Port while making Outgoing calls and
while receiving Incoming calls.
Status of 'M' Wire after
seizure and in conversation
in an OG Call
Status of 'M'
Wire when an IC
call is initiated
Immediate
High
Low
High
High
Low
High
Immediate + Wink
High
Wink
High
High
Wink
High
Seizure Pulse
Low
Low
Low
Low
Wink
Low
Express
Low
Low
Low
Radio A
High
Low
High
Radio B
High
Low
Low
901
Orientation Type: Configure the Orientation Type of the E&M Port according to your installation
scenario.
Select 'Station' if the E&M port is to function as Extension. All Extension-related parameters, the
Station Basic Feature Template and the Station Advanced Feature Template, will be applicable to
this port.
Select 'Trunk' as orientation type if the port is to be assigned the feature of a trunk line. All trunk-related
parameters, Trunk Feature Template, will be applicable to this port.
Select 'Tie Line'158 if the E&M port is to function as both a Station and a Trunk. The system will regard
the port as a Station for incoming calls and as a Trunk for outgoing calls. The Station Basic and
Advanced Feature Templates as well as the Trunk Feature Template will be applied on this port.
By default, the Orientation Type of all E&M ports is 'Station'.
Dial Type: Digits can be dialed over E&M Tie Line by two methods:
Tone: In this Dial Type, the DTMF signals will be sent on the "Tx" of the E&M port of the originating
side and it will be received over the "Rx" of the E&M port of the terminating side.
Pulse: In this Dial Type, the dialed digits will be sent on the "M" wire of the E&M port of the
originating side and will be received over the "E" wire of the E&M port of the terminating side.
The way digits are sent varies according to the Trunk Seizure Type, as described for each Seizure Type
below.
Express: the caller can make call by pressing the DSS key.
Immediate: the PBX seizes the Tie Line and the system will start the Pause Timer. On the expiry of the
Pause Timer the PBX sends the digits to the remote PBX.
Seizure Pulse + Wink: on receiving Wink signal from the terminating end, the originating side (which
initiates seizure) will start the Pause Timer and on expiry of Pause Timer it will start sending digits.
Seizure Pulse: the originating PBX system will start the Pause Timer after sending the Seizure Pulse.
On expiry of the Seizure Pulse Timer it will send digits to the terminating end.
Pulse Dial Ratio: This parameter is to be configured if 'Pulse' is selected as Dial Type in the previous
parameter. Select the Pulse Dial Ratio from any of the following values:
10PPS, 1:2
10PPS, 2:3
10PPS, 1:1
20PPS, 1:2
20PPS, 2:3
20PPS, 1:1
The default Pulse Dial Ratio is 10PPS, 1:2
Wait Wink Timer (sec): This parameter is to be configured, if 'Immediate + Wink' or 'Seizure Pulse +
Wink' have been selected as Seizure Type for the E&M port.
The Wait Wink Timer is the Time period for which the system waits for the acknowledgement in the
form of Wink Signal on "E" wire of the E&M port of the ETERNITY to consider it as a successful
seizure.
902
The range of this timer is from 000 to 255 seconds. By default the timer is set to '000' seconds.
Wink Pulse Timer (msec): This parameter is to be configured, if 'Immediate + Wink' or 'Seizure Pulse
+ Wink' have been selected as Seizure Type for the E&M port.
The Wink Pulse Timer defines the width of the Wink Pulse. The range of this timer is from 0000-9999
milliseconds. By default the timer is set to '0000' milliseconds.
Seizure Pulse Timers (msec): This parameter is to be configured, if 'Seizure Pulse' is selected as
Seizure Type for the E&M port.
T1: This is the time period of the first ON period of the 'Seizure Pulse'.
T2: this is the time period of the second ON period of the 'Seizure Pulse'
T3: this is the time period of the third ON period of the 'Seizure Pulse'.
The range of T1, T2 and T3 is from 000-999 milliseconds. By default the timer is set to '000'
milliseconds.
Minimum Pulse Width for Radio Seizure (msec): This Timer is to be configured if 'Radio A' or 'Radio
B' have been selected as the Seizure Type for the E&M ports.
The Minimum Pulse Width for Radio Seizure defines the time for which the ETERNITY will wait to
detect the width of the pulse sent by the Radio Interface Device159 on the 'E' wire of the E&M port and
recognize it as a seizure signal (incoming call indication).
The range of this timer is from 000 to 999 milliseconds. By default the timer is set to '150' milliseconds.
Release Type: ETERNITY supports four methods to 'release' E&M calls based on, which end will
release the call and 'release pulse width'. These are:
None: Select this option if you have selected "Express" as Trunk Seizure Type for the E&M port. It
is advisable to keep the Release Type as "None" in case the protocol does not support any
signaling for disconnecting the E&M port, for the reason that 'Trunk-to-Trunk Inactivity Timer' will be
started if the E&M port with Release type "None" is involved in a Trunk-to-Trunk call.
Release Pulse: Select this option if the specific Pulse width of the Release Pulse is to be used to
disconnect the call. This Pulse width is configurable, as shown below:
159. Connected on the 'M' wire of the E&M port. The Radio Interface Device sends pulse of approximately 100msec or higher.
903
The call can be disconnected by either party by sending Release Pulse. Consider the following
example:
Status Change: Select this option, if Status change of 'M' wire (M wire is low) is to be considered for
release of the E&M call.
By default, 'Status Change' is selected as the Release Type for all E&M ports.
If you selected Immediate with Ack + WInk as Seizure Type for the E&M Port, select Status Change
as Release Type for the port.
Release Pulse Timer (msec): This timer is to be configured if you selected the option 'Release Pulse'
as the Release Type for E&M calls in the previous parameter.
This timer defines the specific Pulse width of the Release Pulse is to be used to disconnect the call.
The range of this timer is from 0000-9999 milliseconds. By default the timer is set to '0000'
milliseconds.
CCS - When End Point: CCS (Compander Control Signal) is a type of signal used by PLCC Networks
to improve the quality of speech transmission. The PLCC network awaits this signal from the PBX
when speech is established. ETERNITY supports CCS. The system sends CCS signal to the PLCC
panel.
This parameter is relevant if the E&M line is being used in a Power Line Communication Network
(PLCC).
This flag should be enabled if the E&M port is used as an Endpoint in a PLCC network. When the E&M
Port is used as an Endpoint, the system sends CCS to the PLCSS panel while making an outgoing call
through the E&M port and when receiving an incoming call on the E&M port.
904
CCS - When Transit Exchange: This flag is to be enabled if the E&M port is used as a Transit
exchange in a PLCC network.
When the E&M Port is used as a Transit Exchange: The system sends CCS to the PLCC panel when
there is an Incoming/Outgoing Transit call through the E&M port.
By default, the flag is enabled.
DTMF Detection: This flag is relevant when the "Dial Type' for the E&M Port is selected as 'Tone'. This
flag is of significance while receiving incoming calls.
Max. OG Pulse Digit Count: This count defines the maximum number of digits that can be dialed out
to make a call. When dialing out the number, if the number of digits exceeds this count, the port which
is used for dialing these numbers is released automatically.
Idle Wait Timer (sec): This timer signifies the time after which the codes could be simply (Station
Numbers or Station Numbers with Exchange ID) dialed over the E&M trunk.
The Idle Wait Timer is useful in two conditions:
When Forced Disconnection is used. For example, two exchanges A and B are connected
through E&M trunk. Extension 2002 of PBX A is talking to extension 3001 of PBX B over the E&M
line.
Extension 2001 of PBX A calls extension 3001 of PBX B and finds it to be busy.
Extension 2001 is allowed to use forced disconnection feature. Extension 2001 issues the forced
disconnection command. The PBX A disconnects the Extension 2002. It then waits for the Idle Wait
Timer to expire and then dials 3001 over the E&M trunk.
To stop any station from grabbing the E&M trunk until call is released. For example, when
Extension 2001 of PBX A goes On-Hook, PBX A sends a release signal over the E&M trunk to PBX
B. In turn, PBX B sends a release signal to PBX A as an acknowledgment.
The E&M Idle Wait Timer set in PBX A does not allow any other extension of PBX A to grab the
E&M trunk. Similarly, E&M Idle Wait Timer set in PBX B does not allow any extension of PBX to
grab the E&M trunk.
Flash Timer (msec): This Timer is significant when the PBX acts as a Transit exchange for a call. The
flash received on one E&M Port is generated on another E&M Port involved in a Transit call.
Pause Timer (msec): This Timer defines the time for which the system waits before dialing the outside
number after grabbing the E&M trunk.
Sometimes an extension user may not get a dial tone immediately on grabbing a trunk, in which case
the extension user may wait for the dial tone before dialing out the number. However, when the system
dials out the number, if there is no pause time, it is possible that the system may dial out the number
before getting the dial tone. This may result in a wrong number being dialed out. The Pause Timer
helps avoid this.
905
Pseudo Answer Supervision Timer (sec): This is the time period after which, the system will
consider the call as matured, irrespective of whether the call was answered or not. At the end of the
Timer, the system will start detecting Disconnect Supervision.
The range of this timer is from 000 to 255 seconds. By default the Timer is set to 030 seconds.
Ring Timer (sec): This is the time for which the extensions connected to ETERNITY ring for incoming
calls.
The Ring Timer is useful in situations where the users may not be able to immediately answer on the
first few rings. The range of this timer is from 000 to 255 seconds. By default the Timer is set to 255
seconds.
Inter-Digit Pause Timer (msec): This Timer defines the time gap to be inserted between digits of a
number string being dialed by the system.
The range of this timer is from 000 to 999 milliseconds. By default the Timer is set to 750 milliseconds.
DTMF Out Dial: This parameter is of significance when the system dials out DTMF digits to enable the
device at the remote end (in this case a PBX) to detect and decode the Tones. You must configure
both the DTMF ON Time and the Level (dB) according to the DTMF digit detection capacity of the
remote PBX.
For example, PBX A and PBX B are connected over E&M Line. PBX B detects DTMF digits only if the
tone remains present (ON) for 100 milliseconds frequency and at a transmit level of 4 dB. The DTMF
Out Dial parameter for PBX A should be configured accordingly. The DTMF Out Dial ON Time should
be set to 100ms and Level to 4dB.
ON Time (msec): This is the Time for which the DTMF digit tone will remain ON, while being dialed
out by the ETERNITY. The range of this timer is 50 to 500 milliseconds. By default the ON Time is
set to 100 milliseconds.
This Timer must be configured according to the DTMF digit detection capacity of the remote device.
906
DTMF Tx Level (dB): This is the Transmit Level of the DTMF digit dialed out by the system. The
range of DTMF Out Dial 'Level' is from 0 to 7. By default DTMF Out Dial Level is set to 3.
MFC R2 Signaling: This parameter is relevant only if you selected 'Immediate with Ack + Wink' as the
Seizure Type. Configure the following timers related to MFCR2 Signaling.
Forward Tone Maximum ON Time (T1) (sec): the range of this timer is from 1 to 99 seconds. By
default it is set to 15 seconds.
Forward Tone Maximum OFF Timer (T2) (sec): The range of this timer is from 1 to 99. By default
it is set to 24 seconds.
Maximum Compelled Cycle Timer (T3) (sec): the range of this timer is from 1 to 99. By default it
is set to 15 seconds.
Pulse Duration for Pulse Signal (msec): The range of this timer is from 001 to 999. By default, it
is set to 150 seconds.
Pulse Signal Maximum Wait Timer (sec): The range of this timer is from 1 to 99. By default, it is
set to 15 seconds.
First Forward Tone Wait Timer (sec): The range of this timer is from 8 to 24. By default, it is set to
15 seconds.
Minimum MF Signal Persist Timer (msec): The range of this timer is from 1 to 255 seconds. By
default, it is set to 20 seconds.
Prefix String: This parameter is useful when the PBXs connected using E&M lines do not send 0
when the user dials a number of another exchange. You must configure this parameter as zero along
with the exchangess ID and appropriately configure the strip digit count parameter in the CUG Table.
For details, see Prefix String Feature.
Category (Logical Partitioning): This parameter assigns the E&M Port to a trunk category for the
purpose of Logical Partitioning. By default all E&M Ports are assigned to Category 3160.
If you have re-defined Category 3 or have assigned E&M ports to a different category, say Category 2,
enter the same number here.
You may configure the call permission between the Category assigned to E&M Ports and other
Categories. Refer the feature description Logical Partition to know more.
160. Trunk ports used to interconnect two PBXs are assigned this category.
907
Now, apply the E&M Feature Template you customized to the E&M Ports and T1E1PRI ports.
908
Apply the E&M Feature Template you customized to the E&M Port by entering the template number in the
E&M Feature Template field of this port.
909
Click the tab of the desired T1E1 trunk port number (1 to 8), you wish to apply the E&M feature Template.
Enter the number of the Template you customized in the field E&M Feature Template of the selected
T1E1 trunk port.
910
For example, you want to Immediate with Ack +Wink in the E&M Template number 2.
Dial 6002-1-2-01-9
Where
2 is for E&M Template number 2
01 is parameter number for Seizure Type
9 is the value of Immediate with Ack+Wink
To restore default values to the Parameters of an E&M Feature Template:
Dial 6001-1-Template Number to restore default values of the parameters of a single template.
Dial 6001-2-Template Number-Template Number to restore the default values of a range of
templates.
Dial 6001-* to restore the default values of all templates.
911
912
Exit SE mode.
Configuring CO Trunks
The ETERNITY supports a maximum of 128 Analog Two-Wire Trunk Lines161. Before you begin configuring the
CO trunk ports, ensure that the CO trunk card has been installed correctly.
You may configure the CO ports from Jeeves and using a Telephone.
Click CO Parameters.
161. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.
913
CO No.: This non-editable field is the number of the software port of the CO Trunk.
Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the CO Card has been
inserted. 'Port' is the number of the CO trunk port on that card.
By default the ETERNITY can detect and assign the hardware slot and port numbers automatically to the
CO (software) ports. However, if required, you may change the Hardware Slot and Port assigned to the CO
software port. In which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
Enable Port: This flag is for enabling or disabling a CO Trunk port. When a CO Trunk port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clearing this check box.
You may disable CO port in case of trouble with the CO line.
Name: You may assign a 'Name' to each CO Trunk to facilitate identification. Whenever there is an
incoming call without CLI on this port, the Name you have programmed will be displayed on the landing
extension.
The Name of the port may be the name of the Service Provider of this Trunk Line (recommended).
The Name may comprise a maximum of 18 characters.
CO Hardware Template: A CO Hardware Template is a set of features that completely define the
behavior of the hardware port of the CO, such as Type of CO Trunk, AC Termination Impedance, PulseTone Dialing, Answer Supervision, Disconnect Supervision, DTMF detection, etc.
Apply a CO Hardware template to the CO trunk port. The ETERNITY offers 50 CO Hardware Templates.
By default, CO Hardware Template number 01 is assigned to all CO Trunks. Refer the topic CO Hardware
Template to know more.
Check if this default template fulfills the feature requirements of the CO Hardware Ports by clicking the 'CO
Hardware Template' link.
If CO Hardware Template 02 fulfills your requirements, and if the same features at to be applied on all CO
trunk ports, retain Template 02. Similarly, if you want only a few changes to be made to Template 02 and
apply it on all CO Ports, make the changes and retain the template.
However, if different sets of features are to be allowed to different CO hardware ports, then prepare
separate CO Hardware Templates and apply them on the ports as required. To do this,
914
Apply the CO Hardware Template you customized to the CO Port by entering the template number in
the CO Hardware Template field of this port.
Repeat the same steps to customize another template and apply it to the CO Port.
To know more about the hardware port features and customizing templates, refer the topic CO Hardware
Template.
Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator, Auto
Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the behavior
of a Trunk. Apply a Trunk Feature Template to the CO Trunk port. By default, Trunk Feature Template 01
is applied on all CO Trunks as well as all other trunk types like ISDN BRI, ISDN T1/E1/PRI, GSM, and
VoIP. Refer the Trunk Feature Template topic to know more.
Click the Trunk Feature Template link to open the page. Check if the default Template 01 fulfills your
requirement for the CO Trunk port.
If the default Template 01 does not fulfill your requirement, you may prepare a different Trunk Feature
Template and apply on all CO Ports. For this,
Go to the CO Software Port Number you want to assign the Template you prepared.
Enter the number of the Template you prepared (02) in the Trunk Feature Template field.
You may also prepare different Templates for different CO Ports, for example Template 02 for certain
ports, Template 03 for others. In which case, follow the steps described above. For each CO Port, enter
the number of the template you have prepared for that port.
To know more about customizing templates, refer the topic Trunk Feature Template.
Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the CO Trunk
port.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a preference
number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same preference must be
assigned the same Cost Factor. Different trunk types can also be assigned the same Cost Factor. These
trunks are used for routing calls.
915
Assign a Cost Factor to the CO Trunk port, for instance 02, and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the CO Trunk Port 001
only,
You must first assign a Cost Factor (01-99) to CO Port 001, for example, 02.
Click the Least Cost Routing - Number Based link to open the page.
916
Advanced Configuration
The above listed parameters fulfill the basic CO trunk port configuration requirements of most users. However, for
users who need advanced features like Call Budget on the CO trunk ports, you may click the Advance button and
program the following parameters:
Call Budget: If you want to enable 'Call Budget on Trunk' feature, configure the following parameters for
this CO trunk port:
Type: Select the type of Call Budget on TrunkAmount, Minutes or Number of Callsto be applied on
this CO trunk port. By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
Calls: If you selected Calls as the Call Budget Type, enter the number of calls in this field. By default,
the number of calls is set to 9999.
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes to be reset on a
particular date of every month.
917
Scheduled (Date): Enter the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select Daily if your plan suggests
so.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset from SE and SA Mode, referred
to as Manual Reset. Refer the feature description Call Budget on Trunk.
Call Back: This parameter is related to the Call Back on Trunk Port feature. If you want to enable the 'Call
Back on Trunk Port' feature on this CO trunk, configure the following parameters:
Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this flag
is disabled on all trunk port types. By default, the flag is disabled.
Call Back Timer: This is the duration for which the system waits for the caller to disconnect before
applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to 10
seconds.
Call Back Mode: Select from the following options how a Call Back call answered by the remote party
should be routed:
Built-in Auto Attendant
PIN Authentication - Multiple Calls
CLI Authentication - Multiple Calls
CLI Authentication - Single Call - Answer Signaling
Operator
By default, Operator is selected as the Call Back Mode.
Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the CLI number. If you want the call back to be made to a different number, select Alternate
Number.
By default, CLI number is selected for Call Back.
Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all CO trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all CO trunks and Trunk Port types, retain this
list.
If you want a different set of number strings to be programmed for this CO Trunk, select a different
Number List, and assign it to the CO trunk port.
You may program the Incoming Number List either from the Number List page or by clicking the
Incoming Number List link to reach the Number List page.
Refer the topic Number List to know more, and for configuration instructions.
918
Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the Incoming Number List, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
example, for the number string programmed at Index 1 in the Incoming Number List, a corresponding
number string at the same Index, Index 1, should be programmed in the Outgoing Number List.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all CO trunks as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this CO Trunk port.
You may program the Outgoing Number List either from the Number List page or by clicking the
Outgoing Number List link to reach the Number List page.
Refer the topic Number List to know more, and for configuration instructions.
Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the Same Port or an Outgoing Trunk Bundle Group (OTGTBG).
Select Same Port if you want the call back to be made using the same port on which the missed call is
received. If you select OGTB Group, the call back will be made using the OGTB Group, which you have
defined.
By default, Same Port is selected.
OGTB Group: If you selected OGTB Group for making the call back in the previous parameter, you
must define the OGTB Group that must be used in this parameter.
By default, OGTB Group 01 is assigned.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTB Group that you define here for Call Back.
If you have completed configuration of all the above listed CO Parameters, click 'Submit' at the bottom of
the page to save your changes.
919
920
For commands to program Call Budget on CO trunks, refer the topic Call Budget on Trunk.
921
Code is
0 for Disable
1 for Enable
Default: Disabled
To program Call Back Timer on CO port, dial:
3311-1-CO-Call Back Timer to set timer for a single CO trunk.
3311-2 -CO-CO-Call Back Timer to set same timer duration for a range of CO trunks.
3311-*-Call Back Timer to set the same timer duration for all CO trunks.
Where,
CO is the number of the software port of the CO trunk from 001 to 128.
Timer is from 01 to 99 Sec.
Default: 10 Seconds
To select Call Back Mode for CO port, dial:
3312 -1-CO-Call Back Mode to set call back mode for a single CO trunk.
3312 -2-CO-CO-Call Back Mode to set same call back mode for a range of CO trunks.
3312-*-Call Back Mode to set the same call back mode for all CO trunks.
Where,
CO is the number of the software port of the CO trunk from 001 to 128.
Call Back Mode is from 1 to 5
1 for Built-in Auto Attendant
2 for PIN Auth. - Multiple Calls
3 for CLI Auth. - Multiple Calls
4 for CLI Auth. - Single Call - Ans. Sig.
5 for Operator
Default: Operator
To program Call Back On for CO port, dial:
3313-1-CO-Call Back on to program call back on for a single CO trunk.
3313-2-CO-CO-Call Back on to program the same call back on option for a range of CO trunks.
3313-*Call Back on to program the same call back on option for all CO trunks.
Where,
CO is the number of the software port of the CO trunk from 001 to 128.
Call back on is
1 for CLI Number
2 for Alternate Number
Default: CLI Number
To assign Call Back - Incoming Number List to a CO port, dial:
3314-1 -CO-Incoming Number List to assign a list to a single CO trunk.
3314-2-CO-CO-Incoming Number List to assign the same list to a range of CO trunks.
3314-*-Incoming Number List to assign the same list to all CO trunks.
Where,
CO is the number of the software port of the CO trunk from 001 to 128.
Incoming Number List is from 01 to 16.
Default: 15
To assign Call Back - Outgoing Number List to a CO port, dial:
3315-1 -CO-Outgoing Number List to assign a list to a single CO trunk.
3315-2-CO-CO-Outgoing Number List to assign the same list to a range of CO trunks.
3315-*-Outgoing Number List to assign the same list to all CO trunks.
Where,
922
CO is the number of the software port of the CO trunk from 001 to 128.
Incoming Number List is from 01 to 16.
Default: 16
To select Call Back From port for a CO Trunk port, dial:
3316-1-CO-Call Back From to select Call Back From for a single CO trunk.
3316-2-CO-CO-Call Back From to select the same Call Back From for a range of CO trunks.
3316-*-Call Back From to select the same Call Back From for all CO trunks.
Where,
CO is the number of the software port of the CO trunk from 001 to 128.
Call Back From is
1 for Same Port
2 for OGTB Group
Default: Same Port
To assign Call Back - OGTB Group to a CO port, dial:
3317-1-CO-OGTB Group to assign an OGTBG to a single CO trunk.
3317-2-CO-CO-OGTB Group to assign the same OGTBG to a range of CO trunks.
3317-*-OGTB Group to assign the same OGTBG to all CO trunks.
Where,
CO is the number of the software port of the CO trunk from 001 to 128.
OGTB Group is from 01 to 32.
Default: 01
Exit SE mode.
923
How it works
The ISDN numbering plan with ETERNITY revolves around Multiple Subscriber Number (MSN) and Direct
Dialing In (DDI).
With MSN feature, practically all the station users or other specific terminals can be given a unique
telephone number. One station can be assigned a number 2765400, other station can be assigned
2765401, etc. Hence, for one subscriber the Service Provider can assign multiple numbers. Hence this
feature is called Multiple Subscriber Number.
The ETERNITY can be programmed to land all the calls coming through various channels of the BRI line
on the Operator just like in case of normal trunk.
Alternatively, using DDI feature of ISDN the calls can be made to land directly on the desired stations. To
accomplish this requirement, each station should be given a unique directory number. On assigning
directory number, a table is formed internally called DDI table as shown below:
924
Directory Number
000
03
005
04
006
05
008
06
009
07
When a caller calls a MSN number, the call lands on the PBX. The PBX compares the incoming number
with the DDI table. If the incoming number matches with any number in the DDI table, it routes the call to
the specific station. If the incoming number does not match with any number in the DDI table then it is
matched with CLI number. If it matches with any number in the CLI table, the call is routed according to the
CLI table. If the number does not match with either of these tables then the call is routed to the landing
destination.
ISDN
Network
NT 1
ETERNITY
Power
S/T
Interface
U-Interface
(2 wire)
Most of the Service Providers provide the NT1 along with the BRI line.
At the Customer's Premises, the BRI line is terminated on the NT1. The S/T interface of the NT1 is
connected to BRI port of the ETERNITY.
The configuration details of U interface (RJ-45) at NT1 are given below:
Pin Number
Pin Detail
Tx
Rx
The configuration details of S/T interface (RJ-45) on NT1 are given below:
Pin Number
Pin Detail
Rx1
Tx1
Tx2
Rx2
Since 32 BRI ports are supported at present and as per protocol a maximum of 8 terminals can be
connected to the BRI port; at present 64 software ports are supported for ISDN terminals, namely 01 to 64
(Maximum 64 Terminals).
ISDN terminals (software ports) do not have any hardware slot and port Id of their own.
ISDN terminals (software ports) are associated to the BRI software port and the BRI software port has a
hardware slot and port ID of its own.
Each ISDN terminal is assigned an access code (flexible number), station basic template, station
advanced template and a CPU group.
925
The ISDN terminals do not have a direct software port number as other port types have. Rather the ISDN
terminals have a derived software port number. Such derived software port numbers are used while
programming extension name or assigning Station Basic Feature or Station Advanced Feature templates
or programming Routing Group.
For Call routing from the BRI NT:
When call is made from the ISDN terminal the BRI port will check the calling party number sent by the
ISDN Terminal. If the calling party number = Programmed access code of the ISDN Terminal which are
assigned to the BRI port, the BRI Terminal will process the call further as per the Station Basic Feature
Template and Station Advanced Feature Template assigned to the ISDN Terminal.
If the ISDN terminal doesn't send the calling party number while making the OG call, the Station Basic/
Advanced Feature template assigned to the BRI port will be used, for call processing.
When the ISDN terminal doesn't send the calling party number = Access code programmed for the BRI
port on which it is connected, while making the OG call, the Station Basic/Advanced Feature Template
assigned to the BRI port will be applied.
When ISDN terminal sends calling party number which is not programmed for any ISDN terminals
assigned to that particular BRI Port, neither its access code of that particular BRI port, the call will be
dropped, by the BRI port.
BRI port can be connected to a public ISDN exchange, private ISDN exchange or to an ISDN terminal (BRI
Access).
When the BRI port is connected to a Public ISDN exchange, it behaves as a terminal. ETERNITY supports
this function of BRI Access by assigning it a parameter viz. Orientation = Terminal. The signaling used for
this Orientation is Q.931-protocol.
When the BRI port is connected to an ISDN Phone or ISDN Video phone, it behaves as a 'network'.
ETERNITY supports this function of BRI Access by assigning it a parameter viz. Orientation = Network. In
this case, the BRI port behaves as network. The signaling used for this Orientation is Q.931- protocol.
When the BRI port is connected to a private ISDN (the main application of this configuration is CUG,
feature transparency, etc.), the BRI port is used as a pipe of 128Kbps. It is of no significance which end
acts as network and which acts as terminal (User) since the role of the BRI port in such case will be to
route the call depending on the signaling protocol applied on the BRI port (128Kbps link). ETERNITY
supports this function of BRI Access by assigning it a parameter viz. Orientation = Tie-line. In this case, the
BRI port behaves as 128Kbps link. The signaling used for this Orientation can be any of the Interexchange signaling protocol for BRI Access. The most commonly used is QSIG.
926
BRI port supports both automatic and fixed TEI negotiation, so that it can be integrated with a PBX of
another vendor. If 'Fixed' mode is selected, its value is required to be configured using SE commands or
Jeeves. TEI is 'Terminal Endpoint Identifier' protocol for negotiation used while connecting to the BRI port
with remote BRI port.
Video Phone
Making Data Call
Video Phone
An important application of BRI-NT is establishing a Video Call using Video phone. It is used just like other
ISDN Phones.
Video Phone can be used with ETERNITY supporting a BRI connection. ETERNITY does not support call
transfer from one Video Terminal to another Terminal. But the call routing is implemented by preparing
suitable OG Trunk Bundle Group (OGTBG).
The OGTBG for the Video Phone should be so formed by the SE that a BRI channel will be allotted by the
Call Processing logic (preferably).
If the OGTBG formed by the SE contains a non-ISDN trunk and if by the OGTBG logic this non-ISDN trunk
is to be allotted to the ISDN phone which has requested 2 B-channels then the call will be dropped. The
non-ISDN trunk is allowed only if the ISDN user makes an audio call. The PBX will allot first channel even
if other channels are available.
The ETERNITY will support supplementary feature viz. Hold only. If the terminal equipment uses Hold
feature to hold a video call, it will be applicable.
The ETERNITY does not support logic for converting audio call to video call.
Instead, the Video Phone user will have to use one Trunk Access Code (and hence OGTBG) to make and
receive Video call and another TAC to make an audio call.
User can program normal DDI routing logic to route the IC call to the Video Phone using the first or
additional channels.
Depending upon terminal equipment call, the call will be answered by the Operator and transferred to the
Video Phone.
927
Video Conference
As shown in the figure ETERNITY supports Video conference from BRI-NT port.
This feature works only if it is supported by the Service Provider. Video Conference is established mainly
by the Video Conferencing (VC) equipment.
A Video Conference system (H.320) can be connected to any of the BRI-NT port.
For Video conferencing, three BRI-NT ports of ETERNITY are connected to the three ports of the VC
equipment (The remaining one port of the Video Conference system remains free).
Thus user will assign at least 6 B-channels to the OGTBG that is to be assigned to the VC equipment.
(User can assign BRI-NT software port nos. 01 to 03 to the Video Conferencing equipment).
The user will program the routing of IC calls to the Video Conferencing equipment using DDI routing table
for placing IC video conferencing calls.
Video Conferencing generally requires 6 B-channels. But Video Conferencing can also be done at
lower bit rates also using the "aggregation" of 6 B channels which must be supported by the VC
equipment.
Please note that a Video Conference call cannot be transferred or kept on Hold.
928
For data-communication, connect the Router supporting ISDN-BRI or a Computer with ETERNITY ME
Card BRI to the BRI-NT port as shown below:
The computers connected in the LAN can browse the net through the BRI.
Remote LAN Access the Computers in the LAN can access the computer/computers in LAN at the
remote end (Branch office/Home office).
The data call can be made by the router requesting desired number of channels. This establishes a live
connection between the Router and the ISP through the PBX. The users on the LAN can browse the net as
normal using Internet Explorer or Netscape Navigator.
For this, the ETERNITY will allocate data channels only on the BRI-TE port so as to leave other channels
for speech calls when the system detects the call to be a data call.
Similarly, a Remote Computer can be accessed (Remote LAN Access) by dialing the Remote users'
number (The remote end PBX should be so programmed that the call made to a number lands directly on
the Router.) This establishes a permanent connection between the two Routers (and hence two LAN
networks).
Now the user at PBX-A can access the computer in LAN at the remote end in the same way as accessing
another computer on the same network.
Program the Trunk Landing Group such that all data calls will land on the PRI-NT port to which the Router
is connected using CLI based Routing logic or DDI based routing logic.
However, while routing call on the BRI-NT port, the PBX will check that the data channels reserved for data
communication on the BRI-NT port are enough to establish the call. Otherwise the call will be rejected.
The call will be rejected if the number of channels reserved for data calls, are already busy with one datacall.
929
ETERNITY will assign the Hardware Slot-Port automatically, when any card is inserted in the system.
Hardware slot is the number of the Universal slot of ETERNITY in which the BRI Card is inserted. Range of
slot number is 01-16. Port is the number of BRI hardware port on the card to which the BRI line is
connected. Range of Port is from 00-99.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields. By default, Hardware slotPort is 0000.
You may assign a Name to the BRI Port for identification of the port. The Name may consist of a maximum
of 12 characters. By default, it is blank.
Set Orientation Type for BRI Port. You may select from the following options:
Terminal
Network
Tie Line
By default, Orientation Type is Terminal.
930
Enable Power Feed flag, if you want ETERNITY to feed power to the terminal equipment. By default, it is
disabled.
Power Feed is supported only in Hardware Version V2R1 and onwards in ETERNITY BRI8 Card.
The TEI Negotiation mode will be set automatically as per the Interface Type selected for the BRI Port.
If the Point-to-Point is selected as the Interface Type, the TEI Negotiation will be set as Fixed.
You must also configure TEI Negotiation Value (for Fixed Mode). TEI Value range is from 00 to
63. By default, it is 00. TEI Value of BRI (TE) port of ETERNITY should match with the TEI Value
expected by the NT equipment at other end.
If Point-to-Multipoint is selected as the Interface Type, the TEI Negotiation will be set as Auto.
When you change the TEI mode on any port, the BRI card will reset.
You may configure the port to Treat Incoming call as Trunk or Station.
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The calls
will be routed as per the Trunk Feature Template assigned to the BRI Port.
If you select Station, you must also assign a Station Basic Feature Template and Station Advanced
Feature Template to the BRI Port.
When you select Station, the system will treat the calling party as an extension user. The user will have
access to all the features and facilities of the system, as per the Station Basic Feature Template and
Station Advanced Feature Template assigned to the BRI Port.
By default, Trunk is selected.
If Point-to-Point is selected as the Interface Type, you can select the option Trunk or Station for the
parameter Treat Incoming call as.
If Point-to-Multipoint is selected as the Interface Type, only Station can be set as the option for the
parameter Treat Incoming call as.
If Station is selected as the option for Treat Incoming call as, the user will only be able to:
Dial Flexible Numbers
Dial Operator Code
Dial Trunk Access Code for making outgoing calls
Access the Global Directory
Make calls within the Closed User Group
931
Different countries use specific type of ISDN switch. The type of switch determines various factors such as
how many ISDN devices would be handled, which B-channel will support voice, video, data etc. Select
ISDN Switch Variant supported by your country. You may select from the following options:
ATT_4ESS
ATT_5ESS
AUSTRALIA
AUTO CONFIG
DMS_100
ETSI_NET3
NTT_INS64
SWV_HONG_KONG
US_NI1
US_NI2
VN_X
By default, it is ETSI_NET3.
Configure Outgoing (OG) Reference ID for working hours, non-working hours and break hours. By
default, OG Reference ID is 00.
To know more about OG Reference ID, see OG Reference Table.
Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By default,
IC Reference ID is 00.
To know more about OG Reference ID, see IC Reference Table.
Assign Trunk Feature Template to the BRI Port. Trunk Feature Template is a set of general features that
define the behavior of a Trunk Port. By default, Template 01 is assigned to all BRI Ports.
For more details, see Trunk Feature Template.
Assign a Cost Factor to the BRI Port. By default, all the BRI Ports are assigned Cost Factor 01.
For more details, see Cost Factor.
Station Basic Feature Template assigned to the BRI Port is displayed in this field. Station Basic Feature
Template is a set of general features that define the basic behavior of a station. By default, Template 01 is
assigned to all BRI Ports.
For more details, see Station Basic Feature Template.
Station Advanced Feature Template assigned to the BRI Port is displayed in this field. Station Advanced
Feature Template is a set of advanced features, to be applied on extensions such as CLIP, Floor Service,
Walk-in Class of Service. By default, Template 01 is assigned to all BRI Ports.
For more details, see Station Advanced Feature Template.
932
If you have set Terminal as the Orientation Type, you must select the type of network with which the port
is to be Interfaced With. Select the type of network from the following:
Public ISDN
Private ISDN
Default: Public ISDN
If you want ETERNITY to display the called party number as the CLI for incoming calls, select the Display
Called Party Number as CLI check box. By default, Display Called Party Number as CLI option is
disabled.
This parameter is useful when a single BRI line connection and Operator are shared by more than one
organization. To enable the Operator to handle calls more efficiently, you must enable Display Called Party
Number as CLI and configure the names and corresponding numbers of the organizations sharing line in
the Global Directory of ETERNITY.
When there is an incoming call, ETERNITY matches the number with the numbers in the Global Directory.
If a match is found ETERNITY displays the company name configured for that entry to the Operator.
You can configure the Display Called Party Number as CLI option only from Jeeves.
Select Layer 1 Mode depending upon the type of line terminated on BRI port of ETERNITY.
The Public ISDN provides different types of lines in different countries as mentioned below:
On Demand
Always ON
In 'On Demand' type of line, layer 1 (physical layer) remains 'down' when the line is idle. When the network
places incoming call, it activates the layer 1 and places the call. When the terminal makes out going call,
the layer 1 gets activated automatically.
In 'Always ON' type of line, layer 1 is always 'up', in normal condition. In this type of line, 'layer 1 down'
indicates fault condition and calls get failed.
By default, Layer 1 Mode is 'Always ON'.
When Layer 1 mode is set as 'Always ON', the ETERNITY uses this BRI to place call only if the layer 1
is 'up'. When layer 1 goes 'down', ETERNITY considers the line as un-healthy and will not use this BRI
as destination port. ETERNITY will place the call using the alternate port programmed in the same
routing group.
When Layer 1 mode is set as 'On Demand', ETERNITY will not check the Layer 1 condition while
placing call through this BRI.
The default Layer 1 Mode is 'Always ON'. Hence, if the interfaced line is of the type 'On Demand', the
calls will not get routed through BRI port, unless the 'Layer 1 Mode' is changed to 'On Demand'.
When the port is un-healthy, the ETERNITY routes the call using other healthy port. However, this
depends upon the member selection method and other ports programmed in the OG Trunk Bundle
Group (OGTBG). Refer chapter OG Trunk Bundle Group for more details.
Select Priority for the BRI Port. Priority is the precedence given to certain trunks and extensions over
others in being answered by the destination extension. You can select from 1 to 9. By default, Priority 5Normal is set for all BRI Ports.
For know more about Priority feature, see Priority.
Select Return Call to Original Caller (RCOC) check box to enable this feature on the BRI Port. By
default, RCOC flag is disabled.
933
For know more about RCOC feature, see RCOC (Return Call to Original Caller).
Set Overlap Receiving Timer for BRI Port. This timer is relevant while receiving the called party number
information in overlap receiving mode. By default, it is set to 15 seconds.
When the SETUP Message is sent by ETERNITY to the network (ISDN exchange), the exchange
responds by sending SETUP ACK (Acknowledgement), and dial tone is played to the caller. The time
taken by the exchange to respond to the SETUP message may vary from exchange to exchange. Set the
SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP message
and play dial tone to the caller.
Valid Range of the timer is 01 to 20 seconds. By default it is set to 4 seconds.
Change the default settings only if required. If the time you set is less than the time taken by the exchange
to respond, no dial tone will be played to the caller.
You can configure the SETUP Message Timer only from Jeeves.
Configure Idle Code for the BRI Port. Range of Idle Code is from 000 to 255. By default, it is 127 (7F).
The binary equivalent of the configured value (000 to 255) is sent on the channel to signify that the channel
is idle (or Used). This setting depends on the network. Most commonly applicable values are 7F and FF
(Binary equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
Select the number of Channels to be reserved for Data Communication. Channel count is from 0 to 2.
By default, number of channels reserved for Data Communication is 02.
Select the number of Channels to be reserved for Outgoing Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Outgoing Calls is 02.
Select the number of Channels to be reserved for Incoming Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Incoming Calls is 02.
Select the required option for sending the Caller- Type of Numbering Plan (TON) from the following:
Unknown
International
National
Network Specific
Subscriber
Abbreviated
Reserved
By default, Unknown is selected.
934
Select the required option for sending the Caller- Numbering Plan Identification (NPI) from the following:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
Select the required option for sending the Called-Type of Numbering Plan (TON) from the following:
Unknown
International
National
Network Specific
Subscriber
Abbreviated
Reserved
By default, Unknown is selected.
Select the required option for sending the Called-Numbering Plan Identification (NPI) from the following:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
By default, ISDN Numbering is selected.
Select the Bearer Service supported by your service provider. You can select from:
Speech
3.1 KHz Audio
By default, Speech is selected.
Configure Call Budget parameters for the BRI Ports. Call Budget is an expense control feature of
ETERNITY that allows you to keep track of the cost of phone calls made from the BRI Port.
Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the BRI Port.
By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By default
the number of calls is set to 9999.
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan suggests
so.
935
Call Back: This parameter is related to the Call Back on Trunk Port feature. If you want to enable the 'Call
Back on Trunk Port' feature on this BRI Port, configure the following parameters:
Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this flag
is disabled on all trunk port types. By default, the flag is disabled.
Call Back Timer (sec): This is the duration for which the system waits for the caller to disconnect
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to
10 seconds.
Call Back Mode: Select from the following options how a Call Back call answered by the remote party
should be routed:
Built-in Auto Attendant
PIN Authentication - Multiple Calls
CLI Authentication - Multiple Calls
CLI Authentication - Single Call - Answer Signaling
Operator
By default, Operator is selected as the Call Back Mode.
Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the CLI number. If you want the call back to be made to a different number, select Alternate
Number.
By default, CLI number is selected for Call Back.
Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all BRI Ports as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all BRI Ports and Trunk Port types, retain this
list.
If you want a different set of number strings to be programmed for this BRI Port, select a different
Number List, and assign it to the BRI Port.
You may program the Incoming Number List either from the Number List page or by clicking the
Incoming Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Outgoing Number List: Program the number strings that are to be called back in this List. For each
number string you programmed in the Incoming Number List, you must program in the corresponding
index in the Outgoing Number List a number to which the call back is to be made. For example, for the
number string programmed at Index 1 in the Incoming Number List, a corresponding number string at
the same Index, Index 1, should be programmed in the Outgoing Number List.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all BRI Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this BRI Port.
936
You may program the Outgoing Number List either from the Number List page or by clicking the
Outgoing Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an OG Trunk Bundle Group.
Select Same Port if you want the call back to be made using the same port on which the missed call is
received. If you select OGTB Group, the call back will be made using the OGTB Group, which you have
defined.
By default, Same Port is selected.
OGTB Group: If you selected OGTB Group for making the call back in the previous parameter, you
must define the OGTB Group that must be used in this parameter.
By default, OGTB Group 01 is assigned.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTB Group that you define here for Call Back.
Configure Pause Timer for the BRI Port. Range of Pause Timer is from 1 to 9 seconds. By default, it is set
to 3 seconds for all BRI Ports.
This Timer is required to insert delay between the digits while dialing out DTMF digits on the BRI port. One
of the applications for using this parameter is Multi-stage dialing. Refer chapter Multi-Stage Dialing.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the ETERNITY will
out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
Configure DTMF On Time for the BRI Port. Range of DTMF On Time is from 051 to 255 ms. By default, it
is set to 102 ms for all BRI Ports.
The DTMF On Time is the time for which the DTMF digit which is to be outdialed by the ETERNITY remain
On. One of the applications for using this parameter is Multi-stage dialing. Refer chapter Multi-Stage
Dialing.
Configure DTMF Inter Digit Pause Timer for the BRI Port. Range of Inter Digit Pause Timer is from 051 to
255 ms. By default, it is set to 102 ms for all BRI Ports.
Inter Digit Pause Timer is the time for which the system will wait while receiving the dialing digits to
consider it as end-of-dialing.
Select Category (Logical Partition) for the BRI Port. You may select from the following options:
CO
Leased
Private
By default, CO is selected for all BRI Ports.
937
Enter appropriate Debug Code (Level 1 to 4) to obtain debug information of various parts of BRI Card on
the COM Port. By default, debug is off for all BRI Ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000
to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for BRI Port are as specified below:
Level 1
Unused
Flow
NLS
Code
Meaning
001
Primitives
002
State
004
Variables
008
SVC Primitives
016
LAP
032
NLS
064
Flow Debug
000
Debug Off
LAP
SVC Primitives
Variables
State
Primitives
Level 2
Unused
Unused
Code
Meaning
004
Layer 4
016
DTMF Digit
000
Debug Off
Unused
DTMF Digit
Unused
Unused
HDLC D-Channel
Layer 4
Unused
Unused
Level 3
Unused
Code
938
Unused
Unused
Unused
Unused
Unused
Meaning
016
HDLC D-Channel
000
Debug Off
Level 4
Unused
Code
Unused
Unused
Unused
Unused
Unused
OS Task
NI
Meaning
001
002
OS Task
000
Debug Off
Example:
If '004' decimal value is entered as Debug Code for Level 1, then its binary equivalent '100' (0000100)
indicates that debug for "Variables" will get enabled.
If "007" decimal value is entered as a Debug Code for the Level 1, then its binary equivalent "111"
(00000111) indicates that debug for "Primitives", "States" and "Variables" will get enabled.
The TEI Negotiation mode will be set automatically as per the Interface Type selected for the BRI Port.
If the Point-to-Point is selected as the Interface Type, the TEI Negotiation will be set as Fixed.
You must also configure TEI Negotiation Value (for Fixed Mode). TEI Value range is from 00 to
63. By default, it is 00. TEI Value configured in the BRI (NT) should match with the TEI Value
configured in the Terminal Equipment connected with it.
If the Point-to-Multipoint is selected as the Interface Type, the TEI Negotiation will be set as Auto.
When you change the TEI mode on any port, the BRI card will reset.
You may configure the port to Treat Incoming call as Trunk or Station.
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The calls
will be routed as per the Trunk Feature Template assigned to the BRI Port.
If you select Station, you must also assign a Station Basic Feature Template and Station Advanced
Feature Template to the BRI Port.
When you select Station, the system will treat the calling party as an extension user. The user will have
access to all the features and facilities of the system, as per the Station Basic Feature Template and
Station Advanced Feature Template assigned to the BRI Port.
By default, Trunk is selected.
939
If Point-to-Point is selected as the Interface Type, you can select the option Trunk or Station for the
parameter Treat Incoming call as.
If Point-to-Multipoint is selected as the Interface Type, only Station can be set as the option for the
parameter Treat Incoming call as.
If Station is selected as the option for Treat Incoming call as, the user will only be able to:
Dial Flexible Numbers
Dial Operator Code
Dial Trunk Access Code for making outgoing calls
Access the Global Directory
Make calls within the Closed User Group
Different countries use specific type of ISDN switch. The type of switch determines various factors such as
how many ISDN devices would be handled, which B-channel will support voice, video, data etc. Select
ISDN Switch Variant supported by your country. You may select from the following options:
ATT_4ESS
ATT_5ESS
AUSTRALIA
AUTO CONFIG
DMS_100
ETSI_NET3
NTT_INS64
SWV_HONG_KONG
US_NI1
US_NI2
VN_X
By default, it is ETSI_NET3.
Configure Outgoing (OG) Reference ID for working hours, non-working hours and break hours. By
default, OG Reference ID is 00.
To know more about OG Reference ID, see OG Reference Table.
Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By default,
IC Reference ID is 00.
To know more about OG Reference ID, see IC Reference Table.
Assign Trunk Feature Template to the BRI Port. Trunk Feature Template is a set of general features that
define the behavior of a Trunk Port. By default, Template 01 is assigned to all BRI Ports.
For more details, see Trunk Feature Template.
Assign a Cost Factor to the BRI Port. By default, all the BRI Ports are assigned Cost Factor 01.
For more details, see Cost Factor.
940
Station Basic Feature Template assigned to the BRI Port is displayed in this field. Station Basic Feature
Template is a set of general features that define the basic behavior of a station. By default, Template 01 is
assigned to all BRI Ports.
Station Advanced Feature Template assigned to the BRI Port is displayed in this field. Station Advanced
Feature Template is a set of advanced features, to be applied on extensions such as CLIP, Floor Service,
Walk-in Class of Service. By default, Template 01 is assigned to all BRI Ports.
For more details, see Station Advanced Feature Template.
If you want ETERNITY to display the called party number as the CLI for incoming calls, select the Display
Called Party Number as CLI check box. By default, Display Called Party Number as CLI option is
disabled.
This parameter is useful when a single BRI line connection and Operator are shared by more than one
organization. To enable the Operator to handle calls more efficiently, you must enable Display Called Party
Number as CLI and configure the names and corresponding numbers of the organizations sharing line in
the Global Directory of ETERNITY.
When there is an incoming call, ETERNITY matches the number with the numbers in the Global Directory.
If a match is found ETERNITY displays the company name configured for that entry to the Operator.
You can configure the Display Called Party Number as CLI option only from Jeeves.
Select Layer 1 Mode depending upon the type of line terminated on BRI port of ETERNITY.
The Public ISDN provides different types of lines in different countries as mentioned below:
On Demand
Always ON
In 'On Demand' type of line, layer 1 (physical layer) remains 'down' when the line is idle. When the network
places incoming call, it activates the layer 1 and places the call. When the terminal makes out going call,
the layer 1 gets activated automatically.
In 'Always ON' type of line, layer 1 is always 'up', in normal condition. In this type of line, 'layer 1 down'
indicates fault condition and calls get failed.
By default, Layer 1 Mode is 'Always ON'.
When Layer 1 mode is set as 'Always ON', the ETERNITY uses this BRI to place call only if the layer 1
is 'up'. When layer 1 goes 'down', ETERNITY considers the line as un-healthy and will not use this BRI
as destination port. ETERNITY will place the call using the alternate port programmed in the same
routing group.
When Layer 1 mode is set as 'On Demand', ETERNITY will not check the Layer 1 condition while
placing call through this BRI.
The default Layer 1 Mode is 'Always ON'. Hence, if the interfaced line is of the type 'On Demand', the
calls will not get routed through BRI port, unless the 'Layer 1 Mode' is changed to 'On Demand'.
When the port is un-healthy, the ETERNITY routes the call using other healthy port. However, this
depends upon the member selection method and other ports programmed in the OG Trunk Bundle
Group (OGTBG). Refer chapter OG Trunk Bundle Group for more details.
941
Select Priority for the BRI Port. Priority is the precedence given to certain trunks and extensions over
others in being answered by the destination extension. You can select from 1 to 9. By default, Priority 5Normal is set for all BRI Ports.
For know more about Priority feature, see Priority.
Select Return Call to Original Caller (RCOC) check box to enable this feature on the BRI Port. By
default, RCOC flag is disabled.
For know more about RCOC feature, see RCOC (Return Call to Original Caller).
Set Overlap Receiving Timer for BRI Port. This timer is relevant while receiving the called party number
information in overlap receiving mode. By default, it is set to 15 seconds.
When the SETUP Message is sent by ETERNITY to the network (ISDN exchange), the exchange
responds by sending SETUP ACK (Acknowledgement), and dial tone is played to the caller. The time
taken by the exchange to respond to the SETUP message may vary from exchange to exchange. Set the
SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP message
and play dial tone to the caller.
Valid Range of the timer is 01 to 20 seconds. By default it is set to 4 seconds.
Change the default settings only if required. If the time you set is less than the time taken by the exchange
to respond, no dial tone will be played to the caller.
You can configure the SETUP Message Timer only from Jeeves
Configure Idle Code for the BRI Port. Range of Idle Code is from 000 to 255. By default, it is 127 (7F).
The binary equivalent of the configured value (000 to 255) is sent on the channel to signify that the channel
is idle (or Used). This setting depends on the network. Most commonly applicable values are 7F and FF
(Binary equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
Select the number of Channels to be reserved for Data Communication. Channel count is from 0 to 2.
By default, number of channels reserved for Data Communication is 02.
Select the number of Channels to be reserved for Outgoing Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Outgoing Calls is 02.
Select the number of Channels to be reserved for Incoming Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Incoming Calls is 02.
Select the required option for sending the Caller- Type of Numbering Plan (TON) from the following:
Unknown
International
National
Network Specific
Subscriber
Abbreviated
Reserved
By default, Unknown is selected.
942
Select the required option for sending the Caller- Numbering Plan Identification (NPI) from the following:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
By default, ISDN Numbering is selected.
Select the required option for sending the Called-Type of Numbering Plan (TON) from the following:
Unknown
International
National
Network Specific
Subscriber
Abbreviated
Reserved
By default, Unknown is selected.
Select the required option for sending the Called-Numbering Plan Identification (NPI) from the following:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
By default, ISDN Numbering is selected.
Select the Bearer Service supported by your service provider. You can select from:
Speech
3.1 KHz Audio
By default, Speech is selected.
Configure Call Budget parameters for the BRI Ports. Call Budget is an expense control feature of
ETERNITY that allows you to keep track of the cost of phone calls made from the BRI Port.
Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the BRI Port.
By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By default
the number of calls is set to 9999.
943
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan suggests
so.
Configure Call Back parameters for the BRI Ports. Call Back is used to respond to missed calls from
particular numbers on the BRI Port.
Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, the flag
is disabled.
Call Back Timer (sec): This is the duration for which the system waits for the caller to disconnect
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to
10 seconds.
Call Back Mode: From the following options select how a Call Back call answered by the remote party
should be routed:
Built-in Auto Attendant
PIN Authentication - Multiple Calls
CLI Authentication - Multiple Calls
CLI Authentication - Single Call - Answer Signaling
Operator
By default, Operator is selected as the Call Back Mode.
Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the CLI number. If you want the call back to be made to a different number, select Alternate
Number.
By default, CLI number is selected for Call Back.
Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all the BRI Ports as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all the BRI Ports and Trunk Port types, retain
this list.
If you want a different set of number strings to be programmed for this CO Trunk, select a different
Number List, and assign it to the CO trunk port.
You may program the Incoming Number List either from the Number List page or by clicking the
Incoming Number List link to reach the Number List page.
Refer the topic Number List to know more, and for configuration instructions.
944
Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the Incoming Number List, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
example, for the number string programmed at Index 1 in the Incoming Number List, a corresponding
number string at the same Index, Index 1, should be programmed in the Outgoing Number List.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all BRI Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this BRI port.
You may program the Outgoing Number List either from the Number List page or by clicking the
Outgoing Number List link to reach the Number List page.
Refer the topic Number List to know more, and for configuration instructions.
Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an OG Trunk Bundle Group.
Select Same port if you want the call back to be made using the same port on which the missed call is
received. If you select OGTBG, the call back will be made using the OGTBG, which you have defined.
By default, Same port is selected.
OGTB Group: If you selected OGTBG for making the call back in the previous parameter, you must
define the OGTBG that must be used in this parameter.
By default, OGTBG 01 is assigned.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTBG that you define here for Call Back.
Configure Pause Timer for the BRI Port. Range of Pause Timer is from 1 to 9 seconds. By default, it is set
to 3 seconds for all BRI Ports.
This Timer is required to insert delay between the digits while dialing out DTMF digits on the BRI port. One
of the applications for using this parameter is Multi-stage dialing. Refer chapter Multi-Stage Dialing.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the ETERNITY will
out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
Configure DTMF On Time for the BRI Port. Range of DTMF On Time is from 051 to 255 ms. By default, it
is set to 102 ms for all BRI Ports.
The DTMF On Time is the time for which the DTMF digit which is to be outdialed by the ETERNITY remain
On. One of the applications for using this parameter is Multi-stage dialing. Refer chapter Multi-Stage
Dialing.
Configure DTMF Inter Digit Pause Timer for the BRI Port. Range of Inter Digit Pause Timer is from 051 to
255 ms. By default, it is set to 102 ms for all BRI Ports.
Inter Digit Pause Timer is the time for which the system will wait while receiving the dialing digits to
consider it as end-of-dialing.
Select Category (Logical Partition) for the BRI Port. You may select from the following options:
1
945
2
3
Enter appropriate Debug Code (Level 1 to 4), to obtain debug information of various parts of BRI Card on
the COM Port. By default, debug is off for all BRI ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000
to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for BRI Port are as specified below:
Level 1
Unused
Flow
NLS
Code
Meaning
001
Primitives
002
State
004
Variables
008
SVC Primitives
016
LAP
032
NLS
064
Flow Debug
000
Debug Off
LAP
SVC Primitives
Variables
State
Primitives
Level 2
Unused
Unused
Code
Meaning
004
Layer 4
016
DTMF Digit
000
Debug Off
Unused
DTMF Digit
Unused
Unused
HDLC D-Channel
Layer 4
Unused
Unused
Level 3
Unused
946
Unused
Unused
Unused
Unused
Unused
Code
Meaning
016
HDLC D-Channel
000
Debug Off
Level 4
Unused
Code
Unused
Unused
Unused
Unused
Unused
OS Task
NI
Meaning
001
002
OS Task
000
Debug Off
Hardware Slot-Port
Use following command to assign hardware ID to a BRI software port:
1106-BRI-Slot-Port offset
Where,
BRI is from 01 to 32.
Slot is the number of the universal slot, where the BRI card is installed, from 01 to 16.
Port is the number of the BRI port on the card, from 00 to 99.
Use following command to de-assign the hardware slot and the hardware port assigned to the BRI software port.
1106-BRI-00-00
Name
Use following command to assign a name to the BRI port:
5405-1-BRI-Name
5405-2-BRI-BRI-Name
5405-*-Name
Where,
BRI is from 01 to 32.
Name is an alpha-numeric string of 12 characters. Terminate the command with #*.
By default, Name field is Blank.
947
Likewise, when the port is configured in NT mode is disabled, the IC call will not be allowed to land on this port. The
port will be treated as absent and accordingly other activities will be performed like routing the call to other stations
in the group, etc.
Orientation Type
Use following command to program Orientation Type for the BRI port:
6204-1-BRI-Orientation Type
6204-2-BRI-BRI-Orientation Type
6204-*-Orientation Type
Where,
Orientation Type
Meaning
Terminal
Network
Tie Line
By default Type = 1.
When Orientation=Terminal, the port will be regarded as trunk. All the trunk related parameters will be applied.
When Orientation=Network, the port will be regarded as station. All the station related parameters will be applied.
When Orientation = Tie-line, the port will be regarded as station for all incoming calls on it and as a trunk for all
outgoing calls made through it.
Power Feed
Use following command to Feed Power to the BRI port:
6227-1- BRI Port- Feed Power to Port
6227-2- BRI Port-BRI Port- Feed Power to Port
6227-*- Feed Power to Port
Where,
BRI Port is from 01 to 32
Feed Power to port is 0 for disable, 1 for enable.
Default it is disabled.
948
6226-2-BRI-BRI-Interface Type
6226-*-Interface Type
Where,
Interface Type is
1 for Point-to-Point
2 for Point-to-Multipoint
Default =Point to Point
When ISDN Phones are to be connected with BRI NT of the ETERNITY, this parameter should be set as 'Point to
Multipoint'.
When an ISDN PBX is to be interfaced with BRI-NT of the ETERNITY, this parameter should be set as 'Point to
Point'.
Meaning
01
ATT_4ESS
02
ATT_5ESS
03
AUSTRALIA
04
AUTO CONFIG
05
DMS_100
06
ETSI_NET3
07
NTT_INS64
08
SWV_HONG_KONG
09
US_NI1
10
US_NI2
11
VN_X
OG Reference ID-WH
Use following command to program an OG Reference ID to a BRI port:
6231-1-BRI-OG Reference ID
6231-2-BRI-BRI-OG Reference ID
6231-*-OG Reference ID
Where,
BRI is from 01 to 32.
OG Reference ID is from 00 or 01 to 99.
By default, OG Reference ID for BRI is 00.
949
OG Reference ID-BH
Use following command to program an OG Reference ID to a BRI port:
6241-1-BRI-OG Reference ID
6241-2-BRI-BRI-OG Reference ID
6241-*-OG Reference ID
Where,
BRI is from 01 to 32.
OG Reference ID is from 00 or 01 to 99.
By default, OG Reference ID for BRI is 00.
OG Reference ID-NH
Use following command to program an OG Reference ID to a BRI port:
6242-1-BRI-OG Reference ID
6242-2-BRI-BRI-OG Reference ID
6242-*-OG Reference ID
Where,
BRI is from 01 to 32.
OG Reference ID is from 00 or 01 to 99.
By default, OG Reference ID for BRI is 00.
This command is significant only if the BRI port is configured in TE mode.
This command is significant only if the BRI port is configured in TE mode.
IC Reference ID-WH
Use following command to program an IC Reference ID-WH on the BRI port:
6232-1-BRI-IC Reference ID
6232-2-BRI-BRI-IC Reference ID
6232-*-IC Reference ID
Where,
BRI is from 01 to 32.
IC Reference ID is from 01 to 99.
By default, IC Reference ID-WH for BRI is 00.
This command is significant only if the BRI port is configured in TE mode.
IC Reference ID-BH
Use following command to program an IC Reference ID-BH on the BRI port:
6233-1-BRI-IC Reference ID
6233-2-BRI-BRI-IC Reference ID
6233-*-IC Reference ID
Where,
BRI is from 01 to 32.
IC Reference ID is from 00 or 01 to 99.
By default, IC Reference ID-BH for BRI is 00.
This command is significant only if the BRI port is configured in TE mode.
950
IC Reference ID-NH
Use following command to program an IC Reference ID-NH on the BRI port:
6234-1-BRI-IC Reference ID
6234-2-BRI-BRI-IC Reference ID
6234-*-IC Reference ID
Where,
BRI is from 01 to 32.
IC Reference ID is from 00 or 01 to 99.
By default, IC Reference ID-NH for BRI is 00.
This command is significant only if the BRI port is configured in TE mode.
Cost Factor
Use following command to assign a Cost Factor to the BRI port:
6202-1-BRI-SP
6202-2-BRI-BRI-SP
6202-*-SP
Where,
BRI is from 01 to 32.
Cost Factory is from 01 to 99.
By default, all the BRI ports are assigned Cost Factor = 01.
This parameter is insignificant when BRI port is configured in NT mode.
951
Where,
BRI is from 01 to 32.
Template is the number of the Station Advanced Feature Template, from 01 to 50.
Default: Template 01 is assigned to all BRI ports.
Layer 1 Mode
This parameter is applicable for BRI port, when orientation type is 'Terminal'.
The Public ISDN provides different types of lines in different countries as mentioned below:
On Demand
Always ON
In 'On Demand' type of line, layer 1 (physical layer) remains 'down' when the line is idle.
When the network places incoming call, it activates the layer 1 and places the call.
When the terminal makes out going call, the layer 1 gets activated automatically.
In this type of line, 'layer 1 down' indicates fault condition and calls get failed.
Meaning
Always ON
On Demand
Select the 'Layer 1 Mode' parameter, depending upon the type of line, terminated on BRI port of ETERNITY.
By default, Layer 1 Mode is 'Always ON'.
TEI Negotiation
SE can select the Automatic or Fixed TEI negotiation for each BRI port as required. If Fixed TEI negotiation is
selected, the value of fixed TEI negotiation is required to be programmed.
Use following command to select TEI Negotiation on a BRI port:
6238-1-BRI-TEI Negotiation Mode
6238-2-BRI-BRI-TEI Negotiation Mode
6238-*-TEI Negotiation Mode
Where,
BRI is from 00 to 32.
TEI Negotiation mode
Meaning
Automatic (Non-fixed)
Fixed
952
When you change the TEI mode on any port, the BRI card will get reset.
If you have selected 'Fixed' mode, program the value as per the value of port connected at remote end as
explained below:
TEI Value programmed in the BRI (NT) should match with the TEI value programmed in the Terminal
equipment connected with it.
TEI value of BRI (TE) port of ETERNITY should match with the TEI value expected by the NT
equipment at other end.
Priority
This command is applicable when BRI port is configured to act as a station.
Use following command to assign priority to BRI:
3916-1-BRI-Priority
3916-2-BRI-BRI-Priority
3916-*-Priority
Where,
BRI is from 01 to 32
Priority is from 1 to 9
By default, Priority for BRI is 5-Normal.
953
Where,
BRI is from 01 to 32
Timer is from 00 to 99 seconds.
Default: 15 seconds.
This timer is relevant while receiving the called party number information in overlap receiving mode. It is not
relevant for overlap sending mode.
Idle Code
Use following command to program the Idle Code for the BRI port:
6207-1-BRI-Idle Code
6207-2-BRI-BRI-Idle Code
6207-*-Idle Code
Where,
BRI is from 01 to 32.
Idle Code is from 000 to 255.
The binary equivalent of the programmed value (000 to 255) is sent on the channel to signify that the channel is
idle. (or Unused) This setting depends on the network. Most commonly applicable values are 7F and FF (Binary
equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
Default: Idle Code for the BRI port is 127 (7F).
By default, Channel is reserved for IC Calls for BRI is 2. Both the channels are used for receiving IC calls.
Meaning
Unknown
International Number
National Number
Subscriber Number
Abbreviated Number
Reserved Number
Meaning
Unknown
ISDN Numbering
Date Numbering
Telex Numbering
National Numbering
Private
Reserved
955
Where,
BRI is from 01 to 32.
Called Party TON
Meaning
Unknown
International Number
National Number
Subscriber Number
Abbreviated Number
Reserved Number
Meaning
Unknown
ISDN Numbering
Date Numbering
Telex Numbering
National Numbering
Private
Reserved
Call Budget
Refer the topic Call Budget on Trunk for command strings.
Call Back
Use the following commands to program Call Back on BRI Trunk ports. To know more about this feature, refer the
topic Call Back on Trunk Ports.
To enable/disable Call Back on BRI port:
6242-1-BRI- Call Back Flag
6242-2-BRI-BRI-Call Back Flag
6242- * - Call Back Flag
Where,
956
BRI is from 01 to 32
Call Back Flag
Meaning
Disable
Enable
Meaning
Operator
Meaning
CLI Number
Alternate Number
957
Pause Timer
Use following command to program Pause Timer:
6209-1-BRI- Pause Timer
6209-2-BRI-BRI-Pause Timer
6209- * - Pause Timer
Where,
BRI is from 01 to 32
Pause Timer is from 1 to 9 seconds
By default, Pause Timer is 3 seconds.
This Timer is used to provide delay for the number dialing by the BRI port. Pause timer will be applicable when any
'P' digit is configured in the DTMF number string which is to be outdialed as DTMF digits on BRI port. One of the
applications for using this parameter is Multi-stage dialing. Refer chapter Multi-Stage Dialing.
958
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the ETERNITY will out dial
the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
959
Meaning
Disable
Enable
#4
#5
#6
#7
**
##
Debug Code
Debug information of various parts of the card can be obtained on the COM port of the card.
Use following command to get appropriate debug information for the BRI port:
6291-1-BRI-Level-Code
6291-2-BRI-BRI-Level-Code
6291-*-Level-Code
Where,
BRI is from 01 to 32.
Level is from 1 to 4 (As shown below).
Code is the value for the specified level to turn ON the debug for the parameters Code range = 000 to 255. Code
value '000' for each level will turn off that level's debug.
Level and Code for BRI port are as specified below:
960
Level 1
Unused
Flow
NLS
Code
Meaning
001
Primitives
002
State
004
Variables
008
SVC Primitives
016
LAP
032
NLS
064
Flow Debug
000
Debug Off
LAP
SVC Primitives
Variables
State
Primitives
Level 2
Unused
Unused
Code
Meaning
004
Layer 4
016
DTMF Digit
000
Debug Off
Unused
DTMF Digit
Unused
Unused
HDLC D-Channel
Layer 4
Unused
Unused
Level 3
Unused
Code
Unused
Unused
Unused
Unused
Unused
Meaning
016
HDLC D-Channel
000
Debug Off
Level 4
Unused
Code
Unused
Unused
Unused
Unused
Unused
OS Task
NI
Meaning
001
002
OS Task
000
Debug Off
By Default Debug = off for all BRI ports for all levels.
961
Relevant Topics:
1. Configuring PRI Trunks
1029
2. Multi-Stage Dialing
2054
3. Gateway Application-Answer Signaling
4. Logical Partition
2019
5. Call Budget on Trunk
1570
1964
962
Configuring E1 Trunks
Whats this?
Digital Signal Level 1 (T1E1) trunks use Bit-Oriented Signaling (BOS) and multiplexes 24 channels (T1 service) or
32 channels (E1 service) into a single data stream. T1E1 can be used for voice or voice-grade data and for datatransmission protocols. T1 trunk service multiplexes 24 channels into a single 1.544-Mbps data stream. E1 trunk
service multiplexes 32 channels into a single 2.048-Mbps stream. Both T1 and E1 provide a digital interface for
trunk groups.
Signaling Modes
Common Channel Signaling (CCS) is an industry-standard technique where any one of a group of channels carries
the signals for the other channels. Matrix uses the 24th channel of a group for signaling. This signaling technique
differs from 24-channel signaling. When the system is configured for Facility-Associated Signaling, 24-channel
signaling uses the 24th channel in a T1E1 facility to carry signals. This technique also is called clear channel, outof-band or alternate voice data (AVD) signaling.
Channel Associated Signaling (CAS) is similar to common-channel signaling and is used only when the Bit Rate is
2.048 Mbps (the trunk is used with an E1 interface). Signaling is carried on the 16th channel.
Common-channel signaling and channel associated signaling provide a maximum transmission rate of 64 Kbps for
bearer channels.
ROBBED-BIT signaling is a per-channel in-band signaling technique for transmitting signaling bits on each channel
in a T1E1 facility. The least-significant bit in every 6th transmitted information frame is removed and replaced by a
signaling bit. This technique is also called in-band signaling. The maximum transmission rate for each bearer
channel with ROBBED-BIT signaling is 56 Kbps.
ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps (T1) connection and on the 16th channel for a
2.048 Mbps (E1) connection.
QSIG is an ISDN based protocol for signaling between nodes of a Private Integrated Services network.
Any of the common trunks, except for PCOL (Personal Central Office Line) trunks, can be analog or digital. (PCOL
trunks can only be analog.) Administering a digital trunk group is very similar to administering its analog
counterpart, but digital trunks must connect to a T1E1 port and this port must be administered separately.
User interface for E1_PRI and E1_CAS channels:
In case of ISDN_E1_PRI and ISDN_E1_CAS the protocol supports 32 Channels ranging from 00 to 31,
out of which 2 channels (channel no. 00 and 16) are used for framing/signaling. So effectively user has 30
channels for OG/IC calls.
For better understanding of the user the channel IDs are mapped as shown below. Thus for the E1_PRI
and E1_CAS the T1E1 port supports total 30 channels ranging from 01 to 30, which you can use for
making and receiving calls.
Channel ID as
00
per Protocol
01
02
03
Channel ID for
01
User Interface
02
03
04
04
14
14
15
16
15
16
17
17
18
31
30
963
The system Debug is for trouble shooting and so the channel ID in debug will be as the actual channel ID
as supported by the E1_PRI and E1_CAS protocols.
Similarly, in case of T1 PRI, Protocol supports 24 channels (from 01 to 24), in which channel no. 24 is used
for the signaling, so effectively there are 23 Voice channels are available.
But in case of T1 RBS, Protocol supports 24 channels (from 01 to 24) and the protocol doesnt consume
any channel for signaling so that there are total 24 channels available for the users.
Select the T1E1 Port number you want to configure by clicking the respective tab, and program the
following port parameters:
ETERNITY will assign the Hardware Slot-Port automatically, when any card is inserted in the system.
Hardware slot is the number of the Universal slot of ETERNITY in which the T1E1 Card is inserted.
Range of slot number is 01-16. Port is the number of T1E1 hardware port on the card to which the
T1E1 line is connected. Range of Port is from 00-99.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields. By default, Hardware slotPort is 0000.
964
You may assign a Name to the T1E1 Port for identification of the port. The Name may consist of a
maximum of 12 characters. By default, it is blank.
Select the Carrier type as E1. The Carrier type will automatically be assigned to the port when you
select the region. You may it if required.
Select Signal Type. Signal Type signifies the type of signaling to be used on E1 line. The E1 signalling
supported are:
PRI
CAS
QSIG
E&M
Default: PRI
If you select PRI or QSIG as the Signal Type, configure the PRI/QSIG Parameters.
If you select CAS as the Signal Type, configure the CAS Parameters.
If you select E&M as the Signal Type, configure the E&M Parameters.
Set the Orientation Type. Select the orientation type from the following options:
Terminal
Network
Tie-Line
Default: Terminal
If you have set Terminal as the Orientation Type, you must select the type of network with which the
port is to be Interfaced With. Select the type of network from the following:
Public ISDN
Private ISDN
Default: Public ISDN
You may configure the port to Treat Incoming call as Trunk or Station.
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The
calls will be routed as per the Trunk Feature Template assigned to the T1E1 Port.
If you select Station, you must also assign a Station Basic Feature Template and Station Advanced
Feature Template to the T1E1 Port.
When you select Station, the system will treat the calling party as an extension user. The user will have
access to all the features and facilities of the system, as per the Station Basic Feature Template and
Station Advanced Feature Template assigned to the T1E1 Port.
By default, Trunk is selected.
965
If Point-to-Point is selected as the Interface Type, you can select the option Trunk or Station for the
parameter Treat Incoming call as.
If Point-to-Multipoint is selected as the Interface Type, only Station can be set as the option for the
parameter Treat Incoming call as.
If Station is selected as the option for Treat Incoming call as, the user will only be able to:
Dial Flexible Numbers
Dial Operator Code
Dial Trunk Access Code for making outgoing calls
Access the Global Directory
Make calls within the Closed User Group
Line coding is a pattern that data assumes as it is propagated over a communication channel. Select
the Line Coding Mechanism from the following:
AMI-Basic
HDB3
NRZ (Fiber Optic)
Default: HDB3.
Framing means to form a set of 24 or 32, 8 bits time slot that is to be treated as single transmission
unit. The Framing Modes supported by ETERNITY are:
CEPT1 MF (No CRC)
CEPT1 MF (Forced CRC)
CEPT1 MF (Auto CRC)
Default: CEPT1 MF (Auto CRC).
Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By
default, IC Reference ID is 00.
To know more about OG Reference ID, see IC Reference Table.
Assign Trunk Feature Template to the T1E1 Port. Trunk Feature Template is a set of general features
that define the behavior of a Trunk Port. By default, Template 01 is assigned to all T1E1 Ports.
For more details, see Trunk Feature Template.
Station Basic Feature Template assigned to the T1E1 Port is displayed in this field. Station Basic
Feature Template is a set of general features that define the basic behavior of a station. By default,
Template 01 is assigned to all T1E1 Ports.
For more details, see Station Basic Feature Template.
Station Advanced Feature Template assigned to the T1E1 Port is displayed in this field. Station
Advanced Feature Template is a set of advanced features, to be applied on extensions such as CLIP,
Floor Service, Walk-in Class of Service. By default, Template 01 is assigned to all T1E1 Ports.
For more details, see Station Advanced Feature Template.
966
Select Priority for the T1E1 Port. Priority is the precedence given to certain trunks and extensions over
others in being answered by the destination extension. You can select from 1 to 9. By default, Priority
5-Normal is set for all T1E1Ports.
For know more about Priority feature, see Priority.
Assign a Cost Factor to the T1E1 Port. By default, all the T1E1 Ports are assigned Cost Factor 01.
For more details, see Cost Factor.
ISDN glare occurs if the system initiates an outgoing call on a B-Channel at the same time the network
initiates an incoming call on that same B-channel. You may configure the Glare Option as Proceed or
Held Back. While processing a glare condition, the configured Glare Option on T1E1 port will be
considered.
Select Category (Logical Partition) for the T1E1 Port. You may select from the following options:
1
2
3
By default, 1 is selected for all T1E1 Ports.
Idle Code is the 8-bit sequence that occupies the time slot on a E1/T1 trunk channel when it is not
being used. By default, 127 is configured as the Idle Code.
If you want ETERNITY to display the called party number as the CLI for incoming calls, select the
Display Called Party Number as CLI check box. By default, Display Called Party Number as CLI
option is disabled.
This parameter is useful when a single T1E1 line connection and Operator are shared by more than
one organization. To enable the Operator to handle calls more efficiently, you must enable Display
Called Party Number as CLI and configure the names and corresponding numbers of the organizations
sharing line in the Global Directory of ETERNITY.
When there is an incoming call, ETERNITY matches the number with the numbers in the Global
Directory. If a match is found ETERNITY displays the company name configured for that entry to the
Operator.
You can configure the Display Called Party Number as CLI option only from Jeeves
Select Return Call to Original Caller (RCOC) flag to enable this feature on the T1E1 Port. By default,
RCOC flag is disabled.
For know more about RCOC feature, see RCOC (Return Call to Original Caller).
In Channel Reserved for Data Call, configure the channels you want to reserve for Data Calls. By
default, 00 channels are reserved.
In Channel Reserved for Outgoing Call, configure the channels you want to reserve for making
outgoing calls. By default, 30 channels are reserved.
In Channel Reserved for Incoming Call, configure the channels you want to reserve for receiving
incoming calls. By default, 30 channels are reserved.
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When a caller dials the trunk access code or selective trunk access code for dialing the number directly
on the trunk port, the caller waits for the dial tone before dialing the number. But some exchanges do
not give Dial Tone for the T1E1 Port. For Example, when T1E1 port as E1CAS type is used in Delhi, it
is observed that the exchange does not give dial tone when direct dialing on the trunk is used.
Enable the Feed Dial Tone flag. ETERNITY will provides the dial tone to the caller when the T1E1 Port
is accessed.
It is applicable only when Online dialing is used as, Store and Forward dialing, the dial tone is given to
the caller. The dial tone is played as per the Dial Tone Timer.
By default, Feed Dial Tone flag is disabled.
When dial tone flag is disabled, user will hear the dial tone of the exchange if provided, otherwise, user
will hear the silence.
If the user is making the call from the FXS port and dial tone is not provided by the exchange, user will
not know when to start dialing the number. In this case, it is possible that some digits are not out dialed
on the port and wrong number is dialed out because system will out dial the number only if Outgoing
call Acknowledge is received from the ETERNITY ME Card T1E1 and user will not know about this
condition. Hence, it is required to enable this flag, if exchange is not providing the dial tone.
When Online dialing or Store and Forward dialing are used, some exchanges do not provide any tone
while routing/processing the call. Thus, the caller does not know whether the call is being processed or
not as there is silence. In such case, You must enable the Feed Routing Tone flag, ETERNITY will
play the routing tone to the caller. ETERNITY stops playing the routing tone when an alert message or
connect message or disconnect message is received from the T1E1 Card.
By default, the Feed Routing Tone flag is disabled.
To customize the pulse width option and set the pulse shapes configure the Custom Pulse
parameters. ETERNITY generates pulse shapes which match the country standard, where it is
installed. However, if the standard pulse shape does not match, ETERNITY enables you to customize
the pulse width to match your exact requirements.
To use customize pulse width option and set the pulse shape in 1 to 4 phases, keep the T1/E1 Custom
Pulse Width (CPW) flag enabled.
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Word 1: Set the pulse width for setting pulse shape in the 1st phase. The range of Custom Pulse
Width-Word 1 is 001 to 127. Default: 109.
Word 2: Set the pulse width for setting pulse shape in the 2nd phase. The range of Custom Pulse
Width-Word 2 is 001 to 127. Default: 107.
Word 3: Set the pulse width for setting pulse shape in the 3rd phase. The range of Custom Pulse
Width-Word 3 is 001 to 127. Default: 64.
Word 4: Set the pulse width for setting pulse shape in the 4th phase. The range of Custom Pulse
Width-Word 4 is 001 to 127. Default: 64.
Select the Bearer Service supported by your service provider. You can select from:
Speech
3.1 KHz Audio
The Overlap Receiving Timer is relevant while receiving the called party number information in
overlap receiving mode. It is not relevant for the port in overlap sending mode.
Range of Overlap Receiving Timer is from 01 to 99 seconds. Default: 15 seconds.
Configure Pause Timer for theT1E1 Port. Range of Pause Timer is from 1 to 9 seconds. By default, it
is set to 3 seconds for all T1E1 Ports.
This Timer is required to insert delay between the digits while dialing out DTMF digits on the T1E1 port.
One of the applications for using this parameter is Multi-stage dialing. Refer chapter Multi-Stage
Dialing.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the ETERNITY
will out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
When the SETUP Message is sent by ETERNITY to the network (ISDN exchange), the exchange
responds by sending SETUP ACK (Acknowledgement), and dial tone is played to the caller. The time
taken by the exchange to respond to the SETUP message may vary from exchange to exchange. Set
the SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP
message and play dial tone to the caller.
Valid Range of the timer is 01 to 20 seconds. By default it is set to 4 seconds.
Change the default settings only if required. If the time you set is less than the time taken by the exchange
to respond, no dial tone will be played to the caller.
You can configure the SETUP Message Timer only from Jeeves.
Configure DTMF On Time for theT1E1 Port. Range of DTMF On Time is from 051 to 255 ms. By
default, it is set to 102 ms for all T1E1 Ports.
The DTMF On Time is the time for which the DTMF digit which is to be outdialed by the ETERNITY
remain On. One of the applications for using this parameter is Multi-stage dialing. Refer chapter MultiStage Dialing.
Configure DTMF Inter Digit Pause Timer for the T1E1 Port. Range of Inter Digit Pause Timer is from
051 to 255 ms. By default, it is set to 102 ms for all T1E1 Ports.
Inter Digit Pause Timer is the time for which the system will wait while receiving the dialing digits to
consider it as end-of-dialing.
Configure the Minimum ON Time (msec) for which the DTMF signal should be present in order to be
detected. The valid range of this time is 10 to 200 milliseconds. By default, Minimum ON Time is set to
20 milliseconds.
Configure the Minimum OFF Time (msec). This parameter signifies the minimum time period between
successive DTMF digits. The valid range of this time is 10 to 200 milliseconds. By default, Minimum
OFF Time is set to 20 milliseconds.
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Configure the Minimum Level (dB) for the DTMF digit to be considered as valid. The valid range of
this time is 0 to -36.5 dB. By default, Minimum levels is set to -36.5dB.
Make sure you have installed T1E1 Card with version V4R3.1 or higher, to ensure DTMF detection on the
T1E1 trunk.
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Configure Call Budget parameters for the T1E1 Ports. Call Budget is an expense control feature of
ETERNITY that allows you to keep track of the cost of phone calls made from the T1E1 Port.
Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the T1E1
Port. By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field.
By default the Amount is set to 999999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this
field. By default the number of minutes is set to 999999.
Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By
default the number of calls is set to 9999.
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to
be reset on a particular date of every month.
Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan
suggests so.
Call Back: This parameter is related to the Call Back on Trunk Port feature. If you want to enable the
'Call Back on Trunk Port' feature on this T1E1 Port, configure the following parameters:
Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this
flag is disabled on all trunk port types. By default, the flag is disabled.
Call Back Timer (sec): This is the duration for which the system waits for the caller to disconnect
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set
to 10 seconds.
Call Back Mode: Select from the following options how a Call Back call answered by the remote
party should be routed:
Built-in Auto Attendant
PIN Authentication - Multiple Calls
CLI Authentication - Multiple Calls
CLI Authentication - Single Call - Answer Signaling
Operator
By default, Operator is selected as the Call Back Mode.
Call Back on: This parameter allows you to select if the call back should be made to the same
number that was received or to a different number. If you want the call back to be made to the same
number select the CLI number. If you want the call back to be made to a different number, select
Alternate Number.
By default, CLI number is selected for Call Back.
Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all T1E1 Ports as well as all other Trunk port types. If you want
the same numbers strings to be programmed commonly for all T1E1 Ports and Trunk Port types,
retain this list.
If you want a different set of number strings to be programmed for this T1E1 Port, select a different
Number List, and assign it to the T1E1 Port.
You may program the Incoming Number List either from the Number List page or by clicking the
Incoming Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the Incoming Number List, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made.
For example, for the number string programmed at Index 1 in the Incoming Number List, a
corresponding number string at the same Index, Index 1, should be programmed in the Outgoing
Number List.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all T1E1 Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this T1E1 Port.
You may program the Outgoing Number List either from the Number List page or by clicking the
Outgoing Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Call Back from: This parameter determines the trunk port to be used to make the call back. The
call back can be made using the Same Port or an OG Trunk Bundle Group.
Select Same Port if you want the call back to be made using the same port on which the missed
call is received. If you select OGTB Group, the call back will be made using the OGTB Group, which
you have defined.
By default, Same Port is selected.
OGTB Group: If you selected OGTB Group for making the call back in the previous parameter, you
must define the OGTB Group that must be used in this parameter.
By default, OGTB Group 01 is assigned.
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If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTB Group that you define here for Call Back.
FDL is used for communicating general maintenance information or for transmitting user defined
information within the T1 link. General maintenance information is in the form of Performance Message
Report which is generated by the ETERNITY ME Card T1E1PRI and depending upon the FDL
Protocol, the Performance Message Report is sent every second or on request.
Select the FDL flag to enable. This parameter is applicable only if Framing = ESF. If the Network
(Public or Private) to which the ETERNITY is connected does not support FDL then FDL will be
disabled. By default, the T1 FDL is disabled.
If you have enabled FDL, configure the FDL Protocol. ETERNITY supports ANSI T1.403 and AT&T
54016 protocols of reporting the performance monitoring. By default, the T1 FDL Protocol is ANSI
T1.403.
For more information, see T1 Maintenance
Enter appropriate Debug Code (Level 1 to 4), to obtain debug information of various parts of T1E1
Card on the COM Port. By default, debug is off for all T1E1 ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from
000 to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for T1E1 Port are as specified below:
Level 1:
Unused
Unused
001
CAS
002
MFC R2
004
CAS DSP
008
Layer 4
000
Debug Off
Unused
Unused
Layer 4
Unused
HDLC (D-Channel)
CAS DSP
MFC R2
CAS
Level 2:
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Unused
Unused
001
Alarms
002
Counters
004
ABCD Bits
008
FDL
016
HDLC (D
Channel)
FDL
ABCD Bits
Counters
Alarms
000
Debug Off
Level 3:
Unused
Flow Debug
NLS Debug
001
Primitives
002
State
004
Variables
008
SVC Primitives
016
LAP Debug
032
NLS Debug
064
Flow Debug
000
Debug Off
LAP Debug
SVC Primitives
Variables
State
Primitives
Level 4:
Unused
Unused
001
OS Task
002
NI Debug
000
Debug Off
Unused
Unused
Unused
Unused
NI Debug
OS Task
973
PRI/QSIG Parameters
974
Under T1E1 Configuration, click PRI/QSIG Signalling and configure the PRI/QSIG parameters.
ISDN Switch Variant: ISDN supports a variety of service provider switches. Different countries use
specific type of ISDN switch. This switch is designed using ISDN standard protocol. The type of switch
determines various factors such as how many ISDN devices would be handled, which B-channel will
support voice, video, data etc. Select the ISDN Switch Variant from the list. By default, ETSI NET5 is
selected as the ISDN Switch Variant.
Offer continuous Bearer Channel Mapping(01-30): Select this check box for continuous Bearer
Channel Mapping for E1-QSIG/E1-PRI.
D-Channel: Enter the Channel number that is used for Data signaling. By default, D-Channel is 16.
Valid Range is 1 to 31.
Send Called Party Number Using: Select the appropriate option from the following for Send Called
Party Number Using:
Called Party Number IE (Information Element)
Keypad Facility IE (Information Element)
By default, Called Party Number IE is selected.
Dialing Type for Called Party Number: Select the type of dialing supported by your exchange. You
can select:
Enbloc
Digit -by - Digit
Any
By default, Any is selected as the Dialing Type for Called Party Number.
Caller - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the calling party:
Unknown
International
National
Network Specific
Subscriber
Abbreviated
Reserved
Default: Unknown.
Caller- Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the calling party:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
Default: ISDN Numbering.
Called - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the called party:
Unknown
International
National
Network Specific
Subscriber
Abbreviated
Reserved
Default: Unknown.
Called - Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the called party:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
Default: ISDN Numbering.
Receive Equalization Mode: You can set the Receive Equalization Mode as Auto or Manual. By default,
Auto is selected as the Receive Equalization Mode.
Receive Equalization Parameters: This field increases the strength of incoming signals by a fixed
amount to compensate for line losses. Select the required option from the list. By default, the receive
equalization parameters of T1E1 is 8dB.
Feed Inband Tones on T1E1-NT, before sending DISCONNECT: Select this flag, if you want to feed
inband tones on T1E1-NT before sending DISCONNECT message. This flag is applicable only when
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T1E1 port is configured as 'Network'. When this flag is enabled, inband tones shall be feed for 15 seconds
(fixed, non programmable) before sending DISCONNECT message.
When this flag is disabled, inband tones (Busy/Error as applicable for the state of the call) shall not be feed
before sending the DISCONNECT message. However when DISCONNECT message is sent from T1E1NT port, inband tones will always be sent with 'progress indicator 8'. By default, this flag is disabled.
E1 CAS Parameters
Under T1E1 Configuration, click E1 CAS Signaling and configure the CAS parameters.
E1 Line Signaling Variant: The E1 Line Signaling Variant supported by ETERNITY is ITU T Q.400Q.490.
E1 Register Signaling Variant: Select the appropriate option from the following for the E1 Register
Signaling Variant:
DTMF - DNIS/ANI is transmitted in the corresponding speech channel using the DTMF signals as per
ITU-T Q.23.
MFC R2 - DNIS/ANI is transmitted in the corresponding speech channel using the MFC R2 signals as
per ITU-T Q.400-Q490.
By default, E1 Register Signaling Variants is MFC R2.
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Forward Tone Maximum On Timer (T1) (Seconds): This timer signifies the maximum time for which the
forward signal remains ON, from the outbound end.
Matrix ETERNITY System Manual
The range of Forward Time Maximum ON Timer is from 01 to 99 seconds. Default: 15 seconds.
The DSP (Digital Signaling Processor) device will send the forward tone for this timer and will expect
backward signal within this timer. If no backward signal is received during this time, a timeout condition will
occur in which case, an alert signal will be sent to the ETERNITY ME Card Master, error tone be issued to
the calling party and a clear forward signal will be sent on the line.
Forward Tone Maximum Off Timer (T2) (Seconds): This timer signifies the maximum time between two
out going forward signals. During this time the forward tone will remain OFF. If the outbound end does not
send a forward signal for this time, the inbound end will interpret it as per its condition and shall take action
accordingly.
The range of Forward Time Maximum OFF Timer is from 01 to 99 seconds. Default: 24 seconds.
Maximum Compelled Cycle Timer (T3) (Seconds): This timer signifies the maximum time within which
one compelled signaling cycle shall end.
The range of Maximum Compelled Cycle Timer is from 01 to 99 seconds. Default: 15 seconds.
Pulse Duration for Pulse Signal (Milliseconds): Backward signals A-3, A-4, A-6 and A-15 are pulsed to
the outbound end. Pulse duration of these signals vary from country to country.
The range of Pulse Duration for Pulsed Signals is from 001 to 999 ms. Default: 150 ms.
It is recommended that tolerance be fixed at +/- 25 ms.
Pulsed Signal Maximum Wait Timer (Seconds): This timer signifies the time for which the outbound end
waits for the pulsed signal. If the pulsed signal is not received during this time, the compelling signaling is
said to be complete.
The range of Pulsed Signal Maximum Wait Timer is from 01 to 99 seconds. Default: 15 seconds.
First Forward Tone Wait Timer (Seconds)162: This timer signifies the time between receipt of line
seizure signal and the first forward signal.
The range of the First Forward Tone Wait Timer is from 08 to 24 seconds. Default:15 seconds.
Minimum MF Signal Persist Timer (Milliseconds): This timer signifies the minimum time for which the
forward/backward signal shall be sustained on the line by the receiving end.
The range of Minimum MF Signal Persist Timer is from 001 to 255 ms. Default: 20 ms.
Outbound Parameters
Dialed Number Identification Signal (DNIS) End Type: This parameter is applicable only when DNIS
length is set to 99 (that is, variable). The outbound end indicates end of DNIS using a group I tone or using
time out.
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Range of DNIS End Type is from 00, 11 to 15; where 00 indicates End of DNIS as time out. 11 to 15
indicates group I tone to declare End of DNIS. Default: 15.
Address Number Information (ANI) Send Position: This parameter signifies the number of DNIS digits
after which address number information is to be sent. Address number information is usually sent on
receiving the backward tone Send next digit or Send next ANI digit.
If send next address number information tone is received then this parameter is not applicable. But if same
tone is used by the inbound end to request the next ANI digit and the next DNIS digit, ANI is sent after the
number of digits as set in this field.
The range of ANI Send Position is from 00 to 99. Default: 00.
Is Address Number Information (ANI) Available: This parameter indicates Group A tone (received from
the inbound tone) that is to be interpreted as a question by the inbound end asking the outbound end
whether the outbound end has ANI digits to be sent.
The range of Is ANI Available, Group A tone is from 01 to 15. Default: 05.
Positive Response to Is ANI Available: This parameter signifies the Group 1 tone that the outbound end
will send to the inbound end as a response to Is ANI Available tone from the inbound end. The tone
defined in this parameter indicates the Group 1 tone with which the Outbound end will respond to the
inbound end to indicate that it has ANI digits to be sent.
The range of Positive Response to Is ANI Available, Group 1 tone is from 01 to 15. Default: 01.
Negative Response to Is ANI Available163: This parameter signifies the Group 1 tone that the outbound
end will send to the inbound end as a response to Is ANI Available tone from the inbound end. The tone
defined in this parameter indicates the Group 1 tone with which the Outbound end will respond to the
inbound end to indicate that it does not have ANI digits to be sent.
The range of Negative Response to Is ANI Available, Group 1 tone is from 01 to 15. Default: 10.
ANI End Tone Presentation Allowed: This parameter signifies the Group 1 tone used to signify end of
ANI digits with Presentation Allowed.
The range of End of ANI with Presentation Allowed, Group 1 tone is from 00, 11 to 15. Default:15.
ANI End Tone Presentation Restricted: This parameter signifies the Group 1 tone used to signify end of
ANI digits with Presentation Restricted.
The range of End of ANI with Presentation Restricted, Group 1 tone is from 00, 11 to 15. Default: 00.
Inbound Parameters
Dialed Number Identification Signal (DNIS) End Type: This parameter is applicable only when the DNIS
length is set to 99 (that is, variable). The outbound end indicates end of DNIS using a group I tone or using
time out.
The range of DNIS End Type is from 00, 11 to 15; where 00 indicates End of DNIS as time out, 11 to 15
indicates group I tone to declare End of DNIS. Default:15.
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Dialed Number Identification Signal (DNIS) Digit Length: This parameter signifies the number of DNIS
digits required by inbound end to indicate the Called party number during MFC R2 signaling.
The range of DNIS Length is from 01 to 99. Default: 99.
DNIS Digit length (01 to 98) will be expected by the inbound end. (Practical value would be 01 to 10)
DNIS Digit length 99 indicates DNIS length is variable. Further action is taken after timeout or on receipt of
I-15. Refer parameter 'DNIS End Type (Inbound)'.
Address Number Information (ANI) Request Position: The inbound end may or may not request ANI
digits. It may request ANI digits after receiving the first DNIS or after receiving second DNIS or even after
receiving all the DNIS digits.
The range of ANI Request Position is as follows:
ANI Request
Meaning
00
01-98
99
Default: 99.
Address Number Information (ANI) Length: This parameter signifies the number of ANI digits that would
be expected by the inbound side as Calling Party Number during MFC R2 signaling. This parameter at the
inbound side guides the inbound register to switch from requesting ANI digits back to requesting DID
digits.
The range of ANI Length is from 00 to 99. Default: 99.
Ask Address Number Information (ANI): This parameter specifies the backward group A tone used to
ask the outbound end whether it has ANI digits to be sent. This parameter is also known as Request ANI
Category.
The range of Ask ANI is from 00, 01 to 15. If no tone is sent by the inbound end, set this parameter to 00.
For India, this parameter is set to 04. Default: 05.
Positive Response to Ask ANI: This parameter specifies that the Group 1 forward tone is to be received
by the inbound end from the outbound which in turn indicates that outbound end has ANI digits to be sent.
This parameter is also known as ANI category.
The range of Positive Response to Ask ANI is from 01 to 15. Default: 01.
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For example, In India I-1 or I-10 is sent by the outbound end. In Kuwait, I-6 is sent. This parameter cannot
be zero because; Is ANI Available request will be made by the inbound end only if the country supports this
protocol.
Negative Response to Ask ANI: This parameter specifies the Group 1 forward tone to be received by the
inbound end from the outbound which would indicate that outbound end has ANI digits to be sent. This
parameter is also known as ANI category.
The range of Negative Response to Ask ANI is from 01 to 15. Default: 10.
For example, in India I-1 or I-10 is sent by the outbound end. In Kuwait, I-6 is sent. This parameter cannot
be zero because; Is ANI Available request will be made by the inbound end only if the country supports this
protocol.
ANI End Tone Presentation Allowed: This parameter specifies the Group I tone that the inbound end
should expect from the outbound end to consider End of ANI digits with information that the Presentation
of ANI by the outbound end is allowed.
The range of ANI End Tone Presentation Allowed is 00 or from 11 to 15. Default: 15.
If no tone is sent, set this parameter to 00. For India use A-4, for China use A-1.
ANI End Tone Presentation Restricted: This parameter specifies the group I tone that the inbound end
shall expect from the outbound end to consider End of ANI digits with an information that the Presentation
of ANI by the outbound end is Restricted.
The range of ANI End Tone Presentation Restricted is 00 or from 11 to 15. Default: 00.
If no tone is sent, set this parameter to 00. For India use A-4, for China use A-1.
Ask Calling Party Sub Category: This parameter specifies the group 1 tone that the inbound end shall
expect from the outbound end to consider End of ANI digits with an information that the Presentation of
ANI by the outbound end is Restricted. Select this flag to enable. Default: Disabled.
Forward Group II
Ordinary Subscriber: This parameter specifies the forward group II tone used to inform the inbound end
that the calling party is an Ordinary Subscriber. This signal is sent in response to Calling Party Category
signal Request from the inbound end.
Ordinary Subscriber is 00, 01 to 15. Default: 01. If this parameter is not applicable, assign 00.
Priority Subscriber: This parameter specifies the forward group II tone used to inform the inbound end
that the calling party is a Priority Subscriber. This signal is sent in response to Calling Party Category
signal Request from the inbound end.
Priority Subscriber is 00, 01 to 15. Default: 02. If this parameter is not applicable, assign 00.
Maintenance Equipment: This parameter specifies the forward group II tone used to inform the inbound
end that the calling party is Maintenance equipment.
Maintenance Equipment is 00, 01 to 15. Default: 03. If this parameter is not applicable, assign 00.
980
Operator: This parameter specifies the forward group II tone used to inform the inbound end that the
calling party is Operator.
Operator is from 00, 01 to 15. Default: 05. If this parameter is not applicable, assign 00.
Pay Phone: This parameter specifies the forward group II tone used to inform the inbound end that the
calling party is Pay Phone (Coin box).
Pay Phone is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign 00.
Data Transmission: This parameter specifies the forward group II tone used to inform the inbound end
that the call is a Data Call.
Data Transmission is from 00, 01 to 15. Default: 06. If this parameter is not applicable, assign 00.
Interception Operator: This parameter specifies the forward group II tone used to inform the inbound end
that the call is from Interception Operator.
Interception Operator is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign 00.
Backward Group A
Send Next Digit (N+1) (DNIS): This parameter specifies the backward group A tone used to request next
digit. Be it ANI digit or DNIS digit.
Send next Digit range is 00, 01 to 15. Default: 01. If you do not want to use any tone, assign 00.
For India, use A-1 to signify Send DNIS Digit event.
Send Last But One Digit (N-1) (DNIS): This parameter specifies the backward group A tone used to
request last but one digit that is, N-1 digit. Be it ANI digit or DNIS digit.
Send last but one digit range is 00, 01 to 15. Default: 02. If you do not want to use any tone, assign 00.
For India, use A-9 to signify Send last but one digit event.
Send Last But Two Digits (N-2) (DNIS): This parameter specifies the backward group A tone used to
request last but two digits that is, N-2 digit. Be it ANI digit or DNIS digit.
Send last but two digits range is 00, 01 to 15. Default: 07. If you do not want to use any tone, assign 00.
For India, use A-7 to signify Send last but two digits event.
Send Last But Three Digits (N-3) (DNIS): This parameter specifies the backward group A tone used to
request last but three digits that is, N-3 digit. Be it ANI digit or DNIS digit.
Send last but three digits range is 00, 01 to 15. Default: 08. If you do not want to use any tone, assign 00.
For India use A-8 to signify Send last but three digits event.
Send Caller Party Category and ANI Digit: It is to send calling partys category requests transmission of
a group II signal. Range is 00 to 15. By Default, it is 05.
981
Address Completed, Change over of Group B: This parameter specifies the backward group A tone
used to inform the inbound end that the incoming register at the inbound end needs no additional address
digit and is about to go over to transmission of a group B signal conveying the status of equipment at the
subscriber at the inbound end.
Address-Complete, Changeover to reception of Group B signal range is 00, 01 to 15. Default: 03. If you do
not want to use any tone, assign 00.
Send Calling Party Category and Change to Group C: This parameter specifies the backward group A
tone used by the inbound end to request Calling Party Category from the outbound end. This tone also
informs the outbound end to change to reception of Group C signal.
Send Calling Party Category and Change to Group C range is from 00, 01 to 15. Default: 00. If you do not
want to use any tone, assign 00.
Congestion in the National Network: This parameter specifies the backward group A tone used to
inform the congestion at the inbound end.
Congestion in National Network range is 00, 01 to 15. Default: 04. If you do not want to use any tone,
assign 00.
Send Calling Party Category: This parameter specifies the backward group A tone used to request
calling party category.
Send calling party's category range is 00, 01 to 15. Default: 05. If you do not want to use any tone, assign
00.
For India, use A-7 to signify 'Send calling party's category' event.
Address Completed, Charge, Set Speech Condition: This parameter specifies the backward group A
tone used to inform the inbound end that the incoming register at the inbound end needs no additional
address digit, but will not send Group B signals. Also charge the call on answer.
The range of Address-Complete, Charge, Set-up Speech conditions is 00, 01 to 15. Default: 06. If you do
not want to use any tone, assign 00.
Repeat DNIS Digits from Beginning: This parameter specifies the backward group A tone used to inform
the outbound end to send all the DNIS digits from the beginning.
The range of Repeat DNIS digits from beginning is 00, 01 to 15. Default: 00. If you do not want to use any
tone, assign 00.
Send Next ANI Digits: This parameter specifies the backward group A tone used to request next (first)
ANI digit.
The range of Send Next ANI Digit is 00 or 01 to 15.Default: 00. If no such tone is sent, set this parameter
to 00.
A few countries use different tone to request next ANI digit and next DNIS digits. For example, India uses
A-4, China uses A-1.
982
Backward Group B
Send Special Information Tone164: This parameter specifies the backward group B tone used to inform
the outbound end that the call cannot be made through because of reasons beyond those which are
considered by the Protocol. Hence Special Information tone will be sent to the calling party. ETERNITY will
send only the Group B signal and then disconnect the call.
The range of Send Special Information Tone is from 00, 01 to 15. Default: 02. If you do not want to use any
tone, assign 00.
Send Special Information Tone and Setup Speech Conditions: This parameter specifies the backward
group B tone used to inform the outbound end that the call cannot be made through because of reasons
beyond those which are considered by the Protocol. Hence, Special information tone will be sent to the
calling party and request the outbound end to setup speech conditions.
In this case, ETERNITY shall connect the calling party to the voice message of the system informing the
caller that the call cannot be connected.
The range of Send Special Information Tone and setup speech conditions is from 00, 01 to 15. Default: 02.
If you do not want to use any tone, assign 00.
Subscriber Line Busy: This parameter specifies the backward group B tone used to inform the outbound
end that the called subscriber is busy.
Subscriber Line busy range is from 00, 01 to 15. Default: 03. If you do not want to use any tone, assign 00.
Subscriber Line Free, Charge: This parameter specifies the backward group B tone used to inform the
outbound end that the called subscriber is free and the call is to be charged on answer.
The range of Subscriber Line free, Charge is from 00, 01 to 15. Default: 06. If you do not want to use any
tone, assign 00.
Subscriber Line Free, No charge: This parameter specifies the backward group B tone used to inform
the outbound end that the called subscriber is free, but the call is not to be charged on answer. This signal
permits non-chargeable calls without the need for transferring no charge information by line signals.
The range of Subscriber Line free, NO Charge is 00, 01 to 15. Default: 07. If you do not want to use any
tone, assign 00.
Congestion: This parameter specifies the backward group A tone used to inform that congestion is
encountered after changeover from Group-A to Group-B signals.
Congestion range is from 00, 01 to 15. Default: 04. If you do not want to use any tone, assign 00.
Unallocated Number: This parameter specifies the backward group B tone used to inform the outbound
end that the number received is not in use.
The range of Unallocated Number is from 00, 01 to 15. Default: 05. If you do not want to use any tone,
assign 00.
983
Subscriber Line Out of Order: This parameter specifies the backward group B tone used to inform the
outbound end that the called subscriber's line is out of order.
Range of Subscriber's Line out of order is from 00, 01 to 15. Default: 08. If you do not want to use any
tone, assign 00.
Call Rejected, No Indication165: This parameter specifies the Group B backward tone used to inform the
outbound end that the call is rejected but there is no indication of cause.
Call rejected, No indication range is from 00, 01 to 15. Default: 00. If this parameter is not applicable,
assign 00.
Alternative Answer Tone166: This parameter specifies the Group B backward tone used to inform the
outbound end that the call is accepted and the speech path is made through.
Alternative Answer Tone range is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign
00.
Changed Number167: This parameter specifies the Group B backward tone used to inform the outbound
end that the number dialed by the calling party is changed. However, this parameter is rarely used.
The range of Changed Number (announcement on line) is from 00, 01 to 15. Default: 00. If this parameter
is not applicable, assign 00.
Backward Group C
Send Next ANI Digit: This parameter specifies the backward group C tone to request next (even first) ANI
digit from the outbound end.
The range of Send next ANI digit (Group C) is from 00, 01 to 15. Default: 00. If this parameter is not
applicable, assign 00.
Request Transition Back to Group A and Restart from First DNIS (Group C): This parameter specifies
the backward group C tone to restart from the first DNIS and request transition to Group A.
The range of Request transition to Group A and restart from first DNIS is from 00, 01 to 15. Default: 00. If
this parameter is not applicable, assign 00.
Address Completed, Change to Reception of Group B: This parameter specifies the backward group C
tone used to signify Address completed, change to reception of Group B signal.
The range of Address completed, change to reception of Group B signal is from 00, 01 to 15. Default: 00. If
this parameter is not applicable, assign 00.
Congestion168: This parameter specifies the backward group C tone used to signify Congestion.
The range of Congestion is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign 00.
165.
166.
167.
168.
984
Request Transition Back to Group A and Sent Next DNIS: This parameter specifies the backward
group C tone used to signify request transition back to group A, and send next DNIS.
The range of Request transition back to group A, and send next DNIS is from 00, 01 to 15. Default: 00. If
this parameter is not applicable, assign 00.
Request Transition Back to Group A and Repeat the Last DNIS: This parameter specifies the
backward group C tone used to signify request transition back to group A, and repeat the last DNIS.
The range of Request transition back to group A, and repeat the last DNIS is from 00, 01 to 15. Default: 00.
If this parameter is not applicable, assign 00.
C & D Bits: This parameter indicates the default values of C and D bits when the T1/E1 Port transmits line
signals.
CD Bits
00 (C=0, D=0)
01
10
11
Invert Bit A Flag: This parameter signifies whether A-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit A.
Default: Disabled (Do Not Invert Bit A).
Invert Bit B Flag: This parameter signifies whether B-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit B.
Default: Disabled (Do Not Invert Bit B1)
Invert Bit C Flag: This parameter signifies whether C-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit C.
Default: Disabled (Do Not Invert Bit C).
Invert Bit D Flag: This parameter signifies whether D-bit is to be inverted before transmitting and on
receiving. Select the check box to enable, that is to Invert Bit D.
Default: Disabled (Do Not Invert Bit D).
E1 Metering Bit: This parameter signifies the bit used by the network to signal metering pulses.
You can select from the following options:
None
Bit-A
985
Bit-B
Bit-C
Bit-D
Default: Bit-A.
E1 Metering Pulse Minimum Timer (Milliseconds): This timer signifies the minimum time for which the
metering bit is changed, to be recognized as a genuine metering pulse subject to E1 Metering Pulse
Minimum timer.
All Changes occurred for time less than this timer is ignored. The range of E1 Metering Pulse Minimum
timer is from 20ms to 1000ms. Default: 150ms.
Clear Back Signal: This parameter signifies the signal used to signify that the called party has
disconnected the line first. This is indicated in two ways: Release Guard (Ab =1) or Forced Release (Bb =
0). This parameter is country specific.
Default: Release Guard.
Release Timer (Milliseconds): This timer signifies the time for which the clear back signal should persist
on the line to be recognized as a genuine clear back signal. This is also known as Clear Back timer.
The range of Release Timer is from 20ms to 1000ms. Default: 400 ms.
Line Seizure Acknowledge Wait Timer (Milliseconds): This timer signifies the time for which the
outbound end waits for seizure acknowledgement from the inbound end after sending the line seizure
signal. On expiry of this timer, clear forward signal is sent by the outbound end. Alarm is to be generated.
This timer is applicable only when acting as outbound end.
The range of Line Seizure acknowledge Wait Timer is from 0001ms to 9999 ms. Default: 200ms.
Release Guard Timer (Milliseconds): This timer signifies the time for which inbound register waits before
declaring the channel idle (sending idle signal) when clear forward line signal is received from the
outbound end. This timer is applicable for Forced Release signal. This timer is applicable only when acting
as inbound end. This timer depends on the speed of switching and processing.
The range of Release Guard Timer is from 0000 ms to 9999 ms. Default: 200ms.
986
E&M Signalling
Under T1E1 Configuration, click E&M Signalling and configure the E&M parameters.
E&M Feature Template: Assign an E&M Feature Template to the T1E1 Port. The E&M Feature Template
is a set of features specific to E&M signaling, which define the behavior of the E&M ports, according to
their 'Orientation Type', whether they are functioning as Stations, Trunks or Tie-Lines. By default,
Template 01 is assigned to all T1E1 Ports.
For more details, see E&M Feature Template.
B Bit Pattern: Select the Bit Pattern from Same as Bit A or Fixed Value. By default, the Code is 1 (Same
as A bit).
B Bit Value: Configure the B bit value, the value can be 0 or 1. By default, B bit value is 0.
CD Bit Value: Configure the CD bit value, the valid range of the value is 1 to 3. By default, B bit value is 0.
Invert Bit A Flag: This parameter signifies whether A-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit A.
Default: Disabled (Do Not Invert Bit A).
Invert Bit B Flag: This parameter signifies whether B-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit B.
Default: Disabled (Do Not Invert Bit B1)
Invert Bit C Flag: This parameter signifies whether C-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit C.
Default: Disabled (Do Not Invert Bit C).
Invert Bit D Flag: This parameter signifies whether D-bit is to be inverted before transmitting and on
receiving. Select the check box to enable, that is to Invert Bit D.
Default: Disabled (Do Not Invert Bit D).
987
Port Parameters
T1E1-1
Hardware Slot-Port
Use following command to assign hardware ID to a T1E1 software port.
1107-T1E1-Slot-Port offset
Where,
T1E1 is from 1 to 8.
Slot is the number of the universal slot, where the T1E1 card is installed, from 01 to 16.
Port is the number of theT1E1 port on the card, from 01 to 32.
Use following command to de-assign the hardware slot and the hardware port assigned to the T1E1 software port.
1106-T1E1-00-00
Port Status
Used to enable/disable the port. When the Port is disabled, it will not be allotted to the user on grabbing the port.
Instead the user will get error tone.
Use the following command to enable/disable the port:
6101-1-T1E1-Port Status
6101-2-T1E1-T1E1-Port Status
6101-*-Port Status
Where,
T1E1 is from 1 to 8.
Port Status
Meaning
Disable
Enable
Name
Use the following command to assign a name to the port:
5407-1-T1E1-Name
5407-2-T1E1-T1E1-Name
5407-*-Name
Where,
T1E1 is from 1 to 8.
Name can be of upto 18 characters.
988
Carrier
Use the following command to select the carrier:
6108-1-T1E1-Carrier Type
6108-2-T1E1-T1E1-Carrier Type
6108-*-Carrier Type
Where,
T1E1 is from 1 to 8.
Carrier
Type
Meaning
E1
T1
Line type
Use following command to program signaling type/ Line type of a T1E1:
6105-1-T1E1-Line Type
6105-2-T1E1-T1E1-Line Type
6105-*-Line Type
Where,
T1E1 is from 1 to 8.
Line Type
Meaning
PRI
QSIG
CAS
E&M
DDI Routing is not supported on T1/E1 trunk line if you have selected E&M as the Signal Type.
Orientation Type
Use following command to program 'Orientation Type' for the T1E1 port:
6106-1-T1E1-Orientation Type
6106-2.T1E1-T1E1-Orientation Type
6106-*-Orientation Type
Where,
T1E1=1 To 8.
Orientation
Meaning
Terminal
Network
Tie Line
989
By default Type = 1.
When Orientation = Terminal, the port will be regarded as trunk. All the trunk related parameters will be applicable.
When Orientation = Network, the port will be regarded as station. All the station related parameters will be
applicable.
When Orientation = Tie-line, the port will be regarded as station for all IC calls to it and as trunk for all OG calls to
be made through it.
Meaning
AMI-Basic
HDB3
Framing Mode
Use following command to program the Framing Mode for the T1E1 port:
6104-1-T1E1-Framing
6104-2-T1E1-T1E1-Framing
6104-*-Framing
Where,
Framing
Meaning
DDI Routing
OG Reference ID
Use the following command to assign OG Reference ID to T1E1 port:
6131-1-T1E1-OG Reference ID
6131-2-T1E1-T1E1-OG Reference ID
6131-*-OG Reference ID
Where,
T1E1 is from 1 to 8.
OG Reference ID is from 00 to 99.
By default, OG Reference ID is 00.
990
Templates
Trunk Feature Template
Use the following command to assign a Trunk Feature Template to the T1E1 Trunks, dial:
5806-1-T1E1- Trunk Feature Template Number to assign a template to a single T1E1 port.
5806-2-T1E1- Trunk Feature Template Number to assign the same template to a range of T1E1 ports.
5806-*- Trunk Feature Template Number to assign the same template to all T1E1 ports.
Where,
T1E1is the Software Port number of the port from 1 to 8.
Template Number is the number of the customized Trunk Feature Template, from 01 to 50. Default: Trunk Feature
Template 01.
991
Others
Priority
To assign priority to T1E1
3914-1-T1E1-Priority to assign a template to a single T1E1 port.
3914-2-T1E1PRI-T1E1PRI-Priority to assign the same priority to a range of T1E1 ports.
3914-*-Template Number to assign the same Priority to all T1E1 ports.
Where,
T1E1PRI is the number of the T1E1PRI Software port, from 1 to 8.
Priority is from 1 to 9. Default: 5-Normal.
Cost Factor
Assign a Cost Factor to the T1E1 port. This is useful in LCR.
Use following command to assign a name to the T1E1 port:
6102-1-T1E1-SP
6102-2-T1E1-T1E1-SP
6102-*-SP
Where,
T1E1 is from 1 to 8.
SP is from 01 to 99.
By default, Service Provider is 01.
Glare Option
Use following command to program Glare Option for the T1E1 port:
6112-1-T1E1-Glare Option
6112-2-T1E1-T1E1-Glare Option
6112-*-Glare Option
Where,
T1E1 is from 1 to 8.
Glare Option
Meaning
Proceed
Held Back
992
Idle Code
Use the following command to program the Idle Code of a T1E1:
6113-1-T1E1-Idle Code
6113-2-T1E1-T1E1-Idle Code
6113-*-Idle Code
Where,
T1E1 is from 1 to 8.
Idle Code is from 000 to 255 (corresponding to 8 bits).
By default, the idle code is 127 (7F).
Use message mode of the Digital Switch IC to send the idle channel code.
RCOC
To enable RCOC on T1E1 Trunk, dial:
6145-1-T1E1-Code to enable the feature on a single trunk.
6145-2-T1E1-T1E1-Code to enable the feature on a range of trunks.
6145-*-Code to enable the feature on all trunks.
Where,
T1E1 is the software port number of the trunk from 1 to 8.
Code is
0 for Disable
1 for Enable
Default: Disable
Channels
Return Call to Original Caller (RCOC)
Use the following command to enable RCOC on T1E1 Trunk:
6145-1-T1E1-Code to enable the feature on a single trunk.
6145-2-T1E1-T1E1-Code to enable the feature on a range of trunks.
6145-*-Code to enable the feature on all trunks.
Where,
T1E1 is the software port number of the trunk from 1 to 8.
Code is
0 for Disable
993
1 for Enable
Default: Disable
Tone
Feed Dial Tone
Use following command to program the dial tone flag for T1E1 port:
6115-1-T1E1-Flag
6115-2-T1E1-T1E1-Flag
6115-*-Flag
Where,
T1E1 is from 1 to 8.
Flag
Meaning
Disable
Enable
994
By default, Dial Tone Flag is '0' for all the T1E1 ports.
Meaning
Disable
Enable
By default, Routing Tone Flag is '0' for all the T1E1 ports.
Custom Pulse
T1/E1 Custom Pulse Width (CPW)
Use following command to enable/disable Custom Pulse Width (CPW) Flag for the T1E1 port for T1 signaling:
6171-1-T1E1-Flag
6171-2-T1E1-T1E1-Flag
6171-*-Flag
Where,
T1E1 is from 1 to 8.
Flag
Meaning
Disable
Enable
995
Timer
Pause Timer
Use following command to program Pause Timer:
6109-1-T1E1- Pause Timer
6109-2-T1E1-T1E1-Pause Timer
6109-*-Pause Timer
Where,
T1E1 is from 1 to 8
Pause Timer is from 1 to 9 seconds
By default, Pause Timer is 3 seconds.
DTMF ON Time
Use following command to program DTMF ON Time:
6117-1-T1E1-DTMF ON Time
6117-2-T1E1-T1E1-DTMF ON Time
6117-*-DTMF ON Time
Where,
T1E1 is from 1 to 8.
DTMF ON Time is from 051 to 255 msec.
996
Gateway
Use Gateway Application - Answer Signaling?
Use following command to set flag for 'Gateway Application-Answer Signaling' on T1E1 trunk:
6119-1-T1E1-Gateway Application-Answer Signaling flag
6119-2-T1E1-T1E1-Gateway Application-Answer Signaling flag
6119-*- Gateway Application-Answer Signaling flag
Where,
T1E1 is from 1 to 8.
Flag
Meaning
Disable
Enable
#4
#5
#6
#7
**
997
Digit
##
Call Budget
Call Budget Type
To program Call Budget Type on T1E1 Port, dial:
6122-1-T1E1-Budget Type to program call budget type for a single trunk port.
6122-2-T1E1-T1E1-Budget Type to program the same call budget type for a range of trunk ports.
6122-*-Budget Type to program the same call budget type for all trunk ports.
Where,
T1E1is the number of the T1E1 software port from 1 to 8.
Budget Type is
0 for None
1 for Amount
2 for Minutes
3 for Number of Calls
By default, Budget Type is None.
Call Back
Use the following commands to program Call Back on T1E1 Trunk ports. To know more about this feature, refer the
topic Call Back on Trunk Ports.
To enable/disable Call Back on T1E1 port:
6176-1-T1E1- Call Back Flag
6176-2-T1E1-T1E1-Call Back Flag
6176- *-Call Back Flag
Where,
T1EI is from 1 to 8
Call Back Flag
Meaning
Disable
Enable
999
Meaning
Operator
Call Back On
To program Call Back On method for T1E1 port:
6179-1-T1E1-Call Back on selection
6179-2-T1E1-T1E1-Call Back on selection
6179-*-Call Back on selection
Where,
T1E1 is from 1 to 8
Call back on selection is
Call Back on
Meaning
CLI Number
Alternate Number
1000
OGTB Group
To assign a Call Back - OGTB Group for a T1E1 port:
6149-1-T1E1-OGTB Group
6149-2-T1E1-T1E1-OGTB Group
6149-*-OGTB Group
Where,
T1E1 is from 1 to 8
OGTB Group is from 01 to 32
By default, OGTBG is 01.
FDL
FDL Flag - copied from T1 Maintenance
T1 FDL can be enabled/disabled. This parameter is applicable only if Framing = ESF. If the Network (Public or
Private) to which the ETERNITY is connected does not support FDL then T1 FDL will be disabled.
Use following command to enable/disable T1 FDL on a T1E1PRI port:
6164-1-T1E1PRI-T1 FDL
6164-2-T1E1PRI-T1E1PRI-T1 FDL
6164-*-T1 FDL
Where,
T1E1PRI is from 1 to 8.
T1 FDL
Meaning
Disable
Enable
1001
FDL Protocol
ETERNITY will support both the protocols of reporting the performance monitoring. This parameter is applicable
only if T1 FDL is enabled and Framing = ESF. This parameter will match the protocol expected by the other end of
the link.
Use following command to program the T1 FDL Protocol for a T1E1PRI port:
6165-1-T1E1PRI-T1 FDL Protocol
6165-2-T1E1PRI-T1E1PRI-T1 FDL Protocol
6165-*-T1 FDL Protocol
Where,
T1E1PRI is from 1 to 8.
T1 FDL Protocol
Meaning
Disable
AT&T 54016
ANSI T1.403
Debug
ETERNITY supports debug of parameters (debug codes) depending on the Level of debug. On issuing this
command the ETERNITY ME Card T1E1 will send the debug details to the COM port of the T1E1 port.
Please note following command change as per Version of Software used. First set of Command '6191' is
used up to Version V6R0.12 and Commands at the end of 'Level 4' are used for Version V6R0.13
onwards. Here 'XXX' is the Code as mentioned in Tables for Level1 to level 4.
Till S/W V6R12
Debug Level-1
6191-1-T1E1-1-XXX
6191-1-T1E1-1-XXX
Debug Level-2
6191-1-T1E1-2-XXX
6191-1-T1E1-2-XXX
Debug Level-3
6191-1-T1E1-3-XXX
6192-1-T1E1-1-XXX
Debug Level-4
6191-1-T1E1-4-XXX
6192-1-T1E1-2-XXX
Use following command to start/stop debug the parameters for the T1E1 port:
6191-1-T1E1-Level-Debug Code
6191-2-T1E1-T1E1-Level-Debug Code
6191-*-Level-Debug Code
Where,
T1E1 Port is from 1 to 8.
Level is from 1 to 4 (As shown below).
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000 to 255.
Code value 000 for each level will turn off that levels debug.
1002
Level 1:
Unused
Unused
001
CAS
002
MFC R2
004
CAS DSP
008
Layer 4
000
Debug Off
Unused
Unused
Layer 4
Unused
HDLC (D-Channel)
CAS DSP
MFC R2
CAS
Level 2:
Unused
Unused
001
Alarms
002
Counters
004
ABCD Bits
008
FDL
016
HDLC (D
Channel)
000
Debug Off
FDL
ABCD Bits
Counters
Alarms
Level 3:
Unused
Flow Debug
NLS Debug
001
Primitives
002
State
004
Variables
008
SVC Primitives
016
LAP Debug
032
NLS Debug
064
Flow Debug
000
Debug Off
LAP Debug
SVC Primitives
Variables
State
Primitives
Level 4:
Unused
Unused
001
OS Task
002
NI Debug
Unused
Unused
Unused
Unused
NI Debug
OS Task
1003
000
Debug Off
Default: Debug Code = Debug OFF for all T1E1 ports for all levels.
Refer T1 RBS Parameters
Meaning
ATT 4ESS
ATT 5ESS
Australia
DMS
ETSI NET5
NTT INS64
SWV Hongkong
US NI12
QSIG E1
10
QSIG T1
1004
Meaning
Unknown: This is used when the user or network has no a prior information about
the numbering plan. In this case, the Address Value field is organized according to
the network dialing plan. For example, prefix or escape digits might be present.
International Number.
Abbreviated Number.
Reserved Number.
Meaning
Unknown
Telex Numbering
1005
Source NPI
Meaning
Meaning
Unknown: This is used when the user or network has no a prior information about
the numbering plan. In this case, the Address Value field is organized according to
the network dialing plan. For example, prefix or escape digits might be present.
International Number.
Abbreviated Number.
Reserved Number.
Meaning
Unknown
Telex Numbering
1006
Destination NPI
Meaning
Meaning
Manual
Auto
Meaning
None
8 dB
16 dB
24 dB
32 dB
40 dB
48 dB
1007
6130-2-T1E1-T1E1-Flag
6130-*-Flag
Where,
Flag
Meaning
No
Yes
Default = NO.
E1 CAS Signaling
T1E1-1
E1 Line Signaling Variant
The following command is applicable when the Bit Rate=E1.
Use following command to program E1 Line Signaling Variant for the T1E1 port:
6152-1-T1E1-E1 Line Signaling Variant
6152-2-T1E1-T1E1-E1 Line Signaling Variant
6152-*-E1 Line Signaling Variant
Where,
T1E1 is from 1 to 8.
E1 Line Signaling Variant
01
Meaning
ITU T Q.400-Q.490
Meaning
DTMF
MFC R2
1008
DTMF: DNIS/ANI is transmitted in the corresponding speech channel using the DTMF signals as per ITUT Q.23.
MFC R2: DNIS/ANI is transmitted in the corresponding speech channel using the MFC R2 signals as per
ITU-T Q.400-Q490.
1009
T1E1 is from 1 to 8.
Pulsed Signal Maximum Wait Timer is from 01 to 99 seconds.
By default, Pulsed Signal Maximum Wait Timer is 15 secs.
Outbound Parameters
Dialed Number Identification Signal (DNIS) END Type
Use following command to program to set DNIS END Type (outbound) for T1E1 port:
7108-1-T1E1-End of DNIS
7108-2-T1E1-T1E1-End of DNIS
7108-*-End of DNIS
Where,
T1E1 is from 1 to 8.
End of DNIS is from 00, 11 to 15.
00 indicates End of DNIS as time out.
01 to 15 indicates group 1 tone to declare End of DNIS.
By default, DNIS End Type (Outbound) is 15.
1010
1011
Inbound Parameters
Dialed Number Identification Signal (DNIS) End Type
Use following command to program DNIS End Type:
7109-1-T1E1-DNIS End Type
7109-2-T1E1-T1E1-DNIS End Type
7109-*-DNIS End Type
Where,
T1E1 is from 1 to 8.
DNIS End Type is from 00, 11 to 15.
00 indicates End of DNIS as time out.
01 to 15 indicates group 1 tone to declare End of DNIS.
By Default, DNIS End Type (Inbound) is 15.
DNIS Digit length (01 to 98) will be expected by the inbound end. (Practical value would be 01 to 10)
DNIS Digit length 99 indicates DNIS length is variable. Further action is taken after timeout or on receipt of
I-15. Refer parameter 'DNIS End Type (Inbound)'.
ANI Length = 99 indicates ANI Length is variable. If ANI length is variable, the logic waits for End of ANI
from the outbound side. The inbound end will sense for I-12 and I-15. I-12 is used to signify that no ANI
digits are available whereas I-15 is used to signify end of ANI digits. Some countries like China use I-15 to
signify both the events viz. End of ANI and no ANI digits available.
This cannot be zero. This is because; Is ANI Available request would be made by the inbound end only if
the country supports this protocol. In such event, Is ANI Available request will be responded to.
For example, In India I-1 or I-10 is sent by the Outbound end. In Kuwait, I-6 is sent.
1013
ANI End Tone Presentation Allowed is 00 or 11 to 15. If no such tone is sent, set this parameter to 00. For
For example, India uses A-4, China uses A-1, etc.
Forward Group B
Ordinary Subscriber
Use following command to program the Ordinary Subscriber:
7124-1-T1E1-Ordinary Subscriber
7124-2-T1E1-T1E1-Ordinary Subscriber
7124-*-Ordinary Subscriber
Where,
T1E1 is from 1 to 8.
Ordinary Subscriber is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Ordinary Subscriber is 01.
Priority Subscriber
Use following command to program the Priority Subscriber:
7125-1-T1E1-Priority Subscriber
7125-2-T1E1-T1E1-Priority Subscriber
7125-*-Priority Subscriber
Where,
T1E1 is from 1 to 8.
Priority Subscriber is from 00, 01 to 15. Use '00' when this parameter is not applicable.
1014
Maintenance Equipment
Use following command to program the Maintenance Equipment:
7126-1-T1E1-Maintenance Equipment
7126-2-T1E1-T1E1-Maintenance Equipment
7126-*-Maintenance Equipment
Where,
T1E1 is from 1 to 8.
Maintenance Equipment is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Maintenance Equipment is 03.
Operator
Use following command to program the Operator:
7127-1-T1E1-Operator
7127-2-T1E1-T1E1-Operator
7127-*-Operator
Where,
T1E1 is from 1 to 8.
Operator is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Operator is 05.
Pay Phone
Use following command to program the Pay Phone:
7128-1-T1E1-Pay Phone
7128-2-T1E1-T1E1-Pay Phone
7128-*-Pay Phone
Where,
T1E1 is from 1 to 8.
Operator is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Pay Phone is 00.
Data Transmission
Use following command to program the Data Transmission:
7129-1-T1E1-Data Transmission
7129-2-T1E1-T1E1-Data Transmission
7129-*-Data Transmission
Where,
T1E1 is from 1 to 8.
Data Transmission is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Data Transmission is 06.
Interception Operator
Use following command to program the Interception Operator:
7130-1-T1E1-Interception Operator
7130-2-T1E1-T1E1-Interception Operator
7130-*-Interception Operator
Where,
T1E1 is from 1 to 8.
Interception Operator is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Interception Operator is 00.
1015
Backward Group A
Send next Digit (N+1) (DNIS)
Use following command to program the Send next Digit:
7131-1-T1E1-Send Next Digit
7131-2-T1E1-T1E1-Send Next Digit
7131-*-Send Next Digit
Where,
T1E1 is from 1 to 8.
Send next Digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Next Digit is 01.
1016
T1E1 is from 1 to 8.
Send Caller Party Category and ANI Digit is 00 to 15.
By default, Send Last But Three Digit is 08.
1017
Address-Complete, Charge, Set-up Speech Conditions is 00, 01 to 15. 00 is used for No tone.
By default, Address-Complete, Charge Set-up Speech Conditions is 06.
Few countries use different tone to request next ANI digit and next DNIS digits.
Backward Group B
Send Special Information Tone
Use following command to program the Send Special Information Tone:
7143-1-T1E1-Send Special Information Tone
7143-2-T1E1-T1E1-Send Special Information Tone
7143-*-Send Special Information Tone
Where,
T1E1 is from 1 to 8.
Send Special Information Tone is from 00, 01 to 15. 00 is used for No tone.
By default, Send Special Information Tone is 02.
1018
Congestion
Use following command to program Congestion:
7148-1-T1E1-Congestion
7148-2-T1E1-T1E1-Congestion
7148-*-Congestion
Where,
T1E1 is from 1 to 8.
Congestion is from 00, 01 to 15. 00 is used for No tone.
By default, Congestion is 04.
Unallocated Number
Use following command to program the Unallocated Number:
7149-1-T1E1-Unallocated Number
7149-2-T1E1-T1E1-Unallocated Number
7149-*-Unallocated Number
Where,
T1E1 is from 1 to 8.
Unallocated Number is from 00, 01 to 15. 00 is used for No tone.
By default, Unallocated Number is 05.
1019
Changed Number
Use following command to program the Changed Number (announcement on line):
7153-1-T1E1-Changed Number
7153-2-T1E1-T1E1-Changed Number
7153-*-Changed Number
Where,
T1E1 is from 1 to 8.
Changed Number (announcement on line) is from 00, 01-15. Use '00' when this parameter is not applicable.
By default, Changed Number (announcement on line) is 00.
Backward Group C
Send Next ANI digit
Use following command to program the Send next ANI digit:
7154-1-T1E1-Send Next ANI Digit
7154-2-T1E1-T1E1-Send Next ANI Digit
7154-*-Send Next ANI Digit
Where,
T1E1 is from 1 to 8.
Send next ANI digit (Group C) is from 00, 01 to 15. Use '00' when this parameter is not applicable.
This parameter is applicable in Mexico only.
1020
Congestion
This parameter specifies the backward group C tone used to signify Congestion
Use following command to program the tone for Congestion:
7157-1-T1E1-Congestion
7157-2-T1E1-T1E1-Congestion
7157-*-Congestion
Where,
T1E1 is from 1 to 8.
Congestion is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Congestion is 00.
1021
00 (C=0, D=0)
01
10
11
By default, CD Bits is 1.
The C and D bits received during an IC call are ignored by the ETERNITY.
Invert Bit A
Use following command to program to invert/don't invert Bit A for the T1E1 port:
7162-1-T1E1-Invert Bit A
7162-2-T1E1-T1E1-Invert Bit A
7162-*-Invert Bit A
Where,
T1E1 is from 1 to 8.
Invert Bit A
Meaning
Disable
Enable
Invert Bit B
Use following command to program to invert/don't invert Bit B for the T1E1 port:
1022
7163-1-T1E1-Invert Bit B
7163-2-T1E1-T1E1-Invert Bit B
7163-*-Invert Bit B
Where,
T1E1 is from 1 to 8.
Invert Bit B
Meaning
Disable
Enable
Invert Bit C
Use following command to program to invert/don't invert Bit C for the T1E1 port:
7164-1-T1E1-Invert Bit C
7164-2-T1E1-T1E1-Invert Bit C
7164-*-Invert Bit C
Where,
T1E1 is from 1 to 8.
Invert Bit B
Meaning
Disable
Enable
Invert Bit D
Use following command to program to invert/don't invert Bit D for the T1E1 port:
7165-1-T1E1-Invert Bit D
7165-2-T1E1-T1E1-Invert Bit D
7165-*-Invert Bit D
Where,
T1E1 is from 1 to 8.
Invert Bit D
Meaning
Disable
Enable
E1 Metering Bit
Use following command to program the E1 Metering Bit for the T1E1 port:
7166-1-T1E1-E1 Metering Bit
7166-2-T1E1-T1E1-E1 Metering Bit
7166-*-E1 Metering Bit
Where,
1023
T1E1 is from 1 to 8.
E1 Metering Bit
Meaning
None
Bit-A
Bit-B
Bit-C
Bit-D
Meaning
Release Timer
Use following command to program the Release Timer for the T1E1 port:
7169-1-T1E1-Release Timer
7169-2-T1E1-T1E1-Release Timer
7169-*-Release Timer
Where,
T1E1 is from 1 to 8.
Release Timer is from 20ms to 1000ms.
By default, Release Timer is 400 ms.
1024
E&M Signaling
E&M Feature Template
Assign E&M Feature Template to T1E1 using following command:
Use Following command to assign E&M Feature Template to T1E1 Port
6004-1-T1E1-Template Number
6004-2-T1E1-T1E1- Template Number
6004-*- Template Number
Where,
T1E1 is from 1 to 8.
Template Number is 01 to 50.
By default, Template 01 is assigned to T1E1.
B Bit Pattern
Use following command to select B Bit Pattern
7191-1-T1E1-Code
7191-2-T1E1-T1E1-Code
7191-*-Code
Where,
T1E1 is from 1 to 8.
Code
Meaning
Same as A bit
Fixed Value
B Bit Value
Use following command to program B Bit Value
1025
CD Bit Value
Use following command to program CD Bit Value
7193-1-T1E1-CD Bit Value
7193-2-T1E1-T1E1-CD Bit Value
7193-*- CD Bit Value
Where,
T1E1 is from 1 to 8.
CD bit value can be 1 or 3.
By default, CD bit value is 1.
Invert Bit A
Use following command to program to invert/don't invert Bit A for the T1E1 port:
7162-1-T1E1-Invert Bit A
7162-2-T1E1-T1E1-Invert Bit A
7162-*-Invert Bit A
Where,
T1E1 is from 1 to 8.
Invert Bit A
Meaning
Do not invert
Invert
Invert Bit B
Use following command to program to invert/don't invert Bit B for the T1E1 port:
7163-1-T1E1-Invert Bit B
7163-2-T1E1-T1E1-Invert Bit B
7163-*-Invert Bit B
Where,
T1E1 is from 1 to 8.
Invert Bit B
Meaning
Do not invert
Invert
Invert Bit C
Use following command to program to invert/don't invert Bit C for the T1E1 port:
7164-1-T1E1-Invert Bit C
7164-2-T1E1-T1E1-Invert Bit C
1026
7164-*-Invert Bit C
Where,
T1E1 is from 1 to 8.
Invert Bit C
Meaning
Do not invert
Invert
Invert Bit D
Use following command to program to invert/don't invert Bit D for the T1E1 port:
7165-1-T1E1-Invert Bit D
7165-2-T1E1-T1E1-Invert Bit D
7165-*-Invert Bit D
Where,
T1E1 is from 1 to 8.
Invert Bit D
Meaning
Do not invert
Invert
1027
You can also view the T1E1 Trunk Status from the Status link. To view, click the T1E1 link under Status.
1028
This topic explains the connection of PRI line, and application of PRI.
PRI port of the ETERNITY configured for NT mode can be connected to the PRI Port of another PBX
configured for TE mode. In such case, the ETERNITY will behave as Transit Exchange.
Dialing method on the PRI port is not programmable. The PRI port (configured for TE mode) will send the
called party number in Enbloc mode.
All the switch variants are applicable to the PRI port whether programmed in TE or NT mode.
Applications:
Applications for PRI-NT Port are as described below:
Video conferencing system connected to the PRI-NT port with IC/OG calls.
Data Calls support.
Networking of PBXs.
PBX
ISDN
PRI TE
PRI NT
Video
Conferencing
System
Connect the Video Conference system (H.320) to the PRI-NT port of the ETERNITY.
This feature works only if it is supported by the Service Provider. Video Conference is established mainly
by the Video Conferencing (VC) equipment.
Video Conferencing requires 6 B-channels. But Video Conferencing can also be done at lower bit rates
also using the aggregation of 6 B channels which must be supported by the VC equipment.
The VC equipment uses two methods:H.221 and H.242 BONDING, for aggregation.
More than one Video conferencing system can be connected to the ETERNITY to each PRI-NT port.
It also supports internal calling like, one Video Conferencing system to another.
1029
OG calls from VC
Program the OGTBG (OG Trunk Bundle Group) such that the user gets at least 6 B-channels of the same
PRI port.
Go OFF-Hook.
If VC at the called party responds, the call is established which occupies 6-B Channels.
If 6 B-channels on the same PRI port are not available, user at VC will get busy tone.
The user at calling partys VC gets dial tone of the ISDN exchange.
The user at VC starts dialing the destination number. The destination number is sent in keypad IE by the
VC user to the PRI-NT port whereas it is dialed on the PRI-TE port in the method programmed for the port
that is, Enbloc.
Rest of the signaling is done between the VC equipment and the called partys VC equipment.
If VC equipment supports Phone Book feature, user can make call using the feature.
At the end of VC, all the 6 B-channels are freed and available for other users.
It is preferable that the SE will assign an OGTBG to the Video Conferencing equipment in such a manner
that the same group is not assigned to any other station. This is to allow Video Conferencing call at any
time.
IC Video Call:
Program the system to place IC video calls to the PRI-NT port using DDI Routing Table.
For this, program the DDI Routing Table. Refer related chapter.
You can also prepare the Trunk Landing Group (Routing Group) to land the call to PRI-NT port. For this
purpose, Routing Group is to be modified.
1030
For data-communication, connect the Router supporting ISDN-PRI or a Computer with ETERNITY ME
Card T1E1PRI to the PRI-NT port as shown below:
PBX
ISDN
PRI TE
Hub
PRI NT
Router
The computers connected in the LAN can browse the net through the PRI.
Remote LAN Access the Computers in the LAN can access the computer/computers in LAN at the remote
end (Branch office/Home office).
The data call can be made by the router requesting desired number of channels. This establishes a live
connection between the Router and the ISP through the PBX. The users on the LAN can browse the net
as normal using Internet Explorer or Netscape Navigator.
For this, the ETERNITY will allocate data channels only on the PRI-TE port so as to leave other channels
for speech calls when the system detects the call to be a data call.
Similarly, a Remote Computer can be accessed (Remote LAN Access) by dialing the Remote users
number (The remote end PBX should be so programmed that the call made to a number lands directly on
the Router.) This establishes a permanent connection between the two Routers (and hence two LAN
networks).
Now the user at PBX-A can access the computer in LAN at the remote end in the same way as accessing
another computer on the same network.
Program the Trunk Landing Group such that all data calls will land on the PRI-NT port to which the Router
is connected using CLI based Routing logic or DDI based routing logic.
However, while routing call on the PRI-NT port, the PBX will check that the data channels reserved for data
communication on the PRI-NT port are enough to establish the call. Otherwise the call will be rejected.
The call will be rejected if the number of channels, reserved for data calls are already busy with one datacall.
1031
Networking of PBXs:
One of the applications is also for connecting multiple PBX using PRI-NT and PRI-TE ports. A simple connection of
only two PBXs is explained below. Refer following figure for connecting another PBX to the PRI-NT port.
PBX-A
2001
PBX-B
ISDN
ISDN
3001
PRI TE
PRI TE
PRI NT
PRI TE
A station 2001 of PBX-A can call a station 3001 of PBX-B by dialing 3001.
Such a call can be routed using CUG table of PBX-A. Refer chapter Closed User Group (CUG).
Program (for PBX-A) an OGTBG containing the PRI-NT port and assign Trunk access code to it.
Grab the PRI-NT by dialing access code. User of 2001 will get dummy dial tone of the PBX-A.
Method 2
User of 2001 can dial trunk access code of PBX-A, and dial a trunk access code of PBX-B. He will be
connected to ISDN network through PBX-B.
Then he can make OG call on PRI-NT port as per method programmed for the PRI-NT port. (For example,
PRI-NT port can be programmed as a Trunk port and when station user of PBX-B grabs the PRI-NT port,
his call will be placed on the destination programmed for PRI-NT port in the Trunk Landing Group).
1032
Using CUG feature of PBX-B, 3001 can call 2001 through PRI-NT port. Refer chapter Closed User Group
(CUG) for more details.
3001 can also dial Trunk Access Code to grab its PRI-TE port. He will get dial tone of PRI-NT Port of PBXA. Now 3001 can dial 2001 or trunk access code to make a call to the ISDN network connected to PBX-A.
How to configure
For programming of the PRI, please refer topic Configuring E1 Trunks and Configuring T1 Trunks for more
details.
Trunk software ports are automatically assigned to the PRI port by the system depending on the slot in
which they are inserted.
Please take care to terminate the PRI line on the HDSL interface only of the ISDN modem.
The DDI Routing and Routing Tables shall be programmed as explained in chapters on DDI.
1033
Configuring T1 Trunks
Whats this?
Digital Signal Level 1 (T1E1) trunks use Bit-Oriented Signaling (BOS) and multiplexes 24 channels (T1 service) or
32 channels (E1 service) into a single data stream. T1E1 can be used for voice or voice-grade data and for datatransmission protocols. T1 trunk service multiplexes 24 channels into a single 1.544-Mbps data stream. E1 trunk
service multiplexes 32 channels into a single 2.048-Mbps stream. Both T1 and E1 provide a digital interface for
trunk groups.
Signaling Modes
Common Channel Signaling (CCS) is an industry-standard technique where any one of a group of channels carries
the signals for the other channels. Matrix uses the 24th channel of a group for signaling. This signaling technique
differs from 24-channel signaling. When the system is configured for Facility-Associated Signaling, 24-channel
signaling uses the 24th channel in a T1E1 facility to carry signals. This technique also is called clear channel, outof-band or alternate voice data (AVD) signaling.
ROBBED-BIT signaling is a per-channel in-band signaling technique for transmitting signaling bits on each channel
in a T1E1 facility. The least-significant bit in every 6th transmitted information frame is removed and replaced by a
signaling bit. This technique is also called in-band signaling. The maximum transmission rate for each bearer
channel with ROBBED-BIT signaling is 56 Kbps. T1 RBS Protocol supports 24 channels (from 01 to 24) and the
protocol doesnt consume any channel for signaling so that there are total 24 channels available for the users.
ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps (T1) connection and on the 16th channel for a
2.048 Mbps (E1) connection. In case of T1 PRI Protocol supports 24 channels (from 01 to 24), in which channel no.
24 is used for the signaling, so effectively there are 23 Voice channels are available.
QSIG is an ISDN based protocol for signaling between nodes of a Private Integrated Services network.
Any of the common trunks, except for PCOL (Personal Central Office Line) trunks, can be analog or digital. (PCOL
trunks can only be analog.) Administering a digital trunk group is very similar to administering its analog
counterpart, but digital trunks must connect to a T1E1 port and this port must be administered separately.
1034
Select the T1E1 Port number you want to configure by clicking the respective tab, and program the
following port parameters:
ETERNITY will assign the Hardware Slot-Port automatically, when any card is inserted in the system.
Hardware slot is the number of the Universal slot of ETERNITY in which the T1E1 Card is inserted.
Range of slot number is 01-16. Port is the number of T1E1 hardware port on the card to which the
T1E1 line is connected. Range of Port is from 00-99.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields. By default, Hardware slotPort is 0000.
You may assign a Name to the T1E1 Port for identification of the port. The Name may consist of a
maximum of 12 characters. By default, it is blank.
Select the Carrier type as T1. The Carrier type will automatically be assigned to the port when you
select the region. You may it if required.
Select Signal Type. Signal Type signifies the type of signaling to be used on E1 line. The E1 signalling
supported are:
PRI
RBS
QSIG
E&M
1035
Default: PRI
If you select PRI or QSIG as the Signal Type, configure the PRI/QSIG Parameters.
If you select RBS as the Signal Type, configure the RBS Parameters.
If you select E&M as the Signal Type, configure the E&M Parameters.
Set the Orientation Type. Select the orientation type from the following options:
Terminal
Network
Tie-Line
Default: Terminal
If you have set Terminal as the Orientation Type, you must select the type of network with which it is
to be Interfaced With. The network may be:
Public ISDN
Private ISDN
Default: Public ISDN
You may configure the port to Treat Incoming call as Trunk or Station.
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The
calls will be routed as per the Trunk Feature Template assigned to the T1E1 Port.
If you select Station, you must also assign a Station Basic Feature Template and Station Advanced
Feature Template to the T1E1 Port.
When you select Station, the system will treat the calling party as an extension user. The user will have
access to all the features and facilities of the system, as per the Station Basic Feature Template and
Station Advanced Feature Template assigned to the T1E1 Port.
By default, Trunk is selected.
1036
If Point-to-Point is selected as the Interface Type, you can select the option Trunk or Station for the
parameter Treat Incoming call as.
If Point-to-Multipoint is selected as the Interface Type, only Station can be set as the option for the
parameter Treat Incoming call as.
If Station is selected as the option for Treat Incoming call as, the user will only be able to:
Dial Flexible Numbers
Dial Operator Code
Dial Trunk Access Code for making outgoing calls
Access the Global Directory
Make calls within the Closed User Group
Line coding is a pattern that data assumes as it is propagated over a communication channel. Select
the Line Coding Mechanism from the following:
AMI-Basic
B8ZS
NRZ (Fiber Optic)
Default: AMI-Basic.
Framing means to form a set of 24 or 32, 8 bits time slot that is to be treated as single transmission
unit. The Framing Modes supported by ETERNITY are:
SF (D4)
ESF
Default: SF (D4)
Line Buildout Parameter field reduces the outgoing signal strength by a fixed amount. The
appropriate level of loss depends on the distance between your switch (measured by cable length from
the smart jack) and the nearest repeater. Where another switch is at the end of the circuit, as in campus
environments, use the cable length between the 2 switches to select the appropriate setting from the
list. This field is relevant if the Near-end CSU type field is integrated. By default, 0-133ft is selected as
the Line Buildout Parameter.
Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By
default, IC Reference ID is 00.
To know more about OG Reference ID, see IC Reference Table.
Assign Trunk Feature Template to the T1E1 Port. Trunk Feature Template is a set of general features
that define the behavior of a Trunk Port. By default, Template 01 is assigned to all T1E1 Ports.
For more details, see Trunk Feature Template.
Station Basic Feature Template assigned to the T1E1 Port is displayed in this field. Station Basic
Feature Template is a set of general features that define the basic behavior of a station. By default,
Template 01 is assigned to all T1E1 Ports.
For more details, see Station Basic Feature Template.
Station Advanced Feature Template assigned to the T1E1 Port is displayed in this field. Station
Advanced Feature Template is a set of advanced features, to be applied on extensions such as CLIP,
Floor Service, Walk-in Class of Service. By default, Template 01 is assigned to all T1E1 Ports.
For more details, see Station Advanced Feature Template.
Select Priority for the T1E1 Port. Priority is the precedence given to certain trunks and extensions over
others in being answered by the destination extension. You can select from 1 to 9. By default, Priority
5-Normal is set for all T1E1Ports.
For know more about Priority feature, see Priority.
Assign a Cost Factor to the T1E1 Port. By default, all the T1E1 Ports are assigned Cost Factor 01.
For more details, see Cost Factor.
ISDN glare occurs if the system initiates an outgoing call on a B-Channel at the same time the network
initiates an incoming call on that same B-channel. You may configure the Glare Option as Proceed or
Held Back. While processing a glare condition, the configured Glare Option on T1E1 port will be
considered.
1037
Select Category (Logical Partition) for the T1E1 Port. You may select from the following options:
1
2
3
By default, 1 is selected for all T1E1 Ports.
Idle Code is the 8-bit sequence that occupies the time slot on a E1/T1 trunk channel when it is not
being used. By default, 127 is configured as the Idle Code.
If you want ETERNITY to display the called party number as the CLI for incoming calls, select the
Display Called Party Number as CLI check box. By default, Display Called Party Number as CLI
option is disabled.
This parameter is useful when a single T1E1 line connection and Operator are shared by more than
one organization. To enable the Operator to handle calls more efficiently, you must enable Display
Called Party Number as CLI and configure the names and corresponding numbers of the organizations
sharing line in the Global Directory of ETERNITY.
When there is an incoming call, ETERNITY matches the number with the numbers in the Global
Directory. If a match is found ETERNITY displays the company name configured for that entry to the
Operator.
You can configure the Display Called Party Number as CLI option only from Jeeves.
Select Return Call to Original Caller (RCOC) flag to enable this feature on the T1E1 Port. By default,
RCOC flag is disabled.
For know more about RCOC feature, see RCOC (Return Call to Original Caller).
In Channel Reserved for Data Call, configure the channels you want to reserve for Data Calls. By
default, 00 channels are reserved.
In Channel Reserved for Outgoing Call, configure the channels you want to reserve for making
outgoing calls. By default, 30 channels are reserved.
In Channel Reserved for Incoming Call, configure the channels you want to reserve for receiving
incoming calls. By default, 30 channels are reserved.
When a caller dials the trunk access code or selective trunk access code for dialing the number directly
on the trunk port, the caller waits for the dial tone before dialing the number. But some exchanges do
not give Dial Tone for the T1E1 Port. For Example, when T1E1 port as E1CAS type is used in Delhi, it
is observed that the exchange does not give dial tone when direct dialing on the trunk is used.
Enable the Feed Dial Tone flag. ETERNITY will provides the dial tone to the caller when the T1E1 Port
is accessed.
It is applicable only when Online dialing is used as, Store and Forward dialing, the dial tone is given to
the caller. The dial tone is played as per the Dial Tone Timer.
By default, Feed Dial Tone flag is disabled.
1038
When dial tone flag is disabled, user will hear the dial tone of the exchange if provided, otherwise, user
will hear the silence.
If the user is making the call from the FXS port and dial tone is not provided by the exchange, user will
not know when to start dialing the number. In this case, it is possible that some digits are not out dialed
on the port and wrong number is dialed out because system will out dial the number only if Outgoing
call Acknowledge is received from the ETERNITY ME Card T1E1 and user will not know about this
condition. Hence, it is required to enable this flag, if exchange is not providing the dial tone.
When Online dialing or Store and Forward dialing are used, some exchanges do not provide any tone
while routing/processing the call. Thus, the caller does not know whether the call is being processed or
not as there is silence. In such case, You must enable the Feed Routing Tone flag, ETERNITY will
play the routing tone to the caller. ETERNITY stops playing the routing tone will be stopped when an
alert message or connect message or disconnect message is received from the T1E1 Card.
By default, the Feed Routing Tone flag is disabled.
To customize the pulse width option and set the pulse shapes configure the Custom Pulse
parameters. ETERNITY generates pulse shapes which match the country standard, where it is
installed. However, if the standard pulse shape does not match, ETERNITY enables you to customize
the pulse width to match your exact requirements.
To use customize pulse width option and set the pulse shape in 1 to 4 phases, keep the T1/E1 Custom
Pulse Width (CPW) flag enabled.
Word 1: Set the pulse width for setting pulse shape in the 1st phase. The range of Custom Pulse
Width-Word 1 is 001 to 127. Default: 109.
Word 2: Set the pulse width for setting pulse shape in the 2nd phase. The range of Custom Pulse
Width-Word 2 is 001 to 127. Default: 107.
Word 3: Set the pulse width for setting pulse shape in the 3rd phase. The range of Custom Pulse
Width-Word 3 is 001 to 127. Default: 64.
Word 4: Set the pulse width for setting pulse shape in the 4th phase. The range of Custom Pulse
Width-Word 4 is 001 to 127. Default: 64.
Select the Bearer Service supported by your service provider. You can select from:
Speech
3.1 KHz Audio
By default, Speech is selected.
The Overlap Receiving Timer is relevant while receiving the called party number information in
overlap receiving mode. It is not relevant for the port in overlap sending mode.
Range of Overlap Receiving Timer is from 01 to 99 seconds. Default: 15 seconds.
Configure Pause Timer for theT1E1 Port. Range of Pause Timer is from 1 to 9 seconds. By default, it
is set to 3 seconds for all T1E1 Ports.
1039
This Timer is required to insert delay between the digits while dialing out DTMF digits on the T1E1 port.
One of the applications for using this parameter is Multi-stage dialing. Refer chapter Multi-Stage
Dialing.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the ETERNITY
will out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
When the SETUP Message is sent by ETERNITY to the network (ISDN exchange), the exchange
responds by sending SETUP ACK (Acknowledgement), and dial tone is played to the caller. The time
taken by the exchange to respond to the SETUP message may vary from exchange to exchange. Set
the SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP
message and play dial tone to the caller.
Valid Range of the timer is 01 to 20 seconds. By default it is set to 4 seconds.
Change the default settings only if required. If the time you set is less than the time taken by the exchange
to respond, no dial tone will be played to the caller.
You can configure the SETUP Message Timer only from Jeeves.
Configure DTMF On Time for theT1E1 Port. Range of DTMF On Time is from 051 to 255 ms. By
default, it is set to 102 ms for all T1E1 Ports.
The DTMF On Time is the time for which the DTMF digit which is to be outdialed by the ETERNITY
remain On. One of the applications for using this parameter is Multi-stage dialing. Refer chapter MultiStage Dialing.
Configure DTMF Inter Digit Pause Timer for the T1E1 Port. Range of Inter Digit Pause Timer is from
051 to 255 ms. By default, it is set to 102 ms for all T1E1 Ports.
Inter Digit Pause Timer is the time for which the system will wait while receiving the dialing digits to
consider it as end-of-dialing.
Configure the Minimum ON Time (msec) for which the DTMF signal should be present in order to be
detected. The valid range of this time is 10 to 200 milliseconds. By default, Minimum ON Time is set to
20 milliseconds.
Configure the Minimum OFF Time (msec). This parameter signifies the minimum time period between
successive DTMF digits. The valid range of this time is 10 to 200 milliseconds. By default, Minimum
OFF Time is set to 20 milliseconds.
Configure the Minimum Level (dB) for the DTMF digit to be considered as valid. The valid range of
this time is 0 to -36.5 dB. By default, Minimum levels is set to -36.5dB.
Make sure you have installed T1E1 Card with version V4R3.1 or higher, to ensure DTMF detection on the
T1E1 trunk.
1040
Configure Call Budget parameters for the T1E1 Ports. Call Budget is an expense control feature of
ETERNITY that allows you to keep track of the cost of phone calls made from the T1E1 Port.
Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the T1E1
Port. By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field.
By default the Amount is set to 999999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this
field. By default the number of minutes is set to 999999.
Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By
default the number of calls is set to 9999.
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to
be reset on a particular date of every month.
Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan
suggests so.
Call Back: This parameter is related to the Call Back on Trunk Port feature. If you want to enable the
'Call Back on Trunk Port' feature on this T1E1 Port, configure the following parameters:
Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this
flag is disabled on all trunk port types. By default, the flag is disabled.
Call Back Timer (sec): This is the duration for which the system waits for the caller to disconnect
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set
to 10 seconds.
Call Back Mode: Select from the following options how a Call Back call answered by the remote
party should be routed:
Built-In Auto Attendant
PIN Authentication - Multiple Calls
CLI Authentication - Multiple Calls
CLI Authentication - Single Call - Answer Signaling
Operator
By default, Operator is selected as the Call Back Mode.
Call Back on: This parameter allows you to select if the call back should be made to the same
number that was received or to a different number. If you want the call back to be made to the same
number select the CLI number. If you want the call back to be made to a different number, select
Alternate Number.
By default, CLI number is selected for Call Back.
Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
1041
Number List 15 is also assigned to all T1E1 Ports as well as all other Trunk port types. If you want
the same numbers strings to be programmed commonly for all T1E1 Ports and Trunk Port types,
retain this list.
If you want a different set of number strings to be programmed for this T1E1 Port, select a different
Number List, and assign it to the T1E1 Port.
You may program the Incoming Number List either from the Number List page or by clicking the
Incoming Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the Incoming Number List, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made.
For example, for the number string programmed at Index 1 in the Incoming Number List, a
corresponding number string at the same Index, Index 1, should be programmed in the Outgoing
Number List.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all T1E1 Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this T1E1 Port.
You may program the Outgoing Number List either from the Number List page or by clicking the
Outgoing Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Call Back from: This parameter determines the trunk port to be used to make the call back. The
call back can be made using the Same Port or an OG Trunk Bundle Group.
Select Same Port if you want the call back to be made using the same port on which the missed
call is received. If you select OGTB Group, the call back will be made using the OGTB Group, which
you have defined.
By default, Same Port is selected.
OGTB Group: If you selected OGTB Group for making the call back in the previous parameter, you
must define the OGTB Group that must be used in this parameter.
By default, OGTB Group 01 is assigned.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTB Group that you define here for Call Back.
1042
FDL is used for communicating general maintenance information or for transmitting user defined
information within the T1 link. General maintenance information is in the form of Performance Message
Report which is generated by the ETERNITY ME Card T1E1PRI and depending upon the FDL
Protocol, the Performance Message Report is sent every second or on request.
Select the FDL flag to enable. This parameter is applicable only if Framing = ESF. If the Network
(Public or Private) to which the ETERNITY is connected does not support FDL then FDL will be
disabled. By default, the T1 FDL is disabled.
If you have enabled FDL, configure the FDL Protocol. ETERNITY supports ANSI T1.403 and AT&T
54016 protocols of reporting the performance monitoring. By default, the T1 FDL Protocol is ANSI
T1.403.
For more information, see T1 Maintenance
Enter appropriate Debug Code (Level 1 to 4), to obtain debug information of various parts of T1E1
Card on the COM Port. By default, debug is off for all T1E1 ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from
000 to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for T1E1 Port are as specified below:
Enter appropriate Debug Code (Level 1 to 4), to obtain debug information of various parts of T1E1
Card on the COM Port. By default, debug is off for all T1E1 ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from
000 to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for T1E1 Port are as specified below:
Level 1:
Unused
Unused
001
CAS
002
MFC R2
004
CAS DSP
008
Layer 4
000
Debug Off
Unused
Unused
Layer 4
Unused
HDLC (D-Channel)
CAS DSP
MFC R2
CAS
Level 2:
Unused
Unused
001
Alarms
002
Counters
004
ABCD Bits
008
FDL
016
HDLC (D
Channel)
000
Debug Off
FDL
ABCD Bits
Counters
Alarms
1043
Level 3:
Unused
Flow Debug
NLS Debug
001
Primitives
002
State
004
Variables
008
SVC Primitives
016
LAP Debug
032
NLS Debug
064
Flow Debug
000
Debug Off
LAP Debug
SVC Primitives
Variables
State
Primitives
Level 4:
Unused
Unused
001
OS Task
002
NI Debug
000
Debug Off
Unused
Unused
Unused
Unused
NI Debug
OS Task
PRI/QSIG Parameters
If you have selected PRI as Signal Type on the Port Parameters page,
1044
Under T1E1 Configuration, click PRI/QSIG Signalling and configure the PRI/QSIG parameters.
ISDN Switch Variant: ISDN supports a variety of service provider switches. Different countries use
specific type of ISDN switch. This switch is designed using ISDN standard protocol. The type of switch
determines various factors such as how many ISDN devices would be handled, which B-channel will
support voice, video, data etc. Select the ISDN Switch Variant from the list. Default:ETSI NET5.
Offer continuous Bearer Channel Mapping(01-30): Select this check box for continuous Bearer
Channel Mapping for E1-QSIG/E1-PRI.
D-Channel: Enter the Channel number that is used for Data signalling. By default, D-Channel is 16.
Valid Range is 1 to 31.
Send Called Party Number Using: Select the appropriate option from the following for Send Called
Party Number Using:
Called Party Number IE (Information Element)
Keypad Facility IE (Information Element)
Default: Called Party Number IE is selected.
Dialing Type for Called Party Number: Select the option from the following as supported by your
exchange:
Enbloc
Digit-by-Digit
Any
Default: Any.
Caller - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the calling party:
Unknown
International
1045
National
Network Specific
Subscriber
Abbreviated
Reserved
Default: Unknown.
1046
Caller- Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the calling party:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
Default: ISDN Numbering.
Called - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the called party:
Unknown
International
National
Network Specific
Subscriber
Abbreviated
Reserved
Default: Unknown.
Called - Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the called party:
Unknown
ISDN Numbering
Data Numbering
Telex Numbering
National Numbering
Private
Reserved
Default: ISDN Numbering.
Receive Equalization Mode: You can set the Receive Equalization Mode as Auto or Manual. By default,
Auto is selected as the Receive Equalization Mode.
Receive Equalization Parameters: This field increases the strength of incoming signals by a fixed
amount to compensate for line losses. Select the required option from the list. By default, the receive
equalization parameters of T1E1 is 8dB.
Feed Inband Tones on T1E1-NT, before sending DISCONNECT: Select this flag, if you want to feed
inband tones on T1E1-NT before sending DISCONNECT message. This flag is applicable only when
T1E1 port is configured as 'Network'. When this flag is enabled, inband tones shall be feed for 15 seconds
(fixed, non programmable) before sending DISCONNECT message.
When this flag is disabled, inband tones (Busy/Error as applicable for the state of the call) shall not be feed
before sending the DISCONNECT message. However when DISCONNECT message is sent from T1E1NT port, inband tones will always be sent with 'progress indicator 8'. By default, this flag is disabled.
E&M Signalling
Under T1E1 Configuration, click E&M Signalling and configure the E&M parameters.
E&M Feature Template: Assign an E&M Feature Template to the T1E1 Port. The E&M Feature Template
is a set of features specific to E&M signaling, which define the behavior of the E&M ports, according to
their 'Orientation Type', whether they are functioning as Stations, Trunks or Tie-Lines. By default,
Template 01 is assigned to all T1E1 Ports.
For more details, see E&M Feature Template.
B Bit Pattern: Select the Bit Pattern from Same as Bit A or Fixed Value. By default, the Code is 1 (Same
as A bit).
B Bit Value: Configure the B bit value, the value can be 0 or 1. By default, B bit value is 0.
CD Bit Value: Configure the CD bit value, the valid range of the value is 1 to 3. By default, B bit value is 0.
Invert Bit A Flag: This parameter signifies whether A-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit A.
Default: Disabled (Do Not Invert Bit A).
Invert Bit B Flag: This parameter signifies whether B-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit B.
Default: Disabled (Do Not Invert Bit B1)
1047
Invert Bit C Flag: This parameter signifies whether C-bit is to be inverted before transmitting and on
receiving. Select the check box to Invert Bit C.
Default: Disabled (Do Not Invert Bit C).
Invert Bit D Flag: This parameter signifies whether D-bit is to be inverted before transmitting and on
receiving. Select the check box to enable, that is to Invert Bit D.
Default: Disabled (Do Not Invert Bit D).
RBS Parameters
Under T1E1 Configuration, click T1 RBS Signalling and configure the RBS parameters.
1048
Line Signaling Variant: Select the T1 Line Signaling Variant from following options:
SLT Loop Start
CO Loop Start
SLT Ground Start
CO Ground Start
E&M Immediate Dial/Start
E&M Wink Start
E&M Wink Start FGD
Default: E&M Wink Start FGD.
Wink Timer (milliseconds): Wink timer refers to the momentary Off-Hook condition to acknowledge
end of making an outgoing call. The Wink Timer ranges from 0001 ms to 9999 ms. Default: 160 msec.
Wink Wait Timer (milliseconds): Wink Wait Timer signifies the maximum time the system should wait
before sending a wink start signal after an incoming seizure is detected. Wink Wait Timer ranges from
0001 to 9999 msec. Default: 30msec.
Ensure that this timer is greater than the Wink Wait Timer of the other end.
Wait Wink Timer (milliseconds): Wait Wink Timer signifies the time for which ETERNITY will wait for
receiving the DNIS after sending the outgoing seizure signal. Wait Wink Timer ranges from 001 to 999
msec. Default: 5000 msec.
Make sure that this timer is greater than the Wait Wink Timer of the other end.
Delay Duration (milliseconds): This duration signifies the time after which the DNIS information is to
be sent while making an outgoing call. Range of the Delay Duration is from 0001 to 9999 msec.
Default: 100 msec.
Start Delay Timer: Start Delay Timer signifies the time for which ETERNITY waits for receiving DNIS
from the network. This timer is loaded on receiving the Off-hook (I/C Seizure) on the receive channel
(while receiving an incoming call). The Start Delay Timer ranges from 0001 to 9999 ms. Default: 20
msec.
Register Signaling Variant: The Register Signaling Variant for T1/E1 Ports is set as DTMF.
Inbound ANI/DNIS Format: Select the Inbound ANI/DNIS Format for T1/E1 Ports from the following
options:
ANI
DNIS
?ANI?
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
Default:?ANI?DNIS?.
Inbound Delimiter (?) Character: Define the Inbound Delimiter Character in this field. Characters
supported in this field are 0-9, #, *, A, B, C and D. Default: *
Outbound ANI/DNIS Format: Select the Outbound ANI/DNIS Format for T1/E1 Port from the following
options:
ANI
DNIS
?ANI?
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
Default:?ANI?DNIS?.
Outbound Delimiter (?) Character: Define the Outbound Delimiter Character in this field. Characters
supported in this field are 0-9, #, *, A, B, C and D. Default: *
1049
Port Parameters
T1E1-1
Hardware Slot-Port
Use following command to assign hardware ID to a T1E1 software port.
1107-T1E1-Slot-Port offset
Where,
T1E1 is from 1 to 8.
Slot is the number of the universal slot, where the T1E1 card is installed, from 01 to 16.
Port is the number of theT1E1 port on the card, from 01 to 32.
Use following command to de-assign the hardware slot and the hardware port assigned to the T1E1 software port.
1106-T1E1-00-00
Port Status
Use the following command to enable/disable the port:
6101-1-T1E1-Port Status
6101-2-T1E1-T1E1-Port Status
6101-*-Port Status
Where,
T1E1 is from 1 to 8.
Port Status
Meaning
Disable
Enable
Name
Use the following command to assign a name to the port:
5407-1-T1E1-Name
5407-2-T1E1-T1E1-Name
5407-*-Name
Where,
T1E1 is from 1 to 8.
Name can be of upto 18 characters.
Carrier
Use the following command to select the carrier:
6108-1-T1E1-Carrier Type
6108-2-T1E1-T1E1-Carrier Type
6108-*-Carrier Type
1050
Where,
T1E1 is from 1 to 8.
Carrier
Type
Meaning
E1
T1
Line type
Use following command to program signaling type/ Line type of a T1E1:
6105-1-T1E1-Line Type
6105-2-T1E1-T1E1-Line Type
6105-*-Line Type
Where,
T1E1 is from 1 to 8.
Line Type
Meaning
ISDN_E1_PRI
ISDN_T1_PRI
ISDN_E1_CAS
ISDN_T1_RBS
ISDN_E1_QSIG
ISDN_T1_QSIG
ISDN_E1_E&M
ISDN_T1_E&M
DDI Routing is not supported on T1/E1 trunk line if you have selected E&M as the Signal Type.
Orientation Type
Use following command to program 'Orientation Type' for the T1E1 port:
6106-1-T1E1-Orientation Type
6106-2.T1E1-T1E1-Orientation Type
6106-*-Orientation Type
Where,
T1E1=1 To 8.
Orientation
Meaning
Terminal
Network
Tie Line
1051
By default Type = 1.
When Orientation = Terminal, the port will be regarded as trunk. All the trunk related parameters will be applicable.
When Orientation = Network, the port will be regarded as station. All the station related parameters will be
applicable.
When Orientation = Tie-line, the port will be regarded as station for all IC calls to it and as trunk for all OG calls to
be made through it.
Meaning
AMI-Basic
B8ZS
CMI
Framing Mode
Use following command to program the Framing Mode for the T1E1 port:
6104-1-T1E1-Framing
6104-2-T1E1-T1E1-Framing
6104-*-Framing
Where,
Framing
Meaning
SF (D4) for T1
ESF for T1
1052
6162-*-Code
Where,
T1E1 is from 1 to 8.
Code
Meaning
0-133ft
133-266ft
266-399ft
399-533ft
533-665ft
-7.5dB or equidistance
-16dB or equidistance
-22.5dB or equidistance
DDI Routing
OG Reference ID
Use the following command to assign OG Reference ID to T1E1 port:
6131-1-T1E1-OG Reference ID
6131-2-T1E1-T1E1-OG Reference ID
6131-*-OG Reference ID
Where,
T1E1 is from 1 to 8.
OG Reference ID is from 00 to 99.
By default, OG Reference ID is 00.
1053
Templates
Trunk Feature Template
Use the following command to assign a Trunk Feature Template to the T1E1 Trunks, dial:
5806-1-T1E1- Trunk Feature Template Number to assign a template to a single T1E1 port.
5806-2-T1E1- Trunk Feature Template Number to assign the same template to a range of T1E1 ports.
5806-*- Trunk Feature Template Number to assign the same template to all T1E1 ports.
Where,
T1E1is the Software Port number of the port from 1 to 8.
Template Number is the number of the customized Trunk Feature Template, from 01 to 50. Default: Trunk Feature
Template 01.
Others
Priority
To assign priority to T1E1
3914-1-T1E1-Priority to assign a template to a single T1E1 port.
3914-2-T1E1PRI-T1E1PRI-Priority to assign the same priority to a range of T1E1 ports.
3914-*-Template Number to assign the same Priority to all T1E1 ports.
Where,
T1E1PRI is the number of the T1E1PRI Software port, from 1 to 8.
1054
Cost Factor
Use following command to assign a name to the T1E1 port:
6102-1-T1E1-SP
6102-2-T1E1-T1E1-SP
6102-*-SP
Where,
T1E1 is from 1 to 8.
SP is from 01 to 99.
By default, Service Provider is 01.
Glare Option
Use following command to program Glare Option for the T1E1 port:
6112-1-T1E1-Glare Option
6112-2-T1E1-T1E1-Glare Option
6112-*-Glare Option
Where,
T1E1 is from 1 to 8.
Glare Option
Meaning
Proceed
Held Back
Idle Code
Use the following command to program the Idle Code of a T1E1:
6113-1-T1E1-Idle Code
6113-2-T1E1-T1E1-Idle Code
6113-*-Idle Code
Where,
T1E1 is from 1 to 8.
Idle Code is from 000 to 255 (corresponding to 8 bits).
1055
RCOC
To enable RCOC on T1E1 Trunk
Dial 6145-1-T1E1-Code to enable the feature on a single trunk.
Dial 6145-2-T1E1-T1E1-Code to enable the feature on a range of trunks.
Dial 6145-*-Code to enable the feature on all trunks.
Where,
T1E1 is the software port number of the trunk from 1 to 8.
Code is
0 for Disable
1 for Enable
Default: Disable
Channels
Return Call to Original Caller (RCOC)
Use the following command to enable RCOC on T1E1 Trunk:
6145-1-T1E1-Code to enable the feature on a single trunk.
6145-2-T1E1-T1E1-Code to enable the feature on a range of trunks.
6145-*-Code to enable the feature on all trunks.
Where,
T1E1 is the software port number of the trunk from 1 to 8.
Code is
0 for Disable
1 for Enable
Default: Disable
1056
Channel Count (OG) is from 00 to 30. "It specifies the number of channels to be reserved for making an OG calls.
For example, If OG channel count is programmed as 15, simultaneous 15 (maximum) OG calls can be made from
the T1E1 port".
By default, OG Channel Count is 30.
Tone
Feed Dial Tone
Use following command to program the dial tone flag for T1E1 port:
6115-1-T1E1-Flag
6115-2-T1E1-T1E1-Flag
6115-*-Flag
Where,
T1E1 is from 1 to 8.
Flag
Meaning
Disable
Enable
By default, Dial Tone Flag is '0' for all the T1E1 ports.
Meaning
Disable
Enable
By default, Routing Tone Flag is '0' for all the T1E1 ports.
1057
Custom Pulse
T1/E1 Custom Pulse Width (CPW)
Use following command to enable/disable Custom Pulse Width (CPW) Flag for the T1E1 port for T1 signaling:
6171-1-T1E1-Flag
6171-2-T1E1-T1E1-Flag
6171-*-Flag
Where,
T1E1 is from 1 to 8.
Flag
Meaning
Disable
Enable
1058
Timer
Pause Timer
Use following command to program Pause Timer:
6109-1-T1E1- Pause Timer
6109-2-T1E1-T1E1-Pause Timer
6109-*-Pause Timer
Where,
T1E1 is from 1 to 8
Pause Timer is from 1 to 9 seconds
By default, Pause Timer is 3 seconds.
DTMF ON Time
Use following command to program DTMF ON Time:
6117-1-T1E1-DTMF ON Time
6117-2-T1E1-T1E1-DTMF ON Time
6117-*-DTMF ON Time
Where,
T1E1 is from 1 to 8.
DTMF ON Time is from 051 to 255 msec.
By default, DTMF ON Time is 102 msec.
Gateway
Use Gateway Application - Answer Signaling?
Use following command to set flag for 'Gateway Application-Answer Signaling' on T1E1 trunk:
1059
Meaning
Disable
Enable
#4
#5
#6
#7
**
##
Call Budget
Call Budget Type
To program Call Budget Type on T1E1 Port, dial:
6122-1-T1E1-Budget Type to program call budget type for a single trunk port.
6122-2-T1E1-T1E1-Budget Type to program the same call budget type for a range of trunk ports.
6122-*-Budget Type to program the same call budget type for all trunk ports.
Where,
T1E1is the number of the T1E1 software port from 1 to 8.
Budget Type is
0 for None
1 for Amount
1060
2 for Minutes
3 for Number of Calls
By default, Budget Type is None.
1061
Call Back
Use the following commands to program Call Back on T1E1 Trunk ports. To know more about this feature, refer the
topic Call Back on Trunk Ports.
To enable/disable Call Back on T1E1 port:
6176-1-T1E1- Call Back Flag
6176-2-T1E1-T1E1-Call Back Flag
6176- *-Call Back Flag
Where,
T1EI is from 1 to 8
Call Back Flag
Meaning
Disable
Enable
1062
Meaning
Meaning
Operator
Call Back On
To program Call Back On method for T1E1 port:
6179-1-T1E1-Call Back on selection
6179-2-T1E1-T1E1-Call Back on selection
6179-*-Call Back on selection
Where,
T1E1 is from 1 to 8
Call back on selection is
Call Back on
Meaning
CLI Number
Alternate Number
1063
T1E1 is from 1 to 8
Call Back From is
1 for Same Port
2 for OGTB Group
By default, Same Port is selected as Call Back From.
OGTB Group
To assign a Call Back - OGTB Group for a T1E1 port:
6149-1-T1E1-OGTB Group
6149-2-T1E1-T1E1-OGTB Group
6149-*-OGTB Group
Where,
T1E1 is from 1 to 8
OGTB Group is from 01 to 32
By default, OGTBG is 01.
FDL
FDL Flag - copied from T1 Maintenance
Use following command to enable/disable T1 FDL on a T1E1PRI port:
6164-1-T1E1PRI-T1 FDL
6164-2-T1E1PRI-T1E1PRI-T1 FDL
6164-*-T1 FDL
Where,
T1E1PRI is from 1 to 8.
T1 FDL
Meaning
Disable
Enable
FDL Protocol
Use following command to program the T1 FDL Protocol for a T1E1PRI port:
6165-1-T1E1PRI-T1 FDL Protocol
6165-2-T1E1PRI-T1E1PRI-T1 FDL Protocol
6165-*-T1 FDL Protocol
Where,
T1E1PRI is from 1 to 8.
T1 FDL Protocol
Meaning
Disable
AT&T 54016
ANSI T1.403
1064
Refer T1 Maintenance
Debug
ETERNITY supports debug of parameters (debug codes) depending on the Level of debug. On issuing this
command the ETERNITY ME Card T1E1 will send the debug details to the COM port of the T1E1 port.
Please note following command change as per Version of Software used. First set of Command '6191' is
used up to Version V6R0.12 and Commands at the end of 'Level 4' are used for Version V6R0.13
onwards. Here 'XXX' is the Code as mentioned in Tables for Level1 to level 4.
Till S/W V6R12
Debug Level-1
6191-1-T1E1-1-XXX
6191-1-T1E1-1-XXX
Debug Level-2
6191-1-T1E1-2-XXX
6191-1-T1E1-2-XXX
Debug Level-3
6191-1-T1E1-3-XXX
6192-1-T1E1-1-XXX
Debug Level-4
6191-1-T1E1-4-XXX
6192-1-T1E1-2-XXX
Option 1
Use following command to start/stop debug the parameters for the T1E1 port:
6191-1-T1E1-Level-Debug Code
6191-2-T1E1-T1E1-Level-Debug Code
6191-*-Level-Debug Code
Where,
T1E1 Port is from 1 to 8.
Level is from 1 to 4 (As shown below).
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000 to 255.
Code value 000 for each level will turn off that levels debug.
Level 1:
Unused
Unused
001
CAS
002
MFC R2
004
CAS DSP
008
Layer 4
000
Debug Off
Unused
Unused
Layer 4
Unused
HDLC (D-Channel)
CAS DSP
MFC R2
CAS
Level 2:
Unused
Unused
001
Alarms
002
Counters
004
ABCD Bits
008
FDL
FDL
ABCD Bits
Counters
Alarms
1065
016
HDLC (D
Channel)
000
Debug Off
Level 3:
Unused
Flow Debug
NLS Debug
001
Primitives
002
State
004
Variables
008
SVC Primitives
016
LAP Debug
032
NLS Debug
064
Flow Debug
000
Debug Off
LAP Debug
SVC Primitives
Variables
State
Primitives
Level 4:
Unused
Unused
Unused
001
OS Task
002
NI Debug
000
Debug Off
Unused
Unused
Unused
NI Debug
OS Task
Default: Debug Code = Debug OFF for all T1E1 ports for all levels.
1066
Meaning
ATT 4ESS
ATT 5ESS
Australia
DMS
ETSI NET5
Meaning
NTT INS64
SWV Hongkong
US NI12
QSIG E1
10
QSIG T1
Meaning
Unknown: This is used when the user or network has no a prior information about
the numbering plan. In this case, the Address Value field is organized according to
the network dialing plan. For example, prefix or escape digits might be present.
International Number.
1067
Source TON
Meaning
Abbreviated Number.
Reserved Number.
Meaning
Unknown
Telex Numbering
1068
Meaning
Unknown: This is used when the user or network has no a prior information about
the numbering plan. In this case, the Address Value field is organized according to
the network dialing plan. For example, prefix or escape digits might be present.
International Number.
Destination TON
Meaning
Abbreviated Number.
Reserved Number.
Meaning
Unknown
Telex Numbering
Meaning
Manual
Auto
1069
Meaning
None
8 dB
16 dB
24 dB
32 dB
40 dB
48 dB
Meaning
No
Yes
Default = NO.
E&M Signaling
E&M Feature Template
Assign E&M Feature Template to T1E1 using following command:
Use Following command to assign E&M Feature Template to T1E1 Port
6004-1-T1E1-Template Number
6004-2-T1E1-T1E1- Template Number
6004-*- Template Number
Where,
T1E1 is from 1 to 8.
Template Number is 01 to 50.
By default, Template 01 is assigned to T1E1.
Now proceed to program other parameters for E&M on T1E1 using following commands:
B Bit Pattern
Use following command to select B Bit Pattern
7191-1-T1E1-Code
1070
7191-2-T1E1-T1E1-Code
7191-*-Code
Where,
T1E1 is from 1 to 8.
Code
Meaning
Same as A bit
Fixed Value
B Bit Value
Use following command to program B Bit Value
7192-1-T1E1-B Bit Value
7192-2-T1E1-T1E1- B Bit Value
7192-*- B Bit Value
Where,
T1E1 is from 1 to 8.
B bit value can be 0 or 1.
By default, B bit value is 0.
CD Bit Value
Use following command to program CD Bit Value
7193-1-T1E1-CD Bit Value
7193-2-T1E1-T1E1-CD Bit Value
7193-*- CD Bit Value
Where,
T1E1 is from 1 to 8.
CD bit value can be 1 or 3.
By default, CD bit value is 1.
Invert Bit A
Use following command to program to invert/don't invert Bit A for the T1E1 port:
7162-1-T1E1-Invert Bit A
7162-2-T1E1-T1E1-Invert Bit A
7162-*-Invert Bit A
Where,
T1E1 is from 1 to 8.
Invert Bit A
Meaning
Disable
Enable
Invert Bit B
Use following command to program to invert/don't invert Bit B for the T1E1 port:
7163-1-T1E1-Invert Bit B
7163-2-T1E1-T1E1-Invert Bit B
1071
7163-*-Invert Bit B
Where,
T1E1 is from 1 to 8.
Invert Bit B
Meaning
Disable
Enable
Invert Bit C
Use following command to program to invert/don't invert Bit C for the T1E1 port:
7164-1-T1E1-Invert Bit C
7164-2-T1E1-T1E1-Invert Bit C
7164-*-Invert Bit C
Where,
T1E1 is from 1 to 8.
Invert Bit C
Meaning
Disable
Enable
Invert Bit D
Use following command to program to invert/don't invert Bit D for the T1E1 port:
7165-1-T1E1-Invert Bit D
7165-2-T1E1-T1E1-Invert Bit D
7165-*-Invert Bit D
Where,
T1E1 is from 1 to 8.
Invert Bit D
Meaning
Disable
Enable
RBS Signaling
To assign a Line Signaling Variants to the T1E1 port, dial:
6181-1-T1E1-T1 Line Signaling Variants
6181-2-T1E1-T1E1-T1 Line Signaling Variants
6181-*-T1 Line Signaling Variants
Where,
1072
T1E1 is from 1 to 7.
T1 Line Signaling Variants
Meaning
CO Loop Start
CO Ground Start
1073
Use the following command to program the T1 Start Delay Timer for T1E1:
6186-1-T1E1-Start Delay Timer
6186-2-T1E1-T1E1-Start Delay Timer
6186-*-Start Delay Timer
Where,
T1E1 is from 1 to 8.
Start Delay Duration is from 001 to 255 seconds.
By default, T1 Start Delay Timer is 020 seconds.
To assign a Register Signaling Variant to the T1E1 port, dial:
6161-1-T1E1-T1 Register Signaling Variant
6161-2-T1E1-T1E1-T1 Register Signaling Variant
6161-*-T1 Register Signaling Variant
Where,
T1E1 is from 1 to 8.
T1 Register Signaling Variant
1
Meaning
T1 RBS DTMF
T1 RBS DTMF: DNIS is transmitted in the corresponding speech channel using the DTMF signals as per the ITU-T
Q.23.
To select the Inbound ANI/DNIS Format , dial:
6166-1-T1E1-Code
6166-2-T1E1-T1E1-Code
6166-*-Code
Where,
T1E1 is from 1 to 8.
Code is
T1 Register Signaling Variant
Code
ANI
DNIS
?ANI?
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
1074
Code
ANI
DNIS
?ANI?
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
1075
You can also view the T1E1 Trunk Status from the Status link. To view, click the T1E1 link under Status.
1076
T1 RBS Parameters
Whats this?
Some countries like North America support the standard of 1.544Mbps of PCM trunk. This is known as T1 Trunks.
The T1 type of PCM Trunks use Robbed Bit Signaling. ROBBED-BIT signaling is a per-channel signaling technique
for transmitting signaling bits on each channel in a T1E1 facility. The least-significant bit in every 6th transmitted
information frame is removed and replaced by a signaling bit. This technique is also called in-band signaling. The
maximum transmission rate for each bearer channel with ROBBED-BIT signaling is 56 Kbps.
ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps connection and on the 16th channel for a 2.048
Mbps connection. There are two types of parameters:
Line Signaling (ABCD Bits)
Register Signaling
Line signaling is described by following types:
E&M Wink Start FGD
E&M Wink Start
E&M Immediate Dial/Start
SLT Ground start
SLT Loop Start
CO Loop Start
CO Ground Start
T1 Line signaling type is applicable when the Line Type is programmed as T1 RBS for the T1E1 Port.
The Register signaling supported by ETERNITY is DTMF.
T1 Line signaling is applicable when the Line Type is programmed as ISDN_T1_RBS for the T1E1 Port.
Refer chapter Configuring T1 Trunks to program Line Type as T1 RBS.
The various Line Signaling Variant are explained below:
Making an OG Call
The far end (network) sends a wink (a momentary OFF-Hook for 200ms), that is, bits A, B, C and D on the
receive channel receive a pulse (Active High) of 200ms.
On receipt of the wink signal, DNIS (Dialed Number Identification Service) is sent on the speech channels
using the Register Signaling type (DTMF or Decadic (A-Bit) or R1 MFC or R2 MFC).
DNIS is Dialed Number In Service. It is the ISDN number that is being dialed. This is provided by the telco in
the call setup messages. DNIS can be used to provide differentiated service to dialing users.
The call goes through when the called party answers the call.
1077
Receiving an IC Call
The far end sends the DNIS in the speech channels using Register Signaling.
Disconnect
By the Network-Bits A and B on the receive channel are 0. Bits C and D are also 0 in ESF. Following this,
the bits on the transmit channel are set to 0 by the PBX.
By the PBX-Bit A and B on the transmit channel are set to 0. Bit C and D are also set to 0 in case of ESF.
Following this, the bits A and B(C and D in ESF) are received as 0 on the receive channel.
Bits A and B are set to 1 to indicate OFF-Hook. Bits A and B are set to 0 to indicate ON-Hook. Bits C and
D follow bits A and B (Incase of ESF).
Making an OG Call
The far end (network) sends a wink (a momentary OFF-Hook for 200ms.), that is, bits A, B, C and D on the
receive channel receive a pulse (Active High) of 200ms.
On receipt of the wink signal, DNIS (Dialed Number Identification Service. It is the ISDN number that is
being dialed. This is provided by the telco in the call setup messages. DNIS can be used to provide
differentiated service to dialing users.) is sent on the speech channels using the Register Signaling type.
The call goes through when the called party answers the call.
Receiving an IC Call
The far end sends the DNIS in the speech channels using DTMF.
The PBX goes off hook when the call is answered by the called party.
Disconnect
By the Network-Bits A and B on the receive channel are 0. Bits C and D are also 0 in ESF. Following this,
the bits on the transmit channel are set to 0 by the PBX.
By the PBX-Bit A and B on the transmit channel are set to 0. Bit C and D are also set to 0 in case of ESF.
Following this, the bits A and (C and D in ESF) are received as 0 on the receive channel.
Please note that while using T1 RBS, only DID is being sent/received and not the CLI/ANI.
1078
State
Transmit
ON-Hook
Transmit
OFF-Hook/Loop Closed
Receive
ON-Hook
Receive
OFF-Hook
Receive
Ringing
Making an OG Call
To transmit OFF-Hook, bits A (and Bit C if ESF) on the transmit channel is set to 1.
The ETERNITY sends the DNIS (Dialed Number Identification Service) on the speech channels using the
DTMF signals.
The call goes through when the called party answers the call.
Receiving an IC Call
The Network sends the DNIS on the speech channel using DTMF signals.
The PBX detects toggling of bit B. When the called station of the PBX answers, the PBX transmits Offhook state by changing bit-A from 0 to 1.
Disconnect
By the Network:
No indication from the Network. The PBX will detect error tone to detect a disconnect from the network. On
detecting on-hook from the network, the PBX transmits on-hook by setting Bit A from 1 to 0.
By the PBX:
1079
State
Transmit
ON-Hook/Loop Open
Transmit
Ground on Ring
Transmit
OFF-Hook/Loop Closed
Receive
Receive
OFF-Hook/TIP Ground
Receive
Ringing
Making an OG Call
To transmit Off-hook, bits A (and Bit C if ESF) and B (and Bit C if ESF) on the transmit channel is set to 0.
The network detects this change and goes off-hook. The A-bit on the receive channel goes from 1 to 0.
The B-bit is set to 1.
PBX detects dial tone and sends DNIS (DTMF digits) on the corresponding speech channel.
The call goes through when the called party answers the call.
Receiving an IC Call
The Network sends the DNIS on the speech channel using DTMF signals.
When the called station of the PBX answers, the PBS transmits Off-hook state by changing bit-A from 0 to
1.
Disconnect
1080
By the Network-A bit on the receive channel of the PBX goes from 0 to 1. On detecting on-hook from the
network, the PBX transmits on-hook by setting Bit A from 1 to 0.
By the PBX-The PBX transmits on-hook by setting Bit A from 1 to 0. On detecting on-hook from the PBX,
the network transmits on-hook by setting Bit A from 0 to 1.
CO Loop Start or CO Ground Start is used when the ETERNITY is connected to the Network, that is, when
the T1E1 port is configured for Terminal mode (Connection mode).
Whereas FXS Loop Start or FXS Ground Start is used when the ETERNITY is connected to another
ETERNITY, that is, when the T1E1 port is configured for Network mode.
CO Loop Start-This is complementary to FXS Loop Start explained above.
CO Ground Start-This is complementary to FXS Ground Start explained above.
Click T1 RBS Parameters to open the page and configure the following:
Line Signaling Variant: Select the T1 Line Signaling Variant from following options:
SLT Loop Start
CO Loop Start
SLT Ground Start
CO Ground Start
E&M Immediate Dial/Start
E&M Wink Start
E&M Wink Start FGD
Default: E&M Wink Start FGD.
Wink Timer (milliseconds): Wink timer refers to the momentary Off-Hook condition to acknowledge
end of making an outgoing call. The Wink Timer ranges from 0001 ms to 9999 ms. Default: 160 msec.
Wink Wait Timer (milliseconds): Wink Wait Timer signifies the maximum time the system should wait
before sending a wink start signal after an incoming seizure is detected. Wink Wait Timer ranges from
0001 to 9999 msec. Default: 30msec.
Ensure that this timer is greater than the Wink Wait Timer of the other end.
Wait Wink Timer (milliseconds): Wait Wink Timer signifies the time for which ETERNITY will wait for
receiving the DNIS after sending the outgoing seizure signal. Wait Wink Timer ranges from 001 to 999
msec. Default: 5000 msec.
Make sure that this timer is greater than the Wait Wink Timer of the other end.
Delay Duration (milliseconds): This duration signifies the time after which the DNIS information is to
be sent while making an outgoing call. Range of the Delay Duration is from 0001 to 9999 msec.
Default: 100 msec.
Start Delay Timer: Start Delay Timer signifies the time for which ETERNITY waits for receiving DNIS
from the network. This timer is loaded on receiving the Off-hook (I/C Seizure) on the receive channel
(while receiving an incoming call). The Start Delay Timer ranges from 0001 to 9999 ms. Default: 20
msec.
1081
Register Signaling Variant: The Register Signaling Variant for T1/E1 Ports is set as DTMF.
Inbound ANI/DNIS Format: Select the Inbound ANI/DNIS Format for T1/E1 Ports from the following
options:
ANI
DNIS
?ANI?
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
Default:?ANI?DNIS?.
Inbound Delimiter (?) Character: Define the Inbound Delimiter Character in this field. Characters
supported in this field are 0-9, #, *, A, B, C and D. Default: *
Outbound ANI/DNIS Format: Select the Outbound ANI/DNIS Format for T1/E1 Port from the following
options:
ANI
DNIS
?ANI?
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
Default:?ANI?DNIS?.
Outbound Delimiter (?) Character: Define the Outbound Delimiter Character in this field. Characters
supported in this field are 0-9, #, *, A, B, C and D. Default: *
Meaning
CO Loop Start
CO Ground Start
1082
1083
T1E1 is from 1 to 8.
T1 Register Signaling Variant
1
Meaning
T1 RBS DTMF
T1 RBS DTMF: DNIS is transmitted in the corresponding speech channel using the DTMF signals as
per the ITU-T Q.23.
Code
ANI
DNIS
?ANI?
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
1084
Code
ANI
DNIS
?ANI?
Code
?DNIS?
?ANI?DNIS?
?DNIS?ANI?
Exit SE mode.
Relevant Topics:
1. T1 Maintenance
2364
2. Configuring T1 Trunks
1034
1085
You can customize mobile port parameters and select the network using Jeeves and a Telephone. However,
mobile port status can be viewed using Jeeves only.
169. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.
1086
Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the mobile card has been
inserted. 'Port' is the number of the Mobile port in which you have installed the SIM card.
The number of the Universal Slot will depend on the model of ETERNITY you are using. The number of
the Mobile Port depends on the configuration of the Mobile card. For instance, if you have installed
GSM8 card, the number of the ports will be from 1 to 8.
By default the ETERNITY can detect and assign the hardware slot and port numbers automatically to
the mobile (software) ports. However, if required, you may change the Hardware Slot and Port
assigned to the mobile software port. In which case, enter the desired Hardware Slot and Port number
in this field.
If you want to de-assign the Hardware Slot and Port, enter '00' in both fields.
Enable Port: This flag is for enabling or disabling a Mobile Trunk port. When a Mobile Trunk port is
disabled, neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clicking the check box.
Name: You may assign a 'Name' to each Mobile Trunk to facilitate identification. Whenever there is an
incoming call without CLI on this port, the Name you have programmed will be displayed on the landing
extension.
The Name of the port may be the name of the Service Provider of the SIM Card you have installed on
this port (recommended) or the phone number assigned to the SIM card on this port.
The Name may comprise a maximum of 18 characters.
Band Selection (MHz): The Frequency Band supported by the mobile networks varies from country to
country. ETERNITY's mobile card supports frequency bands of most countries. Select the Frequency
Band used by your GSM/3G Provider for the mobile port.
By default, Mobile Frequency Band 'All Bands is selected.
Frequency Band selection not required if the Mobile Card has SIMCOM 3G module.
When you change the Frequency Band, the change will be effected after the next system restart or the
next Mobile Port restart.
Preferred Network Mode: Select the Preferred Network Mode for the Mobile Port as Dual, GSM or
UMTS. By default, Dual Mode is selected.
SIM PIN: If you have enabled PIN protection and changed the SIM PIN of the card to the default value
'1234' using a mobile handset170, you can assign a new PIN to the SIM card from the ETERNITY.
Make sure that the PIN stored on the SIM card and that of the system are the same.
170. Refer the topic Installing the Mobile Card for instructions. If you have not enabled PIN protection before installing the GSM card,
you will not be able to change the SIM PIN.
1087
You must click Submit after you enter the new PIN. Wait for 5 seconds, and then refresh this page to
view the new SIM PIN.
If you have enabled PIN protection, and the SIM PIN on the Card and the SIM PIN programmed in the
ETERNITY are not the same, the SIM card may get blocked and would require the Personal
Unblocking Number (PUK) from the Service Provider to reactivate it again.
The SIM PIN will not be set to default value, when you restore the default settings of the system.
Incoming Calls: Select the Incoming Call Mode on the Mobile Port from the options:
Accept': incoming calls will be allowed and incoming call logic is applied.
Ignore': incoming calls will not be processed further and call logic will not be applied.
Reject': incoming calls will be rejected immediately and mobile port will be freed (released).
By default, incoming calls are accepted. This feature is particularly useful in outbound call centers for
blocking incoming calls on a SIM number (mobile port) of the ETERNITY.
Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator,
Auto Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the
behavior of a Trunk. Apply a Trunk Feature Template to the Mobile Trunk. By default, Trunk Feature
Template 01 is applied on all Mobile Trunks as well as all other trunk types. Refer the topic Trunk
Feature Template to know more.
Click the 'Trunk Feature Template link to open the page. Check if the default Template 01 fulfills your
requirement for the mobile port.
If the default Template 01 does not fulfill your requirement, prepare another Trunk Feature
Template171, and enter the newly prepared Template number for the Mobile port.
CLIR: Enable this flag if you want to activate CLIR for all outgoing calls made through the Mobile Port.
When CLIR is enabled, the called party will not be able to see the subscriber number of the Mobile
Port.
By default CLIR is disabled for all Mobile ports.
This feature will work only if subscribed/supported by your mobile service provider.
Return Call to Original Caller (RCOC): Enable this flag if you want to apply the RCOC feature.
If this feature is enabled on the Mobile trunk port, the system routes calls returned by remote parties
back to the extensions that originally made the call from this port (the original callers' extensions). To
know more, refer the feature description for Return Call to Original Caller (RCOC).
Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the
mobile port.
171. The default template is applied on the ports of all trunk types supported by ETERNITY. Changes to the default template will be
applied on all trunk types also. So, you are advised to prepare a new template and apply it to the desired trunk types.
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Cost Factor is a number assigned to each trunk for identification. This number also serves as a
preference number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same
preference must be assigned the same Cost Factor. Different trunk types can also be assigned the
same Cost Factor. These trunks are used for routing calls.
Assign a Cost Factor to the Mobile Trunk port, for example, 03 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '9' through the SIM installed in
Mobile Port Number 01 only,
You must first assign a Cost Factor (01-99) to Mobile Port 01, for example, 03.
Enter '9' in the 'Number' column, Cost Factor '03' as Preference 1, 2, 3 and 4.
All outgoing calls assigned Cost Factor trunk 03 will be made from Mobile Port 01.
Advanced Configuration
The above listed parameters fulfill the basic mobile trunk port configuration requirements of most users. However, it
is anticipated that some users may need to configure other less commonly used features on the mobile ports, such
as Call Budget, Call Back on Mobile Port, or they may want to use the Mobile port as a Gateway application.
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For such users, you may click the 'Advance' button and program the following parameters:
N/w Registration Retry Count: The mobile port is programmed to automatically locate and register with
the Network that supports the SIM card installed on it. Also, at each power ON, the mobile port (SIM) will
automatically register with the Network that supports the SIM on it.
However, if the Mobile port fails to register, it will restart the process of network registration on the expiry of
the Network Registration Retry Timer172. On the expiry of this timer, the system will retry registration for
the programmed Count (number of times) and with each re-try attempt, the count will be decremented by
one.
By default the Retry Count is set to 2. If required you may change the Count to the desired value.
Mobile Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive
Gain) and Outgoing Speech (Transmit Gain) on the Mobile port by changing the Rx Gain and Tx Gain to
the desired level. Different levels can be set for each port type in the Mobile Gain Settings Template. By
default, Mobile Gain Template 1 is assigned. If you want to assign a different Template, you must
customize the Mobile Gain Settings Template first and then assign the number of the Mobile Gain Settings
Template here. To customise the Mobile Gain Settings, see Gain Settings.
172. The Network Registration Retry Timer defines the time for which the Mobile port, which has failed to register with the network,
should wait before attempting to re-register with the network. Network registration retry timer is 2 minutes and is non-programmable.
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If you change the Tx or Rx Gain during an active call, the change you made will not apply on the current
call. It will be applied on the next call.
Call Back: This parameter is related to the Call Back on Trunk Port feature. If you want to enable the 'Call
Back on Trunk Port' feature on this Mobile trunk, configure the following parameters:
Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this flag
is disabled on all trunk port types. By default, the flag is disabled.
Call Back Timer: This is the duration for which the system waits for the caller to disconnect before
applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to 10
seconds.
Call Back Mode: Select from the following options how a Call Back call answered by the remote party
should be routed:
Built-In Auto Attendant
PIN Authentication - Multiple Calls
CLI Authentication - Multiple Calls
CLI Authentication - Single Call - Answer Signaling
Operator
By default, Operator is selected as the Call Back Mode.
Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the CLI number. If you want the call back to be made to a different number, select Alternate
Number.
By default, CLI number is selected for Call Back.
Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all Mobile trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all Mobile trunks and Trunk Port types, retain
this list.
If you want a different set of number strings to be programmed for this Mobile Trunk, select a different
Number List, and assign it to the Mobile trunk port.
You may program the Incoming Number List either from the Number List page or by clicking the
Incoming Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the Incoming Number List, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
example, for the number string programmed at Index 1 in the Incoming Number List, a corresponding
number string at the same Index, Index 1, should be programmed in the Outgoing Number List.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all Mobile trunks as well as all other Trunk port types.
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You may program the default number list, or a different number list and assign it to this Mobile Trunk
port.
You may program the Outgoing Number List either from the Number List page or by clicking the
Outgoing Number List link to reach the Number List page.
Refer the topic Number Lists to know more, and for configuration instructions.
Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an OG Trunk Bundle Group.
Select Same port if you want the call back to be made using the same port on which the missed call is
received. If you select OGTBG, the call back will be made using the OGTBG, which you have defined.
By default, Same port is selected.
OGTB Group: If you selected OGTBG for making the call back in the previous parameter, you must
define the OGTBG that must be used in this parameter.
By default, OGTBG 01 is assigned.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost Routing
on the OGTBG that you define here for Call Back.
Call Budget: If you want to enable 'Call Budget on Trunk' feature, configure the following parameters for
this mobile trunk port:
Type: Select the type of Call Budget on TrunkAmount, Minutes or Number of Callsto be applied on
this mobile trunk port. By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
Calls: If you selected Calls as the Call Budget Type, enter the number of calls in this field. By default,
the number of calls is set to 9999.
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
on a particular date of every month.
Scheduled (Date): Enter the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be every month. You may select Daily if your plan suggests so.
The consumed Call Budget can be from SE and SA Mode, referred to as Manual Reset. Refer the feature
description Call Budget on Trunk.
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Accept Anonymous Calls: The flag is for accepting calls without CLI that land on the mobile port. By
default the flag is enabled. You may disable this flag to disallow calls without CLI on this Mobile port.
Pause Timer: This Timer is used for providing delay in number dialing from the Mobile port. The Pause
Timer will be applicable when the digit 'P' is configured in the DTMF number string which is to be out dialed
as DTMF digits on the Mobile port.
For example, if PPP2 is to be out dialed and Pause timer is programmed as 3 seconds, the ETERNITY will
out dial the digit 2 after 9 seconds, that is, after a delay of individual P (3+3+3 =9). The range of this time is
from 1 to 9. By default the Timer is set to 1 seconds.
This parameter is used for the Multi-Stage Dialing feature.
DTMF Outdial Option: You can select whether to send the DTMF digits from the Mobile Ports Inband or
through signaling, that is, AT Command. By default, DTMF Outdial Option is Inband.
When you select DTMF Outdial using AT Command, the length of the DTMF digits will be determined by
the DTMF ON Time you set.
DTMF ON Timer: This parameter determines the time for which the DTMF digit will remain ON, while
being out dialed by the ETERNITY. This parameter finds its application in the feature Multi-Stage Dialing'
and in DTMF Outdialing using AT Command. By default, DTMF ON Timer is 100 ms.
DTMF Detection Mode: You can select whether to detect the DTMF digits using the GSM Modules or
through DSP. Default: Using DSP.
When you select DTMF Detection Using Module, the length of the DTMF digits will be determined by the
DTMF Detection Timer you set.
DTMF Detection Duration: This is the minimum ON to consider the DTMF digit as a valid digit. The valid
range of this timer is 20 to 100ms. Default: 30ms.
In ETERNITY ME GSM8 Card with firmware Version 1, only the DTMF ON Timer is supported. If you have
installed this card in your system, configure DTMF ON Timer.
Min. Level (dB): This parameter signifies the minimum level (dB) of the DTMF digit to be considered as
valid. By default, Minimum level is set to -30dB.
If the Min. Level set is very low, the DTMF digits might be detected in Voice and if it is very high, the DTMF
digits may be lost.
If received DTMF digit level is higher than or equal to the set value of Min. Level (dB), the system will
accept the DTMF.
If received DTMF digit level is lower than the set value of Min. Level (dB), the system will ignore the DTMF.
Category (Logical Partition): This parameter assigns the Mobile Port to a trunk category for the purpose
of Logical Partitioning. By default all Mobile Ports are assigned to Category 1. Do not change the default
setting.
If you want to change the call permission between the mobile port and other trunks, click the 'Category' link
to open the Logical Partitioning page. You may program the call permission between Category 1 (assigned
to Mobile Trunk Ports) and other Categories. Refer the feature description Logical Partition to know
more.
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Gateway Application-Answer Signaling: This parameter is to be programmed if the Mobile Trunk Port is
being used in a gateway application as a source port (from where calls originate). The calls originated on
the source port (mobile port) are routed using another Trunk port, the terminating port, which may be any
trunk port, for example: T1E1. When call made from the terminating port gets matured, this is signaled to
the source port in the form of DTMF digits.
Enable: Enable this flag if you want the Mobile port to be used in a Gateway Application.
DTMF String (max. 4 digits): Program the DTMF digits to be sent to signal call maturity to the source
port.
SMS Parameters: If you want to use this Mobile Trunk Port for the SMS Server application to send/receive
SMS, configure the following parameters:
SMS Center Number: The system displays the default SMS Center Number of the network. If
required, you can change the SMS Center Number. The SMS Center Number can be a maximum of 16
digits. Valid Range: 0 to 9 and +
Send SMS: Enable this check box if you want to use this Mobile Trunk Port to send SMS.
Number of SMS to be sent (Total): Enter the maximum number of SMS that can be sent using this
Mobile Trunk Port. Default: 0999999999. Valid Range: 1 to 4294967296.
Number of SMS to be sent (Daily):Enter the maximum number of SMS that can be sent using this
Mobile Trunk Port daily. Default: 20000. Valid Range: 1 to 20000.
Receive SMS: Enable this check box if you want to use this Mobile Trunk Port to receive SMS.
Debug: Enable this flag by selecting the check box if you want to initiate debugging for the Mobile Port. By
default, debugging is disabled.
If you have completed configuration of all the above listed Mobile Port Parameters, click Submit at the
bottom of the page to save your changes.
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1095
SIM PIN is the new SIM PIN, any combination of 4 to 8 digits. Terminate the command with '#*' if SIM
PIN is fewer than 8 digits.
To select Incoming Call mode on the Mobile port, dial:
8005-1-Mobile-Mode to select the mode for a single mobile port.
8005-2-Mobile-Mobile-Mode to select the same mode for a range of mobile ports.
8005-*-Mode to select the same mode for all mobile ports.
Where,
Mode is
1 for Accept Incoming Calls.
2 for Ignore Incoming Calls
3 for Reject Incoming Calls
Default: Accept
To assign a Trunk Feature Template to the Mobile Port, dial:
5807-1-Mobile Port Number-Template Number to assign a template to a single mobile port.
5807-2-Mobile Port Number-Mobile Port Number-Template Number to assign the same template to
a range of mobile ports.
5807-*-Template Number to assign the same template to all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64
Template Number is the number of the Trunk Feature Template from 01 to 50.
To enable/disable CLIR on the Mobile Port, dial:
8031-1-Mobile Port Number-CLIR to enable/disable CLIR on a single mobile port.
8031-2-Mobile Port Number-Mobile Port Number-CLIR to enable/disable CLIR on a range of mobile
ports.
8031-*- CLIR to enable/disable CLIR on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64
CLIR is
0 for Disable
1 for Enable
Default: Disable
To enable RCOC on Mobile Port, dial:
8030-1-Mobile -Code to enable the feature on a single mobile port.
8030-2-Mobile-Mobile-Code to enable the feature on a range of mobile ports.
8030-*-Code to enable the feature on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64
Code is
0 for Disable
1 for Enable
Default: Disabled
To assign Cost Factor to a Mobile Port, dial:
8001-1-Mobile-Cost Factor to assign cost factor to a single mobile port.
8001-2-Mobile-Mobile-Cost Factor to assign the same cost factor to a range of mobile ports.
8001-*-Cost Factor to assign the same cost factor to all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64
Cost Factor is from 01 to 99.
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For Advanced Configuration of the Mobile Ports, use the following commands:
To set Network Registration Retry Count, dial:
8004-1-Mobile-N/w Registration Retry Count to set the count for a single mobile port.
8004-2-Mobile-Mobile-N/w Registration Retry Count to set the same count for a range of mobile
ports.
8004-*-N/w Registration Retry Count to set the same count for all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64
N/w Registration Retry Count is from 001 - 255: Default: 002
For commands to program Call Budget on Mobile Ports, refer the topic Call Budget on Trunk.
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Where,
Mobile Port Number is the number of the software port from 01 to 64.
Call Back on is
1 for CLI Number
2 for Alternate Number.
To assign a Call Back - Incoming Number List to a Mobile Port, dial:
8034-1-Mobile-Incoming Number List to assign a list to a single mobile port.
8034-2-Mobile-Mobile-Incoming Number List to assign the same list to a range of mobile port.
8034-*-Incoming Number List to assign the same list to all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
Incoming Number List is from 01 to 16
Default: 15
To assign a Call Back - Outgoing Number List to a Mobile Port, dial:
8035-1-Mobile-Outgoing Number List to assign a list to a single mobile port.
8035-2-Mobile-Mobile-Outgoing Number List to assign the same list to a range of mobile port.
8035-*-Outgoing Number List to assign the same list to all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
Outgoing Number List is from 01 to 16
Default: 16
To select Call Back From for a Mobile Port, dial:
8036-1-Mobile-Call Back From to select call back from for a single mobile port.
8036-2-Mobile-Mobile-Call Back From to select the same call back from option for a range of mobile
ports.
8036-* -Call Back From to select the same call back from option for all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
Call Back From is
1 for Same Port
2 for OGTB Group
Default: Same Port
To assign a Call Back - OGTB Group for a Mobile Port, dial:
8037-1-Mobile-OGTB Group to assign an OGTBG to a single mobile port.
8037-2-Mobile-Mobile-OGTB Group to assign the same OGTBG to a range of mobile ports.
8037-*-OGTB Group to assign the same OGTBG to all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
OGTB Group is from 01 to 32
Default: 01
To enable/disable Accept Anonymous Calls on Mobile Port, dial:
8029-1-Mobile-Code to enable/disable Anonymous Calls on Mobile Port on a single mobile port.
8029-2-Mobile-Mobile-Code to enable/disable Anonymous Calls on Mobile Port on a range of mobile
ports.
8029-*-Code to enable/disable Anonymous Calls on Mobile Port on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
Scheduled Reset Date is from 01 to 31.
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1099
Default: 1
To enable/disable Gateway Application on the Mobile Port, dial:
8016-1-Mobile-Gateway Application flag to enable/disable on a single mobile port.
8016-2-Mobile-Mobile-Gateway Application flag to enable/disable on a range of mobile ports.
8016-*-Gateway Application flag to enable/disable on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
Gateway Application flag is.
0 for Disable
1 for Enable
Default: Disable
To program the DTMF String for the Gateway Application on the Mobile Port, dial:
8017-1-Mobile-DTMF String to program the string on a single mobile port.
8017-2-Mobile-Mobile-DTMF String to program the same string on a range of mobile ports.
8017-*-DTMF String to program the same string on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
DTMF String is a maximum of 4 digits. Default: CCC
To enable/disable Debug on a Mobile Port, dial:
8028-1-Mobile port number-Debug Code to enable/disable debug on a single mobile port.
8028-2-Mobile port number-Mobile port number-Debug Code to enable/disable debug on a range
of mobile ports.
8028-*-Debug Code to enable/disable debug all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64
Debug Code is
1 for Enable
0 for Disable
Default: Disable
To program SIP Rx Gain on the Mobile Port, dial:
8041-1-Mobile-SIP Rx Gain to program the SIP Rx Gain on a single mobile port.
8041-2-Mobile-Mobile-SIP Rx Gain to program the same SIP Rx Gain on a range of mobile ports.
8041-*-SIP Rx Gain to program the same SIP Rx Gain on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
Rx and Tx Gain is 01 to 63 dB: Default: 4.
For Rx and Tx Gain at SIP codes, see the tables below.
To program SIP Tx Gain on the Mobile Port, dial:
8042-1-Mobile-SIP Tx Gain to program the SIP Tx Gain on a single mobile port.
8042-2-Mobile-Mobile-SIP Tx Gain to program the same SIP Tx Gain on a range of mobile ports.
8042-*-SIP Tx Gain to program the same SIP Tx Gain on all mobile ports.
Where,
Mobile Port Number is the number of the software port from 01 to 64.
Rx and Tx Gain is 01 to 63 dB: Default: 4.
1100
Exit SE mode.
Network Selection
After the Mobile card is successfully installed and powered on, the mobile port is programmed to automatically
locate and register with the Network that supports the SIM card installed in. Also, at each power ON, the mobile
port (SIM) will automatically register with the Network that supports the SIM on it.
However, if the Mobile port fails to register, it will restart the process of network selection on the expiry of the
Network Registration Retry Timer173.
If the ETERNITY is located in a border area where more than one Network Operator is available, it is possible that
the SIM card may register with another available network and result in 'Roaming' charges. To avoid this, you must
disable automatic network selection and program manual network selection.
When you enable manual network selection, you must program the Network Operator Priority Table. This table
requires you to program the Network Operator Codes (MCC-MNC)174 in order of priority for a Mobile Port. So,
whenever you register with the network manually, select the Network Operator that matches in order of priority. If
the Mobile port fails to register, it will restart the process of network selection on the expiry of the Network
Registration Retry Timer.
If no match is found, the Mobile port (SIM) will not get registered with any of the available network operators and no
calls can be made or received on this port.
173. The Network Registration Retry Timer defines the time for which the Mobile port, which has failed to register with the network,
should wait before attempting to re-register with the network. Network registration retry timer is 2 minutes and is non-programmable.
174. The Network Operator Code comprises of the Mobile Country Code (MCC) appended by the Mobile Network Code (MNC). The
MCC is usually a 3-digit code that identifies a country. A single country may be assigned more than one MCC. For example the
MCC assigned to India is 404, but same code applies to all network operators in the country.
The MNC is usually a 2/3-digit code. The MCC-MNC combination uniquely identifies the home network of the mobile terminal or
the mobile user. For example, AirTel, a GSM network operator in India, has different MNC assigned to its networks in various
states. The MNC for AirTel in the state of Maharashtra is 90, while the same for the state of Gujarat is 98.
1101
1102
Enter the Network Operator Codes (MCC-MNC) in order of priority. The codes must not exceed 8 digits.
You can store up to 9 Network Operator Codes in the order of priority.
Repeat the same steps to set network selection mode for other mobile ports.
When you change the Network Selection Mode to Manual and the Network Operator Code manually,
the change you made will not come into effect until you have restarted the Mobile Port.
Exit SE mode.
Wait for 4-5 seconds after the Port Status page is opened.
1103
1104
The Port status page will display the following parameters for all Mobile ports that have been enabled:
Port Name: This is the name by which the Mobile Port is programmed.
Port Status: This is the status of the connection - showing Initialization with the Network, Registering
with the Network, Idle or Busy state of the network. It also shows errors and alerts when SIM is absent,
the wrong SIM PIN has been entered, SIM PUK is required.
IMEI: This is the unique identification number of (the GSM engine) each Mobile port.
Network Operator Code: This is the MCC-MNC code of the network with which the mobile port is
registered.
Network Operator Name: This is the name of the service provider/network operator with which the
Mobile Port is registered.
SMSC Number: This is the number of the SMS Center of the network operator.
Signal Strength (dBm): This is the signal strength in '-dBm' as received from the network with which
the Mobile port is registered.
Bit Error Rate (BER): BER is Bit Error Rate which defines the quality of the channel.
Ec/Io (dB): This is the ratio of the received energy per chip (= code bit) and the interference level,
given in dB. In case no true interference is present, the interference level is equal to the noise level.
This parameter is significant only when the GSM engine is registered with the 3G network.
The range of this parameter can be from 0 to 63dB. A ratio of 10dB to 14dB is normal, numbers going
higher than that is progressively worse.
Call Duration: This is the total call duration of matured outgoing calls175 on the Mobile port. This data
is used for calculating Answer Seizure Ratio (ASR) for the port. It is displayed in MMMMMM:SS format.
Dialed Calls: This is the total number of outgoing calls176 made from the Mobile port. This data is used
for calculating Answer Seizure Ratio (ASR) for the port.
Successful Calls: This is the total number of matured outgoing calls made from the Mobile port. This
data is used for calculating Answer Seizure Ratio (ASR) and Average Call Duration for the port.
ASR % : This is the Answer Seizure Ratio (ASR) calculated by the system for the Mobile port, in terms
of percentage. ASR is the sum of all outgoing matured calls from the Mobile port, divided by the total
number of outgoing calls made from the Mobile port, multiplied by 100. The system calculates ASR
after the completion of the outgoing call.
ACD: This is the Average Call Duration (ACD) of outgoing calls made from the Mobile port. It is an
indicator for monitoring the network condition. Decreasing ACD is indicative of trouble in the network
condition.
The system calculates ACD after the completion of the outgoing calls, by dividing the total call duration
by the number of outgoing matured calls.
Reset ASR and ACD: This field allows the System Engineer to manually the ASR and the ACD of the
Mobile port.
The parameters Total Call duration, Number of matured calls, Total Number of OG Calls, ASR and
ACD are saved in the configuration, and are not on Power OFF condition. The system maintains the
statistics for the last 999 calls. When the total number of outgoing calls exceeds 999, the system will
stop calculating ACD and ASR and will display ASR and ACD calculated on the basis of the last 999
calls only.
Therefore, the System Engineer must manually ASR and ACD when the total number of calls reaches
999. When you ASR and ACD the number of call matured and the number of calls dialled is to 0.
ASR and ACD can be anytime, even when the total number of calls is less than 999.
When ACD is , only the 'Total Call Duration' maintained for the ACD calculation will be . The 'Total Call
Duration' of the Call Budget, the consumed minutes maintained for the Call Budget on the mobile port will
remain unaffected.
SIM ID: This is the Integrated Circuit Card ID (ICC-ID) of the SIM Card inserted in the Mobile port. Each
SIM is internationally identified by its ICC-ID. ICC-IDs are stored in the SIM Card and are also printed
on the SIM card body.
IMSI: International Mobile Subscriber Identity (IMSI) is a unique number stored in the SIM card.
175. Matured calls are outgoing calls for which 'CONNECT' message was received from the network.
176. The total number of outgoing calls made includes the number of times the ATD has been sent from the Mobile port to the network.
1105
Cell ID: This is the 16-bit identifier that identifies the cell. The cell is the radio coverage area given by
one BTS (Base Transceiver Station).
Location Area Code (LAC): The LA (Location Area) is a group of cells defined by the Operator. The
LAC (Location Area Code) uniquely identifies a LA within a PLMN (Public Land Mobile Network).
Call Budget Type: This shows the Call Budget Type, whether Amount, Minutes or Number of Calls,
are set on the Mobile port.
Allotted Amount (Rs.) / Minutes/Calls: This shows the sum/number of minutes/number of calls
allotted as Call Budget on the Mobile port.
Consumed Amount (Rs.) / Minutes/Calls: This shows the sum/number of minutes/number of calls of
the allotted Call Budget that has been used up on the Mobile port.
Call Budget Reset Schedule (Date): This shows whether the consumed Call Budget on the Mobile
port is to be Daily or on a particular date of a month.
Reset Consumed Amount/Minutes/Calls: This editable field allows the you to reset the consumed
Call Budget Amount/Minutes/Calls at any time, manually.
SMS Budget Scheduled Reset: This shows if you have enabled Scheduled Reset for the SMS budget.
SMS Budget Reset Schedule (Date): This shows the date on which the SMS Budget will be reset.
SMS Budget Consumed SMS (Total): This shows the number of SMS of the allotted SMS Budget that
has been used up on the Mobile port.
Consumed SMS(Daily): This shows the number of SMS of the allotted Daily SMS Budget that has
been used up on the Mobile port.
Reset Consumed SMS (Total) Budget : This editable field allows the you to reset the consumed SMS
Budget, manually. It will reset both Consumed SMS (Total) and Consumed SMS (Daily).
The Consumed SMS Budget can be reset from the System Engineer mode as well as the System
Administrator mode manually at any time or on a scheduled date from the System Administrator mode
only. Refer Resetting SMS Budget.
Registered with Network: This shows the type of network with which the Mobile port is registered,
whether GSM, GSM Compact, 3G or UMTS.
Firmware Version of Engine: This shows the firmware version of the mobile Engine.
You can also view the Mobile Port Status from the Status link. To view, click the Mobile link under Status.
.
1106
To reset the SMS Budget from the SA mode, follow the steps given below:
Consumed SMS (Total): This shows the number of SMS of the allotted SMS Budget that has been
used up on the Mobile port.
Consumed SMS (Daily): This shows the number of SMS of the allotted Daily SMS Budget that has
been used up on the Mobile port.
Scheduled reset consumed SMS (Total) budget: Enable this check box if you want only the
Consumed SMS (Total) Budget to be reset on a particular date of every month.
Budget Reset Schedule (Date): Select the date of the month (Daily or 1-31) on which you want the
SMS Budget to be reset every month.
Reset Consumed SMS (Total) Budget : Enable this checkbox to reset the consumed SMS Budget,
manually. It will reset both Consumed SMS (Total) and Consumed SMS (Daily).
Exit SE mode.
The System Administrator can also
view the ID of the Mobile Network Operator with which the Mobile Port is currently registered.
check Signal Strength of a mobile port, whenever there is trouble placing calls over a Mobile port to rule
out weak signal as the cause.
To do this,
1107
When the network responds, you the Mobile Network Signal Strength will be displayed on the LCD of
your DKP and programming beeps will be played.
The Signal Strength values are in -dBm. '-113' indicates weak signal, whereas '-51' indicates maximum
signal strength.
1108
The Radio Port transmits the audio signal to the Radio device and the extension user gets Ring Back Tone
for 3 seconds.
Since the Radio Port is always in the receiving mode, the system connects the speech path between the
Radio device user (connected to the Radio Port) and the extension user.
The extension user (caller) can now converse with the Radio device user.
When the Radio device user wants to talk, s/he must press the PTT button of the Radio device and talk.
The extension user can now listen to the Radio device user.
If the Radio port is busy, the extension user will hear the Busy Tone.
1109
The Radio extension user (3000) dials another Radio extension user (3001).
Audio signal is detected on the Radio port (3001) and the Radio extension user (3000) gets Ring Back
Tone for 3 seconds.
Since the Radio Port (3001) is always in the receiving mode, the system connects the speech path
between both the Radio extension users.
Both users must press the PTT button of their Radio device to talk. At a time only one user can speak.
If the system detects silence for one minute, it disconects the call.
If the Radio port is busy, the extension user will hear the Busy Tone.
You must configure the Radio parameters so as to use the Radio functionality in the ETERNTY. You can configure
Radio parameters only through Jeeves.
Enable Port: This flag is for enabling or disabling a Radio port. When a Radio port is disabled, neither
incoming nor outgoing calls can be made from that port.
By default, the port is enabled. Clear the check box to disable the ports.
1110
Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the Radio Card has been
inserted. 'Port' is the number of the hardware port on the card to which the Radio device is connected.
By default the ETERNITY can detect and assign the hardware slot and port numbers automatically to
the Radio (software) ports. However, you may change the Hardware Slot and Port assigned to the
Radio software port. If required, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, enter '00' in both fields.
Access Code: Assign Station Access Codes to the Radio Port. Station Access Codes are commonly
referred to as Extension Numbers. These may be number strings of a maximum 6 digits, which are to
be dialed to call the Radio port to which they are assigned.
To assign Station Access Codes according to your preference and requirment to a range of Radio
Ports, see Assigning Access Codes to a Range of Extensions.
By default, the Station Access Codes are blank for all Radio ports.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with
any other Access Code in the 'Dial' phase. Refer the topics Access Codes and Conflict Dialing to
know more.
Name: Assign a 'Name' to the Radio port. The name may be of the person who will use the Radio
device or the name of the department or location of the device. This name will be displayed on the LCD
of the Operator/extension user's phone, if it is equipped with Caller ID.
You can program a name of maximum 18 alphanumeric characters.
On Detecting Voice: Select Route to Operatror, to route incoming calls on the Radio Port to the
Operator.
Select Greet to Dial, if you want the callers to dial the desired extension number. The call will be routed
to the dialed number. This option must be selected only when you have a Dial Pad connected to the
Radio Deivce. You can customise the message played to the callers, if required. For detailed
instructions, see Voice Message Applications.
Station Basic Feature Template: As the Radio Port functions as a station, assign a Station Basic
Feature Template to the Radio port.
Only the following features of the Station Basic Feature Template are applied on the Radio Port:
Time Table
Operator
Class of Service (Only Global Directory Part 1, 2 and 3, Basic Features and PRivacy from Built-In
Auto Attendant)
Call Budget
Call Privilege
Toll Control-Call Budget consumed
Outgoing Trunk Bundle Group (WH, BH and NH)
Store Outgoing Calls
Store Incoming Calls
By default, Station Basic Feature Template number 01 is also applied on Radio ports.
1111
Check if the default settings of the features applied on the Radio ports (Time Table, Operator, Class of
Service, Storage of Outgoing and Incoming Calls etc) match your requirements for the Radio ports. If
yes, retain the default Station Basic Feature Template 01.
If you want to change any of the feature settings in for the Radio Ports, you may prepare a different
Template177, for example, Template 14 and apply it on the Radio Ports.
Also, if different feature settings are to be applied on different Radio Ports prepare separate Station
Basic Feature Templates and apply them on the ports. To do this,
Click the Station Basic Feature Template link to open the page.
Customize the Radio Port related features (listed above) in Template number 14 and click Submit
to save.
Enter the number of the Template you customized, Template 14 in the 'Station Basic Feature
Template' field of the Radio Port, for example: Port 1, on which you want to apply this template. If
you want to apply this template to other ports too, like 2, 3, and 4, assign the Template 14 to all
these ports.
If required, repeat the same steps to customize and assign a different Template to another Radio
port.
Refer the topic Station Basic Feature Template to know more about customizing the templates and
applying on the ports.
Station Advance Feature Template: Assign a Station Advanced Feature Template to the Magneto
Port.
Only the following features in Station Advance Feature Templates are applied on Magneto Ports.
DDI IC Routing
Send DDI Number as CLI?
Internal Calls Storage
CDC Table
Call Tapping
Caller Category
By default Station Advanced Feature Template 01 is assigned to all the Radio ports. Check if the
default settings of the features applied on the Radio ports (Internal Call Storage flag and Call Forward
Ring Timer) fulfill the feature requirements of the Radio ports, by opening the 'Station Advanced
Feature Template' link.
If the default template fulfills your requirement, retain the default Station Basic Feature Template 01 for
the Radio ports.
177. This is recommended because changing the values of the default Template will be applied on all other extension types to which the
Template is assigned.
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However, if you want to change any of the feature settings in for the Radio Ports, you may prepare a
different Template178, for example, Template 05 and apply it on the Radio Ports. To do this,
Click the Station Advanced Feature Template link to open the page.
Enter the number of the Template you customized, Template 05 in the 'Station Advanced Feature
Template' field of the Radio Port, for example, 1, on which you want to apply this template. If you
want to apply this template to other terminals too, like 2, 3, and 4, assign the Template 05 to all
these ports.
Repeat the same steps to customize and assign a different Template to another Radio Port.
Also refer the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the station ports.
Priority: Select a Priority Level for the Radio Port from 1, 2, 3... to 9, with '1' being lowest Priority and '9'
being highest Priority. Whenever an extension (Radio Port) with higher priority calls an extension with
lower priority, a triple ring is placed on the called extension. To know more, read the feature description
Priority.
By default, the Priority of all Radio ports is set to '5-Normal'.
Advanced Configuration
The above listed parameters fulfill the basic Radio extension port configuration requirements of most users.
However, it is anticipated that some users may need to configure the more advanced features like Radio VAD
178. This is recommended because changing the values of the default Template will be applied on all other extension types to which the
Template is assigned.
1113
Threshold Level, Minimum Silence Time for valid IC call (msec), Minimum Speech Time to activate PTT (msec) etc.
For such users, you may click the 'Advanced' button.
Scroll with the horizontal bar, and configure the following advanced Radio Parameters:
Radio VAD Threshold Level: This parameter defines the level below which the Radio port would not
validate the audio signal as valid speech packet. Set the Radio VAD Threshold Level to the desired
value. The range of this level is 0 to -96 dBm. By default, it is -15 dBm.
Minimum Silence Time for valid IC call (msec):Set the duration for which the system may detect and
accept silence before actual speech packets, to treat this silence as a valid incoming call. The range is
100 to 9999 msec. By default, it is 1200 msec.
Minimum Speech Time for valid IC call (msec): Set the duration for which the system must receive
speech packets for a valid incoming call. The range is 100 to 9999 msec. By default, it is 500 msec.
All the above parameters will be taken into account to determine an incoming call on the port. Once the call
matures, these parameters shall not be checked.
1114
Minimum Speech Time to activate PTT (msec): Set this timer to ensure continuous communication,
once the call matures. As per the set time, the PTT will be activated and speech will be transmitted.
The range is 100 to 9999 msec. By default, it is 700 msec.
Minimum Silence Time to deactivate PTT (msec): Set the duration of silence after which the system
must deactivate PTT to disconnect the call. The range is 100 to 9999 msec. By default, it is 400 msec.
1115
When your VoIP Card is installed in a Private Network, you may have to change the IP Address and
Subnet Mask of the WAN Port of the card, before connecting it to the LAN Switch/Hub. However, this will
not be necessary, if there is a DHCP server on the LAN which will automatically assign an IP Address that
does not conflict with any other device on the LAN.
Depending on your installation scenario, configure the VoIP Port Parameters using Jeeves or dialing commands
from a Telephone.
1116
Select the VoIP Port number you want to configure by clicking the respective tab, and program the
following port parameters:
Hardware Slot: This is the number of the Universal Slot in which the VoIP card has been inserted. The
ETERNITY can automatically detect and assign the Hardware Slot numbers to the VoIP card (LAN and
WAN Port). If required, you may change the Hardware Slot number.
Name: You may assign a 'Name' to each VoIP Port to facilitate easy identification. The Name may
comprise a maximum of 18 characters.
LAN Port
MAC Address: This non-editable field shows the MAC Address of the LAN port.
IP Address: Enter the IP Address to be assigned to the LAN Port of the VoIP Card. The default IP
Address is 192.168.002.031. You can assign only Static IP to the LAN Port.
Subnet Mask: Enter the Subnet Mask to be assigned to the LAN Port. The default Subnet Mask is
255.255.255.0
1117
WAN Port
MAC Address: This non-editable field displays the MAC Address of the WAN port.
Use MAC Cloning: MAC Cloning is required when you want the WAN Port to use a MAC Address other
than its own unique MAC Address as source MAC Address.
When MAC Address Cloning is disabled, the WAN Port will use its unique MAC Address as the source
MAC Address on all Ethernet Frames. When MAC Cloning is enabled, the WAN Port will use the cloned
MAC Address on all Ethernet frames.
Select the check-box to enable cloning of the MAC Address of the WAN Port. By default, MAC Address
Cloning is disabled.
Clone MAC Address: If you have enabled MAC Cloning, enter the MAC Address to be cloned in this field.
Connection Type: Select the appropriate Connection Type for the WAN port, according to the IP
Addressing scheme of your installation scenario. Consult your LAN Administrator also in this regard.
Static: Select this option if the connection type is Static. This is also the default connection type for all
WAN ports.
When you select this option, you must
assign an IP Address to the WAN Port.
change the Subnet mask of the WAN Port as appropriate.
program the Router's LAN Interface IP Address as the Gateway IP Address.
program the DNS Address/Domain Name provided by your ISP or ask your LAN Administrator for
the DNS Address and Domain Name.
1118
DHCP: Select this option if the connection type DHCP. As the DHCP Server will automatically assign IP
Address, Subnet Mask, Gateway Address to the WAN Port, you need not configure any of these.
PPPoE: Select this option if the connection type is PPPoE. As the PPPoE server will automatically
assign the IP Address, Subnet Mask and Gateway Address to the WAN Port, you need not change any
of these. You must program the User ID, Password and PPPoE Service Name as provided by your ISP.
Program the Service Name only if it has been provided. You must set DNS address.
PPPoE: The parameter is relevant if you have selected PPPoE as the Connection Type. Configure the
following PPPoE parameters:
User ID: Enter the User ID provided by the Internet Service Provider. The User ID may be a maximum
of 64 characters.
Password: Enter the User Password provided by the Internet Service Provider. The password may be
a maximum of 64 characters.
Service Name: Enter the PPPoE Service Name, if provided by your Internet Service Provider. The
Service Name may consist of a maximum of 64 characters. If Service Name is not required, leave this
field blank.
IP Address: You must enter the IP Address, only if you selected 'Static' as the Connection Type.
If you configure the PPPoE User ID, Password and Service Name using commands, you can configure a
maximum of 16 characters only.
If you selected DHCP or PPPoE as the Connection Type, the IP Address assigned by the DHCP/PPPoE
server will be displayed here.
Subnet Mask: You must enter the Subnet Mask, only if you selected 'Static' as the Connection Type.
If you selected DHCP/PPPoE as the Connection Type, the Subnet Mask assigned by the DHCP/PPPoE
server will be displayed here.
Default Gateway: You must enter the Gateway IP Address, only if you selected 'Static' as the Connection
Type.
If you selected DHCP/PPPoE as the Connection Type, the Gateway IP Address assigned by the DHCP/
PPPoE server will appear here in non-editable format.
Domain Name Server (DNS): Configure the following DNS Connection settings for the WAN Port:
DNS Address Assignment: If you selected 'Static' as your network Connection Type (IP Addressing),
you can select only 'Static' as the DNS Address Assignment.
If you selected DHCP as your network Connection Type, and the DHCP server provides DNS Address,
set the DNS Address Assignment to 'Automatic'. If the DHCP server does not provide DNS Address,
set DNS Address Assignment as 'Static' and program the DNS Server Address/DNS Name provided by
your ISP.
If you selected PPPoE as your network Connection Type, and the PPPoE server provides DNS
Address, set the DNS Address Assignment as 'Automatic'. If the PPPoE server does not provide DNS
Address, set the DNS Address Assignment as 'Static' and program the DNS Server Address/DNS
Name provided by your ISP.
DNS Address: This field will be editable only if you selected DNS Address Assignment as 'Static'.
Enter the DNS IP Address here. The DNS Address can be a maximum of 15 characters.
If you selected DNS Address Assignment as 'Automatic', the DNS Address assigned by the DHCP/
PPPoE server will appear here.
DNS Domain Name: Program DNS Domain Name if provided by your ITSP/LAN Administrator.
Otherwise, keep this field blank. The Domain Name may be a maximum of 40 characters.
Dynamic DNS (DynDNS.org): This parameter is applicable only when you are going to configure SIP
Extensions on the VoIP Card.
When the VoIP Card is assigned dynamic IP Address using DHCP or PPPoE, SIP-enabled devices
registered with the Card as SIP Extensions need to change their configuration whenever a new IP Address
is assigned to the VoIP Card. Dynamic DNS resolves this.
ETERNITY VoIP Card supports Dynamic DNS Server client of the Service Provider Dynamic DNS.org. If
you want to use the DNS Service of DynDNS.org, program these parameters:
Enable Dynamic DNS: If you have taken the services of DynDNS.org, you must enable this flag. By
default this flag is disabled.
1119
Update IP Address at Power ON: When your VoIP Card is registered with the DynDNS.org, the
DynamicDNS server stores the mapping between hostname and IP Address, which can be updated
periodically. However, if the VoIP card frequently sends IP Address update request to the DDNS server,
the server is likely to block the hostname in its database and terminate the DDNS services provided to
you.
So, if you restart the ETERNITY frequently, there is a great chance that DDNS server will block the
hostname programmed in the system. This will in turn affect the ability of the system to receive the calls
using DDNS host name since the entry (mapping between host name and IP Address) in the DNS
server will be deleted during those scenarios.
The VoIP Card offers you control over whether the system should update the IP Address in the DDNS
server at each Power ON or not. It also allows you to update the IP Address at any time, as required.
If you do not want to update the IP Address in the DDNS Server at each Power ON, set this flag to 'No'.
By default, the flag is set to 'No'.
User ID: Enter the User ID created by you with DynDNS.org here. A maximum of 40 characters,
including all ASCII characters are allowed.
Password: Enter the Password created by you for your User ID with DynDNS.org here. The password
may be not more than 24-characters long.
Host Name: Enter the Host Name created by you with DynDNS.org here. A maximum of 40
characters. All ASCII characters except < > and (double quote) are allowed.
Retry Trials: This count defines the number of attempts that the VoIP Card should make to send the IP
Address Update Request to the Dynamic DNS Server. The Retry Count may be set from 1 to 9. By
default the count is set to 1.
Update IP Address Now?: Enable this flag whenever you want to update the IP Address in the DDNS
server. By default, this flag is disabled.
You can use this flag to update the IP Address in the DDNS server, if you have disabled Update IP
Address during each Power ON.
VoIP Server Domain: This parameter is of relevance only if you are configuring SIP Extensions on the
VoIP Card.
The VoIP Card is capable of maintaining a domain for registering SIP clients (any SIP-enabled device) as
SIP Extensions.
Program the Server Domain if you want SIP clients to register with the Registrar Server of the VoIP card
using the domain handled by the VoIP Card179.
179. SIP clients can be registered with the VoIP Card either using the domain handled by the VoIP Card or using the WAN or LAN Port
IP Address.
If domain is programmed, VoIP card will listen for the SIP message which is redirected to the programmed domain only. It will also
listen for SIP messages on the WAN IP address and LAN IP address.
But if domain is not programmed, the VoIP card will listen for SIP messages only on the WAN IP Address and LAN IP address.
1120
If you program Server Domain for registration of SIP clients, you must also map the Domain name and the
IP Address of the WAN Port of the VoIP Card to the DNS Server in the network.
Advanced Configuration
The above listed parameters fulfill the basic VoIP Port configuration requirements of most users. To configure other
parameters such as Quality of Service (QoS), NAT, STUN, SIP UDP/TCP Port, RTP Listening Port, VLAN, MAC
Cloning, click the Advance button at the bottom.
Quality of Service (QoS): This refers to priority of IP packets on network layer. QoS is programmed for
both signaling (SIP) and media (RTP). following types of QoS can be configured:
SIP DiffServe/ToS: The VoIP Card sends all the SIP signaling messages with this QoS setting which
is selected here. This field defines the priority bits for SIP messages. The Valid DiffServe range is from
00-63, default: 26
RTP DiffServe/ToS: The VoIP Card sends all the RTP packets with the QoS setting which is selected
here. This field defines the priority bits for RTP packet. The Valid DiffServe range is from 00-63, default:
46
1121
QoS parameters are applicable for all packets (SIP/ RTP) leaving both LAN and WAN port as well as TCP
connection.
Simple Traversal of UDPs through NATs (STUN):This parameter is to be configured only if the VoIP
Port (WAN) ETERNITY is located behind a NAT Router and SIP Messages need to be forwarded to the
public internet.
Simple Traversal of UDP through NAT (STUN) specifies the mechanism required for NAT traversal in SIP
messages. The STUN Server facilitates traversing through most NATs, except symmetric NATs. If your
router has symmetric NAT, do not program this parameter. If your router as asymmetric NAT, configure the
following STUN parameters:
STUN Server Address: Enter the STUN Server Address, a maximum of 40 characters.
STUN Port: Enter the Listening Port of the STUN Server. The valid range for this field is from 102465535. The default STUN Port is 03478.
Use SIP Port fetched using STUN: This flag is enabled by default to allow SIP Port Number to be
fetched using STUN in the SIP message. Disable this flag if you are using Port-Forwarding in the
Router for SIP messages.
You also need to select 'Use IP Address fetched using STUN'' as the 'Source Port IP Address' in the 'SIP
Extension General Parameters', when you configure SIP Extensions'. Refer the topic "Configuring SIP
Extensions".
Since STUN does not work with symmetric NAT, as an alternative to STUN you can use the Router's Public
IP Address as NAT Traversal mechanism. Ask your Network Administrator about the NAT Traversal
mechanism that suits best for your voice network and program this parameter.
Router's Public IP Address: This parameter is of relevance if the VoIP Port (WAN) of the ETERNITY is
located behind a NAT Router and SIP Messages are to be forwarded to the public internet.
Router's Public IP Address specifies the fixed IP Address of your NAT router required for NAT Traversal in
SIP messages.
You also need to select 'Router's Public IP Address' as the 'Source Port IP Address' in the 'SIP Extension
General Parameters', when you configure SIP Extensions'. Refer the topic "Configuring SIP Extensions".
You can also use STUN as an alternative to the Router's Public IP Address as NAT Traversal mechanism.
Ask your Network Administrator about the NAT Traversal mechanism that suits best for your voice network
and program this parameter.
Disconnect on Silence Detection: With this parameter, any matured incoming or outgoing call can be
disconnected automatically, if silence (No RTP Packets) is detected for more than a specified duration of
time.
1122
Enable Disconnection on Silence Detection?: If you want to use this feature, click the check box to
enable it. By default, Disconnect on Silence Detection is enabled.
Detection Time (sec): This Timer defines the duration for which, if silence is detected continuously,
the call will be disconnected. The valid range of this Timer is from 001 to 999 seconds. By default, it is
set to 999 seconds.
Channels Reserved for SIP Trunks: The VoIP Card supports up to 32 voice channels (depending on the
model of ETERNITY), which can be used by SIP Extensions and SIP trunks.
It may happen that SIP Extension users use up most of the channels of the VoIP card, leaving too few or
none for making/receiving SIP Trunk calls.
This can be avoided by reserving some voice channels exclusively for SIP trunk calls.
Specify the minimum number of voice channels you want to reserve for SIP Trunk calls. By default, no
channel is reserved.
SIP 100rel: This parameter is to be configured if you want to support reliable transmission of (SIP)
provisional responses. Enable 100rel by selecting the check box, if you want the VoIP Port to use 100rel
for reliable transmission of SIP provisional responses and to use PRACK (Provisional Acknowledgement).
By default, the flag is disabled.
SIP Over TCP: ETERNITY supports transporting of SIP messages over User Datagram Protocol (UDP),
Transfer Control Protocol (TCP) as well as Transport Layer Security connection (TLS) Despite the
advantages that SIP over TCP and SIP over TLS offer, it is more common to use UDP to transport SIP
messages.
By default, SIP Over TCP is enabled. To be able to send SIP messages over TCP, you must configure
TCP as the Default Transport for Outgoing Messages on the SIP Trunk Parameters page.
If you do not want to transport SIP messages using TCP, clear the SIP Over TCP check box.
SIP Over TLS: ETERNITY supports transporting of SIP messages over Transport Layer Security. TLS
offers secure SIP signaling.
By default, SIP Over TLS is enabled. To be able to send SIP messages over TLS, you must configure
TLS as the Default Transport for Outgoing Messages on the SIP Trunk Parameters page.
If you do not want SIP messages to be transported using TLS, clear the SIP Over TLS check box.
SIP UDP Port: This port defines the port on which the VoIP Port of ETERNITY listens for SIP messages
transported over UDP. This port is also used as the source port for sending SIP messages to the remote
peer. The valid range for this port is 1025-65535. Default: 05060.
SIP TCP Port: This port defines the port on which the VoIP Port of ETERNITY listens for SIP messages
transported over TCP. This port is also used as the source port for sending SIP messages to the remote
peer. The valid range for this port is 1025-65535. The default SIP TCP Port is 05060.
SIP TLS Port: This port defines the port on which the VoIP Port of ETERNITY listens for SIP messages
transported over TLS. This port is also used as the source port for sending SIP messages to the remote
peer. The valid range for this port is 1025-65535. The default SIP TLS Port is 05061.
RTP Listening Port: This port defines the port on which the VoIP Port of ETERNITY listens for RTP
Packets. This port is also used as the source port for sending RTP packets to the remote peer. The valid
range for this port is 1025-65278. The default RTP Listening Port is 08000.
1123
Layer 2 VLAN/CoS: This parameter is to be configured if the VoIP Port (WAN) of ETERNITY is to be
connected in VLAN network.
This parameter enables the ETERNITY to add VLAN header to the packets generated by it. The VLAN
header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of
traffic180.
VLAN Tag is applied on all packets generated by system (SIP, RTP, DNS, ARP, etc.), whereas CoS bits
are applied only for SIP and RTP packets generated by system.
The corresponding meaning of CoS bits with respect to traffic type is as follows:
COS
Traffic Type
Best Effort
Background
Spare
Excellent Effort
Controlled Load
Video
Voice
Network Control
Enable Layer 2 VLAN/CoS: When this flag is enabled, all packets generated by the system (SIP, RTP,
DNS, ARP, etc.) will be tagged with VLAN ID as programmed. The CoS bits as programmed for SIP
and RTP packets will be included in the VLAN header. By default, this flag is disabled.
VLAN ID: Consult your network administrator and program the VLAN ID. The valid range for this is
from 0 - 4094. Default: 1.
SIP CoS: Define the CoS (priority) bits in all SIP packets. The range of CoS bits is from 0 to 7. Default:
3.
RTP CoS: Define the CoS (priority) bits in all RTP packets. The range of CoS bits is from 0 to 7.
Default: 6.
UDP NAT Keep Alive: This parameter is to be configured when the VoIP Port is connected behind a NAT
router181 and SIP messages are transported over UDP. UDP NAT Keep Alive messages must be sent to
refresh the UDP binding in the NAT router.
Enable UDP NAT Keep Alive: Enable this flag to send UDP NAT Keep Alive messages periodically to
refresh the binding in the NAT router. By default, this flag is disabled.
180. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), better handling
of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred to
as Class of Service (CoS) which is defined by IEEE 802.1P.
181. Network Address Traversal (NAT) allows multiple hosts in the network to share the single public routable IP address. Means all the
hosts in the private network shall be identified by single public IP address in the global IP cloud.
1124
Interval (sec): Select Time period after which the VoIP Port should send UDP NAT Keep Alive
messages. This time period should be less than the UDP Binding Timer of the router. The valid range is
001-999 seconds. By default it is set to 180 seconds.
Type of Message: Select the type of message type to be sent when UDP NAT Keep Alive is enabled.
Select either REGISTER or NOTIFY. By default, NOTIFY is selected.
TCP NAT Keep Alive: This parameter is to configured when the VoIP (WAN) Port is connected behind a
NAT router and SIP messages are transported over TCP. TCP NAT Keep Alive messages must be sent to
refresh the TCP binding in the NAT router.
Enable TCP NAT Keep Alive: Enable this flag to send TCP NAT Keep Alive messages periodically to
refresh the binding in the NAT router. By default, this flag is disabled.
Interval: Select Time period after which the VoIP Port should send TCP NAT Keep Alive messages.
This time period should be less than the Binding Timer of the router. The valid range is 0001-9999
seconds. By default it is set to 120 seconds.
SIP Invite Timer (sec): This is the time in seconds that the VoIP Port waits for a response from the called
party after ending INVITE message. This timer starts after sending INVITE message to the called party
and stops on receipt of the provisional response or the final response or when the user disconnects the
call. On expiry of the timer, the call process is terminated by the ETERNITY and an error tone is played to
the user. The range of the SIP Invite Timer is 010-180 seconds. By default it is set to 30 seconds.
SIP Provisional Timer (sec): This is the time in seconds that the VoIP Port waits for the final response
after receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called party
or when the user disconnects the call. On expiry of the timer, the ETERNITY terminates the call process
and plays error tone to the user. The range of SIP provisional Timer is 010-180. By default, the timer is set
to 60 seconds.
General Request Timer (sec): The time in seconds for which the VoIP Port waits for response for a
transaction request. This timer starts on the initiation of a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the VoIP Port clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. By default the timer
is set to 20 seconds.
Assign LED for SIP Trunk: You can monitor the status and functioning of any one of the SIP trunks with
this parameter. This is LED2 on the VoIP card, which is assigned to any one of the SIP trunks that you
want to monitor. When assigned to a SIP trunk, the LED indicates the functioning of this trunk. By default
the LED is assigned to SIP trunk 01. Enter the number of the SIP trunk you want to be monitored here.
If you have completed configuring the required parameters for the VoIP Ports, click 'Submit' at the bottom
of the page to save your VoIP Port settings.
To program the parameters or another port, click the VoIP Port number on the top of the screen. Repeat
the same steps as above. Click Submit at the bottom of the page to save your port parameter settings.
1125
1126
0 for Disable
Default: Disable
To configure Cloned MAC Address on WAN Port, dial:
7774-1-VoIP Port-Cloned MAC Address
Where,
VoIP Port is the number of the software port from 01 to 16.
Cloned MAC Address is a string of a maximum of 16 characters in hexadecimal format, for example:
00:50:C2:55:B0:10. Only Hexadecimal characters (0-9, A-F, '.')
Repeat this command to program the Cloned MAC Address on another VoIP Port.
When you enter the number string of the Cloned MAC Address, alphanumeric dialing will be automatically
enabled on the DKP. Do not enter '.' (dot/period) in the MAC Address. For example, to program MAC
Address 00.50.C2.55.B0.10, enter 0050C255B010 from the DKP.
To prevent the possible collision and loss of data, do not clone the MAC address, which is already used by
another device in the same network. Also, use MAC address as per IANA (Internet Assigned Numbers
Authority) standards.
To select the Connection Type for the VoIP Port, dial:
7751-1-VoIP Port-Connection Type to select connection type for a single VoIP port.
7751-2-VoIP Port-VoIP Port-Connection Type to select the same connection type for a range of VoIP
ports.
7751-*-Connection Type to select the same connection type for all ports.
Where,
VoIP Port is the number of the software port from 01 to 16
Connection Type is
1 for DHCP
2 for PPPoE
3 for Static IP
Default: Static IP.
To configure PPPoE User ID for PPPoE Connection Type, dial:
7752-1-VoIP Port-PPPoE User ID to program User ID for a single VoIP port.
7752-2-VoIP Port-VoIP Port-PPPoE User ID to program the same User ID for multiple ports.
7752-*-PPPoE User ID to program the same User ID for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16
PPPoE User ID is a string of maximum 16 characters with extended ASCII character set.
To configure PPPoE Password for PPPoE Connection Type, dial:
7753-1-VoIP Port-PPPoE Password to program password for a single VoIP port.
7753-2-VoIP Port-VoIP Port-PPPoE Password to program the same password for a range of VoIP
ports.
7753-*-PPPoE Password to program the same password for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16
PPPoE Password is a string of maximum 16 characters, with extended ASCII character set.
To configure the Service Name for a PPPoE Connection Type, dial:
7787-1-VoIP Port-PPPoE Service Name to program service name for a single VoIP port.
1127
7787-2-VoIP Port-VoIP Port-PPPoE Service Name to program the same service name for a range of
VoIP ports.
7787-*-PPPoE Service Name to program the same service name for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16
PPPoE Service Name is a string of maximum 16 characters, with extended ASCII character set.
To configure IP Address for VoIP Port having 'Static' as Connection Type, dial:
7754-1-VoIP Port-IP Address
Where,
VoIP Port is the number of the software port from 01 to 16
IP Address is from 000-255.
Enter each octet in full. For example: To program IP address 192.168.10.11, enter 192168010011.
To configure Subnet Mask for the VoIP Port having 'Static' as Connection Type, dial:
7755-1-VoIP Port-Subnet Mask to program the Subnet of a single VoIP port.
7755-2-VoIP Port-VoIP Port-Subnet Mask to program the same Subnet for a range of VoIP ports.
7755-*-Subnet Mask to program the same Subnet for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16
Subnet Mask is from 000-255. Enter each octet in full. For example to program Subnet Mask
255.255.255.0, enter 255255255000.
To configure Default Gateway IP Address for the VoIP Port having 'Static' as Connection Type, dial:
7756-1-VoIP Port-Gateway Address to program Gateway IP Address for a single VoIP port.
7756-2-VoIP Port-VoIP Port-Gateway Address to program the same Gateway IP Address for a range
of VoIP ports.
7756-*-Gateway Address to program Gateway IP Address for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16
Gateway IP Address is from 000-255. Enter each octet in full. For example: Enter the Gateway Address
192.168.10.10 as 192168010010.
1128
Where,
VoIP Port is the number of the software port from 01 to 16.
DNS Server Address is a maximum of 15 characters, with extended ASCII character set.
Dynamic DNS
To enable/disable Dynamic DNS flag, dial:
7823-1-VoIP Port-Flag to program the flag of a single VoIP port.
7823-2-VoIP Port-VoIP Port-Flag to program the same flag for a range of VoIP ports.
7823-*-Flag to program the same flag for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
Flag is
0 for disable
1 for enable
Default: disabled.
To enable/disable Update IP Address at Power ON flag, dial:
7824-1-VoIP Port-Flag to program the flag of a single VoIP port.
7824-2-VoIP Port-VoIP Port-Flag to program the same flag for a range of VoIP ports.
7824-*-Flag to program the same flag for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
Flag is
0 for disable
1 for enable
Default: disabled.
To program User ID for Dynamic DNS, dial:
7825-1-VoIP Port-User ID for Dynamic DNS.
Where,
VoIP Port is the number of the software port from 01 to 16.
User ID for Dynamic DNS is maximum 40 characters.
To program Password for Dynamic DNS, dial:
7826-1-VoIP Port-Password for Dynamic DNS
Where,
VoIP Port is the number of the software port from 01 to 16.
Password for Dynamic DNS is maximum 24 characters.
To program Host Name for Dynamic DNS, dial:
7827-1-VoIP Port-Host Name for Dynamic DNS
1129
Where,
VoIP Port is the number of the software port from 01 to 16.
Host Name for Dynamic DNS is maximum 40 characters.
To program Retry Trials for Dynamic DNS, dial:
7828-1-VoIP Port-Retry Trial for Dynamic DNS
Where,
VoIP Port is the number of the software port from 01 to 16.
Retry Trial for Dynamic DNS is from 1 to 9.
Default: 1.
To enable/disable Update IP Address Now? flag, dial:
7829-1-VoIP Port-Flag to program the flag of a single VoIP port.
7829-2-VoIP Port-VoIP Port-Flag to program the same flag for a range of VoIP ports.
7829-*-Flag to program the same flag for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
Flag is
0 for disable
1 for enable
Default: disabled.
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1131
SIP 100rel
To enable/disable Use 100rel Response for a VoIP Port, dial:
7783-1-VoIP Port-Code to enable/disable 100rel Response for a single VoIP port.
7783-2-VoIP Port-VoIP Port Code to enable/disable 100rel Response for a range of VoIP ports.
7783-*-Code to enable/disable 100rel Response for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
Code is
0 for Disable
1 for Enable
Default: Disable
SIP/RTP Ports
To define SIP UDP Port for a VoIP Port, dial:
7768-1-VoIP Port-SIP UDP Port to define SIP UDP Port for a single VoIP port.
7768-2-VoIP Port-VoIP Port-SIP UDP Port to define the same SIP UDP Port for a range of VoIP ports.
7768-*-SIP UDP Port to define the same SIP UDP Port on all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP UDP Port is from 1025 to 65535.
Default: 5060
To define SIP TCP Port for a VoIP Port, dial:
7784-1-VoIP Port-SIP TCP Port to define SIP TCP Port for a single VoIP port.
7784-2-VoIP Port-VoIP Port-SIP TCP Port to define the same SIP TCP Port for a range of VoIP ports.
7784-*-SIP TCP Port to define the same SIP TCP Port on all VoIP ports.
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Where,
VoIP Port is the number of the software port from 01 to 16.
SIP TCP Port is from 1025 to 65535.
Default: 5060
To define RTP Listening Port, dial:
7769-1-VoIP Port-RTP Listening Port to define RTP Listening on a single VoIP port.
7769-2-VoIP Port-VoIP Port-RTP Listening Port to define the same RTP Listening port on a range of
VoIP ports.
7769-*-RTP Listening Port to define the same RTP Listening port on all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
RTP Listening Port is from 1025 to 65535.
Default: 8000.
Layer 2 VLAN/CoS
To enable/disable Layer2 VLAN/CoS for a VoIP Port, dial:
7775-1-VoIP Port-Flag to enable/disable Layer2 VLAN/CoS for a single VoIP port.
7775-2-VoIP Port-VoIP Port-Flag to enable/disable Layer2 VLAN/CoS for a range of VoIP ports.
7775-*-Flag to enable/disable Layer2 VLAN/CoS for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
Flag is
0 for Disable
1 for Enable
Default: Disable
To assign a VLAN ID to the VoIP Port, dial:
7776-1-VoIP Port-VLAN ID to assign VLAN ID for a single VoIP port.
7776-2-VoIP Port-VoIP Port-VLAN ID to assign the same VLAN ID for a range of VoIP ports.
7776-*-VLAN ID to assign the same VLAN ID for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
VLAN ID is from 0 to 4094
Default: 0001
To program SIP CoS for a VoIP Port, dial:
7788-1-VoIP Port-SIP CoS to program SIP CoS for a single VoIP port.
7788-2-VoIP Port-VoIP Port-SIP CoS to program the same SIP CoS for a range of VoIP ports.
7788-*-SIP CoS to program the same SIP CoS for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP CoS is COS priority bits from 0 to 7. The corresponding meaning of the bits with respect to Traffic
type is as follows:
0 is for Best Effort
1 is for Background
2 is for Spare
3 is for Excellent Effort
4 is for Controlled Load
5 is for Video
6 is for Voice
7 is for Network Control
Default: 3, Excellent Effort
Matrix ETERNITY System Manual
1133
1134
Timers
To configure the SIP Invite Timer, dial:
7770-1-VoIP Port-SIP Invite Timer to configure the invite timer for a single VoIP port.
770-2-VoIP Port -VoIP Port-SIP Invite Timer to configure the same timer duration for a range of VoIP
ports.
7770-*-SIP Invite Timer to configure the same timer duration for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP Invite Timer is from 010 to 180 seconds.
Default: 30 seconds.
To configure the SIP Provisional Timer, dial:
7771-1-VoIP Port-SIP Provisional Timer to configure the provisional timer for a single VoIP port.
7771-2-VoIP Port-VoIP Port-SIP Provisional Timer to configure the same duration of the provisional
timer on a range of VoIP ports.
7771-*-SIP Provisional Timer to configure the same duration of the provisional timer on all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP Provisional Timer is from 010 to 180 seconds.
Default: 60 seconds.
To configure the SIP General Request Timer for a VoIP Port, dial:
7779-1-VoIP Port-SIP General Request Timer to configure the request timer for a single VoIP port.
7779-2-VoIP Port-VoIP Port-SIP General Request Timer to configure the same timer duration for a
range of VoIP ports.
7779-*-SIP General Request Timer to configure the same timer duration for all VoIP ports.
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP General Request Timer is from 10 to 60 seconds.
Default: 20 seconds.
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LED Indication
To assign Registration LED to a SIP trunk:
7772-1-VoIP Port-SIP trunk
Where,
VoIP Port is the number of the software port from 01 to 16.
SIP trunk is the number of the SIP trunk you want to assign the LED, from 01 to 32.
Default: SIP trunk 01.
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The Port status page will display the following parameters for all VoIP Ports you have configured:
1137
Select the T1E1 Port tab, for which you want to configure the data parameters.
By default, both Framing and Signaling are enabled. You can divide the E1 channels to route Voice calls
as well as for Data communication.
1138
Select the number of E1 Channels to be used for routing data from E1 to the desired Data port in Start
Channel and End Channel. Default: Start and End Channel are blank.
In Data Port Number, select the desired Data Port to which the E1 Port is to be mapped. Default: 1.
The E1 Channels not configured above to route Data, shall be used to route Voice Calls.
By default, Framing is enabled. You can use this E1 Port for both Voice Calling and Data Communication.
Clear this check box, if you want to use this E1 Port for Data Communication only. When you clear this
check box, Framing is disabled and you can allot all the E1 channels to a single data port only.
In Assign all channels of this E1 Port to Data Port Number, select the desired Data Port to which you
want to assign all the E1 channels.
By default, Signaling is enabled. You can use this E1 Port for both Voice Calling and Data
Communication.
Clear this check box, if you want to use this E1 Port for Data Communication only. When you clear this
check box, Signaling is disabled and you can allot different E1 channels to different data ports.
Select the number of E1 Channels to be used for routing data from E1 to the desired Data port in Start
Channel and End Channel. Default: Start and End Channel are blank.
In Data Port Number, select the desired Data Port to which the E1 Port is to be mapped. Default: 1.
Repeat the same steps to map another E1 port to a Data port and selelct the channels to be used for
routing data between the two ports.
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Program the SIP ID, registrar Server Address, Registrar Server Port, Authentication ID, Authentication
Password as provided by your ITSP.
If your ITSP uses Outbound Proxy, Enable the Outbound Proxy for the SIP trunk and also program the
Outbound Proxy Server Address and Port as provided by your ITSP.
The VoIP Ethernet Ports are assigned Software Port numbers from 01 to 16. Configuration of all VoIP
parameters are done for the Software Ports. Each of these software ports must first be assigned a Hardware
ID. A hardware ID of the Software Port shows the physical location of the port on the system.
Please read Software Port and Hardware ID to know about know more about this topic.
The maximum number of SIP trunks supported by ETERNITY varies by model.
ETERNITY ME supports 32 SIP Trunks.
ETERNITY GE supports 16 SIP Trunks.
ETERNITY PE supports 16 SIP Trunks.
If you have installed as single VoIP card, you can configure all SIP trunks on the same card.
If you have installed more than one VoIP card, you may program the SIP trunks on the VoIP Ethernet Ports in
a distributed manner. For example: if you have installed four VoIP cards in your ETERNITY ME, you can either
program 8 Trunks each on the 4 VoIP Ports OR you may program 12 Trunks on the VoIP Port of the first card,
8 on the VoIP Port of the second card, 6 Trunks each on the VoIP Ports of the third and the fourth card.
1140
Configure the following SIP Trunk parameters for the SIP trunks:
SIP Trunk No.: This non-editable field is the number of the SIP trunk. The SIP trunks are numbered
from 01 to 32.
VoIP Port No.: This is the software port number of the VoIP Ethernet Port, from 00 to 16, to which you
want to configure the SIP trunk.
It is possible to configure more than one SIP trunk on a single VoIP Ethernet Port. To do this, assign
the same VoIP Ethernet Port Number to all those SIP trunks you want to configure on that VoIP Port.
For example, if you want to configure SIP trunks 03, 04 and 05 on VoIP Ethernet Port No. 02, enter this
number (02) in this field for SIP trunk number 03, 04, and 05.
Enable SIP Trunk: This flag is for enabling or disabling the SIP trunk. To be able to make incoming
and outgoing calls from the SIP trunk, click to enable the SIP trunk. By default, the SIP trunk is
disabled, disallowing incoming and outgoing calls. You may disable SIP trunks that are not functioning.
Name: You may assign a 'Name' to each SIP trunk to facilitate identification. Whenever there is an
incoming call without CLI on this port, the Name you have programmed will be displayed on the landing
extension, provided it is a DKP.
The Name assigned to the SIP trunk may consist of a maximum of 18 characters. The Name of the port
may be the name of the ITSP the SIP trunk is subscribed with (recommended).
1141
SIP ID: Enter the SIP User ID provided by the ITSP. SIP User ID is the ID that callers will use to call this
SIP trunk.
The SIP User ID may be a number or text for remote parties to call on the SIP trunk. For example, if
SIP URI provided by the ITSP is 12345@abc.com, enter 12345 in this field. SIP User ID may consist of
a maximum of 40 characters. All ASCII characters are allowed.
SIP Trunk Mode: You may select SIP Trunk Mode as Proxy or Peer-to-Peer, according to your
requirement.
If you are using the services of an Internet Telephony Service Provider (ITSP), select Proxy to register
this SIP trunk with the ITSP.
If you are not using this service, select Peer-to-Peer.
During Incoming Call Routing to select a SIP trunk, ETERNITY compares the SIP ID received in the
Request URI of the INVITE message with the SIP ID configured on the SIP Trunk.
If you do want ETERNITY to check the SIP ID received in the Request URI of the INVITE message,
clear the Check SIP ID During Incoming Call check box. Default: Enabled.
Treat Incoming Peer-to-Peer calls as: If you select Peer-to-Peer as the SIP Trunk mode, you may
configure the trunk to Treat Incoming Peer-to-Peer call as: Trunk or Station.
If you select Trunk, the incoming calls will be routed as per the Trunk Feature Template assigned to
the SIP Trunk.
If you select Station, the system will route the incoming call as follows:
When only a number is received in the To: field of the INVITE message, ETERNITY will check the
number in the Closed User Group Table. If a match is found in the CUG table the call will be routed
as per the corresponding Outgoing Trunk Bundle Group.
If the CUG Table is not configured or if no match is found for the number received in the To: field
of the INVITE message, the system will check if there is an extension number that matches with the
number received in the To: field of the INVITE message. If a match is found the call is routed to
the desired extension number.
When a Trunk Access Code and a number is received in the To: field of the INVITE message, the
system will route the call as per the Outgoing Trunk Bundle Group assigned in the Station Basic
Feature Template of the SIP Trunk.
1142
If Station is selected as the option for Treat Incoming call as, the user will only be able to:
Dial Flexible Numbers
Dial Operator Code
Dial Trunk Access Code for making outgoing calls
Access the Global Directory
Make calls within the Closed User Group
Registrar Server Address: Enter the Proxy/Registrar Server Address. The Server Address may be an
IP Address or a Domain name, of maximum 40 characters.
Registrar Server Port: Enter the Registrar Server Listening Port. The valid rang is from 1025 to
65535. By default, 5060 is set as the Listening Port of the Registrar Server.
Re-registration Time (sec): The Registrar Server deletes an entry of its client from its database on
expiry of a fixed timer, which is set by the Registrar Server. ETERNITY VoIP Card sends a registration
request before this Timer expires to remain registered on the server.
Enter the value of the Timer after which the system should send registration request to maintain
registration binding with the server. The valid range of this timer is from 00001- 65535. By default the
Timer is set to 3600 seconds.
Registration Retry Time (sec): This Timer stands for the period between retries for registration. If the
registration attempt fails, ETERNITY sends the registration request on the expiry of this Timer again.
The system continues to send the registration request till it gets registered. The valid range of this timer
is from 00001- 65535. By default the Timer is set to 00010 seconds.
Authentication User ID: Enter the Authentication ID provided by the ITSP for this SIP trunk. The
Authentication User ID may be a string of 40 characters (maximum), including ASCII characters.
Authentication Password: Enter the Authentication Password provided by the ITSP for this SIP trunk.
The Authentication Password may be a string of 24 characters (maximum), including ASCII characters.
Outbound Proxy: This parameter is relevant only if the ITSP has a SIP outbound server to handle
voice calls. If yes, program the following parameters:
Enable: Select this check box to enable Outbound Proxy. By default the flag is disabled.
Server Address: Enter the Outbound Proxy Server's address. It may be an IP Address or Domain
name. A maximum of 48 Characters, including ASCII characters are allowed.
Server Port: Enter the Outbound Proxy Server's Listening Port. The valid range for this is 102565535. By default the Server Port is 5060.
SIP Hardware Template: Assign a SIP Hardware Template to the SIP Trunk. The SIP Hardware
Template contains voice quality related features such as Voice Codec selection, Tx and Rx Gains,
Echo Cancellation, Jitter Buffer and, Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 assigned to all SIP Trunks. Template
number 01 is also assigned to all SIP Extensions.
First, check if the values in Template 01 fulfill the feature requirements of the SIP Trunks. Retain this
template, if it fulfills the feature requirements of all SIP Trunks and if the same features are to be
allowed to all SIP Trunks.
If different sets of SIP hardware features are to be allowed to different SIP Trunks, then prepare
separate SIP Hardware Templates and apply them on the SIP Trunks. To do this,
1143
Enter the number of the Template you customized, Template 03 in the 'SIP Hardware Template'
field of the SIP Trunk on which you want to apply this template.
Repeat the same steps to customize and assign a different SIP Hardware Template to another SIP
Trunk.
Also, refer the topic SIP Hardware Template to know more about customizing the templates and
applying on the SIP Trunks.
Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator,
Auto Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the
behavior of a Trunk. Apply a Trunk Feature Template to the SIP trunk. By default, Trunk Feature
Template 01 is applied on all SIP trunks as well as all other trunk types. Refer the topic Trunk Feature
Template to know more.
Click the 'Trunk Feature Template link to open the page. Check if the default Template 01 fulfills your
requirement for the SIP trunk.
If the default Template 01 does not fulfill your requirement, prepare another Trunk Feature
Template182, and enter the newly prepared Template number for the SIP trunk.
Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the SIP
trunk.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a
preference number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same
preference must be assigned the same Cost Factor. Different trunk types can also be assigned the
same Cost Factor. These trunks are used for routing calls.
Assign a Cost Factor to the SIP trunk, for instance, 02 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '9' through the SIP trunk 01
only,
You must first assign a Cost Factor (01-99) to SIP trunk 01, for example, 02.
Click the 'Least Cost Routing - Number Based' link to open the page.
Enter '9' in the 'Number' column, Cost Factor '02' as Preference 1, 2, 3 and 4.
Simultaneous Calls: This parameter is for defining the number of Simultaneous Calls to be allowed on
the SIP trunk. ETERNITY ME and GE support up to 32 simultaneous calls, while ETERNITY PE
supports 16 simultaneous calls. You can configure as many simultaneous calls as supported on the
SIP trunk by the ITSP.
182. The default template is applied on the ports of all trunk types supported by ETERNITY. Changes to the default template will be
applied on all trunk types also. So, you are advised to prepare a new template and apply it to the desired trunk types.
1144
If the ITSP supports less than 8 simultaneous calls on SIP Trunks, you must program this parameter
accordingly. However, if you ITSP supports more than 8 simultaneous calls, you need not change this
parameter.
If you have configured the above parameters for the desired VoIP Ethernet Ports, click Submit to save
your configuration settings.
Return Call to Original Caller (RCOC): Enable this flag if you want to apply the 'Return Call to Original
Caller' on this SIP trunk.
If this feature is enabled on the SIP trunk, the system will route calls returned by remote parties back to the
extensions that originally made the call from this Trunk (the original callers' extensions). To know more,
refer the feature description for RCOC (Return Call to Original Caller).
Station Basic Feature Template: Assign a Station Basic Feature Template to the SIP trunk. There are
50 different templates to choose from. Each template can also be altered to suit your requirement and
preferences. Default: Template number 01 assigned to all SIP trunks.
Station Advanced Feature Template: Assign a Station Advanced Feature Template to the SIP trunk.
There are 50 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. Default: Template number 01 assigned to all SIP trunks.
Add 'rinstance' in Register: 'rinstance' is any random value which can be used by the VoIP Ethernet Port
to fetch its own contact binding, that is, to know the Registration Expiry Timer assigned by the server. By
default, this flag is enabled.
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Send REGISTER Message: With this parameter you can select whether or not the system should send
REGISTER message from the SIP trunk. By default, this flag is enabled allowing REGISTER message to
be sent from the SIP trunk.
Allow OG Calls without Registration: This parameter is to be enabled to allow outgoing calls to be made
from the SIP trunk, even when the SIP trunk is not registered. If this flag is disabled the system will not
allow outgoing calls to be made if the status of the SIP trunk is 'not registered'. By default, this flag is
disabled.
Message Wait Indication183: If you have subscribed for Voice Mail from the ITSP of the SIP trunk, and
you want to subscribe for Message Wait Indication on the SIP Trunk for the voice mail service, configure
the following:
Message Retrieval No.: Enter the Message Wait Retrieval number provided to you by your ITSP in
this field. This number is used for retrieval of voice mail on the SIP trunk. The Message Retrieval
Number may consist of a maximum of 16 digits. Default: Blank.
Send MWI notification on: Assign the destination extension on which Message Wait Indication is to
be sent whenever there is a new message on the SIP trunk. The extension can be an SLT, a DKP, an
ISDN Terminal or a SIP Phone. Select the Port Type and Number of the destination extension. Default:
SLT-001.
Port Types
SLT
01
001 512
DKP
02
001 128
ISDN Terminal
28
001 64
SIP Extension
34
001 999
To know more about this feature, see Message Wait Indication on SIP Trunks.
Send OPTIONS as Heartbeat: With this parameter you can select whether or not the system should send
the OPTIONS message periodically to the Proxy Server to monitor its availability. Calls can be made and
received only if the Proxy Server is alive.
If the Proxy Server is unavailable, like no response is received, the status of the SIP Trunk will display
Heartbeat Failed along with the Reason for Failure.
To view status of the Proxy Server, go to SIP Trunk Status.
The Send OPTIONS as Heartbeat will work only if you have disabled Send REGISTER message.
If you enable Send OPTIONS as Heartbeat, you must configure the Heartbeat Interval.
Heartbeat Interval: Define the Heartbeat Interval (Seconds), the time period, from 10 to 999
seconds, after which ETERNITY should send the OPTIONS message to the Proxy Server. Default: 30
seconds.
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Send CLI Option: This parameter allows you to configure the CLI of the SIP Trunk to be sent to the
remote party on outgoing calls made using the SIP trunk. You may select any of the following options as
desired:
CLIR: Select this option if you do not want the CLI to be sent.
SIP ID: You may select this option if you want the SIP ID programmed on the SIP Trunk to be sent as
CLI.
Calling Party Wise: Select this option if you want to send the Calling Extension Number (the number
of the extension making the outgoing call through the SIP trunk) as CLI.
When reverse DDI is programmed on the SIP Trunk, the DDI number of the calling extension will be
sent, instead of its extension number.
If the calling extension has disabled the parameter Send DDI as CLI in its Station Advanced Feature
Template, then its Pilot number configured in the Outgoing Reference Table will be sent as CLI.
If calling extension has enabled CLIR, no CLI will be sent on the SIP Trunk.
Fixed Number: Select this option if you want a specific number to be sent as CLI. When you select this
option, you must also define the number to be sent as CLI.
You may select this option you wish to send any of your trunk line numbers as CLI on the SIP Trunk so
as to enable the called party to call back the calling party using this CLI.
Since it is not possible to call back a SIP ID, Fixed Number offers you a solution, using which you can
send a trunk line number as CLI on the SIP Trunk. Using this CLI, the called party can call back the
calling party.
The Fixed Number may consist of a maximum of 40 characters, including all ASCII characters.
By default, SIP ID is set as the Send CLI option for all SIP Trunks.
When extension number of the calling extension is blank, and the Send CLI Option programmed for the
the SIP Trunk is other than "SIP ID", then also SIP ID will be sent as CLI.
Accept Anonymous Calls?: The flag is for accepting calls without CLI that land on the SIP trunk. By
default the flag is enabled. You may disable this flag to disallow calls without CLI on this SIP trunk port.
Source Port IP Address: Select the Source Port IP Address for the SIP trunk. You may select from any of
the following options, as applicable to your installation scenario.
Use VoIP Ethernet Port IP Address: Select this option, if the VoIP Ethernet Port on which the SIP
trunk is configured is connected directly to the public Internet.
Use IP Address fetched using STUN: Select this option, if the VoIP Ethernet Port is located behind a
NAT router other than Symmetric. If you select this option, make sure to disable 'Outbound Proxy'.
Use Router's Public IP Address: Select this option if the VoIP Ethernet Port is located behind a NAT
router (any type). If you select this option, make sure to disable 'Outbound Proxy'.
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Use Symmetric RTP?: The use of Symmetric RTP makes it possible for a SIP device to send the RTP on
the same connection on which it is listening for RTP. This is done only on peer to peer SIP trunks.
Enable this flag, if the VoIP Ethernet Port is located on a public IP and you want outgoing calls to the SIP
Client located behind the NAT Router. OR if you need to receive incoming calls from the SIP Client located
behind the NAT router. By default, Use of Symmetric RTP is disabled.
Digest Authentication: This flag is to be enabled, if you want the feature 'Digest Authentication' to be
enabled on the SIP trunk.
When Digest Authentication is enabled, incoming calls from callers will be allowed only after the callers
have authenticated themselves (with their User ID and Password). If the caller enters invalid authentication
information, the system will re-challenge authentication once more, and reject the call, if the authentication
attempt fails again.
When you enable Digest Authentication, make sure that the Digest Authentication Table is also configured
first. To do this,
Enter the User ID to be authenticated along with its corresponding Password against each Index on the
Table.
The User ID may be a maximum of 40 characters, including ASCII characters. The User Password may
be a maximum of 16 characters, including ASCII characters.
If you have finished entering the User IDs and their corresponding User Passwords in this Table, click
Submit at the bottom of the page.
Default Transport for Outgoing Message: Configure this parameter for Proxy SIP Trunks only. The VoIP
card of the ETERNITY supports three options for transporting outgoing SIP messages:
UDP: Outgoing messages are transported using UDP. By default this option is selected.
TCP: Outgoing messages are transported using TCP. If you select this option, you must enable SIP
Over TCP on the VoIP Port Parameters page.
TCP (Fallback to UDP): TCP is used for outgoing messages. However, if the TCP connection fails, the
system will attempt to send the message again over UDP.
TLS: Outgoing messages are transported using TLS. If you select this option, you must enable SIP
Over TLS on the VoIP Port Parameters page.
By default, UDP is selected as the Default Transport for Outgoing Message.
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The Default Transport for Outgoing Message options are checked only if you have enabled SIP over TCP
or SIP over TLS.
.
If the SIP over TCP and SIP over TLS are disabled, all outgoing SIP messages will be transported over
UDP only.
SRTP Mode: ETERNITY supports SRTP (Secure Real Time Protocol) for secure conversations over SIP.
The VoIP card of ETERNITY supports the following options:
Disable: ETERNITY uses normal RTP for transporting the speech packets.
Optional: ETERNITY uses SRTP for transporting the speech packets. If the remote user does not
support SRTP, ETERNITY will use normal RTP for transporting the speech packets.
If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
By default, AVP is selected as the SRTP Media Type.
Forced: ETERNITY uses only SRTP (SAVP) for transporting the speech packets. If the remote user
does not support SRTP, ETERNITY will reject incoming calls from and drop outgoing calls made to
such users.
By default, SRTP Mode is Disabled.
Call Budget: If you want to enable Call Budget on Trunk feature on this SIP trunk, configure the
following parameters:
Type: Select the type of Call Budget on TrunkAmount or Minutes or Callsto be applied on this SIP
trunk. By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 99999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 99999.
Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By default
the number of calls is set to 9999.
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan suggests
so.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset also using SA and SE
commands, referred to as Manual Reset. Refer the feature description Call Budget on Trunk.
Call Back: This parameter is related to the Call Back on Trunk Port feature. If you want to enable the
'Call Back on Trunk Port' feature on this SIP trunk, configure the following parameters:
Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this flag
is disabled on all trunk port types. By default, the flag is disabled.
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Call Back Timer: This is the duration for which the system waits for the caller to disconnect before
applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to 10
seconds for all SIP Trunks.
Call Back Mode: Select from the following options how a Call Back call answered by the remote party
should be routed:
Built-In Auto Attendant
PIN Authentication - Multiple Calls
CLI Authentication - Multiple Calls
CLI Authentication - Single Call - Answer Signaling
Operator
By default, Operator is selected as the Call Back Mode.
Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the CLI number. If you want the call back to be made to a different number, select Alternate
Number.
By default, CLI number is selected for Call Back.
Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all SIP trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all SIP trunks and Trunk Port types, retain this
list.
If you want a different set of number strings to be programmed for this SIP Trunk, select a different
Number List, and assign it to the SIP trunk port.
You may program the Incoming Number List either from the Number List page or by clicking the
Incoming Number List link to reach the Number List page.
Refer the topic Number List to know more, and for configuration instructions.
Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the Incoming Number List, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
example, for the number string programmed at Index 1 in the Incoming Number List, a corresponding
number string at the same Index, Index 1, should be programmed in the Outgoing Number List.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all SIP trunks as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this SIP Trunk port.
You may program the Outgoing Number List either from the Number List page or by clicking the
Outgoing Number List link to reach the Number List page.
Refer the topic Number List to know more, and for configuration instructions.
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Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an Outgoing Trunk Bundle Group (OGTBG).
Select Same port if you want the call back to be made using the same port on which the missed call is
received. If you select OGTBG, the call back will be made using the OGTBG, which you have defined.
By default, Same port is selected.
OGTB Group: If you selected OGTBG for making the call back in the previous parameter, you must
define the OGTBG that must be used in this parameter.
By default, OGTBG 01 is assigned to all SIP trunks.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTBG that you define here for Call Back.
Incoming (IC) Reference ID: Assign an Incoming Reference ID for the SIP trunk for Working Hours,
Break Hours, Non-Working Hours. By default, 00 is assigned as Incoming Reference ID for all three time
zones.
Outgoing (OG) Reference ID: Assign an Outgoing Reference ID for the SIP Trunk, from 00 to 99. By
default, 00 is assigned as Outgoing Reference ID.
Pause Timer (sec): This Timer is required for inserting delay while digits of a number string are out dialed
from the SIP trunk. The Pause Timer will be applicable when the letter 'P' is configured in the DTMF
number string which is to be out dialed as DTMF digits on the SIP trunk. The range of this timer is from 1 to
9 seconds. By default the Timer is set to 3 seconds.
For example, if 'PPP3' is to be out dialed and Pause timer is programmed as 3 seconds, the ETERNITY
will out dial the digit 3 after 9 seconds, after a delay of individual P (3+3+3 =9). The range of this Timer is
from 1 to 9.
This parameter is used for the Multi-Stage Dialing feature.
DTMF ON Timer: This is the time for which the DTMF digit will remain ON, while being out dialed by the
ETERNITY. This parameter finds its application in the feature Multi-Stage Dialing. The range of this timer
is from 051 to 255 milliseconds. By default, ON Time is defined as 102 msec.
DTMF Inter Digit Pause (msec): This is the time for which the ETERNITY will wait before dialing the
successive digits.
Gateway Application - Answer Signaling: This parameter is to be programmed if the SIP trunk is being
used in a gateway application as a source port (from where calls originate). The calls originated on the
source port (SIP trunk) are routed using another Trunk port, the terminating port, which may be any trunk
port, like T1E1. When call made from the terminating port gets matured, this is signaled to the source port
in the form of DTMF digits.
Use?: Enable this flag if you want the SIP trunk to be used in a Gateway Application.
DTMF String: Program the DTMF digits to be sent to signal call maturity to the source port.
Send Re-INVITE over SIP Trunk on Hold: With this parameter you can select whether or not the system
should send Re-INVITE message from the SIP Trunk to the Remote Peer, when an external call over the
SIP Trunk is put on hold by the extension user. Set this parameter as per the requirement of the Remote
Peer. The Remote Peer can be a Proxy Server or a SIP Device.
1151
By default, Send Re-Invite over SIP Trunk on Hold is disabled. For more information see Call Hold.
Delayed Offer: Enable this flag, if you want the SIP Trunk to generate INVITE without SDP. Default:
Disabled.
Fetch Called Party Number From: ETERNITY extensions may be assigned DDI numbers provided by
the ITSP. When INVITE is received from ITSP, the ITSP may send the DDI number either in the Request
URI of the INVITE message or in the To: field of the INVITE message.
By default, Request URI is selected. Ask your ITSP if you need to change this parameter.
Called Party Number as CLI : If you want ETERNITY to display the called party number received in the
INVITE message as the CLI, select the Display Called Party Number as CLI check box. By default, Display
Called Party Number as CLI option is disabled.
This parameter is useful when a single SIP Trunk having DDI Numbers and Operator are shared by more
than one organization. To enable the Operator to handle calls more efficiently, you must enable Display
Called Party Number as CLI and configure the names and corresponding numbers of the organizations
sharing the SIP Trunk in the Global Directory of ETERNITY.
When there is an incoming call, ETERNITY matches the number with the numbers in the Global Directory.
If a match is found ETERNITY displays the company name configured for that entry to the Operator.
You can configure the Display Called Party Number as CLI option only from Jeeves.
On Connecting Media Send: If Built-In Auto Attendant or DISA is enabled on the SIP Trunk, select the
response you want the system to send on Connecting Media, that is 200 OK or 183 Session Progress.
Answer Source Trunk on Receiving: When a call received on any trunk of ETERNITY is routed through
the SIP Trunk, select the response after which the call on the source trunk must be answered. You can
select Early Media or 200 OK.
If you have configured the above parameters, click Submit at the bottom of the page to save your
configuration settings.
1152
For each SIP trunk (number), the following settings will be displayed:
SIP Trunk Number
Status
Registration Time
Registration Retry Count
Reason for Failure
Call Budget Type
Allotted Amount/Minutes/Calls
Consumed Amount/Minutes/Calls
Scheduled Reset
Budget Reset Scheduled (Date)
Reset Consumed Budget (this is not a status indicator. It is for resetting the Consumed Call Budget
manually)
You can also view the SIP Trunk Status from the Status link. To view, click the SIP Trunk link under
Status.
1153
SIP is the number of the software port of the SIP trunk from 01 to 32.
VoIP Ethernet Port is the number of the port from 00 to 16.
Default: no SIP trunks are assigned VoIP Ethernet Port.
To de-assign the SIP trunk from a VoIP Ethernet Port:
7701-1-SIP trunk-00
To enable/disable the SIP trunk, dial:
7702-1-SIP-Code to enable/disable a single SIP trunk.
7702-2-SI-SIP-Code to enable/disable a range of SIP trunks.
7702-*-Code to enable/disable all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Code is
1 for Enable
0 for Disable
To assign a name to the SIP trunk, dial:
5410-1-SIP-Name-#* to assign the name to single SIP trunks.
5410-2-SIP-SIP-Name-#* to assign the same name to a range of SIP trunks.
5410-*-Name-#* to assign the same name to all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Name is a string of 18 alphanumeric characters. Terminate the command with #* if the Name has fewer
than 18 characters.
To clear the name of a SIP trunk, dial:
5410-1-SIP-#* to clear the name of a single SIP trunk.
5410-2-SIP-SIP-#* to clear the name of a range of SIP trunks.
5410-*-#* to clear the names of all SIP trunks.
To configure SIP User ID for a SIP trunk, dial:
7704-1-SIP-SIP User ID
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
SIP User ID is a string of 40 characters (maximum), with extended ASCII characters
To configure Registrar Server's Address, dial:
7705-1-SIP-Registrar Servers' Address to configure the server address for a single SIP trunk.
7705-2-SIP-SIP-Registrar Servers' Address to configure the same address of a range of SIP trunks.
7705-*-Registrar Servers' Address to configure the same address for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
SIP Registrar Servers' Address is a string of 40 characters (maximum) with extended ASCII characters.
To define Registrar Server's Listening Port, dial:
7706-1-SIP-Registrar Server's Port to define the listening port for a single SIP trunk.
7706-2-SIP-SIP-Registrar Server's Port to define the listening port for a range of SIP trunks.
7706-*-SIP Registrar Server's Port to define the listening port for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
SIP Registrar Server's Port is from 1025-65535. Default: 5060
1154
1155
7716-*-Outbound Proxy Server Address to configure the same Server Address for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Outbound Proxy Server Address is a string of a maximum 40 characters, with extended ASCII
character set.
1156
For Advanced Configuration of the SIP trunks, use the following commands:
To enable/disable RCOC on SIP trunk, dial:
7743-1-SIP-Code to enable/disable RCOC on a single SIP trunk.
1157
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Flag is
0 for restrict OG Calls without Registration
1 for allow OG Calls without Registration
Default: 1
To subscribe for Message Wait Indication on SIP Trunk, dial:
7951-1-SIP-Flag to enable/disable this flag on a single SIP trunk.
7951-2-SIP-SIP-Flag to enable/disable this flag on a range of SIP trunks.
7951-*-Flag to enable/disable this flag on all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Flag is
0 for Disable
1 for Enable
Default: Disabled
To configure Message Retrieval Number for the SIP Trunk, dial:
7952-1-SIP-Message Retrieval No. to configure the Message Retrieval Number for a single SIP trunk.
7952-2-SIP-SIP-Message Retrieval No. to configure the Message Retrieval Number for a range of SIP
trunks.
7952-*- Message Retrieval No. to configure the Message Retrieval Number for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Message Retrieval Number may be a maximum of 16 digits.
Default: Blank
To configure the destination to send Message Wait Indication notification for the SIP Trunk, dial:
7953-1-SIP-Port Type-Port No. to configure the destination for a single SIP trunk.
7953-2-SIP-SIP-Port Type-Port No. to configure the destination for a range of SIP trunks.
7952-*- Port Type-Port No. to configure the destination for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Port Types and Port Numbers are:
Port Type
SLT
01
001 512
DKP
02
001 128
ISDN Terminal
28
001 64
SIP Extension
34
001 500
Default: SLT-001
To select a 'Send CLI' option for a SIP trunk, dial:
7718-1-SIP-Send CLI Option to select 'Send CLI' option for a single SIP trunk.
7718-2-SIP-SIP-Send CLI Option to select 'Send CLI' option for a range of SIP trunks.
7718-*-Send CLI Option to select Send CLI option for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Send CLI Option is
1158
1159
SIP is the number of the software port of the SIP trunk from 01 to 32.
Digest Authentication is
0 for Disable
1 for Enable
Default: Disable
To configure Digest Authentication Table, dial:
4118-Index-User ID to configure User ID in the table
4119-Index-User Password to configure the corresponding password for the User ID.
Where,
Index is the location on the Authentication Table from 01 to 32.
User ID may be a maximum of 40 characters, including ASCII characters. Terminate the command with
#* if User ID has fewer than 40 characters.
User Password may be a string of a maximum of 16 characters, including ASCII characters. Terminate
the command with #* if User Password is less than 16 characters.
The Index number should remain the same for each User ID and its corresponding password. For
example, if you configured a User ID at Index number 03, the corresponding password for this ID should
also be configured at Index number 03.
To configure Default Transport for Outgoing Messages for a SIP trunk, dial:
7731-1-SIP-Default Transport for OG Message to select the transport mode for a single SIP trunk.
7731-2-SIP-Default Transport for OG Message to select the transport mode for a range of SIP trunks.
7731-*-SIP-Default Transport for OG Message to select the transport mode for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Default Transport for OG Message is
1 for UDP
2 for TCP
3 for TLS
Default: UDP
For SE Commands for configuring Call Budget on SIP Trunks, refer the feature description Call
Budget on Trunk.
To enable / disable Call Back on SIP Trunk port, dial:
7712-1-SIP-Code to enable/disable Call Back on a single SIP trunk.
7712-2-SIP-SIP-Code to enable/disable Call Back on a range of SIP trunks.
7712-*-Code to enable/disable Call Back on all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Code is
0 for Disable
1 for Enable
Default: Disabled
To program Call Back Timer on SIP Trunk port, dial:
7713-1 -SIP-Call Back Timer to set timer for a single SIP trunk.
7713-2 -SIP-SIP-Call Back Timer to set same timer duration for a range of SIP trunks.
7713-*-Call Back Timer to set the same timer duration for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Timer is from 01 to 99 Sec.
Default: 10 Seconds (for SIP trunk 01)
1160
1161
1162
To configure the DTMF Inter-Digit Pause Timer for the SIP trunk, dial:
7726-1-SIP-DTMF Inter-Digit Pause Timer to configure the timer for a single SIP trunk.
7726-2-SIP-SIP-DTMF Inter-Digit Pause Timer to configure the timer for a range of SIP trunks.
7726-*-DTMF Inter-Digit Pause Timer to configure the timer for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
DTMF Inter-Digit Pause Timer from 051 to 255 milliseconds. Default: 102 msec.
To enable/disable Gateway Application-Answer Signaling on the SIP trunk, dial:
7727-1-SIP-Gateway Application to enable/disable gateway application for a single SIP trunk.
7727-2-SIP-SIP-Gateway Application to enable/disable gateway application for a range of SIP trunks.
7727-*-Gateway Application to enable/disable gateway application for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Gateway Application is
0 for Disable
1 for Enable
Default: Disable
To configure DTMF String for Gateway Application-Answer Signaling on the SIP trunk, dial:
7728-1-SIP-Gateway Application DTMF String to configure string for a single SIP trunk.
7728-2-SIP-SIP-Gateway Application DTMF String to configure string for a range of SIP trunks.
7728-*-Gateway Application DTMF String to configure string for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Gateway Application DTMF String is a string of maximum 4 digits. Default: CCC.
To enable/disable the flag Send Re-Invite over SIP Trunk on Hold, dial:
5324-1-SIP- Flag to configure Send Re-Invite over SIP Trunk on Hold for a single SIP trunk.
5324-2-SIP-SIP-Flag to configure Send Re-Invite over SIP Trunk on Hold for a range of SIP trunks.
7726-*-Flag to configure Send Re-Invite over SIP Trunk on Hold for all SIP trunks.
Where,
SIP is the number of the software port of the SIP trunk from 01 to 32.
Flag is
0 to Disable
1 to Enable
Default: Disabled.
Exit SE mode.
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Connect SPARSH VP248, the Extended IP Phone for ETERNITY supplied by Matrix. The Matrix Extended
IP Phone functions just like a DKP.
Connect SPARSH 330184, the Touch Screen Extended IP Phones for ETERNITY supplied by Matrix.
SIP Extensions function like any normal DKP/SLT extension of the ETERNITY. SIP Extension users can make and
receive calls to any extension user of the ETERNITY as well as to external numbers over PSTN, GSM, VoIP and
E&M lines, depending on the Logical Partition configured in the System.
The number of SIP Extensions supported by the different models of ETERNITY are:
ETERNITY ME: 999 SIP Extensions
ETERNITY GE: 500 SIP Extensions
ETERNITY PE: 50 SIP Extensions
SIP Extensions are a licensed feature. Decide the number of SIP Extensions you will require and buy the
license. Refer the topic License Management to know more.
You can register a SIP Extension at three different locations as a single SIP Extension for Call Forking.
SIP Extension Settings and the Extended Phone Settings as per the type of Extended Phone you have
connected, see Configuring SIP Extension Settings as per the Extended Phone Type.
Voice Mail Settings, if you want to provide mailbox to the SIP extensions. See Configuring Voice Mail
Settings for SIP Extensions.
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VoIP Port No.: This is the software port number of the VoIP Card.
Hardware Slot: This is the number of the hardware slot in which the VoIP Card is located.
Name: This is the name you have assigned to the VoIP Card, when you configured the VoIP Port
parameters.
As the Source Port IP Address, select the NAT Traversal mechanism for SIP messages from the
following options:
Use VoIP Ethernet Port IP Address: Select this option, if your VoIP Card is not located behind a
NAT Router.
Use IP Address fetched using STUN: Select this parameter, if your VoIP Card is located behind a
NAT Router, and you have set 'Use IP Address fetched using STUN' as the NAT Traversal
mechanism in the VoIP Port Parameters.
Use Router's Public IP Address: Select this option, if your VoIP Card is located behind a NAT
Router, and you have set 'Router's Public IP Address' as the NAT Traversal mechanism in the VoIP
Port Parameters
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You may set the Maximum Registration Timer (sec) as required. This is the Maximum Expiry Timer,
which the VoIP Card will accept in the REGISTER request received. If the value of Maximum Expiry
Timer received in the REGISTER request is greater than the value you have set here, the system will
send the value you have set in the SIP message. The same timer is used for handling SUBSCRIBE
requests. The valid range of this timer is from 10 to 99999 seconds. By default it is set to 3600
seconds.
You may set the Minimum Registration Timer (sec), as required. This is the Minimum Expiry Timer,
which the User Agent should send in its REGISTER request. If the expiry value in the REGISTER
message is less than this value, the request will be rejected. The valid range of this timer is from 10 to
99999 seconds. By default, it is set to 45 seconds.
In the Private Key field, enter the MD5 authentication key the VoIP Card of ETERNITY should use to
encrypt/decrypt the SIP messages. The Private Key may consist of a maximum of 24 characters. By
default, the field is blank.
You can restore the default values of any one or all the parameters on this page by clicking the Default
One and the Default button respectively.
If you have connected the Matrix SPARSH VP248 as SIP Extensions, for configuration instructions see
Configuring Matrix SPARSH VP248 - Extended IP Phone.
If you have connected the Matrix SPARSH VP330186 as SIP Extensions, for configuration instructions see
Configuring Matrix SPARSH VP330
If you have registered the Matrix SPARSH MS Android/iPhone Application187 as SIP Extensions, for
configuration instructions see Configuring Matrix SPARSH MS Android/iPhone Application.
If you have connected Open SIP Phones or SIP enabled devices as SIP Extensions, for configuration
instructions see Configuring Open SIP Phones
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Configure the following voice mail settings for the SIP extensions you wish to assign voice mail.
SIP Extension: This is the software port number of the SIP extension.
Access Code: This is the SIP ID (that is, extension number) assigned to the SIP Extension in the SIP
Extension Settings. The SIP IDs you assigned on the SIP Extension Settings page will appear here.
Name: This is the name assigned to the SIP extension in the SIP Extension Settings.
Personal Mailbox: Select the check box to assign Personal Mailbox to the SIP extension.
By default, Personal Mailbox is not assigned to extensions in the Enterprise mode. In the Hotel mode,
Personal mailbox is assigned to all extension phones, by default.
Mailbox Size (min.): You may increase or decrease the size of the personal mailbox assigned to the
SIP extension, by changing the default Mailbox size of 5 minutes. You may change the mailbox size to
any desired value from 00001 to 60000 minutes. Default: Enterprise mode: 5 minutes; Hotel mode: 999
minutes.
Maximum Message Length (sec): You can define the length of each message (in seconds) callers
are to be allowed to record in the mailbox. You may change the maximum message length to any
desired value from 0001 to 9999 seconds. Default: Enterprise mode: 15 seconds; Hotel mode: 120
seconds.
The VMS card will stop recording the message of the callers, if it exceeds the maximum message
length, and will store only that part of the message recorded within the maximum message length limit.
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New Message Delivery Option in Mailbox Full condition: When the personal mailbox is full, you
may select one of the following options for delivery of new messages:
Do not offer to record a message: The VMS will not allow the caller to record a message by
declining delivery of the message.
Deliver new message to General Mailbox: The VMS will record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their Class of Service (COS) are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the SIP Extension. Refer Class of Service (COS) for instructions.
Overwrite old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS starts overwriting the oldest message first.
By default, Deliver to General mailbox is selected.
Play message details after delivery of message: After the extension user has finished listening to a
message in the mailbox, you can also have the VMS play message details such as Date and Time
when the message was recorded, the callers number188, and the extension number dialed by the
caller189 to the extension user.
You may select from one of the following options for Play message details:
Never: The VMS will not play message details to the mailbox owner after playing the message.
Always: The VMS will play message details to the mailbox owner after playing each message.
On Demand: The VMS will play message details to the mailbox owner only when the mailbox
owner requests it. On completion of each message, the VMS will prompt the extension user to
press a digit for date and time stamp. When the mailbox owner presses the digit, the VMS will play
the message details.
Ask Password to Access Mailbox: By default, access to the mailbox is password protected. The
User Password is required to access the mailbox. Whenever the mailbox owner accesses the
mailbox, the VMS will ask for the (user) password.
If you want to remove password protection, clear this check box.
Since a Mailbox can be accessed using the default User Password, 1111, extension users who are assigned
a mailbox are recommended to change their User Password to a unique 4 digit number to prevent
unauthorized access to their mailbox.
188. The number of person who left the message in the mailbox.
189. The number of the extension user for whom the message is intended.
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Allow Mailbox Management: Mailbox Management allows the extension user to change mailbox
settingsrecord Extension Name for the mailbox, redirect messages from the mailbox, delete all old
messages from the mailbox, record greeting messages for the mailbox . By default, Mailbox
Management is enabled.
If you do not want to allow the extension user to change the mailbox settings, in Allow Mailbox
Management, select No.
To know more about this feature, see Mailbox Settings
Auto Delete Messages: Select the type of messages you want the VMS to automatically delete from
your mailbox. You can select All or Old. Default: None.
Days for Auto Delete Messages: Select the number of days after which you want the VMS to
automatically delete the messages in your mailbox. Default: 90 days.
Department Group Mailbox: You can assign the Mailbox of a Department Group to SIP extensions,
even to those SIP extensions that are not included in the Department Group. Refer the topic
Department Call.
To assign the Department Group mailbox to a SIP extension, select the number of the Department
Group (1 to 16) from the box.
If you do not want to assign Department Group mailbox to a SIP extension, select None.
Default: None.
Voice Mail Auto Attendant Features: This parameter is applicable only if you are using the VMS Auto
Attendant for Auto Attendant.
Voice Mail Auto Attendant Profile: Select a Voice Mail Auto Attendant profile for the SIP
extension. The Auto Attendant profile determines the welcome message to be played to mailbox
owners when they reach the home node. It also determines whether or not the user should be taken
to the root node directly.
Abbreviated Name: When the VMS is used as Auto Attendant, the callers can be prompted to Dial
by Name of the desired extension users instead of their extension numbers.
To allow callers to reach the SIP extension using Dial By Name, abbreviate the extension users
name to three letters and enter it in this field.
Announce Name: If you want the VMS to announce the extension users name to the caller when
transferring the call to the extension, select the check box to enable Announce Name. By default,
Announce Name is disabled.
If you enable Announce Name, make sure you record the extension users name on the VMS. Refer
Recording Extension Names for instructions.
Call Transfer: Select the desired method for transferring the call answered by the VMS Auto
Attendant to the SIP extension. You may select any of the following methods of call transfer for
each time zone, Working Hours (WH), Break Hours (BH) and Non-working Hours (NH):
None: When the caller dials the extension number, the VMS Auto Attendant will check if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
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Blind: When the caller dials the extension number, the VMS Auto Attendant will transfer the call
on the extension without checking whether it is busy or free.
Wait for Ring: When the caller dials the extension number, the VMS Auto Attendant will wait for
the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the
extension, if assigned, or take the caller back to the home node.
Wait for Answer: When the caller dials the extension number, the VMS Auto Attendant will
transfer the call when the extension answers (goes OFF-Hook).
If the extension does not answer190, the VMS Auto Attendant will transfer the call to the mailbox
of the extension, if assigned, or take the caller back to the home node.
Screen: The VMS Auto Attendant prompts the caller to record his/her name. It puts the caller on
hold and places the call on the desired extension. If the extension is free and answers the call,
the VMS announces the callers name to the extension user and prompts the extension user to
choose whether or not to speak to the caller. If the extension user chooses to talk, the VMS
transfers the call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the
extension user, if assigned, and asks the caller to leave a message.
By default, Wait for Answer is selected as Call Transfer method for all time zones.
Message Wait Indication: This parameter allows you to select the type of indication to be given to the
extension user for new messages in the mailbox and message wait set by another extension user.
Select Stuttered Dial Tone/Voice Message, to receive a notification or select No Notification, if you
do not want to receive any notification. Default: No Notification.
Stuttered Dial Tone/Voice Message: When the extension user goes OFF-Hook, s/he will hear a
voice message, if a pre-recorded Voice Module has been assigned for Message Wait Notification. If
no voice module is recorded and assigned, the extension user will hear a stuttered dial tone
instead.
If you want voice message to be played as message wait notification, record and assign a Voice
Module. Refer Voice Message Applications for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait Notification
will not be played, if there are already 4 being played simultaneously. In which case, Stuttered Dial Tone will
be played for Message Wait Notification, when the extension user goes OFF-Hook.
The Notification Types - LED Lamp and Ring are not supported on SIP Extensions.
When the extension user answers the call, the VMS informs the user of the new message and
allows the extension user to access it.
Refer the feature description Message Wait to know more.
190. The VMS will wait for the duration of the Wait for Answer Timer (default: 30 seconds; the timer is configurable). If the call is not
answered before this timer expires, it is treated as No Reply.
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Message Wait Notification via Call: The message wait notification will be sent to a number
(destination number). This number can be an internal or an external number. To use this feature,
configure the following parameters:
Type: If you want the notifications to be sent as soon as a new message arrives in the mailbox of
the extension user, select Immediate.
If you want the notification to be sent at fixed time schedules, select Scheduled.
If do not want to set message wait notification via call, select None. Default: None.
Profile: Assign the Profile according to which you want the system to send the notifications. The
Profile determines the time intervals during which the notifications must be sent to the destination
number.
Destination Number: Enter the number on which you want the system to send the notification
calls.
The destination number can be an internal or an external number. The destination number can be a
maximum of 16 digits. Valid digits are 0 to 9, # and *.
When the notification call is answered, the VMS informs the callee of the new message and allows
the callee to access it.
Refer the feature description Message Wait Notification via Call to know more.
Message Wait Notification via E-mail: The message wait notification will be sent to the e-mail
address of the extension user. To use this feature, configure the following parameters:
Notification: If you want the message wait notification to be mailed to the extension user along with
the new voice message as attachment, select the option Send With Attachment.
If you want only the notification to be mailed, select the option Send Without Attachment.
If do not want to set message wait notification via e-mail, select Do not send. Default: Do not send.
E-mail Address: Enter the e-mail ID of the extension user to which the notification is to be sent. Email ID may consist of up to 64 characters. Default: blank.
Extensions users will receive notifications only for the mailbox memory utilization, if you configure the E-mail
Address and select Do not sent as the Notification option.
Refer the feature description Email Based Notification to know more.
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The SIP Extension Status page will open and display the following for each SIP Extension,
Contact 1
Contact 2
Contact 3
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General Parameters
To select Source Port IP Address, dial:
7830-1-VoIP Port-Source Port IP Address Option
Where,
VoIP Port is the software port number of the VoIP Port from 01 to 16.
Source Port IP Address Options are
1 for Use IP Address of VoIP Ethernet Port.
2 for Use IP Address Fetched using STUN.
3 for Use Routers Public IP Address.
By default, Use IP Address of VoIP Ethernet Port is selected.
To set the Maximum Registration Timer, dial:
7831-1-VoIP Port - Maximum Registration Timer
Where,
VoIP Port is the software port number from 01 to 16.
Maximum Registration Timer is from 10 to 99999 seconds.
By default, it is set to 3600 seconds.
To set the Minimum Registration Timer, dial:
7832-1-VoIP Port-Minimum Registration Timer
Where,
VoIP Port is the software port number from 01 to 16.
Minimum Registration Timer is from 10 to 99999 seconds.
By default, it is set to 45 seconds.
To configure Private Key, dial:
7833-1-VoIP Port-Private Key
Where,
VoIP Ethernet Port is the software port number from 01 to 16.
The Private Key is any string of a maximum of 24 characters. All ASCII characters are allowed.
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1 - With Attachment
2 - Without Attachment
Default - Disabled.
To program E-mail ID for a mailbox, dial:
7914-1-SIP Extension-E-mail ID-#* to assign an E-mail ID for a mailbox to a single SIP Extension.
7914-2-SIP Extension-SIP Extension-E-mail ID-#* to assign an E-mail ID for a mailbox to a range of SIP
Extensions.
7914-*-E-mail ID-#* to assign an E-mail ID for a mailbox to all SIP Extensions.
Where.
SIP Extension is from 001 to 999.
Where E-mail ID is 64 ASCII Characters.
Default: Blank.
Exit SE mode.
1177
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You may select the SIP Extension number you want to configure.
The parameters of the SIP Extension number you selected will appear on this page.
In the VoIP Port No. field, select the software port number of the VoIP Port to which you want to assign the
SIP Extension. For example, you want to assign SIP Extension 1 to VoIP Port number 2, select 02 as port
number from the list.
Upto 250 SIP Extensions can be registered with a single VoIP Card.
Select the Use SIP Extension check box to enable the SIP extension. Default: enabled.
You may clear this check box, when you want to deactivate the SIP extension.
In the Name field, enter a name for the SIP Extension, which may be the name of the person who will use
the SIP Extension or the name of a Department. The name you enter here will be displayed as the Caller
ID of the SIP Extension on the remote user's phone, when the SIP Extension user makes calls.
The name may consist of a maximum of 18 alphanumeric characters.
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Enter the SIP ID for the extension. The SIP ID is necessary for registering the SIP Extension with the
Registrar of the VoIP Card. It is the number with which you can call the SIP Extension. Any extension user
of the ETERNITY can call a SIP Extension by dialing the SIP ID assigned to the SIP extension. SIP ID of
each SIP Extension must be a unique number string of a maximum of 6 digits. Any combination of digits
from 0 to 9 and the characters * and # are allowed.
You cannot assign the same SIP ID to more than one extension.
To assign SIP IDs according to your preference and requirment to a range of SIP Extensions, see
Assigning Access Codes to a Range of Extensions.
By default, the SIP IDs are Blank.
The SIP ID will be set to default value (blank), when you restore the default settings of the system.
In the Authentication ID field, enter the number which you want the VoIP Card to use for user
authentication of the SIP messages received from the SIP Extension. The number may be a string of
maximum 6 alphanumeric characters. All ASCII characters except < > and (double quote) are allowed.
You must configure the Authentication ID, if any of the SIP Message Authentication Options, namely
REGISTER or INVITE or SUBSCRIBE or PUBLISH, is enabled.
If you do not configure the Authentication ID and the Authentication Password, by default, the system will
consider the SIP ID as the Authentication ID and 1234 as Authentication Password. You may change the
Authentication ID and the Authentication password as per your requirement.
In the Authentication Password field, enter the password to be used by the VoIP Card to authenticate
the SIP messages received from the SIP Extension. You can enter a maximum of 24 alphanumeric
characters. All ASCII characters except < > and (double quote) are allowed.
In Call Appearances, define the maximum number of simultaneous incoming calls that the SIP Extension
user should be allowed to receive. You can set up to 10 call appearances for a SIP Extension. Default: 2.
When Call Appearance is set to 2, the SIP Extension can receive 2 calls at a time.
Under Authentication, enable Authentication of any or all of the following SIP Message Options by
selecting the respective check boxes:
REGISTER Request
INVITE Request
SUBSCRIBE Request
By default, the SIP Message Options REGISTER, INVITE, SUBSCRIBE are enabled.
Make sure that the Authentication ID for the SIP Extension has been programmed, when any of the above
SIP Message Options are enabled.
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For secure conversations over SIP, enable SRTP Mode. The VoIP card of ETERNITY supports the
following options:
Disable: ETERNITY uses normal RTP for transporting the speech packets.
Optional: ETERNITY uses SRTP for transporting the speech packets. If the remote user does not
support SRTP, ETERNITY will use normal RTP for transporting the speech packets.
If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
By default, AVP is selected as the SRTP Media Type.
Forced: ETERNITY uses only SRTP (SAVP) for transporting the speech packets. If the remote user
does not support SRTP, ETERNITY will reject incoming calls from and drop outgoing calls made to
such users.
By default, SRTP Mode is Disabled.
Select a Key Template for the extension. The Key Template may be of the Operator, Executive, Guest
or Hotel Attendant, according to the key map you want to assign to this extension. Default: Operators
Template.
Like the DKP, the Extended Phone will function as Operator, Executive, Hotel Attendant, and Hotel Guest
extension, according to the key map template you assign. For example, if the Extended Phone is to be
used by the Operator, select Operator's Template. The phone will be assigned the key template with the
special features required by Operators, such as more DSS keys for Trunk Access and Call Appearances,
a Call Release Key, etc.
Similarly, if the user of the Extended IP Phone is a Hotel Attendant, select 'Hotel Attendant's Template'.
The key map with the specific Front Desk User features such as Check-In, Check-Out, Guest In/Out,
Change Room Clean Status, Room Shift, will be automatically assigned to the Extended IP Phone.
To know more about key templates, and for instructions on customizing them, read the DSS Keys
Programming topic.
If you want to customize the key map of this Extended IP Phone instead of applying a key template, select
the option Personalized, and configure the Phone Key Settings. See Configuring Matrix Extended Phone
Settings using Jeeves for instructions.
The template you assign will be applied on the Extended IP Phones registered at all three locations.
Assign a SIP Hardware Template to the SIP Extension. Default: 01. The SIP Hardware Template
contains voice quality related features such as Voice Codec selection, Tx and Rx Gains, Echo
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 assigned to all SIP Extensions as well as to
SIP Trunks.
Check if the values in this template fulfill requirements of the SIP Extension. If Template 01 fulfills the
feature requirements, retain Template 01.
If a different set of SIP hardware features are to be allowed to this SIP Extensions, prepare another
template and assign it to this extension. To do this,
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Select the number of the Template you customized, Template 02 in the SIP Hardware Template
field.
Also see the topic SIP Hardware Template to know more about customizing the templates and
applying on the SIP Extensions.
Assign a Station Basic Feature Template to the SIP Extension. Default: The Station Basic Feature
Template has a set of features like Time Table, Class of Service, Toll Control, Operator, Storage of
Incoming and Outgoing Calls, Outgoing Trunk Bundle groups. There are 50 different templates to
choose from. Each template can also be altered to suit your requirement and preferences.
If the default Station Basic Feature Template 01 fulfills the feature requirements of the SIP Extension
(Class of Service (COS), Toll Control, OG Trunk Bundle Group, etc.) retain this template, you may
also customize this template. If you want to assign a different set of features to this SIP Extension,
prepare a different Station Basic Feature Template and apply it to this extension. To do this,
Select the number of the Template you customized, Template 05, in the Station Basic Feature
Template field.
Also, see the topic Station Basic Feature Template to know more about customizing the templates
and applying on extensions.
Assign a Station Advanced Feature Template to the SIP Extension. Default: Template 01. The
Station Advanced Feature Template has a set of advanced features for extensions such as Message
Wait Notification and Alarm Notification settings, Routing of Incoming Auto Attendant Calls, Call
Duration Control, Floor Service, etc. There are 50 different templates to choose from. Each template
can also be altered to suit your requirement and preferences.
Check if the default template fulfills the feature requirements of the SIP Extension by clicking the
Station Advanced Feature Template link.
You may retain this template and customize it further, or customize another template if a different set of
features are to be allowed to this SIP Extension. To customize/prepare another template,
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Select the Template number, for example 02, and customize this template.
In the Station Advanced Feature Template field, select the number of the template you
customized.
Also see the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the extensions.
If you want to provide other features like Personal Directory, Priority, or assign a Station Type to the SIP
Extension, click the Advanced button at the bottom of the page.
Enter the Mobile Number of the extension user you wish to store. The Number can be a maximum of 16
digits.
Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of 64
characters.
You can assign the extension user to a Group. The system clubs together extension users assigned the
same Group. The Group can be a maximum of 16 characters. Default: Blank.
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Call Pick Up allows the SIP Extension user to 'pick up' (answer) calls ringing on any other extension, by
dialing a feature code, without physically going to the ringing extension. It also allows incoming calls for the
SIP Extension to be answered by the other extensions assigned the same Call Pick-Up group.
For this to work, both the ringing extension and the extension picking up the call must be in the same 'Call
Pick Up Group'. Refer Call Pick Up for instructions on how to create groups. You can create as many as
99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SIP Extension in this field.
If using the system in the Hotel Mode, select the Station Type for the SIP Extension as Administration or
Guest.
You may assign a Personal Directory number to the SIP Extension. Default: 00.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes ("Out Going Trunk Bundle Group Index"). The Personal
Directory is necessary for using the features Abbreviated Dialing and Dial By Name.
When a Personal Directory is assigned to a SIP Extension, make sure you also configure this directory.
The Personal Directory can be programmed by the SIP Extension users and by the System Engineer.
Refer the topic Abbreviated Dialing for instructions on programming the Personal Directory.
If Personal Directory is not to be assigned, enter 00 in this field.
Select a Priority Level for the SIP Extension from 1 to 9. Default; 5-Normal.
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an
extension with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description Priority.
If this SIP extension is assigned to Operator, you may want to set a higher priority for this extension.
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Select the check box Enable Matrix Extended Phone Mode. Default: Disabled.
Enter the MAC Address191 of the SPARSH VP248 connected at this location in hexadecimal format:
00:50:C2:55:B0:10. Default: blank.
ETERNITY validates the Extended Phone on the basis of the MAC Address, and provides configuration on
validation.
As ETERNITY allows registration of the SIP Extension from three different locations, it identifies the SIP
Extension in each location by the programmed MAC address.
Select the appropriate Registrar Server Address to register the SPARSH VP248 with the SIP Registrar of
ETERNITY, according to your installation scenario:
If the SPARSH VP248 is connected on the WAN network, select Use WAN Port IP Address as
Registrar Server IP Address.
191. MAC address is the address of the electronic hardware devices such as a computer, which is hard-coded into the device during
manufacture and cannot be modified. No two devices can have similar MAC address and thus it uniquely identifies your phone.
MAC address is assigned as per the IANA standard. The MAC Address of the phone will be used as source MAC address on all
Ethernet frames.
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If the SPARSH VP248 is connected on the LAN network, select Use LAN Port IP Address as
Registrar Server IP Address.
If the SPARSH VP248 is connected in the Global Network and ETERNITY is located behind a Router,
or behind a NAT Router and STUN is programmed, select Use Router/STUN's IP Address as
Registrar Server IP Address.
Make sure you configure either the Routers Public IP Address or Simple Traversal of UDPs
through NATs (STUN) in the VoIP Port Parameters. See Configuring VoIP Network.
If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as
Registrar Server IP Address.
By default, Use WAN Port IP Address is selected as the Registrar Server IP Address.
To set the call progress tone generation standards of the country where the SPARSH VP248 is installed,
select the Call Progress Tone - Region. Default: Region 1.
See Call Progress Tones to know more.
To display the Date and Time of the country where the SPARSH VP248 is installed, select the Date and
Time - Region. Default: India.
If you want to enable Daylight Saving Time (DST) on the phone, set Apply DST? to Yes. Default: No.
The Daylight Saving Time convention followed in the country/region you selected will be automatically
applied. The SPARSH VP248 will change its date and time settings according to the DST convention of the
selected country/region.
Select the CO CLIP Pattern for the SPARSH VP248. This is the type of Calling Line Presentation on the
phone for incoming calls from trunks. You can select any of these options:
Name Only (only the name of the caller will be displayed).
Number Only (only the number of the caller will be displayed).
Number + Name (both the name and the number of the caller will be displayed).
Default: Number + Name.
Select a Ringer Mode for the phone from the four options:
Ring immediately (it rings immediately as a fresh calls lands on the phone).
Ring if idle (rings only if the phone is idle).
Ring after a delay (if the call is still not answered).
Silent.
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If you selected Ring after a delay as Ringer Mode, set the Ring Delay Timer (sec), if required, to the
desired value.
The Ring Delay Timer is the time in seconds the system waits on receiving a call before ringing on the
phone. The range of this timer is 0 to 99 seconds. Default: 10 seconds.
If you want to enable Ringer Auto Acknowledge mode, set the Acknowledge Timer (sec) to the desired
value.
The Ringer Auto Acknowledge mode determines when to stop the ring on the phone. There are two
options for Ringer Auto Acknowledge:
Stop only when the call is answered.
Stop after a delay.
To stop the ring on the phone after a delay, the Acknowledge Timer must be configured. The range of this
timer is 01 to 99 seconds. Default: 00 seconds.
To stop the ring only when the Call is answered or manually acknowledged, the Acknowledge Timer must
be set to '00'. By default, Ring Auto Acknowledge is turned OFF.
To assign the Ring Destination for the SPARSH VP248, select the desired destination for Play Ring on.
You may choose
Speakerphone: The ring will be played on the Speakerphone.
Headset: The ring will be played on the Headset.
Default: Speakerphone.
When you select the Headset as the destination, make sure that you set the flag Headset Connected? to
Yes, connect a Headset to the SPARSH VP248.
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Select the desired Ring Tune according to your/SPARSH VP248 users preference. Default: 1.
Set the Ringer Volume to the desired level, from 0 to 7, according to your preference. Default: 4.
To increase/decrease the volume of outgoing speech (Transmit Gain) on the handset of the SPARSH
VP248, set the Handset Transmit Volume Level to the desired level, from 0 to 7. Default: 4.
To increase/decrease the volume of incoming speech (Receive Gain) on the handset of the SPARSH
VP248, set the Handset Receive Volume Level to the desired level, from 0 to 7. Default: 4.
To increase/decrease the volume of outgoing speech (Transmit Gain) on the headset of the SPARSH
VP248, set the Headset Transmit Volume Level to the desired level, from 0 to 7. Default: 4.
To increase/decrease the volume of outgoing speech (Receive Gain) on the headset of the SPARSH
VP248, set the Headset Receive Volume Level to the desired level, from 0 to 7. Default: 4.
To change the Transmit Gain of the Speakerphone MIC Volume, set Speaker Transmit Volume Level to
the desired level, from 0 to 7. Default: 4.
To change the Receive Gain of the Speakerphone MIC Volume, set Speaker Receive Volume Level to
the desired level, from 0 to 7. Default: 4.
To use a Headset with the SPARSH VP248, select the Headset Connected? check box. Default:
Disabled.
Make sure that you connect a Headset to the SPARSH VP248, if you enable this option.
Select the Auto Answer check box to enable this feature on the SPARSH VP248. Default: Disabled.
When you set the Auto Answer feature on the SPARSH VP248, the phone goes OFF-Hook automatically
after a preset period of time, without the extension user having to pick up the handset or press the speaker
or headset key. When you enable Auto Answer, you must configure the Auto Answer Timer.
If you enabled Auto Answer on the phone, set the Auto Answer Timer (sec) to the desired value.
This timer defines the time in seconds that the SPARSH VP248 should wait before going OFF-Hook to
auto answer a call. The range of this timer is 1 to 9 seconds. Default: 1 second.
Adjust the Backlight brightness of the phones LCD display, by setting the LCD Backlight Level to the
desired value, from 1 to 4. Default: 3.
Set the Back Light Off Timer (sec) to the desired value, if required, from 000 to 999 seconds. Default: 10
seconds.
Set the LCD Contrast Level to a level from 1 to 4 that is comfortable to you. Default: 3.
To personalize the key map of the SPARSH VP248192, click the Phone Key Settings link.
The key map of the Extended Phone opens in a new window on your screen.
192. To personalize the phone key settings, you select Personalized Key Template for the SIP Extension.
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In the Select Function Type list, select the function to be performed by the key. For example, you want to
use the key to call the Operator.
The Operator function is a Feature, so select the option FEATURE from the Select Function Type list
box.
From the Select Offset drop down list, all the features that can be assigned to keys are listed.
Select Operator from the list of features in the Select Offset box.
Click OK.
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To take a second example, if you want to assign Remote DND to the key currently assigned CO 2 key,
click the key.
In the Select Function Type list box, select the option SA Command, as Remote DND is a System
Administrator (SA) Command.
In the Select Offset box, select the option Set DND for remote station.
Click OK. The box closes. Remote DND feature will appear in abbreviated form as R-DND on the key
label.
Follow the same instructions to assign features to other DSS keys. Selecting the appropriate Function
Type and the Offset for each feature/function.
If you want assign a feature, select FEATURE as function type, and select the desired feature as Offset.
If you want to use the key to call a DKP or a SIP extension, select DKP or SIP Extension as Function
Type and select the number of the extension as Offset.
To assign direct access to a mobile trunk, select MOBILE as Function Type and the desired port number 1
or 2 as Offset.
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To assign direct access to a SIP Trunk, select SIP as Function Type and the desired trunk number from 1
to 4 as Offset.
Click OK, each time you select a Function Type and Offset in the dialog box.
You can reinstate default key assignment any time, by clicking the Default button at the bottom of the
window.
If you assign/re-assign functions to the following keys, the Phone will restart:
Speaker
Headset
Ringer
Acknowledge
Local Menu
If you have upgraded your SPARSH VP248 to an Extended IP Phone with firmware V5Rx, the capsense
key labels listed in the table below will have the following functions:
Key Label
Fwd Busy
Voice Mail
Fwd NR
DND
Forward
Reject
Release
RTP Port
RTP Listening Port: This is the port on which the SPARSH VP248 listens for SIP messages over TCP.
This port is also used as the source port for sending RTP packets. This port is also used as the source
port for sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default:
8000.
Quality of Service
Set the SIP Quality of Service (QoS) for SIP signaling as:
If the SPARSH VP248 is connected behind a NAT router, configure NAT Keep Alive.
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Select the check box Enable NAT Keep Alive to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
The time period you define should be less than the binding timer of the router.
Timers
SIP INVITE Timer (sec): This is the time in seconds that the phone waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the
called party and stops on receipt of the provisional response or the final response or when the user
disconnects the call. On expiry of the timer, the phone terminates the call process and gives an error
tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
SIP Provisional Timer (sec): This is the time in seconds that the phone waits for final response after
receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called
party or when the user disconnects the call. On expiry of the timer, the IP phone terminates the call
process and gives error tone to the user. The range of SIP Provisional Timer is 10-180 seconds.
Default: 60 seconds.
General Request Timer (sec): This is the time in seconds for which the phone waits for response of a
transaction request. This timer starts on initiating a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the phone clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20
seconds.
Debug
To debug using Syslog Client supported by the SPARSH VP248, configure Debug parameters:
Enter the IP Address and port of the remote Syslog Server and as Syslog Server Address and
Server Port.
The address of the Listening Port of the Syslog Server is from 1025-65535;514. Default: 514. Syslog
uses the UDP as transport protocol and listens on the port 514 (the default listening port).
You may select the Debug Level from the following options, by selecting the respective check box:
SIP
System
Hardware
Call
Network
VoPP
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You may select any or all of these debug levels. The Syslog Client will send only the debug messages
for the selected level to the remote server on the IP network. For example, if the debug log of 'Call's is
required, you can select this option, and disable all others.
If you have completed the configuration of the SPARSH VP248 Settings at Location 1, follow the same
steps as described above to configure the SPARSH VP248 at Location 2 and Location 3.
When you change any of the parameters listed below in the SIP Extension at Location 1, 2, 3, the phone
will restart automatically, if registered:
VoIP Port Number
Use SIP Extension
SIP ID
Authentication ID
Authentication Password
Registrar Server IP Address
MAC Address
Enable Matrix Extended Phone Mode
Extended Phone Type
Key Map in the Key Template assigned to phone
Call Progress Tone
Date and Time
Apply DST?
QoS
SIP/RTP Ports
NAT Keep Alive
SIP Timers
The SIP Extension registered at Location 1, 2, 3, will also restart, if:
The SE Password of ETERNITY is changed
The VoIP Card Parameters are changed
You restart the System
You restart the VoIP Card
Set the System to Default
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Ringer Mode is
1=Ring Immediately
2=Ring if idle
3=Ring after delay
4=Ring Off
Default: Ring Immediately
To configure the Ring Delay Timer (Sec), dial:
7911-1-SIP Extension - Location - Ring Delay Timer
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Ringer Timer is from 01 to 99 (00=Immediate)
Default: 10
To configure the Acknowledge Timer (Sec), dial:
7912-1-SIP Extension - Location - Acknowledge Timer
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Ack. Timer is from 01 to 99
Default: 00
To configure the Play Ring on option, dial:
7913-1-SIP Extension - Location - Ring Destination
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Ring Destination is
1=Speaker Phone
2=Headset
Default: Speaker Phone
To select the Ring Tune, dial:
7914-1-SIP Extension - Location - Ring Tune
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Ring Tune is from 0 to 9
Default: 1
To select Ring Volume, dial:
7915-1-SIP Extension-Location-Ringer Volume
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Ring Volume is 0 to 7.
Default: 4
To set Handset Transmit Volume level, dial:
7916-1-SIP Extension-Location-Handset Transmit volume
Where,
SIP Extension is from 001 to 999.
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Location is 1 to 3.
Handset Transmit Volume level is 0 to 7.
Default: 4
To set Handset Receive Volume level, dial:
7917-1-SIP Extension-Location-Handset Receive volume
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Handset Receive Volume level is 0 to 7.
Default: 4
To set Headset Transmit Volume level, dial:
7918-1-SIP Extension-Location-Headset Transmit volume
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Headset Transmit Volume level is 0 to 7.
Default: 4
To set Headset Receive Volume level, dial:
7919-1-SIP Extension-Location-Headset Receive volume
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Headset Receive Volume level is 0 to 7.
Default: 4
To set Speaker Transmit Volume level, dial:
7920-1-SIP Extension-Location-Speaker Transmit volume
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Speaker Transmit Volume level is 0 to 7.
Default: 4
To set Speaker Receive Volume level, dial:
7921-1-SIP Extension-Location-Speaker Receive volume
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Speaker Receive Volume level is 0 to 7.
Default: 4
To enable/disable Headset Connectivity, dial:
7922-1-SIP Extension-Location-Headset Connectivity flag
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
Headset connectivity flag is 0 for Disable, 1 for Enable.
Default: Disable
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Default: Disable
Exit SE mode.
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You may select the SIP Extension number you want to configure.
The parameters of the SIP Extension number you selected will appear on this page.
In the VoIP Port No. field, select the software port number of the VoIP Port to which you want to assign the
SIP Extension. For example, you want to assign SIP Extension 1 to VoIP Port number 2, select 02 as port
number from the list.
Upto 250 SIP Extensions can be registered with a single VoIP Card.
Select the Use SIP Extension check box to enable the SIP extension. Default: enabled.
You may clear this check box, when you want to deactivate the SIP extension.
In the Name field, enter a name for the SIP Extension, which may be the name of the person who will use
the SIP Extension or the name of a Department. The name you enter here will be displayed as the Caller
ID of the SIP Extension on the remote user's phone, when the SIP Extension user makes calls.
The name may consist of a maximum of 18 alphanumeric characters.
Enter the SIP ID for the extension. The SIP ID is necessary for registering the SIP Extension with the
Registrar of the VoIP Card. It is the number with which you can call the SIP Extension. Any extension user
of the ETERNITY can call a SIP Extension by dialing the SIP ID assigned to the SIP extension. SIP ID of
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each SIP Extension must be a unique number string of a maximum of 6 digits. Any combination of digits
from 0 to 9 and the characters * and # are allowed.
You cannot assign the same SIP ID to more than one extension.
To assign SIP IDs according to your preference and requirment to a range of SIP Extensions, see
Assigning Access Codes to a Range of Extensions.
By default, the SIP IDs are Blank.
The SIP ID will be set to default value (blank), when you restore the default settings of the system.
In the Authentication ID field, enter the number which you want the VoIP Card to use for user
authentication of the SIP messages received from the SIP Extension. The number may be a string of
maximum 6 digits. Any combination of digits from 0 to 9 and the characters * and # are allowed in the
string.
You must configure the Authentication ID, if any of the SIP Message Authentication Options, namely
REGISTER or INVITE or SUBSCRIBE or PUBLISH, is enabled.
If you do not configure the Authentication ID and the Authentication Password, by default, the system will
consider the SIP ID as the Authentication ID and 1234 as Authentication Password. You may change the
Authentication ID and the Authentication password as per your requirement.
In the Authentication Password field, enter the password to be used by the VoIP Card to authenticate
the SIP messages received from the SIP Extension. You can enter a maximum of 24 digits as password.
The valid digits for the password are 0 to 9, * and #. Default: Blank.
In Call Appearances, define the maximum number of simultaneous incoming calls that the SIP Extension
user should be allowed to receive. You can set up to 10 call appearances for a SIP Extension. Default: 2.
When Call Appearance is set to 2, the SIP Extension can receive 2 calls at a time.
Under Authentication, enable Authentication of any or all of the following SIP Message Options by
selecting the respective check boxes:
REGISTER Request
INVITE Request
SUBSCRIBE Request
By default, the SIP Message Options REGISTER, INVITE, SUBSCRIBE are enabled.
Make sure that the Authentication ID for the SIP Extension has been programmed, when any of the above
SIP Message Options are enabled.
For secure conversations over SIP, enable SRTP Mode. The VoIP card of ETERNITY supports the
following options:
Disable: ETERNITY uses normal RTP for transporting the speech packets.
Optional: ETERNITY uses SRTP for transporting the speech packets. If the remote user does not
support SRTP, ETERNITY will use normal RTP for transporting the speech packets.
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If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
Forced: ETERNITY uses only SRTP (SAVP) for transporting the speech packets. If the remote user
does not support SRTP, ETERNITY will reject incoming calls from and drop outgoing calls made to
such users.
By default, SRTP Mode is Disabled.
Select a Key Template for the extension. The Key Template may be of the Operator, Executive, Guest
or Hotel Attendant, according to the key map you want to assign to this extension. Default: Operators
Template.
Like the DKP, the SPARSH VP330 will function as Operator, Executive, Hotel Attendant, and Hotel Guest
extension, according to the key map template you assign. For example, if the SPARSH VP330 is to be
used by the Operator, select Operator's Template. The phone will be assigned the key template with the
special features required by Operators, such as more DSS keys for Trunk Access and Call Appearances,
a Call Release Key, etc.
Similarly, if the user of the SPARSH VP330 is a Hotel Attendant, select 'Hotel Attendant's Template'. The
key map with the specific Front Desk User features such as Check-In, Check-Out, Guest In/Out, Change
Room Clean Status, Room Shift, will be automatically assigned to the Extended IP Phone.
To know more about key templates, and for instructions on customizing them, read the DSS Keys
Programming topic.
If you want to customize the key map of this SPARSH VP330 instead of applying a key template, select the
option Personalized, and configure the Phone Key Settings. See Configuring Matrix Extended Phone
Settings using Jeeves for instructions.
The template you assign will be applied on the SPARSH VP330 Phones registered at all three locations.
Even if you assign keys for the following feature in the Key Templates, these features will not function:
Function Type
Offset
Macro
SA Command
Special Keys
Digit Pause
Digit A
Digit B
Digit C
Digit D
Enter
Local Menu
Feature
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Function Type
Offset
Minibar Details
Emergency Conference
Self Ring Test
SA Command Prefix
PMS - User Defined Fields
Door Phone - Call Routing Mode
Door Phone - Destination
DOP - Turned ON/ Turned OFF
Department Group Call Forward
Assign a SIP Hardware Template to the SIP Extension. Default: 01. The SIP Hardware Template
contains voice quality related features such as Voice Codec selection, Tx and Rx Gains, Echo
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 assigned to all SIP Extensions as well as to
SIP Trunks.
Check if the values in this template fulfill requirements of the SIP Extension. If Template 01 fulfills the
feature requirements, retain Template 01.
If a different set of SIP hardware features are to be allowed to this SIP Extensions, prepare another
template and assign it to this extension. To do this,
Select the number of the Template you customized, Template 02 in the SIP Hardware Template
field.
Also see the topic SIP Hardware Template to know more about customizing the templates and
applying on the SIP Extensions.
Assign a Station Basic Feature Template to the SIP Extension. Default: The Station Basic Feature
Template has a set of features like Time Table, Class of Service, Toll Control, Operator, Storage of
Incoming and Outgoing Calls, Outgoing Trunk Bundle groups. There are 50 different templates to
choose from. Each template can also be altered to suit your requirement and preferences.
If the default Station Basic Feature Template 01 fulfills the feature requirements of the SIP Extension
(Class of Service (COS), Toll Control, OG Trunk Bundle Group, etc.) retain this template, you may
also customize this template. If you want to assign a different set of features to this SIP Extension,
prepare a different Station Basic Feature Template and apply it to this extension. To do this,
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Select the number of the Template you customized, Template 05, in the Station Basic Feature
Template field.
Also, see the topic Station Basic Feature Template to know more about customizing the templates
and applying on extensions.
Assign a Station Advanced Feature Template to the SIP Extension. Default: Template 01. The
Station Advanced Feature Template has a set of advanced features for extensions such as Message
Wait Notification and Alarm Notification settings, Routing of Incoming Auto Attendant Calls, Call
Duration Control, Floor Service, etc. There are 50 different templates to choose from. Each template
can also be altered to suit your requirement and preferences.
Check if the default template fulfills the feature requirements of the SIP Extension by clicking the
Station Advanced Feature Template link.
You may retain this template and customize it further, or customize another template if a different set of
features are to be allowed to this SIP Extension. To customize/prepare another template,
Select the Template number, for example 02, and customize this template.
In the Station Advanced Feature Template field, select the number of the template you
customized.
Also see the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the extensions.
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If you want to provide other features like Personal Directory, Priority, or assign a Station Type to the SIP
Extension, click the Advanced button at the bottom of the page.
Enter the Mobile Number of the extension user you wish to store. The Number can be a maximum of 16
digits.
Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of 64
characters.
You can assign the extension user to a Group. The system clubs together extension users assigned the
same Group. The Group can be a maximum of 16 characters. Default: Blank.
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For this to work, both the ringing extension and the extension picking up the call must be in the same 'Call
Pick Up Group'. Refer Call Pick Up for instructions on how to create groups. You can create as many as
99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SIP Extension in this field.
If using the system in the Hotel Mode, select the Station Type for the SIP Extension as Administration or
Guest.
You may assign a Personal Directory number to the SIP Extension. Default: 00.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes ("Out Going Trunk Bundle Group Index"). The Personal
Directory is necessary for using the features Abbreviated Dialing and Dial By Name.
When a Personal Directory is assigned to a SIP Extension, make sure you also configure this directory.
The Personal Directory can be programmed by the SIP Extension users and by the System Engineer.
Refer the topic Abbreviated Dialing for instructions on programming the Personal Directory.
If Personal Directory is not to be assigned, enter 00 in this field.
Select a Priority Level for the SIP Extension from 1 to 9. Default; 5-Normal.
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an
extension with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description Priority.
If this SIP extension is assigned to Operator, you may want to set a higher priority for this extension.
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Select the check box Enable Matrix Extended Phone Mode. Default: Disabled.
Enter the MAC Address194 of the SPARSH VP330 connected at this location in hexadecimal format:
00:50:C2:55:B0:10. Default: blank.
ETERNITY validates the SPARSH VP330 on the basis of the MAC Address, and provides configuration on
validation.
As ETERNITY allows registration of the SIP Extension from three different locations, it identifies the SIP
Extension in each location by the programmed MAC address.
Select the appropriate Registrar Server Address to register the SPARSH VP330 with the SIP Registrar of
ETERNITY, according to your installation scenario:
If the SPARSH VP330 is connected on the WAN network, select Use WAN Port IP Address as
Registrar Server IP Address.
If the SPARSH VP330 is connected on the LAN network, select Use LAN Port IP Address as
Registrar Server IP Address.
194. MAC address is the address of the electronic hardware devices such as a computer, which is hard-coded into the device during
manufacture and cannot be modified. No two devices can have similar MAC address and thus it uniquely identifies your phone.
MAC address is assigned as per the IANA standard. The MAC Address of the phone will be used as source MAC address on all
Ethernet frames.
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If the SPARSH VP330 is connected in the Global Network and ETERNITY is located behind a Router,
or behind a NAT Router and STUN is programmed, select Use Router/STUN's IP Address as
Registrar Server IP Address.
Make sure you configure either the Routers Public IP Address or Simple Traversal of UDPs
through NATs (STUN) in the VoIP Port Parameters. See Configuring VoIP Network.
If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as
Registrar Server IP Address.
By default, Use WAN Port IP Address is selected as the Registrar Server IP Address.
To set the call progress tone generation standards of the country where the SPARSH VP330 is installed,
select the Call Progress Tone - Region. Default: Region 1.
See Call Progress Tones to know more.
To display the Date and Time of the country where the SPARSH VP330 is installed, select the Date and
Time - Region. Default: India.
If you want to enable Daylight Saving Time (DST) on the phone, set Apply DST? to Yes. Default: No.
The Daylight Saving Time convention followed in the country/region you selected will be automatically
applied. The IP phone will change its date and time settings according to the DST convention of the
selected country/region.
To personalize the key map of the SPARSH VP330195, click the Phone Key Settings link.
195. To personalize the phone key settings, you select Personalized Key Template for the SIP Extension.
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The key map of the Extended Phone opens in a new window on your screen.
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In the Select Function Type list, select the function to be performed by the key. For example, you want to
use the key to call the Operator.
The Operator function is a Feature, so select the option FEATURE from the Select Function Type list
box.
From the Select Offset drop down list, all the features that can be assigned to keys are listed.
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Select Operator from the list of features in the Select Offset box.
Click OK.
Follow the same instructions to assign features to other DSS keys. Selecting the appropriate Function
Type and the Offset for each feature/function.
If you want assign a feature, select FEATURE as function type, and select the desired feature as Offset.
If you want to use the key to call a DKP or a SIP extension, select DKP or SIP Extension as Function
Type and select the number of the extension as Offset.
To assign direct access to a mobile trunk, select MOBILE as Function Type and the desired port number 1
or 2 as Offset.
To assign direct access to a SIP Trunk, select SIP as Function Type and the desired trunk number from 1
to 4 as Offset.
Click OK, each time you select a Function Type and Offset in the dialog box.
You can reinstate default key assignment any time, by clicking the Default button at the bottom of the
window.
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The phone will enter the Auto Configuration mode, when you assign/re-assign certain features in the key
maps. To know more, refer to the SPARSH VP330 User Guide.
Select the protocol to be used to transport the SIP messages. You can select the Transport Mode as TCP
or TLS.
If you select TCP, make sure the SIP Over TCP check box is selected in VoIP Port Parameters.
If you select TLS, make sure the SIP Over TLS check box is selected in VoIP Port Parameters.
For secure conversations over SIP, select the Enable SRTP? check box. The SIP messages will be
transported over SRTP only.
RTP Port
RTP Listening Port: This is the port on which the phone listens for SIP messages over TCP. This port
is also used as the source port for sending RTP packets. This port is also used as the source port for
sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default: 8000.
Quality of Service
Set the SIP Quality of Service (QoS) for SIP signaling as:
If the SPARSH VP330 is connected behind a NAT router, configure NAT Keep Alive.
Select the check box Enable NAT Keep Alive to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
The time period you define should be less than the binding timer of the router.
Timers
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SIP INVITE Timer (sec): This is the time in seconds that the phone waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the
called party and stops on receipt of the provisional response or the final response or when the user
disconnects the call. On expiry of the timer, the phone terminates the call process and gives an error
tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
SIP Provisional Timer (sec): This is the time in seconds that the phone waits for final response after
receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called
party or when the user disconnects the call. On expiry of the timer, the IP phone terminates the call
process and gives error tone to the user. The range of SIP Provisional Timer is 10-180 seconds.
Default: 60 seconds.
General Request Timer (sec): This is the time in seconds for which the phone waits for response of a
transaction request. This timer starts on initiating a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the phone clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20
seconds.
If you have completed the configuration of the SPARSH VP330 Phone Settings at Location 1, follow the
same steps as described above to configure the SPARSH VP330 Phone at Location 2 and Location 3.
When you change any of the parameters listed below in the SIP Extension at Location 1, 2, 3, the phone
will go in Auto Configuration mode automatically, if registered:
Name
VoIP Port Number
Use SIP Extension
SIP ID
Authentication ID
Authentication Password
Registrar Server IP Address
MAC Address
Enable Matrix Extended Phone Mode
Extended Phone Type
Key Map in the Key Template assigned to phone
Language
Call Progress Tone
Date and Time
Apply DST?
QoS
SIP/RTP Ports
NAT Keep Alive
SIP Timers
The SIP Extension registered at Location 1, 2, 3, will also restart, if:
The SE Password of ETERNITY is changed
The VoIP Card Parameters are changed
You restart the System
You restart the VoIP Card
Set the System to Default
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Location is 1 to 3.
RTP Diffserve/ToS is 00 to 63.
Default: 46
To enable/disable NAT Keep Alive, dial:
7933-1-SIP Extension-Location-NAT Keep Alive flag
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
NAT KEEP Alive flag is 0 for Disable, 1 for Enable.
Default: Disabled.
To set NAT Keep Alive Interval, dial:
7934-1-SIP Extension-Location-NAT Keep Alive Interval
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
NAT KEEP Alive Interval is 001 to 999 seconds.
Default: 120.
To set SIP INVITE Timer, dial:
7935-1-SIP Extension-Location-SIP INVITE Timer
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
SIP INVITE Timer is 10 to 180 seconds.
Default: 30.
To set SIP Provisional Timer, dial:
7936-1-SIP Extension-Location-SIP Provisional Timer
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
SIP Provisional Timer is 10 to 180 seconds.
Default: 60.
To set SIP General Request Timer, dial:
7937-1-SIP Extension-Location-SIP General Request Timer
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
SIP General Request Timer is 10 to 60 seconds.
Default: 20.
Exit SE mode.
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You may select the SIP Extension number you want to configure.
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The parameters of the SIP Extension number you selected will appear on this page.
In the VoIP Port No. field, select the software port number of the VoIP Port to which you want to assign the
SIP Extension. For example, you want to assign SIP Extension 1 to VoIP Port number 2, select 02 as port
number from the list.
Upto 250 SIP Extensions can be registered with a single VoIP Card.
Select the Use SIP Extension check box to enable the SIP extension. Default: enabled.
You may clear this check box, when you want to deactivate the SIP extension.
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In the Name field, enter a name for the SIP Extension, which may be the name of the person who will use
the SIP Extension or the name of a Department. The name you enter here will be displayed as the Caller
ID of the SIP Extension on the remote user's phone, when the SIP Extension user makes calls.
Enter the SIP ID for the extension. The SIP ID is necessary for registering the SIP Extension with the
Registrar of the VoIP Card. It is the number with which you can call the SIP Extension. Any extension user
of the ETERNITY can call a SIP Extension by dialing the SIP ID assigned to the SIP extension. SIP ID of
each SIP Extension must be a unique number string of a maximum of 6 digits. Any combination of digits
from 0 to 9 and the characters * and # are allowed.
You cannot assign the same SIP ID to more than one extension.
To assign SIP IDs according to your preference and requirment to a range of SIP Extensions, see
Assigning Access Codes to a Range of Extensions.
By default, the SIP IDs are Blank.
The SIP ID will be set to default value (blank), when you restore the default settings of the system.
In the Authentication ID field, enter the number which you want the VoIP Card to use for user
authentication of the SIP messages received from the SIP Extension. The number may be a string of
maximum 6 digits. Any combination of digits from 0 to 9 and the characters * and # are allowed in the
string.
You must configure the Authentication ID, if any of the SIP Message Authentication Options, namely
REGISTER or INVITE or SUBSCRIBE or PUBLISH, is enabled.
If you do not configure the Authentication ID and the Authentication Password, by default, the system will
consider the SIP ID as the Authentication ID and 1234 as Authentication Password. You may change the
Authentication ID and the Authentication password as per your requirement.
In the Authentication Password field, enter the password to be used by the VoIP Card to authenticate
the SIP messages received from the SIP Extension. You can enter a maximum of 24 digits as password.
The valid digits for the password are 0 to 9, * and #. Default: Blank.
In Call Appearances, define the maximum number of simultaneous incoming calls that the SIP Extension
user should be allowed to receive. You can set up to 10 call appearances for a SIP Extension. Default: 2.
When Call Appearance is set to 2, the SIP Extension can receive 2 calls at a time.
Under Authentication, enable Authentication of any or all of the following SIP Message Options by
selecting the respective check boxes:
REGISTER Request
INVITE Request
SUBSCRIBE Request
By default, the SIP Message Options REGISTER, INVITE, SUBSCRIBE are enabled.
Make sure that the Authentication ID for the SIP Extension has been programmed, when any of the above
SIP Message Options are enabled.
For secure conversations over SIP, enable SRTP Mode. The VoIP card of ETERNITY supports the
following options:
Disable: ETERNITY uses normal RTP for transporting the speech packets.
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Optional: ETERNITY uses SRTP for transporting the speech packets. If the remote user does not
support SRTP, ETERNITY will use normal RTP for transporting the speech packets.
If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
By default, AVP is selected as the SRTP Media Type.
Forced: ETERNITY uses only SRTP (SAVP) for transporting the speech packets. If the remote user
does not support SRTP, ETERNITY will reject incoming calls from and drop outgoing calls made to
such users.
By default, SRTP Mode is Disabled.
Assign a SIP Hardware Template to the SIP Extension. Default: 01. The SIP Hardware Template
contains voice quality related features such as Voice Codec selection, Tx and Rx Gains, Echo
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 assigned to all SIP Extensions as well as to
SIP Trunks.
Check if the values in this template fulfill requirements of the SIP Extension. If Template 01 fulfills the
feature requirements, retain Template 01.
If a different set of SIP hardware features are to be allowed to this SIP Extensions, prepare another
template and assign it to this extension. To do this,
Select the number of the Template you customized, Template 02 in the SIP Hardware Template
field.
Also see the topic SIP Hardware Template to know more about customizing the templates and
applying on the SIP Extensions.
Assign a Station Basic Feature Template to the SIP Extension. Default: The Station Basic Feature
Template has a set of features like Time Table, Class of Service, Toll Control, Operator, Storage of
Incoming and Outgoing Calls, Outgoing Trunk Bundle groups. There are 50 different templates to
choose from. Each template can also be altered to suit your requirement and preferences.
If the default Station Basic Feature Template 01 fulfills the feature requirements of the SIP Extension
(Class of Service (COS), Toll Control, OG Trunk Bundle Group, etc.) retain this template, you may
also customize this template. If you want to assign a different set of features to this SIP Extension,
prepare a different Station Basic Feature Template and apply it to this extension. To do this,
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Select the number of the Template you customized, Template 05, in the Station Basic Feature
Template field.
Also, see the topic Station Basic Feature Template to know more about customizing the templates
and applying on extensions.
Assign a Station Advanced Feature Template to the SIP Extension. Default: Template 01. The
Station Advanced Feature Template has a set of advanced features for extensions such as Message
Wait Notification and Alarm Notification settings, Routing of Incoming Auto Attendant Calls, Call
Duration Control, Floor Service, etc. There are 50 different templates to choose from. Each template
can also be altered to suit your requirement and preferences.
Check if the default template fulfills the feature requirements of the SIP Extension by clicking the
Station Advanced Feature Template link.
You may retain this template and customize it further, or customize another template if a different set of
features are to be allowed to this SIP Extension. To customize/prepare another template,
Select the Template number, for example 02, and customize this template.
In the Station Advanced Feature Template field, select the number of the template you
customized.
Also see the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the extensions.
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If you want to provide other features like Personal Directory, Priority, or assign a Station Type to the SIP
Extension, click the Advanced button at the bottom of the page.
Enter the Mobile Number of the extension user you wish to store. The Number can be a maximum of 16
digits.
Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of 64
characters.
You can assign the extension user to a Group. The system clubs together extension users assigned the
same Group. The Group can be a maximum of 16 characters. Default: Blank.
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For this to work, both the ringing extension and the extension picking up the call must be in the same 'Call
Pick Up Group'. Refer Call Pick Up for instructions on how to create groups. You can create as many as
99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SIP Extension in this field.
If using the system in the Hotel Mode, select the Station Type for the SIP Extension as Administration or
Guest.
You may assign a Personal Directory number to the SIP Extension. Default: 00.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes ("Out Going Trunk Bundle Group Index"). The Personal
Directory is necessary for using the features Abbreviated Dialing and Dial By Name.
When a Personal Directory is assigned to a SIP Extension, make sure you also configure this directory.
The Personal Directory can be programmed by the SIP Extension users and by the System Engineer.
Refer the topic Abbreviated Dialing for instructions on programming the Personal Directory.
If Personal Directory is not to be assigned, enter 00 in this field.
Select a Priority Level for the SIP Extension from 1 to 9. Default; 5-Normal.
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an
extension with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description Priority.
If this SIP extension is assigned to Operator, you may want to set a higher priority for this extension.
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Select the check box Enable Matrix Extended Phone Mode. Default: Disabled.
Enter the IMEI Number198 of the Matrix SPARSH MS Android Application Phone.
If you are using an iPhone, enter the Device ID here. Default: blank.
ETERNITY validates the Matrix SPARSH MS Android/iPhone Application Phone on the basis of the IMEI
Number/Device ID, and provides configuration on validation.
198.
IMEI Number is the unique identification number of the GSM engine used in the Mobile handset.
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As ETERNITY allows registration of the SIP Extension from three different locations, it identifies the SIP
Extension in each location by the programmed IMEI Number/Device ID.
Select the appropriate Internal Registrar Server Address to register the Matrix SPARSH MS Android/
iPhone Application Phone with the SIP Registrar of ETERNITY within a private network. Select the
appropriate option as per your installation scenario:
If you want the Matrix SPARSH MS Android/iPhone Application Phone to register using the WAN
network, select Use WAN Port IP Address as the Internal Registrar Server Address.
If you want the Matrix SPARSH MS Android/iPhone Application Phone to register using the LAN
network, select Use LAN Port IP Address as Registrar Server Address.
If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as
Internal Registrar Server Address.
By default, Use WAN Port IP Address is selected as the Internal Registrar Server Address.
Select the appropriate External Registrar Server Address to register the Matrix SPARSH MS Android/
iPhone Application Phone with the SIP Registrar of ETERNITY from a public network. Select the option
according to your installation scenario:
If you want the Matrix SPARSH MS Android/iPhone Application Phone to register using the WAN
network, select Use WAN Port IP Address as External Registrar Server Address.
If the Matrix SPARSH MS Android/iPhone Application Phone is connected in the Public Network and
ETERNITY is located behind a Router, or behind a NAT Router and STUN is programmed, select Use
Router/STUN's IP Address as External Registrar Server Address.
Make sure you configure either the Routers Public IP Address or Simple Traversal of UDPs
through NATs (STUN) in the VoIP Port Parameters. See Configuring VoIP Network.
If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as
External Registrar Server Address.
By default, Use WAN Port IP Address is selected as the External Registrar Server Address.
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Select the protocol to be used to transport the SIP messages. You can select the Transport Mode as TCP
or TLS.
If you select TCP, make sure the SIP Over TCP check box is selected in VoIP Port Parameters.
If you select TLS, make sure the SIP Over TLS check box is selected in VoIP Port Parameters.
For secure conversations over SIP, select the Enable SRTP? check box. The SIP messages will be
transported over SRTP only.
RTP Port
RTP Listening Port: This is the port on which the phone listens for SIP messages over TCP. This port
is also used as the source port for sending RTP packets. This port is also used as the source port for
sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default: 8000.
Quality of Service
Set the SIP Quality of Service (QoS) for SIP signaling as:
If the Matrix SPARSH MS Android/iPhone Application Phone is connected behind a NAT router, configure
NAT Keep Alive.
Select the check box Enable NAT Keep Alive to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
The time period you define should be less than the binding timer of the router.
Timers
SIP INVITE Timer (sec): This is the time in seconds that the phone waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the
called party and stops on receipt of the provisional response or the final response or when the user
disconnects the call. On expiry of the timer, the phone terminates the call process and gives an error
tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
SIP Provisional Timer (sec): This is the time in seconds that the phone waits for final response after
receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called
party or when the user disconnects the call. On expiry of the timer, the IP phone terminates the call
process and gives error tone to the user. The range of SIP Provisional Timer is 10-180 seconds.
Default: 60 seconds.
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General Request Timer (sec): This is the time in seconds for which the phone waits for response of a
transaction request. This timer starts on initiating a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the phone clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20
seconds.
If you have completed the configuration of the Matrix SPARSH MS Android/iPhone Application Phone
Settings at Location 1, follow the same steps as described above to configure the Matrix SPARSH MS
Android/iPhone Application Phone at Location 2 and Location 3.
When you change any of the parameters listed below in the SIP Extension at Location 1, 2, 3, the phone
will go in Auto Configuration mode automatically, if registered:
Name
VoIP Port Number
Use SIP Extension
SIP ID
Authentication ID
Authentication Password
Registrar Server IP Address
IMEI Number
Enable Matrix Extended Phone Mode
Extended Phone Type
Language
QoS
SIP/RTP Ports
NAT Keep Alive
SIP Timers
The VoIP Card Parameters are changed
You restart the System
You restart the VoIP Card
The System is set to Default
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Where,
Transport Mode
1=TCP,
2=UDP,
Default: TCP
To define the SIP Listening Port, dial:
7929-1-SIP Extension-Location-SIP Listening Port
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
SIP Listening Port is 1025 to 65535.
Default: 5060
To define the RTP Listening Port, dial:
7930-1-SIP Extension-Location-RTP Listening Port
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
RTP Listening Port is 1025 to 65278.
Default: 5060
To set the QoS - SIP Diffserve/ToS level, dial:
7931-1-SIP Extension-Location-SIP Diffserve/ToS level
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
SIP Diffserve/ToS is 00 to 63.
Default: 26
To set the QoS - RTP Diffserve/ToS level, dial:
7932-1-SIP Extension-Location-RTP Diffserve/ToS level
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
RTP Diffserve/ToS is 00 to 63.
Default: 46
To enable/disable NAT Keep Alive, dial:
7933-1-SIP Extension-Location-NAT Keep Alive flag
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
NAT KEEP Alive flag is 0 for Disable, 1 for Enable.
Default: Disabled.
To set NAT Keep Alive Interval, dial:
7934-1-SIP Extension-Location-NAT Keep Alive Interval
Where,
SIP Extension is from 001 to 999.
Location is 1 to 3.
NAT KEEP Alive Interval is 001 to 999 seconds.
Default: 120.
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Exit SE mode.
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You may select the SIP Extension number you want to configure.
The parameters of the SIP Extension number you selected will appear on this page.
In the VoIP Port No. field, select the software port number of the VoIP Port to which you want to assign the
SIP Extension. For example, you want to assign SIP Extension 1 to VoIP Port number 2, select 02 as port
number from the list.
Upto 250 SIP Extensions can be registered with a single VoIP Card.
Select the Use SIP Extension check box to enable the SIP extension. Default: enabled.
You may clear this check box, when you want to deactivate the SIP extension.
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In the Name field, enter a name for the SIP Extension, which may be the name of the person who will use
the SIP Extension or the name of a Department. The name you enter here will be displayed as the Caller
ID of the SIP Extension on the remote user's phone, when the SIP Extension user makes calls.
The name may consist of a maximum of 18 alphanumeric characters.
If no name is assigned to the SIP Extension, the system will display the name received in the INVITE
message from the SIP Extension user when making outgoing calls.
Enter the SIP ID for the extension. The SIP ID is necessary for registering the SIP Extension with the
Registrar of the VoIP Card. It is the number with which you can call the SIP Extension. Any extension user
of the ETERNITY can call a SIP Extension by dialing the SIP ID assigned to the SIP extension. SIP ID of
each SIP Extension must be a unique number string of a maximum of 6 digits. Any combination of digits
from 0 to 9 and the characters * and # are allowed.
You cannot assign the same SIP ID to more than one extension.
By default, the SIP IDs are Blank.
The SIP ID will be set to default value (blank), when you restore the default settings of the system.
In the Authentication ID field, enter the number which you want the VoIP Card to use for user
authentication of the SIP messages received from the SIP Extension. The number may be a string of
maximum 6 digits. Any combination of digits from 0 to 9 and the characters * and # are allowed in the
string.
You must configure the Authentication ID, if any of the SIP Message Authentication Options, namely
REGISTER or INVITE or SUBSCRIBE or PUBLISH, is enabled.
If you do not configure the Authentication ID and the Authentication Password, by default, the system uses
the SIP ID as the Authentication ID and 1234 as Authentication Password. You may change the
Authentication ID and the Authentication password as per your requirement.
In the Authentication Password field, enter the password to be used by the VoIP Card to authenticate
the SIP messages received from the SIP Extension. You can enter a maximum of 24 digits as password.
The valid digits for the password are 0 to 9, * and #. Default: Blank.
In Call Appearances, define the maximum number of simultaneous calls that the SIP Extension user
should be allowed to make/receive. You can set up to 10 call appearances for a SIP Extension. Default: 2.
When Call Appearance is set to 2, the SIP Extension can make/receive 2 calls at a time.
Under Authentication, enable Authentication of any or all of the following SIP Message Options by
selecting the respective check boxes:
REGISTER Request
INVITE Request
SUBSCRIBE Request
By default, the SIP Message Options REGISTER, INVITE, SUBSCRIBE are enabled.
Make sure that the Authentication ID for the SIP Extension has been programmed, when any of the above
SIP Message Options are enabled.
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If you are going to register this SIP extension with the same SIP ID at more than one location199, you may
enable the Shared Call Appearance Subscription check box on this SIP Extension. Default: Disabled.
Shared Call Appearance provides notification on call states to all the phones registered with the same SIP
ID at different locations. To know more about this feature, see Shared Call Appearance.
To provide voice mail facility to the SIP Extension, select the Voice Mail Subscription check box. Default:
disabled.
To allow the SIP Extension to monitor the status of another extension or Trunk, select the Busy Lamp
Field200 Subscription check box. Default: disabled. SeeBusy Lamp Field for Trunks to know more.
When extension's state is changed from Ringing (early state as defined in BLF) to Mature (confirm state,
as defined in BLF) state, because of implementation of ETERNITY, it will send 'Terminate' state while
moving from ringing to mature state. The interpretation of terminate message will vary from terminal to
terminal.
To allow the SIP Extension user to view the status of other SIP-enabled Terminals, whether they are
available or not, select the Presence Subscription check box. Default: disabled.
The SIP Extension, for which you have enabled Presence Subscription, will be able to view Presence of
only those SIP Extensions which have PUBLISH enabled.
To allow the SIP Extension user to publish presence using the PUBLISH feature supported by the SIP
Client, select the PUBLISH Enable check box. Default: disabled.
By default, Authentication for PUBLISH message is enabled. You may disable if you do not want to use
Authentication.
You must configure the Authentication ID, if you have enabled both Publish and Authentication.
For secure conversations over SIP, enable SRTP Mode. The VoIP card of ETERNITY supports the
following options:
Disable: ETERNITY uses normal RTP for transporting the speech packets.
Optional: ETERNITY uses SRTP for transporting the speech packets. If the remote user does not
support SRTP, ETERNITY will use normal RTP for transporting the speech packets.
If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
By default, AVP is selected as the SRTP Media Type.
199. ETERNITY allows you to register SIP phones with the same SIP ID at three different locations.
200. Busy Lamp Field (BLF), a typical feature supported by PCM/TDM PBX and Key Telephone Systems, is also supported on SIP
Extensions.
In PCM/TDM PBX and Key Telephone Systems, this feature is typically used by the Operator to monitor the status of another
extension, that is, whether it is available, ringing or busy. The status of the other extensions is indicated on the special function
keys programmed on the Operator's console. This helps the Operator decide whether to place the call, or transfer the call to that
extension, or pick up the call ringing on that extension.
With BLF Subscription enabled on the SIP Extension, the user can monitor the status of another extension or trunk.
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Forced: ETERNITY uses only SRTP (SAVP) for transporting the speech packets. If the remote user
does not support SRTP, ETERNITY will reject incoming calls from and drop outgoing calls made to
such users.
By default, SRTP Mode is Disabled.
Key Templates are not applicable to Open SIP Phones registered with ETERNITY.
Assign a SIP Hardware Template to the SIP Extension. Default: 01. The SIP Hardware Template
contains voice quality related features such as Voice Codec selection, Tx and Rx Gains, Echo
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 assigned to all SIP Extensions as well as to
SIP Trunks.
Check if the values in this template fulfill requirements of the SIP Extension. If Template 01 fulfills the
feature requirements, retain Template 01.
If a different set of SIP hardware features are to be allowed to this SIP Extensions, prepare another
template and assign it to this extension. To do this,
Assign a Station Basic Feature Template to the SIP Extension. Default: The Station Basic Feature
Template has a set of features like Time Table, Class of Service, Toll Control, Operator, Storage of
Incoming and Outgoing Calls, Outgoing Trunk Bundle groups. There are 50 different templates to
choose from. Each template can also be altered to suit your requirement and preferences.
If the default Station Basic Feature Template 01 fulfills the feature requirements of the SIP Extension
(Class of Service (COS), Toll Control, OG Trunk Bundle Group, etc.) retain this template, you may
also customize this template. If you want to assign a different set of features to this SIP Extension,
prepare a different Station Basic Feature Template and apply it to this extension. To do this,
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Select the number of the Template you customized, Template 05, in the Station Basic Feature
Template field.
Also, see the topic Station Basic Feature Template to know more about customizing the templates
and applying on extensions.
Assign a Station Advanced Feature Template to the SIP Extension. Default: Template 01. The
Station Advanced Feature Template has a set of advanced features for extensions such as Message
Wait Notification and Alarm Notification settings, Routing of Incoming Auto Attendant Calls, Call
Duration Control, Floor Service, etc. There are 50 different templates to choose from. Each template
can also be altered to suit your requirement and preferences.
Check if the default template fulfills the feature requirements of the SIP Extension by clicking the
Station Advanced Feature Template link.
You may retain this template and customize it further, or customize another template if a different set of
features are to be allowed to this SIP Extension. To customize/prepare another template,
Select the Template number, for example 02, and customize this template.
In the Station Advanced Feature Template field, select the number of the template you
customized.
Also see the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the extensions.
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If you want to provide other features like Personal Directory, Priority, or assign a Station Type to the SIP
Extension, click the Advanced button at the bottom of the page.
If using the system in the Hotel Mode, select the Station Type for the SIP Extension as Administration or
Guest.
You may assign a Personal Directory number to the SIP Extension. Default: 00.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes ("Out Going Trunk Bundle Group Index"). The Personal
Directory is necessary for using the features Abbreviated Dialing and Dial By Name.
When a Personal Directory is assigned to a SIP Extension, make sure you also configure this directory.
The Personal Directory can be programmed by the SIP Extension users and by the System Engineer.
Refer the topic Abbreviated Dialing for instructions on programming the Personal Directory.
If Personal Directory is not to be assigned, enter 00 in this field.
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Select a Priority Level for the SIP Extension from 1 to 9. Default; 5-Normal.
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an
extension with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description Priority.
If this SIP extension is assigned to Operator, you may want to set a higher priority for this extension.
General Parameters
To select Source Port IP Address, dial:
7830-1-VoIP Port-Source Port IP Address Option
Where,
VoIP Port is the software port number of the VoIP Port from 01 to 16.
Source Port IP Address Options are
1 for Use IP Address of VoIP Ethernet Port.
2 for Use IP Address Fetched using STUN.
3 for Use Routers Public IP Address.
By default, Use IP Address of VoIP Ethernet Port is selected.
To set the Maximum Registration Timer, dial:
7831-1-VoIP Port - Maximum Registration Timer
Where,
VoIP Port is the software port number from 01 to 16.
Maximum Registration Timer is from 10 to 99999 seconds.
By default, it is set to 3600 seconds.
To set the Minimum Registration Timer, dial:
7832-1-VoIP Port-Minimum Registration Timer
Where,
VoIP Port is the software port number from 01 to 16.
Minimum Registration Timer is from 10 to 99999 seconds.
By default, it is set to 45 seconds.
To configure Private Key, dial:
7833-1-VoIP Port-Private Key
Where,
VoIP Ethernet Port is the software port number from 01 to 16.
The Private Key is any string of a maximum of 24 characters. All ASCII characters are allowed.
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a Trunk - works like a trunk interface when any of the extensions of the PBX makes an outgoing call
through it.
OR
a Tie Line - takes on a dual personality: functioning as both a station and a trunk. The E&M port works like
an extension interface for incoming calls. It works like a trunk interface when any extension makes an
outgoing call through it.
This dual function is used in PBXs that are used as Transit Exchanges as in a PLCC Network. Read
PLCC-An Introduction to know more.
E&M Trunk Seizure Type201: Immediate, Immediate with Ack, Immediate + Wink, Immediate with
Ack+Wink (MFCR2)202, Seizure Pulse, Seizure Pulse + Wink, Express, and Radio.
Address Signaling: Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The E&M Interface (Type IV and Type V connection) and the Speech Interface (2-wire speech or 4-wire)
are selected at the time of installation by changing the Jumper settings.
201. This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.
202. Currently supported only on ETERNITY GE E&M4 Card.
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Configure the following parameters for each E&M port on this page:
E&M Port No.: This non-editable field is the software port number of the E&M Port. Refer the topic
Software Port and Hardware ID to know more.
H/w (Hardware) Slot - Port: 'Slot' is the number of the number of the Universal Slot in which the E&M
Card is inserted. 'Port' is the number of the E&M hardware port on which the Tie Line equipment (PBX,
Router, Leased Line, etc.) is connected.
The ETERNITY can automatically detect and assign the hardware slot and port numbers automatically
to the E&M software ports.
For example: if you have inserted the card E&M8 in Slot 05 and E&M4 card in Slot 06 of ETERNITY
ME16S, the system will assign the Hardware Slot 05 and port numbers 01-08 to the E&M Software
Ports from 001 to 008 respectively. The system will assign hardware Slot 6 and port numbers 01-04 to
the E&M Software Ports 009 to 012. Refer the topic Software Port and Hardware ID to know more.
However, if required, you may change the Hardware Slot and Port assigned to the E&M software port.
In which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
Enable Port: This flag is for enabling or disabling an E&M port. When an E&M port is disabled, neither
incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning.
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Name: You may assign a Name to the E&M Port. Whenever there is an incoming call on this Port, this
name will be displayed on the destination extension, while receiving the call.
The Name may comprise a maximum of 18 characters.
E&M Feature Template: This parameter is applicable to all E&M Ports. The E&M Feature Template is
a complete set of E&M features to be applied on E&M Ports according to their 'Orientation Type',
whether they are Stations, Trunks or Tie-Lines.
By default E&M Feature Template 01 is applied on all E&M Ports. This template has 'Station' as the
default Orientation Type.
If all the E&M Ports are to be programmed as 'Stations' retain this template.
If all the E&M Ports are to be programmed as 'Trunks' use the default E&M Feature Templates 09 and
10 which have 'Trunks' as Orientation Type.
If some of the E&M Ports are to be programmed as Stations, some as Trunks and yet others as Tie
Lines, prepare different E&M Feature Template for each Orientation Type and apply them to the related
ports.
Apply the E&M Feature Template you customized to the E&M Port by entering the template number
in the 'E&M Feature Template' field of this port.
Repeat the same steps to customize another template and apply it to another E&M Port.
Refer the topic E&M Feature Template for more details on customizing the templates and applying
them on E&M Ports.
Trunk Feature Template: This parameter is relevant only if the E&M Port is to be programmed to
function as a Trunk or a Tie-Line203. To know more, refer E&M Feature Template.
Assign a Trunk Feature Template to the E&M Port. A Trunk Feature Template is a set of features like
Time Table, Operator, Auto Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR
Storage, etc., that defines the behavior of a Trunk. Apply a Trunk Feature Template to the E&M Port.
203. To program the E&M Port as a Trunk or a Tie-Line, you must set the 'Orientation Type' of the E&M Port to 'Trunk' or 'Tie-Line' in the
E&M Feature Template applied on the port.
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By default, Trunk Feature Template 01 is applied on all E&M (trunk) Ports. This Template is also the
default template commonly applied on all other trunk types (CO, ISDN BRI, ISDN T1E1PRI, GSM, and
VoIP).Refer the topic Trunk Feature Template to know more.
Click the link Trunk Feature Template to open the page. Check if the default Template 01 fulfills your
requirement for the E&M Trunk port.
If not, you may prepare a different Trunk Feature Template and apply on all E&M Ports. For this,
If the default Template 01 does not fulfill your requirement,
You may prepare a different Trunk Feature Template and apply on all E&M Ports. For this,
Go to the E&M Software Port Number you want to which you want to assign the Template you
prepared.
Enter the number of the Template you prepared (04) in the Trunk Feature Template field.
You may also prepare different Templates for different E&M Ports, for example Template 04 for certain
ports, Template 05 for others. In which case, for each E&M Port, enter the number of the template you
have prepared for that port.
To know more about customizing templates, refer the topic Trunk Feature Template.
Station Basic Feature Template: This parameter is applicable on when the E&M Port is to be
programmed to function as a Station (Orientation type = Station or Tie-Line).
Assign a Station Basic Feature Template to the E&M Port functioning as a Station. By default, Station
Basic Feature Template 01 is assigned to all extensions, that includes SLT and DKP ports.
Check if the default template fulfills the feature requirements (like Class of Service (COS), Toll
Control, OG Trunk Bundle Group, etc.) of the E&M Ports functioning as Stations.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all E&M (station) ports, retain Template 01.
If different sets of features are to allowed to different E&M (station) Ports, then prepare separate
Station Basic Feature Templates and apply them on the ports. To do this,
Click the link Station Basic Feature Template to open the page.
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Customize Template number 05 and click Submit at the bottom of the page.
Enter the number of the Template you customized, Template 05 in the 'Station Basic Feature
Template' field of the E&M Port (for example: E&M No. 003) on which you want to apply this
template. If you want to apply this template to other ports too, like E&M No. 004, 005, and 006,
assign the Template 05 to all these ports.
Repeat the same steps to customize and assign a different Template to another E&M port.
Also, refer the topic Station Basic Feature Template to know more about customizing the templates
and applying on the ports.
Station Advanced Feature Template: This parameter is applicable only when the E&M Port is to be
programmed to function as a Station.
By default Station Advanced Feature Template 01 is assigned to all extensions, that includes SLT and
DKP ports as well as E&M ports with the orientation type 'Station'.
Check if this default template fulfills the feature requirements of the E&M Ports (with 'station' as
orientation type) by selecting the 'Station Advanced Feature Template' link.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to
all E&M (station) ports, retain Template 01.
If different sets of features are to be allowed to different E&M (station) Ports, then prepare separate
Station Advanced Feature Templates and apply them on the ports.
To do this,
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Click the Station Advanced Feature Template link to open the page.
Enter the number of the Template you customized, Template 03 in the 'Station Basic Feature
Template' field of the E&M Port (for example, E&M No. 003) on which you want to apply this
template. If you want to apply this template to other ports too, like E&M No. 004, 005, and 006,
assign the Template 03 to all these ports.
Repeat the same steps to customize and assign a different Template to another E&M (Station) port.
Also refer the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the ports.
Priority: This parameter is applicable only when the E&M Port is to be programmed to function as a
Station204. To know more, refer E&M Feature Template.
Each station of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls
an extension with lower priority, a triple ring is placed on the called extension. To know more, read the
feature description Priority.
By default, the Priority of all E&M Ports functioning as Stations is set to '9-Highest'. So, decide what
Priority Level you will assign to each of the E&M Ports functioning as Stations and set the desired level
for each port.
Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the E&M
Trunk port.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a
preference number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same
preference must be assigned the same Cost Factor. Different trunk types can also be assigned the
same Cost Factor. These trunks are used for routing calls.
Assign a Cost Factor to the E&M Trunk port, for example, 03 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the E&M Trunk Port
001 only,
You must first assign a Cost Factor (01-99) to E&M Trunk Port 001, for example 03.
Click the Least Cost Routing - Number Based link to open the page.
Enter '6' in the 'Number' column, Cost Factor '03' as Preference 1, 2, 3 and 4.
If you have completed configuration of all the above listed E&M Parameters, click 'Submit' at the
bottom of the page to save your changes.
204. Recall that E&M ports can function as trunks, as stations and have both functions. When an E&M Port is programmed as a Station
Interface, it can only receive incoming calls. To program the E&M Port as a Station, you must set the 'Orientation Type' of the E&M
Port to 'Station' in the E&M Feature Template applied on the port.
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Where,
E&M is the Software Port number of the E&M port from 001 to 128.
Slot is Slot number in which the E&M Card is inserted from 01 to 16
Port Offset is the number of the E&M Port on the card from 01 to 99.
To clear the Hardware ID assigned to an E&M Software Port, dial:
1105-E&M-00-00
To enable/disable an E&M Port, dial:
3321-1-E&M-Code to enable/disable a single E&M port.
3321-2-E&M-E&M-Code-#* to enable/disable a range of E&M ports.
3321-*-Code to enable/disable all E&M ports.
Where,
E&M is the Software Port number of the port from 001 to 128.
Code is
0 for Disable
1 for Enable
Default: Enable
To program a Name for an E&M Port, dial:
5406-1-E&M-Name-#* to program a name for a single E&M port.
5406-2-E&M-E&M-Name-#* to program the same name for a range of E&M ports.
5406-*-Name-#* to program the same name for all E&M ports.
Where,
E&M is the Software Port number of the port from 001 to 128.
Name is a string of alphanumeric characters 18 characters (maximum).
Terminate the command with #* if the name string has fewer than 18 characters.
To clear a name assigned to an E&M port, dial:
5406-1-E&M-#* to clear the name of a single E&M port.
5406-2-E&M-E&M-#* to clear the names for a range of E&M ports.
5406-*-#* to clear the names of all E&M ports.
To assign an E&M Feature Template to an E&M Port, dial:
6003-1-E&M-Template Number to assign a template to a single port.
6003-2-E&M-E&M-Template Number to assign the same template to a range of ports.
6003-*-Template Number to assign the same template to all ports.
Where,
E&M is the Software Port number of the port from 001 to 128.
Template Number is from 01 to 50.
Default: E&M Feature Template 01.
To assign a Trunk Feature Template to an E&M Port, dial:
5805-1-E&M-Template Number to assign a template to a single port.
5805-2-E&M-E&M-Template Number to assign the same template to a range of ports.
5805-*-Template Number to assign the same template to all ports.
Where,
E&M is the Software Port number of the port from 001 to 128.
Template Number is from 01 to 50.
Default: Trunk Feature Template 01.
To assign a Station Basic Feature Template to an E&M Port, dial:
5505-1-E&M-Template Number to assign a template to a single port.
1268
Exit SE mode.
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Ring is played on the Magneto Field Telephone connected to the Magneto Port.
The extension user gets Ring Back Tone for the duration of the Ring Back Tone Timer.
The extension user must press # or the Magneto Ring Enable (MRE) Key (on the DKP) before the expiry of
the Ring Back Tone Timer (programmable; default: 45 seconds) to check if the Magneto User has
answered the call205.
The system considers pressing of the # or MRE Key as call maturity and connects the speech path
between the Magneto Field Telephone (connected to the Magneto Port) and the extension phone.
If the Magneto User has answered the call, the extension user may start speech on hearing the Magneto
User's voice.
If there is a period of silence, the extension user may press the # or MRE Key again to generate Ring on
the Magneto Telephone, and press the # and MRE Key once again during the Ring Back Tone Timer to
check speech with the Magneto User.
205. As there is no Answer Signaling or Call Disconnect feature on the Magneto line, there is no way for the Extension user to know
whether speech has been established with the Magneto User, but to wait to hear the Magneto User's voice.
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The extension user either goes ON-Hook while in speech. Error Tone will be played to the Magneto
user for the duration of the Error Tone Timer and the system releases the Magneto Port.
The Magneto User cranks the hand generator (Ring Down Signal) while in speech to disconnect the
call. The system detects the Ring Current and releases the Magneto Port.
If the Magneto port is busy, the extension user will hear the Busy Tone.
If the extension user goes ON-Hook while the Magneto Phone is ringing, the system will stop the ring
on the Magneto phone.
The Magneto Field Telephone user cranks the hand generator to generate ringing current.
The system detects the ringing current. If the ring current is present for more than 500msec, the system
treats it as a Call request from the Magneto Port.
The system rings the Operator extension assigned to the Magneto Port for the duration of the Ring Back
Tone Timer and plays Ring Back Tone to the Magneto Telephone user.
If the Operator goes OFF-Hook, before the expiry of the Ring Back Tone Timer, speech is established
between the Operator and the Magneto User.
The Operator can now transfer the call to another extension or external number.
The Operator can go ON-Hook while in speech to disconnect. Error Tone will be played to the Magneto
user.
OR
The Magneto User can cranks the hand generator again (Ring Down Signal) while in speech to
disconnect the call.
The system releases the Magneto Port and the port with which the Magneto User is in speech.
If neither the Operator nor any of the two Magneto station users in speech disconnects the call, the system
will automatically disconnect the call if silence (no speech) is detected for more than a specified duration of
time.
For this, the flag 'Enable Silence Detection on Magneto' must be enabled in the System Parameters and
the 'Magneto Silence Disconnection Timer (default: 60 seconds) must be programmed. When Silence
Disconnection is enabled on Magneto port, the system will start the Magneto Silence Disconnection Timer
as soon as it detects silence. If the continuous silence is detected till the expiry of the timer the system
considers the conversation as over and releases the Magneto stations. The call is disconnected. However,
if speech is detected during this timer, the timer will be stopped.
The call can be disconnected also using Forced Release by any extension user (Operator or any other
extension user) having the feature 'Forced Release' in its Class of Service.
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The Magneto user may crank the hand generator again to send Ring Current on the expiry of the Ring
Back Tone Timer. However, if the Magneto user sends the Ring Current again during the Ring Back
Tone Timer, the system will treat it as a call disconnect event. It will stop ringing the Operator's
extension and release the Magneto Port.
The Magneto Ring Enable (MRE) Key should be programmed on the DKP extensions that are to be
allowed access to call Magneto ports.
On SLT extensions and ISDN Terminals, the # key will serve the function of the MRE key.
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Magneto No.: This non-editable field is the number of the software port of the Magneto port.
HW Slot-Port: 'Slot' is the number of the Universal Slot in which the Magneto Card is inserted. 'Port' is
the number of the Magneto hardware port on which the Magneto Field telephone instrument is
connected.
The ETERNITY can automatically detect and assign the hardware slot and port numbers automatically
to the Magneto software ports.
However, if required, you may change the Hardware Slot and Port assigned to the Magneto software
port. In which case, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
Enable Port: This flag is for enabling or disabling a Magneto port. When a Magneto port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by selecting the check
box.
Access Code: Assign Station Access Codes to the Magneto Port. Station Access Codes are
commonly referred to as Extension Numbers. These may be number strings of a maximum 6 digits,
which are to be dialed to call the Magneto port to which they are assigned.
A maximum of 6 digits are allowed in an Access Code. By default, the Station Access Codes are blank
for all Magneto ports.
To assign Station Access Codes according to your preference and requirment to a range of Magneto
Ports, see Assigning Access Codes to a Range of Extensions.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics Access Codes and Conflict Dialing to know
more.
Name: Assign a 'Name' to the Magneto port. The name may be of the person who will use the Magneto
telephone or the name of the department or location of the telephone. This name will be displayed on
the LCD of the Operator/extension user's phone, if it is equipped with Caller ID.
You can program a name of a maximum of 18 alphanumeric characters.
Station Basic Feature Template: As the Magneto Port functions as a station, assign a Station Basic
Feature Template to the Magneto port.
Only the following features of the Station Basic Feature Template are applied on the Magneto Port:
Time Table
Operator
Class of Service
By default, Station Basic Feature Template number 01 is assigned to all extension types of the
ETERNITY (SLT, DKP, ISDN Terminals, E&M Lines with Station as Orientation Type). Template 01 is
also applied on Magneto ports by default.
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Check if the default settings of the features applied on the Magneto ports (Time Table, Operator, Class
of Service, Storage of Outgoing and Incoming Calls) match your requirements for the Magneto ports. If
yes, retain the default Station Basic Feature Template 01.
If you want to change any of the feature settings in for the Magneto Ports, you may prepare a different
Template206, for example, Template 14 and apply it on the Magneto Ports.
Also, if different feature settings are to be applied on different Magneto Ports prepare separate Station
Basic Feature Templates and apply them on the ports. To do this,
Click the Station Basic Feature Template link to open the page.
Customize the Magneto Port related features (listed above) in Template number 14 and click
Submit to save.
Enter the number of the Template you customized, Template 14 in the 'Station Basic Feature
Template' field of the Magneto Port, for example: MAG-001, on which you want to apply this
template. If you want to apply this template to other ports too, like MAG-002, 003, and 004, assign
the Template 14 to all these ports.
If required, repeat the same steps to customize and assign a different Template to another Magneto
port.
Refer the topic Station Basic Feature Template to know more about customizing the templates and
applying on the ports.
Station Advance Feature Template: Assign a Station Advanced Feature Template to the Magneto
Port.
Only the following features in Station Advance Feature Templates are applied on Magneto Ports.
By default Station Advanced Feature Template 01 is assigned to all extensions of the ETERNITY,
which also includes DKP ports, SLT ports, ISDN Terminals and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the Magneto Ports by opening the link
'Station Advanced Feature Template'.
Check if the default settings of the features applied on the Magneto ports (Internal Call Storage flag
and Call Forward Ring Timer) fulfill the feature requirements of the Magneto ports, by opening the
'Station Advanced Feature Template' link.
If the default template fulfills your requirement, retain the default Station Basic Feature Template 01 for
the Magneto ports.
206. This is recommended because changing the values of the default Template will be applied on all other extension types to which the
Template is assigned.
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However, if you want to change any of the feature settings in for the Magneto Ports, you may prepare a
different Template207, for example, Template 05 and apply it on the Magneto Ports. To do this,
Click the Station Advanced Feature Template link to open the page.
Enter the number of the Template you customized, Template 05 in the 'Station Advanced Feature
Template' field of the Magneto Port, for example, MAG-001, on which you want to apply this
template. If you want to apply this template to other terminals too, like MAG-002, 003, and 004,
assign the Template 05 to all these ports.
Repeat the same steps to customize and assign a different Template to another Magneto Port.
Also refer the topic Station Advanced Feature Template for instructions on customizing these
templates and applying them on the station ports.
Priority: Assign a Priority Level from 1 to 9 to the Magneto Port, with '1' being lowest Priority and '9'
being highest Priority. Whenever a Magneto Port with higher priority calls an extension with lower
priority, a triple ring is placed on the called extension. To know more, read the feature description
Priority.
By default, the Priority of all Magneto Ports is set to '5-Normal'.
If you have completed configuration of the desired Magneto Ports, click Submit at the bottom of the
page to save your settings.
Now, configure the Magneto Ring Enable (MRE) Key for DKP extensions.
Assign the MRE Key function in the DKP Key Template assigned to the DKPs, by customizing a template
and assigning it to the DKPs.
OR
Assign the MRE Key function individually in each DKP, by selecting 'Personalized' Key map for each DKP.
207. This is recommended because changing the values of the default Template will be applied on all other extension types to which the
Template is assigned.
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Enable Silence Detection on Magneto: By default this flag is enabled. If not, select the check box to
enable this flag.
Magneto Silence Disconnection Timer: Set the timer to the desired value. The range of this timer is
from 001 to 255 seconds. By default it is set to 30 seconds.
Magneto VAD Threshold Level: Set the level to the desired value. The range of this level of the 0 to
-96 dBm. By default it is -25 dBm.
1277
1278
Exit SE mode.
SLT8-Magneto8 Card
1279
CO8-Magneto8 Card
Port Swapping can be done from the System Administrator (SA) mode using Jeeves or dialing SA commands from
a telephone.
Before you swap an SLT/CO port and a Magneto port, disconnect the SLT/CO and the Magneto Telephone
connected to these two ports, and connect them according to the pin position in the Swapped Mode. Refer
to the above illustrations of the pin details of the SLT/CO-Magneto Cards in the 'Normal' and 'Swapped'
mode.
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To swap ports on the SLT-Magneto Card, click the SLT8-MAG8 link to open the page.
To swap ports of the CO-Magneto Card, click the CO8-MAG8 link to open the page.
The card connector details will appear according to the Slot Number in which the card is installed. Click the
link of the Slot Number of the card for which you want to use port swapping.
The connector details of the selected card will appear on the page in the normal mode. The page displays
each connector on the card, with the wire-pair colors, the connection status, and the port name you have
programmed for that port.
Go to the connector number for which you want to use port swapping.
Select the Swap check box to enable port swapping for the connector.
When you finish enabling port swapping for the desired connectors on the card, click Submit at the bottom
of the page.
The page will be refreshed and the connector details of the 'Swapped' mode will appear on the page.
If you want to restore Normal mode, clear the Swap check box.
Repeat the same steps if you want to use port swapping for another SLT/CO-Magneto Card, installed in
the ETERNITY.
Exit SA mode.
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LD Port No.: This non-editable field is the number of the software port assigned to the LD port.
H/w Slot-Port: 'Slot' is the number of the Universal Slot in which the LD port Card has been inserted.
'Port' is the number of the LD port on that card.
By default, the ETERNITY will automatically detect and assign the hardware slot and port numbers to
the LD (software) ports. However, if required, you may change the Hardware Slot and Port assigned to
the LD software port.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
208. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.
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Enable Port: This flag is for enabling or disabling a LD trunk port. When an LD trunk port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clearing the check
box.
Name: You may assign a 'Name' to each LD trunk port to facilitate identification. Whenever there is an
incoming call on this port, the Name you have programmed will be displayed on the landing extension.
The Name may comprise a maximum of 18 characters.
Station Basic Feature Template: Assign a Station Basic Feature Template to the LD Trunk port, as
the port functions as an extension for incoming calls.
By default, Station Basic Feature Template 01 is assigned to all extensions of the system (SLT ports,
DKP ports, ISDN Terminals, and E&M Lines with Station as Orientation Type).
Check if the default template fulfills the feature requirements (like Class of Service (COS), Toll
Control, OG Trunk Bundle Group, etc.) of the LD Port when functioning as an SLT.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all LD trunk ports functioning as SLTs, retain Template 01.
If not, customize a Station Basic Feature Templates and assign it to the LD trunk port. To do this,
Click the Station Basic Feature Template link to open the page.
Customize Template number 11 and click Submit at the bottom of the page.
Enter the number of the Template you customized, Template 11 in the 'Station Basic Feature
Template' field of the LD Port, for example, LD-01, on which you want to apply this template.
Repeat the same steps to customize and assign a different Template to another LD trunk port.
Also, refer the topic Station Basic Feature Template to know more about customizing the templates
and applying on the ports.
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SLT Hardware Template: Assign an SLT Hardware Template to the LD Trunk port. By assigning an
SLT Hardware Template the LD port will be configured with hardware features of SLTs like AC
Impedance, Answer Signaling type, Speech Transmit and Receive Gains, Open Loop Disconnect,
Loop Current, and Fax connectivity.
There are 50 SLT Hardware Templates that can be customized and assigned to the LD ports. By
default SLT Hardware Template Number 01 is assigned to all the SLTs of the system.
Check if the values in this template fulfill your requirements. If the default SLT Hardware Template 06
fulfills the feature requirements and if the same features are to be allowed to all LD trunk ports, retain
Template 06.
If different sets of hardware features are to be allowed to different LD ports, then customize separate
SLT Hardware Templates and apply them on the LD ports. To customize SLT hardware templates,
Customize Template number 03 and click 'Submit' at the bottom of the page.
Enter the number of the Template you customized, Template 03 in the SLT Hardware Template
field of the LD Ports on which you want to apply this template.
Repeat the same steps to customize and assign a different SLT Hardware Template to another LD
port.
Also, refer the topic SLT Hardware Template to know more about customizing the templates and
applying on the LD trunk ports.
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Apply the CO Hardware Template you customized to the CO Port by entering the template number
in the CO Hardware Template' field of this port.
Repeat the same steps to customize another template and apply it to the LD trunk port.
To know more about the hardware port features and customizing templates, refer the topic CO
Hardware Template.
Cost Factor: This parameter is of relevance only if Least Cost Routing feature is applied on the LD
trunk port for outgoing calls.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a
preference number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same
preference must be assigned the same Cost Factor. Different trunk types can also be assigned the
same Cost Factor. These trunks are used for routing calls.
Assign a Cost Factor to the LD trunk port, for example, 02 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the LD trunk port 01
only,
You must first assign a Cost Factor (01-99) to LD trunk port 01, for example, 02.
Click the Least Cost Routing - Number Based link to open the page.
Enter '6' in the 'Number' column, Cost Factor '02' as Preference 1, 2, 3 and 4.
SMDR-OG Storage: This flag is used to enable or disable the storage of details of outgoing calls from
the LD trunk port. Please refer the topic Station Message Detail Recording-Storage for more details.
By default, storage of outgoing calls is enabled.
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SMDR IC Storage: This flag is used to enable or disable storage of details of incoming calls on the LD
trunk port. Please refer the topic Station Message Detail Recording-Storage to know more. By
default, storage of incoming calls is enabled.
Hold on DSS Key Press: This flag defines the 'Hold' state of the external called party, when an
extension user presses a DSS key to dial another port.
For example, the DKP extension user (on DKP-001 port) is in the middle of speech with an external
party on LD trunk port-02.
If extension user of DKP-001 presses a DSS key to call another extension port DKP-003, two situations
are possible, depending on whether the Hold on DSS Key Press flag is enabled or disabled:
When the Hold Flag is enabled: LD-02 will be played music-on-hold. DKP-001 will hear Ring Back
Tone and the call will be placed on DKP-003.
When Hold Flag is disabled: LD-02 will be disconnected. DKP-001 will hear Ring Back Tone, and
call will be placed on DKP-003.
Call Cost Calculation Pulse Rate Option: This parameter is to be configured only if you want to apply
the Call Cost Calculation (CCC) feature on the LD trunk ports.
You can program four options for Pulse Rate Types. Select from Pulse Rate Type for Pulse Rate
Option 1 to 4 which you want to apply on the LD trunk ports.
Call Cost Calculation Time Schedule: This parameter is to be configured only if you want to apply the
Call Cost Calculation (CCC) feature on the LD trunk ports.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases,
you must first define the Time Zone (time of the day) for which a particular Pulse Rate should be
applied and the Time Schedule for each Time Zone.
You can configure up to four different Time Zones - T1, T2, T3 and T4 with different Pulse rates in the
Configuring Pulse Rate Types.
Now, configure the Call Cost Calculation Time Schedule, by specifying the Start Time and the End time
(in 24hours: minutes format) for each Time Zone.
The default Time Schedule (starts and end time) for each Time Zone Index are as follows:
Time Zone Index
Start Time
End Time
T1
00:00
23:59
T2
00:00
23:59
T3
00:00
23:59
T4
00:00
23:59
If your service provider offers the same Pulse Rate for the entire day,
1286
program only one Time Zone Index with the Pulse Rate, for example, T1, in the CCC-Normal Pulse
Rate Table.
Now program the Time Schedule for Time Zone, T1, with the start and end time in Hours: Minutes
format;
set the start and end time of the other Time Zone Index, T2 to T4, to 00:00 (hours: minutes).
Similarly, if your service provider supports two different Pulse Rates in a day, program the Start and the
End time for two Time Zones and set the other two to 00:00.
If you have programmed all the Parameters, click Submit at the bottom of the page to save your
settings.
When you have finished configuring the desired number of LD ports, you may log out of Jeeves or
continue with other configuration tasks.
1287
To assign Station Basic Features Template (SBFT) for LD trunk port, dial:
5512-1-LD-SBFT to assign a template to a single port.
5512-2-LD-LD Trunk-SBFT to assign the same template to a range of ports.
5512-*-SBFT to assign the same template to all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
SBFT is from 01 to 50.
By default, SBFT 01 is assigned.
To assign the SLT Hardware Template for LD trunk port, dial:
5704-1-LD-SLT Hardware Template to assign a template to a single port.
5704-2-LD-LD-SLT Hardware Template to assign the same template to a range of ports.
5704-*-SLT Hardware Template to assign the same template to all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
SLT Hardware Template is from 01 to 50.
By default, the SLT Hardware Template 01 is assigned.
To program the CO Hardware Template for LD trunk port, dial:
5904-1-LD-CO Hardware Template to assign a template to a single port.
5904-2-LD-LD-CO Hardware Template to assign the same template to a range of ports.
5904-*-CO Hardware Template to assign the same template to all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
CO Hardware Template is from 01 to 50.
By default, CO Hardware Template 01 is assigned.
To program the Cost Factor for LD trunk port, dial:
3942-1-LD-Cost Factor to assign cost factor to a single port.
3942-2-LD-LD-Cost Factor to assign the same cost factor to a range of ports.
3942-*-Cost Factor to assign the same cost factor to all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
Cost Factor is from 01 to 99.
By default, Cost Factor 01 is assigned.
To enable/disable the SMDR OG Storage flag for LD trunk port, dial:
3943-1-LD-Flag to enable/disable flag for a single port.
3943-2-LD-LD-Flag to enable/disable flag for a range of ports.
3943-*-Flag to enable/disable flag for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
Flag is
0 for Disable
1for Enable.
By default, flag is enabled.
1288
1289
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The Start Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, Start Time is set to 00:00.
To program the Call Cost Calculation Time Schedule-T2-End Time for LD trunk port, dial:
3950-1-LD-End Time to program the end time for a single port.
3950-2-LD-LD-End Time to program the same end time for a range of ports.
3950-*-End Time to program the same end time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The End Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, End Time is set to 23:59.
To program the Call Cost Calculation Time Schedule-T3-Start Time for LD trunk port, dial:
3951-1-LD Trunk-Start Time to program the start time for a single port.
3951-2-LD Trunk-LD Trunk-Start Time to program the same start time for a range of ports.
3951-*-Start Time to program the same start time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The Start Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, Start Time is set to 00:00.
To program the Call Cost Calculation Time Schedule-T3-End Time for LD trunk port, dial:
3952-1-LD-End Time to program end time for a single port.
3952-2-LD-LD-End Time to program the same end time for a range of ports.
3952-*-End Time to program the same end time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The End Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, End Time is set to 23:59.
To program the Call Cost Calculation Time Schedule-T4-Start Time for LD trunk port, dial:
3953-1-LD-Start Time to program start time for a single port.
3953-2-LD-LD-Start Time to program the same start time for a range of ports.
3953-*-Start Time to program the same start time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The Start Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, Start Time is set to 00:00.
To program the Call Cost Calculation Time Schedule-T4-End Time for LD trunk port, dial:
3954-1-LD-End Time to program end time for a single port.
3954-2-LD-LD-End Time to program the same end time for a range of ports.
3954-*-End Time to program the same end time for all ports.
Where,
LD is the number of software port of the LD trunk port, from 01 to 32.
The End Time is in 24 hours, HH:MM; range of HH is 00 to 23, range of MM is from 00 to 59.
By default, End Time is set to 23:59.
1290
Exit SE mode.
Configuring LCR
Least Cost Routing (also referred to as Automatic Route Selection) is an expense control feature of ETERNITY.
Least Cost Routing (LCR) is useful when there are different trunk lines for making outgoing calls, and the service
providers of these trunks offer different tariffs for calls made to certain locations or numbers or during a particular
time of the day.
When a call is made from an extension of the ETERNITY, LCR recognizes where the call is going. Depending upon
how the LCR is programmed, the system routes the call through the assigned trunks.
The system can be programmed to select the most cost effective trunk for the time of the day when the call is made
from the extension or to select the most cost effective trunk for the destination number dialed from the extension or
to select the most cost effective trunk considering both time of the day and destination number.
Accordingly, ETERNITY supports four types of LCR which can be programmed, namely:
1. Time-based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers a different tariff according to the time of the day.
For example, Service Provider 1 offers a lower tariff for calls made between 9am to 8pm, while Service
Provider 2 offers a lower tariff for calls made between 8pm to 9am.
When Time-based LCR is programmed, the system uses the Online-dialing logic, whereby digits dialed by
the user are directly passed on to the trunk.
2. Number based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers different tariffs according to the area or distance, or phone numbers dialed. For
instance, Service Provider 1 provides cheaper calling rates for calls made from City A to City B, than
Service Provider 2 and Service Provider 3.
3. Time and Number based LCR: This type of LCR is a combination of number and time based LCR, that is,
the service providers offer different tariffs according to the time of the day as well as area/distance.
For example, Service Provider 1 offers lower rates for calls made from City A to City B during peak hours
9am to 8pm, as compared to Service Provider 2, whereas Service Provider 2 offers cheaper rates for calls
made from City A to City B during off peak hours (8pm to 9 am).
When Time+Number-based LCR is programmed, the system uses Store and Forward dialing logic,
whereby digits dialed by the user are first stored at a memory location in the system, and then dialed out
on the assigned trunk.
4. Service Provider-based LCR: This type of LCR may be used when the same Service Providers offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network. For example, Service Provider 1 offers lower rates to call a Service
Provider1 number in City A and in City B, than for calling numbers of Service Provider 2 in the same cities.
This type of LCR may also be used when the same Service Providers apply different charges for different
subscriber services provided by them. For example, Service Provider 1 offers both Fixed Line as well as
GSM services and applies different charges for fixed line and GSM services.
1291
When Service Provider-based LCR is programmed, whenever a number is dialed out, the system ignores
the area code, checks the number in the 'Service Provider-based LCR table', and routes the call according
to the trunk programmed for that number.
ETERNITY also supports LCR based on Carrier Pre-Selection. This type of LCR is useful where there exist
different service providers for local and long distance calls. Refer the topic Least Cost Routing-Carrier
Pre-Selection to know more.
Cost Factor
For LCR to work, all trunks that are allotted to extensions for making outgoing calls, must first be assigned a Cost
Factor.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a preference number
for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same preference must be assigned the same
Cost Factor. Different trunk types can also be assigned the same Cost Factor. These trunks are used for routing
calls.
By default all trunks are assigned Cost Factor number 01.
After assigning Cost Factor to Trunks, you must configure the Type of LCR to be used on Trunks in the Outgoing
Trunk Bundle Group (OGTBG) allotted to the extensions for making calls.
1292
On a sheet of paper, make a list of all the trunks assigned to extensions for making outgoing calls.
Make a table of the trunk types and assign a cost factor to each trunk type, as shown below.
Trunk Type and Number
Service Provider
Cost Factor
CO-001
BSNL
01
CO-002
BSNL
02
MOB-001
Reliance
03
MOB-002
BSNL
04
MOB-003
Airtel
05
MOB-004
Vodafone
06
BRI-01
BSNL
07
BRI-02
Reliance
08
Program the Cost Factor number you assigned to the Trunk types in their respective trunk parameters. For
instance, assign Cost Factor 01 to CO-001 and Cost Factor 002 to CO-002 in the Configuring CO
Trunks. Similarly, assign Cost Factor 03 to Mobile Trunk 001, Cost factor 04 to Mobile Trunk 002, Cost
Factor 05 to Mobile Trunk port 003, and Cost Factor 06 to Mobile Trunk port 004 in the Mobile Port
Parameters.
For programming instructions, refer the topics Configuring CO Trunks and Configuring Mobile Trunks.
You can configure Time-based LCR for as many as 8 different Time Zones.
Define the Time Zone, that is, the start and end time, when the LCR should be applied for the outgoing
calls. The Time Zone you define is stored at an Index number from 1 to 8.
For each Time Zone that you define, select the Trunk as your first preference, that is, Preference 1. Select
the trunk of your second, third and fourth preference. When the trunk you selected as first preference is
busy, the system will route the call through the next trunk you have set that is free.
Refer to the table you prepared for assigning Cost Factor to trunks.
For example, you want calls made during 9am to 8pm to be routed through BSNL CO trunks (CO-001 and
CO-002). If these trunks are busy, you want the system to route calls through the BRI line of BSNL trunk.
When this line is busy, you want the system to attempt to route calls through the BRI line of Reliance.
You want calls made between 8pm to 9am to be routed through BSNL CO trunk 001 only.
At Time Zone Index 1, define the Time Zone start and end time in 24 Hours:Minutes format, enter the Cost
Factor you assigned to CO-001 (01) and CO-002 (02) as Preference 1 and Preference 2 respectively.
Enter the cost factor you assigned to BRI-01 (07) and BRI02 (08) as Preference 3 and Preference 4
respectively.
Time Zone
Index
Time Zone
Cost Factor
Start Time
(HH:MM)
End Time
(HH:MM)
Preference
1
Preference
2
Preference
3
Preference
4
09:00
20:00
01
02
07
08
20:01
08:59
01
01
01
01
3
4
5
6
7
8
Similarly, at Time Zone Index 2, define the Time Zone in 24 Hours: Minutes format. Enter the Cost factor
you assigned to CO-001, that is, 01 as Preference 1, 2, 3, and 4. When calls are made during this time
period, they will be routed through CO-001 only.
If you have finished defining Time Zones and the preferred trunks for the time zones, configure the Timebased LCR using Jeeves or a Telephone.
1293
Enter the values of the Time-based LCR you prepared on the sheet of paper in the appropriate fields.
Enter SE mode.
To program Time Zone at a Time Zone index, dial:
3402-Time Zone Index-Start Time-End Time
Where,
Time Zone Index is from 01 to 08.
Start Time is the time in HH:MM format when the Time zone starts.
End Time is the time in HH:MM format when the Time zone ends.
For example to program 09:00 to 20:00 hours at Time Zone Index 1, dial 3402-01-0900-2000
By default, Time Zone is 00.00 to 23.59.
To program the Cost Factor (Service Provider preference) for the Time Zone, dial:
3403-Time Zone Index-CF1-CF2-CF3-CF4
Where,
Time Zone Index is from 01 to 08.
CF1is the first preferred (the cheapest) service provider.
1294
Exit SE mode.
You can configure Number-based LCR for as many as 99 different Numbers, which are stored against
Index numbers from 01 to 99.
Enter each of the number strings at an Index number from 01 to 99. A Number string may be a complete
telephone number, a truncated phone number or an area code.
For each number string you enter, select a Trunk as your first preference, that is, Preference 1. Select the
trunk of your second, third and fourth preference. When the trunk you selected as first preference is busy,
the system will route the call through the next trunk you have set that is free.
Refer to the table you prepared for assigning Cost Factor to trunks.
For example, you want all mobile numbers to be routed through the Mobile Trunk ports, all local numbers
to be routed through the CO ports.
All mobile numbers start with the number '9', which is prefixed with a '0' when making long distance mobile
calls, so enter '9' and '09' as the number strings. For '9' as well as '09', select the Mobile trunks through
which the calls should be made in order of preference.
Similarly, all local numbers start with 2, so enter this number in the number string column, and select the
CO trunk in the order of preference. As in this example, you have only two CO trunks, so you may keep the
same two trunks as your preference.
Index
Number
Cost Factor
Preference 1
Preference 2
Preference 3
Preference 4
04
03
05
06
09
04
06
05
03
01
02
01
02
4
5
1295
Index
Number
Cost Factor
Preference 1
Preference 2
Preference 3
Preference 4
:
:
99
If you have finished entering the number strings, and selecting the preferred trunks for the numbers,
configure the Number-based LCR using Jeeves or a Telephone.
1296
Enter the values of the Number-based LCR you prepared on the sheet of paper in the appropriate fields.
Enter SE mode.
To program a number at number Index in the Time-based LCR table, dial:
3411-Number Index-Number String-#*
Where,
Number Index is from 02 to 99.
Number String can be a complete telephone number, a truncated telephone number or an area code.
Number string may be a maximum of 16 digits. Terminate the command with #* if the number string
has fewer than 16 digits. Terminate the command with #* if the number string has fewer than 16 digits.
By default, Number String is 'Blank'.
For example to program '9' at Index 02, dial 3411-02-9-#*
To clear a number string programmed at a number index, dial:
3411-Number Index-#*
Where,
Number Index is from 02 to 99.
To program Cost Factor (Service Provider preference) for the each Number, dial:
3412-Number Index-CF1-CF2-CF3-CF4
Where,
Number Index is from 01 to 99.
CF1is the first preferred (the cheapest) service provider.
CF2 is the second preferred (second cheapest) service provider.
CF3 is the third preferred service provider.
CF4 is the fourth preferred service provider.
For example, to program Cost Factor for number '9' at Index 01, dial 3412-01-04-03-05-06
It is mandatory to complete this command with CF1 to CF4. If you have only one service provider, program
the same as CF1, CF2, CF3, CF4.
To default the Number-based LCR table, dial:
3410
This is a combination of the Time Zone-based and Number-based LCR. You may use this feature if your
service providers offer lower call rates for calls made to certain numbers during a certain time of the day.
Define the Time Zones when the service providers offer lower tariff. You can define up to 8 time zones.
For each Time Zone you define, specify the Number strings on which lower tariff is applied during that
Time Zone.
For each Number string you enter for a particular time zone, assign Cost Factor. Select a trunk as your first
preference. Select trunks of your second, third and fourth preference. Refer to the table you prepared for
assigning Cost Factor.
1297
You can enter up to 99 different number strings, which are stored at Index numbers from 01 to 99. The
Number strings may be complete telephone numbers, truncated phone numbers or area codes.
When the trunk you selected as first preference is busy, the system will route the call through the next
trunk you set as preference if it is free.
For example, service provider of CO-001 and CO-002 (assigned Cost Factor 01 and 02) offers the lowest
rate for calls made to Area Code 022 between 8am to 12pm, followed by service providers of Mobile
Trunk-02 (assigned cost factor 04) and Mobile Trunk-01 (assigned cost factor 03).
Assign Cost Factor preference for the number string in this sequence: 01, 02, 04, 03
Time Zone1
Start Time
End Time
Time Zone2
Number
HH
MM
HH
MM
HH
MM
08
00
12
00
09
00
12
00
18
00
20
00
Cost Factor
Index
Time Zone3
Cost Factor
Cost Factor
Prefe
rence
1
Prefe
rence
2
Prefe
rence
3
Prefere
nce 4
Prefe
rence
1
Prefe
rence
2
Prefe
rence
3
Prefere
nce 4
Prefe
rence
1
Prefe
rence
2
Prefe
rence
3
Prefere
nce 4
04
03
02
01
03
05
06
04
03
05
06
04
022
01
02
04
03
011
01
02
04
05
080
01
02
05
06
Time
Zone
8
:
99
If you have finished defining the time zones, entering the number strings, and selecting the preferred
trunks for the number strings, configure the Number and Time-based LCR using Jeeves or a Telephone.
1298
Enter the values of the Time+Number-based LCR you prepared on the sheet of paper in the appropriate
fields.
Enter SE mode.
To define Time Zone for Time+Number-based LCR, dial:
3421-Time Zone Index-Start Time-End Time
Where,
Time Zone Index is from 01 to 08.
Start Time is the time in HH:MM format when the Time zone starts.
End Time is the time in HH:MM format when the Time zone ends.
By default, Time Zone is 00.00 to 23.59.
For example, to define 08:00 to 12:00 as start and end time of Time Zone 1, dial 3421-1-0800-1200
To program the number string for Time+Number-based LCR, dial:
3422-Number Index-Number String-#*
Where,
Number Index is from 02 to 99.
Number String can be a complete telephone number, a truncated telephone number or an area code.
Number string is of maximum 16 digits. Terminate the command with #* if the number string has fewer
than 16 digits.
By default, Number String is 'Blank'.
For example, to program Number string '022' at Number Index 02, dial 3422-02-022-#*
1299
In Service Provider-based LCR, whenever a number is dialed out, the system ignores the area code, and
starts checking the numbers in the 'Service Provider-based LCR table' and routes the call according to the
trunk programmed for that number. For this, you must program the two parameters Area Code and Ignore
Digit Count in the Area Code Table.
Number
Area
Code
Ignore
Digit Count
Preference 1
Preference 2
Preference 3
Preference 4
01
080
08
07
01
02
02
022
07
01
02
08
99
01
02
01
02
:
03852
As you can see, the Service Provider-based LCR Table is similar to the Number-based LCR table.
You can program as many as 99 different numbers which are stored against Index numbers from 01 to 99.
The number strings may be the complete telephone number, a truncated phone number or the first digit of
the phone number.
For each number string that you enter against an Index number, you must also specify the Area Code and
the Ignore Digit Count.
1300
Cost Factor
Index
No.
The Ignore Digit Count is the number of digits in the area code that the system should ignore before
checking the Service Provider-based LCR table. For each area code that you enter, the corresponding
Ignore Digit Count will be the number of digits in the area code. For example, the area code for the number
starting with '3' is 080, which consists of 3 digits. So, the Ignore Digit Count for the number/area code 080
will be 3.
For each number string and area code that you enter, assign the Trunk of the service provider that you
prefer as your first, second, third and forth preference for dialing that number/area code. Refer the table
you prepared for assigning Cost Factor to trunks.
If you have finished entering the number strings, their corresponding area codes and the Ignore Digit
Count, and the preferred trunks, configure Service Provider-based LCR using Jeeves or a Telephone.
To program Area Code and Ignore Digit Count, Under Configuration, click Call Cost Calculation.
Enter the Area Codes and the corresponding Ignore Digit Counts from the sheet you prepared for Service
Provider based-LCR. You may also enter the respective name for each area code, if desired.
Now, click the Least Cost Routing (LCR). The links for the LCR options appear.
1301
Enter the values of the Service Provider-based LCR you prepared on the sheet of paper in the appropriate
fields.
Enter SE mode.
To program Area Code in the Area Code Table, dial:
2620-Area Code Index-Area Code-#*
Where,
Area Code Index is from 001 to 999.
Area Code is a string of maximum 7 digits. Terminate the command with #* if the number string has
fewer than 7 digits.
By default, Area Code is 'Blank'.
For example, to program Number string '080' at Number Index 001, dial 2620-001-080-#*
To clear an Area Code in the Area Code Table, dial:
2620-Area Code Index-#*
To program Ignore Digit Count for an Area Code, dial:
2623-Area Code Index-Ignore Digit Count
1302
Where,
Area Code Index is from 001 to 999.
Ignore Digit Count is from 0 to 9.
By default, Ignore Digit Count is '0'.
Refer the topic "Area Code Table" under Call Cost Calculation to know more.
To program a number at number Index in the Service Provider-based LCR table, dial:
3441-Number Index-Number String-#*
Where,
Number Index is from 02 to 99.
Number String can be a complete telephone number, a truncated telephone number or an area code.
Number string may be a maximum of 16 digits. Terminate the command with #* if the number string
has fewer than 16 digits.
By default, Number String is 'Blank'.
For example to program '3' at Index 02, dial 3441-02-3-#*
To clear a number string programmed at a number index, dial:
3441-Number Index-#*
Where,
Number Index is from 02 to 99.
To program Cost Factor (Service Provider preference) for the each Number, dial:
3442-Number Index-CF1-CF2-CF3-CF4
Where,
Number Index is from 01 to 99.
CF1is the first preferred (the cheapest) service provider.
CF2 is the second preferred (second cheapest) service provider.
CF3 is the third preferred service provider.
CF4 is the fourth preferred service provider.
For example, to program Cost Factor for number '3' at Index 01, dial 3412-01-08-07-01-02
It is mandatory to complete this command with CF1 to CF4. If you have only one service provider, program
the same as CF1, CF2, CF3, CF4.
To default the Number-based LCR table, dial:
3440
Exit SE mode.
1303
For each OGTBG number assigned to extensions, select the desired LCR Type: Time-based, Numberbased, Time+Number based, Service Provider-based (Cost Factor).
You can find the OGTBG number assigned to each extension from the Station Basic Feature Template
assigned to the extension.
Enter SE mode.
To select LCR type in OG Trunk Bundle Group, dial:
1404-1-OGTBG-LCR Type to select an LCR Type for a single group.
1404-2-OGTBG-OGTBG-LCR Type to select the same LCR Type for a range of OGTB groups.
1404-*-LCR Type to select the same LCR Type for all OGTB groups.
Where,
OGTBG is from 01 to 32.
LCR Type is
0 for No LCR
1 for Time zone-based LCR
2 for Number-based LCR
3 for Time+Number-based LCR
4 for Service Provider-based (Cost Factor) LCR
By default, 'No LCR' is selected as LCR Type.
1304
Exit SE mode.
Emergency
Number 1
Emergency
Number 2
Emergency
Number 3
Emergency
Number 4
Emergency
Number 5
Emergency
Number 6
Emergency
Number 7
Emergency
Number 8
Emergency
Number 9
Emergency
Number 10
Australia
000
106
112
OGTB
Group
Number
32
30
31
Banglade
sh
999
1
OGTB
Group
Number
Belgium
101
100
112
OGTB
Group
Number
Bhutan
110
112
113
OGTB
Group
Number
Canada
911
OGTB
Group
Number
32
China
110
120
119
OGTB
Group
Number
Germany
110
112
1305
Emergency
Number 1
Emergency
Number 2
Emergency
Number 3
Emergency
Number 4
Emergency
Number 5
Emergency
Number 6
Emergency
Number 7
Emergency
Number 8
Emergency
Number 9
Emergency
Number 10
100
101
108
112
110
118
119
113
112
113
118
115
112
OGTB
Group
Number
Jordan
191
199
OGTB
Group
Number
Kazakhsta
n
03
OGTB
Group
Number
Feature
OGTB
Group
Number
India
OGTB
Group
Number
Indonesia
OGTB
Group
Number
Italy
Kenya
999
OGTB
Group
Number
Kuwait
777
OGTB
Group
Number
Malaysia
999
112
OGTB
Group
Number
Maldives
102
108
OGTB
Group
Number
Mauritius
999
115
144
1306
Emergency
Number 1
Emergency
Number 2
Emergency
Number 3
Emergency
Number 4
Emergency
Number 5
Emergency
Number 6
Emergency
Number 7
Emergency
Number 8
Emergency
Number 9
Emergency
Number 10
OGTB
Group
Number
Mexico
911
OGTB
Group
Number
Namibia
911
OGTB
Group
Number
Feature
Nepal
100
OGTB
Group
Number
New
Zealand
111
112
911
08
OGTB
Group
Number
Oman
9999
OGTB
Group
Number
Pakistan
15
115
16
911
112
OGTB
Group
Number
117
911
112
OGTB
Group
Number
Poland
997
999
998
112
OGTB
Group
Number
Russia
02
03
01
112
OGTB
Group
Number
999
995
112
911
Philippine
s
Singapore
1307
Emergency
Number 1
Emergency
Number 2
Emergency
Number 3
Emergency
Number 4
Emergency
Number 5
Emergency
Number 6
Emergency
Number 7
Emergency
Number 8
Emergency
Number 9
Emergency
Number 10
10111
10117
112
091
061
080
085
112
119
110
111
110
112
113
OGTB
Group
Number
Sweden
112
OGTB
Group
Number
Taiwan
110
119
OGTB
Group
Number
Thailand
191
1669
199
OGTB
Group
Number
Turkey
155
112
110
OGTB
Group
Number
999
998
997
112
999
112
Feature
OGTB
Group
Number
South
Africa
OGTB
Group
Number
Spain
OGTB
Group
Number
Sri Lanka
OGTB
Group
Number
Sudan
UAE
OGTB
Group
Number
UK
1308
Emergency
Number 1
Emergency
Number 2
Emergency
Number 3
Emergency
Number 4
Emergency
Number 5
Emergency
Number 6
Emergency
Number 7
Emergency
Number 8
Emergency
Number 9
Emergency
Number 10
USA
911
112
OGTB
Group
Number
32
Feature
OGTB
Group
Number
For example, if you selected USA as Region, the default Emergency Number Table would look like this:
Default Emergency Number Table for USA
Index
Emergency Number
01
911
32
02
112
01
03
01
04
01
05
01
06
01
07
01
08
01
09
01
10
01
If you selected Australia as Region, the default Emergency Number Table would look like this:
Default Emergency Number Table for Australia
Index
Emergency Numbers
01
000
32
02
106
30
03
112
31
04
01
05
01
06
01
07
01
08
01
09
01
10
01
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Each Number is stored at an index from 01 to 10. The Emergency Number fields from index 01 to 5 are noneditable, but you can select a different OGTBG for each of these default Emergency Numbers.
You can add Emergency Numbers at index 06 to 10 in the table, and select the OGTBG as required.
Click the Default button, the system will upload the default Emergency Numbers as per the Region you
have selected.
The first five entries, at Index 01 to 05 on this table, are uneditable. These fields will be populated with the
default Emergency Numbers of your country (which you selected as Region).
If the Emergency Numbers loaded by default are applicable for your region/country, all you need to do is
re-assign, if required, the Outgoing Trunk Bundle Group assigned by default to each number.
If the Emergency Numbers loaded by default are not applicable for your region/country, you may add the
Emergency Numbers and their OGTBG at index 06 to 10 in this table.
Make sure that the trunks configured in the OGTBG for each Emergency Number belong to the correct
network and the ports through which the calls are to be routed are not disabled. For example, '112' is the
default Emergency Number for the mobile network. So, make sure that the Mobile Trunk to be used for
dialing this number is included in the OGTBG you assign to this number in the Table.
1310
Exit SE mode.
1311
1312
CHAPTER 10
Make sure you have installed the VMS Card correctly and observed the Reset Cycle. Before you begin
configuration of the VMS related parameters, consider the following points:
Decide which extension users are to be provided voice mail. Make a list of these extensions by their port
type and (software) port number, access codes.
To upload or download Software, configuration files, mailbox messages, you will need to access the
embedded FTP server of the VMS Card. Connect a PC to the VMS Ethernet Port to a PC. If you connect a
LAN PC, you must change the IP Address of the VMS Ethernet Port and configure VMS Ethernet Port
Parameters.
1313
1314
Select the Connection Type according to the type of IP Addressing scheme used on the LAN to which
the VMS Ethernet Port is connected. The IP Addressing used on the LAN may be:
Static
DHCP
PPPoE
Static IP Parameters: If you selected Static as Connection type, ask your LAN Administrator and
assign the following to the VMS Ethernet Port:
IP Address
Subnet Mask
Gateway IP Address
PPPoE Parameters: If you selected PPPoE as Connection type, ask your LAN Administrator and
enter the details for the following in their respective fields:
User ID
Password
Service Name (if provided)
DNS Setting:
DNS Server - Static - Automatic
DNS Address
DNS Domain Name
MAC Address:
Unique: The MAC Address of the VMS Ethernet Port is displayed in this box.
Clone: Select this radio button if you want the MAC Address to be cloned. Enter the MAC Address
to be cloned in this field. Default: Blank.
Click Submit to save your settings. The VMS Card will restart.
Continue configuration.
209. This would not be required if there is a Dynamic Host Configuration Protocol (DHCP) server on the LAN.
1315
1316
Default:Static.
To program DNS IP Address, dial
5069- Address-Address-Address-Address-#*
Default: Blank
To program DNS Domain Name, dial:
5070-DNS Domain Name-#*
Default: Blank
To enable/disable MAC Address Cloning, dial:
5071-MAC Address Type-#*
Where,
MAC Address Type is
0 for Disable Cloning
1 for Enable Cloning
Default: 0
To program Clone MAC Address, dial:
5072-Clone MAC Address-#*
Exit SE mode.
1317
The status of the following port parameters will appear on your screen:
Description
DHCP option is set and application broadcasts the DHCP request to get
WAN parameter and is waiting for response from the DHCP Server.
PPPoE option is set and application broadcasts PPPoE request to get WAN
parameter and is waiting for response from the PPPoE Server.
When DHCP option is set and the DHCP Client in the system sends an IP
renew request to the DHCP Server. In returm from the DHCP Server the
client receives a new IP Address with renewal.
Success
Option may be set as DHCP, PPPoE and Static and All Network Parameters
are Set and Read Successfully.
Failed
1318
Check the default values of the following parameters, and change them, if required, to the desired values:
1319
Extension Number Validation: The VMS Auto Attendant allows callers to reach directly the desired party
in an organization, by giving them the option of dialing the extension number of the desired party.
When the Extension Number Validation flag is enabled, the VMS compares the extension number dialed
by the external caller with the extension numbers configured in the system. If no match is found, the VMS
responds with a message Invalid Number (Invalno.wav).
By default, Extension Number Validation is enabled when ETERNITY is operating the Enterprise mode.
This flag is disabled when ETERNITY is operating in the Hospitality mode.
Send Message as per Distribution List: For Sending Messages to a group of persons at the same
time, the VMS allows you to create Distribution Lists210 to which voice messages along with email
notification can be sent.
Using Distribution Lists, an extension user can send voice message along with email notification to as
many as 500 extensions of ETERNITY. Distribution Lists take up a sizeable part of the system resources,
and may slow down the process of sending voice messages to Distribution Lists.
The VMS resolves this with the flag Abort All Processes and send Email to Extension Number of
Distribution List.
When you enable this flag, the VMS stops all other processes - drops calls, stops sending and receiving of
emails over the SMTP client - and sends the voice message to the Distribution List. Thus, the process of
sending messages to Distribution Lists happens much faster. Once the message is sent, the VMS
resumes the other processes.
By default, this flag is disabled.
Home Node Code for Graph: By default, 0 is defined as the function code for Home Node in a graph.
Dialing 0 takes the caller to the home position in a graph. If required, you may assign a different function
code for Home Node in this field. Refer the topic Graphs and Nodes to know more.
1320
Code to Stop Message Recording: The VMS requires callers/extension users to signal end of message
recording by dialing a code. By default, # is the programmed as the code to Stop Message Recording. If
required, you may assign a different code.
Memory Usage Notification to SE: The VMS allows notifications to be sent to the System Engineer via
email. Enable this flag, if you want notifications to be sent via email to the System Engineer. Make sure
that you have configured the SMTP Settings, for more information, see SMTP Settings.
The table below displays the events for which notifications will be sent to the SE:
Event
Message
Description
You must also specify the email address to which the notifications are to be sent.
SE Email ID: Enter the email address on which the notifications should be sent to the System Engineer.
The System Engineer must have access to this email ID. The email ID may consist of a maximum of 64
characters.
Mailbox for Call Taping: Enter the Access Code of any SLT, DKP, SIP Extension, ISDN Terminals,
Department Group, General Mailbox or Extensions over QSIG, whose mailbox you want to assign for Call
Taping.
Make Message Notification calls using TAC: Select the Trunk Access Code to be used by the system to
make outgoing notification calls to external numbers.
When Mailbox is not assigned, Route Station Calls to: You can route station calls either to a Mailbox
or to the Home Position, when Mailbox is not assigned to an extension.
Channel Reserved for Voice Mail Auto Attendant: Select the number of channels of the VMS that you
wish to reserve for the Voice Mail Auto Attendant. These channels will be used to answer incoming calls
landing on Voice Mail Auto Attendant enabled Trunks only. These channels even if free will not be
available to extension users to access their Mailbox.
1321
Home Node
Home Node or Home Position is the point from where the journey of the caller/extension user starts in a graph of
the VMS. The VMS greets the caller/extension user and takes them to the Home Position. At the Home position,
Welcome message is played, the VMS offers different options to the caller, like go to Operator, dial the name of the
extension user, go to Home Position, disconnect the call, leave a message.
The VMS instructs the caller/extension user to dial different digits (function codes) for each option offered in the
Home Position. For each of the Home Node options listed below, you may change the default function codes to
match your requirement.
Operator Code: By default, 9 is the function code for Operator. The VMS instructs the caller to dial 9 to
reach the Operator.
Dial by Name Code: By default, 7 is the function code for Dial by Name. The VMS instructs the caller to
dial 7 to dial by name of the extension user.
Root Node Code: By default, 0 is the function code for reaching the Root Node (Home Position). The
VMS instructs the caller to dial 0 to reach the Home Position.
Mailbox Management Code: By default, 8 is the function code for Mailbox management. The VMS
instructs the extension users (mailbox owner) to dial 8 to reach their mailbox settings.
Disconnect Code: By default, # is the function code for disconnecting the call. The VMS instructs the
caller to dial # to disconnect the call.
1322
Leave Message Code: By default, 6 is the function code for leaving messages. The VMS instructs the
callers to dial 6 to leave a message for the extension user.
The VMS offers extension users the facility to customize the greetings messages played to callers for certain
conditionsbusy, no reply or unconditional/unregistered Call Forward. Extension users can record a different
message for each condition. These greetings are known as Conditional Greetings. For detailed instructions to
record the greetings, see Recording Conditional Greetings.
After these greetings are played, the VMS prompts the caller to dial the digits to direct the caller. The digits dialed
can be from 1 to 9, 0 and #. For each digit dialed by the caller you can select the action will be taken by the system.
The VMS offers the following options that can be set by you for the digits dialed by the caller:
Leave Message
Dial another Extension
Dial another Extension using Dial by Name
Transfer Call to Operator
Transfer Call to Mobile/Alternate Number
Transfer Call to Assistance
Go to Home Position
Disconnect Call
If you select the options Transfer Call to Mobile/Alternate Number or Transfer Call to Assistance, you must
make sure that the relevant numbers are configured by the Extensions users. The Extension users can
configure these numbers through their Mailbox Settings. For detailed instructions, see Mailbox Settings.
It is possible that the caller does not dial any digit, then the you can select any one of the following as the Leave
Message Option for Caller on No Digit Dialed option:
Disconnect
Transfer to Operator
Leave message
Go to Home Position
Make sure Allow Mailbox Management is enabled by you for the Extension users, so that they can:
record Conditional Greetings.
change the settings of their mailbox and configure the Mobile/Alternate Number or/and Assistance
Number.
1323
When the VMS transfers a call, it plays different voice messages, informing the caller about the status of the called
party, asking the caller to leave a message. By default, the VMS plays the following messages. You may choose
whether or not the VMS should play the following messages to the caller:
Call could not be Attended Message: This message is played to the caller when the called party does
not answer the transferred call.
Called Party Busy Message: This message is played to the caller when the called party is found to be
busy.
Called Party Not Available Message: This message is played to the caller when the called party is not
available.
Leave Message: This message is played to the caller to leave a message for the called party. If you
disable this option, the VMS will not offer the caller the option of leaving a message for the extension user.
Call could not be Attended Message, Called Party Busy Message, Called Party Not Available Message and
Leave Message options will be applicable only if, Conditions Greetings are not recorded by the Extension
users.
1324
Message Verification: This message is played to the caller after they have recorded their message. If you
disable this option, the VMS will not give the caller the option of verifying the recorded message.
Timers
First Digit Wait Timer for External Caller: Configure the First Digit Wait Timer for External Caller. When
the Voice Mail Auto Attendant answers the external call and if the caller does not dial any digit, the system
waits for the expiry of this timer to process the call further.
First Digit Wait Timer for Internal Caller: Configure the First Digit Wait Timer for Internal Caller. When
the Voice Mail Auto Attendant answers the internal call and if the user does not dial any digit, the system
waits for the expiry of this timer to process the call further.
This is the time for which the system plays the Dial tone to the extension user.
Digit to Digit Wait Timer for External Caller: Configure the Digit to Digit Wait Timer for External Caller.
This is the time for which the system waits for the external caller to dial the next digit. On the expiry of this timer,
the system considers it as the end of number dialing.
Digit to Digit Wait Timer for Internal Caller: Configure the Digit to Digit Wait Timer for Internal Caller.
This is the time for which the system waits for the user to dial the next digit. On the expiry of this timer, the
system considers it as the end of number dialing.
Wait for Answer Timer: Configure the Wait for Answer Timer. This is the time for which the destination
extension rings.
If you have completed configuring the VMS General Parameters, click Submit to save changes.
1325
1 for Enabled
0 for Disabled
Default: Enabled.
To enable/disable Memory Usage Notification to SE, dial:
5082-Code-#*
Where,
Code is
1 for Enabled
0 for Disabled
Default: Disabled
To configure Email ID of the System Engineer for System Alerts, dial:
5083-SE Email ID-#*
Where,
Email ID is upto 64 characters.
Default: Blank.
To select Menu Node option when No Digit is dialed by the caller, dial:
5096-Code-#*
Where,
Code is
1 to Disconnect
2 to Transfer to Operator
To assign Code to Stop Message Recording, dial
5087-Code-#*
Where,
Code is a number string of upto 3 digits (max), consisting of the digits 0 to 9,*,#
To assign codes for functions at the Home Node, dial:
5088-Function-Code-#*
Where,
Function is
1 for Dial by Name
2 for Root Node
3 for Mailbox Management
4 for Disconnect
5 for Leave Message
Code is any single digit from 0 to 9.
Default function codes:
7 for Dial by Name
0 for Root Node
8 for Mailbox Management
# for Disconnect
6 for Leave Message
To enable/disable playing of Call Could Not be Attended message, dial:
5090-Code-#*
Where,
Code is
1 for Enabled
0 for Disabled
1326
Default: Enabled.
To enable/ disable playing of 'Extension busy message', dial:
5091-Code-#*
Where,
Code is
1 for Enabled
0 for Disabled
Default: Enabled.
To enable/ disable playing of 'No reply message', dial:
5092-Code-#*
Where,
Code is
1 for Enabled
0 for Disabled
Default: Enabled.
To enable/ disable playing of 'Leave message' option, dial:
5093-Code-#*
Where,
Code is
1 for Enabled
0 for Disabled
Default: Enabled.
To enable/disable Message Verification option, dial:
5094-Code-#*
Where,
Code is
1 for Enabled
0 for Disabled
Default: Enabled
Exit SE mode.
1327
A Graph: It is the logical path that the caller/extension user takes within the VMS to reach the desired
destination (person or mailbox). A graph starts from the Root Node and traverses through different nodes
these may be Menu Node, Information Node, Transfer Node, Message Node. At each node the VMS offers
the caller the choice of performing some action like deciding what option to choose (menu node), leaving a
message (message node), accessing information (information node), or reaching an extension number
(transfer node).
The VMS plays voice messages to prompt callers at every step, and the caller/extension user dials single
digit codes to decide which path to take and reach the desired destination. To know more, refer the topic
Graphs and Nodes.
You can create 4 different graphs in the VMS, and assign a different graph to each Voice Mail Auto
Attendant Profile.
Operator: The incoming calls on the VMS which need to be transferred to the Operator, are transferred to
this Operator group. For each profile you can select a different operator group.
System Greetings: It is played to the caller/extension user when the VMS answers the call. System
Greetings are played according to the time of the day, Morning, Afternoon, Evening. By default, the caller/
extension user is played the messages Good Morning, Good Afternoon, Good Evening according to
the time of the day.
You can set the start time for the System Greetings. Refer Greeting Message Time in System
Parameters for instructions.
You can also record custom message as System Greetings. Refer the topic Recording Voice Messages.
Welcome Message: It is played to the caller when the VMS answers the incoming call according to the
current Time Zone: Working Hours or Non-Working Hours. By default, the welcome message for each time
zone are:
Working Hours: Welcome! Please dial the extension number Or to dial by name press 7. To leave a
message press 6. To go to operator, press 9. For more options, press 0. To disconnect, press # (hash/
pound).
Non-Working Hours: Welcome! We are closed due to holiday. To leave a message, press 6. For
Assistance press 9. To disconnect press # (hash/pound).
You can also record custom messages as welcome messages for Working and Non-Working Hours. Refer
the topic Recording Voice Messages.
1328
Directly Route to Root Node: This option takes the caller/extension user directly to the Root Node in the
graph, without playing the System Greeting or Welcome Message. This option can be selected according
to the Time Zone-Working, Break and Non-working Hours. The caller/extension user is taken to the Root
Node in the graph, if the option Route to Root Node is selected for the current Time Zone.
Home Node option for Caller on No Digit Dialed: The VMS plays the voice messages to instruct callers
to navigate through the Graph. The VMS waits for the caller/extension user to dial a digit for a certain time
period. If the caller/extension user fails to dial the digit, the VMS times out and connects the caller to the
Operator or terminates the call, or it can be configured to take the caller to the Root Node. This option can
be selected according to the Time Zone-Working, Break and Non-working Hours.
Make sure the Directly Route to Root Node check box is disabled to route calls according to the Home
Node option you select.
You can create 16 different Voice Mail Auto Attendant Profiles, by changing the values of the parameters
described above.
Before you begin configuration of the Voice Mail Auto Attendant Profile, decide how many different Voice Mail Auto
Attendant Profiles you want to create.
For each Profile you want to create, decide:
the path and options you want to give to external callers and to extension users, and accordingly create
the graphs. Refer Graphs and Nodes for instructions.
the System Greeting you want to play to callers in the morning, afternoon and evening.
the Welcome Greeting is to be played for Working, Break and Non-Working hours.
if you want to customize System Greetings or Welcome Messages or any Graph-Nodes messages, and
record the messages. Upload the new voice message files using the embedded FTP server of the VMS
card. Recording Voice Messages.
whether you want the VMS to take the external callers and extension users directly to the Root Node
(without playing the System Greetings or Welcome Message).
whether you want the VMS to take the callers and extension users to the Root Node or Operator or to
Disconnect the call, when they do not dial any digit.
when incoming calls on the VMS need to be transferred to the operator, the group to which these call must
be transferred. Refer Configuring 'Operator' for instructions to create operator groups.
For each Voice Mail Auto Attendant Profile that you decide to create, give a number from 1 to 16.
1329
1330
Click the Voice Mail Auto Attendant Profile sub-link. The Voice Mail Auto Attendant Profile page opens.
Select a Profile number. You may refer to the number you gave to each profile you created.
select the desired Graph you want to use for this profile. By default, Graph 1 is assigned to all profiles.
select the Operator to which the VMS must transfer the calls to. You can select a different operator
group for each profile.
select the number of the Greeting message to be played for the Morning Time, Afternoon Time and
Evening Time. By default, Greeting message 01 (Good Morning), Greeting Message 02 (Good
Afternoon) and Greeting Message 03 (Good Afternoon) are selected as morning, afternoon and
evening greeting messages respectively.
select the number of the Welcome Message to be played during Working hours, Break hours and
Non-Working hours.
By default, Welcome Message 01, 02 and 03 are selected for Working, Break and Non-Working hours
respectively.
Welcome Message 01: Welcome! Please dial the extension number Or to dial by name press 7. To
leave a message press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #
Welcome Message 02: Welcome! We are closed at this moment. To leave a message, press 6. For
Assistance press 9. To disconnect press #
Welcome Message 03: Welcome! We are closed due to holiday. To leave a message, press 6. For
Assistance press 9. To disconnect press #
Select the Directly Route to Root Node check box, if you want the VMS to take the caller/extension
user directly to the Root Node in the graph during Working hours, Break hours and Non-Working hours.
By default, this parameter is disabled in all the time zones.
When Directly Route to Root Node is disabled for a time zone, the VMS will play System and Welcome
Greetings to callers and extension users.
Home Node option for Caller on No Digit Dialed: The VMS plays the voice messages to instruct
callers to navigate through the Graph. The VMS waits for the caller/extension user to dial a digit for a
certain time period. If the caller/extension user fails to dial the digit, the VMS times out and connects
the caller to the Operator or terminates the call, or it can be configured to take the caller to the Root
Node. You can select one of the following option for Working Hours, Break Hours and Non-working
Hours, when no digit is dialed by the caller.
Select Operator, if you want the VMS to take the caller to the Operator when they do not dial any
digit.
Select Disconnect, if you want the VMS to disconnect the call, when the caller does not dial any
digit.
Select Root Node, if you want the VMS to take the caller back to the Root Node in the graph (from
which they originally entered). When the caller is taken to the root node, they will be able to
navigate the graph all over again.
1331
according to the Voice Mail Auto Attendant Profile assigned to the extension, the VMS greets the caller
with the Greeting relevant to the time of the day, plays the Welcome message programmed for the current
Time Zone (Working hours, Break hours, Non-working hours), and offers the different options as per the
graph assigned.
if the incoming call to the VMS need to be transferred to the Operator, they are transferred according to the
option you select as the Operator.
if the Route to Root Node is enabled in the Profile for the current Time Zone, the VMS takes the caller
directly to the root node, without playing the greeting and welcome message.
if no digit is dialed by the caller, the VMS takes the caller to the option you select in Home Node option for
Caller on No Digit Dialed.
You can assign different Voice Mail Auto Attendant Profiles to different extensions and to Department Groups.
To assign Voice Mail Auto Attendant Profile to extensions, configure the Voice Mail Settings on the extensions
and Department Groups.
For SLT extensions, see Voice Mail for SLT Extensions under Configuring SLT Extensions for
instructions.
For DKP extensions, see Voice Mail for DKP Extensions under Configuring DKP Extensions for
instructions.
For ISDN Terminals, see Voice Mail for ISDN Terminal under Configuring ISDN Terminals for
instructions.
For SIP extensions, see Configuring Voice Mail Settings for SIP Extensions under Configuring SIP
Extensions for instructions.
For Department groups, see Voice Mail for Department Group under the feature description for
Department Call for instructions.
1332
first select Voice Mail Auto Attendant as the destination for Routing Incoming Calls during the Working,
Break or Non-Working Hours.
the VMS answers the external incoming call landing on the trunk.
According to the Voice Mail Auto Attendant Profile assigned to the trunk, the VMS greets the caller with the
Greeting relevant to the time of the day, plays the Welcome message programmed for the current Time
Zone (working hours, break, non-working hours), and offers the different options as per the graph assigned
in the Voice Mail Auto Attendant profile.
If the Directly Route to Root Node is enabled in the Profile for the current Time Zone (working hours, nonworking hours), the VMS takes the caller directly to the root node, without playing the greeting and
welcome message.
If calls are to the transferred to the operator, you can select the Operator.You can select a different
operator for each profile.
You can select a different Home Node option when callers do not dial any digit. You can route such calls to
the operator, root node or disconnect them.
You can assign different Voice Mail Auto Attendant Profiles (Graph number, Operator, Greetings and
Welcome Message, Directly Route to Root Node option, Home Node Option for Caller on No Digit Dialed)
to different trunks.
You must configure the Voice Mail System Parameters in the Trunk Feature Template. For instructions,
see Trunk Feature Template Parameters in Configuring Trunks.
You must assign the customized template to the desired trunk port types. For instructions, see
Customizing Trunk Feature Templates in Configuring Trunks.
1333
To program welcome message for all three time slots of the port, dial:
5033-1-Profile Number-Time Zone-Welcome Message#*
Where
Profile Number is from 01 to 16
Greeting Index is 1 for Working Hours, 2 for Break Hours and 3 for Non-Working Hours
Greetings Number is from 1 to 12 and None
To program whether to route the incoming call directly to Root Node for all three time slots of the port, dial:
5034-1-Profile Number-Time Zone-Code#*
Where
Profile Number is from 01 to 16
Greeting Index is 1 for Working Hours, 2 for Break Hours and 3 for Non-Working Hours
Code is 1 to enable and 0 to disable
1334
Exit SE mode.
Decide on which trunks you want to enable Voice Mail Auto Attendant, and the make a list of these trunks
by their port type (CO, Mobile, etc.) and their (software) port number.
configure welcome messages. You may record custom welcome messages in WAV format, or use the
default pre-recorded welcome messages of the VMS, as per your requirements.
configure Auto Attendant parameter (Voice Mail Auto Attendant) for the time zones in the Trunk Feature
Template(s) of the trunks on which you want to use the VMS Auto Attendant.
Auto Attendant: This parameter is to be configured if you want to enable Auto Attendant on the trunk
ports on which you will apply the template.
Auto Attendant can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break
Hours (BH) and Non-Working Hours (NH).
For each Time Zone, you may select the desired Auto Attendant option from the following:
OFF: Select this option if you want to disable Auto Attendant for the Time Zone.
Built-In Auto Attendant: Select this option if you want the calls to be answered by the built-in Auto
Attendant of the ETERNITY. In Built-In Auto Attendant, ETERNITY answers the call using Voice
Modules, if assigned, or it answers the call and plays the appropriate call progress tone - Dial tone,
Ring Back tone, Busy tone - for each call state.
If you select this option, make sure you also configure the Built-In Auto Attendant related Timers and
Flags, record and assign the Built-In Auto Attendant related Voice Message. Refer the topics Auto
Attendant and Voice Message Applications for instructions.
Voice Mail Auto Attendant: Select this option if you want the calls to be answered by the Auto
Attendant of the Voice Mail System. The Voice Mail System of ETERNITY answers calls and
processes them according to the Voice Mail Auto Attendant Profile assigned to the trunk.
1335
If you select this option, make sure you also select and assign the desired Voice Mail Auto Attendant
Profile to the trunk.
By default, Auto Attendant is disabled (OFF) for all the Time Zones.
Auto Attendant Delayed Timer: Set this Timer, if you want to enable Delayed Auto Attendant on the
trunk.
When you enable Delayed Auto Attendant, ETERNITY routes the incoming call on the trunk to the Trunk
Landing Group assigned to this trunk. It waits for the duration of the Auto Attendant Delayed Timer for any
of the extensions the Trunk Landing Group to answer the call.
If none of the extensions in the Trunk Landing Group answers the call before the expiry of the Auto
Attendant Delayed Timer, ETERNITY processes the call according to the type of Auto Attendant - Built-In
Auto Attendant or Voice Mail Auto Attendant - set for the trunk.
To enable Delayed Auto Attendant, set the timer to the desired value. The range of this timer is from 01 to
99 seconds. By default, the timer is set to 10 seconds.
To disable Delayed Auto Attendant, select Never. By default, it is disabled.
Voice Mail Auto Attendant Profile: Select the desired Voice Mail Auto Attendant Profile from 1 to 16, if
you have enabled Voice Mail Auto Attendant for a time zone. By default, 1 is selected.
You may configure VMS Auto Attendant related parameters using Jeeves or by dialing commands from a
Telephone.
1336
Enter the Trunk Landing Group Number for all the time zones - WH, BH and NH. Refer Trunk Landing
Group (TLG) for instructions on configuring trunk landing groups. Also refer the topic Routing Group.
Set the Auto Attendant Delayed Timer, if you want to enable Delayed Auto Attendant.
Select the Voice Mail Auto Attendant Profile. Refer Voice Mail Auto Attendant Profile for more details.
Now, you must assign this customized Trunk Feature Template on the different trunk port types. For
instructions, refer Configuring Trunks.
You may log out of Jeeves if you have completed all configuration tasks.
1337
Check the default values of the following parameters, and change them, if required, to the desired values:
1338
Access Code: By default, 1176 is the access code for the General Mailbox. Extension users must dial
this number, if they want to access the General Mailbox.
If required, you may assign a different access code to the General Mailbox. The Access Code you
assign may consist of a maximum of 16 digits. Digits 0-9, # and * are allowed.
Name: You can assign a Name to the General Mailbox. The name you assign may consist of a
maximum of 18 characters.
Mailbox Size (min.): You may increase or decrease the size of the General Mailbox, by changing the
default Mailbox Size of 999 minutes. You may change the mailbox size to any desired value from
00100 to 60000 minutes. Default: Enterprise mode: 999 minutes; Hotel mode: 999 minutes.
Maximum Message Length (sec): You can define the length of each message (in seconds) callers
are to be allowed to record in the General Mailbox. You may change the maximum message length to
any desired value from 0001 to 9999 seconds. Default: Enterprise mode: 999 seconds; Hotel mode:
999 seconds.
The VMS card will stop recording the message of the callers if it exceeds the maximum message
length, and will store only that part of the message recorded within the maximum message length limit.
New Message Delivery Option in Mailbox Full condition: When the General Mailbox is full, you may
select one of the following options for delivery of new messages:
Do not offer to record a message: The VMS will not allow the caller to record a message by
declining delivery of the message.
Overwrite old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS starts overwriting the oldest message first.
By default, Overwrite old messages is selected.
Play message details after delivery of message: After the extension user has finished listening to a
message in the mailbox, you can also have the VMS play to the extension user, message details such
as Date and Time when the message was recorded, the callers number212, and the extension number
dialed by the caller213.
You may select from one of the following options for Play message details:
Never: The VMS will not play message details to the extension user accessing the General
Mailbox.
Always: The VMS will play message details to the extension user accessing the General Mailbox,
after playing each message.
On Demand: The VMS will play message details to the extension user accessing the General
Mailbox only when the extension user requests for it. On completion of each message, the VMS will
prompt the extension user to press a digit for date and time stamp. When the extension user
presses the digit, the VMS will play the message details.
212. The number of the person who left the message in the mailbox.
213. The number of the extension user for whom the message is intended.
1339
Ask Password to access Mailbox: By default, any extension user can access the General Mailbox by
dialing the General Mailbox Access Code. To prevent unauthorized access to the General Mailbox,
select this check box.
Mailbox Password: If you have selected the Ask Password to access Mailbox check box,
extensions users can access the General Mailbox by dialing the default Password, 1111. You can
change this password, if required. The password you assign may consist of a maximum of 4 digits.
Valid Range: 0000 to 9999.
Auto Delete Old Messages (days): Select the number of days after which you want the VMS to
automatically delete old messages (read messages) in your mailbox.
For example, a message is received on July 20, read on July 21. If you set Auto Delete Old Messages
as 5. The VMS will automatically delete the read message on July 26, 5 days after the message has
been read.
Email Address: Enter the email ID to which the notification is to be sent. The notification will be sent to
this email address when, either 80% or 100 % of the mailbox memory has been consumed. The Email
Address may consist of up to 64 characters. Default: blank.
For more information and instructions on how to access the General Mailbox, see Accessing the General
Mailbox.
1340
1
2
Message
Node
2
3
Information
Node
3
4
Transfer
Node
4
5
6
Transfer
Node
Transfer
Node
Callers must dial appropriate digits while traversing through the graph to reach the desired destination. Single digit
codes are used to navigate within a graph.
While traversing through the graph, the VMS uses voice messages to prompt callers at every step so as to enable
them to reach the desired destination.
Let us take the example of the 6-node graph illustrated above to understand graphs and nodes. In this 6-node
graph, the caller reaches the Root Node by dialing the digit '0'. The Root Node is the entry point of the graph.
At the Root Node, the VMS offers the caller different options like leaving a message, accessing Company
information, have the call transferred to a mailbox, or a particular extension, operator, or department.
For each option, the VMS prompts the caller to dial a particular digit. Each option takes the caller to a particular
node.
Dialing '1' at the Root Node takes the caller to the Message Node.
Dialing '2' at the Root Node takes the caller to the Information Node.
Dialing '3', '4' or '5' at the Root Node will take the caller to the Transfer Node.
The caller decides which path to take and dials the single digit code for the desired path announced by the VMS in
the voice prompt.
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Menu Node
This node is the point in the graph, at which the VMS presents the callers with different Options in the voice prompt.
The caller is given a menu to select from, hence the name Menu Node. The caller must choose an option offered
in the voice prompt to navigate through the graph further.
There may be several menu nodes in a graph. In this sense, the Root Node is the first Menu node in a graph (see
6-node graph).
A different voice prompt can be recorded and played to the caller for each Menu node.
At a menu node, a caller can be offered 12 different options. The digits 0 to 9, * and # can be programmed for
selecting the options. The caller dials the digit for the desired option to traverse from one node to another. If the
caller does not dial any digit, you can configure a node of your choice to redirect the caller.
Information Node
The Information node is a point in the graph at which the VMS delivers information to the caller. When the caller
reaches the Information node, the VMS plays the recorded message.
The Information node is a useful when you want to provide information related to the company/organization,
company profile, products, new launches, distribution network, etc.
If the caller does not dial any digit, you can configure a node of your choice to redirect the caller or you can
configure the VMS to replay the message for the desired number of times. After playing the message for the
desired number of times, you may either redirect the caller to operator or disconnect the call.
The Information node can be reached only from a Menu node.
A different voice prompt can be recorded for each information node.
Transfer Node
This node is a point in the graph from where the VMS takes the caller to a specific extension. Each transfer node is
linked to an extension. When a caller reaches the transfer node, the VMS transfers the call to the destination
extension which is assigned to the transfer node.
1342
Thus, the transfer node can be used for allowing callers to reach an extension of a department group, a help line,
an information desk, etc.
A voice prompt may be played to the caller at the Transfer node. The VMS then transfers the call to the extension
linked to the node.
Transfer node can be reached only from a Menu node. A graph may terminate at the Transfer node.
Disconnect Node
This node is a point in the graph at which the VMS disconnects the call after playing an appropriate message and a
disconnect prompt to the caller.
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Menu
Node
Menu
Node
2 3
1 2 3
Information
Node
Information
Node
Information
Node
11
Message
Node
Menu
Node
13
12
Message
Node
Message
Node
3
Menu
Node
Menu
Node
Menu
Node
10
14
15
Transfer
Node
2 3
1 2 3
16
Transfer
Transfer Node
Node
17
20
18
19
Transfer
Node
21
Transfer
Node
22
Transfer
Node
A graph initiates from the Root Node and traverses through many menu nodes. It may terminate at an information
node or a message node or a transfer node, but never at a menu node.
1344
New
Customer
(MN)
Existing
Customer
(MN)
Representative
for Complaints
(TN)
Insurance
(IN)
Mutual
Funds
(IN)
Stock
Market
(IN)
Representative
for Stock booking
(TN)
Information
on Shares
(TN)
Representative
(IN)
Representative
for Insurance
(TN)
Representative
for Mutual Funds
(TN)
Information on
Services Provided
(IN)
Information
on Insurance
(TN)
Information
on Mutual Funds
(TN)
214. These may be incoming trunk calls answered by the VMS Auto Attendant, or calls made by extension users to the VMS to access
their mailbox.
1345
Click Graph.
The page of Graph 1 opens. To select a graph number, you may click the tab.
1346
A graph has maximum 64 nodes. For each Node Number, select the Node Type.
If you want to give more options to the caller/extension user at this point in the graph, select Menu as
Node Type.
The respective field of the Extension Number column will become un-editable. At the same time, in
the respective row the fields Destination Node for the Digits, the Destination Node for No digit
dialed/Message play over (Information node) and the Auto-Attendant will become editable.
If you want to give some information to the caller/extension user, select Information as Node Type.
The respective fields of both Extension Number and Destination Node for the Digits columns will
become un-editable. At the same time, the Destination Node for No digit dialed/Message play over
(Information node) field will become editable.
If you want to give the caller/extension user the option of leaving a message for a predefined user,
select Message to predefined user as Node Type.
The respective fields of both Destination Node for the Digits and the Destination Node for No digit
dialed/Message play over (Information node) columns will become un-editable. At the same time,
the Extension Number field will become editable.
In the Extension Number field, enter the number of the extension, whose mailbox is to be assigned to
the Message to predefined user node.
If you want to give the caller/extension user the option of reaching the desired extension of ETERNITY,
select Transfer as Node Type.
The respective fields of both Destination Node for the Digits and the Destination Node for No digit
dialed/Message play over (Information node) columns will become un-editable. At the same time,
the Extension Number field will become editable.
In the Extension Number field, enter the number of the extension to which the call is to be transferred
from this node.
If you want to give the caller the option of reaching the desired extension user by dialing first three
digits of the name, select Dial by Name as Node Type.
The respective fields of Destination Node for the Digits, the Destination Node for No digit dialed/
Message play over (Information node) and the Extension Number columns will become uneditable.
If you want to give the caller the option to manage their mailbox, select Mailbox Management as Node
Type.
The respective fields of Destination Node for the Digits, the Destination Node for No digit dialed/
Message play over (Information node) and the Extension Number columns will become uneditable.
If you want to give the caller the option to leave a message for any extension user, select Leave
Message as Node Type.
The respective fields of Destination Node for the Digits, the Destination Node for No digit dialed/
Message play over (Information node) and the Extension Number columns will become uneditable.
1347
If you want to give the caller the option to disconnect the call, select Disconnect as Node Type.
The respective fields of Destination Node for the Digits, the Destination Node for No digit dialed/
Message play over (Information node) and the Extension Number columns will become uneditable.
You must configure the desired Extension Number, if you select Message to predefined user or
Transfer as the Node Type.
Select Menu as the Node Type, the option Auto-Attendant will be available. Enable the Auto-Attendant, if
you want to allow callers to dial extension numbers and Operator code while they are in the Menu node.
To enable the Auto-Attendant, select Yes. Default: Disabled.
The start digit of the Extension Number/Operator Code dialed by the caller at the Menu Node may be
the same as the Digit programmed for going to the next node, resulting in conflict. To resolve conflicting
digits, you may configure the Conflict Dialing Timer. For instructions, see System Timers and Counts.
If the Digit programmed for going to the next node and the digit of the Extension Number/Operator
Code dialed by the caller match, the preference will be given to the Digit programmed for going to the
next node.
In the Destination Node for the Digits fields, assign destination nodes to which you want the VMS to take
the caller/extension user on pressing the digits 0 to 9, # and * from the Menu node.
By default, the destination node number is '00' for all the digits at menu node. To de-assign a destination
node for a particular digit at menu node, enter '00' for that digit.
If you select Menu as the Node Type, in the Destination Node for No digit dialed/Message play over
(Information node) field, you can configure a node of your choice to redirect the caller, if the caller does
not dial any digit.
You may also configure the Count for No digit dialed/Message play over (Information node). The VMS
will redirect the caller to the Node configured in the Destination Node for No digit dialed/Message play
over (Information node) field, for the number of counts configured by you.
You can select to Disconnect or Transfer to operator as the Option for caller when count is over.
If you select Information as the Node Type, in the Destination Node for No digit dialed/Message play
over (Information node) field, you may configure the same node to replay the message or any other
node.
You may also configure the Count for No digit dialed/Message play over (Information node). The VMS
will repeat the message for the number of counts configured.
If you want the VMS to play the message only once, configure Destination Node for No digit dialed/
Message play over (Information node) as 00.
You can select to Disconnect or Transfer to operator as the Option for caller when count is over.
1348
Menu
Information
Transfer
Dial by Name
Mailbox Management
Leave Message
Disconnect
By default, node type for 01 node is 'Menu' and for all other nodes it is 'Transfer'.
To assign an Extension number for a message/transfer node, dial:
5038-Graph-Node-Extension Number-#*
Where,
Graph is from 1to 4.
Node is from 01 to 64.
Extension Number is the access code you have assigned to the extension.
Default:
To assign unique digit for each option of a menu node, dial:
5037-Graph-Node-Digit-Destination Node-#*
Where,
Graph is from 1to 4.
Node is from 01 to 64.
Digit is from 1 to 9.
Destination Node is from 01 to 64.
Default: 00
To enable the Auto-Attendant
5039-Graph-Node-Auto-Attendant option -#*
Where,
Graph is from 1to 4.
Node is from 01 to 64.
Code is
1 for Enabled,
0 for Disabled,
Default is Disabled.
Exit SE mode.
1349
SMTP Settings
You can use the VMS to send emails for the following functions:
notification to the extension users about the arrival of new messages and the memory usage status of their
mailbox (Refer Email Based Notification).
memory usage notification to SE (Refer Configuring VMS General Parameters).
For email transmission, the VMS uses Simple Mail Transfer Protocol (SMTP). If you intend to use Email Based
Notification and send Memory Usage Notification to SE, you must:
configure the parameter Message Wait Notification via Email215 in the VMS settings of the extension.
configure the parameter Memory Usage Notification to SE in the VMS General Parameters.
configure the SMTP Settings.
Requires Authentication?: Select 'Yes' if SMTP server requires authentication for the e-mail service.
By default, it is set to 'No'.
If you enable Requires Authentication, you must also enter the User ID and Password.
215. You can also have the new voice message mailed as an attachment with the message wait notification.
1350
Enable Secure Socket Layer (SSL)?: Select 'Yes' to enable SSL, if all data to the SMTP server are to
be transmitted over secure layer. By default, it is set to 'No'.
Display Name: Enter the name to be displayed to the mail recipient in this field. The Display Name
may consist of up to 24 characters (maximum). By default, this field is blank.
E-mail ID: Enter the Email ID provided by your SMTP server in this field. This e-mail ID will appear to
recipients as the originator of the e-mail. The e-mail ID may consist of 64 characters (maximum). By
default, it is blank.
User ID: Enter the User ID provided the SMTP server. User ID is required if you have enabled
Requires Authentication? The User ID may consist of a maximum 40 characters. By default, the field is
blank.
Password: Enter the Password provided by the SMTP server. The User Password can be of maximum
24 characters. By default, it is blank.
SMTP Server Address: Enter the SMTP Server Address here. The SMTP Server Address may have a
maximum of 46 characters. By default, this field is blank.
SMTP Server Port: Enter the SMTP Server Port Service Provider in this field. The valid Port range is
from 1 to 65535. By default, SMTP Server Port is 25.
Timers
Connection Timeout Interval: This is the time duration for which the VMS will wait for a response
from the SMTP server. You may change the Connection Timeout Interval timer, if required. The range
of Connection Timeout Interval timer is 01 to 99 seconds. By default, it is set to 60 seconds.
Reconnection Interval: This is the time duration for which the VMS will wait before attempting to
reconnect with the SMTP server. You may change the Reconnection Interval timer, if required. Range
of Reconnection Interval timer is 01 to 10 seconds. By default, it is set to 10 seconds.
Click the button 'Click to Test SMTP to check if the SMTP Parameters have been configured
correctly.
When you click this button, the alert message will appear: "Testing SMTP can take up to 99 seconds.
Would you like to continue?" Click 'OK' button.
The message "Please refresh the web browser after few seconds to check the test mail status" will
appear. Click 'OK' button.
Refresh the web browser after a few seconds. The Test Result will be displayed in 'Test Status' field.
Test Status: Any one of the results listed below may appear in this field:
Test Status Message
Description
1351
Description
1352
1353
Distribution List
A Distribution List enables extension users to send the same message to a group of extensions at the same time.
Any extension with a mailbox can be included in a Distribution List. You can create upto 30 Distribution Lists of 50
members each.
How to configure
To configure Distribution Lists,
It would be more helpful if you also gave a name to the distribution list, for example, sales, marketing,
customer care, etc.
Draw a two-column table on a piece of paper for each Distribution List. In one column, write the number of
the member in serial order. You may write the numbers corresponding to the number of members you wish
to include in this list.
Now, in the next column, for each member number, write the extension number (this is the access code
you have assigned to the extension).
Distribution List for Marketing
Members
Distribution List 1
01
02
03
:
010
Distribution List 2
01
02
03
:
08
1354
Now, refer to the Distribution lists you created on paper, and enter the same information in the appropriate
fields on this page.
1355
Announce Name can be enabled on all extension types, SLT, DKP, ISDN Terminals, SIP Extensions, Department
Goups. For instructions see Configuring SLT Extensions, Configuring DKP Extensions, Configuring ISDN
Terminals, Configuring SIP Extensions, Department Call.
Station Names can be recorded by the System Administrator as well as by the extension users. The instructions
provided here are for the System Administrator only. Instructions for extension users may be found under the topic
Mailbox Settings.
Lift the receiver of any SLT extension or DKP/Extended IP Phone connected to the ETERNITY.
Enter SA mode.
The VMS will prompt you to start recording the Station Name after the beep and press # (hash/pound) sign
to indicate end of message.
As you record the names of each extension, the VMS will create audio files with unique file names, StnName.wav.
This file will be stored in the Mailbox of the extension user. Refer Download Mailbox Messages under VMS
Maintenance for uploading Station Names using FTP.
1356
The status of the VMS Card memory appears on your screen under Voice Mail Memory Status.
1357
Mailbox Status
You can also view the status of individual mailboxes on this page under Mailbox Status.
To view status of individual mailboxes, in Enter Extension Number to view Mailbox Status, enter the
extension number whose mailbox status you wish to view.
Click OK.
1358
Extension Number: This is the extension number you selected for viewing mailbox status.
Personal Mailbox assigned?: If mailbox is assigned to the extension, it displays Yes. If no mailbox is
assigned No will appear.
Mailbox Size (minutes): This is the size of the mailbox in minutes assigned to the extension.
Redirect Set: It displays Yes, if the extension user has redirected the messages of his/her mailbox to
another extension users mailbox. It displays No, if redirection has not been set.
Redirect Station: It displays the number of the station to which the messages have been redirected.
Assistance Number: This field displays the Assistance number configured by the extensions user.
Mobile/Alternate Number: This field displays the Mobile/Alternate Number configured by the
extension user.
Memory Consumed by Mailbox User : This field displays the current status of the mailbox memory of
the extension.
New Messages: This is the number of new (unread) messages in the mailbox.
You can also view the Voice Mail Memory Status from the Status link. To view, click the VMS Memory link
under Status.
In the Delete Messages of Extensions From box, enter the extension number from which you want the
VMS to start deleting the messages. In the To box, enter the last extension number till which you want the
VMS to delete the messages.
The valid digits are 0 to 9, * and #. A combination of # and * cannot be used to define a range of
extensions. For example, you cannot define a range as *2001 to #2010 or vise versa.
To delete messages from the mailbox of an individual extension user, in Delete Messages of Extensions
From and To, enter the same extension number.
1359
1360
From the Delete Messages list, select the type of messages you want to delete No, All, New or
Old.
Select the Delete Personal Greetings check box, to delete the Personal Greetings of the mailbox
user.
Select the Delete Conditional Greetings check box, to delete the Conditional Greetings of the
mailbox user.
Click OK.
A confirmation message Do you really want to Delete All/New/Old Messages in the Mailboxes? appears
on the screen. Click OK to delete the messages.
If you select All from the Delete Message list, the Station Name of the extension users will not be
deleted.
If you select the Delete Personal Greetings check box, the Personal Greetings recorded for all the
time zones will be deleted.
If you select the Delete Conditional Greetings check box, the Conditional Greetings recorded for all
the conditions will be deleted.
You can delete Personal Greetings of a particular Time Zone or Conditional Greetings of a particular
condition using FTP only. For instructions to use FTP, see Uploading Custom Greetings, Messages
and Voice Prompts
Using SA Commands
You can delete messages for a range of extension users using SA command only.
1361
Where,
Start and End Extension Number can be the numbers of SLT , DKP or SIP Extensions, ISDN
Terminals, Department Groups or the General Mailbox.
Message Types is
1 - All Messages
2 - Old Messages
3 - New Messages.
1362
Exit SA mode.
System Greetings: These are voice messages played when a new call lands on the VMS. Callers are
greeted according to the time of the day - morning, afternoon, evening (Time Zone). You can cuteness the
Time Zones as per your requirement. For detailed instructions, see Greeting Message Time. A different
System Greeting can also be played to callers on holidays.
System Greetings are played to callers when the VMS Auto Attendant feature is enabled on trunks.
Personal Greetings: These messages are played to callers when they are diverted to the extension
users mailbox to leave a message. Extension users can record personal mailbox greeting messages of
their choice.
Conditional Greetings: These messages are played to callers when they are diverted to the extension
users mailbox for certain conditionsbusy, no reply or unconditional/unregistered Call Forward.
Extension users can record a different message for each condition.
Welcome Messages: These are voice guidance messages played to the callers who call the VMS.
Welcome messages help the callers navigate through the VMS (graph and its nodes). Welcome messages
are played according to the time of the day, that is, the Time Zone programmed in the system.
Welcome Messages are played to callers when the VMS Auto Attendant feature is enabled on trunks.
Holiday Messages: These are voice guidance messages played to the callers who call the VMS on a
Holiday. These messages are played in place of the Welcome messages and help the callers navigate
through the VMS (graph and its nodes). A different message can be played for each holiday.
Holiday Messages are played to callers when the VMS Auto Attendant feature is enabled on trunks and
the Holiday Table is configured.
Graph-Node Messages: These are voice guidance messages that are played to help the caller take some
action once they have entered the VMS (graph). Refer the topic Graphs and Nodes to know more.
Prompts/Responses: These are voice guidance messages that are played to the caller in response to the
action taken (that is, when the caller dials a digit).
For all of these message types, audio files containing the appropriate recorded voice guidance messages are
loaded in the configuration of the VMS Card. The VMS plays the messages related to the function it is performing.
For example, if Voice Mail Auto Attendant (the VMS Auto Attendant feature) is enabled on a trunk, the VMS plays
messages relevant to the Voice Mail Auto Attendant Profile programmed for the current Time Zone. This helps the
caller navigate through the graph and nodes assigned to the Voice Mail Auto Attendant Profile for the Time Zone.
At each node, the VMS plays the related message, as explained below:
The VMS plays the default System Greeting and Welcome message to the caller according to the time of
the day, e.g.: Good Morning. Welcome. Please dial the extension number or to dial by name, press 7. To
leave a message, press 6. To go to Operator, press 9. For more options, press 0. To disconnect, press #
(hash).
1363
As the caller navigates, the VMS plays the pre-recorded voice messages related to the particular node the
caller has reached. If the caller dials 6 to leave a message, the VMS takes the caller to the message node,
and plays voice message for this node, e.g.: Dial 1 to leave a message, 2 to disconnect, 0 to go to
Home Position.
If the caller dials 1 to leave a message, the VMS plays the prompt: Record your Message after the beep
and press any digit to end.
After the caller has recorded the message and dialed a digit, the VMS plays back the recorded message:
The message you recorded is....
The VMS responds with the option of message verification: To re-record the message, press 1, to confirm
press 2.
If the caller dials 2 to confirm the recorded message, the VMS prompts the caller: Dial 1 to disconnect, 0
to go to Home Position.
If the caller fails to dial any digit, and the call is timed out, the VMS ends the call with the response: Thank
you for your call.
In the same way, when you set a voice guided alarm, the VMS plays the alarm-related voice prompts, like: Enter
the time, HH MM in twenty four hour format. Thus, for every voice mail related function or feature, the VMS plays
the appropriate voice message.
The VMS gives you the option of either using the default voice guidance messages loaded in the VMS Card
configuration, or recording custom messages that better suit your purpose.
All VMS default voice messages are in English only. If you want, you may record voice messages in your local
language.
No special programming is required for using the default voice messages. However, if you want to use custom
messages, you must first:
record the message (Greetings, Welcome Messages, Holiday Messages, Graph Node messages, Voice
Guidance prompt).
upload the new recorded message file in the VMS configuration.
Each voice message file has a unique file name, e.g.: Thankyou.wav containing the message Thank you
for your call; Extbusy.wav containing the message The person you called is busy.
The voice message file is tagged with a unique number, referred to a Prompt Number, e.g. Thankyou.wav
is tagged with the prompt number 108, Extbusy.wav is tagged with the prompt number 100.
The complete list of voice messages by their prompt numbers and file names is provided at the end of this
topic.
There are two ways to record custom voice messages. You may either:
1. record voice messages using a Telephone connected to ETERNITY,
or
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2. record voice messages from any other source, and upload the audio files of the Voice messages on to
Voice messages are in WAV format, so the custom messages must also be in the same format.
When you record voice messages from a Telephone, the VMS stores the message you record in the
required file format, with its unique file name and prompt number. All you need to do is define the
Prompt Number when you record your message.
When you record voice messages from any other source and upload the them in the VMS configuration
files, make sure that the audio files are recorded in .wav file format, with the attributes listed below:
Bit Rate: 128 kbps
Audio Sample Size: 16 bit
Channels: 1 (mono)
Audio Sample Rate: 8 KHz
Audio Format: PCM
The audio file of the custom message you have recorded must have the same unique file name as the
existing default audio file.
Voice Messages can be recorded and played back from the System Administrator mode.
Extension users can record their personal mailbox greetings on their own. Refer the topic Recording
Personal Greetings to know more.
Extension users can record the conditional greetings on their own. Refer the topic Recording Conditional
Greetings to know more.
Lift the receiver of any SLT or DKP extension phone connected to the ETERNITY.
Enter SA mode.
Prompts/ Response
01
02
03
04
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System Greeting
Filename
Number
Prompts/ Response
05
06
07
08
09
10
11
12
Only three System Greeting files are programmed. Greeting number 04 to 12 are blank. You may record
your custom greetings on any of these files.
The VMS will prompt you to start recording your messages after the beep and press # (hash/pound) sign
to indicate end of message.
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Prompts/ Response
01
Welmsg01.wav
02
Welmsg02.wav
03
Welmsg03.wav
04
Welmsg04.wav
(Hotel)
Welcome! Please dial the room number. To leave a message for the
guest, press 6. To go to Operator, press 9. To disconnect, press # (hash).
05
Welmsg05.wav
06
Welmsg06.wav
Welcome! We are closed for a short break. Please press 9 for assistance
or press # (hash) to disconnect.
07
Welmsg07.wav
08
Welmsg08.wav
Number Filename
Prompts/ Response
09
Welmsg09.wav
10
Welmsg10.wav
(Hotel)
Welcome! Please dial the room number. To leave message for the guest,
press 6. To go to operator, press 0. To disconnect, # (pound).
11
Welmsg11.wav
12
Welmsg12.wav
Welcome! We are closed for a short break. Please press 0 for assistance
or press # (pound) to disconnect.
The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
To check the recorded Welcome Message, dial:
Filename
Prompts/ Response
HolidayMsg01.wav
HolidayMsg02.wav
HolidayMsg03.wav
HolidayMsg04.wav
5
6
7
8
9
10
11
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Holiday
Message
Number
Filename
Prompts/ Response
12
Only four Holiday Message files are pre-recorded. Holiday Message number 05 to 12 are blank. You may
record your custom messages on any of these files.
The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
Follow VMS prompt to record your message.
To check Mailbox Greeting you recorded for a time zone for an extensions mailbox, dial:
1072-309-Extension-Timezone Index
The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
To check the Voice Prompt you recorded for a node in a graph, dial:
1368
1072-303-Graph-node
The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
1072-313-Prompt Number
The audio file is recorded in the prescribed format (.wav) and attributes.
The audio file of the custom message you have recorded has the same unique file name as the existing
default audio file.
You may recall that each voice message file has a unique file name, and is tagged with a unique Prompt
Number. For example: You have customized the thank you messages as: Thank you for calling (default
message: Thank you for your call). Make sure that the new message is saved with the same unique file
name Thankyou.wav.
You can upload Custom Voice Messages using Windows FTP or FireFTP. If you want to use FireFTP make sure
FireFTP Add-on is installed in your browser.
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window.
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The existing audio files, with their unique names and .wav extension appear.
Go to the location (CD, Pen Drive or Local disk) where you have stored the custom voice messages.
Click the desired folder. The Thank you message is a response, so open the System Prompts and
Responses folder.
Right click, and paste the file you copied in the System Prompts and Responses folder.
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As you are pasting a file with the same file name, you will be prompted if you want to replace the existing
file. Click OK.
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CHAPTER 11
SMS Server
How it works
For this feature to work,
you must have the SMS Server license. See License Management.
make sure your Email Server uses SMTP to send messages and POP3 to receive messages.
you must configure the SMTP Client and POP3 Client parameters in SMS Server of ETERNITY. See SMS
Server - Mail Settings
the users must have valid Email IDs.
you must define the Mobile port through which the messages are to be sent/received (Fixed/LCR). See
SMS Routing.
configure the SMS parameters and the SMS Budget parameters (if required) on the respective Mobile
ports, see Configuring Mobile Trunks.
you must configure the SMS Server parameters as well as the multipart SMS parameters for sending/
receiving SMS (if required). See How to Configure.
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Here,
To forward an SMS to an Email ID, the SMS received on the Mobile Port must be in a specific format.
Illustrated below is as example of the format.
The sender must send the message to the SIM Number of the Mobile Port of ETERNITY, that is
9898012345.
When the SMS is received on Mobile Port (9898012345) in the above format, the system checks the
senders number (9898214345 - John) in the Denied numbers list of the SMS Server.
If the SMS is received in multiple parts the SMS Server combines it into a single SMS.
.
If a match is found in the Denied list, the SMS will be rejected. If no match is found, then it checks the
destination where the SMS is to be forwarded as an email.
The destination can be the recipients Extension number, Name/Group Name or Email ID. The To field in
the message body specifies the destination. To send the same message to multiple destinations, enter the
destinations separating them with a comma or semicolon.
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When the destination is an Email ID, the system will forward the SMS to the recipients Email ID. In this
case, James@matrixrd.org
When the destination is a Name or Extension Number, the system will search for the Name/Number in its
database. When a match is found, the message will be sent to their corresponding Email IDs. In this case
the Email ID of Smith is Smith@matrixrd.org and the Email ID of the Extension user 2001 is
Bella@matrixrd.org.
If a match is not found for the Email ID, Name/Group Name or Extension Number, by default the system
rejects the message. The system also provides you the option to send the message to a specific recipient
(Send to Default recipient), if you do not want to reject the message.
The SMS Server will convert this SMS to an Email. The email will be sent by the SMS Server to the Email
Server and the Email Server will finally deliver it to the recipients.
When the recipients download their emails, it will be as per the format displayed below.
In this Email, the From field contains the Email address of the SMS Server. The To field contains the Email IDs of
the recipients, the Subject contains the Name and/or Number of the sender and the Body of the message contains
the message for the recipients.
The Name will be displayed only if it has been configured and it is found in the system database.
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To forward an Email as an SMS, the sender must send the Email to the SMS Server in a specific format.
Illustrated below is as example of the format.
Here,
1. The sender of the Email is John, John@matrixrd.org.
2. The To field must be the Email ID of the SMS Server, smsserver@domain.com.
3. The Subject must contain the destination where the SMS to be delivered.
The destination can be a Name/Numbers of the users to whom the SMS is to be sent. To send the SMS to
multiple destinations, enter the destinations separating them with a comma or semicolon. Here the
destination, that is the recipients are James, Smith, Bella, 9898981212, 9845544544.
4. The body must contain the message to be sent to the recipients, that is Due to an urgent task tomorrows
meeting is postponed.
5. The senders Email ID or Name will be displayed to the recipients, if you have configured the parameter
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This mail will be sent to the Email Server and then the Email Server forwards it to the SMS Server.
The SMS Server can receive emails from extensions users (system users) or from external users. You can
allow or deny emails from users as per your requirement.
If you want to receive Emails from extension users only, select the Enable Email to SMS forwarding
check box.
If you want to receive emails from extension users as well as external users, select the Enable Email to
SMS forwarding for External Users check box.
Then, the Server checks the Email ID of the sender in the Denied Email list.
If a match is found in the Denied list, the Email will be rejected. If no match is found, then it checks the
destination where the SMS is to be forwarded.
The destination can be the recipients Number or Name. The Subject field of the message specifies the
destination. To send the same SMS to multiple destinations, enter the destinations separated by comma or
semicolon.
If the destination is a Number, the server will check the number in the Denied list. If a match is found the
email will be rejected. If a match is not found, the Server will send the SMS directly to the number using the
Mobile Port of ETERNITY.
If the destination is a Name, the system will search for the Name in its database. When a match is found,
the SMS will be sent to the corresponding Number. In this case, the Numbers of James, Smith, Bella.
The Server sends a return email as well as a delivery status report to the SA/Sender or to Both, informing
that the SMS has been delivered/not delivered.
If a match is not found for the Name, a reply mail is sent to the Sender, informing that the Name is not
found.
The SMS Server converts the Email to a SMS. In this case the SMS delivered to each recipient, will appear
as given below.
From:9898012345
Due to an urgent task tomorrows
meeting is postponed.
From:John@matrixrd.org
Options
Cancel
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In the SMS sent to the recipients, the From field contains the SIM Number through which the SMS Server sent the
SMS. The Body of the message contains the message for the recipients and the Footer/Signature that is, the Email
ID/Name of the sender (if the Signature has been added).
It is recommended that you configure the parameter Send Footer/Signature in SMS, so that the recipient
knows the sender of the message. If this parameter is not configured, only the mobile port SIM number will
be displayed to the recipient.
Bulk SMS
The SMS Server supports Bulk SMS, that is a single message can be sent to multiple numbers. The message can
be sent using the email client by specific users.
How it works
For this feature to work,
make sure you have enabled Email to SMS Forwarding. See Email to SMS Forwarding.
Allowed-Denied Numbers for sending SMS. See Email to SMS Forwarding.
Minimum time delay between sending two consecutive SMS. See Email to SMS Forwarding.
configure the Bulk SMS parameters and the Email IDs of the users allowed to send Bulk SMS. See Bulk
SMS
you must define the Mobile port through which the messages are to be sent (Fixed or LCR). See SMS
Routing
make sure your Email Server uses SMTP to send messages and POP3 to receive messages.
you must configure the SMTP Client and POP3 Client parameters in the SMS Server of ETERNITY. See
SMS Server - Mail Settings
users must have valid Email IDs.
To send the Bulk SMS the email must be received in the following format:
the email must be sent with the Subject Bulk SMS.
the content received in the email will be considered as the SMS text.The text in the email must not exceed
160 characters.
the mail must contain an attachment of the numbers only or numbers and names in csv format. The csv file
can have a maximum of 1000 numbers. The csv file can have two columns Name and Number or have
only one column with numbers. Make sure the columns do not have any header.
When the system receives the email from the user who has requested for Bulk SMS, the system will check this
Email ID in the Allowed Email IDs List for sending Bulk SMS. If a match is found, the system will serve the request.
If a match is not found the Bulk SMS request will be rejected.
The system will check the Allowed Email IDs List for sending Bulk SMS list, only if you have enabled
the Allowed Email IDs to send Bulk SMS check box. If the Allowed Email IDs List for sending Bulk
SMS option is disabled, all the system users (extension users) will be able to send Bulk SMS requests.
The system will serve only one Bulk SMS request at a time. If one Bulk SMS is in progress and the system receives
another request the system will reject it. Bulk SMS supports a single SMS of 160 characters only. It does not
support multi-part SMS.
While the Bulk SMS request is in progress, the system also provides you the option to stop the process at any point
of time if required. The sender must contact the System Administrator to do so.
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After the Bulk SMS request has been served, the system updates the csv file and adds another column, Status. For
each entry the Status of the SMS is updated. The Status column may contain any one of the following:
Sent - When any SMS is sent successfully by the system i.e OK is received from GSM engine.
Failed - When GSM engine gives any error response
Invalid - For any entry where data is found other than number in the .csv file
Denied - When SMS number is found it the Denied list
Limit Exceed - When Daily or Monthly limit is exceeded for all the Mobile ports used for sending the SMS
for any entry.
No Port Available - When all Mobile Ports are configured as "Not allowed to send SMS" which are to be
used for sending the SMS for any entry.
This report is then emailed to the user who had requested for Bulk SMS. The Subject of the report email is the
same as received for Bulk SMS request.
In certain cases the system may not be able to serve the Bulk SMS request. In such cases the system sends a
reply email with the Error message to the user who requested for Bulk SMS. The possible Error conditions and
messages are mentioned below:
Condition
Error Message
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How to Configure
For the SMS Server to function, you need to configure the following parameters,
General Parameters
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By default, the SMS Server is disabled. To use the SMS Server feature, select Enable.
SMS Configuration
Click SMS Configuration to expand.
You can Send SMS through certain fixed Mobile Ports or through different Mobile Ports according to
specific numbers and time.
To send messages through fixed Mobile Ports, select Using Fixed Mobile Port and configure the SMS
Routing-Fixed Port table. For detailed information, see Fixed Port Routing.
To send messages to specific numbers through certain preferred Mobile Port/s during a defined time
interval, select Based on specific Time-Number and configure the SMS Routing-LCR table. For detailed
information, see Least Cost Routing.
Default: Using Fixed Mobile Port.
Configure the Allowed-Denied Numbers for sending SMS list, if you want to allow or restrict sending of
messages from specific numbers. To do this,
Select the Allowed-Denied Numbers for sending SMS check box.
Click on Allowed-Denied Numbers for sending SMS and the Allowed-Denied Numbers for
sending SMS table opens. You can configure upto 999 numbers.
In the Allowed column, enter the numbers from which messages can be sent and in the Denied column,
enter the numbers from which messages cannot be sent.
You can also configure this list by clicking the Allowed-Denied Numbers for sending SMS link under
SMS Server.
Default: Disabled
Configure the Allowed-Denied Numbers for receiving SMS list, if you want to allow or restrict receiving
messages on specific numbers. To do this,
Select the Allowed-Denied Numbers for receiving SMS check box.
Click on Allowed-Denied Numbers for receiving SMS and the Allowed-Denied Numbers for
sending SMS table opens. You can configure upto 999 numbers.
In the Allowed column, enter the numbers on which messages can be received and in the Denied
column, enter the numbers on which messages cannot be received.
You can also configure this list by clicking the Allowed-Denied Numbers for receiving SMS link
under SMS Server.
Default: Disabled
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Select the Number of parts allowed for sending SMS from the list. Default:1
If the SMS length is more than number of parts allowed, you can select either Ignore remaining
part of the SMS or Do not send SMS and send error report to sender. Default: Do not send SMS
and send error report to sender.
If you select Ignore remaining part of the SMS, you can select the Send error report to sender
when SMS length is more than allowed parts check box, if you want to send an error report to the
sender.
If you enable Send error report to sender when SMS length is more than allowed parts, enter the
message you want to send to the sender in the email in Reply error report as an Email to sender
containing text. Default text: Some texts of message are ignored as length of mail is more than
allowed characters.
Configure the Minimum time delay between sending two consecutive SMS (sec) as supported by the
network. Default: 05.
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By default the Enable SMS to Email forwarding is selected (enabled). If you do not want the SMS Server
to forward the SMS as Emails to the users, clear the check box.
Configure the Default recipient of Email when recipient is not specified in SMS. When an SMS is
received without any recipient, the system delivers the SMS as an Email to the default recipient/s
configured here. You can configure upto 5 Email IDs. Default: Blank.
Select the desired option for, When recipient is specified in SMS then forward Email to. You can select
Recipient specified in SMS or Default Recipient. Default: Recipient specified in SMS.
If you want all incoming SMS to be delivered as Email to a specific recipient only, select Default
Recipient.
If you want the incoming SMS to be delivered as Email to the recipients specified in the SMS, select
Recipient specified in SMS
Select the desired option for, If recipient (Name/Number/Email Id) not found in database. You can
select Reject SMS or Send Email to default recipient. Default: Reject SMS.
Select the desired option for, If conflict occurs for recipient Name. You can select Reject SMS or Send
Email to default recipient. Default: Reject SMS
By default the Enable Email to SMS forwarding check box is selected (enabled). Emails received from
extension users only will be forwarded as SMS. If you do not want the SMS Server to forward Emails as
SMS to the users, clear the check box.
By default the Enable Email to SMS forwarding for External Users check box is clear (disabled). Select
this check box, if you want the SMS Server to receive Emails from extension users as well as external
users and then forward them as SMS.
Configure the Allowed-Denied Email IDs to send SMS list, if you want to allow or restrict certain Email
IDs to send SMS. To do this,
Select the Allowed-Denied Email IDs to send SMS check box.
Click on Allowed-Denied Email IDs to send SMS and the Allowed-Denied Email ID list table opens.
You can configure upto 999 Email IDs.
Select the desired option Allow all except programmed in Denied List or Deny all except
programmed in Allowed List.
If you select Allow all except programmed in Denied List, in Denied Email ID column, enter the
Email IDs from which SMS cannot be sent.
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If you select Deny all except programmed in Allowed List, in Allowed Email ID column, enter the
Email IDs from which SMS can be sent.
You can also configure this list by clicking the Allowed-Denied Email ID link under SMS Server.
Default: Disabled.
Select the Send an Email if SMS is delivered to phone check box, if you want a confirmation email to be
sent when the delivery report is received by the Server from the network. Default: Disabled.
If you have enabled Send an Email if SMS is delivered to phone, in Send delivery report status to,
select the recipient to whom the delivery status report must be sent. You can select Sender, System
Administrator or Both. Default: Sender.
If you select System Administrator, the email will be sent to the email IDs configured in Send copy of
SMS.
In Email Reply Text, enter the message you want to send in the Email. The message can a maximum
of 100 characters. Default text: Your Message is delivered to Receiver.
Select the Send an Email if SMS is not delivered to phone check box, if you want a confirmation email
to be sent when the delivery report is not received by the Server from the network. Default: Disabled.
If you have enabled Send an Email if SMS is not delivered to phone, in Send delivery report
status to, select the recipient to whom the delivery status report must be sent. You can select Sender,
System Administrator or Both. Default: Sender.
If you select System Administrator, the email will be sent to the email IDs configured in Send copy of
SMS.
In Email Reply Text, enter the message you want to send in the Email. The message can a maximum
of 100 characters. Default text: Your Message is not delivered to Receiver.
Enable the Send Footer/Signature in SMS check box and select the desired option to be sent:
Specific Text: Enter the text/message to be sent to the recipient as signature in the SMS.
Send Email ID of Sender: Enter the Email ID of the Sender to be sent to the recipient as signature in
the SMS.
Send Name of Sender: Enter the Name of the Sender to be sent to the recipient as signature in the
SMS.
It is recommended that you configure the parameter Send Footer/Signature in SMS, so that the recipient
knows the sender of the message. If this parameter is not configured, only the mobile port SIM number will
be displayed to the recipient.
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Bulk SMS
Click Bulk SMS to expand.
By default Bulk SMS is disabled. If you want the SMS Server to send Bulk SMS, select the check box.
Select the Allowed Email IDs to send Bulk SMS check box and configure the Allowed Email IDs to
send Bulk SMS list. Only these users will be able to send Bulk SMS. To configure the Email IDs,
Click on Allowed Email IDs to send Bulk SMS and the Allowed Email IDs List for sending Bulk
SMS table opens. You can configure upto 64 Email IDs.
Enter the Email IDs of the users who can send Bulk SMS.
You can also configure this list by clicking the Allowed Email IDs List for sending Bulk SMS link
under SMS Server.
Click Submit to save the entries.
Default: Disabled.
Condition/ Activity/Event
Error Cause 1
Error Cause 2
Error Cause 3
Error Cause 4
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Error Cause
Condition/ Activity/Event
Error Cause 5
Error Cause 6
Error Cause 7
Error Cause 8
Error Cause 9
Error Cause 10
If required you can modify the Email reply text as per your requirement. To do this,
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The Error Cause table displays the Error Cause number with the corresponding Email reply text.
Select the Error Cause Number for which you need to edit/change the reply text.
Click Submit.
You can generate a report of all the errors, see SMS Server Reports for more information. These Error Causes
are also logged into the Fault Log. See System Fault Log for more information.
Select the Send copy of each SMS to Email ID check box and configure the Email IDS to which this
copy of SMS must be sent by the SMS Server.
You can configure upto 5 Email IDs. The Email IDs can maximum of 64 characters.
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SMS Routing
The SMS Server can send SMS using any of the following methods:
Fixed Port Routing - through a single/fixed group of Mobile Ports.
Least Cost Routing - through selective preferred Mobile Ports grouped together in order to utilise the
benefits offered by the service providers, such as 1000 free SMS in a month, reduced rates to send
messages etc.
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Click SMS Routing-Fixed Port. The SMS Fixed Port table opens.
Rotation: By default, all the messages will be sent using the first Mobile Port enabled by you.
Select the Rotation check box, when you want the first SMS to be sent through the first Mobile Port,
the subsequent SMS through the next Mobile Port and so on. For example, if three Ports MOB Port 1,
MOB Port 2, MOB Port 3 are selected and there is a request to send 5 SMS. Then, the first SMS will be
sent through MOB Port 1, second SMS through MOB Port 2, third SMS through MOB Port 3, fourth
SMS through MOB Port 1 and so on.
If the Rotation check box is cleared, all the SMS will be sent using the first Mobile Port enabled by you.
For example, if three Ports MOB Port 1, MOB Port 2, MOB Port 3 are selected and there is a request to
send 5 SMS. Then, each SMS will be sent through MOB Port 1 only.
Name: This is the name assigned to the Mobile Port. This will be displayed only if you have assigned a
name to the Mobile Port on the Mobile Port Parameters page. See Configuring Mobile Trunks.
Enable: Select the Enable check box corresponding to the Mobile Port you want to use for sending the
SMS.
Click Submit.
Make sure that you have selected the Send SMS check box for this Mobile Port. See Configuring Mobile
Trunks.
If you have configured only the Time, the system will check the time while sending the SMS and then route
it according to the selected preference. For example you want to send SMS from one group of trunks
during 9:00 a.m. to 2:00 p.m. and from 3:00 p.m. to 8:00 a.m. through another group.
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If you have configured only the Numbers, the system will check the destination number while sending the
SMS and then route it according to the selected preference. For example SMS to numbers that begin with
99 and 97 can be routed through different trunk groups.
If you have configured both, the time and number, the system will check the time as well as the destination
number while sending the SMS and then route it according to the selected preference.
Configuring LCR
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Time Zone 1 to 4: Configure the Start Time and End Time for each time zone. You can configure four
different time zonesTime Zone1, 2, 3 and 4 as per your requirement.
Number: Configure the destination numbers to which the messages are to be sent.
Preference1 to 4: Select the trunks in the order of preference through which you want to send the SMS.
You can select upto 4 preferences.
How to configure
SMTP Configuration
Contact your Network Administrator for the following information and configure the parameters as per the
configurations done in the Email Server to register the SMS Server as an SMTP Client.
Click the SMTP Configuration link to expand.
If your Email Server uses authentication, select Requires Authentication as Yes. Default: No. If you have
enabled authentication, you must also configure the User ID and the Password.
To transport all data in a secure manner, select Enable Secure Socket Layer (SSL) as Yes. All the data
to the Email Server will be transported over secure layer. Default: No.
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Configure the Display Name. This name will be displayed to the mail recipient. You can configure a
maximum of 24 characters. Default: Blank.
Configure the Email ID for the account registered with the Email Server, as provided by your network
administrator. This Email ID will appear to the recipient as the originator of the email (that is in the FROM
field). The Email ID you configure may consist of a maximum of 64 characters. Default: Blank.
If you have enabled authentication, configure the User ID and the Authentication Password as provided to
you by your network administrator. The User ID may consist of a maximum of 40 characters and the
Password can be a maximum of 24 characters. Default: Blank.
Configure the SMTP Server Address and SMPT Server Port. This is the Servers IP Address and Port
number that is used to send outgoing mails. If port is not programmed, use the default port value equal to
25. The Server Address can be a maximum of 40 characters.
Timers
Connection Timeout Interval is the time period for which SMS Client of the SMS Server tries to connect
with the Email Server, for the first time. Configure this time period in Connection Timeout Interval. Valid
Range: 01 to 99 seconds. Default: 60 seconds.
Reconnection Interval is the time interval after which SMTP Client of the SMS Server tries to reconnect
automatically with the Email Server, when the connection is not established with the Email Server the first
time. Configure the time interval in Reconnection Interval. Valid Range is 01 to 10 seconds. Default: 10
seconds.
After you have configured the SMTP parameters and submitted them, the Click to Test SMTP Settings
button appears.
Click the 'Click to Test SMTP button to check if the SMTP parameters have been configured correctly.
When you click this button, the alert message appears: "Testing SMTP can take up to 99 seconds. Would
you like to continue?" Click the OK button.
The message "Please refresh the web browser after few seconds to check the test mail status" appears.
Click OK button.
Refresh the web browser after a few seconds. The Test Result will be displayed in the 'Test Status' field.
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Test Status: Any one of the results listed below may appear in this field:
Test Status Message
Description
Description
POP3 Configuration
Contact you Network Administrator for the following information and configure the parameters as per the
configurations done in the Email Server to register the SMS Server as a POP3 Client.
Click the POP3 Configuration link to expand.
If the Email Server uses authentication, select Requires Authentication as Yes. Default: No. If your Email
Server uses authentication, you must also configure the User ID and the Password.
To transport all data in a secure manner, select Enable Secure Socket Layer (SSL) as Yes. All the data
to the Email Server will be transported over secure layer. Default: No.
Configure the Email ID for the account registered with the Email server, as provided by your network
administrator. The Email ID you configure may consist of a maximum of 64 characters. Default: Blank.
This Email ID will be received in the FROM header to the users, whenever any mail is received by the
system.
If you have enabled authentication, configure the User ID and the Authentication Password as provided to
you by your network administrator. The User ID may consist of a maximum of 40 characters and the
Password can be a maximum of 24 characters. Default: Blank.
Configure the POP3 Server Address and POP3 Server Port. This is the Servers IP Address and Port
number that is used to download incoming mails. For example, Email Server address is 192.168.1.1 and
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port is 1400, then configure the Server Address as 192.168.1.1 and port as 1400. If port is not
programmed, use the default port value equal to 110. The Server Address can be a maximum of 46
characters.
Click Submit to save settings.
Timer
Download interval Timer (min.) is the time interval after which the POP3 Client of the SMS Server retries
to fetch new mail from the Email Server. Valid Range is 01 to 99 minutes. Default: 01 minute.
After you have configured the POP3 parameters and submitted them, the Click to Test POP3 Settings
button appears.
Click the 'Click to Test POP3 Settings button to check if the POP3 parameters have been configured
correctly.
When you click this button, the alert message appears: "Testing POP3 can take up to 99 seconds. Would
you like to continue?" Click the OK button.
The message "Please refresh the web browser after few seconds to check the test mail status" appears.
Click OK button.
Refresh the web browser after a few seconds. The Test Result will be displayed in the 'Test Status' field.
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Test Status: Any one of the results listed below may appear in this field:
Test Status Message
Description
ETERNITY allows you to set a variety of filters for printing the SMS Server Reports. ETERNITY supports Syslog
Client for SMS Server Reports. The Syslog Client enables the system to send records in syslog format to the
remote Syslog Server. You can view the records on the remote server and print.
The Report contains the following information for each transaction:
Index
Email
Direction
Email Address
Status
Date
Time
SMS
Direction
Number
Status
Date
Time
Part of SMS
Mobile Port
Text
To generate the reports you must configure the following parameters:
General Report Settings
Scheduled Report Settings
Error Report Settings
Report Backup Parameters
Report Filters
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In the Destination IP:Port, enter the IP Address and Port of the remote Syslog Server.
If you have opted for Scheduled Reports, in When scheduled backup is done, send an email to,
enter the desired email ID. The report will be generated and sent to this email ID.
In the Destination IP:Port, enter the IP Address and Port of the remote Syslog Server.
How to use
You can print Reports whenever you want or schedule printing of the report from the System Administrator
mode.
You must set the filters as per your requirement before you print the Report. To do this,
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Open Jeeves.
Click Filter Report to set the various filters as per your requirement:
Select the Direction. You can select SMS to Email or Email to SMS or Both.
Select the Mobile Port/s using which the SMS are sent/received. You can select the desired range in
the From and To fields.
Select the Dates during which the SMS/Emails are sent/received.You can select the desired range in
the From and To fields.
Select the Time duration during which the SMS/Emails are sent/received.You can set the desired
range in the From and To fields.
Select the type of SMS Status. You can select from the following:
All
Pending
Sent
Delivered
Not Delivered
Received
Failed
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Select the type of Mail Status. You can select from the following:
All
Sent
Received
Pending
Failed
The minimum and maximum number of parts in which an SMS can be sent is from 1 to 8. Select the
Part of SMS for which you want to generate the report. After you have selected the Part of SMS value,
select the desired filterAll, Equal to, Less Than or More thanto be applied to that value. For
example is you select 5 as the Part of the SMS and More than as filter, the report will be generated for
all the SMS sent in more than 5 parts.
If you want reports to be generated for certain numbers, select the Filter Numbers check box and
configure the desired numbers in the table.
If you want reports to be generated for certain Email IDs, select the Filter Email Ids check box and
configure the desired Email IDs in the table.
Click Submit.
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To Generate schedule backup, select the desired option a particular day, day of the week, or day of the
month.
To generate the report manually, click the Report under SMS Server.
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Click Export, if you wish to save the report at the desired location.
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In Current status of Bulk SMS process, the status of the Bulk SMS process is displayed.
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If the Current status of Bulk SMS process displays Running, click the Abort button to stop the ongoing
process midway.
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CHAPTER 12
Land the call on a group of extensions, referred to as Trunk Landing Group (TLG).
Route the call to the Built-In Auto Attendant.
Route call to the Voice Mail Auto Attendant.
Greet the caller with Trunk Auto Answer, and then route the call to the Trunk Landing Group.
Route the call to a specific extension on the basis of the CLI received.
It is also possible to configure a different routing option for each time zoneWorking Hours, Break Hours, Nonworking hoursaccording to your call routing requirement.
Besides the type of routing you configure, there are certain features supported by ETERNITY on trunks that also
determine how an incoming call will be routed, when these features are enabled on the trunk.These features are:
Call Back on Trunk Ports: Missed calls received on a trunk are returned to the same or to an alternative
number.
Direct Inward System Access (DISA): From a remote location, users can make a call to an extension of
ETERNITY. The call is routed to an extension after authentication and the users can use the system
resources (make calls, access features, configure the system).
RCOC (Return Call to Original Caller)217: When an unanswered outgoing call is made by an extension is
returned by the called party, this call is routed to the very extension that originally made the outgoing call.
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How it works
When a call lands on a trunk, the system checks the trunk features and the routing option configured for the current
time zone on the trunk in the following sequence and accordingly routes the call.
1. Call Back on Trunks
2. DISA - CLI Authentication
3. RCOC (Return Call on Original Caller)
4. DDI Routing and DISA - CLI Authentication
5. DDI Routing and DISA - PIN Authentication
6. DDI Routing
7. DISA - PIN Authentication
8. Auto Attendant - Built-In or Voice Mail
9. Trunk Auto Answer
10. Trunk Landing Group
In the default Incoming Call Routing configuration, incoming calls on a trunk of ETERNITY are routed to a Trunk
Landing Group (TLG).
The call is landed on the The Trunk Landing Group (TLG) assigned to the trunk for the current time zone.
By default, Trunk Landing Group number 01 assigned to all trunks for all time zones and has DKP software
port 001, SLT software ports 001 and 002 as members.
The member extensions in the TLG start ringing in the sequence in which they are arranged in the TLG.
Each member extension rings for the duration of the Ring Timer (programmable; default: 15 seconds).
If Continuous Ring is enabled, each extension will ring continuously till the call is answered. The extension
continues to ring even as other extensions in the group are hunted.
If the call remains unanswered even after the last extension in the group has been hunted, the system will
loop back and start hunting from the first extension, all over again.
If Rotation is enabled on a TLG, for each new call on a trunk, the system will land the call on the extension
next to the one that received the last call. Thus, ensuring an equal distribution of incoming calls on all
member extensions of the TLG.
When Rotation is disabled, for each new call on a trunk, the system will land the call on the first free
extension of the TLG.
To know more about how TLG works, see Trunk Landing Group (TLG).
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You can also route calls that remain unanswered by extensions in the Trunk Landing Group to the Voice
Mail Auto Attendant.
When a call lands on the trunk, the system checks the trunk features and the routing option configured for
the current time zone on the trunk in the sequence mentioned earlier.
When Auto Attendant is configured for the current time zone, the system checks for the type of Auto
Attendant configured for the current time zone.
If Built-In Auto Attendant is selected, the call is routed accordingly. See description forBuilt-In Auto
Attendant in this topic.
If Voice Mail Auto Attendant is selected the call is processed as per the Voice Mail Auto Attendant
Profile assigned to the trunk.
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When a call lands on the trunk, the system checks the trunk features and the routing option configured for
the current time zone on the trunk in a sequence mentioned earlier.
When Trunk Auto Answer is enabled on a trunk, the system checks for the type of Trunk Auto Answer set
on the trunk.
Feature Interactions
As mentioned earlier, certain trunk features also determine incoming call routing. Consider the following feature
interactions between Trunk Auto Answer and other trunk features when configuring Trunk Auto Answer as a routing
option on a trunk.
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Call Back on Trunks: This feature should be disabled when using Trunk Auto Answer.
DISA - CLI Authentication: This feature should be disabled when using Trunk Auto Answer.
Return Call to Original Caller (RCOC): If both Trunk Auto Answer and RCOC are enabled on the same
trunk, the call will first be answered as per Trunk Auto Answer, and then placed on the destination as per
RCOC.
CLI Based Routing: If both Trunk Auto Answer and CLI Based Routing are enabled on the same trunk,
the call will first be answered as per the Trunk Auto Answer, and then placed on the destination as per the
CLI Based Routing Table.
DISA - PIN Authentication: If both Trunk Auto Answer and DISA PIN Authentication are enabled on the
same trunk, only DISA - PIN Authentication will work. The call will be treated as a DISA call and routed as
per the DISA logic.
Auto Attendant: If both Trunk Auto Answer and Auto Attendant are enabled on the same trunk, only Auto
Attendant will work. The call will be processed by the Auto Attendant.
When CLI Based Routing is configured for the current time zone, the system checks the entries of the CLI
Based Routing Table.
If the calling partys number is found in the CLI Based Routing Table, the call is placed on the
corresponding landing destination.
If the calling partys number does not exist in the CLI Based Routing Table, the call will be routed
according to the incoming call routing option you have configured.
For a detailed feature description, see CLI Based Routing.
DDI Routing works on the basis of the IC Reference Table and the DDI Routing Table.
The T1E1PRI, BRI and SIP trunks ports are assigned IC Reference ID, which may be different for each
time zone. The IC Reference ID is the reference number that acts as an identifier to the translation logic
programmed in the IC Reference Table. When a call lands on the ISDN/SIP trunk, the system checks the
IC Reference ID assigned to it. It checks the IC Reference Table for the corresponding DDI Routing
Reference ID for call resolving.
Using the DDI Routing Reference ID, the system checks the DDI Routing Table for mapping the received
DDI number to a flexible number (target extension).
The system first compares the received DDI number (called party number) with the DDI numbers
programmed in the DDI Routing Table.
Once a perfect match is found, the system checks for the Route on First Destination flag in the IC
reference table. If the flag is enabled, the call lands on the first extension of the MSN number in the DDI
Routing Table. If the flag is disabled, the call is routed to the identified target extension.
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Once the target extension is identified, the system checks the DDI IC routing flag of the extension in its
Station Advanced Feature Template. If the flag is enabled, the call lands on the extension else the call is
routed to the TLG assigned to the trunk.
When the call lands on the DDI extension, the caller gets the Ring Back Tone. The extension rings for
duration of the DDI Ring Timer.
If the call is not answered within the DDI Ring Timer, the system checks for the When No reply option in
the IC reference Table for action to be taken.
Similarly, if the DDI extension is busy, the caller gets Busy Tone. The system checks the When Busy
option in the IC Reference Table for action to be taken.
The options for action to be taken the DDI extension does not No Reply or is Busy are:
Disconnect the call
Route the call to Trunk Landing Group
Answer the call automatically, greet the caller with a voice message, and on completion of message
disconnect the call.
Answer the call, greet the caller with a voice message, and on completion of the message route the call
to Trunk Landing Group.
Route the call to Voice Mail; to the DDI extensions Mail Box.
To know more, refer the topics Direct Dialing-In (DDI), IC Reference Table, DDI Routing Table,
DDI Routing will not work if Auto Attendant or DISA are enabled on a trunk.
To do this, you must configure the desired outbound trunks in an Outgoing Trunk Bundle and create an Outgoing
Trunk Bundle Group (OGTB). You must then include this OGTB as a member of the Trunk Landing Group you
assign to the SIP/ISDN T1E1PRI/ISDN BRI/E&M trunk.
When there is an incoming call on the SIP/ISDN T1E1PRI/ISDN BRI/E&M trunk,
the system checks the trunk features and the routing option configured for the current time zone on the
trunk in a sequence as mentioned earlier.
On finding OGTB as member in the TLG of the current time zone, the call is routed to the OGTB number
assigned.
The system out dials the Called Number received in the incoming call from the OGTB.
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How to configure
Incoming call routing options for a trunk are to be configured on the Trunk Feature Template assigned to the trunk.
First configure a Trunk Landing Group. You may configure the same Trunk Landing Group for working,
break and non-working hours or you different Trunk Landing Group for all three time zones.
To land calls on the Operator extension first, make sure the Operator extension is included in the TLG you
configure. Also make sure that the Operator extension is the first member in the group.
You can also keep the same Routing Group you configured as Operator group as the Trunk Landing
Group.
Assign the Trunk Landing Group to the trunk in the Trunk Feature Template for each time zone.
To have a number of extensions in the group ring simultaneously, enable Continuous Ring on these
extensions and set the Ring Timer for these extensions to 00 seconds.
To set equal distribution of incoming calls on all extensions in the group, enable Rotation for the entire
group (default: disabled).
Make a list of the trunks by their port type (CO, Mobile, SIP, BRI, T1E1PRI) and port number on which you
want to use the Built-In Auto Attendant.
Configure a Trunk Feature Template with Built-In Auto Attendant enabled for the desired time zones.
Assign this Trunk Feature Template to the desired trunks. The calls landing on this trunk will be answered
by the Built-In Auto Attendant. See Trunk Feature Template, for more information.
Set the Start Time for the Morning, Afternoon and Evening Greeting Messages. Refer Greeting Message
Time in System Parameters for instructions.
Assign Voice Modules for Built-In Auto Attendant Messages. To play to callers pre-recorded voice
messages as Built-In Auto Attendant greetings and to play voice prompts at each stage of the call, you
need to assign Voice Modules for the following Built-In Auto Attendant Messages:
For detailed instructions, see How to configure in Auto Attendant.
If you want the incoming calls to first land on extensions, you must configure the Auto Attendant Delayed
Timer (sec). The call will be placed fist on the extensions in the Trunk Landing Group. If the call remains
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unanswered till the expiry of the timer, it shall be answered by the Built-In Auto Attendant. This is known as
Delayed Auto Attendant on trunks.
For detailed instructions, see How to configure in Auto Attendant.
To route the call to the Voice Mail Auto Attendant, you need to configure:
Make a list of the trunks by their port type (CO, Mobile, SIP, BRI, T1E1PRI) and port number on which you
want to use the Voice Mail Auto Attendant.
Configure a Trunk Feature Template with Voice Mail Auto Attendant. enabled for the desired time zones
and assign the Voice Mail Auto Attendant Profile.
Assign this Trunk Feature Template to the desired trunks. The calls landing on this trunk will be answered
by the Voice Mail Auto Attendant. See Trunk Feature Template, for more information.
For detailed instructions, see How to configure in Auto Attendant.
Configure Welcome and Greeting messages. You may either use the default, pre-recorded welcome
messages of the VMS, or record the custom welcome messages that meet your requirements, in .WAV
file format.
For more information and instructions, see the Configuring Voice Mail System.
If you want the incoming calls to first land on extensions, you must configure the Auto Attendant Delayed
Timer (sec). The call will be placed fist on the extensions in the Trunk Landing Group. If the call remains
unanswered till the expiry of the timer, it shall be answered by the Voice Mail Auto Attendant. This is known
as Delayed Auto Attendant on trunks.
For detailed instructions, see How to configure in Auto Attendant.
Configure the CLI Based Routing Table. Enter the numbers of the calling parties and the numbers of the
corresponding destination extensions.
Enable CLI Based Routing on the desired trunks according to time zones in their Trunk Feature
Template.
For more details, see How to configure in CLI Based Routing.
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Select the trunk on which you want to enable Trunk Auto Answer.
Enable Trunk Auto Answer in the Trunk Feature Template of the desired trunk.
Select the Trunk Auto Answer Greeting message, the Trunk Auto Answer Ring Back Tone Message,
and the Trunk Auto Answer Busy Bye Message for the Working Hours, Break Hours and Non-Working
Hours.Create a Trunk Landing Group with the desired extensions.
Configure the Trunk Auto Answer related Timers, if required. The following Timers are of relevance to the
Trunk Auto Answer Feature:
The DID Inactivity Timer (default: 60 seconds)
The Ring Back Tone Timer (default: 45 seconds)
The Busy Tone Timer (default: 7 seconds)
You may change the duration of these timers from theSystem Timers and Counts page.
The Ring Back Timer and the Busy Tone Timer are also applicable for the Ring Back Tone and the Busy
Tone played for internal calls.
Record and assign Voice Modules for the following Voice Messages related to this feature:
Trunk Auto Answer Greeting Message
Trunk Auto Answer Ring Back Tone Message
Trunk Auto Answer Busy Bye Message
For each of these messages, you can record four different messages.
See the topic Voice Message Applications for instructions on recording and assigning voice modules to
greeting messages.
For detailed instructions, see How to configure in Trunk Auto Answer.
DDI can be enabled on ISDN T1E1PRI, BRI as well on SIP Trunks. Select the trunks on which you want to
enable DDI.
Make a list of the Extensions to whom you want to assign DDI Numbers.
Assign an Incoming (IC) Reference ID in the respective port parameters of the trunks. For instructions, see
Configuring E1 Trunks, Configuring T1 Trunks, Configuring BRI Trunks and Configuring SIP Trunks.
Configure the DDI Routing Table. For instructions, see DDI Routing Table in Direct Dialing-In (DDI).
Configure the Incoming Reference Table. For instructions, see IC Reference Table.
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You must make the Outgoing Trunk Bundles. See Outgoing Trunk Bundle
Make an Outgoing Trunk Bundle Group by assigning these Outgoing Trunk Bundles. See OG Trunk
Bundle Group for detailed instructions.
You may configure the same Trunk Landing Group for working, break and non-working hours or you
different Trunk Landing Group for all three time zones.
Assign the Trunk Landing Group to the trunk in the Trunk Feature Template for each time zone.
The Class of Service assigned to the extension user. Class of Service (CoS) defines the permission an
extension will have on a PBX. It defines the set features of the PBX that the extension is to be allowed
access to.
Feature requirements vary among users and with time. Similarly, certain features that are required during
working hours may not be required during break or non-working hours.
For details, see Class of Service (COS).
The Call Budget assigned to the extension user: The system keeps a tab on the total cost of phone calls
made by extension users. If the budget assigned to the user has been consumed, the system will not allow
the user to make further outgoing calls, but will be able to make internal calls.
The extension user can be assigned a fresh budget, after which s/he can resume making calls.
Call Budget can be enabled on all the extensions as well as on selected extensions. Each extension can
be assigned a different amount depending on the user requirement.
For details, see Call Budget.
The Call Privilege assigned to the extension. In Call Privilege you can define the Toll Control Levels. Toll
Control (or Toll Restriction) is an expense control feature of ETERNITY. It enables you to program the
system so that each extension has a designated calling permission referred to as 'Call Privilege'.
Each type Call Privilege allows the extension to call certain areas and restricts it from calling others. The
extension can also be restricted from the dialing of specific telephone numbers.
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A Trunk Access Code is a short digit sequence dialed from an extension phone instructing the PBX to
assign a trunk line or any trunk line from a group of trunks (OG Trunk Bundle Group) to the user to dial an
external number. For making outgoing calls each extension user must dial a Trunk Access Code.
The outgoing calls are routed through the OG Trunk Bundle Groups assigned to the extensions in the
Station Basic Features Template. All the trunks connected to the system can be bunched in different
groups called OG Trunk Bundle Groups and these OG Trunk Bundle Groups can be allotted to each
extension.
Each OG Trunk Bundle Group consists of a single OG Trunk Bundle or multiple OG Trunk Bundles. The
Outgoing (OG) Trunk Bundle is set of parameters that completely define the grouping of similar channels/
trunks. Bundles of similar trunks/channels only can be formed. The system can hunt for a free trunk within
the bundle as per the set option - Ascending, Descending or Cyclic. For more information, see Outgoing
Trunk Bundle.
An Extension can be allotted different OG Trunk Bundle Group during different timings of the day.
Since there are different trunk lines for making calls and the service providers of these trunks offer different
tariffs for calls made to certain locations or numbers or during a particular time of the day, you can enable
Least Cost Routing on the trunks.
When a call is made from an extension of the ETERNITY, using LCR the system selects the lowest cost
trunk from among all the trunks allotted to that extension to make the outgoing call, depending upon the
type of LCR configured.
For more information, see Configuring LCR.
For LCR to work, all trunks that are allotted to extensions for making outgoing calls, must first be assigned
a Cost Factor.
Cost factor is used for grading trunks in the order of increasing cost of routing calls, from 01 to 99, where
01 signifies least cost and 99 signifies the highest cost. Thus you can grade up to 99 trunks according to
the increasing cost of routing calls.
After assigning Cost Factor to Trunks, you must configure the Type of LCR to be used on Trunks in the
Outgoing Trunk Bundle Group (OGTBG) allotted to the extensions for making calls.
If LCR is not enabled, the outgoing call will be routed through the free trunk from the OG Trunk Bundle
Group.
For more information, see Cost Factor in Configuring LCR.
The Call Budget assigned to the Trunk. Each trunk can be allotted a 'budget' limit for outgoing calls. This
budget limit can be programmed to be reloaded manually each time it is exceeded or at a scheduled date,
either daily or at a particular date of the month.
For more information, see Call Budget on Trunk.
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How it works
When the user dials a number without grabbing a trunk, that is, without dialing a Trunk Access Code, the system
checks:
The Closed User Groups table, if a match is found the call is routed through the OGTBG assigned in the
table. For more information, see Closed User Group (CUG).
If no match is found, the checks for an internal extension number. For the Extension user it checks for the
features enabled in the Class of Service (CoS). If Basic Features are disabled the extension user will not
be able to make internal calls. If in the CoS the Basic Features are enabled and a match is found in the
internal group, the call is placed on the dialed extension.
If still the system is unable to find a match, the systems plays an error tone to the caller.
When the user dials a number after grabbing a trunk line, that is after dialing a Trunk Access Code, the system
checks:
The Station Basic Feature Template assigned to the user. In this template it checks for the type of Toll
Control and Call Budget.
The Outgoing Trunk Bundle Group assigned to the user for making outgoing calls, for the Trunk Access
Code dialed.
The features enabled on the Trunks in the OGTBG, that is Least Cost Routing and Call Budget.
If LCR is not enabled the system allots a free trunk from the OG Trunk Bundle Group to route the call.
If LCR is enabled, the system checks for the type of LCR enabled and routes the call using the
cheapest free trunk from the group. If the cheapest trunk is not free, the system hunts for the second
cheap trunk in group and routes the call. If none of the trunks are free, the system plays a busy tone to
the user.
After checking LCR, for the free trunk the system checks the type of Call Budget enabled on the trunk.
The calls will be routed using this trunk as long as the budget limit set for the trunk (i.e. the Amount or
Minutes or the maximum number of Calls) is not crossed. As soon as the limit is crossed the Trunk is
automatically disabled.
How to configure
For the extension user configure the following in the Station Basic Features Template:
The Basic Features are enabled in the Class of Service (CoS). If you want to restrict dialing of internal calls
by the user, disable the Basic Features. Select a CoS group number, for example 19 and disable the Basic
Features. In the Station Basic Features Template assigned to the user, enter 19 as the CoS for each time
zone.
To restrict calling, enable Call Budget and configure the following parameters for the feature to work:
select the Call Budget check box to enable
select the Toll Control-Call Budget Consumed option
set the Preset Call Budget Amount
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For detailed instructions to assign a CoS, Call Budget, Call Privilege and Outgoing Trunk Bundle Group to an
extension user, see Station Basic Feature Template and Configuring DKP Extensions, Configuring SLT
Extensions, Configuring ISDN Terminals, Configuring SIP Extensions.
You need to provide access of the trunks to the users for making calls, to do so, follow the steps given below:
To create an Outgoing Trunk Bundle put similar trunk types are put together.
Assign the Outgoing Trunk Bundles as members in an Outgoing Trunk Bundle Group. See Configuring
using Jeeves in Outgoing Trunk Bundle Group.
For the Outgoing Trunk Bundle Group, enable Least Cost Routing, if required. Select the type of LCR
required and also configure the respective LCR tables. For detailed instructions, see Configuring LCR.
Enter this Outgoing Trunk Bundle Group number, as per the time zone in the Station Basic Features
Template assigned to the extension user.
Determine the cost factor you want to assign each trunk. Assign the Cost Factor to each trunk type in their
respective trunk parameters.
Enable Call Budget on Trunks. Select the type of Call Budget you want to set on the trunk, in their
respective trunk parameters.
For detailed instructions on assigning the Cost Factor and Call Budget to the trunk, see Configuring CO
Trunks, Configuring Mobile Trunks, Configuring E1 Trunks, Configuring T1 Trunks,Configuring BRI
Trunks, , Configuring SIP Trunks.
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CHAPTER 13
Abbreviated Dialing
What's this?
Abbreviated Dialing is the use of short codes (abbreviated numbers), typically 2-3 digits, to dial out long-digit
numbers. It is also referred to as Memory Dialing.
Abbreviated Dialing allows you to dial quickly and easily, frequently called, long-digit numbers.
This feature requires you to store the frequently called, long-digit numbers218 and their corresponding short codes
in special lists, known as 'directories'. These directories may be 'personal' or 'global'.
ETERNITY supports two types of Abbreviated Dialing based on the type of directory used: Personal Abbreviated
Dialing and Global Abbreviated Dialing.
Abbreviated Dialing forms the basis of two other features of the ETERNITY: Dialed Number Directory and Quick
Dial.
218.These may be numbers of your branch offices, your clients, as also numbers of emergency services such as fire, police.
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For example: personal directory number 02 is assigned to extension 2001. The number 02652630555 is stored at
Index number 16 of this directory. The user of extension 2001 can call this number by simply dialing '8' (feature
access code) followed by '16' (the index number).
ETERNITY will automatically dial out the number using the trunk access code ID specified for this number in the
personal directory.
When an extension user dials an abbreviated number from the Personal Directory, the system first
checks OG Trunk Bundle Group (OGTBG) and Toll Control Level (Call Privilege) of that extension and
then dials out the number.
Each extension can access only the personal directory assigned to it.
Personal Directory can be programmed by the System Engineer, as well as extension users. Extension
users can add contacts to the Personal Directory assigned to them from their extensions phones (DKP/
SLT/ISDN phone/IP Phone).
The Global Directory has Memory Location codes starting from 100 to 999. The telephone numbers along with their
corresponding names are stored against Memory Location codes.
Whenever extension users of ETERNITY want to use Global Abbreviated Dialing, all they needs to do is dial the
feature access code ('8' or '6') and the Memory Location code at which the desired number is stored.
For example: the number 02652630566 is stored at Memory Location 102 of the Global Directory. Now, extension
users of ETERNITY can call this number by simply dialing the '8' or '6' (feature access code for Abbreviated
Dialing) followed by '102' (Memory Location code at which the desired number, 02652630566, is stored).
The ETERNITY will dial out the number using any of the trunks in the OG Trunk Bundle Group assigned to it in the
Memory Location.
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Extensions can use Global Abbreviated Dialing only if this feature is included in the Class of Service
(COS) allowed to them.
Further, an extension can access only that part of the Global Directory which is allowed to it in the CoS.
For instance, if extension 2001 is allowed Global Directory Part 1 in its CoS, the user of extension 2001
can dial out only those numbers contained in Global Directory Part 1.
Matrix ETERNITY System Manual
So, to be able to access the entire Global Directory, extensions must be assigned all three parts of the
directory in their Class of Service. By default, only Global Directory Part 1 is included in the CoS of all
extensions.
Global Directory can be programmed by the System Engineer, and by Digital Keyphone extension
users who have Global Directory Programming allowed to them in the Class of Service.
While the System Engineer can program all three parts of the Global Directory, digital keyphone
extension users who are allowed Global Directory Programming in their Class of Service can configure
only Global Directory Part 1.
How to configure
For both Personal and Global Abbreviated Dialing to work, the System Engineer must:
Assign Personal Directory to the desired extensions (which may be different: SLT, DKP, ISDN Terminals,
IP Phones).
Enable Global Directory Part 1/2/3 as desired in the Class of Service (CoS) group allowed to the
extensions.
Enable Global Directory Part 1 and Global Directory Programming in the CoS group of the digital key
phone extension users, who are to be allowed to program (add, delete, edit) contacts in the Global
Directory Part 1 from their digital keyphones.
All the above parameters can be programmed by the System Engineer using Jeeves as well as a telephone.
Ask the extension users the numbers they would like to be included in the personal directory of their
extension.
Make separate lists of numbers along with their corresponding names and trunk access codes, for each
personal directory. You may draw four-column tables on paper and enter the Numbers and corresponding
names and trunk access codes against each Index number. For example:
Personal Directory 01
Index No.
Number
Name
TAC ID
1
2
:
1419
Index No.
Number
Name
TAC ID
Number
Name
TAC ID
25
Personal Directory 02
Index No.
1
2
25
Compile the numbers to be included in the global directory. Numbers that are commonly dialed by all
extensions can be included in the global directory.
Draw a four-column table on paper and enter the telephone numbers along with their names, and the
Outgoing Trunk Bundle Group (OGTBG) at each Memory location. For example:
Global Directory
Memory Location
OGTBG
Number
Name
100
101
999
Prepare the Global Directory keeping in mind that is divided into three parts: Part 1 (100 to 799), Part 2
(800 to 899), and Part 3 (900 to 999). As Part 1 is allowed to all extensions in their default CoS, you may
include the numbers allowed to all extensions in this part of the directory.
1420
To upload Personal Contacts CSV file using Jeeves, see Upload Personal Directory CSV files .
For Global directory contacts, the format of the CSV file must be as follow: For example 121, 9867985489, Sean
Gilbert, Sean@hotmail.com, Accounts, 01,000
Where,
121 is the Index Number at which you want the entry to be stored (mandatory)
9867985489 is the Contact Number (mandatory)
Sean Gilbert is the Contact Name (optional)
Sean@hormail.com is the Email ID (optional)
Accounts is the Group (optional)
01 is the Outgoing Trunk Bundle Group (OTBG) to route the call (optional)
000 is the Alternate Number Group (optional)
To upload Global Contacts CSV file using Jeeves, see Upload Global Directory CSV file.
1421
You can configure upto 50 Personal Directories. Select the directory number by clicking the required
number tab above the table.
For each directory configure the following parameters,
Enter the Number you wish to store against an Index Number. Enter the contact's Name against the
number.
The length of the Number field is limited to 16 digits. The length of the Name field is limited to 12
alphanumeric characters. All ASCII characters except < > and (double quote) are allowed. Ensure
that the number and the name are programmed within this limit.
Each directory has a limit of 25 entries. You may enter up to 25 Numbers and Names in each Personal
Directory.
Change the TAC Index (TAC ID), if required.
Keep a print of each personal directory for your record and for the record of the extension user to whose
phone the personal directory is assigned. This will also help you take care of overlaps and include some of
the numbers that are dialed by all users in the Global Directory instead of the Personal Directory.
1422
In Personal Directory Number, select the number of the personal directory in which you want to
upload the contacts from the CSV file.
Select the Clear all other indices of the selected Personal Directory, which are not specified in
the .csv file being uploaded check box, to overwrite the existing contacts in the Personal directory
with the contacts of CSV file. Default: Disabled.
Click the Browse button to Select the .csv file to be uploaded from the location on the local disk.
All the contacts of the CSV file will be uploaded in the selected Personal Directory. To view, click the
respective Personal Directory link.
1423
In Personal Directory Number, select the number of the personal directory from which you want to
download the contacts.
You will get a prompt with an option to open the Opening Personal_Directory_01.csv file or save the
file to a location. Save the file on the local disk.
Open the Personal_Directory_01.csv file from the location on the local disk to view the contacts in
CSV format.
Click the Advance button.Go to Personal Directory column of the SLT to which the directory is to be
assigned.
Enter the number of the Personal Directory. For example, to assign Personal Directory No. 02 to SLT 2001
(software port 001, connected on hardware slot 03, hardware port 09), enter '02' in the 'Personal Directory'
column for SLT 2001.
1424
Click the Advance button. Go to Personal Directory column of the DKP to which the directory is to be
assigned.
Enter the number of the Personal Directory. For example, to assign Personal Directory No. 01 to DKP
3001 (software port 001, connected on hardware slot 17, hardware port 01), enter '01' in the 'Personal
Directory' column for DKP 3001.
Click the Advance button. Go to Personal Directory column of the ISDN Terminal to which the directory
is to be assigned.
For each ISDN Terminal that is to be assigned a Personal Directory, enter the number of the directory in
this column.
Click Submit at the bottom of the page to save changes.
Select the software port number of the SIP extension you want to assign the Personal Directory.
Scroll to Personal Directory, and select the Personal Directory number you want to assign to this
extension.
1425
If a Personal Directory is not assigned, for the extension user then the system sends a reply mail to the sender with
the subject: SMS Server Personal Directory Configuration and the message in the body: Personal Directory is not
assigned, can't add contact Name: XXXXX Number: XXXXX. Where name/number is as received in mail for
directory configuration.
If a match is found for the user and the user is assigned a Personal Directory, the system accepts and takes the
necessary action as requested by the user. The request may be for adding, viewing, deleting or editing a contact.
The following validations are applicable when you want to configure contacts via Email:
The Number can be a maximum of 16 digits.
The Name can be a maximum of 12 characters.
The Email ID can be a maximum of 64 characters.
The Group can be a maximum of 16 characters.
Adding a contact
You can add a single contact, multiple contacts or a group via Email. To add a contact you must send an Email to
the SMS Server in a specific format. Given below are the various examples and formats of adding a contact.
1426
Make sure you do not enter any space before and after "=".
The contact is added at a first free index, between 01 to 25 in the Personal Directory and a confirmation mail is sent
to the sender, with the Subject: SMS Server Personal Directory Configuration and the message in the body: New
entry is successfully added at index-XX, Name: XXXXX, Number: XXXXX. Where XXXXX=is the information
received in the mail for configuration. A separate mail is sent for each contact that is added.
To add a Group
To add a group HDFC Customer Care with three numbers, you must send a mail to the SMS Server in the following
format.
To: smssever@domain.com
Subject:HDFC Customer Care=+919898985400 (James), +919897894512 (John), +91898954045 (Steve)
Here,
The To field contains the Email ID of the SMS Server.
The Subject contains the details of the contacts.
The Group Name is HDFC Customer Care
The names to be added in this group are James, John, Steve and their numbers are 919898985400,
+919897894512, +91898954045 respectively.
The contact is added at a first free index, between 01 to 25 in the Personal Directory and a confirmation mail is sent
to the sender with the Subject: SMS Server Personal Directory Configuration and the message in the body: New
entry is successfully added at index-XX, Group: XXXXX, Name: XXXXX, Number: XXXXX. A separate mail is sent
for each contact that is added.
If the Directory is full and the system cannot add the contact, a reply mail is sent with the Subject: SMS Server
Personal Directory Configuration and the message in the body as Personal Directory is full.
1427
James D: 9426712345
James S: 9426921345
Similarly, if you want the details of all the names/groups beginning with J, then return mail will contain all names
starting with character "J" and each name will be displayed in a separate row, followed by their numbers.
James D: 9426712345
James S: 9426921345
John S :9996565123
If no match is found for the Name/Group Name, the system sends a reply mail with the Subject: SMS Server Error
Cause and the message in the body: X not found in Personal Directory, where X is the actual name.
Deleting a Contact
To delete an entry from the Personal Directory, the mail to be sent to the SMS Server must be in the format as
given below.
To: smssever@domain.com
Subject: HDFC Customer Care=
Here,
The To field contains the Email ID of the SMS Server.
The Subject contains the Name/Group Name that you want to delete.
In this case, there are three contacts +919898985400 (James), +919897894512 (John), +91898954045
(Steve) that will be deleted.
If the system is able to delete the entry, a reply mail with Subject: SMS Server Personal Directory Configuration and
the message in the body: X is deleted from Personal Directory', where X is the particular name which is received for
deletion of the contact/group name. In this case, the reply Email will be HDFC Customer Care is deleted from
Personal Directory.
If the SMS Server received a delete request for a contact and the same is not configured in the Personal Directory,
then a reply mail is sent to the sender, with the Subject: SMS Server Error Cause and the message in the body: X
not found in Personal Directory. Where, X is the actual name.
Modify/Edit a Contact
You cannot edit any contact in the Personal Directory via Email.
To modify any contacts details, you must first delete the existing contact from the Personal Directory. Then you can
add the same contact with new contact details.
Enter SE mode.
To program a telephone number in a personal directory, dial:
1428
1429
Exit SE mode.
1430
Each page of has 100 entries. To go to the next 100 entries click the links above the table '200-299'
Enter the Number you wish to store against a Memory Location Code. Enter the contact's Name
against the number.
The length of the Number field is limited to 16 digits. The length of the Name field is limited to 12
alphanumeric characters. All ASCII characters except < > and (double quote) are allowed. Ensure
that the number and the name are programmed within this limit.
Change the OGTB, if required. See OG Trunk Bundle Group.
1431
Select the Clear all other indices of the Global Directory, which are not specified in the .csv file
being uploaded check box, to overwrite the existing contacts in the Global directory with the contacts
of CSV file. Default: Disabled.
Click the Browse button to Select the .csv file to be uploaded from the location on the local disk.
All the contacts of the CSV file will be uploaded in the Global Directory. To view, click the Global
Directory link.
1432
You will get a prompt with an option to open the Opening Global_Directory.csv file or save the file to
a location. Save the file on the local disk.
Open the Global_Directory.csv file from the location on the local disk to view the contacts in CSV
format.
Make sure that the feature Global Directory is enabled in the CoS of the extensions to which you are
assigning the Global Directory.
By default, Global Directory Part 1 is allowed to all extensions in their CoS.
If the entire directory is to be assigned to all extensions, you may simply enable 'Global Directory Part 2
and Part 3 in the default CoS group 01 in the default Station Basic Feature Template 01 assigned to the
extensions.
However, if selected extensions are to be allowed Global Directory Part 2/Part 3, follow these steps:
1433
Refer the topics Class of Service (COS) and Station Basic Feature Template for further instructions.
Decide which of the DKP Extension users are to be allowed Global Directory Programming (of Global
Directory Part 1) and allow this feature in their Class of Service.
By default, Global Directory Programming is disabled in the default CoS group 01 in the default Station
Basic Feature Template 01 assigned to all extensions of ETERNITY. This means none of the extensions
can program Global Directory.
If you want to allow Global Directory Programming to all DKP extension users, simply enable this feature in
the CoS group of the Station Basic Feature Template assigned to them.
If you want to allow Global Directory Programming to only selected extensions, then follow these steps:
Define a CoS group with Global Directory Programming enabled.
Make sure this CoS also has Global Directory Part 1 enabled.
Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
Assign this template to the DKP extensions to which Global Directory Programming is to be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for programming
instructions.
If Global Directory Part 1, 2 or 3 is assigned to an extension user, ETERNITY will not check for Toll
Control.
When you assign Global Directory Programming to a DKP extension user, the user can program any
number in Global Directory Part 1, this includes numbers denied to the extension user in the Call
Privilege defined in the Toll Control level of this extension user.
Since the system does not check for Toll Control for numbers dialed out from Global Directory Part 1,
there is a possibility of extension users programming numbers not allowed to them in their Toll Control
level in the Global Directory Part 1, inadvertently or intentionally.
Hence, the System Engineer is advised to exercise caution when allowing this feature to DKP
extension users.
Click Submit at the bottom of the pages on which you make changes to save your settings.
If you have finished configuration, you may log out of Jeeves. Or you may continue, as required.
Enter SE mode.
To program a telephone number in the global directory, dial:
1801-Location Code-Number
Where,
Location Code is from 100 to 999.
Number is the telephone number, max. 16 digits. If the number has fewer than 16 digits, you must dial
#* to terminate the command.
To clear a telephone number from a location in the global directory, dial:
1801-Location Code-#*
1434
Exit SE mode.
Refer the topics Class of Service (COS) and Station Basic Feature Template for instructions on how to use
SE commands to
apply the template with the newly programmed CoS group to the extensions.
If you are using an SLT, you will not be able to program the Name of the contact in the directory.
Dial 1071.
Enter Personal Memory Index (001 to 025)
Enter Number of the contact (max. 16 digits).
Press 'Enter' key.
1435
Pick up handset.
Dial 1071.
Dial Personal Memory Index (001 to 025).
Dial Number of the contact (max. 16 digits).
Press #*.
Dial Trunk Access Code.
You get confirmation tone.
Replace handset
Extension users can only add, delete and edit names and numbers of contacts in Global Directory Part
1. However, they cannot program the Outgoing Trunk Bundle Group (OGTBG) for the contacts in the
directory.
When an extension user programs Global Directory Part 1, the system will automatically assign the
number and name to a free Memory Location. The system will use the OGTBG assigned to that
Memory Location by the System Engineer to dial out the number added by the extension user.
By default, OGTBG 01 is assigned to all Memory Location Codes in the Global Directory.
To simplify configuration for both the System Engineer and the extension users, the System Engineer
is recommended to assign the same OGTBG number uniformly to all Memory Location Codes, and
enable Least Cost Routing on this OGTBG.
If no OGTBG has been assigned to a Memory Location in the Global Directory (that is, the field is
blank), and an extension user adds a contact to this Memory Location, the number will not be dialed
out.
To program Global Directory Part 1 from the digital key phone, follow these steps:
1436
Adding a contact
To add a contact, select Add and press Enter key.
Enter your contacts name on the prompt: Name:
A maximum of 12 characters are allowed.
Press Enter key to save name.
Enter your contacts number on the prompt: Number:
A maximum of 16 digits are allowed.
Press Enter key to save number.
You will get the confirmation tone and the confirmatory message: Stored at Index xxx.
Editing a contact
To edit a contact,
Scroll to Contacts in the phone menu and press Enter key.
Select Edit and press Enter key.
You get the prompt: 'Name:'
Enter the initial letters of the contact's name.
The number of matching entries that will appear at a time on your phone's display will vary according to
your phone's LCD display capacity.
Scroll with the Up/Down navigation keys to reach the desired contact's name on the list.
Press 'Enter' key to select the name.
The system displays the name you selected.
To delete a character, use the Back/Forward navigation key to place the cursor under the character you
want to delete.
Press the Cancel key to delete the character you selected with the cursor.
To enter a character, use the Back/Forward navigation key to place the cursor in the position you want to
enter the character.
Enter the desired character by pressing the relevant digit pad keys in quick succession.
After you have finished editing the name/ number, press Enter key.
The number of the contact whose name you edited will be displayed.
Repeat the same steps as you did for editing the name.
After you have finished editing the number, press Enter key.
You will get the confirmation tone and the confirmatory message: Stored at Index xxx.
Deleting a contact
To delete a contact,
Scroll to Contacts in the phone menu and press Enter key.
Select Delete and press Enter key.
You get the prompt: 'Name'
Enter the initial letters of the contact's name.
The number of matching entries that will appear at a time on your phone's display will vary according to
your phone's LCD display capacity.
Scroll with the Up/Down navigation keys to reach the desired contact's name on the list.
Press 'Enter' key to delete the name.
You will get the confirmation tone and the confirmatory message: Deleted.
1437
How to use
Personal Abbreviated Dialing
For EON and Extended IP Phone Users
1438
OR
1439
Access Codes
What's this?
Access codes are short digit sequences dialed from an extension phone to instruct the PBX to perform a function
such as:
Calling an extension.
Grabbing a trunk line or any trunk line from a group of trunks (OG Trunk Bundle Group).
Station Codes: Codes used for calling extensions, Analog Input Port, Digital Output Port. These codes
are also commonly referred to extension numbers, phone numbers. For the purpose of this document,
station codes are referred to as Flexible Numbers.
Default station codes: the factory-set default values for SLT extensions are from 2001 to 2512; for DKP
extensions from 3001 to 3128; for Analog Output Ports from 3921, 3922.
Logical Group Codes: codes used for calling a group of stations as in a Department group, a group of
trunks as in Outgoing Trunk Bundle Group.
Default logical group codes: the factory-set codes for Department Numbers start from 3901, 3902.
Outgoing Trunk Bundle Groups from 61, 62, etc.
How it works
Whenever an access code is dialed from an extension, the system matches each digit in the code with the access
codes programmed within the system to determine the instruction, that is, whether it is an extension it must call, or
a trunk line it must grab, a port it has to activate, etc. The system processes the instruction when a match is found.
1440
For example:
When the first digit '1' is dialed, the system finds a match. As several default access codes begin with '1'
the system waits for the next digit to be dialed.
When the second digit '3' is dialed, the system finds a match for '13'.
As '13' is common for all Call Forward options219, the system waits for the next digit to be dialed
When the user dials the third digit '1', the system finds a match for '131'.
If there is more than one access codes matching with '131', e.g. '1311', '1314', '1315' the system will wait
for the next digit to be dialed.
If no further digit is dialed on expiry of the Inter Digit Wait Timer, the system understands the instruction as
'Call Forward - Unconditional' and waits for the destination phone number to be dialed.
Access Codes are related to various phases of a call. When a call is processed by a PBX, it goes through a
number of pre-defined phases.
Typically a call passes through the different phases as shown below:
Idle
Dial
Routing
Blocked
Placed
Matured
2-Way
Matured
3-Way
Denied
No
activity.
Digits are
pressed on
the phone
keypad/dialed
from the
rotary.
The system
is processing
the call. The
call is neither
placed nor
blocked.
The dialed
extension
is busy.
The dialed
extension is
ringing.
Connected
with the
dialed
extension.
Connected
with two
extensions.
No reply
from dialed
extension.
Dial tone is
played.
Beeps are
played.
Busy tone
is played.
Ring Back
Tone is
played.
Two-way
speech.
Three-way
speech.
Error Tone
is played.
Different access codes are dialed at different call phases. Station Codes and Logical Group Codes are dialed in the
'Dial' phase.
As different features are invoked in each call phase, Feature Access Codes are dialed at different call phases. For
example:
Auto Call Back code is dialed at the 'Blocked' phase' as well as 'Placed' phase.
219. Call forwarding options: Unconditional, When Busy, When No Reply, When Busy or No Reply.
1441
'Idle' phase is when no code is dialed. In the 'Denied Phase' no code is allowed to be dialed.
Each access code in a single call phase may be of different lengths, but must be unique. For example, the
same access code cannot be used for two different features like Call Forward and Redial, since both these
features are invoked in the 'Dial' phase.
However, the same access code can be used for features in different call phases. For example, '4' is the
default feature access code for DND Override (Routing Phase), Call Pick-Up-Group (Dial Phase) and BargeIn (Blocked Phase).
Similarly, Station and Logical Group Codes too must be unique and should not match with any of the features
invoked in the 'Dial' phase. Refer the topics Flexible Numbers and OG Trunk Bundle Group to know more.
How to configure
ETERNITY provides default Access Codes for stations, logical groups - department and trunk groups - and
features.
It also provides country-specific default Access Codes which are applied automatically when you select the
'Region' to configure the system.
The default Access Codes for India are presented in the table below. The default Access Code tables also indicate
the call phase in which each feature is invoked.
Call Phases
Feature
Number
Access
Code
Dial
Enter SE Programming
Mode
1#91
Enter SA Programming
Mode
1#92
12
102
Redial
17
1070
Personal Directory
Programming
10
1071
Abbreviated Dialing
11
Operator
12
Call Forward
13
13
Dynamic Lock
14
14
Hotline
15
15
Feature
1442
Routing
Blocked
Placed
Matured
2-way
Matured
3-way
Call Phases
Feature
Number
Access
Code
Dial
Alarm
16
161
Do Not Disturb
17
18
Interrupt Request
18
Barge-In
19
Raid
20
Trunk Reservation
21
Call Toggle
22
Conference
23
*3
24
Dial-In Conference
25
*19
Call Park
26
115
27
116
Room Monitor
28
1073
29
1092
Voice Help
30
1090
31
111
32
114
Paging
33
1074
DISA Login
34
1079
35
##
36
1051
37
69
Flashing on Trunk
38
User Absent/Present
39
104
40
1058
41
1059
Background Music
42
1099
Meet Me Paging
43
1093
Hot Desk
44
1091
45
Presence
46
1097
47
1094
Conversation Recording
48
1095
Forced Release
49
#*
Feature
Routing
Blocked
Placed
Matured
2-way
Matured
3-way
Y
Y
Y
Y
Y
1443
Call Phases
Feature
Feature
Number
Access
Code
Transfer
50
51
1098
Forced Answer
52
53
1054
54
1055
Minibar Details
55
1056
Mute
56
1052
Emergency Conference
57
1177
58
1057
Call Chaining
59
1050
SA Command Prefix
60
1072
61
Floor Service
62
Keypad Lock
Dial
Routing
Blocked
Placed
Matured
2-way
Y
Y
Y
38
63
CLI Restriction
64
103
65
1075
Reminder
66
162
67
163
68
164
69
1078
70
1076
71
1077
72
1096
73
1171
74
1172
Open a Door
75
1173
76
1174
77
Invoke RCOC
78
**
79
1075
80
*37
81
*38
1444
Matured
3-way
Y
Y
Y
Y
Call Phases
Feature
Number
Access
Code
Dial
82
1179
General Mailbox
83
1176
Intercom
84
*5
Terminate Conference
86
190
87
191
88
*13
61
62
63
64
Voice Mail
3931
Feature
Routing
Blocked
Placed
Matured
2-way
Matured
3-way
You can either use the default Access Codes or change them to suit your preferences.
1445
To disable an access code, delete the existing code and leave the field blank.
To change Station Access Codes (for SLT and DKP), Department Groups, Trunks and Trunk Groups, refer
the topics Flexible Numbers, Department Call, and OG Trunk Bundle Group.
1446
If you try to assign a number string that is already used to access an extension or use a feature then
the system will not accept the command and will play error tone.
To default access codes, dial:
3161-1-Feature Number to default access codes for a single feature.
3161-2-Feature Number-Feature Number to default access codes of a range of features.
3161-* to default access codes of all features
Where,
Feature Number is from 01 to 88.
E.g.: To default the prefix of SA command '107' to '1072' dial: 3111-1-60
A default trunk access codes table is given below:
OGTBG Index
61
62
63
64
1447
1448
Exit SE mode.
Account Codes
What's this?
Account Codes are very useful feature for organizations such as business consultants, law firms, advertising and
media agencies, and the like that cater to several clients, interacting with third parties on behalf of their clients.
Such organizations need to keep track of calls made to and on behalf of each client.
An 'Account Code' is a unique three-digit number that an organization can assign to each of its clients. Each
Account Code may be given a name and programmed in the Account Name List.
Doing so, whenever calls are made to the client or to a third party on behalf of the client,
The extension user dials the Account Code or Name assigned to the client.
Details of these calls are recorded by the Account Code dialed in the Station Message Detail Recording
Report (SMDR) for Outgoing Calls.
The SMDR report can be printed using the Account Code as filter.
This way, the organization can know the details of calls made to and on behalf of each client.
How it works
For example, an advertising media agency makes nearly 100 calls every day to and on behalf of its clients that
includes 'Midas Business Solutions', 'Jet-Set Holidays', 'Bacchus Vineyard'.
Assign a three-digit account code to Midas Business Solutions, for instance '001' and the name code
'Midas Biz' in the Account Name List.
Assign a three-digit account code to Jet-Set Holidays, for instance '002' and the name code 'Jet' in the
Account Name List.
A, a person from advertising media agency makes a call to Midas Business Solutions and talks to the
secretary B. During an ongoing conversation A dials the account code 001. In between, A needs to consult
the manager C. Therefore, A presses the Transfer key to put B on Consultation hold and dials the
number of C. Account code 001 will be applicable to the second call made to C also.
A, a person from advertising media agency makes a call to Midas Business Solutions and talks to the
secretary B. During an ongoing conversation A dials the account code 001. In between, A needs to talk to
another client, say Jet-Set Holidays D. Therefore, A pressed the Call Hold key to put B on Exclusive/
1449
Global hold and dials the number of D. While in speech with D, A dials account code 002. Here, Account
Code 001 will be applicable to Midas Business Solutions and Account Code 002 will be applicable to JetSet Holidays.
To use Account Codes, this feature must be included in the Class of Service (CoS) group allowed to
the extensions.
If you want to use Account Names, you must program the Account Name List.
When the Forced Account Code is enabled on an extension and trunk, the system will ask the user to
enter the account code irrespective of the method of dialing: Global Abbreviated dialing, Personal
abbreviated dialing, Least Cost Routing, or Selective Trunk Access.
However, if Forced Account Code is enabled on the selected trunk, and the number is dialed using
Selective Trunk Access, the system will dial out the number using Store and Forward dialing.
In the case of Abbreviated Dialing or Direct Dialing, if the extension user fails to dial the Account Code,
an error message will be displayed on the extension user's DKP.
How to configure
For Account Code to work, the you must:
1. Enable Account Codes feature in the Class of Service (CoS) of the extensions to which this feature is to be
allowed.
2. Prepare and program the Account Name List, if it is to be used.
3. If Forced Account Code is to be used, you must enable 'Forced Account Code' flag in
the Station Advanced Feature Template applied to the extensions from which calls using account
codes are to be made.
the Trunk Feature Template applied on the trunks through which calls using account codes are to be
made.
All the above feature parameters can be programmed using Jeeves or by dialing commands from a
Telephone.
1450
Write Account Codes on one column. Account codes may be any three-digit number between 001 and
999.
Write the Account Names, that is, names of the clients on the second column, against their respective
Account Codes.
The names must not exceed 12 characters. All ASCII characters except < > and (double quote) are
allowed. For example:
Account Code
001
Midas Biz
002
Jet Set
:
010
Bacchus
You need not follow a cardinal numbering sequence when assigning Account Codes.
You may assign any code to any client. For instance, you can assign code '111' to Midas Business
Solutions, '222' to Jet-Set Holidays, '333' to Bacchus Vineyard.
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Enter the Names of the clients against the account codes you have assigned to them. Refer the paper with
the two-column table you created.
Now, include the feature Account Codes in the Class of Service of the extensions.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all extensions of the
ETERNITY. Template 01 has the feature Account Codes in the default CoS Group (Number 01). So, all extensions
of ETERNITY can use this feature.
If Account Codes is to be allowed only to selected extensions, follow these steps:
1. Define a CoS group with 'Account Codes' enabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
3. Assign this new Template to the extensions to which Account Codes is to be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions.
If Forced Account Code is to be used, enable the 'Forced Account Code' flag in the Station Advanced Feature
Template of the extensions and in the Trunk Feature Template assigned to the trunks.
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When you enable 'Forced Account Code' in a template, this feature will be enabled on all DKP, SLT and
SIP extensions that are assigned this template. If necessary, create a separate template with this feature
and assign this template only to those extensions that are to be assigned this feature.
To enable Forced Account Code flag on Trunks,
Go to the parameters page of the type of trunk you want to program. For example: CO Parameters page
to program CO Trunk, Mobile Parameters page to program GSM Trunk, SIP Parameters page to
program SIP Trunks etc.
Click the Trunk Feature Template link in the column. The feature template page will open.
By default Trunk Feature Template 01 is assigned to all Trunk Types.
If you want to enable this feature on all trunks, enable it in Trunk Feature Template 01.
If you want to enable this feature only on select trunks, program a different Template number with this
feature.
Select the Forced Account Code check box in the template number assigned to the trunk.
Now change the Trunk Feature Template number of the trunk you want to program. This number should
be the same as the template in which you have enabled the Forced Account Code check box.
Enter SE mode.
To create the Account List:
Dial command 4851-1-Account Code-Account Name
Where,
Account Code is 001 to 999.
Account Name is a string of 12 characters.
Terminate Account Name with #*, if it is less than 12 characters.
For example: to program account code '001' with the name 'Midas Biz', dial
4851-1-001-MIDAS BIZ-#*
Press the digit key '1' twice to give space between characters.
To enable Forced Account Code flag in Station Advanced Feature Template:
Dial command 5602-1-Template Number-Feature Number-Flag Code
Where,
Template Number is Station Advanced Feature Template from 01 to 50. Default: 01
Feature Number for Forced Account Code is 09.
Flag Code is
0 for Disable
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1 for Enable
For example: To enable Forced Account Code Flag in Station Advanced Feature Template 02: Dial
5602-1-02-09-1
To apply the Station Advanced Feature Template now programmed with the Forced Account Code on
stations, refer the topic Customizing Station Advanced Feature Template using a Telephone.
To enable Forced Account Code flag on a Trunk:
Dial command 5802-1-Template Number-Feature Number-Flag Code
Where,
Template Number is from 01 to 50.
Feature Number for Forced Account Code flag is 29.
Flag Code is
0 for Disable
1 for Enable
For example: To enable Forced Account Code Flag in Trunk Feature Template 01: Dial 5802-1-01-29-1
To apply the Trunk Feature Template now programmed with the Forced Account Code on different
types of trunks, refer the topic Customizing Trunk Feature Template using a Telephone.
Exit SE mode.
How to use
Account Codes can be dialed in two ways: by Number and by Names.
Account codes, that is, number and names, can be dialed:
Print and hand out copies of the Account Code List to everyone in the organization for reference while
making calls.
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Dial 1058
Enter Account Code
Dial Trunk Access Code
Dial 1058
Enter Account Code
Speech will be resumed.
To enter Account Code Number when Forced Account Code Flag is enabled:
Dial 1058
Enter the Account Code Number.
You get dial tone.
Dial Trunk Access Code followed by the number of the client.
If you dial the Trunk Access Code to grab a trunk, without dialing the Forced Account Code, you will get an
error tone. Go ON-hook and then go OFF-hook. Now follow the steps in the sequence mentioned above.
Go OFF hook.
Dial Account Code first.
Dial Trunk Access Code followed by the number of the client.
If you dial the Trunk Access Code to grab a trunk, without dialing the Forced Account Code, you will get an
error tone. Go ON-hook and then go OFF-hook. Now follow the steps in the sequence mentioned above.
Dial 1059.
Enter the initial letter of the client's name.
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The Account Name List will be displayed on your DKP, alphabetically with the corresponding account
codes.
Scroll to select the desired client name and press Enter key.
Dial Trunk Access Code.
Dial the client's number.
Dial 1059.
Dial the initial letter of the client's name.
The Account Name List will be displayed on your phone, alphabetically with the corresponding account
codes.
Scroll to select the desired client name and press Enter key.
Speech will be resumed with the called party.
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AC Impedance Test
What's this?
ETERNITY supports the AC Impedance Test for clear, audible and echo-free speech over the CO Trunks. This test
helps you to set the most appropriate values for the CO Trunk parameters AC Impedance, CO Termination and
CO Line Type to correct the line impedance mismatch between the AC Termination Impedance presented by the
CO port of ETERNITY to the line and the CO Termination Impedance presented by the Central Office to the line.
a telephone with a valid number. You are recommended to use -a mobile phone with Mute function.
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In Enter Phone Number to which call should be made, enter the phone number on which you want
to make test call. The number can be a landline or a mobile number. We recommend you to use a
mobile number for the test call.
If you are using a mobile phone number, be sure the handset of the configured number supports the Mute
function.
In Make call using CO Port, select the CO trunk using which you want to make the test call. This must
be the same CO trunk for which AC Impedance is to be set.
Select the Test Mode. You may select Reliable (Recommended) or Accurate.
The Reliable Test mode suggests the AC Impedance settings on the basis of most commonly used AC
Impedances, CO Terminations and CO Line Types across the globe. The test using Reliable Test
mode takes approximately 5 minutes to complete.
The Accurate Test mode suggests the AC Impedance settings on the basis of all the possible AC
Impedances, CO Terminations and CO Line Types across the globe. The test using the Accurate Test
mode takes 1 hour and 20 minutes to complete.
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Click the Start Test button. The system calls the phone number you configured. The message Setting
up the Test.... appears on your screen.
While the test is being conducted, in Eternity ME and GE you will hear pulsating tone on all the ports of the
same card. In Eternity PE you shall hear this tone on all the ports of the system.
Answer the test call from your mobile phone. You will hear the Music-on-Hold as per the type of Answer
Supervision you have configured in the CO Hardware Template assigned to the CO trunk you are
testing.
By default, the Answer Supervision selected in the template is Pseudo Answer and the Pseudo Answer
Supervision Timer is set to 10 seconds. If you have not changed this default setting, you will hear
Music-on-Hold after 10 seconds of answering the call.
As the Music-on-Hold begins to play, Mute the microphone of your mobile phone.
If you are making the test call on a landline number, mute the call using the Mute key of the phone. If
your phone does not have a Mute key, unplug the handset cable from the phone body. This is to
prevent test signals from reflecting back into the mic of the handset.
After 5 seconds of Music-on-Hold, you will hear the test signals being transmitted by the system for the
duration of the test. The message Test running successfully... appears on your screen.
On completion of the test, the system will automatically disconnect the call. The message Test
completed appears on your screen.
However, if you wish to abort the test midway, you may click the Abort Test button.
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A new window opens. A list of ongoing calls will be displayed. Select a call on which you want to
perform the test.
Select the Test Mode. You may select Reliable (Recommended) or Accurate.
The Reliable Test mode suggests the AC Impedance settings on the basis of most commonly used AC
Impedances, CO Terminations and CO Line Types across the globe. The test using Reliable Test
mode takes approximately 5 minutes to complete.
The Accurate Test mode suggests the AC Impedance settings on the basis of all the possible AC
Impedances, CO Terminations and CO Line Types across the globe. The test using the Accurate Test
mode takes 1 hour and 20 minutes to complete.
Click the Start Test button. The message Setting up the Test.... appears on your screen.
While the test is being conducted, in Eternity ME and GE you will hear pulsating tone on all the ports of the
same card. In Eternity PE you shall hear this tone on all the ports of the system.
After 5 seconds the message Test running successfully... appears on your screen.
On completion of the test, the system will automatically disconnect the call. The message Test
completed appears on your screen.
However, if you wish to abort the test midway, you may click the Abort Test button.
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At the end of the test, the page displays the Suggested Impedance Settings for the AC Impedance,
CO Termination, CO Line Type and Return Loss.
You may now apply the suggested AC Impedance settings to the CO Trunk. To apply these settings,
select the desired CO Trunks and click on the Apply button.
Verify the settings by making a trial call from a DKP. There should be no echo and speech should be
audible and clear.
If you still hear echo during the trial call, you may re-run the test using the Accurate Test mode.
After you have determined the best matching AC Impedance, CO Termination, CO Line Type and
Return Loss by running the tests, apply the same suggested settings on the CO Trunk you are testing.
You may configure the same settings to all other CO Trunks, you have subscribed from the same CO
exchange.
It is possible that the CO trunks subscribed from the same exchange differ in their AC Impedance settings,
in such a case, you must run the test for each CO trunk separately and configure a different CO Hardware
Template for each of these trunks.
To generate the detailed test report, click the Generate Test Report button.
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The detailed test report appears in a new window. The Suggested Impedance Setting will appear in
bold.
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You may print the report by clicking the Print in the test report window.
You can also save the report in PDF format by selecting the PDF creator in your Printer options.
Repeat the above steps to conduct further tests as per your requirement.
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Alarms
What's this?
Alarms are an efficient and user-friendly feature available to all extensions of the ETERNITY.
Alarms can be set and canceled
Once Only - A one-time call, where the extension phone rings at the set time.
Daily - A repeat call, where the extension phone rings at the set time everyday.
Personalized - The Operator greets the extension user to serve the alarm request.
Automated - The system serves the alarm request by playing a voice message or music.
How it works
Personalized Alarm
When the Alarm serving mechanism is configured as 'Personalized',
The Operator phone rings first220, displaying the number of the extension to which the alarm is to be
served.
When the Operator answers this call, a call is placed on the extension on which the alarm is set.
The extension rings for the duration of the Alarm Ring Timer.
When the extension user answers the call, the Operator greets the extension user with the time and alarm
message.
This event is recorded in the Hotel-Motel Activity Log as 'Wake-up Alarm of <HH:MM> Answered on
<phone number>'.
If the extension user does not answer the call till the Alarm Ring Timer has elapsed, the Operator phone
will display a text message notifying 'No Reply' from the extension. The Alarm is now considered as
served.
220. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the ETERNITY will
make two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.
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This event is recorded in the Hotel-Motel Activity log as 'Wake-up Alarm of <HH:MM> No Reply on <Phone
Number>'.
If the extension is busy221, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
inform the extension user about the alarm in person or send someone to do it.
Automated Alarm
When the Alarm serving mechanism is configured as 'Automated',
The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is
from the EON series, an Alarm message will appear on its display.
When the extension user answers the call, s/he may be played music-on-hold, or a pre-recorded voice
message, or a music/message from an external source222, or be connected to a routing group, depending
upon the Alarm Notification Type programmed by the System Engineer.
The System Engineer may consult with the Enterprise to decide which of these options is to be
programmed as the Alarm Notification Type.
If the extension user does not answer the alarm call, the ETERNITY makes two more attempts (in all, 3
attempts) at an interval of 5 minutes between each attempt, to call the extension. (Each attempt is
recorded in the Hotel-Motel Activity log as 'Wake-up Alarm of <HH:MM> No Reply on <Phone Number>'.
If all Alarm attempts go unanswered, the ETERNITY places the call on the Operator phone. The Operator
phone rings till the end of the Alarm Ring Timer. The Operator phone displays the extension number with
the message 'No Reply'. The Alarm is now considered as served. (This event is recorded as "Alarm
Notification to Front Desk for <Phone Number>").
If the extension phone is busy ETERNITY will continue to make Alarm Attempts at the Alarm Interval
programmed. When all Alarm Attempts go unanswered, the ETERNITY will place a call on the Operator
phone. The Operator phone will display the number of the extension phone with the message 'Busy'.
The Snooze function can be added to Automated-Alarms to ensure that the extension user answers the
call. Snooze is a system-wide feature; when set, this function will be added to all Automated Alarms.
When Snooze is activated,
The extension phone rings for the Number of Alarm Attempts programmed, at set Alarm Attempt
Intervals.
221. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'. This event is recorded in the Hotel-Motel Activity Log as "Alarm not Served, <phone
number> is Busy".
222. This device can be connected to the Analog Input Port of ETERNITY.
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The extension stops ringing, when the extension user answers the call and dials the Code '0' to
acknowledge the Alarm. Please note that this Alarm Acknowledgement Code is non-programmable.
ETERNITY can register as many as 960 Alarm requests set by the Operator and extension users.
Multiple Alarms can be set for an extension by the Operator and/or by the extension user. For example,
Daily Alarm at 09:00am is set for an extension. The extension user wants to change the alarm time to
08:30am for a day. The extension user/Operator can set another alarm, that is, a Once Only Alarm, at
08:30am without disturbing the daily alarm. Both the Alarms will ring at the set time.
When multiple alarm requests have been set on an extension, if the Operator/extension user cancels
an alarm set for an extension, the system cancels all alarms set for the extension. It is not possible to
cancel any of these alarms selectively.
It is not possible to modify an alarm request. Instead, the alarm request should be canceled and a new
one should be made.
The duration of Alarm Ring Timer, the Number of Alarm Attempts and the Alarm Attempt Interval are
programmable.
Alarms can be set for all extensions of the ETERNITY, including the Operator phone also.
All the Alarm events are logged in the "Hotel-Motel Activity Log".
Alarm settings will be retained in the system during power down and system upgrades.
How to configure
The following parameters play an important role in the functioning of the Alarm feature. These parameters carry
default values. The default values have been selected keeping the larger user base in mind. However, these values
can be changed by the System Engineer at the time of installation or afterwards as per users' requirements.
1. Alarm Ring Timer - The duration for which the system rings the extension to serve an Alarm call. By
default, the Alarm Ring Timer is set to 45 seconds. This timer can be set between 001 to 255 seconds.
This timer also signifies the duration for which the Operator phone rings to notify that an Alarm call has not
been answered or the extension phone is busy.
2. Number of Alarm Attempts - Number of times the system attempts to place an Alarm call on the
extension phone before notifying the Operator that the call is not answered or the phone is busy. By
default, the Number of Alarm Attempts is set to '3'. The Number of Alarm Attempts can be set between 1
and 9.
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3. Alarm Attempt Interval - The time period between each Alarm Call attempt. By default, the Alarm Attempt
Interval is set to 5 minutes. The Alarm Attempt Interval can be set between 1 and 9.
4. Use Alarm with Snooze - Snooze is a functionality which forces the extension user to acknowledge the
Alarm call. With snooze enabled, the system expects the user to answer the Alarm call by going OFF-Hook
and dial Acknowledgement code '0'. With snooze disabled, the system considers the Alarm as answered
when the extension user simply answers the alarm call by going OFF-Hook (dialing acknowledgement
code is not mandatory). Users may choose whether or not to enable snooze. By default, snooze is
disabled.
5. Configurable Alarm Type - When the Operator and extension user set an Alarm call request, the system
gives them the choice of setting 'Once Only' or 'Daily' Alarm calls.
User experience however, shows that 'Once Only' Alarm call requests are more common than 'Daily' Alarm
requests. So, ETERNITY allows you the flexibility of setting 'Once Only' as the default Alarm Type, by
disabling the 'Configuring Alarm Type' flag.
When this flag is disabled the system will prompt the Operator/Extension user to enter the Time of the
Alarm call and consider the Alarm Type as 'Once Only'.
By default, this flag is disabled.
6. Configurable Alarm Category - When the Operator sets an Alarm call for an extension, the system
prompts the Operator to select an Alarm Type (Once Only or Daily) and to select the alarm serving
mechanism - 'Automated or Personalized'.
If the Enterprise wishes to offer only 'Automated' Alarms to its extension users, ETERNITY allows the
flexibility to set 'Automated' as the default Alarm call serving mechanism. This can be done by disabling
the 'Configurable Alarm Category' flag.
When this flag is disabled, the system will consider the Alarm call serving mechanism as 'Automated' and
will prompt the Operator only for the Time of the Alarm call.
By default, this flag is disabled.
When both flags 'Configurable Alarm Type' and 'Configurable Alarm Category' are disabled, the system
will set and serve 'Once Only - Automated' alarms only.
If the 'Configurable Alarm Type' flag is disabled, but the 'Configurable Alarm Category' flag is enabled,
the system will set 'Once Only' alarm calls, but give the option of selecting 'Automated' or
'Personalized' as the serving mechanism.
Similarly, if 'Configurable Alarm Type' is enabled, but the 'Configurable Alarm Category' flag is disabled,
the system will allow both 'Once Only' and 'Daily' alarms to be set, but the serving mechanism will be
'Automated'.
7. Voice Guided Alarm Verification: For Voice-guided Alarms, the VMS of ETERNITY allows you to
enable/disable the Alarm Verification for alarms and reminders, allowing extension users who want to use
alarms and reminders to confirm the Time set for an alarm and Date and time set as a reminder. By
default, this flag is enabled.
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The flags Configurable Alarm Type and Configurable Alarm Category are not applicable for Voice-guided
Alarms. In the case of Voice-guided Alarms, the Operator/Extension user will be prompted to select the
Alarm type and serving mechanism, each time, even when both aforementioned flags are disabled.
8. Alarm Notification Type - This is the means of notifying the extension user about the Alarm call. The
extension user can be played Music-On-Hold, Live Music, Pre-recorded Voice Message, Weather
information, Date and Time, etc. The ETERNITY supports four types of Alarm Notifications:
Voice Message: Selecting this option would play a message recorded in the Voice Module to the
extension user when s/he answers the Alarm call.
Music-On-Hold: Selecting this option would play music-on-hold to the extension user when s/he
answers the Alarm call.
External Music: Selecting this option would connect the extension user to live music when he answers
the Alarm call.
However, for this option to work the System Engineer should connect a live music source to the Analog
Input Port of ETERNITY. The live music source can be replaced by any other form of music. Please
refer the 'Technical Specifications' of Analog Input Port.
Routing Group: Selecting this option would connect the extension user to the stations programmed in
the Alarm Notification Group. The System Engineer may connect a device which can play customized
alarm greetings with date, time, weather conditions, traffic conditions, a marketing message, etc. on the
stations programmed in the Alarm Notification Group.
If Voice Mail Auto Attendant Profile is selected as a Routing Group member, the system will place the call
on the Voice Mail System.
Voice Mail: Selecting this option would connect the extension user to the Voice mail System. Use this
option only if you have VMS installed in the system.
9. Macros - This is a short code for simulating the Alarm call. The SLTs with special function keys send a
fixed string to the system, when each function key is pressed. The system interprets this string and
translates it into a string that can be understood by the system. For example, the SLT has a special
function key for Alarm calls which sends the string '51' to the system. The system can be programmed to
translate '51' into the feature access code for Alarm calls, '*161'.
All the above listed parameters can be programmed using Jeeves and a Telephone.
To select Alarm Notification Type for extensions, under Configuring Extensions, seeStation Advanced
Feature Template.
If you select Voice Message as Alarm Notification type, ensure that you assign a voice module to
'Alarm' voice message application. Please refer topic Voice Message Applications for more details.
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If you select the External Music as Alarm Notification Type, make sure you connect a music source to
the 'Analog Input Port' of the system. Please refer 'Technical Specifications' of the Analog Input Port for
more details.
If you select Voice Mail as the Alarm Notification Type, make sure you have installed the Voice Mail
System Card.
If you have selected Routing Group as Alarm Notification Type, you must create a Routing Group and
assign this Routing Group number in the Station Advanced Feature Template of the extensions. See
Routing Group and Station Advanced Feature Template for instructions on applying the template to
SLTs, DKPs, ISDN Terminals, Virtual Extensions.
To program SLTs with special Alarm function key and to create macro for a DKP key, see Macros.
To use a customized alarm messaging device, see Configuring Customized Alarm Messaging Devices
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Where,
Template Number is Station Advanced Feature Template from 01 to 50.
Default: 01.
Alarm Notification Type is:
1 for Music on Hold
2 for Voice Message (Voice Modules)
3 for External Music
4 for Routing Group
5 for Voice Mail
For example: To program Routing Group as Notification Type in Station Advanced Feature Template
02: Dial 5602-1-02 -12-4
To apply the Station Advanced Feature Template now programmed with the Alarm Notification Type to
extension phones, refer the topic Customizing Station Advanced Feature Template using a
Telephone.
To program Macros, dial the following commands:
Dial command 3115-1-Macro Index-Access Code to program Access code (that is, number sent to
the system by the SLT)
Where,
Macro Index is from 01 to 25 Access Code is a string of 4-digits.
If the length of the Access code is less than 4-digits terminate the command with #*
For example: To program access code '53' for the macro for Alarm dial: 3115-1-02-53-#*
Dial command 3115-1-Macro Index to clear the Access code for the macro.
Exit SE mode.
Connect the devices for customized alarm greetings to SLT ports only.
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Select the default Routing Group 31 used for Alarm Notification Group, or any other Routing Group
number.
Select SLT as the Member Type and enter the SLT Port Number where the device is connected. It is
possible to configure 32 members in a single routing group.
If only one device is connected, disable all other members from 02-32 by setting Member Type to None.
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Open the Station Advanced Feature Template page. By default Template Number 01 is applied to all
stations. The template has Voice Message as default notification type. It is recommended that you
program another Template.
Select Routing Group as the Alarm Notification Type in the template you have selected for configuration.
Enter the number of the Alarm Notification Routing Group (default group: 31) in which you have
programmed the device (SLT port).
Apply the Advanced Feature Template now configured with Routing Group as Alarm Notification Type and
the number of the Alarm Notification Routing Group to the stations.
Refer the section Station Advanced Feature Template for instructions on applying this template to
extension phones.
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You can cancel any of the unserved Alarm calls by selecting the check-box and clicking the Cancel
Selected Alarms button on this page.
You can also print this page by clicking the Print button on this page.
How to use
Alarms can be set by the extension users by themselves. The extension users can also ask the Operator to set the
alarm for them.
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To cancel Alarms,
Using Commands:
To set Alarm for the extension user,
To cancel Alarms,
Using Jeeves:
Log in as System Administrator.
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Search the extension on which you want to set the alarm by the Extension Number or the Extension
Name.
To cancel Alarms,
Dialing Commands:
To set Alarm,
To cancel Alarms,
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To cancel Alarms,
Extension users can set only automated alarms from their phones. For personalized alarms, they must
request the Operator.
If there are multiple alarms set, alarms cannot be canceled selectively. Only the Operator can cancel
alarms selectively from SA mode.
Alarms set on an extension will be served, even if DND is also set on the same extension.
Press the key assigned the 'Print Alarm Report' function (if programmed).
OR
Dial 1072-913.
You get a confirmatory text message and a confirmation tone.
Go idle.
Wakeup/Alarm Report
AS ON 10-10-2013(Thu) AT 23:49
------------------------------------------------------------------------------Room#
Phone# Wakeup D P
Room#
Phone# Wakeup D P
------------------------------------------------------------------------------3001
12:18
2017
14:12
3001
12:21
2017
12:14
------------------------------------------------------------------------------* indicates Daily Alarm and + indicates Personal Alarm
Page : 1
---End of Report---
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How it works
This feature works as an extension of the features Last Number Redial and Auto Redial. It requires you to
program Alternate Number Groups in the Global Directory first. With the alternate numbers programmed in the
Global Directory, all you need to do is to use Last Number Redial or Auto Redial, every time you want the system to
try Alternate Number Dialing.
For example: Midas Business Solutions has four telephone numbers: 2640459, 2631235, 2635589 and 2565590.
To be able to use Alternate Number Dialing, you must first program all four numbers as Alternate Number Group in
the Global Directory.
Now, when you dial one of these numbers, '2640459', and get a busy tone, you can either initiate Last Number
Redial or set an Auto Redial request.
When you initiate Last Number Redial,
The system will dial an alternative number for the dialed number.
If the redialed number is busy, you can set Last Number Redial again.
If the second alternative number is also busy, you can set Last Number Redial again.
This process will be repeated each time you set Last Number Redial, until the call gets through.
If the alternative number is busy, the system will redial another alternative number.
The system will dial a different (alternative) number on each auto redial attempt223, until the call gets
through.
223. The number of auto redial attempts depends on the Auto Redial Count programmed in the system. By default, the system will
make 5 redial attempts if Auto Redial 'normal' is set. If Auto Redial 'Priority' is set, the system will make 20 redial attempts.
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(for the number of redial attempts programmed), until the call gets through.
when any of the alternate numbers gets through, the system will give a ring on your extension.
(Busy)
( Bu
sy)
Calling Party
2630555
2630556
2630557
ETERNITY
Alternate Number Dialing will work only on extensions that are allowed the features Last Number
Redial in their Class of Service (COS)
Also, Alternate Number Dialing will work only for those numbers that exist in the Global Directory
assigned to each extension. The Global Directory is divided into three parts, 100-399 (Part 1), 400-699
(Part 2), and 700-999 (Part 3). If an extension is assigned only Global Directory Part 2, Alternate
Number Dialing will work only for those numbers grouped as Alternate Number Groups in Global
Directory Part 2.
Alternate Number Dialing will work also with Abbreviated Dialing. For example, an extension user
dials the abbreviated code 8100, and the dialed out number is busy. When the extension user sets
Redial or Auto Redial, the ETERNITY will try the alternate numbers related to 8100.
How to configure
For Alternate Number Dialing to work, the System Engineer must:
1. Make a List of Alternate Numbers.
2. Create Alternate Number Groups.
3. Program Alternate Number Groups in the Global Directory.
4. Enable the features 'Last Number Redial', 'Global Directory', in the Class of Service (CoS) group of the
extensions to which Alternate Number Dialing facility is to be provided. If desired, 'Auto Redial', 'Auto
Redial Priority' may also be enabled in the CoS of these extensions.
All of the above parameters can be programmed using Jeeves or dialing SE Commands from a telephone.
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To create Alternate Number Groups, the alternate numbers must exist in the Global Directory. If any of
the alternate numbers do not exist in the Global Directory, first program the numbers in the directory,
before you begin creating Alternate Number Groups. Refer the topic Abbreviated Dialing for
instructions on programming the Global Directory.
As Alternate Number Dialing works only for the Alternate Number Groups in the Global Directory
assigned to each extension, ensure that the relevant Global Directory with the Alternate Number
Groups is allowed in the CoS of the extensions.
Write the name of the contact on one column and the Alternate Numbers for the contact on the other
column.
Make a list of the numbers which need to be grouped as alternate numbers. For example:
Name of the Contact
Alternate Numbers
022281110001, 022281110002
Bacchus Vineyard
2640075, 2640076
GoodLife Inn
2788856, 2788896
Taking the above example further, the Alternate Number Groups on the list may be numbered as follows:
Name of the Contact
Alternate Numbers
001
022281110001, 022281110002
002
Bacchus Vineyard
2640075, 2640076
003
GoodLife Inn
2788856, 2788896
004
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Enter the number of the Alternate Number Group in the last column of the page.
For example, you have assigned Alternate Number Group '001' to all the numbers of the contact Midas
Business Solutions, enter this number against each number belonging to this contact.
Similarly, enter Alternate Group number '004' against the numbers belonging to the 'GoodLife Inn' to which
it is assigned.
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Memory
Location
Outgoing Trunk
Bundle Group
Number
Name
Alternate
Number Group
100
01
2640459
Midas Biz
001
101
01
2631235
Midas Biz
001
102
01
2635589
Midas Biz
001
103
01
2565590
Midas Biz
001
104
01
2788856
GoodLife Inn
004
105
01
022281110001
Jet Set
002
Memory
Location
Outgoing Trunk
Bundle Group
Number
Name
Alternate
Number Group
106
01
022281110002
Jet Set
002
107
01
033298765432
R. Mendez
000
108
01
2640075
Bacchus
003
109
01
2640076
Bacchus
003
129
01
:
2788896
GoodLife Inn
004
The numbers of the contacts may not necessarily appear alphabetically or in a sequence. It is possible that
the numbers of the same contact may be programmed at different memory locations in the Global
Directory.
In the above example, one number of the GoodLife Inn is programmed at memory location Index 104 and
the other on Index 129. Since these two numbers are grouped and assigned the number alternate group
number '004', this number must be entered against the GoodLife Inn numbers at the respective memory
location Index.
After assigning Alternate Number Groups, click Submit at the bottom of the page to save changes.
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Enable the features Last Number Redial and Global Directory, in the Class of Service (CoS) group of
the extensions to which Alternate Number Dialing facility is to be provided. If desired, Auto Redial, Auto
Redial Priority may also be enabled in the CoS of these extensions.
By default Station Basic Feature Template Number 01 is assigned to all extensions of ETERNITY. The
default CoS Group 01 in this template has 'Redial' enabled in the set of 'Basic Features', so all extensions
of ETERNITY can use Last Number Redial.
However, the default CoS Group 01 has only Global Directory Part 1 enabled.
Recall that Alternate Number Dialing will work only for those numbers that exist in the Global Directory
assigned to each extension. So, the Global Directory Part containing the Alternate Number Groups must
be allowed to the extensions in their Class of Service. For example, if Alternate Number Groups are
programmed in Global Directory Part 2, extensions must have Global Directory Part 2 in their Class of
Service.
If all extensions are to be allowed the Alternate Number Dialing facility, simply enable the Global
Directories containing Alternate Number groups in the default CoS group 01.
However, if Alternate Number Dialing is to be allowed to select extensions only, define a new CoS group
and prepare a new Station Basic Feature Template with this CoS group and apply it to the desired
extensions.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions.
The Station Basic Feature Template 01 does not have the features Auto Redial and Auto Redial Priority in
the default CoS group 01. If these features are also to be allowed to the extensions, enable them in the
CoS you prepare.
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If you have a continuous sequence of numbers that need to be programmed in the same group, you may
dial:
1804-2-Memory Location Code-Memory Location Code-Alternate Group Number
Here you enter a sequence of Memory Location Codes, from 100 to 999, and the number of the
Alternate Number Group.
For example: To assign the numbers of Midas Biz to an Alternate Number Group, dial: 1804-2-100103-001
The numbers stored at Index 100 to 103 will be assigned to Alternate Number Group 001, which is the
number assigned to Midas Biz.
To clear the Alternate Number Groups, dial the following commands:
1804-1-Memory Location Code-000 to clear group of a single number.
1804-2-Memory Location Code-Memory Location Code-000 to clear group of a continuous
sequence of numbers.
1804-*-000 to clear all Alternate Number Groups
For example: To clear the Alternate Number Group assigned to a number of GoodLife Inn, dial: 1804-1104-000
The Alternate Number Group assigned to the GoodLife Inn number '2788856' stored at Index 104 will
be cleared.
For example: To clear the Alternate Number Group assigned to the sequence of numbers of Midas Biz,
dial: 1804-2-100-103-000
The Alternate Number Group assigned to the Midas Biz numbers stored in a continuous sequence
starting from Index 100 to 103 will be cleared.
Exit SE mode.
How to use
Confirm with your System Engineer that
Alternate Number Groups are programmed in the Global Directory allowed to your extension.
'Basic Features' (these include Redial) are enabled in the Class of Service allowed to your extension.
Now, follow the instructions for using the feature Last Number Redial.
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Authority Codes
What's this?
Authority Code is a unique password-protected code with an associated Class of Service, Toll Control and Call
Budget, which can be assigned to extension users. With Authority Codes, extension users of ETERNITY can make
calls or access features from any other extension of the system as per the Class of Service and Toll Control
assigned to their code.
This feature is useful when you want a group to extension users to use a single extension, but at the same time you
want to keep an account of the calls made by each user. If required, you can assign a call budget to each Authority
code to control call cost.
To make outgoing calls or access features, extension users must 'Walk-In' from any extension port: DKP, SLT, SIP,
ISDN Terminal, E&M (with Station as Orientation Type), and then dial their Authority Code and Password.
How it works
An 'Authority Code' is a unique three-digit number, protected by a four-digit password. The default Authority
Password is 1111. To be able to use an Authority Code, the password must be changed to another value. The
ETERNITY supports as many as 999 Authority Codes.
Each Authority Code has an associated Class of Service and Toll Control, which is configured in the Station Basic
Feature Template assigned to the Code.
To make calls using an Authority Code,
User A is assigned 222 as Authority Code and is provided a unique 4-digit Authority Password.
To access features or make calls as per the CoS and Toll Control assigned to Authority Code 222, User A
must do the following:
Dial the feature code for Walk-In Class of Service (default 111) from any extension of ETERNITY.
Dial the code for Walk-In by Authority followed by the Authority Code and then the Authority Password.
To make calls or access features according to the Authority Code dial 111, the feature code for 'Walk-In
Class of Service' from any extension of ETERNITY.
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If the Walk Out mode set for A is One CallA will automatically be logged out from the extension after one
call.
If the Walk Out mode set for A is Multiple Calls, A can make as many calls as desired, and remains
'walked-in' until the A dials the feature code to 'Walk-Out', or until another extension user walks into the
same extension.
If a call budget has been assigned to As Authority Code, A will be able to make calls till the assigned
amount is consumed.
Details of the calls made by A are recorded by the Authority Code in the Station Message Detail Recording
Report (SMDR) for Outgoing Calls.
The SMDR report can be printed using the Authority Code as filter.
This way, the organization can know the details of calls made by each user as well as have control over the
expenses.
You can also use Authority Codes from a remote location using DISA. To do so, you must make a DISA call. Then
you must Walk In and dial your Authority Code and Password. To know more, see Direct Inward System Access
(DISA).
How to configure
For Authority Codes to work, you must:
Configure the Authority Code Table.
Assign a Station Basic Feature Template to the Authority Code, with the desired Class of Service and Toll
Control.
Enable the parameter Store Outgoing Calls in the Station Basic Feature Template assigned to the
Authority Code, if you want details of outgoing calls made using Authority Codes.
Assign a Station Advanced Feature Template to the Authority Code with the desired Walk-Out Mode
Assign each user a Call Budget, if required.
Change the Retry Counts for the Authority Code Password, if required.
Make a list of users you want to assign Authority Codes to and their respective passwords.
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Assign an Authority Password. The password can be a minimum of 4 digits and a maximum of upto 10
digits. Valid digits are 0 to 9, * and #. Default: 1111.
Enter the Name of the user. The name acts as an identifier. Default: Blank.
By default, Station Basic Feature Template number 41 and Station Advanced Feature Template
number 41 are assigned to all Authority codes.
If you want to change the calling permission, allow/deny features to users, you must customize the Class
of Service and Toll Control in the Station Basic Feature Template according to your requirement.
You may set the 'Walk-Out Mode in as One Call or Multiple Calls in the Advanced Feature Template.
Station Advanced Feature Template. Refer to Station Basic Feature Template and Station Advanced
Feature Template for detailed instructions.
You can assign/change the Name and Password from the SA mode also. To do this,
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Assign/change Authority Password. The password can be a minimum of 4 digits and a maximum of upto
10 digits. Valid digits are 0 to 9, * and #. Default: 1111.
Enter the Name of the user against the authority code you have assigned to them. The name acts as an
identifier. Default: Blank.
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In the Allot an Amount column, enter the amount you want to assign to the user as budget limit for
outgoing calls. The amount you allot here will be displayed as Allotted Amount.
If you are re-assigning a new amount before the previous balance is consumed, make sure you add the
available balance to the new amount. Enter this amount in Allot an Amount.
For example, you have allotted an amount of Rs.1000 and the consumed amount is Rs.600. The available
balance is Rs.400. Now, if you want to assign a new amount of Rs.500. In Allot an Amount you must enter
900 (Available Balance + New = 400 + 500).
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The Allotted Amount column displays the amount allotted to the user for making outgoing calls.
The Consumed Amount column displays the call budget amount consumed by the user.
How to use
For EON Users and Extended IP Phone Users
To use Authority Code from any extension:
Go OFF-Hook.
Press DSS Key assigned to Walk-in COS.
OR
Dial 111
Select the Walk-in by Authority Code option and press the Enter key.
Dial the Authority Code, followed by the Password.
You will hear the confirmation tone, followed by dial tone.
Dial the desired number.
Go OFF-Hook.
Dial 111
Press 2 to select Walk-in by Authority Code
Dial the Authority Code, followed by the Password.
You will hear the Confirmation tone, followed by dial tone.
Dial the desired number.
enable Store Outgoing Calls in the Station Basic Feature Template of the extension user.
set the Calls made using Authority Code filter in Outgoing Call Print Filters
configure the destination port for SMDR-Outgoing Call Report.
Refer the section Station Message Detail Recording-Report, for detailed instructions on printing reports using
filters.
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Auto Attendant
Whats this?
Auto Attendant allows external callers to reach an extension directly without the intervention of the Operator.
If Auto Attendant is enabled on a trunk, whenever an external call lands on that trunk, the Built-In Auto Attendant or
the Voice Mail Auto Attendant of ETERNITY (if Voice Mail System card is installed) greets the caller and prompts
the caller to dial the desired extension number. The call is then placed to the extension number dialed by the caller.
ETERNITY offers Delayed Auto Attendant, whereby incoming calls routed to the Operator or the Trunk Landing
Group, can be answered by the Built-In Auto Attendant or the Voice Mail Auto Attendant, if none of the landing
extensions answers the call within a certain time period.
Regular callers who know the extension numbers of ETERNITY, can use the Auto Attendant to reach the desired
extensions without Operator assistance. Thus, this reduces call traffic on the Operator extension, saves callers the
time for call set-up and transfer. The Auto Attendant is particularly useful during non-working hours and holidays,
and it helps project a professional image of the organization.
Built-In Auto Attendant will not work, when the dialed extension has Privacy from Built-In Auto
Attendant enabled in its Class of Service. So, if you want to prevent external callers from accessing
certain extensions, you must enable Privacy from Built-In Auto Attendant in their Class of Service. To
know more, see Privacy.
How it works
Auto Attendant can be configured on all trunk types, for the three time zones (working hours, break hours and nonworking hours).
When configuring Auto Attendant on a trunk, you may choose to have calls answered by the Built-In Auto Attendant
or the Voice Mail Auto Attendant.
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The system waits for the period of the Built-In Auto Attendant Answer Wait Timer (default: 05 seconds) to
answer the call during this period. The caller gets Ring Back Tone from the CO.
The system greets the caller with the pre-recorded Time-based Greetings. These are: Built-In Auto
Attendant-Morning Greetings/Afternoon Greeting/ Evening Greeting. This is then followed by the Built-In
Auto Attendant-Welcome Greeting for the current time zone (working hours, break hours, non-working
hours). A Voice Module must be assigned for the Built-In Auto Attendant Time-based as well as the
Welcome Greeting.
The Built-In Auto Attendant Time-based and Welcome Greeting messages are played once.
If no voice module is assigned as Welcome Greeting, the system will play music-on-hold after answering
the call. It will play music-on-hold until the end of the Built-In Auto Attendant Music Timer (default: 5
seconds).
On the completion of the Welcome Greeting or music-on-hold at the end of the Built-In Auto Attendant
Music Timer, the system plays the Built-In Auto Attendant Dial Message to prompt the caller to dial the
desired extension number.
The Built-In Auto Attendant Dial Message is played once and the caller gets Beeps. The system waits for
the Built-In Auto Attendant Beeps Timer (default: 10 seconds) to expire.
If the caller does not dial any number before the Built-In Auto Attendant Beeps Timer expires, the system
plays the Built-In Auto Attendant Call Transfer to Operator message and transfers the call to the Operator.
The system waits for the duration of the Built-In Auto Attendant Inactivity Timer (default: 60 seconds) for
the Operator to answer the call. If there is no answer at the end of this timer, the system releases the trunk.
If the caller fails to dial digits, you can have the call disconnected instead of having it routed to the
Operator. For this, you need to enable the Disconnect Built-In Auto Attendant call, when caller does
not dial any digit flag in the System Parameters. When this flag is enabled, the system will play the
Built-In Auto Attendant No Dial Voice message to the caller. If the caller fails to dial a digit within the BuiltIn Auto Attendant Beeps Timer, the system will disconnect the call.
If the caller dials the extension number, the system checks if the number is valid.
If the dialed digits are invalid, the system plays the Wrong Dial voice message to the caller. This message
is played once. The system waits for the duration for the Built-In Auto Attendant Error Tone Timer (default:
5 seconds).
If the Wrong Dial Voice Message is not programmed, the system plays Error Tone to the caller for the
duration of the Built-In Auto Attendant Error Tone Timer, followed by the Built-In Auto Attendant Dial
Prompt.
If the number dialed by the caller is valid, the system checks if the dialed extension is free.
If the dialed extension is busy, the system plays the Built-In Auto Attendant Busy Message to the caller.
The message is played once.
If no Built-In Auto Attendant Busy Message is programmed, the caller will hear Busy Tone. The Busy Tone
is played for duration of the Built-In Auto Attendant Busy Tone Timer (default: 15 seconds), followed by the
Built-In Auto Attendant Dial Prompt.
To have the call disconnected if the dialed extension is busy, you may enable the Disconnect Built-In
Auto Attendant Call, when dialed number is busy flag in the System Parameters.
The dialed extension is free. The system calls the extension and plays Built-In Auto Attendant Ring Back
Tone Message (if programmed) or Ring Back Tone to the caller. This message is played until the dialed
extension is ringing.
The system waits for the period of the Built-In Auto Attendant Ring Timer for the dialed extension to
answer the call.
When the dialed extension answers the call, the caller gets connected to the extension.
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If the dialed extension does not answer before the expiry of the Built-In Auto Attendant Ring Timer, the
system prompts the caller to dial again with the Built-In Auto Attendant Dial Prompt message to the caller.
The system diverts the call to the Operator. When the call is transferred to the Operator, the system plays
the Built-In Auto Attendant Call Transfer to Operator voice message (if programmed) or plays Ring Back
Tone to the caller.
If there is no reply from the dialed extension, you can have the call disconnected instead of having it routed
it to the Operator by enabling the Disconnect Built-In Auto Attendant call, when dialed number is not
responding flag in the System Parameters.
The Voice Mail System (VMS) installed in the ETERNITY answers the call.
The VMS greets the caller with the Welcome message and the Greeting Message selected for the current
time zone (working hours, break hours and non-working hours).
If the system detects the day as a holiday, the VMS plays the Holiday Message. To know more, see
Holiday Table.
The VMS plays prompts to the caller to process the call further.
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as a call lands on a trunk, the system checks the incoming call routing configured for the current time zone
for the trunk.
on finding Delayed Auto Attendant enabled, the system rings on the destination extensions (Operator and
Trunk Landing Group) for the duration of time defined for ringing the extensions (default: 10 seconds).
if no reply is received from the extensions, the system routes the call to the auto attendant you selected,
which may by the Built-In Auto Attendant or the Voice Mail Auto Attendant.
How to configure
To use the Built-In Auto Attendant on trunks, do the following:
1. Make a list of the trunks by their port type (CO, Mobile, SIP, T1E1PRI, BRI) and port number on which you
messages as Built-In Auto Attendant greetings and to play voice prompts at each stage of the call, you
need to assign Voice Modules for the following Built-In Auto Attendant Messages:
Built-In Auto Attendant-Welcome Greeting: Played to callers when answering the call. Different
welcome greetings can be programmed for Working Hours, Break Hours and Non-working Hours. The
Built-In Auto Attendant Welcome Greeting message is played once.
Built-In Auto Attendant-Dial Prompt: Played after the Welcome greeting message to prompt the
caller to dial the desired extension number. This message is played once.
Built-In Auto Attendant-Ring Back Tone: Played after the caller has dialed the number and the
system is ringing the dialed extension. This message is played continuously as the dialed extension
rings.
Built-In Auto Attendant-Wrong Dial message: Played when the caller dials a wrong number or the
number dialed by the caller does not match with any extension number of ETERNITY. This message is
played once.
Built-In Auto Attendant-Destination Busy: Played when the dialed extension is busy. This message
is played once.
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Built-In Auto Attendant-Destination No Reply: Played when the dialed extension does not respond.
This message is played once.
Built-In Auto Attendant-No Dial: Played when the caller has not dialed any number. This message is
played once.
Built-In Auto Attendant-Call Transfer to Operator: Played to the caller when the call is being
transferred to the Operator. This message is played once.
Pre-recorded Built-In Auto Attendant voice messages are provided in .WAV file format on the CD-ROM
provided to you with the ETERNITY.
The default Voice Module numbers assigned to Built-In Auto Attendant messages and the messages
recorded on each module are:
Voice
Module
Number
Voice Message
02
Good Morning!
03
Good Afternoon!
Greeting
04
Good Evening!
05
Welcome!
06
08
09
message
10
message
11
Ringing message
(Ring Back Tone)
12
Reply message
13
You may customize these Built-In Auto Attendant voice messages by recording messages of your
choice and assigning them to the voice modules. For instructions on recording messages on the voice
modules and assigning voice modules to different functions, see Voice Message Applications.
If you do not use any of the above voice modules, the system will play the Call Progress Tone for each call
state.
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enable Auto Attendant by selecting Voice Mail Auto Attendant for the desired time zones.
Configure Welcome and Greeting messages. You may either use the default, pre-recorded welcome
messages of the VMS, or record the custom welcome messages that meet your requirements, in .WAV
file format.
For more information and instructions, see the Configuring Voice Mail System.
To use Delayed Auto Attendant on trunks, do the following:
1. Make a list of the trunks by their port type and port number on which you want to enable Delayed Auto
Attendant.
2. In the Trunk Feature Template assigned to these trunks,
if you want to use the VMS Auto Attendant for Delayed Auto Attendant, select Voice Mail Auto
Attendant and complete the voice mail related configuration. For more information and instructions,
see the Configuring Voice Mail System.
if you want to use the Built-In Auto Attendant for Delayed Auto Attendant, select Built-In Auto
Attendant, and assign the Voice Modules for Built-In Auto Attendant Messages, as described
earlier.
If required, you may also change the default values of the following Built-In Auto Attendant related Timers and set
them to the desired values.
To know more about these timers and for configuration instructions, see System Timers and Counts.
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You may also configure the following Built-In Auto Attendant related flags, as required:
Disconnect Built-In Auto Attendant call, when dialed number is busy: When this flag is enabled, if
the dialed extension is found busy, the system will disconnect the call instead of routing it to the
Operator. Default: disabled.
Disconnect Built-In Auto Attendant call, when dialed number is not responding: When this flag is
enabled, if there is no reply from the landing destination extensions, the system will disconnect the call
instead of routing it to the Operator. Default: disabled.
Disconnect Built-In Auto Attendant call, when caller does not dial any digit: When this flag is
enabled, if the caller fails to dial a digit within the Built-In Auto Attendant Beeps Timer, the system will
disconnect the call instead of routing it to the Operator. Default: disabled.
These flags may be used in Hotels that provide 'Limited Services' and do not want to receive unanswered/busy
calls on the guest phones. For instructions on enabling or disabling these flags, see System Parameters.
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Auto Answer
What's this?
Auto Answer allows incoming calls to be answered without any manual interventions by the extension users.
This feature is particularly useful for Operators in high call traffic settings, as it saves them the effort of picking up
the handset or pressing the speaker key repeatedly.
This feature works on digital key phones (DKP) as well as in Extended IP Phones.
How it works
With Auto Answer set on an extension DKP/Extended IP Phone, whenever a call lands on the DKP/Extended IP
Phone extension,
the extension rings for the duration of the Auto Answer Timer224. This timer is programmable, and by
default it is set to 1 second.
on the expiry of the Auto Answer Timer the system plays a beep to the user.
the DKP goes OFF-Hook to answer the call, without any intervention by the extension user such as picking
up the handset or pressing the speaker or the headset key.
If a headset is connected, and headset connectivity is enabled on the DKP, the incoming speech audio will
be diverted to the headset automatically.
Auto Answer works only if the DKP is in idle state; the phone must not be busy with an active call or using a feature.
How to configure
For Auto Answer to work, you are required to do the following:
1. Enable Auto Answer in the DKP Parameters.
2. Change Auto Answer Timer, if required. The range of this timer is 1 to 9 seconds. By default, the Auto
All of the above can be programmed by the System Engineer using Jeeves and a Telephone.
The DKP extension users can also program the above parameters using the Phone Menu of EON. See "How
to use" Auto Answer later in this topic.
224. This timer defines the time in seconds that the DKP should wait before going OFF-Hook to answer incoming calls.
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1500
If the DKPs are already installed and configured, identify the DKP you wish to provide Auto Answer feature
by its Hardware Port/Slot Number, Access Code or Name.
Go to the column Auto Answer of the DKP parameters using the horizontal scroll bar. Enable the flag by
selecting the check box.
To cancel Auto Answer, disable the flag by selecting the check box.
Now, go to the column Auto Answer Timer, and set the Timer as required. By default the Timer is set to 1
second.
If Headset is to be used by the DKP, go to the column Headset Connected? and click the check box to
enable the flag.
Repeat the above steps to program Auto Answer parameters for each DKP that is to be provided this
feature.
Enter SE mode.
To set auto answer on DKP, dial:
1214-1-DKP-Auto Call Answer Mode to set Auto Answer on a single DKP.
1214-2-DKP-DKP-Auto Call Answer Mode to set Auto Answer on a range of DKPs.
1214-*-Auto Call Answer Mode to set Auto Answer on all DKPs.
Where,
DKP is the Software port number of the DKP, from 001 to 128.
Auto Call Answer Mode is
0 for Manual mode
1 for Auto Answer mode.
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Exit SE mode.
How to use
Extensionuserscanset/cancelAutoAnswerandenableHeadsetconnectivityfromtheirDKP
/ by navigating the Menu of EON.
To set Auto Answer:
225. This function must have been programmed by the System Engineer on a DSS Key of EON. Refer "Digital Key Phone - Keys Programming" for instructions.
1502
226. This function must have been programmed by the System Engineer on a DSS Key of EON. Refer "Digital Key Phone - Keys Programming" for instructions.
1503
How it works
When you set Auto Call Back,
As soon as both extensions, yours and the remote extension, are available, the system will ring first on
your extension for the duration of the Auto Call Back Ring Timer. This timer is set by default to 30 seconds
and is programmable.
When you go OFF-Hook, the system will ring on the remote extension (provided it is also available at that
moment) for the duration of the Auto Call Back Ring Timer.
When the remote extension user goes OFF-Hook, your call will get connected.
However, if the remote extension gets busy before the system can ring on it, the system will continue to try
again.
Auto Call Back set for a busy trunk works the same way. As soon as the busy trunk port you are trying to
access is available, the system will ring your extension. When you go OFF-Hook you will be connected to
the trunk port.
Each extension of the ETERNITY can set only one Auto Call Back request at a time. If you set another
Auto Call Back request, before the first one has been served, the system will override the first request
and serve the second.
The ETERNITY has the capacity to serve 300 Auto Call Back requests from its extensions at a time.
The service duration for each request is 60 minutes. Requests that are not served within 60 minutes
are automatically cancelled by the system. Also, the system will not serve any more requests if all the
300 requests are pending. In such a case, the system will play an error tone, when an extension
attempts to make a request.
Auto Call Back request set by you will be cleared by the system if:
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it was successfully served, that is, your extension was connected to the remote extension or the trunk you
were trying to reach.
you do not answer the Auto Call Back ring, before the expiry of the Ring Timer, that is, within 30 seconds
(default setting).
the remote extension does not answer the Auto Call Back ring before the expiry of the Ring Timer.
Auto Call Back works for internal calls and for accessing trunk ports only.
Internal calls include calls between PBXs that are networked using Q-SIG.
How to configure
Auto Call Back is a Class-of-Service dependant feature. An extension user can set/cancel Auto Call Back only if it
is enabled in the extension's Class of Service.
The only programming involved in this feature is enabling/disabling Auto Call Back in the Class of Service and
changing the duration of the Auto Call Back Ring Timer, if required.
Both these can be programmed using Jeeves and a Telephone.
Refer the topics Class of Service (COS) and Station Basic Feature Template for instructions on how to
enable/disable a feature in a CoS group, how to prepare a Station Basic Feature Template with a new CoS
group and assign the new template to SLT, DKP and ISDN Terminal extensions using Jeeves.
If the User wants to increase or decrease the duration of the of the Auto Call Back ring on both extensions,
that is, the extension requesting Auto Call Back and the destination extension, program the 'Auto Call Back
Ring Timer', according to User preference.
Under Configuration, click System Timers and Counts to open the page.
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Set the Auto Call Back Timer (sec) to the desired duration.
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For example: To enable Auto Call Back when Busy in CoS group 02, dial 1302-1-02-04-1
To enable Auto Call Back when No Reply in CoS group 02, dial 1302-1-02-05-1
To assign the CoS group with Auto Call Back Busy/No Reply to a Station Basic Feature Template, dial:
5502-1-Template Number-Feature Number-Code
Where,
Template Number is from 01 to 50.
Feature Number is
03 for CoS group for Working Hours
04 for CoS group for Break Hours
05 for CoS group for Non-Working Hours
Code is CoS group number from 01 to 20.
For example: To apply CoS group 02 with Auto Call Back Busy and No Reply for each Time zone in
Station Basic Feature Template 02, dial the following commands:
5502-1-02-03-02 for Working hours
5502-1-02-04-02 for Break hours
5502-1-02-05-02 for Non-working hours
To apply the Station Basic Feature Template now programmed with Auto Call Back when Busy and No
Reply, to the extensions, dial the following commands:
If extension is an SLT, dial:
5503-1-SLT-Template Number to apply on a single SLT.
5503-2-SLT-SLT-Template Number to apply on a range of SLTs.
5503-*-Template Number to apply on all SLTs.
Where,
SLT is the Software port number of the SLT, from 001 to 512
Template Number is the number of the Station Basic Feature Template (01 to 50) you have
programmed with the Auto Call Back feature.
If extension is a DKP, dial:
5504-1-DKP-Template Number to apply to a single DKP.
5504-2-DKP-DKP-Template Number to apply to a range of DKPs.
5504-*-Template Number to apply to all DKPs
Where,
DKP is the Software port number of the DKP, from 001 to 128.
Template Number is from 01 to 50.
To program the Auto Call Back Ring Timer:
Dial command 3801-Seconds
Where,
Seconds = 001 to 255 seconds.
Exit SE mode.
1507
How to use
Extension users can set two types of Auto Call Back:
Auto Call Back on Busy - when the extension/trunk they are trying is Busy.
Auto Call Back on No Reply - when there is no reply from the extension they are trying.
Auto Call Back can be set from EON as well as any SLT.
Press the 'Call Back' Key on EON48 on Busy Tone or press the DSS Key assigned to Auto Call Back on
EON310 on Busy Tone.
You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key will
be turned on.
Go idle or you get dial tone after 3 seconds.
Press the 'Call Back' Key on EON48 again or press the DSS Key assigned to Auto Call Back on EON310
again.
You get confirmatory message "Auto Call Back Canceled" on the phone's display. The LED of the DSS
Key will be turned off.
Go idle or you get dial tone after 3 seconds.
Using Command:
To set Auto Call Back on Busy:
Dial 102.
You get confirmatory message 'Auto Call Back Canceled' on the phone's display. The LED of the DSS key
assigned to Auto Call Back will be turned off.
Go idle or you get dial tone after 3 seconds.
1508
On Busy Tone.
Dial 2.
Press the 'Call Back' Key / DSS Key assigned to Auto Call Back on Ring Back Tone.
You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key will
be turned on.
Go idle or you get dial tone after 3 seconds.
Press the 'Call Back' Key / DSS Key assigned to Auto Call Back again.
You get confirmatory message "Auto Call Back Canceled" on the phone's display. The LED of the DSS
Key will be turned off.
Go idle or you get dial tone after 3 seconds.
Using Command:
To set Auto Call Back on No Reply:
Dial 102.
You get confirmatory message 'Auto Call Back Canceled' on the phone's display. The LED of the DSS key
assigned to Auto Call Back will be turned off.
Go idle or you get dial tone after 3 seconds.
1509
1510
Auto Redial
What's this?
The Auto Redial feature retries a call automatically if the dialed number is busy. It repeatedly checks the busy line
till it is free. When the called number is no longer busy, the extension of the caller rings.
Auto Redial saves time and the effort of repeatedly dialing the entire phone number over and over until the called
party gets off the phone.
The Auto Redial feature is supported for external numbers only.
How it works
When an extension user dials a number and gets a busy tone, s/he may set Auto Redial. When Auto Redial is set,
ETERNITY will dial out the requested number and will wait until the 'Ring Back Tone Wait Timer227'
expires to sense the Ring Back Tone from the requested number. This timer is programmable and is set to
60 seconds as default.
If the system does not detect Ring Back Tone for 60 seconds, it releases the trunk and tries again after
some time. If the system detects a busy tone, it releases the trunk and redials the number automatically
after some time. This process is repeated until the system detects the Ring Back Tone.
When the ETERNITY detects the Ring Back Tone instead of the Busy Tone, it will ring on the extension
that set Auto Redial. The extension will ring for the duration of the 'Redial Ring Timer228'. This timer is
programmable and is set to 45 seconds as default.
If the extension is in the middle of any activity such as dialing, ringing or speech, the ETERNITY will
suspend Auto Redial until the extension becomes idle again. After which it dials the requested number
again.
Two types of Auto Redial are supported by the ETERNITY - Auto Redial (normal) and Auto Redial 'Priority' - that
differ from each other in terms of the number of redial attempts and the interval between attempts.
Auto Redial (normal): The system is programmed by default to make 5 attempts to redial at an interval of
45 seconds (default) between each attempt. Both, the number of attempts as well as the duration of the
interval can be changed match User preference, like decreasing the number of attempts to 3 and
increasing the interval to 60 seconds.
Auto Redial 'Priority': the system makes a greater number of attempts to redial and the duration of the
interval between each attempt is less. By default the system is programmed to make 20 redial attempts at
227. Time for which ETERNITY waits to sense the RBT from the PSTN/CO Network after dialing the requested number. This timer is
particularly relevant to CO ports. Valid range of the timer: 000 to 255 seconds. Default: 060 seconds.
228. Time for which the extension that has requested Auto Redial should ring. Valid range of the timer: 000 to 255 seconds. Default:
045 seconds.
1511
intervals of 20 seconds. The number of attempts as well as duration of the interval are programmable; for
instance, the number of attempts can be set to 30 and the interval to 15 seconds.
To change the number of redial attempts and the interval between them, the SE must Auto Redial Count and the
Auto Redial Timer respectively. In addition to these, the system has three other related timers, which can be
programmed to match User preference:
An extension user can request Auto Redial for multiple numbers at a time from the same extension and
more than one extension can attempt auto redial simultaneously.
The system uses the same OG Trunk Bundle Group you used. If you dialed the number on group code
60, the system grabs one of the free trunks from group code 60 for Auto Redial.
If the number was dialed the first time using selective trunk access, the system will use the same trunk
to execute Auto Redial.
If the extension is programmed for 'Dynamic Lock', and you have set the 'Auto Redial', the system will
check the Toll control as per dynamic lock level.
Auto Redial may not work well on Two-wire Trunk lines, as its functioning greatly depends on line
condition. Unlike ISDN, GSM and VoIP trunks, the line condition of CO trunks may not always measure up
to the standard requirement for Auto Redial to function.
How to configure
For Auto Redial to work, the System Engineer must:
1. Enable the features 'Auto Redial' and 'Auto Redial Priority' in the Class of Service (CoS) group of the
preference. This will change the number of redial attempts made by the system and the interval between
them.
3. If required, also change other related Timers such as Auto Redial Dial Tone Wait Timer, Auto Redial Ring
1512
However, if Auto Redial/Auto Redial Priority is to be allowed on only select extensions, follow these steps:
1. Define a CoS group with Auto Redial/Auto Redial Priority enabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
3. Assign this new Template to the extensions to which Auto Redial/Auto Redial Priority is to be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions.
To change Auto Redial Counts and Timers:
Below Auto Redial, change the Count and Timer of the type of Auto Redial - Normal or Priority - as per
your requirement.
You may change any of the related timers - Auto Redial Dial Tone Wait Timer, Auto Redial Ring Back Tone
(RBT) Wait Timer, Auto Redial Ring Timer - as per your preferences on this page.
1513
The Timer for Auto Redial Normal as well as Priority must be set to more than 5 seconds.
ii. The Auto Redial Priority Count should be set to less than 15.
enable a feature (in this case Auto Redial and Auto Redial Priority) in the CoS group.
apply the Template with the on DKP and SLT extensions, using SE commands.
Exit SE mode.
How to use
Auto Redial can be set/canceled from EON/Extended IP Phone as well as SLT.
Using Command:
To set Auto Redial:
1515
How it works
The gadget connected to the DOP can be operated in 9 different ways, referred to as Gadget Operation Modes.
Each of these is described in the table below:
Gadget Operation Modes
Gadget
Mode No.
1
Description
Typical Application
Office Lights.
1516
Gadget
Mode No.
2
Description
Typical Application
Door Lock.
Switch ON and Switch OFF the gadget as per the FlipFlop Interval.
The gadget remains ON for the duration of the Timer
called Flip-Flop Interval and remains switched OFF for
the duration of the same Timer.
Festive illuminations.
Strings of light bulbs can be turned on
and turned off in the intervals of a few
seconds to get a blinking effect.
Glow signboard.
Glow signboards can be turned ON
manually when required. Regardless of
when they were turned on, they can be
turned OFF automatically at a particular
time (for example, 7am).
For example, you can turn on the Glow
signboards in the evenings whenever
you remember to and have them
switched OFF early next morning.
Sprinklers.
You can have Sprinklers started to
water the lawn at a particular time of the
day and turn OFF the sprinklers
manually when you want by dialing a
command.
1517
Gadget
Mode No.
6
Description
Typical Application
A hooter/siren.
A fire alarm or smoke sensor connected
to the DIP senses smoke sends
instigation to the DOP to turn on the
hooter/siren connected to it.
Once the emergency is identified, the
hooter/siren connected to the DOP can
be switched off by dialing a command.
1518
Gadget
Mode No.
10
Description
Typical Application
Feature Commands override the gadget mode set for an application. You can turn ON or turn OFF a
gadget anytime by dialing Feature Commands, even when the Gadget mode selected is based on a
Timer or a Schedule or Instigation from DIP. For example, if you have selected Gadget mode 6 "Switch
ON and Switch OFF the Gadget as per the Schedule", you can at any time switch ON/OFF the gadget
with the Feature Command. The system will give precedence to the command issued and act
accordingly, without waiting for the ON/OFF-Time. Further control of the gadget will be as per schedule,
unless Feature Command is dialed again.
Similarly, when the Gadget mode is Switch ON and OFF on Instigation from DIP, you can still dial the
Feature Command to switch ON. The system will turn ON the gadget without waiting for instigation
from the DIP. The system will wait for instigation from DIP to turn OFF the gadget. But if you dial the
feature Command to switch OFF, the system will turn OFF the gadget without waiting for instigation
from DIP.
ETERNITY supports only one Schedule for all gadgets. If you want to operate more than one gadget
(possible only on ETERNITY PE, as it supports 3 DOPs) as per the "Schedule" mode, the Schedule
you program will be applied commonly to all the gadgets.
Each Time Schedule has on - ON and corresponding OFF Time - stored against an Index number from
1 to 24. You cannot configure the same time againts any Index number. For example, if at Index 1 you
have set the Scheduled ON Time as 12:50:00, then you cannot set this time as the Scheduled ON
Time or Scheduled OFF Time against any Index.
When the DOP is programmed to be switched ON/OFF on instigation from the DIP, the trigger time of
the DIP will be the same as the Event Sense Timer of the DIP. Refer Digital Input Port (DIP) to know
more about this Timer.
Automated Control Applications operated in the 'Preset ON-Time', 'Preset OFF-Time' and 'Scheduled'
modes function on the basis of the Real Time Clock (RTC) of the ETERNITY. Though the RTC circuit
automatically updates the date, day and time values, it may drift over a long period. Check the RTC
values every month and reset the values to correct the drift.
1519
For instructions refer the topics Installing ETERNITY ME/Installing ETERNITY GE/Installing ETERNITY PE as
relevant to your model of ETERNITY. Also refer the topic Digital Output Port (DOP) and Digital Input Port (DIP).
How to configure
Programming of automated control applications involves the following:
Configuring the Digital Output Port to which the gadget is connected. Refer the topic Digital Output Port
(DOP) for instructions.
Configuring the Digital Input Port, if you want to operate the gadget in conjunction with a device connected
to the DIP. Refer the topic Digital Input Port (DIP).
1520
Gadget mode: Select gadget mode in which you want to operate the device. Refer the description of
"Gadget Operation Modes" earlier in this topic.
Depending on the Gadget mode you select, you may have to program the relevant Timer. For example,
if you select the option "Switch ON the gadget using Command and Switch OFF after a Preset interval"
as Gadget mode you may program the Preset Interval. All other Timer fields will be non-editable.
DOP Number: Select the DOP number to which the gadget is connected.
ETERNITY ME and GE have only one DOP. So, you will get only DOP1 as the option. If you are using
ETERNITY PE, select the DOP number (DOP1, DOP2, DOP3) to which the gadget is connected.
1521
Preset Interval (sec): You may program this Timer if you have selected the option "Switch ON the
gadget using Command and Switch OFF after a Preset interval" or the option "Switch ON the gadget
following instigation from DIP and Switch OFF after a Preset Interval" as Gadget mode.
The range of this timer is from 001 to 255 seconds. By default it is set to 10 seconds.
Flip-Flop Interval (sec): You may program this Timer if you have selected the option 'Switch ON and
Switch OFF the gadget as per the Flip-Flop Interval' as Gadget mode.
The range of this timer is from 001 to 255 seconds. By default it is set to 3 seconds.
1522
Preset ON-Time (HH:MM:SS): You may program this Timer, if you have selected the option 'Switch
ON the gadget at Preset ON time and Switch OFF using command' as Gadget mode. The time format
is in 24 Hours: Minutes: Seconds format. Select the hours, minutes and seconds in the respective
boxes.
Preset OFF-Time (HH:MM:SS): You may program this Timer, if you have selected the option 'Switch
ON the gadget using command and Switch OFF at the Preset OFF time' as Gadget mode. The time
format is in 24 Hours: Minutes: Seconds format. Select the hours, minutes and seconds from the
respective boxes.
1523
1524
If you have selected the option Switch ON and Switch OFF the Gadget as per the Schedule as the
Gadget mode, you must also program the start and end time for the schedule. To do this,
The Scheduled ON Time and OFF Time are to be programmed in the 24 Hours, Minutes and Seconds
format (HH:MM:SS).
You can program different ON and OFF Time schedules for 24 hours in a day. So you can run the
control application more than once in a day at different times.
Each Time Schedule has an - ON and corresponding OFF Time - stored against an Index number from
1 to 24. You cannot configure the same time againts any Index number. For example, if at Index 1 you
have set the Scheduled ON Time as 12:50:00, then you cannot set this time as the Scheduled ON
Time or Scheduled OFF Time against any Index.
The system stores each ON and OFF Time Schedule in ascending order of time. If you want to run the
control application more than once in a day, program the Time Schedule in the sequence from earliest
to last against each Index, that is, morning hours should be programmed before evening hours against
each Index number.
Set the DOP Contact Type as Normally Open/Normally Closed. By default it is set to 'Normally Open'.
Click Submit at the bottom of the page to save your DOP settings
Repeat the same steps to program the parameters of Gadget-2 and Gadget-3 (applicable for
ETERNITY PE only).
1525
Enter SE mode.
To select the Gadget Mode for a gadget, dial:
1711-Gadget Number-Gadget Mode
Where,
Gadget is from 1 to 3
Gadget mode is from 1 to 10229.
By default, 1 is selected for all gadgets.
To assign DOP for the Gadget, dial:
1712-Gadget Number-DOP
Where,
DOP is
0 for None
1 for DOP1
2 for DOP2
3 for DOP3
Gadget is from 1 to 3
By default, None is assigned to all gadgets.
To program the Preset Interval, dial:
1713-Gadget Number-Preset Interval
Where,
Gadget is from 1 to 3
Preset Interval is from 001 to 255 seconds.
By default Preset Interval is set to 010 seconds
To program 'Flip-Flop Timer', dial:
1714-Gadget Number-Flip-Flop Timer
Where,
Gadget is from 1 to 3.
Flip-Flop Timer is from 001 to 255 seconds.
By default Flip-Flop Timer is set to 003 seconds
To program 'Preset ON Time', dial:
1715-Gadget Number-Preset ON Time
Where,
Gadget is from 1 to 3
Preset ON Time is to be entered in HH MM SS (24 Hours) format.
To program 'Preset OFF Time', dial:
1716-Gadget Number-Preset OFF Time
Where,
Gadget is from 1 to 3
Preset OFF Time is to be entered in HHMMSS (24 Hours) format
229. Refer the table 'Gadget Operation Modes' at the beginning of this topic for description of the modes and their numbers.
1526
Exit SE mode.
How to use
The gadget programmed to work on the basis of a Timer or a Schedule or on instigation from the DIP, will work on
the set time/instigation received.
Users need to dial Feature Commands if the selected Gadget mode requires it or whenever they want to override
the gadget mode selected to operate the gadget. For this, they must dial feature commands to turn ON and turn
OFF the DOP to which the gadget is connected.
DOP number is the number of the DOP from 1 to 3 to which the gadget is connected.
1527
DOP number is the number of the DOP from 1 to 3 to which the gadget is connected.
DOP number is the number of the DOP from 1 to 3 to which the gadget is connected.
1528
How it works
Automatic Number Translation makes use of the Automatic Number Translation (ANT) Table. The ANT Table
consists of three columns:
Dialed Number: This column contains the numbers you expect the users to dial.
Strip Digit: This column contains the number of digit(s) to be stripped off by the system from the Dialed
Number string before dialing it out.
Add Prefix: This column contains the digit(s) which are to be added as prefix to the Dialed Number string
by the system before dialing it out.
This table is applied on the desired trunk, through which outgoing calls are made.
Upto 8 different ANT tables can be configured and each table can accommodate upto 32 strings.
Here is an example of how this table is to be configured and used:
You want
All 10-digit numbers to be dialed out after adding the prefix 1.
All 7-digit numbers, starting with 2 to be dialed out after adding the prefix 1315.
All numbers beginning with 91 to be stripped off the first 2 digits and 0 to be added as prefix.
When you do not want to specify any numeric digits in the numbers to be modified, use the character $. This
character represents any numeric digit from 0 to 9. For example, a 10-digit number (having the numeric digits from
0 to 9) can be represented using this character as $$$$$$$$$$.
1529
Thus, the entries you will need to make in the ANT table will be as follows:
Dialed Numbers
Strip Digit
Add Prefix
$$$$$$$$$$
2$$$$$$
1315
91
The entry for 10-digit numbers to be dialed out after adding the prefix 1 will be as shown in the first row of
this table. The 10-digit number is represented with the $ character in the Dialed Numbers column. Since no
digit is to be stripped off, 0 is entered in the Strip Digit column. As the prefix 1 is to be added, this number
is entered in the Add Prefix column. The system will add 1 as prefix before dialing out numbers from
0000000000 to 9999999999.
Similarly, the entry for 7-digit numbers starting with 2 to be dialed out after adding the prefix 1315 will be
as shown in the second row of this table. The system will add 1315 as prefix before dialing out numbers
from 2000000 to 2999999.
The entry for all numbers beginning with 91 to be stripped off the first 2 digits and 0 to be added as prefix
will be as shown in the third row of this table. When users dial numbers beginning with 91, the system will
strip off the first two digits and add 0. For example, when a user dials the number 919925801882, the
system will dial out 09925801882.
Automatic Number Translation also forms the basis of Multi-Stage Dialing.
How to configure
To apply Automatic Number Translation on a trunk,
Decide which of the trunks types are to be assigned the Automatic Number Translation (ANT) feature.
Decide the number of ANT tables you need. You can program 8 different tables, with a maximum of 32
entries in each.
Make the tables on a piece of paper by drawing three-column tables. In the first column of a the table, write
the dialed numbers that need to be modified before being dialed out from the trunk. For each dialed
number in the first column, enter the number of digits you want the system to strip off (if required) from this
number in the second column. In the third column enter the number you want the system to add as prefix
(if required) before dialing out the number.
Configure the Automatic Number Translation Table in the system using the tables you prepared.
Enable the Automatic Number Translation flag in the Outgoing Trunk Bundle (OGTB) of the trunk.
Assign the Automatic Number Translation Table you configured to the OGTB of the trunk.
1530
Click Configuration.
In the Dialed Number column of the table you chose, configure the numbers you expect the extension
users to dial. The Dialed Numbers can be a maximum of 16 characters. Default: Blank.
When you want to specify number-length without specifying the numeric digits, use the character $ to
represent the digit. $ represents any number from 0 to 9.
In the Strip Digit column, enter the number of digits you want the system to strip off from the Dialed
Number before the dialing out this number. The valid range is from 00 to 16. Default: 0.
In the Add Prefix column, enter the number you want the system to add as prefix to the Dialed Number
before the system dials out this number. The Prefix can be a maximum of 40 characters. Default: Blank.
1531
In the OG Trunk Bundle of the desired trunk, enable the Automatic Number Translation (ANT) Apply
flag by selecting the check box.
In the ANT Table No. list, select the table number you configured for this trunk.
1532
Code is:
0 for Disable
1 for Enable
Default: Disabled.
For example:
To enable Automatic Number Translation flag in OGTB number 1, dial: 6702-1-001-5-1
To enable the same flag in OGTB numbers 1 to 8, dial: 6702-2-001-008-5-1
To enable the same flag in all OGTBs, dial: 6702-*-5-1
To assign an Automatic Number Translation Table, dial:
6702-1-OG Trunk Bundle Number-Feature Number-Code to assign a Table to a single trunk bundle.
6702-2-OG Trunk Bundle Number-OG Trunk Bundle Number-Feature Number-Code to assign the
same Table to a range of trunk bundles.
6702-*-Feature Number-Code to assign the same Table to all trunk bundles.
Where,
OG Trunk Bundle Number is from 001 to 128.
Feature Number for Automatic Number Table is '6'.
Code is ANT Table number from 1 to 8.
For example:
To assign ANT Table 3 to OGTB number 1, dial: 6702-1-001-6-3
To assign ANT Table 3 to OGTB numbers 1 to 8, dial: 6702-2-001-008-6-3
To assign ANT Table to all OGTBs, dial: 6702-*-6-3
Exit SE mode.
1533
How it works
Background Music on DKP
When the Background Music feature is enabled on a DKP extension,
Music is played after the extension user dials the Background music feature code and goes ON-Hook.
Music is stopped automatically whenever there is an activity on the extension phone, such as:
an incoming call landing on the extension. (Music is stopped and the phone rings).
the extension user going OFF-Hook to make an outgoing call. (Music is stopped and the system dial
tone is played.)
the extension user goes OFF-Hook to access any system feature using the phone.
Volume of the background music can be controlled using the Volume keys of the DKP.
1534
The extension user must first press the Speaker key. The system interprets this as 'OFF-Hook' and plays
dial tone.
The extension user must dial the Background music feature code and press the Speaker key again. The
system interprets this as 'OFF-Hook'. It plays the dial tone and waits for the First Digit Timer to elapse.
Background Music is played only after the First Digit Timer has elapsed.
Music is stopped when there is an incoming call. The extension user is played Ring Back Tone.
The extension user must go ON-Hook by pressing the Speaker key. Ring for incoming call is played.
Once the call has ended, the extension user can go ON-Hook. If the extension goes OFF-Hook, Music will
be played again at the end of the dial tone and the First Digit Timer.
However, if the extension user dials a feature access code/extension number/external number before the
end of the First Digit Timer, music will not be played, until the extension goes OFF-Hook again.
Volume of the music can be controlled using the volume keys of the SLT.
Background Music can be played only on extensions that have this feature enabled in the Class of
Service (COS) assigned to them.
How to configure
For the Background Music feature to work, the System Engineer must:
1. Enable Background Music in the Class of Service of the extensions to which this feature is to be allowed.
2. Connect a compatible external music device to the AIP.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all extensions of
ETERNITY. The Station Basic Feature Template 01 has the feature Background Music enabled in the default
Class of Service (COS) group 01. So, all extension users of the ETERNITY can play Background Music,
provided that their phones are a DKP or an SLT with Speaker.
In case Background Music is to be denied to an extension user, follow these steps:
1. Define a CoS group with Background Music disabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
3. Assign this new Template to the Station to which Background Music is to be denied.
1535
Refer the topics Class of Service (COS) and Station Basic Feature Template for instructions.
How to use
Background Music can be played by extension users whose phone is a DKP or an SLT with Speaker function.
1536
Barge-In
What's this?
Barge-In allows you to break into an on-going conversation between two extension users, between an extension
user and an external caller as well.
Barge-In can be used by Operators to transfer Incoming calls to busy extensions. The Operator can put the caller
on hold, barge into the busy extension to inform about the call, and then transfer the call.
Barge-In can be used by a Boss to interrupt the secretary's busy extension.
ETERNITY offers flexibility to allow/deny Barge-In feature to an extension user, that is, allow the extension user to
barge into on-going conversations. It also provides the flexibility to prevent conversations of extension users from
being barged in, referred to as Privacy against Barge-In.
How it works
C calls A.
C gets Ring Back tone (RBT) and A gets beeps indicating a new call. If A is using EON, C's name and
number appear on Cs phone display.
C gets RBT and A gets beeps for Barge-in timer. (By default, 10 seconds)
If A does not respond till the end of the Barge-In Timer (set to 10 seconds, by default), A gets connected to
C. B is put on hold and is given hold-on music.
If B disconnects while A and C are talking, the held call between A and B is cleared.
If B keeps holding the call and C disconnects, the call between A and C is cleared and A is connected back
to B.
If B keeps holding the call and A disconnects, the call between A and C is cleared and A gets ring. A picksup the handset and gets connected back to B.
1537
Feature Interactions
Call States:
Barge-In works only if the dialed extension is busy. The dialed extension may be busy with another
extension or trunk (external number).
Barge-In works only if the user about to be barged in is in a two-way normal speech with another user
or external party.
It will not work if the busy signal is due to the user being OFF-Hook, or in the middle of dialing, or
accessing a feature of the PBX.
Call Toggle: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
Privacy against Barge-In: If the feature 'Privacy against Barge-in is enabled for an extension, it cannot be
barged into.
Priority: No Interaction with Barge-In. If 'A' has lower priority than 'B' but has Barge-In enabled; A can
barge in B.
Do Not Disturb (DND): Barge-In will not work if the called user has set DND. If 'A' has set DND. A is
busy with C. B calls A. B cannot barge in A.
DND-Override: Barge-In will work if the calling user is allowed DND-Override and also has higher
'Priority' than the called user. If 'A' has set DND. A is busy with C. B calls A. On busy signal, B dials the
Barge-In code. Barge-In will be successful only if B has DND-Override enabled and has higher priority
than A.
How to configure
The functioning of this feature is controlled by three parameters, 'Barge-In', 'Privacy against Barge-In' and 'Barge-In
Timer'.
1538
b. Prepare an extension Basic Feature Template with this CoS group applicable in all the Time Zones.
c. Assign this new Template to the extensions to which Barge-In is to be allowed.
Repeat the above steps to allow 'Privacy from Barge-In' in the CoS of extensions that are to be exempted from
Barge-In.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions on
programming.
Barge-In Timer
Barge-In Timer is the time after which the caller gets connected to the called party. By default the Timer is set to 10
seconds.
Under Configuration, click System Timers and Counts to open the page.
1539
Enter SE mode.
Dial command 3803-Seconds
Where,
Seconds is from 001 to 255 seconds. Default is 10 seconds.
Exit SE mode.
How to use
For EON and Extended IP Phone Users
Dial an extension.
If the extension is busy, you get Busy Tone.
Press DSS Key assigned to 'Barge-In' function.
OR
Dial 4230.
You get Ring Back Tone.
Wait for the system to connect you to the called extension.
Talk.
Replace the handset after the conversation has ended.
Dial an extension.
If the extension is busy, you get Busy Tone.
Dial 4.
You get Ring Back Tone.
Wait for the system to connect you to the called extension.
Talk.
Replace the handset after the conversation has ended.
230. This default feature access code can be changed to suit your preference. Refer the topic Access Codes.
1540
BCCH Selection
What's this?
BCCH Selection feature enables you to lock the Mobile Port of ETERNITY to a particular cell or channel or BTS
(Base Transceiver Station) for various reasons such as:
How it works
In the GSM network, each BTS is assigned one particular channel called as ARFCN (Absolute Radio Frequency
Channel Number), which is transmitted by BTS in BCCH (Broadcast Control Channel).
Now, when ETERNITY is switched on, the Mobile Port gets registered with the network on a particular BTS which
has the highest signal strength. However, the signal strength is not consistent. It keeps fluctuating, resulting in call
drop or poor voice quality.
Therefore, to avoid this, ETERNITY enables you to lock the Mobile Port to a particular cell or channel manually
after checking Signal Strength and Signal Quality of each cell.
How to configure
You can lock Mobile Port to a cell or a channel only through Jeeves.
1541
1542
Mobile Port Number: This is number of the Mobile port for which BCCH Selection status is displayed.
You can choose a different Mobile Port number from the drop down list. The page will display the
BCCH Selection related parameters for the selected mobile port.
Mobile Port Status: The current state of the Mobile Port is displayed in this field. Given below is the
description of the various status indication messages that will appear in this field.
STATUS
DESCRIPTION
Disabled
GSM
Initialization
Displayed when GSM module is in initialization state, that is, before SIM
detection.
SIM Absent
SIM PUK
required
Registering
Displayed when the Mobile Port is in registration process with the Network.
Idle
Displayed when the Mobile Port is registered with the Network and it is
free.
Busy
BCCH Locking Status: The current BCCH Locking status of the mobile port is displayed in this field.
Given below is a description of the various BCCH Locking status indication messages that will appear
in this field.
STATUS
DESCRIPTION
Trying to Lock
Displayed when user selects Manual BCCH Locking as 'No' from 'Yes' and
module is in initialization process after system or module restart.
Trying to lock on
BCCH xxxxx
Displayed when BCCH Locking is selected as Manual and the Mobile Port is in
the registration process with the Network. xxxxx is the BCCH selected by the
user for locking the cell.
Manually Locked
on BCCH xxxxx
Auto Locked on
BCCH xxxxx
Main Cell- Bit Error Rate (%): Bit Error Rate of the Main Cell is displayed in this field. Bit Error Rate
(BER) is the percentage of received bits on a digital link that are in error relative to the number of bits
received. Bit Error Rate is calculated from the received signal quality.
Manual BCCH Locking: This parameter allows you to lock the Mobile Port to a particular cell of your
preference. By default, manual BCCH locking is set to 'No'. When manual BCCH locking is set to 'No',
Mobile Port gets locked to the cell as per the highest signal strength. Select 'Yes' if you want to lock the
Mobile Port to the particular cell selected by you.
Auto Refresh: Click this button to refresh BCCH Selection page. All parameters on this page will be
downloaded automatically after every 15 seconds. By default, Auto Refresh button is enabled.
Stop Auto Refresh: By clicking this button, you can stop the system from automatically refreshing the
BCCH Selection page every 15 seconds. When you stop Auto Refresh, you must click 'Refresh' at the
bottom of this page to refresh the page whenever you want
Cells: Indicates the cells with which the Mobile Port can be locked. You can decide to lock the Mobile Port
with a particular cell after considering the following cell related parameters, which appear on the page:
MCC-MNC: In this field, MCC-MNC of a cell is displayed. Mobile Country Code (MCC) is a three digit
number uniquely identifying a country and Mobile Network Code (MNC) is either a two or three digit
number used to identify a given network from within a specific country.
LAC (Location Area Code): In this field, LAC (Location Area Code) is displayed. LAC uniquely
identifies a location area within a GSM PLMN (Public Land Mobile Network). The maximum length of
LAC is 16 bits ranging from 0 to 65535. LAC is displayed in hexadecimal characters for SIMCOM-2G
and Wavecom-2G engines which ranges from 0000 to FFFF. For SIMCOM-3G engine, LAC is
displayed in decimal digits which ranges from 00000 to 65535.
Cell ID: In this field, Cell ID is displayed. It is a 16-bit identifier that identifies the cell. Cell ID is
displayed in hexadecimal characters for SIMCOM-2G and Wavecom-2G engines which ranges from
0000 to FFFF. For SIMCOM-3G engine, Cell ID is displayed in decimal digits which ranges from 00000
to 65535.
1543
BSIC (Base Station Identification Code): In this field, BSIC (Base Station Identification Code) is
displayed. BSIC allows a mobile station to distinguish between different neighboring base stations.
BSIC is a three-digit value ranging from 0 to 255.
BCCH (Broadcast Control Channel): In this field, the BCCH value of the cell is displayed. BCCH
defines the frequency channel number.
Receive Level: In this field, the Receive Signal Strength level of the cell is displayed. It is the average
Receive Signal Strength of the cell. Its value ranges from -110 dBm to -47 dBm.
Manual Cell Locking: This radio button is for locking a Mobile Port to a selected cell manually.
Select the desired Mobile Port Number from the drop down list.
Set the parameter Manual BCCH Locking to Yes.
Go to the Cell to which you want to lock the Mobile Port you selected.
Select the radio button Manual Cell Locking of that Cell.
Click Submit at the bottom of the page.
The BCCH Locking for the selected Mobile Port will appear on this page, if Auto Refresh is enabled.
If you have stopped Auto Refresh, click 'Refresh' at the bottom of the page to refresh the page and view
the current BCCH Locking settings of the selected Mobile port.
You may now log out of Jeeves.
Example:
Consider the following example when using this feature:
Problem:
ETERNITY is installed in roaming area, where more than one network is available, say A and B.
Mobile Network Selection is set to 'Manual' mode and the first priority is programmed as network A and the
second priority is programmed as network B.
The Mobile Port gets registered with A network. After registration, the user locks the Mobile Port to one of
the cells of A network.
After registration, if the module or the system restarts or gets deregistered from the network, module starts
registration process again.
While re-registering, ETERNITY tries to lock the Mobile Port to the last selected cell of network A.
If network A is unavailable then the Mobile Port will not get registered with the network.
Solution:
1544
In this situation, user should set Manual BCCH locking mode to 'No' to register Mobile Port with the
suitable network automatically.
Later, the user can set the Manual BCCH locking mode to 'Yes' and lock the Mobile Port to the desired cell
after assessing the cell information.
How it works
Consider the following illustration.
21
S1
22
S2
23
S3
T1
T2
T3
T4
T5
T6
S4
T7
S6
S5
PBX A
Sn
31
S2
32
S3
33
PSTN
S7
Tn
ETERNITY
S1
T1
S8
T2
S9
PBX-A is connected behind ETERNITY. In this 'Behind the PBX' configuration, the Trunk Lines T5, T6, T7 of
ETERNITY are connected to the Stations (SLT) S4, S5, S6 of PBX-A.
However, Trunk lines T1 and T2 of PBX-A are connected directly to the PSTN.
In such application scenarios, implementing toll control restrictions for the trunks is a difficult task for ETERNITY.
For example: Extension number 21 of ETERNITY in the above illustration is not allowed the facility of long distance
dialing. It has access to all the CO trunks.
When the user of Extension 21 wants to access T1, T2 or T3 (which are direct trunks from the PSTN to ETERNITY)
the user dials '0' (Trunk Access Code programmed), gets PSTN dial tone. When the user dials the number,
ETERNITY applies Toll Control.
When the user of Extension 21 tries to grab a trunk T5, T6 or T7 (which are connected to stations of PBX-A) by
dialing Trunk Access Code, for example, '0', the user gets the dial tone of PBX A. This means, the user of
Extension 21 must dial '0' again to grab PSTN dial tone of the T1/T2 connected to PBX-A.
But when the user dials '0' again, ETERNITY plays an Error Tone, because ETERNITY has applied Toll Control
and since Extension 21 is not allowed long distance dialing, ETERNITY rejects dialing on trunk and plays error
tone.
1545
This would not have been a problem if Extension 21 were allowed long distance dialing. Since Extension 21 cannot
be allowed long distance dialing, ETERNITY provides a solution for this in the form of a programmable Pre-PSTN
Digit Count (PPDC) for each CO trunk.
The Pre-PSTN Digit Count defines the number of digits to be dialed to reach the PSTN. The system will apply Toll
Control check for the extension only after the programmed PPDC.
PPDC is to be programmed only for trunks that are connected to another PBX, and not for Trunks connected
directly to the PSTN. To take the above illustration further, PPDC must be programmed only for T5, T6, and T7.
PPDC count is to be programmed should have the same number of digits as the Trunk Access Codes programmed
for PBX-A. For example, if the Trunk Access Code is a single digit number, such as '0', the PPDC will be '1'. If Trunk
Access Code is a two-digit number, such as 61, the PPDC will be '2'.
Since PPDC is not applicable on trunks directly connected to the PSTN, it must be programmed as '0' for T1, T2,
T3, T4 of ETERNITY.
How to configure
The 'Pre-PSTN Digit Count' (PPDC) is to be programmed in the CO Hardware Template applied to the CO trunks
of the PBX that are connected to station ports of the other PBX as well as to CO trunks that are directly connected
to the PSTN.
For CO Trunks that are directly connected to the PSTN, PPDC must be programmed as '0'.
For CO Trunks that are connected to the stations of another PBX, PPDC must be programmed as per the
number of digits in the Trunk Access Codes defined for the second PBX.
1546
Scroll with the horizontal scroll bar to reach the PPDC column of the template.
By default CO Hardware Template Number 01 is assigned to all trunks. The default 'PPDC' in this template
is '0'.
For all trunks that are to be assigned PPDC '0' (that is, trunks connected directly to the PSTN), you may
retain this template.
For trunks that are to be assigned a PPDC count from 1 to 6 (that is, trunks connected to the stations of
another PBX), prepare another CO Hardware Template by selecting another template number, for
instance Template 02.
From the drop down list, select the appropriate value. This would depend on the number digits in the Trunk
Access Code defined for the trunks in the other PBX. If the TAC is single digit, select '1'. If TAC is double
or triple digit, select '2' or '3' as applicable as the PPDC.
Enter the number of the template you prepared (Template 02) in the field CO Hardware Template for
each port you want to assign this template.
For Trunks to be assigned PPDC Count 0, retain CO Hardware Template Number 01.
1547
Exit SE Mode.
1548
Building Intercom
What's this?
ETERNITY offers the Building Intercom application as the telecom and security solution for commercial and
residential buildings, such as malls, shopping complexes, residential apartment blocks and gated-communities.
With the Building Intercom application, you can also connect private networks, that is few PBXs with the intercom
application can be connected to each other using VoIP.
Presently, the Building Intercom application is supported in the ETERNITY ME and GE models and their variants.
For installation instructions, under Installing ETERNITY ME, see The Intercom Line Card and under Installing
ETERNITY GE, see The Intercom Line Card.
Caution: When installing ETERNITY ME/GE for Building Intercom, you can install only the Intercom Line
Cards, VoIP Card and the VMS Card in the system. If other Trunk cards or Extension or Combination
cards present in the system the Intercom Line Card will not work.
1549
How to use
Building Intercom has two applications:
1550
Standalone Application
Extended Application
Standalone Application
In this case the system is installed at the site where internal communication is required between internal members
in a same complex only.
Block A
PBX installed
in Block A
4051
4001
4052
4002
4053
4003
4100
4050
Here Block A is a residential complex with 100 flats. Eternity with ILC Cards is installed for internal communication
between the residents.
All the extensions 4001 to 4100 can communicate with each other.
Extended Application
Let us understand how to use the Building Intercom Application with the following illustration:
A residential society has three buildings. Each building has 50 flats. In each building a PBX (having Intercom Line
Cards) is installed. The residents in the same building only, can communicate with each other.
The residents of all the three buildings want to communication with each other.
To overcome this, you must install a VoIP Card in each PBX. All the three buildings will be connected using the IP
network as shown in the figure below.
For installing the VoIP Card see The VoIP Card under Installing ETERNITY ME and The VoIP Card under
Installing ETERNITY GE.
1551
In this case, PBX A, PBX B and PBX C will be connected to form a single group.The residents in Wing 1, 2 and 3
will be able to call each other using extension numbers..
Wing 2
Wing 1
2001
3001
2002
3002
3003
2003
PBX A
PBX B
192.168.1.1
2050
192.168.1.2
3050
IP Network
Wing 3
192.168.1.3
4001
4002
4003
PBX C
4050
Intercom calls can be made between Wing 1, 2 and 3 with suitable configurations of the PBX.
Select a SIP trunk to be used for this application and enable it. For example, SIP Trunk 1.
1552
In the Route Code field of the CUG table, enter the Number that will be dialed to call the users of PBX B
and C. In this case, 3 and 4 respectively. As the system uses the best match logic to match number strings
in the CUG table, you may configure only the prefix of the number to be dialed, instead of configuring the
complete number string.
For the number you entered, in the Dialed Digit Count field, enter 4.
Select the OG Trunk Bundle Group. Configure SIP Trunk1 as the only member in this group. The calls will be
routed through this SIP Trunk only.
Configure the Strip Digit Count as 0 and clear the Self Route check box.
The CUG table you configure in PBX A would look like this:
Index
Route Code
OG Trunk
Bundle
Group
Strip Digit
Count
01
01
Self Route
Dialed Digit
Count
In the Number field of the Peer-to-Peer table, enter the numbers of the extension users of PBX B and
C. In this case, 3 and 4, respectively.As the system uses the best match logic to match number strings
in the Peer-to-Peer table, you may configure only the prefix of the number to be dialed, instead of
configuring the complete number string.
For the number 3001 enter the Domain Address, enter the Domain Name/IP Address of PBX B. In this
case, 192.168.1.2 and for 4001 the Domain Address is 192.168.1.3
The SIP messages can be transported using UDP, TCP or TLS. Select the Default Transport for
Outgoing Message as per your requirement for each index.
The Peer-to-Peer table you configure in PBX A would look like this:
Domain
Address
Name
192.168.1.2
PBX B
TCP
192.168.1.3
PBX C
TCP
Index
Number
No Match
Found
2
3
When 2001 from PBX A dials 3001, the system compares it with the CUG table configured in PBX A.
When a match is found in the CUG table, the system uses SIP Trunk1 to route the call.
SIP Trunk1 is configured as a Peer-to-Peer trunk, hence the system will check the Peer-to-Peer table.
As 3001 is configured in the Peer-to-Peer table, the system fetches Destination address and transports
the SIP messages using the protocol select as the Default Transport for Outgoing Message.
1553
When there is an incoming call on PBX B, the system checks the CUG table first and as 3001 is not
programmed in the CUG table, it checks the flexible number of the extensions.
As 3001 is found in the flexible number list, the call is routed to the extension 3001.
When 2001 dials 4001, the call will be routed as per the above logic.
Similarly you must configure the parameters in PBX B and C.
In certain cases different building may have the same extension numbers, in these cases you must configure the
CUG Table with Exchange ID. For detailed information, see Closed User Group-With Exchange ID.
1554
With BLF subscription and BLF key configured, whenever, there is a change in the state of the monitored trunk,
ETERNITY sends a NOTIFY message to the SIP Extension. The NOTIFY message contains the Call State. On
receiving the NOTIFY message, the SIP Extension updates the LED indication of the BLF key on the SIP phone.
The SIP extensions will indicate the following calls states for the outgoing and incoming calls on the monitored
trunks:
Outgoing Calls
Call State
Description
Trying
Confirmed
When the external party answers the call and speech is established with the
extension user, that is, the call is matured.
Hold
When the call on the trunk has been put on hold by the extension.
Available/Idle
Call State
Description
Early
When an indication is received from the Network that the external party is ringing.
Confirmed
When the incoming call is placed on the SIP Extension as the destination and
speech is established with the extension user, that is, the call is matured.
Hold
When the call on the trunk has been put on hold by the SIP extension.
Available/Idle
SIP phones may differ in the BLF indication (LED color and cadence, text message display) they provide
for the Call States. Refer to the manufacturers documentation for BLF Indication supported on the SIP
phones.
1555
How it works
An Open Standard SIP Phone is registered as a SIP Extension, with the extension number 3301.
As the user of extension 3301 wants to monitor the trunk CO-001, BLF subscription is enabled on
extension 3301 and CO-001 is assigned to the BLF key on the SIP phone.
Extension 3301 makes an outgoing call to an external number 2630555. The BLF key will indicate the
current call state of the CO-001 Trunk as Trying according to the LED indication supported by the SIP
phone for this call state.
If the SIP phone supports text message display for call states, each call state will be displayed on the
phone.
When the external party answers the call, the call between the CO-001 and SIP Extension 3301 gets
matured. The BLF key will indicate the current call state of the CO Trunk as Confirmed according to the
LED indication supported by the SIP phone for this call state.
When the SIP Extension 3301 disconnects the call, the ETERNITY will disconnect the call of external
number and the BLF key will display the call state of the CO as Terminated according to the LED
indication supported by the SIP phone for this call state.
Similarly, the BLF key configured on the SIP Extension 3301 will display the call states of the CO-001 trunk
for incoming calls from external numbers.
Since multiple calls can be made through a single trunk, the BLF key will indicate the status of the first call
detected by the system. When the first call is terminated, the status of the second call (if ongoing) will be
indicated.Similarly the status of all subsequent calls will indicated after the previous call is terminated.
How to configure
To provide BLF to SIP extension users, you must do the following:
1556
Enable Busy Lamp Field Subscription on the SIP Extensions you want to provide this feature. For
instructions, see Configuring SIP Extension using Jeeves under Configuring SIP Extensions.
Assign a BLF Key for the trunk to be monitored on the SIP Phones registered as extensions. For
instructions refer to the manufacturers documentation (Installation Guide/User Guide) for the respective
SIP Phones.
To monitor the trunks, configure the BLF Key as per the table given below:
Trunk
User ID part in
SUBSCRIBE
CO
COxxx
BRI
BRIxxCHy
Remarks
Trunk
User ID part in
SUBSCRIBE
T1E1
T1E1xCHyy
Mobile
MOBxx
SIP
SIPxx
Remarks
1557
This feature requires a license. To use this feature you must purchase the license for the Mobility Feature
Suite. Refer the topic License Management to know more.
How it works
For this feature to work:
1558
The CLI of those callers whom the system should call back must be programmed in the Call Back
Incoming Number List.
The Call Back Timer may be programmed. When the caller disconnects within the Call Back Timer, the
Call Back will be applied for that number.
You must define Call Back on, that is, you must select whether the number which must be called back
should be the same CLI number which the call was received or an alternative number.
The number on which call back is to be made must be programmed in the Call Back Outgoing Number
List, if it is not the same CLI number or if it is an alternative number.
You must select whether the call back should be made using the same trunk port on which the call was
received or an Outgoing Trunk Bundle Group (OGTBG). If you select OGTBG, you must also program the
OGTBG.
You may enable Least Cost Routing (LCR) on the OGTB if you want the system to select the least cost
trunk for calling back the missed call number. Program LCR accordingly.
Select a Call Back Mode, that is, how the call should be routed when the call back is answered by the
remote party; whether it should be routed through Built-In Auto Attendant, DISA or Operator.
Following is an example of a Call Back on a mobile port, when the above parameters are programmed.
The system checks if the Call Back flag is enabled on mobile port 01.
The system matches the CLI of A with the Call Back Incoming Number List assigned to mobile port 01 to
determine if the calling number is eligible for a call back.
The system waits for the period of the Call Back Timer (programmable, default: 10 seconds).
A must disconnect before the expiry of the Call Back Timer so that the system can treat it as a Missed Call.
If A disconnects within the Call Back Timer, the system applies Call Back for As number.
The system checks the Call Back on parameter, whether it has to call back the same number or an
alternative number.
If an alternative number is programmed as Call Back on, the system checks the Outgoing Call Back
Number List for the alternative number. As the CLI of A matches with the number on Index 15 of the Call
Back Incoming Number List, the system checks Index 15 of the Call Back Outgoing Number List for the
corresponding alternative number to this number.
The system checks if the number is to be called from the same port or an OGTBG.
If the same port is programmed, the system will make a call to the number using mobile port 01.
If OGTBG is programmed, the system will check if Least Cost Routing is enabled in the OGTBG and make
the call back accordingly.
The system checks the type of Call Back Mode enabled on mobile port 01 (the port on which the call back
request was made).
Four scenarios are possible:
1. Auto Attendant is enabled as Call Back Mode on mobile port 01.
2. 'Pin Authentication - Multiple Calls' or 'CLI Authentication - Multiple Calls' is enabled as Call Back Mode
231. If the system does not find a match for the CLI of the caller in the Call Back Incoming Number List, the 'Call
Back' feature will not be applicable and the call will be processed according to the normal incoming call logic.
1559
A can now reach any station or trunk of ETERNITY from DISA Mode.
3. 'CLI Authentication - Single Call Answer Signaling' is enabled as Call Back Mode on mobile port 01.
The system lands the call on the Operator extension assigned to mobile port 01.
Read the topics Auto Attendant, Direct Inward System Access (DISA) and Configuring 'Operator' to
know more about the call respective call logic.
Since this feature is essentially for callers, they must be aware of its functioning to be able to use it, that
is, disconnect the call within the Call Back Timer. If the caller does not disconnect within the Call Back
Timer, the call will be processed according to the normal incoming call logic.
ETERNITY supports only one call back request at a time, for one trunk port. The second incoming call
on that trunk port will be processed by the system as per normal incoming call routing.
For call back requests made from an OGTBG, if any of its trunks is busy, ETERNITY will support only
the last call back request in the OGTBG. Previous requests will be processed as per the normal
incoming call management logic.
How to configure
For this feature to function, you must program the following parameters on each Trunk port type (CO, BRI, T1, E1,
Mobile, SIP) on which you want to use this feature:
Enable Call Back: This flag must be enabled on the desired trunk port on which you want to activate the
Call Back on Trunk Port feature. By default, this flag is disabled on all trunk port types.
Call Back Timer: This is the duration for which the system waits for the caller to disconnect the call after
the system has found a matching number for the callers CLI in the Call Back Incoming Number List.
When the caller disconnects within Call Back Timer, the system applies Call Back on the port. If the caller
does not disconnect within the Call Back Timer, the incoming call management logic is applied for the call
on the trunk port.
The range of this timer is from 01 to 99 seconds. By default, it is set to 10 seconds.
Call Back Incoming Number List: This is the list of numbers that are eligible for Call Back. The system
checks the CLI of the caller with this list to determine if the caller is eligible for a call back.
The system compares the number string programmed in the Call Back Incoming List with the number
string received as CLI.
1560
Number string programmed in the 'Call Back Incoming Number List' shall be compared with the actual
received CLI.
The number string programmed in the Call Back Incoming Number List may be shorter than the number
string received as CLI, but only if the programmed number string completely matches with the received
CLI from the right towards left, the system will consider it as a complete match.
For example, if the programmed string is 263055 and the number string received in the CLI is
2652630555, the system will consider it a complete match. If the received CLI 912652630555, the system
will consider this caller too as eligible for a call back. Thus any CLI received with 263055 as the last 7 digits
will be considered as match found.
By default, Number List 15 is assigned to all trunk port types as Call Back Incoming Number List. You
may program this list for all port types, or you may program another Number List and assign it to the
particular trunk port type.
Refer the topic Number Lists for instructions on how to configure the Number List.
Call Back on: For each Trunk port type you have set the Call Back feature, you must define Call Back on,
that is, you must select whether the number which must be called back should be the same number from
which the call was received or a different number.
When missed call is eligible for call back (matches with Incoming Number list), the 'Call Back on'
parameter determines the number on which the call back is to be made, that is, whether on the same
number from which the missed call is received or on a different number.
In countries where CLI received on trunks can be dialed out without any modification, you may select CLI
Number as Call Back on option.
In countries where CLI received on trunks can be dialed only after appropriate modification, you may
select Alternate Number as the Call Back on option. You may also select Alternate Number as Call
Back on when you want the call back to be made to a different number.
Call Back Outgoing Number List: When the system finds a missed call eligible for a call back, it will
make the call back on the basis of the Call Back on option you selected and the Outgoing Number List you
programmed.
If you selected CLI Number as Call Back on option, you do not need to program the corresponding
outgoing number for the CLI received.
However, if the CLI received needs to be modified before being dialed out, then program the modified CLI
in the Outgoing List as the corresponding outgoing number for the CLI received.
The modified CLI or the Alternate number should be programmed at the same index number as the index
number at with the received CLI is programmed in the Call Back Incoming Number List. For example, for
the received CLI number string programmed at Index 15 in the Call Back Incoming Number List, the
corresponding modified CLI/Alternate number string should be programmed at the same Index, 15, in the
Call Back Outgoing Number List.
When the CLI received matches with the number string programmed at Index 15 of the 'Call Back
Incoming Number List', the call back will be made using the (modified/Alternate) number programmed at
Index 15 of the 'Call Back Outgoing Number List'.
1561
By default, Number List 16 is assigned to all trunk port types as Call Back Outgoing Number List. You
may program this list for all port types, or you may program another Number List and assign it to the
particular trunk port type.
Refer the topic Number Lists for instructions on how to configure the Number List.
If you have selected Alternate Number as Call Back on option, but do not want to provide alternative
numbers to call back particular callers (that is, CLI received), in such a case, program the CLI of these
callers in the Incoming Number List but keep the corresponding index numbers in the Outgoing Number
Lists blank.
Call Back from: This parameter determines the trunk port to be used to make call back.The call back can
be made using the same port or an Outgoing Trunk Bundle Group (OTGTBG). Select Same port if you
want the call back to be made using the same port on which the missed call was received. If you select
OGTBG, the call back will be made using the OGTBG, which you have defined.
OGTBG for Call Back: If you selected OGTBG for making the call back in the previous parameter, you
must assign the OGTBG that must be used in this parameter.
By default, OGTBG 01 is selected for Call Back.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost Routing
on the OGTBG that you define here for Call Back.
Call Back Mode: Select from the following options how a Call Back call answered by the remote party
should be routed:
Built-In Auto Attendant: The system will process the call as per the Built-In Auto Attendant call
logic - give a dial tone to the remote party, who can now call any extension. Refer the feature
description for Auto Attendant.
PIN Authentication-Multiple Calls: The system will process the call as per DISA call logic - allow
remote party to enter DISA mode with PIN-Authentication. On successful authentication (DISA
Login) the user is allowed to make calls or use features as allowed to him/her.
CLI Authentication-Multiple Calls: The system will process the call as per DISA call logic,
allowing the remote party to enter DISA mode with CLI Authentication-Multiple calls as
authentication method and level of access.
CLI Authentication-Single Call: The system will process the call as per DISA call logic, allowing
the remote party to enter DISA mode with CLI Authentication-Single call as authentication method
and level of access. Refer the feature description for Direct Inward System Access (DISA).
Operator: When the remote party answers the Call Back call, the system will route the call to the
Operator232.
232. 'Operator' is the station which is assigned to the Mobile port in the Trunk Feature Template. Refer Trunk Feature Template to know
more.
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To program Call Back on different port types, refer the relevant topics mentioned below:
For Call Back on Mobile Ports, refer the topic Configuring Mobile Trunks.
For Call Back on BRI Ports, refer the topic BRI Parameters under Configuring BRI Trunks.
For Call Back on T1E1 Ports, refer the topic Call Back on T1E1 Trunk Ports, under Configuring E1
Trunks and Configuring T1 Trunks.
For Call Back on SIP Trunks, refer the topic Configuring SIP Trunks.
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Call Budget
What's this?
Call Budget is a cost control feature that allows you to keep a tab on the total cost of phone call made by extension
users.
With this feature, each extension can be allotted a 'budget' limit for outgoing calls, which is automatically reloaded
at the start of every month.
Long distance calls form a major part of the increased cost of telephone calls. Though excessive use or misuse of
long distance dialing can be restricted using Toll Control, there may be extension users whose nature of work
requires them to make long distance calls. Instead of denying them the facility, their telephone bill can be limited to
a certain amount using Call Budget.
With a Call Budget allotted to the extension, the user is free to make calls as long as s/he does not cross the budget
limit. Once the user exceeds the budget limit, the extension can be denied access to long distance dialing.
The extension user can be assigned a fresh budget, after which s/he can resume making long distance calls.
Call Budget can be enabled on all the extensions as well as on selected extensions. Each extension can be
assigned a different amount depending on user requirement.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic License Management to know more.
How it works
When an extension allotted Call Budget makes a call,
The system checks the current call budget amount of the extension.
If the consumed amount is within the budget limit allotted to the extension,
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The system allows the extension to make the call as per the Toll Control Levels assigned to it.
After the call ends, the system calculates and adds the call amount to the extension's account. Thus it
calculates and updates the total cost of calls made from the phone.
If the consumed amount exceeds the budget limit allotted to the extension,
The system allows the extension to make the call as per the Toll Control-Call Budget Consumed
assigned to the extension.
After the call ends, the system calculates and adds the call amount to the extension's account.
Until a new Call Budget is allocated to the extension user, the extension user can make calls only as per
Toll Control assigned for the Call Budget Consumed state.
Once a new Call Budget is allocated, the extension user can make calls as per the Toll Control assigned
to the extension.
If the budget exceeds anytime during the month, and if no fresh budget amount is allotted, the system
allows calls to be made as per the Allowed and Denied List of Toll Control-Call Budget Consumed till the
end of the month. From the 1st day of the following month, the system automatically reloads the budget
amount. The extension can now make calls.
The Call Budget allotted to extension is valid for one month. The system automatically reloads the budget
at the start of every month.
The budget amount can be changed or allotted afresh to extensions from the System Administrator (SA)
mode, at any time. The Call Budget allotted by the SA will be reloaded in the following month.
Call Budget is not based on real time (online) call cost calculation. The ETERNITY calculates the call
cost only after the call has ended.
So, if the Call Budget allotted to an extension user gets exhausted in the middle of a call, the call will
not get disconnected, though the budget exceeds. To prevent this from occurring, the System Engineer
may program the Call Duration Control (CDC) feature.
Call Budget is dependent on precise Call Cost Calculation. So, SMDR parameters and long distance
codes must be programmed properly to prevent errors in calculation.
This feature works independent of any Call Accounting Software (CAS) installed with the ETERNITY.
The ETERNITY will calculate cost of phone calls made by extension phones even when no call budget
is allocated233.
How to configure
The working of this feature is controlled by three parameters: Call Budget flag, Toll Control-Call Budget
Consumed and Preset Call Budget Amount (this parameter is applicable only for the Hotel Mode).
These parameters can be programmed using Jeeves and Telephone.
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Refer the topic Station Basic Feature Template for detailed instructions for programming a feature in the template
and assigning templates to extensions.
Scroll to Preset Call Privilege and change the Preset Call Budget Amount to the required value.
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The amount programmed as Preset Call Budget is to be considered as the local currency.
At the time of installation, when the SE selects the Region Code (country code) and defaults the
system, the related Currency Code is applied.
The currency symbol will not be displayed on the Operator's phone, on account of the limited number of
characters that can be displayed.
The local currency symbol will appear at the relevant places in the outgoing SMDR reports.
How to use
Call Budget amount can be allotted to extensions from the System Administrator mode, using Jeeves or by dialing
SA commands from an extension phone.
Click Extension.
Now, Search Extension on which you want to set this feature. You search by entering either Extension
Number or the Extension Name.
Click Submit.
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In Allot Call Budget, enter the amount you want to assign to the user as budget limit for outgoing calls.
To re-assigning a new amount before the previous balance is consumed, make sure you add the
available balance to the new amount. Enter this amount in Allot a Call Budget.
For example, if you have allotted an amount is Rs.1000 and the consumed amount is Rs.600. The
available balance is Rs.400. Now, if you want to assign a new amount of Rs.500. In Allot a Call Budget
you must enter 900 (Balance + New = 400 + 500).
The Allotted Amount/Used displays the amount allotted to the user as well as the call budget amount
consumed by the user for making outgoing calls.
To allot call budget to another extension, follow the same instructions as above.
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1569
Amount: In this type of Call Budget, a fixed amount is assigned to the trunk. By default the amount of
999999 (to be considered in the local currency) is set as Call Budget Amount on trunks. With Amountbased Call Budget you can control the actual expense incurred on making calls from a trunk.
Minutes: In this type of Call Budget, a fixed number of Minutes are assigned to the trunk. By default,
999999 minutes are assigned as Call Budget Minutes on trunks. This type of Call Budget is useful when
the Service Provider offers 'Free' minutes. For example, the Service Provider allows the customer to make
calls for the first 1000 minutes every month. This offer can be availed of by programming Minutes-based
Call Budget on the trunk port.
Number of Calls: In this type of Call Budget, you can define the maximum number of calls that can be
made from a trunk. By default, the maximum number of Call Budget - Calls is set to 9999 calls on the
trunks. This type of Call Budget is useful when the Service Provider offers a certain number of free calls or
a certain number of free calls for a fixed period. For instance, the Service Provider offers 150 free calls per
month.
With a Call Budget allotted to a trunk, the users can make calls from the trunk as long as the budget limit set for the
trunk (that is, the Amount or Minutes or the maximum number of Calls) is not crossed. Once the budget limit is
exceeded, the trunk gets disabled automatically and no outgoing calls are allowed to be made from the trunk.
The consumed Budget can be reset, after which it becomes functional again and allows outgoing calls to be made.
The consumed Call Budget can be reset manually, that is, anytime, as required/desired, or on a scheduled date
either daily or on a particular date of the month.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic License Management to know more.
How it works
Call Budget can be enabled on trunk port types - CO, Mobile, SIP, BRI, T1E1PRI- all at once or on selected trunk
port types from among them. Each trunk can be assigned a different Call Budget, depending on the requirement of
the users.
When Call Budget is enabled on a trunk port, for each outgoing call,
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The system checks the type of Call Budget set on the trunk - Amount, Minutes or number of Calls.
It checks the Call Budget consumed.
When Amount-based Call Budget is selected, the Amount should be assigned to the trunk.
At the end of each outgoing call made from the trunk, the system will calculate the cost of the call on the
basis of the Pulse Rate Type programmed. The system will thus calculate the total amount consumed after
the end of each call. Refer the topic "Call Cost Calculation" to know more.
When Minutes-based Call Budget is set, the total minutes for which calls will be allowed from the trunk port
must be defined.
With the number of Minutes defined, at the end of each call, the system will calculate the duration of the
call on the basis of the units programmed in the Pulse Rate. The system will calculate the consumed
minute on the basis of the duration of the call. Refer the topic "Call Cost Calculation" to know more.
When the Call Budget is based on 'Number of Calls', the maximum number of calls to be allowed from the
trunk port is to be defined.
With the number of calls programmed, the system will maintain a count for the number of matured
outgoing calls made from that trunk port.
Thus for each matured call, the Number of Calls-Count is incremented, irrespective of the actual duration
of the matured call.
When the assigned 'cost' or 'minutes' or 'number of calls' assigned to trunk is exhausted, ETERNITY will:
The consumed Call Budget Amount/Minutes/Calls can be reset manually at any time from the System
Administrator mode or the System Engineer mode or can be programmed to be automatically reset either
daily or on a particular date of the month.
The current Call Budget Amount/Minutes/Calls limit can be changed from the System Administrator (SA)
mode, at any time. If scheduled reset of consumed Call Budget is programmed, then the Call Budget
allotted by the SA will be reloaded on the scheduled date.
Once a new Call Budget is allocated to the trunk, outgoing call facility is resumed on the trunk.
Call Budget on Trunks is not based on real time (online) call cost calculation. The ETERNITY
calculates the call cost only after the call has ended.
If the Call Budget allotted to a Trunk Port gets exhausted in the middle of a call, the call will not
disconnected, though the budget is exceeded.
Call Budget on Trunks is dependent on precise Call Cost Calculation. So, SMDR parameters and long
distance codes must be programmed properly to prevent errors in calculation.
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This feature works independent of any Call Accounting Software (CAS) installed with the ETERNITY.
The ETERNITY will calculate cost of phone calls made by the trunks even when no call budget is
allocated235.
How to configure
Call Budget on Trunks is to be programmed in the Trunk Port Parameters of the trunk type on which you want to
enable this feature. This can be done using Jeeves as well as a telephone
Call Budget parameters must be programmed in the Port Parameters page of the trunk type you want to
configure. For instance, to configure Call Budget on a CO Trunk,
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Call Budget: If you want to enable Call Budget on Trunk feature, configure the following parameters for
this CO trunk port:
Type: Select the type of Call Budget on Trunk, that is, Amount or Minutes or Calls to be applied on this
CO trunk port. By default, no Call Budget type is selected.
Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 999999.
Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 999999.
Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By default
the number of calls is set to 9999.
Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan suggests
so.
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You may program the same Call Budget parameters as listed above for other trunk types:
Click Mobile Port Parameters to program Call Budget on Mobile Ports. Click the Advance button on
this page to reach Call Budget parameters.
Similarly, click Port Parameters under T1E1 Configuration to program Call Budget on T1E1PRI
trunks. Click the Advance button on this page to reach Call Budget parameters.
Click BRI Parameters under BRI Configuration to program Call Budget on BRI trunks. Click the
Advance button on this page to reach Call Budget parameters.
Click VoIP Configuration, click SIP Trunk Parameters to program Call Budget on SIP trunks. Click
the Advance button on this page to reach Call Budget parameters.
The consumed Call Budget on trunk can be reset from the System Engineer mode as well as the
System Administrator mode manually at any time, referred to as Manual Reset.
Manual Reset of Call Budget on Trunks by the System Engineer can be done either from Jeeves or
using a Telephone.
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You may perform manual reset of the consumed Call Budget on the above listed trunk types from the
Status page of each of these trunk types.
Open the Status page under CO Configuration. Select the Reset Consumed Amount/Minutes/
Calls check box of the CO port for which you want to reset the consumed Call Budget.
Similarly, open Status page under Mobile Configuration to enable the same parameter Reset
Consumed Amount/Minutes/Calls of the Mobile port for which you want to reset the consumed Call
Budget.
To manually reset the consumed Call Budget on T1E1PRI trunks, open Status page under T1E1
Configuration. Enable the parameter Reset Consumed Amount/Minutes/Calls for the T1E1 port for
which you want to reset the consumed Call Budget.
To manually reset the consumed Call Budget on BRI trunks, open Status page under BRI
Configuration, and enable the parameter Reset Consumed Amount/Minutes/Calls for the BRI port
for which you want to reset the consumed Call Budget.
Open Status page under VoIP Configuration and enable the parameter Reset Consumed Amount/
Minutes/Calls to manually reset consumed Call Budget on the desired SIP trunks.
Enter SE mode.
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1 for Amount
2 for Minutes
3 for Number of Calls
By default, Budget Type is None.
To program Call Budget Amount on CO, dial:
3302-1-CO-Budget Amount to program amount for a single trunk port.
3302-2-CO-CO-Budget Amount to program the same amount for a range of trunk ports.
3302-*-Budget Amount to program the same amount for all trunk ports.
Where,
CO is the Software Port number of the CO port from 001 to 128.
Budget Amount is of 6 digits max. Use leading zeros if amount to be programmed has fewer than 6
digits.
By default Budget Amount is 999999.
To program Call Budget Minutes on CO, dial:
3303-1-CO-Minutes to program minutes for a single trunk port.
3303-2-CO-CO-Minutes to program the same minutes for a range of trunk ports.
3303-*-Minutes to program the same minutes for all trunk ports.
Where,
CO is the Software Port number of the CO port from 001 to 128.
'Minutes' is of 6 digits max. Use leading zeros if Minutes to be programmed has less than 6 digits.
By default, Minutes is 999999.
To program Call Budget - Number of Calls on CO, dial:
3309-1-CO-Number of calls to program number of calls for a single trunk port.
3309-2-CO-CO-Number of calls to program the same number of calls for a range of trunk ports.
3309-*-Number of calls to program the same number of calls for all trunk ports.
Where,
CO is the Software Port number of the CO port from 001 to 128.
Number of Calls is of 4 digits from 0001 to 9999. Use leading zeros if number of calls to be
programmed has fewer than 4 digits.
By default, Number of calls is 9999.
To program Call Budget Reset Mode for CO, dial:
3304-1-CO-Call Budget Reset Mode to program reset mode for a single trunk port.
3304-2-CO-CO-Call Budget Reset Mode to program the same reset mode for a range of trunk ports.
3304-*-Call Budget Reset Mode to program the same reset mode for all trunk ports.
Where,
CO is the Software Port number of the CO port from 001 to 128.
Reset Mode is
1 for Scheduled reset
2 for Manual reset
By default, Call Budget Reset Mode is Scheduled.
To program the Date for Scheduled Reset mode, dial:
3305-1-CO-Date to program date for a single trunk port.
3305-2-CO-CO-Date to program the same date for a range of trunk ports.
3305-*-Date to program the same date for all trunk ports
Where,
CO is the Software Port number of the CO port from 001 to 128.
Date is
01 to 31 for Scheduled date to reset every month.
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Where,
Mobile is the number of the Mobile software port from 01 to 64.
Number of Calls is of 4 digits from 0001 to 9999. Use leading zeros if number of calls to be
programmed has fewer than 4 digits.
By default, Number of calls is 9999.
To program Call Budget Reset Mode for Mobile trunk port, dial:
8022-1-Mobile-Call Budget Reset Mode to program reset mode for a single trunk port.
8022-2-Mobile-Mobile-Call Budget Reset Mode to program the same reset mode for a range of trunk
ports.
8022-*-Call Budget Reset Mode to program the same reset mode for all trunk ports.
Where,
Mobile is the number of the Mobile software port from 01 to 64.
1 for Scheduled reset
2 for Manual reset
By default, Call Budget Reset Mode is Scheduled.
To program the Date for Scheduled Reset mode for Mobile trunk port, dial:
8023-1-Mobile-Date to program reset date for a single trunk port.
8023-2-Mobile-Mobile-Date to program the same reset date for a range of trunk ports.
8023-*-Date to program the same reset date for all trunk ports.
Where,
Mobile is the number of the Mobile software port from 01 to 64.
Date is
01 to 31 for Scheduled date to reset every month.
00 for Scheduled reset Daily.
By default, Reset date is 1st. of every month.
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Exit SE mode.
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Enter SE mode.
To manually reset consumed Call Budget of CO trunks, dial:
3306-1-CO to manually reset a single trunk port.
3306-2-CO-CO to manually reset a range of trunk ports.
3306-* to manually reset all trunk ports.
Where,
CO is the Software Port number of the CO port from 001 to 128.
To manually reset consumed Call Budget of SIP trunks, dial:
7738-1-SIP to reset a single trunk.
7738-2-SIP-SIP to reset a range of trunks.
7738-* to reset all trunks.
Where,
SIP is the number of the software port of the SIP Trunk from 01 to 32.
To manually reset consumed Call Budget of Mobile trunks, dial:
8024-1-Mobile to reset a single trunk port.
8024-2-Mobile-Mobile to reset a range of trunk ports.
8024-* to reset all trunk ports.
Where,
Mobile is the number of the Mobile software port from 01 to 64.
To manually reset consumed Call Budget of BRI trunks, dial:
6219-1-BRI to reset a single trunk.
6219-2-BRI-BRI to reset a range of trunks.
6219-* to reset all trunks.
Where,
BRI is the number of the BRI software port from 01 to 32.
To manually reset consumed Call Budget of T1E1 trunks, dial:
6140-1-T1E1 to reset a single trunk port.
6140-2-T1E1-T1E1 to reset a range of trunk ports.
6140-* to reset all trunk ports.
Where,
T1E1is the number of the T1E1 software port from 1 to 8.
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Exit SE mode.
Call Chaining
What's this?
Call Chaining is when an external/internal call transferred by the Operator to another extension or external number
is made to return to the Operator's extension after the conversation between the caller and the extension/external
number to which it is transferred has ended.
Call Chaining is useful situations where the Operator intervention is required after the transferred call has ended.
For instance:
The caller needs to take an appointment or requires some information from the Operator after talking to the
desired extension.
A marketing executive who calls his supervisor to consult on a technical problem needs to be informed
about his travel itinerary and ticket booking by the Operator. The Operator can transfer the call to the
supervisor, and use Call Chaining to retrieve the call once the conversation has ended to give the
information to the executive.
How it works
A is an External Caller
B is an extension.
If A disconnects the call with B, the call will be released. It will not return to the Operator.
If the Operator is busy, A will be played music on hold for the duration of the Call Park Release Timer.
If the Operator is busy and the Timer elapses, the call will be released.
Call Chaining can be performed when call is transferred from any extension to another extension or external
number.
How to configure
The only programming involved in the functioning of this feature is assigning this feature to a DSS key which has
LED and programming, if necessary, the Call Park Release Timer.
Refer the topic DSS Keys Programming and Call Park for instructions.
1583
How to use
For EON & Extended IP Phone Users
1584
How it works
For this feature to work,
you must get the tariff details from the Service Providers and configure the same.
determine the outgoing trunk for the calls according to the type of calls, namely, local calls, national calls or
international calls.
When the call is made from a trunk, the system checks the Call Cost Calculation Pulse Rate Option, 1 to
4 assigned to the trunk, on the basis of the Call Cost Calculation Time Schedule configured for the
outgoing call.
Each Call Cost Calculation Pulse Rate option contains a Pulse Rate Type for the Pulse Rate, which is
assigned in the Area Code Table.
The system matches the Number dialed by the extension user with the Area Code Table configured in the
system. When the area code matches with an entry in the table, the system obtains the Pulse rate type
configured for the Call Cost Calculation Pulse Rate option assigned to the trunk.
This Pulse Rate type obtained from the Area Code Table is checked in the Pulse Rate table to obtain
the corresponding duration and cost to be applied for the call duration. The Pulse Rate Table may be the
Normal Pulse Rate Table or Discounted Pulse Rate Table, depending on the day of the call. The
system uses the built-in RTC to determine the day.
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The Pulse Rate Type applied (duration and cost) is divided into two parts for each time zone:
First unit.
Additional units.
Number of Units is derived from the pulse rate at the time of the call and duration of the call. System
acquires the pulse rate type and call duration with the help of in-built RTC.
Total Units = First Unit + Additional Unit.
If the call duration is less than the pulse rate of the first unit then additional unit is zero.
Call Units = (Call duration in seconds)/(Pulse rate in seconds).
The system applies the rates as configured in the Normal Pulse Rate Table for all the days, except when it
detects a day configured in the Discounted Pulse Rate Schedule. These are special days when special
Tariffs are offered.
For the days configured in the Discounted Pulse Rate Schedule as special days, the system checks the
the duration and cost of the First unit and the Additional units configured in the Discounted Pulse Rate
Table.
ETERNITY uses the Cost of the Call for SMDR. This cost is deducted from the Call Budget (Amount), if allotted to
the trunk and also from the Call Budget, if assigned to the extension users.
The logic for call cost calculation is explained with the help of an example:
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Assign Trunk Feature Template 1 to CO-1. Configuring the Call Cost Calculation parameters in the
Trunk Feature Template as follows:
Template
No.
Call
Cost
Calculat
ion
Pulse
Rate
Option
001
T1
T2
T3
Start Time
End Time
Start Time
End Time
Start Time
End Time
HH
MM
HH
MM
HH
MM
HH
MM
HH
HH
00
00
22
00
22
01
23
59
MM
MM
Index
Area
Code
Name
001
26
002
09
Ignore
Digit
Count
Option - 2
Option - 3
Option - 4
Local
03
06
09
10
Mobile
05
03
07
08
Time Zone T2
Time Zone T3
Time Zone T4
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
180
180
60
30
90
30
120
60
02.00
02.00
02.00
02.00
02.00
02.00
02.00
02.00
300
300
300
300
300
300
300
300
01.00
01.00
01.00
01.00
01.00
01.00
01.00
01.00
30
30
30
30
30
30
30
30
01.00
01.00
01.00
01.00
01.00
01.00
01.00
01.00
45
45
45
45
45
45
45
45
01.00
01.00
01.00
01.00
01.00
01.00
01.00
01.00
180
180
180
180
180
180
180
180
03.00
03.00
03.00
03.00
03.00
03.00
03.00
03.00
Durations)
Cost
01
02
03
04
05
32
Duration (sec)
Cost
Duration (sec)
Cost
Durations)
Cost
Duration (sec)
Cost
Duration (sec)
Cost
Duration (sec)
Cost
With this configuration, ETERNITY will calculate the call cost as follows:
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An Outgoing Call is made by an extension user, to the number 2630555 through the trunk, CO-1 at
20:10 hours. ETERNITY will check the Call Cost Calculation parameters assigned on the trunk and
determine the Time Zone as per time of the call.The system will also check the corresponding Pulse
Rate Option configured on the trunk.
In this example, Time Zone for CO-1 at 20:10 Hours would be Time Zone 1, and the Pulse Rate
Option for CO-1 is 1.
ETERNITY will match the dialed number 2630555 in the Area Code table. A best match is found
with the entry configured at index 001 in the Area Code Table.
As per the Area Code Table, the Pulse Rate Type 03 is programmed in Pulse Rate Option 1 for
the matching entry (at Index 001).
(However, if CO-1 would have been assigned Pulse Rate Option 2, the Pulse Rate type 06 would
have been selected as shown in Area Code Table)
Finally, for Pulse Rate Type 03 ETERNITY will check the Normal Pulse Rate Table. ETERNITY will
consider the Cost for the First Unit as 01.00 (As Rs. or $ as per applicable currency) for the duration
of 30 seconds and for the additional unit also, the cost will be considered as 01.00 for the duration
of 30 seconds. This data will be used for calculating the total cost of call based on the total duration
of the call.
Similarly, when there are Special Tariffs offered on certain days, the system will check the Discounted Pulse Rate
Schedule and the Discounted Pulse Rate Table.
The days on which the special rates are to be applied must be configured in the Discounted Pulse Rate Schedule.
The duration and cost of the First unit and the Additional units must be configured in the Discounted Pulse Rate
Table according to the Time Zones.
How to configure
To be able to use Call Cost Calculation, you must do the following:
Define the Unit and Service Charge on the basis of which call cost is to be calculated.
Assign Call Cost Calculation Pulse Rate Option and the Call Cost Calculation Schedule on the trunks on
which you want to apply this feature.
Configure the Pulse Rate Types for the Pulse Rate Option you assign to the trunk.
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For Normal Days configure the pulse rate in the Normal Pulse Rate Table.
For Special days configure the pulse rate in the Discounted Rate Table. If you configure the Discounted
Pulse Rate Table, you must also configure the Discounted Pulse Rate schedule.
Service Charge
By default, no Service Charge is applied on call cost by the system. Service Charge on call cost is generally applied
in Hotels and other organization which charge users for the calls made by them.
If you want to apply Service Charge,
In the Service Charge Type field, select the type of service charge you want to apply from the options:
Fixed for a call: A fixed amount is added as service charge to every call regardless of the cost of that
call.
If you select this option, you must define the Amount to be added as service charge in the Specify
Service Charge field.
per Unit: service charge is added to each unit of the call. For example, if a call worth 10 units was
made, the service charge will be applied on each of the 10 units, instead of the one time service
charge as in the case of Fixed service charge.
If you select this option, you must define the amount to be added as service charge on each unit in the
Specify Service Charge field.
Percentage of call cost: A percentage of the cost of the call is added as a service charge for that call.
If you select this option, you must define the percentage in the Specify Service Charge in % field
which appears.
1589
To assign Call Cost Calculation Pulse Rate Option and configure the Call Cost Calculation Schedule
on the trunks, go to Configuration, and configure these in the Trunk Feature Template assigned to the
different trunk port types. See Configuring CO Trunks, Configuring Mobile Trunks, Configuring BRI
Trunks Configuring E1 Trunks, Configuring T1 Trunks, Configuring Trunksand Configuring SIP
Trunks under Configuring ETERNITY for instructions.
Configure the Pulse Rate Type with rates for the First Unit and the Additional Unit.
Generally service providers offer different call rates for different types of calls, for example: local, national,
international. You can configure different Pulse Rate Types for different types of calls. Thus, each Pulse
Rate Type can have different rates for the First and the Additional unit.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
will need to configure the Call Cost Calculation Schedule for the trunk, by dividing the day into Time
Zones, from 1 to 4, as required, to match the time of the pulse rates offered by your service provider.
1590
If you have configured Time Zones for the Call Cost Schedule on a trunk, you may define the different
Pulse Rate Types for each Time Zone.
Configure the Discounted Pulse Rate Type with rates for the First Unit and the Additional Unit, as
required.
1591
1592
In the Area Code column, enter the number strings (prefix) of the Area Codes, country codes, local
numbers. You can configure as many as 999 area codes in the table.
Do not configure the Ignore Digit Count. This parameter is relevant only for Service Provider Based Least
Cost Routing.
Different service providers offer different pulse rates for the same type of calls. To take care of this,
ETERNITY allows you to assign different Pulse Rate Options for each area code.
For each area code, configure the Pulse Rate to be followed for the desired Pulse Rate Options in the
Pulse Rate Type for Pulse Rate (Option 1 to 4) column.
For Default Area Code Table for the Region-USA at end of the chapter.
1593
To program the unit charge for additional units when 12/16 KHz metering is used, dial:
2601-Unit Charge for Additional Unit
Where,
Unit charge is the amount in XX.XX format in any currency.
By default, Unit charge for additional unit is Rs.1.10.
Example: To program unit change for first unit to Rs. 1.50, dial 2600-0150
To program unit charge for additional unit to US$0.75, dial 2601-0075
Service Charge
To select service charge type, dial:
2602-Service Charge Type
Where,
Service Charge Type
Meaning
Number of Units
Number of Units is derived from the pulse rate at the time of the call and duration of the call. System
acquires the pulse rate type and call duration with the help of in-built RTC.
Total Units = First Unit + Additional Unit.
If the call duration is less than the pulse rate of the first unit then additional unit is zero.
Call Units = (Call duration in seconds)/(Pulse rate in seconds).
Assign parameters; 'CCC Pulse Rate Option' and program 'CCC Time Schedule' for the Trunk Feature Template
which is assigned to the specific trunk used for OG calls.
1594
Program four 'CCC Time Schedule', T1, T2,T3 and T4. Program Start Time and End Time for each.
Pulse Rate
Type
01
02
Time Zone 2
Time Zone 3
Time Zone 4
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
180.00
180.00
180.00
180.00
180.00
180.00
180.00
180.00
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
Cost
Duration
0.00
0.00
0.00
0.00
0.00
0.00
0.00
0.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
Cost
:
32
Time Zone 1
To program duration of first unit for a pulse rate type on normal days, dial:
2607-Pulse Rate Type-Time Zone-Duration of First Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Duration of First Unit is from 000.00 to 999.99.
Pulse Rate
Type
01
Duration
Cost
Time Zone 1
Time Zone 2
Time Zone 3
Time Zone 4
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
180.00
180.00
180.00
180.00
180.00
180.00
180.00
180.00
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1595
Pulse Rate
Type
02
03
04
05
06
07
08
09
10
11
12
32
Time Zone 1
Time Zone 2
Time Zone 3
Time Zone 4
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
300.00
300.00
300.00
300.00
300.00
300.00
300.00
300.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
90.00
90.00
90.00
90.00
90.00
90.00
90.00
90.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
120.00
120.00
120.00
120.00
120.00
120.00
120.00
120.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
30.00
30.00
30.00
30.00
30.00
30.00
30.00
30.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
30.00
30.00
30.00
30.00
30.00
30.00
30.00
30.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
16.00
16.00
16.00
16.00
16.00
16.00
16.00
16.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
3.30
3.30
3.30
3.30
3.30
3.30
3.30
3.30
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
2.10
2.10
2.10
2.10
2.10
2.10
2.10
2.10
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
1.70
1.70
1.70
1.70
1.70
1.70
1.70
1.70
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
2.10
2.10
2.10
2.10
2.10
2.10
2.10
2.10
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
0.00
0.00
0.00
0.00
0.00
0.00
0.00
0.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
Cost
Duration
0.00
0.00
0.00
0.00
0.00
0.00
0.00
0.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
Duration
To program the cost of first unit of a pulse rate type for normal days, dial:
2609-Pulse Rate Type-Time Zone-Cost of First Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Cost of first Unit is from XX.XX.
To program the cost of additional unit of a pulse rate type for normal days, dial:
2610-Pulse Rate Type-Time Zone-Cost of Additional Unit
Where,
1596
2611
CCC Pulse
Rate Type
01
02
03
04
05
06
07
08
09
10
11
12
Time Zone 1
Time Zone 2
Time Zone 3
Time Zone 4
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
180.00
180.00
180.00
180.00
180.00
180.00
180.00
180.00
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
300.00
300.00
300.00
300.00
300.00
300.00
300.00
300.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
90.00
90.00
90.00
90.00
90.00
90.00
90.00
90.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
120.00
120.00
120.00
120.00
120.00
120.00
120.00
120.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
30.00
30.00
30.00
30.00
30.00
30.00
30.00
30.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
30.00
30.00
30.00
30.00
30.00
30.00
30.00
30.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
16.00
16.00
16.00
16.00
16.00
16.00
16.00
16.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
3.30
3.30
3.30
3.30
3.30
3.30
3.30
3.30
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
2.10
2.10
2.10
2.10
2.10
2.10
2.10
2.10
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
1.50
1.50
1.50
1.50
1.50
1.50
1.50
1.50
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
2.10
2.10
2.10
2.10
2.10
2.10
2.10
2.10
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
0.00
0.00
0.00
0.00
0.00
0.00
0.00
0.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
Duration
Cost
Duration
Cost
Duration
Duration
1597
CCC Pulse
Rate Type
16
Time Zone 1
Time Zone 2
Time Zone 3
Time Zone 4
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
First
Unit
Add.
Unit
Duration
0.00
0.00
0.00
0.00
0.00
0.00
0.00
0.00
Cost
1.10
1.10
1.10
1.10
1.10
1.10
1.10
1.10
To program duration for a first unit of a pulse rate type on Special Days, dial:
2612-Pulse Rate Type-Time Zone-Duration of First Unit
Where,
Pulse rate type is from 01 to 32.
Time Zone from 1 to 4.
Duration of First Unit is from 000.00 to 999.99.
To program duration for additional unit for a pulse rate type on Special Days, dial:
2613-Pulse Rate Type-Time Zone-Duration of Additional Unit
Where,
Pulse rate type is from 01 to 32.
Time Zone from 1 to 4.
Duration of Additional Unit is from 000.00 to 999.99.
To program the cost of first unit of a pulse rate type for holidays, dial:
2614-Pulse Rate Type-Time Zone-Cost of First Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Cost of First Unit is from XX.XX.
To program the cost of additional unit of a pulse rate type for Special Days, dial:
2615-Pulse Rate Type-Time Zone-Cost of Additional Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Cost of Additional Unit is from XX.XX.
1598
Special Days
Day
Sun
Mon
Tue
Wed
Thu
Fri
Sat
Code
Meaning
Day
26-01
15-08
02-10
Blank
Blank
1599
Area Code
Area Name
1201
NJ
1202
DC
1203
CT
1204
Manitoba
1205
AL
1206
WA
1207
ME
1208
ID
1209
CA
10
1210
TX
11
1212
NY
12
1213
CA
13
1214
TX
14
1215
PA
15
1216
OH
16
1217
IL
17
1218
MN
18
1219
IN
19
1224
IL
20
1225
LA
21
1226
Ontario
22
1228
MS
23
1229
GA
24
1231
MI
1600
Index
Area Code
Area Name
25
1234
OH
26
1239
FL
27
1240
MD
28
1242
Bahamas
29
1246
Barbados
30
1248
MI
31
1250
BC
32
1251
AL
33
1252
NC
34
1253
WA
35
1254
TX
36
1256
AL
37
1260
IN
38
1262
WI
39
1264
Anguilla
40
1267
PA
41
1268
Antigua
42
1269
MI
43
1270
KY
44
1276
VA
45
1281
TX
46
1284
BVI
47
1289
Ontario
48
1301
MD
49
1302
DE
50
1303
CO
51
1304
WV
52
1305
FL
53
1306
Saskatchewan
54
1307
WY
55
1308
NE
56
1309
IL
57
1310
CA
58
1312
IL
1601
Index
Area Code
Area Name
59
1313
MI
60
1314
MO
61
1315
NY
62
1316
KS
63
1317
IN
64
1318
LA
65
1319
IA
66
1320
MN
67
1321
FL
68
1323
CA
69
1325
TX
70
1330
OH
71
1331
IL
72
1334
AL
73
1336
NC
74
1337
LA
75
1339
MA
76
1340
USVI
77
1345
Cayman
78
1347
NY
79
1351
MA
80
1352
FL
81
1360
WA
82
1361
TX
83
1386
FL
84
1401
RI
85
1402
NE
86
1403
Alberta
87
1404
GA
88
1405
OK
89
1406
MT
90
1407
FL
91
1408
CA
92
1409
TX
1602
Index
Area Code
Area Name
93
1410
MD
94
1412
PA
95
1413
MA
96
1414
WI
97
1415
CA
98
1416
Ontario
99
1417
MO
100
1418
Quebec
101
1419
OH
102
1423
TN
103
1424
CA
104
1425
WA
105
1430
TX
106
1432
TX
107
1434
VA
108
1435
UT
109
1438
Quebec
110
1440
OH
111
1441
Bermuda
112
1443
MD
113
1450
Quebec
114
1456
NANParea
115
1469
TX
116
1473
Grenada
117
1478
GA
118
1479
AR
119
1480
AZ
120
1484
PA
121
1500
NANParea
122
1501
AR
123
1502
KY
124
1503
OR
125
1504
LA
126
1505
NM
1603
Index
Area Code
Area Name
127
1506
NewBrunswick
128
1507
MN
129
1508
MA
130
1509
WA
131
1510
CA
132
1512
TX
133
1513
OH
134
1514
Quebec
135
1515
IA
136
1516
NY
137
1517
MI
138
1518
NY
139
1519
Ontario
140
1520
AZ
141
1530
CA
142
1540
VA
143
1541
OR
144
1551
NJ
145
1559
CA
146
1561
FL
147
1562
CA
148
1563
IA
149
1567
OH
150
1570
PA
151
1571
VA
152
1573
MO
153
1574
IN
154
1575
NM
155
1580
OK
156
1585
NY
157
1586
MI
158
1600
Canada
159
1601
MS
160
1602
AZ
1604
Index
Area Code
Area Name
161
1603
NH
162
1604
BC
163
1605
SD
164
1606
KY
165
1607
NY
166
1608
WI
167
1609
NJ
168
1610
PA
169
1612
MN
170
1613
Ontario
171
1614
OH
172
1615
TN
173
1616
MI
174
1617
MA
175
1618
IL
176
1619
CA
177
1620
KS
178
1623
AZ
179
1626
CA
180
1630
IL
181
1631
NY
182
1636
MO
183
1641
IA
184
1646
NY
185
1647
Ontario
186
1649
T&CIsland
187
1650
CA
188
1651
MN
189
1660
MO
190
1661
CA
191
1662
MS
192
1664
Montsrat
193
1670
CNMI
194
1671
GU
1605
Index
Area Code
Area Name
195
1678
GA
196
1682
TX
197
1684
AS
198
1700
NANParea
199
1701
ND
200
1702
NV
201
1703
VA
202
1704
NC
203
1705
Ontario
204
1706
GA
205
1707
CA
206
1708
IL
207
1709
Newfoundland
208
1710
US
209
1712
IA
210
1713
TX
211
1714
CA
212
1715
WI
213
1716
NY
214
1717
PA
215
1718
NY
216
1719
CO
217
1720
CO
218
1724
PA
219
1727
FL
220
1731
TN
221
1732
NJ
222
1734
MI
223
1740
OH
224
1754
FL
225
1757
VA
226
1758
St.Lucia
227
1760
CA
228
1762
GA
1606
Index
Area Code
Area Name
229
1763
MN
230
1765
IN
231
1767
Dominica
232
1769
MS
233
1770
GA
234
1772
FL
235
1773
IL
236
1774
MA
237
1775
NV
238
1778
BC
239
1779
IL
240
1780
Alberta
241
1781
MA
242
1784
St. V&G
243
1785
KS
244
1786
FL
245
1787
PrtoRico
246
1800
NANParea
247
1801
UT
248
1802
VT
249
1803
SC
250
1804
VA
251
1805
CA
252
1806
TX
253
1807
Ontario
254
1808
HI
255
1809
DomRepub
256
1810
MI
257
1812
IN
258
1813
FL
259
1814
PA
260
1815
IL
261
1816
MO
262
1817
TX
1607
Index
Area Code
Area Name
263
1818
CA
264
1819
Quebec
265
1828
NC
266
1829
DomRepub
267
1830
TX
268
1831
CA
269
1832
TX
270
1843
SC
271
1845
NY
272
1847
IL
273
1848
NJ
274
1850
FL
275
1856
NJ
276
1857
MA
277
1858
CA
278
1859
KY
279
1860
CT
280
1862
NJ
281
1863
FL
282
1864
SC
283
1865
TN
284
1866
NANParea
285
1867
Yukon
286
1868
Tri&Tob
287
1869
St. K&N
288
1870
AR
289
1876
Jamaica
290
1877
NANParea
291
1878
PA
292
1888
NANParea
293
1900
NANParea
294
1901
TN
295
1902
N Scotia
296
1903
TX
1608
Index
Area Code
Area Name
297
1904
FL
298
1905
Ontario
299
1906
MI
300
1907
AK
301
1908
NJ
302
1909
CA
303
1910
NC
304
1912
GA
305
1913
KS
306
1914
NY
307
1915
TX
308
1916
CA
309
1917
NY
310
1918
OK
311
1919
NC
312
1920
WI
313
1925
CA
314
1928
AZ
315
1931
TN
316
1936
TX
317
1937
OH
318
1939
PrtoRico
319
1940
TX
320
1941
FL
321
1947
MI
322
1949
CA
323
1951
CA
324
1952
MN
325
1954
FL
326
1956
TX
327
1970
CO
328
1971
OR
329
1972
TX
330
1973
NJ
1609
Index
Area Code
Area Name
331
1978
MA
332
1979
TX
333
1980
NC
334
1985
LA
335
1989
MI
336
0117
Kazkhstan
337
01120
Egypt
338
01127
South Africa
339
01130
Greece
340
01131
Netherlands
341
01132
Belgium
342
01133
France
343
01134
Spain
344
01136
Hungary
345
01139
VaticanCity
346
01140
Romania
347
01141
Switzerland
348
01143
Austria
349
01144
UK
350
01145
Denmark
351
01146
Sweden
352
01147
Norway
353
01148
Poland
354
01149
Germany
355
01151
Peru
356
01152
Mexico
357
01153
Cuba
358
01154
Argentine
359
01155
Brazil
360
01156
Chile
361
01157
Colombia
362
01158
Venezuela
363
01160
Malaysia
364
01161
Australia
1610
Index
Area Code
Area Name
365
01162
Indonesia
366
01163
Philippines
367
01164
NZ
368
01165
Singapore
369
01166
Thailand
370
01181
Japan
371
01182
Korea
372
01184
VietNam
373
01186
China
374
01190
Turkey
375
01191
India
376
01192
Pakistan
377
01193
Afghanistan
378
01194
Sri Lanka
379
01195
Myanmar
380
01198
Iran
381
011212
Morocco
382
011213
Algeria
383
011216
Tunisia
384
011218
Libya
385
011220
Gambia
386
011221
Senegal
387
011222
Mauritania
388
011223
Mali
389
011224
Guinea
390
011225
IvoryCoast
391
011226
BurkinaFaso
392
011227
Niger
393
011228
Togolese
394
011229
Benin
395
011230
Mauritius
396
011231
Liberia
397
011232
SierraLeone
398
011233
Ghana
1611
Index
Area Code
Area Name
399
011234
Nigeria
400
011235
Chad
401
011236
CenAfrica
402
011237
Cameroon
403
011238
CapeVerde
404
011239
SaoTome
405
011240
Equtl_Guinea
406
011241
Gabonese
407
011242
Congo
408
011243
CongoDem
409
011244
Angola
410
011245
GuineaBissa
411
011246
DiegoGarcia
412
011247
Ascension
413
011248
Seychelles
414
011249
Sudan
415
011250
Rwandese
416
011251
Ethiopia
417
011252
SomalianRep
418
011253
Djibouti
419
011254
Kenya
420
011255
Tanzania
421
011256
Uganda
422
011257
Burundi
423
011258
Mozambique
424
011260
Zambia
425
011261
Madagascar
426
011262
Reunion
427
011263
Zimbabwe
428
011264
Namibia
429
011265
Malawi
430
011266
Lesotho
431
011267
Botswana
432
011268
Swaziland
1612
Index
Area Code
Area Name
433
011269
Comoros
434
011290
StHelena
435
011291
Eritrea
436
011297
Aruba
437
011298
FaroeIsland
438
011299
Greenland
439
011350
Gibraltar
440
011351
Portugal
441
011352
Luxembourg
442
011353
Ireland
443
011354
Iceland
444
011355
Albania
445
011356
Malta
446
011357
Cyprus
447
011358
Finland
448
011359
Bulgaria
449
011370
Lithuania
450
011371
Latvia
451
011372
Estonia
452
011373
Moldova
453
011374
Armenia
454
011375
Belarus
455
011376
Andorra
456
011377
Monaco
457
011378
SanMarino
458
011379
VaticanCity
459
011380
Ukraine
460
011381
Yugoslavia
461
011385
Croatia
462
011386
Slovenia
463
011387
Bosnia
464
011389
Macedonia
465
011420
Czech Repub
466
011421
Slovakia
1613
Index
Area Code
Area Name
467
011423
Liechtenstein
468
011500
Falklands
469
011501
Belize
470
011502
Guatemala
471
011503
El Salvador
472
011504
Honduras
473
011505
Nicaragua
474
011506
CostaRica
475
011507
Panama
476
011508
St.Pierre
477
011509
Haiti
478
011590
Guadeloupe
479
011591
Bolivia
480
011592
Guyana
481
011593
Ecuador
482
011594
FrenchGuyana
483
011595
Paraguay
484
011596
Martinique
485
011597
Suriname
486
011598
Uruguay
487
011599
NethAntilles
488
011670
East Timor
489
011672
Antarctic
490
011673
Brunei
491
011674
Nauru
492
011675
PapuaNewGuin
493
011676
Tonga
494
011677
SolomonIslnd
495
011678
Vanuatu
496
011679
Fiji
497
011680
Palau
498
011681
Wallis Island
499
011682
Cook Islands
500
011683
Niuel Island
1614
Index
Area Code
Area Name
501
011684
AmerSamoa
502
011685
WSamoa
503
011686
Kiribati
504
011687
NewCaledonia
505
011688
Tuvalu
506
011689
FrenchPolyne
507
011690
Tokelau
508
011691
Micronesia
509
011692
MarshalIslnd
510
011850
Korea North
511
011852
Hong Kong
512
011853
Macau
513
011855
Cambodia
514
011856
Laos
515
011870
SatIndlOcn
516
011871
SatEastAtl
517
011872
SatPacific
518
011873
SatIndianOcn
519
011874
SatWestAtl
520
011880
Bangladesh
521
011960
Maldives
522
011961
Lebanon
523
011962
Jordan
524
011963
SyrianArab
525
011964
Iraq
526
011965
Kuwait
527
011966
SaudiArabia
528
011967
Yemen
529
011968
Oman
530
011971
UAE
531
011972
Israel
532
011973
Bahrain
533
011974
Qatar
534
011975
Bhutan
1615
Index
Area Code
Area Name
535
011976
Mongolia
536
011977
Nepal
537
011992
Tajikistan
538
011993
Turkmenistan
539
011994
Azerbaijani
540
011995
Georgia
541
011996
Kyrgyzstan
542
011998
Uzbekistan
543
544
545
546
547
998
999
1616
How to configure
For this feature to work, it must be enabled on the extension by the System Administrator (SA).
To enable 'Call Cost Display' for an extension:
Enter SA mode.
Exit SA mode.
How to use
For EON and Extended IP Phone Users Only
Dial 1075.
Scroll with the up/down navigation keys to view the cost of the last 10 calls.
The display shows the last 10 dialed numbers and their corresponding call cost.
For example: If the call charge is for the dialed number 0014034545247 is $2 and 80 cents' then the
display will show:
001403454 5247
F ri
22 JAN
2.8 0
12:19
1617
1618
How it works
External-Outgoing Calls
It checks whether the flag, Apply CDC to Outgoing Calls, is enabled. It matches B's number with the
entries on the Apply CDC to Number List and the Do Not Apply CDC to Number list in the CDC table.
Three results are possible:
a. The flag is enabled and a match is found for the number in the Apply CDC to Number List. So, CDC is
on the call.
c. The flag is enabled and a match is found in both Number Lists, that is, Apply CDC and Do Not Apply
CDC. The system gives precedence to the Do Not Apply Number List. So, CDC is not applied to the
call.
When CDC is applied to the call (see point a above), the CDC Timer starts as soon as B has answered the
call. This timer is set to 160 seconds as default, but can be programmed to the desired time limit.
At the end of the default/programmed time limit of the CDC Timer, the CDC Goodbye Timer starts. This
timer provides a grace period of 20 seconds for the user to finish the call. This Timer is non-programmable.
At the end of the Goodbye Timer, the call is disconnected, if the 'Disconnect CDC after Timer' flag is
enabled.
If this flag has not been programmed, the call will not be disconnected.
Instead, the CDC Warn Timer will be loaded again for the default/programmed duration. The user can
know how long s/he has been talking.
1619
A is played Warning Beeps. B cannot hear the beeps. This continues until either party disconnects.
External-Incoming Calls
CDC works similarly for incoming calls.
B calls A.
The system checks whether the flag, Apply CDC to Incoming Calls, is enabled and matches B's number
with the entries on the Apply CDC to Number List in the table. If the flag is enabled and a match is found
for the number, CDC is applied on the call.
Internal Calls
A and B are extension users.
A calls B.
The system checks whether the flag, Apply CDC to Internal Calls, is enabled in the CDC Table.
Default: 20 secs.
Feature Interactions:
1620
Call Transfer: In case of transferred call, the CDC timer gets reset and starts again afresh on the
transferred extension.
Emergency Number Dialing: Emergency calls are not affected by this feature, that is, CDC will not be
applied on the dialing of Emergency Numbers.
For Inter PINX or Intra PINX calls (QSIG Calls), the CDC will work only if it is enabled on the source port
(calling extension) irrespective of whether CDC is enabled or disabled on the called extension.
How to configure
This feature is controlled by the 'Call Duration Control Table'. This table is to be programmed as required by the
extensions. The CDC Table is assigned in the Station Advanced Feature Template of those extensions on which
Call Duration Control is to be applied.
To program the Call Duration Control Table,
decide the types of calls - Outgoing, Incoming and Internal - on which CDC is to be enabled.
make a list of numbers on which CDC is to be applied, that is, the Apply CDC to Numbers List.
Make a list of numbers on which CDC is not to be applied, the Do Not Apply CDC to Number List.
The Call Duration Control Table can be programmed using Jeeves and a Telephone.
Under Configuration, click the Call Duration Control to open the page.
The CDC Table will open. There are 8 CDC Tables. By default CDC Table No. 1 is assigned to all
extensions of ETERNITY. If the same CDC is to be assigned to all extensions, program this table.
If different CDC is to be applied to different extensions, program separate CDC tables for these
extensions.
Now program the following parameters in the table you have selected:
Apply CDC to Internal Calls: This flag is to be enabled if CDC is to be applied on internal calls. By
default the flag is disabled.
1621
Apply CDC to Incoming Calls received from Trunk: This flag is to be enabled if CDC is to be applied
to incoming calls external calls. By default this flag is disabled.
Apply CDC to Outgoing Calls made from Trunks: This flag is to be enabled if CDC is to be applied
to outgoing external calls. By default this flag is disabled.
Do Not Apply CDC for calls matching with numbers: This is the list of numbers on which CDC is not
to be applied. By default, Number List 08 is assigned to this parameter. You must program this list with
numbers which you want to be exempt from CDC.
To program the list, click Do Not Apply CDC for calls matching with numbers.
1622
This will lead you to the Number Lists page. Click '001-250' of Number list 07-08.
By default, Number List 08 is assigned to this parameter. You can also program any other Number List
you want. Enter the list of numbers on which CDC is not be applied (refer to the list you prepared).
Click Submit at the bottom of the page to save your list.
Return to the Call Duration Control page. If you have prepared a Number List other than the default
08, then enter the list of that number in the Do Not Apply CDC to Number List column.
Apply CDC for calls matching with numbers: This is the list of numbers on which CDC is to be
applied. By default, Number List 07 is assigned to this parameter. You must program this list with
numbers on which you want CDC to be applied.
To program the list, click Apply CDC for calls matching with numbers. The 'Number Lists' page
opens.
Click '001-250' of Number list 07-08. Follow the same steps as described above for programming the
Do Not Apply CDC Number List.
CDC Timer: This is the time for which the warning beeps are to be played before the system
disconnects the call. The range of the timer is 0001 to 9999 seconds. By default this Timer is set to 160
seconds. Set the CDC Timer to the desired time limit.
1623
Disconnect Call after CDC Timer: This flag is to be enabled if you want the call to be automatically
disconnected on the expiry of the CDC Timer. By default the flag is disabled, which means that calls will
not be disconnected on expiry of the CDC Timer. Enable the flag if required.
Under Configuration, click Station Advanced Feature Template. Ensure that the CDC Table Number
(in this case 01) you have programmed is assigned in the Template you want to apply to the extensions.
To assign the Station Advanced Feature Template with the CDC Table on SLT, DKP and ISDN Terminal
extensions, SIP Extensions go to the respective pages SLT Parameters under SLT Configuration, DKP
Parameters under DKP Configuration and ISDN Terminal Parameters under ISDN Configuration and
SIP Extensions Settings under VoIP Configuration.
Refer Station Advanced Feature Template for instructions on customizing the templates and assigning
them to extensions.
If selected extensions are to be allowed CDC or if different CDC parameters are to be allowed to selected
extensions (for example, 160 seconds duration timer for a few extensions, 360 duration timer for some
other extensions), then follow these steps:
a. Define a new CDC table.
b. Program the different CDC parameters in this table, as required for the extensions.
c. Apply this CDC table on a separate Station Advanced Feature Template.
1624
d. Apply the new Station Advanced Feature Template now programmed with a different CDC table on the
Enter SE mode.
To enable CDC for Outgoing Calls in a CDC Table, dial:
4202-1-CDC Table-Code to enable the flag in a single table.
4202-2-CDC Table-CDC Table-Code to enable the flag in a range of tables.
4202-*-Code to enable the flag in all tables.
Where,
CDC Table is from 1 to 8.
Code is
0 for Disable
1 for Enable.
To enable CDC flag for Incoming Calls in a CDC Table, dial:
4203-1-CDC Table-Code to enable the flag in a single table.
4203-2-CDC Table-CDC Table-Code to enable the flag in a range of tables.
4203-*-Code to enable the flag in all tables.
Where,
CDC Table is from 1 to 8.
Code is
0 for Disable.
1 for Enable.
To enable CDC flag for Internal Call, dial:
4204-1-CDC Table-Code to enable the flag in a single table.
4204-2-CDC Table-CDC Table-Code to enable the flag in a range of tables.
4204-*-Code to enable the flag in all tables.
Where,
CDC Table is from 1 to 8.
Code is
0 for Disable.
1 for Enable.
To assign a number list to Apply CDC to Number List, dial:
4205-1-CDC Table-Number List to enable the flag in a single table.
4205-2-CDC Table-CDC Table-Number List to enable the flag in a range of tables.
4205-*-Number List to enable the flag in all tables.
Where,
CDC Table is from 1 to 8.
Number List is from 01 to 16.
Default Number List is 07.
To assign a number list to Do Not Apply CDC to Number List, dial:
4206-1-CDC Table- Number List to assign Number List to a single table.
4206-2-CDC Table-CDC Table- Number List to assign Number List to a range of tables.
4206-*-Number List to assign Number List to all tables.
Where,
CDC Table is from 1 to 8.
1625
Number List 04 as the Apply CDC to Number List and program the number '0' in this list. Take Number List
05 as the Do Not Apply CDC to Number List and program '022' in this list. Refer the topic "Number Lists"
for programming instructions.
2. Enable CDC for Outgoing call in the table. If using SE Commands, dial 4202-1-5-1.
3. Assign Number List 04 as allowed list and Number List 05 as denied list in table 5. If using SE commands
dial 4205-1-5-04 to assign List 04 and dial 4206-1-5-05 to assign List 05.
4. Change the CDC timer to 240 seconds. If using SE Commands, dial 4207-1-5-240.
1626
5. Enable CDC disconnection Flag in CDC Table 5. If using SE Commands, dial 4208-1-5-1.
6. Assign CDC Table 5 to SLT 202. To do this, change the CDC number in a Station Advanced Feature
1627
How it works
The dialed external number with duration (5-digits in the format of MM:SS) is displayed on the LCD of
EON, when the call is answered.
6 1 6 A M I T PAT E L
F ri
1628
22 JAN
02:52
12:19
Call Forward
What's this?
During a typical workday, it is common for people in an organization to move from one place to another. For
instance, a manager might go on the production floor or remain in the conference room for a few hours; a field
engineer may spend half of the day on site. So, they need to be able to attend their calls even when they are not
present at their desks. The 'Call Forward' feature of ETERNITY ensures this.
Using this feature, calls landing on an extension can be forwarded to another extension, an external number, Voice
Mail, or a Department Group. This way, extension users can ensure that callers can reach them and that they do
not miss calls when they are not present at their extension.
You can also set Call Forward for all the extensions from the SA Mode. See Settings Call Forward for All
Extensions using SA Jeeves.
The Call Forward feature of ETERNITY offers the following forwarding options:
Unconditionally - calls are forwarded to the destination phone number automatically without waiting for a
response from the called party's phone.
If Busy - calls are forwarded to the destination phone number only when the called party's phone is busy.
If No Reply - calls are forwarded to the destination phone number only when the called party does not
answer the phone. Each extension can set a different time after which the call should be forwarded, in
case of no reply. The default time is 30 seconds for all extensions and can be changed by programming
the Call Forward No-Reply Timer.
If Busy or No Reply - calls are forwarded to the destination phone number when the called party's phone
is either busy or does not reply.
Dual Ring237 - when calls are forwarded to another phone number. Both phones, that is, the source phone
(whose calls are forwarded) as well as the destination phone (on which call is forwarded) will ring and the
user can answer from either extension.
Dual Ring is useful to users who may have to be present frequently at two different places. As it is
cumbersome to forward the calls from one extension to another and cancel it repeatedly, extensions users
can set Dual Ring, so they can attend to their calls at either place they are present.
How it works
A has set Call Forward to extension B unconditionally.
The system forwards all calls for A to B, without checking for Busy Tone and without waiting for the Call
Forward No-Reply Timer to expire.
1629
The system waits for the Call Forward No-Reply Timer to expire and forwards all external incoming calls to
the external number.
The system forwards the call for A to B on detecting Busy signal from A.
B belongs to a Department Group and has set Call Forward-If Busy to C within the Department Group.
If the system detects Busy signal on B, it forwards the call for B to C in the Department Group.
However, if the caller has called the Department Group instead of calling B directly, the call will land in the
sequence on all Department group extensions. When it is B's turn, the call will not be forwarded to C, B will
ring instead.
C belongs to a Department Group and has set Call Forward-No Reply to D within the Department Group.
The system waits for the Call Forward No-Reply Timer to expire, and forwards the call for C to D in the
Department Group.
Whenever there is a call for D, if the system does not detect a busy signal from D, it waits for the Call
Forward No-Reply timer to expire.
When there is a call for E, the system rings on both E and the destination F.
Feature Interaction:
1630
Do Not Disturb (DND): When DND and Call Forward-Unconditional are set on an extension, Call Forward
is given priority. If any other type of Call Forward and DND are set on an extension, DND is given priority.
You can select the types of calls, that is, internal calls only, or trunk calls, or both, to be forwarded to
external numbers. You can program the system to forward internal calls only, or trunk calls only or both
trunk calls and internal calls, to the external number. For this, the parameter 'Allow External Call
Forward for' must be programmed in the Station Advanced Feature Template of the extensions that
are allowed Call Forward.
The system supports only single-point Call Forward, which means, if the destination extension is also
forwarded, the call will not follow the forwarding path. For example: Calls for extension A are set to be
Matrix ETERNITY System Manual
forwarded to extension B. Call Forward is also set on extension B with C as the destination number.
Calls for A will land on B only and calls for B will land on C only.
Only one Call Forward Type can be set from an extension. Every new Call Forward Type set overrides
the previous one.
When the calls are forwarded the extension user gets the feature tone on lifting the handset to indicate
that Call Forward is set on his/her extension.
How to configure
The functioning of this feature is controlled by three parameters: 'Class of Service' and 'Call Forward No-Reply
Timer' and Allow External Call Forward for.
Call Forward must be enabled in the Class of Service (COS) group of the extensions to which this feature is to be
allowed.
When Call Forward No-Reply is set, if required the Call Forward No-Reply Timer needs to be programmed.
You may select the types of calls, that is, internal, external, both internal and external calls to be forwarded by
programming the Allow External Call Forward for parameter.
You can set Call Forward for All the Extensions from the SA mode only, see Settings Call Forward for All
Extensions using SA Jeeves.
Refer the topics Class of Service and Station Basic Feature Template for programming instructions.
1631
If you want to set different Timer duration for different extensions, then prepare separate Station Advanced Feature
Templates with the desired Timer durations and assign different Templates (with different Timer durations) to the
extensions as desired.
1632
Under Configuration, click Station Advanced Feature Template to open the page.
Select an Advanced Feature Template number. (by default Template 01 is assigned to all extensions)
Dial command 5602-1-Station Advanced Feature Template Number-02-Call Forward No-Reply Timer
Where,
Station Advanced Feature Template is from 01 to 50. Default: 50.
Timer is from 001 to 255 seconds.
02 is the parameter number for "Call Forward No-Reply Timer" in the Template.
For example: To program Call Forward No-Reply Timer as '60' secs.' in Template number 02, dial 5602-102-02-060
Exit SE mode.
Refer the topic Station Advanced Feature Template for instructions on applying the template to extensions
using Jeeves and from a Telephone.
When Call Forward No-Reply is set on a phone that is programmed in a Trunk Landing Group, the calls will
be forwarded on expiry of 'Call Forward No-Reply Timer' programmed in the routing group for this member
phone. Call Forward No-Reply Timer, programmed in Station Advanced Feature Template will not be
applied in this case.
To set Call Forward of all Extensions to the Voice Mail, select Forward Calls of all Extensions to
Voice Mail. Select the type of Call ForwardUnconditionally, When Busy, When No Reply, When Busy
or No Reply you want to set.
1633
To set Call Forward of all Extensions to an Extension, select Forward Calls of all Extensions to
Extension Configure the Extension Number to which the calls are to be forwarded. Select the type of
Call ForwardUnconditionally, When Busy, When No Reply, When Busy or No Reply you want to set.
How to use
Call Forward can be set/canceled by extension users who are allowed this feature. It can be set/canceled by an
extension user for another extension (refer Call Forward-Remote to know more).
If the call is to be forwarded to an external number, dial Trunk Access Code, then the external phone
number and terminate the command with #*.
1634
For users world wide, Trunk Access Code (TAC) for dialing external numbers are: 0, 5, 61, 62, 63,
64.
For users in USA, TAC for dialing external numbers are: 9, 5, 81, 82, 83, 84.
If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail System. The default
Access Code is 3931. Verify with the System Engineer if the default VMS Access Code has been
changed and use the new code to dial the VMS.
1635
Call Forward-Remote
What's this?
An extension user can set Call Forward for another ('remote') extension from his/her own extension. Thus, Call
Forward set for an extension from another extension is called 'Call Forward-Remote'.
This feature can be used by the Operator or the Receptionist to forward the calls for the Managers and other
extension users to the destinations where they will be available.
This feature is also useful in Hotels, where the Front Desk can set Call Forward for guests. Refer the ETERNITY
Hospitality System Manual to know how this feature can be used in hotels.
Call Forward-Remote is possible only from the System Administration (SA) mode.
How it works
This feature works in the same way as Call Forward. The only difference is that it is set by one extension user for
another extension.
For example:
A needs to forward calls for B's extension to another extension 'C' or an external number or a Voice Mail
System or a Department Number.
A dials the Call Forward-Remote feature code followed by B's extension number, the destination number
where the calls for B should land.
The system routes all incoming calls for B to the destination number.
How to configure
As Call Forward-Remote can be invoked only from the SA mode, either the feature 'SA Mode' or 'SA Extension'
must be enabled in the Class of Service of extensions that are to be allowed this feature.
The feature 'SA Mode' requires a password to be dialed. Users must be provided a password to use this
feature from their extensions. The feature 'SA Extension' allows entry into SA mode, without a password.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
ETERNITY. This Template is assigned CoS group 01 by default. The default CoS group 01 has only 'SA Mode'
enabled. So, all extensions of the ETERNITY can access Call Forward Remote, provided they have the SA
password.
Since 'SA Extension' is disabled in the default CoS group 01, none of the extensions of the ETERNITY can enter
SA mode without password to set Call Forward-Remote.
1636
You may decide which extensions should be allowed Call Forward-Remote feature. In general practice only very
few extensions are allowed this feature.
So, to allow this feature to a few extensions only:
a. Define a CoS group with either 'SA Mode' or 'SA Extension' enabled. Recall that the facility 'SA Mode' is
password protected, so the extensions allowed access to this feature must also be provided an SA
Password.
b. Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
c. Assign this new Template to the extensions to which Call Forward-Remote is to be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions and
programming.
How to use
For EON and Extended IP Phone Users
To set Call Forward-Remote:
Dial 1072-006.
Enter the Destination Phone Number.
Scroll to select the desired Call Forward Type:
All Calls.
If Busy.
If No Reply.
If Busy or No Reply.
Dual Ring.
Press 'Enter' key.
238. If call is to be forwarded to an extension of the ETERNITY, dial the extension number. If call is to be forwarded on an external number, dial Trunk Access Code, then dial the external phone number and terminate the command with #*.
For users world wide, Trunk Access Code (TAC) for dialing external numbers are: 0, 5, 61, 62, 63, 64. For users in USA, TAC for
dialing external numbers are: 9, 5, 81, 82, 83, 84.
239. If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail System. The default Access Code is 3931.
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Dial 1072-006.
Enter Extension Number.
Scroll to select 'Cancel'.
Press 'Enter' key.
You get a confirmation tone and text message for Call Forward canceled.
Go Idle or you get dial tone after 3 seconds.
Lift handset.
Dial 1072-006.
Enter Extension Number.
Dial 1 for All Calls
Dial 2 for If Busy
Dial 3 for If No Reply
Dial 4 for If Busy or No Reply
Dial 5 for Dual Ring
Dial destination Phone Number/Voice Mail System.
You get confirmation tone.
Replace handset.
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Call Forward-Scheduled
What's this?
Extension users may want their calls to be automatically forwarded to a desired destination number during working
hours or non-working hours. To cite an example, a Support Technician spends working hours on the field and
wants all incoming calls on his extension in the office to be forwarded to his cell phone during working hours. During
non-working hours, he wants call calls to be forwarded to his voice mail.
Remembering to set and cancel Call Forward and changing the destination number for each Time Zone, that is,
working hours, non-working hours, break hours, every day proves to be cumbersome for such extension users.
In addition to Call Forward, ETERNITY supports 'Call Forward - Scheduled', which allows extension users to set
call forward for desired Time Zones at one time, and the system automatically forwards the calls to the destination
defined for each Time Zone.
This feature requires a license. To use this feature you must purchase the license for the Mobility Feature
Suite. Refer the topic License Management to know more.
How it works
Call Forward-Scheduled supports all the forwarding options as Call Forward: Unconditionally, If Busy, If No Reply, If
Busy or No Reply, Dual Ring.
Any of these options can be set for the three Time Zones: working hours, break hours and non-working hours.
The destination for Call Forward-Scheduled can be an internal (extension) number or an external number.
Both 'Call Forward' and Call Forward-Scheduled can be set on the same extension. In this case, priority is given to
'Call Forward' over Call Forward-Scheduled.
The logic for forwarding calls to the destination number remains the same as described in the topic Call Forward,
illustrated in the following example.
When there is a call on extension A, the system first checks if there is any 'Call Forward' type (that is,
Unconditional, Busy, No Reply, Busy/No Reply, Call Follow Me) set on extension A.
If 'Call Forward' is set on extension A, the system will follow the logic described in 'How it works' under the
topic 'Call Forward".
If no 'Call Forward' is set on extension A, the system will check if Call Forward-Scheduled is set on A.
Since Call Forward-Scheduled is set on extension A, the system will compare the Time Zone for which the
Call Forward is scheduled with the current Time Zone of extension A.
If the current Time Zone of extension A is the same as the Time Zone set for Call Forward Scheduled, that
is, non-working hours, the call will be forwarded to extension B as per the call forward type set.
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As the Call Forward Type set by A is Unconditional, the system will forward the call to B, without checking
for the Busy Tone and without waiting for the Call Forward No-Reply Timer to expire.
If the current Time Zone of extension A is not the same as Time Zone set for Call Forward-Scheduled, the
call will not be forwarded. The system will consider that no call forward has been set.
Call Forward - Scheduled can be set simultaneously for more than one Time Zone from the same
extension. For example, extension A can set Call Forward-Scheduled for working hours, then again set
Call Forward-Scheduled for non-working hours, and again for break hours.
A different Call Forward Type can be set for a different Time Zone. For example, extension A can set
Call Forward -Unconditional for non-working hours, and Call Forward -Busy for working hours. Also, a
different destination number can be set for forwarding calls in each Time Zone. For example, extension
A can set Call Forward-Unconditional for non-working hours to a mobile number and set extension B as
destination number for working hours.
When more than one Call Forward type is set on the same extension for the same Time Zone, the
latest Call Forward type set for the Time Zone will override the previous Call Forward type set for that
Time Zone. For example, extension A sets Call Forward -Busy for working hours, then sets Call
Forward Busy or No Reply for working hours, the latter will override the former. The system will
consider the latest, that is, Busy or No Reply as the Call Forward type for forwarding calls during
working hours.
Call Forward-Scheduled can be cancelled individually for a desired Time Zone or all at once for all
Time Zones.
Call Forward-Scheduled can be set by extension users as well as for extension users from the System
Administrator mode.
It is also possible to select the types of calls, that is, internal calls only, or trunk calls, or both, to be
forwarded to external numbers. You can program the system to forward internal calls only, or trunk
calls only or both trunk calls and internal calls to the external number. For this, the parameter 'Allow
External Call Forward for' must be programmed in the Station Advanced Feature Template of the
extensions that want to use Call Forward-Scheduled.
Call Forward-Scheduled when set/cancelled from the SA mode, will not depend on the assigned CoS.
How to configure
The programming of this feature involves the same parameters as in Call Forward.
'Call Forward' must be enabled in the Class of Service (CoS) group of the extensions to which this feature is to be
allowed. Refer the topic "Call Forward".
If Call Forward No-Reply is to be set, and if required, the Call Forward No-Reply Timer may be
Programmed in the Station Advanced Feature Template applied on the extensions which are to be allowed this
feature. Refer the topic Call Forward.
The types of calls to be forwarded to the external number may be selected in the parameter "Allow External Call
Forward for" in the Station Advanced Feature Template applied on the extensions which are allowed Call
Forward-Scheduled. You may select from 'Internal Calls', 'Trunk Calls' and 'Internal + Trunk Calls'. By default, only
trunk calls are forwarded to external numbers.
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Extensions that are to be allowed to set Call Forward-Scheduled for other extensions must be allowed either the
feature 'SA Mode' or 'SA Extension' in their COS. Refer the topic Call Forward-Remote.
How to use
Call Forward-Scheduled can be set/canceled by users for their own extension, or for any other extension from the
SA mode.
Click Extension.
Now, Search Extension, by entering either extension number as Extension Number, or by entering the
name of the extension as Extension Name.
Click Submit.
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You can set Call Forward - Scheduled for Working Hours, Break Hours as well as for Non-working Hours
To set Call Forward for the extension for the Working Hours, under Working Hours, select the type of Call
Forward:
To forward calls to voice mail, select the radio button Forward Calls to Voice Mail and the type of call
forward from the drop down list. Default: Unconditionally.
To forward calls of this extension to another extension, select Forward Calls to Phone radio button
and the type of call forward for this option from the drop down list. Default: Unconditionally.
Enter the extension number to which calls must be forwarded in the empty field provided for this option.
To forward calls of this extension to an external number, select Forward Calls to External Number
radio button, and select the type of call forward for this option from the drop down list. Default:
Unconditionally.
Enter the external number to which calls must be forwarded in the empty field provided for this option.
Click the Apply Call Forward button to set Call Forward -Scheduled.
The message Call Forward is set appears.
Click the Dual Ring button to set Call Forward-Scheduled with Dual Ring.
The message Dual ring is On appears.
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To set call forward for Break Hours and Non-working Hours, follow the same instructions as above.
To set Call Forward - Scheduled for another extension, follow the same instructions as above.
The destination number for forwarding calls can be a maximum of 24 digits. Terminate the command
with #* if destination number has fewer than 24 digits.
If the destination number is an external number, enter the Trunk Access Code followed by the
destination number.
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Dial 1072-223.
Enter extension number (from which calls are to be forwarded)
Scroll to the desired Time Zone.
Press Enter key to select Time Zone.
Scroll to the desired Call Forward type for the selected Time Zone.
Press Enter key to select Call Forward type.
Enter Destination Number on prompt.
You get confirmation tone and message showing extension to which Call Forward is set.
Lift handset.
Dial 1175-1-1-Destination Number for CF-Scheduled-Unconditional.
Dial 1175-1-2-Destination Number for CF-Scheduled -Busy.
Dial 1175-1-3-Destination Number for CF-Scheduled -No Reply.
Dial 1175-1-4-Destination Number for CF-Scheduled-Busy/No Reply.
Dial 1175-1-5-1 for CF-Scheduled -Dual Ring.
Dial 1175-1-5-0 to cancel CF-Scheduled -Dual Ring.
Dial 1175-1-0 to cancel CF-Scheduled for Working Hours.
Replace handset.
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Lift handset.
Dial 1175-2-1-Destination Number for CF-Scheduled -Unconditional.
Dial 1175-2-2-Destination Number for CF-Scheduled -Busy.
Dial 1175-2-3-Destination Number for CF-Scheduled -No Reply.
Dial 1175-2-4-Destination Number for CF-Scheduled -Busy/No Reply.
Lift handset.
Dial 1175-3-1-Destination Number for CF-Scheduled -Unconditional.
Dial 1175-3-2-Destination Number for CF-Scheduled -Busy.
Dial 1175-3-3-Destination Number for CF-Scheduled -No Reply.
Dial 1175-3-4-Destination Number for CF-Scheduled -Busy/No Reply.
Dial 1175-3-5-1 for CF-Scheduled -Dual Ring.
Dial 1175-3-5-0 to cancel CF-Scheduled -Dual Ring.
Dial 1175-3-0 to cancel CF-Scheduled for Non-working Hours.
Replace handset.
Lift handset.
Dial 1072-223-Extension number-1-1-Destination Number for CF-Scheduled -Unconditional.
Dial 1072-223-Extension number-1-2-Destination Number for CF-Scheduled -Busy.
Dial 1072-223-Extension number-1-3-Destination Number for CF-Scheduled -No Reply.
Dial 1072-223-Extension number-1-4-Destination Number for CF-Scheduled -Busy/No Reply.
Dial 1072-223-Extension number-1-5-1 for CF-Scheduled -Dual Ring.
Dial 1072-223-Extension number-1-5-0 to cancel CF-Scheduled -Dual Ring.
Dial 1072-223-Extension number-1-0 to cancel CF-Scheduled -for working hours.
Replace handset.
Lift handset.
Dial 1072-223-Extension number-2-1-Destination Number for CF-Scheduled -Unconditional.
Dial 1072-223-Extension number-2-2-Destination Number for CF-Scheduled -Busy.
Dial 1072-223-Extension number-2-3-Destination Number for CF-Scheduled -No Reply.
Dial 1072-223-Extension number-2-4-Destination Number for CF-Scheduled -Busy/No Reply.
Dial 1072-223-Extension number-2-5-1 for CF-Scheduled -Dual Ring.
Dial 1072-223-Extension number-2-5-0 to cancel CF-Scheduled -Dual Ring.
Dial 1072-223-Extension number-2-0 to cancel CF-Scheduled -for break hours.
Replace handset.
Lift handset.
Dial 1072-223-Extension number-3-1-Destination Number for CF-Scheduled -Unconditional.
Dial 1072-223-Extension number-3-2-Destination Number for CF-Scheduled -Busy.
Dial 1072-223-Extension number-3-3-Destination Number for CF-Scheduled -No Reply.
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If Call Forward-Unconditional and Call Forward-When Not Registered, have been set on the same SIP
phone. Call Forward-Unconditional will have priority over Call Forward-When Not Registered.
How to configure
The Call Forward-When Not Registered feature does not require any specific programming except:
ensuring that 'Call Forward' in the Class of Service (COS) group in the Station Basic Feature Template
Parameters applied to the SIP phones.
if required, selecting the types of calls to be forwarded to the external number. By default, only trunk calls
are forwarded to external numbers. If you want to select a different type of call, configure the parameter
Allow External Call Forward for in the Station Advanced Feature Template applied to the SIP phones.
Refer the sub-topic Station Advanced Feature Template, under Configuring Extensions.
If you want to allow Call Forward-When Not Registered to be set only by the System Administrator (SA) for
the extension users, the System Engineer (SE) must disable Call Forward feature in the Class of Service
(CoS) group in the Station Basic Feature Template applied to the SIP phones.
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If you disable Call Forward in the CoS of a SIP phone, the user will not be able to set any other type of
Call Forward.
Click Extension.
Now, Search Extension on which you want to set this feature. You search by entering either Extension
Number or the Extension Name.
Click Submit.
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Select the destination for forwarding calls when the SIP Extension fails to register from the following:
Forward Calls to Voice Mail.
Forward Calls to Extension Number. If you select this option, you must enter the desired
Extension Number in the corresponding box.
Forward Calls to External Number. If you select this option, you must enter the desired external
number in the corresponding box. Also, assign a trunk to route the call by selecting the Trunk
Access Code from the using TAC list.
Click the Apply Call Forward button. The message Call Forward is set appears.
To set time-zone based Call Forward - When Not Registered, click Call Forward When Not RegisteredScheduled to expand.
To set Call Forward When Not Registered for working hours, under Working Hours, select the desired
destination from the following options:
Forward Calls to Voice Mail.
Forward Calls to Extension Number. If you select this option, you must enter the desired
Extension Number in the corresponding box.
Forward Calls to External Number. If you select this option, you must enter the desired number in
the corresponding box, and assign a trunk to route the call by selecting the Trunk Access Code in
the using TAC list.
Click the Apply Call Forward button. The message Call Forward is set appears.
To set call forward for Break Hours and Non-working Hours, follow the same instructions as above.
To set Call Forward When Not Registered - Scheduled for another extension, follow the same
instructions as above.
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Lift handset.
Press DSS key assigned to Call Forward-When Not Registered (if programmed).
OR
Dial *13.
Scroll to the desired option.
On the prompt, Forward to Number, enter the Destination NumberExtension Number/External Number/
Voice Mail System.
The destination number for forwarding calls can be a maximum of 24 digits. Terminate the command
with #* if destination number has fewer than 24 digits.
If the you want to route the calls to the Voice Mail, enter the VMS Access Code as the destination
number.
If the destination number is an external number, enter the Trunk Access Code followed by the
destination number and #*.
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Lift handset.
The destination number for forwarding calls can be a maximum of 24 digits. Terminate the command
with #* if destination number has fewer than 24 digits.
If the you want to route the calls to the Voice Mail, enter the VMS Access Code as the destination
number.
If the destination number is an external number, enter the Trunk Access Code followed by the
destination number and #*.
Lift handset.
Dial *13-1-0.
Replace handset.
1651
Call Hold
What's this?
Call Hold enables you to put an on-going conversation (with an internal or external number) on hold. ETERNITY
offers three types of Call Hold:
Exclusive Hold: An on-going conversation is put on hold from a DKP/Extended IP Phone and is retrieved
from the same DKP/Extended IP Phone that put it on hold.
Global Hold: An on-going conversation is put on hold from a DKP/Extended IP Phone and is retrieved
from any DKP/Extended IP Phone connected to ETERNITY.
Consultation Hold: An on-going conversation is put on hold in order to perform any further activity, such
as Call Transfer, Conference, Call Toggle.
Exclusive Hold and Global Hold are supported on DKPs and Extended IP Phones.
Consultation Hold is supported on the SLTs, DKPs and Extended IP Phones.
ETERNITY supports interoperability with the Polycom IP Phones. When any extension of ETERNITY puts
a SIP Extension on hold (Exclusive, Global or Consultation Hold), ETERNITY will send Re-Invite message
to the SIP Extension put on hold.
How it works
Exclusive Hold
When a call is put on Exclusive Hold,
ETERNITY starts the Exclusive Hold Retrieval Timer (programmable; default: 2 minutes). The call remains
connected to the DKP/Extended IP phone which placed it on hold.
The extension user can retrieve the call within this timer.
If the call is not retrieved before the expiry of this timer, it is returned to the DKP/Extended IP Phone. The
DKP/Extended IP Phone rings and may answer the call.
The returned call is disconnected, if the DKP/Extended IP Phone is not in idle state, or if the call is not
answered by DKP/Extended IP Phone.
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Pressing the Hold key again (when the DKP/Extended IP Phone is idle).
Pressing the Call Appearance key of the call put on hold (when the DKP/Extended IP Phone is busy).
Answering the call, when it returns at the end of the Exclusive Hold Retrieval Timer.
To be able to place calls on Exclusive Hold, you must select 'Exclusive Hold' as the Default Call Hold Type in the
System Parameters of ETERNITY. See System Parameters for instructions.
Global Hold
When a call is placed on Global Hold,
The call remains connected in the system. The call remains on hold for the duration of the Global Hold
Retrieval Timer (programmable; default: 60 seconds).
Any DKP/Extended IP Phone connected to the ETERNITY can pick up the call put on Global hold by:
Pressing the DSS key assigned to the extension put on Global Hold.
If this call is not retrieved before the expiry of the Global Hold Retrieval Timer, the call is returned to the
DKP/Extended IP Phone which put it on hold. The DKP/Extended IP Phone rings and may answer the call.
The returned call is disconnected, if the DKP/Extended IP Phone is not in idle state, or if the call is not
answered by DKP/Extended IP Phone.
To be able to place calls on Global Hold, you must select 'Global Hold' as the Default Call Hold Type in the System
Parameters of ETERNITY. The DKP/Extended IP Phone (which picks up the call) must have a DSS Key to access
the Trunk or the Extension which is put on hold. See System Parameters for instructions.
ETERNITY provides the flexibility to use Exclusive Hold and Global Hold at the same time. You can put
calls on Exclusive Hold even when Global Hold is enabled in the system.
ETERNITY does not support Global Hold on SIP Trunks.
If a call put on Exclusive or Global Hold is to be transferred or included in a Conference, you must first
retrieve the call.
Consultation Hold
During an on-going conversation, any SLT, DKP or Extended IP Phone can place a call on Consultation Hold to
perform any of the following:
Call Transfer
Call Toggle
Conference
Making a Second Call
Call Park
Mute
Call Chaining
Conversation Recording
Door Lock Opener
Flashing on Trunks
Manual Priority Intrusion (for E&M MFCR2 only)
Matrix ETERNITY System Manual
1653
The call is released from the held state once the operation has been performed or cancelled.
How to configure
For Exclusive and Global Hold, you must configure the following parameters:
Class of Service: Call Hold must be enabled in the Class of Service (CoS) of the DKPs/Extended IP
Phones you want to allow this feature.
In the default Station Basic Feature Template 01 assigned to all extensions of ETERNITY, Call Hold is
included in the 'Basic Features' assigned to all Class of Service groups, including the default CoS group
01. So, all extensions of ETERNITY can use this feature.
Refer the topics Class of Service (COS) and Station Basic Feature Template to know more.
Call Hold Type: Enable the desired option, that is, Exclusive Hold or Global Hold in the System
Parameters.
Send Re-INVITE over SIP Trunk on Hold: When an external call over a SIP Trunk is put on hold by any
extension, and you want ETERNITY to send Re-INVITE message over SIP Trunk to the remote end, you
must enable this flag on the SIP Trunk. See Configuring SIP Trunks to know more.
DSS Keys: Program DSS Keys for Trunks and Stations on the DKPs which are allowed to retrieve calls on
Global Hold. Refer the topic DSS Keys Programming, Phone Key Settings in Configuring Matrix
SPARSH VP330 and Phone Key Settings in Configuring Matrix SPARSH VP248 - Extended IP Phone
for instructions.
Global Hold Retrieval Timer: Change the default setting of this timer to the desired duration, if required.
Exclusive Hold Retrieval Timer: Change the default setting of this timer to the desired duration, if
required.
For instructions to change the Timers, see System Timers and Counts.
For Consultation Hold to work, Call Hold must be enabled in the Class of Service (COS) of the SLTs, DKPs/
Extended IP Phones you want to allow this feature.
How to use
Exclusive Hold
For EON and Extended IP Phone Users Only
To put a call on Exclusive Hold, when Exclusive Hold is selected as the Default Call Hold Type:
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Global Hold
For EON and Extended IP Phone Users Only
To put a call on Global Hold, when Global Hold is selected as the Default Call Hold Type:
From any DKP/Extended IP Phone, press the DSS Key of the Trunk/extension put on Global Hold.
Consultation Hold
For EON and Extended IP Phone Users
To put a call on Consultation Hold:
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OR
Press Transfer key.
Press Flash.
OR
Tap the Hook switch of your phone.
For detailed feature description and instructions refer to the topic How to use in
Call Transfer
Call Toggle
Conference-3 Party, Conference-Multiparty, Conference Dial-In
Call Park
Mute
Call Chaining
Conversation Recording
Door Phone
Flashing on Trunks
Priority Calls in E&M MFCR2 Signaling
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Call Park
What is this?
Call Park allows you to place a call on Hold, so it can be retrieved from the same or another extension of the
system.
A call is 'parked' when the extension user temporarily places the call into a location in the system called 'Orbit'. The
user can attend to other calls. The parked call can be retrieved on completion of the current call by dialing the Orbit
number.
Call Parking is useful in offices housed in different parts of a building or multi-storied offices. It is useful in situations
like:
the person who picked up the call is not the desired called party or the desired party is at an unknown
location. The person who picked up the call can then either go to find the desired called party or call other
numbers to find him/her. When found, the desired called party can pick up the call from the same or any
extension by dialing the Orbit number.
the person who picked up the call may have to go to another part of the office to look up a file or consult a
colleague. The person can park the call and continue the conversation from the other part of the office.
Call Park-General Orbit: The extension user can park calls in any of the 8 'general' Orbits, which are like
fictional extensions located in the system. The calls parked in the General Orbit can be picked up from any
extension by dialing the General Orbit Number. At a time, only one call can be parked in each General
Orbit.
Call Park-Personal Orbit: Each telephone instrument (EON/SLT/IP Phone) connected as extension has
one Personal Orbit. Calls parked in personal orbit can be picked up only from where the call is parked. So,
no other person can pick up this call. Multiple calls can be parked in the Personal Orbit at a time.
Extension users can park the call either in the General Orbit or the Personal Orbit by dialing an Orbit Number from
1 to 9, where:
After parking a call, the extension user can continue to make and answer other calls and use other system features.
On SIP extensions, ETERNITY supports Call Park and Retrieve using REFER Message. For a list of IP
phones on which this feature has been tested, see ETERNITY Features tested on IP Phones of different
Brandsin the Appendix.
1657
How it works
A and B are extension users. C, D and E are callers.
C calls B.
A picks up the call.
As B is not present at his extension, A parks the call in General Orbit Number 2 by dialing the Access
Code.
C is played on-hold music.
A tries to find B (by either calling several numbers or by going in person or sending someone).
The parked call remains in orbit for the duration of the Call Park Timer, which is set to 2 minutes by default.
A finds B.
B retrieves the call from another extension by dialing the feature access code for retrieving Call Park and
Orbit number 2.
If A cannot locate B or if B cannot attend the call, A can also retrieve the call from his extension.
However,
If neither A nor B retrieves the parked call within the Call Park Timer, the system will hunt for the extension
that parked the call (A) on the expiry of the Call Park Timer.
Meanwhile, if A is busy, the system again keeps the call parked in orbit number 2 for the period of the Call
Park Timer. This process continues for the duration of the Call Park Release Timer, which is set to 3
minutes by default.
If A is free, the system will ring on A's phone. A gets connected to C again.
If A does not retrieve the parked call till the end of the Call Park Release Timer, C gets disconnected.
How to configure
To provide this feature to extensions,
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Enable the Call Park in the Class of Service (COS) of the Station Basic Feature Template of the
extensions. By default, this feature is enabled in the COS of all extension types for all the time zones.
If required, you may change the duration of the Call Park Timer and the Call Park Release timer. See
System Timers and Counts for instructions.
How to use
For EON and Extended IP Phone Users
To park a call:
You are in speech with extension/external caller.
Press DSS Key assigned to 'Call Park'.
OR
Press Transfer Key and dial 115
Enter Orbit Number (1-9)
(Personal Orbit:1; General: 2-9).
To retrieve a parked call from your phone, when your phone is in idle state:
To retrieve a parked call from your phone, when you are in speech with someone:
To retrieve a parked call from your phone, when your phone is in idle state:
To retrieve a parked call from your phone, when you are in speech with someone:
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Call Logs
What's this?
ETERNITY stores the details of 20 each, of the following types of calls:
Missed calls: incoming calls that were not answered by extension users.
Answered calls: incoming calls answered by extension users.
Dialed calls: calls made by extension users.
The call history of each of the above types of calls is stored by Name, Number, and Date-Time of the Call.
If there is no name in the CLI of the above types of calls, the system stores and displays the Number and the DateTime. In case there is no number in the CLI, the system will display the Port number on/from which the call was
received/made.
The Call Logs contain details of both internal as well as external calls made or received by the extension users.
The Call Logs feature is supported on EON and Extended IP Phones.
Using call logs you can:
view call history: you can see the calls you missed, answered or dialed.
make calls: you can call any number that you have missed, answered, or dialed.
edit the numbers: you can change or modify the number in the call log. This is useful when the CLI
received and stored in the call log is not in the same format that is to be used to make calls.
save the numbers: you can store the external numbers in your call logs in the "Personal Directory" and
use them for Personal Abbreviated Dialing.
The maximum number of calls that can be stored under each Call Log type is 20. The logs will be cleared
automatically using the First-In, First-Out method, that is, the latest call detail will replace the record of the oldest
call detail.
Given the limited Call Log capacity, the system also allows you to choose if you want internal calls to be displayed
or not in the Missed, Answered and Dialed Call Logs. And accordingly it will store internal calls in the logs.
The system stores each Missed, Answered and Dialed call individually even if the same number is received
multiple times.
How to configure
This feature does not require any specific programming, except:
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Selecting whether internal calls should be logged in the Missed, Answered and Dialed Call Logs. This can
be done on the 'System Parameters' page of Jeeves or by using a Telephone.
Programming of a DSS key for the Call Logs feature. For instructions please refer the topic Configuring
DKP Extensions.
You may enable any or all of the following flags by selecting the respective check box:
Store Internal Calls in Missed Call Log
Store Internal Calls in Answered Call Log
Store Internal Calls in Dialed Call Log
Click Submit at the bottom of the page to save your settings.
Log out of Jeeves or continue, as required.
Enter SE mode.
To enable/disable Log Internal Calls in Missed Calls, dial:
5361-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Missed Calls" log)
1 for Enable (Store internal calls in "Missed Calls" log)
By default, Internal Call Logs in Missed Calls is enabled.
To enable/disable Log Internal Calls in Answered Calls, dial:
5362-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Answered" log)
1 for Enable (Store internal calls in "Answered Calls" log)
By default, Internal Call Logs in Answered Calls is enabled.
To enable/disable Log Internal Calls in Dialed Calls, dial:
5363-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Dialed Calls" log)
1 for Enable (Store internal calls in "Dialed Calls" log)
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Exit SE mode.
How to use
The Call Logs feature allows you to view calls and edit numbers, make calls to any number logged, and store
numbers.
If there is no name in the CLI, the Call Log will only display the number.
If you press the 'Enter' key, the system will dial out the number you just viewed.
Press DSS Key programmed for Call Logs, when the phone is idle.
Scroll to select the desired Call Log Call Log: Missed, Answered, Dialed.
Follow the same steps as described for viewing Call Log entries using the Phone Menu.
OR
1662
Press the DSS Key assigned the Call Logs feature, when it glows.
The phone will display the call log details of the last missed call by: <Name> <Date> <HH:MM> (only if
name is received).
Press Enter key.
The phone will display the Number: <XXXXXXXXXXX>
You may exit the Phone Menu by going OFF-Hook or pressing the Cancel key.
You may also edit or store the number.
You may scroll with the < Back navigation key to view the other call logs.
The LED of the Call Logs DSS key will be turned off once you have viewed the missed call.
Go to Call Logs from the Phone Menu or by pressing Call Logs DSS Key. (see instructions given above).
Scroll with the Up/Down Navigation Key to reach the desired Call Log: Missed, Answered, Dialed.
Press Enter key to select the desired Call Log.
The phone will display the call log details by: <Name> <Date> <HH:MM>
Press Enter key.
The phone will display the Number: <XXXXXXXXXXX>
To edit the number, move the cursor with the Front and Back navigation key.
Place the cursor under the digit you want to delete.
Press 'Cancel' key to delete a digit.
To insert a digit, place the cursor where you want to insert the digit, and enter the digit using the dial pad.
The digit will be inserted in the number string accordingly.
Repeat the same to delete/insert another digit.
After editing the number, you may store it in the Personal Directory or dial the edited number by pressing
the Enter key.
The original number (you now changed) will remain unaffected in the Log. However, if you make a call to
the new number (you changed), it will be logged in the "Dialed" call log and the Last Number Redial list.
Go to Call Logs from the Phone Menu or by pressing Call Logs DSS Key.
Scroll with the Up/Down Navigation Key to reach the desired Call Log: Missed, Answered, Dialed.
Press Enter key to select the desired Call Log.
The phone will display the call log details by: <Name> <Date> <HH:MM>
Press Enter key.
The phone will display the Number: <XXXXXXXXXXX>
Press Enter key.
The system will dial out the selected number using the Outgoing Trunk Bundle Group assigned in TAC-1.
The dialed number will be logged in the "Dialed" call log and the Last Number Redial List.
When you store the number in the Personal Directory, the system will automatically assign Trunk
Access Code "TAC-1".
If all 25 Location Index Numbers of the Personal Directory are already programmed, the message
"Memory Full" will appear on your phone's display and you will get an Error Tone. Refer the topic
Abbreviated Dialing to know more.
1663
Call Pick Up
What's this?
Call Pick-Up allows extension users to answer calls ringing on other extensions from their own extension; without
physically going to the ringing extensions.
Extension users can 'pick-up' both internal and trunk calls ringing on other extensions.
As extension users can answer calls of their colleagues or co-workers without physically going to their extensions,
this feature ensures that all incoming calls are answered.
ETERNITY offers two types of Call Pick-Up:
Call Pick Up-Group - extensions are assigned to Pick-Up Groups. Any extension in a Pick-Up Group can
answer calls ringing on other extensions within the same group only.
Call Pick-Up Selective - calls ringing on any extension of the system can be answered.
On SIP extensions, ETERNITY supports Call Pickup-Selective and Call Pickup-Group using Temporary
Subscription. For a list of IP phones on which this feature has been tested, see ETERNITY Features
tested on IP Phones of different Brands in the Appendix.
Eternity will send only first 3 ringing call's information in NOTIFY message to the SIP Extension, which has
requested Group Call Pickup. This feature has been supported in SIP Phones of CISCO and POLYCOM.
SIP Extension which has subscribed for BLF of DKP / SIP Extension of ETERNITY, the Eternity will send
information for the call present in the first call loop only.
How it works
Call Pick-Up Group
1664
Extensions must be assigned to Call Pick-Up Groups. The extensions in a Call Pick-Up group may be SLT,
DKP and ISDN Terminal.
For example, extensions 2007, 2008, 2009, 2010. 2011, 2012, 2013 are assigned to Pick-Up Group
number 03.
When an extension in this group rings, any extension in the group can pick up the call by dialing the
feature access code for Call Pick-Up Group (default: 4).
Whenever an extension in the system rings, the call can be picked up by any extension of the system by
dialing the feature access code and the number of ringing extension.
When more than one extension in a Pick-Up Group is ringing, you can choose which one to answer first,
using Call Pick-Up Selective.
Feature Interactions:
Call States: Call Pick-Up will fail if the ringing extension goes into idle state just when you are dialing the
pick-up access code.
Auto Call Back: Call Pick-Up will fail if the call ringing on the extension is an Auto Call Back request.
Alarms: Call Pick-Up will fail if the call ringing on the extension is an Alarm Call.
How to configure
For this feature to function, Call Pick-Up should be enabled in the Class of Service of extension that are to be
allowed this feature.
Call Pick-Up Groups
On a sheet of paper, list the extensions that are to be grouped into a Call Pick-Up Group. Make as many Call PickUp Groups as required. Assign each group a number.
Call Pick-Up
Group Number
01
SLT Extensions
DKP Extensions
ISDN Terminals
02
03
3205, 3206
:
99
The numbering of Call Pick-Up Groups must start from 01 and end at 99.
Do not assign '00' as Call Pick-Up Group. '00' is the command to de-assign from a Call Pick-Up Group.
To program these groups, you may use Jeeves or issuing SE commands from a telephone.
1665
1666
In the column Call Pick-up Group, assign the group number for SLT extensions. Refer to the sheet of
paper you prepared.
Assign Call Pick-Up Group number to DKP extensions. Refer to the sheet of paper you prepared.
Assign Call Pick-Up Group numbers to ISDN Terminals. Refer to the sheet of paper you prepared.
1667
1668
Exit SE mode.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions and
programming.
How to use
For EON and Extended IP Phone Users
To pick up a ringing extension in your Group:
Dial 4.
Talk.
Go idle.
To pick up any one of several ringing extensions ringing or the extension that is not in your group:
Dial 12.
1669
To pick up any one of several ringing extensions ringing or the extension that is not in your group:
1670
Event
Sound
Duration
Timer
Dial Tone 1
Toooooooooooo
Dial Tone 2
Toooooooooooo
Turroo... Turrroo
Busy Tone
(Engaged Tone)
Tooooooo.........
Toooooooo
Error Tone
(Congestion/
Refusal Tone as
per ITU
TooTooToo
Too
Internal Call
Waiting Tone
(Intrusion Tone
as per ITU)
Beep..
Beep
Interrupt Request
Timer, Barge-In
Timer
1671
CPT
Event
Sound
Duration
Timer
External Call
Waiting Tone
(Call Waiting
Tone as per ITU)
Beep...Beep
......Beep...
Beep
Transfer-On Busy
Timer.
Confirmation
Tone
(Acceptance
Tone as per ITU)
Beep... Beep...
Beep
Confirmation Tone
Timer
Feature Tone
Beep.................
Beep
Programming
Tone
Beep.................
Beep
Programming
Confirmation
Tone
Beep... Beep...
Beep
Programming
Confirmation Timer
Programming
Error Tone
TooTooToo
Too
Programming Error
Tone Timer
a. In Store and Forward dialing, the digits are first stored in a memory location and then these are dialed on the trunk.
For example: When Least Cost Routing (LCR) is enabled, the system will store the dialed digits first, check the trunk
through which the call is to be routed and then dials the number on the appropriate trunk.
Tone standards vary with the country of application. For example, as per ITU standard, the Dial Tone for India
consists of 400Hz modulated by 25Hz, whereas it is 350+440Hz, without modulation, for USA/Canada. Further,
many countries use different frequencies and cadences for the same tone. For example, in the US, five different
frequency and cadence are used for Dial Tone.
ETERNITY offers the flexibility of setting the Call Progress Tone Generation (CPTG) type to match the countryspecific CPT standards established by ITU.
India being the default 'Region' for ETERNITY, the CTPG for India is set as default in the system.
1672
How it works
At the time of installation, when the System Engineer selects the 'Region' (according to the geographical location of
the site where the system is installed) and defaults the system, ETERNITY sets the country-specific CPTG type
defined for the selected 'Region'.
For countries that use different frequencies and cadences for the same tone, for instance, USA, only one
frequency/cadence among the group is considered. See Table "Default CPTG Type".
How to configure
Programming of Call Progress Tones involves configuration of three parameters: CPTG Type (Region), CPT
related Timers, and Dial Tone Type.
The country-specific CPTG type is set automatically by the system when the 'Region' is selected. However, if
required, the System Engineer can change the CTPG type set by the system.
To set the Call Progress Tone Type, select the desired Region from the list.
Under Configuration, click System Timers and Counts to open the page.
1673
CPTG
Region
Code
Region
Dial Tone 2
Freq.
Cadence
(sec)
Freq.
Cadence
(sec)
Freq.
Region1
440
Continuous
350+440
Continuous
350+440
Region2
400
Continuous
400
Continuous
400
Region3
350+440
Continuous
350+440
Continuous
Argentina
425
Continuous
425
Continuous
1674
Cadence
(sec)
Busy Tone
Freq.
Cadence
(sec)
0.4on 0.2off
0.4on 2.0off
440
0.75on 0.75off
0.6on 0.2off
0.2on 2.0off
400
0.5on 0.5off
440+480
2.0on 4.0off
480+620
0.5on 0.5off
425
425
0.3on 0.2off
CPTG
Region
Code
5
Dial tone 1
Region
Australia
Brazil
Canada
Freq.
Cadence
(sec)
425*25
Continuous
Dial Tone 2
Freq.
Cadence
(sec)
Cadence
(sec)
Freq.
425*25
Continuous
400*25
.4on .2off
.4on 2.0off
Busy Tone
Freq.
Cadence
(sec)
425
0.375on
0.375off
425
Continuous
425
Continuous
425
425
0.25on 0.25off
350+440
Continuous
350+440
Continuous
440+480
2.0on 4.0off
480+620
0.5on 0.5off
China
450
Continuous
450
Continuous
450
1.0on 4.0off
450
0.35 on
0.36off
Egypt
425*50
Continuous
425*50
Continuous
425*50
2.0on 1.0off
425*50
1.0on 4.0off
10
France
440
Continuous
440
Continuous
440
1.5on 3.5off
440
0.5on 0.5off
11
Germany
425
Continuous
425
Continuous
425
1.0on 4.0off
425
0.48on 0.48off
12
Greece
425
0.2on 0.3off
0.7on 0.8off
425
0.2on 0.3off
0.7on 0.8off
425
1.0on 4.0off
425
0.3on 0.3off
13
India1
400*25
Continuous
400*25
Continuous
400*25
.4on .2off
.4on 2.0off
400
0.75on 0.75off
14
Indonesia
425
Continuous
425
Continuous
425
1.0on 4.0off
425
0.5on 0.5off
15
Iran
425
Continuous
425
Continuous
425
1.0on 4.0off
425
0.5on 0.5off
16
Iraq
400
0.4on 0.2off
0.4on 1.5off
400
0.4on 0.2off
0.4on 1.5off
400
Continuous
400
1.0on 1.0off
17
Israel
400
Continuous
400
Continuous
400
1.0on 3.0off
400
0.5on 0.5off
18
Italy1
425
Continuous
425
Continuous
425
1.0on 4.0off
425
0.5on 0.5off
19
Japan
400
Continuous
400
Continuous
400*25
1.0on 2.0off
400
.5on .5off
20
Kenya
425
Continuous
425
Continuous
425
0.67on
3.0off 1.5on
5.0off
425
0.2on 0.6off
0.2on 0.6off
21
Korea
350+440
Continuous
350+440
Continuous
440+480
1.0on 2.0off
480+620
0.5on 0.5off
22
Malaysia
425
Continuous
425
Continuous
425
0.4on 0.2off
0.4on 2.0off
425
0.5on 0.5off
23
Mexico
425
Continuous
425
Continuous
425
1.0on 4.0off
425
0.25on 0.25off
24
New
Zealand
400
Continuous
400
Continuous
400+450
0.4on 0.2off
0.4on 2.0off
400
0.5on 0.5off
25
Phillippines
425
Continuous
425
Continuous
425+480
1.0on 4.0off
480+620
0.5on 0.5off
26
Poland
425
Continuous
425
Continuous
425
1.0on 4.0off
425
0.5on 0.5off
27
Portugal
425
Continuous
425
Continuous
425
1.0on 5.0off
425
0.5on 0.5off
28
Russia
425
Continuous
425
Continuous
425
0.8on 3.2off
425
0.4on 0.4off
29
Saudi
Arabia
425
Continuous
425
Continuous
425
1.2on 4.6off
425
0.5on 0.5off
30
Singapore
425
Continuous
425
Continuous
425*24
0.4on 0.2off
0.4on 2.0off
425
.75on .75off
31
South
Africa
400*33
Continuous
400*33
Continuous
400*33
0.4on 0.2off
0.4on 2.0off
400
.5on .5off
32
Spain
33
Thailand
34
Turkey
35
425
Continuous
425
Continuous
425
1.5on 3.0off
425
0.2on 0.2off
400*50
Continuous
400*50
Continuous
400
1.0on 4.0off
400
0.5on 0.5off
450
Continuous
450
Continuous
450
2.0on 4.0off
450
0.5on 0.5off
UAE
350+440
Continuous
350+440
Continuous
400+450
0.4on 0.2off
0.4on 2.0off
400
0.375on
0.375off
36
UK
350+440
Continuous
350+440
Continuous
400+450
0.4on 0.2off
0.4on 2.0off
400
0.375on
0.375off
37
USA
350+440
Continuous
350+440
Continuous
440+480
2.0on 4.0off
480+620
0.5on 0.5off
38
Italy2
400
Continuous
400
Continuous
400
1.0on 2.0off
400
0.5on 0.5off
39
Belgium
425
Continuous
425
1.0on
0.25off
425
1.0on 3.0off
425
0.5on 0.5off
40
India2
350+440
Continuous
350+440
Continuous
350+440
0.4on 0.2off
0.4on 2.0off
400
0.75on 0.75off
1675
CPTG
Region Region
Code
Error Tone
Cadence
Freq.
(sec)
Confirmation Tone
Freq.
Feature Tone
Cadence
Freq.
(sec)
Cadence
(sec)
CCWT
Cadence
(sec)
Freq.
ICWT
Freq.
Cadence
(sec)
Region1
440
0.25on
0.25 off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
350+440
0.1on
0.1off
0.1on
2.7off
440
0.1on
2.9off
Region2
400
0.25on
0.25 off
400
0.1on
0.1off
400
1.5on
0.1off
400
0.2on
4.8off
400
0.2on
4.8off
Region3
440
0.25on
0.25 off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
440+480
0.1on
0.1off
0.1on
2.7off
440
0.1on
2.9off
Argentina
425
0.3on
0.4off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.3on
10.0off
425
0.1on
2.9off
Australia
425
0.375on
0.375off
425*25
0.1on
0.1off
425*
25
0.1on
0.9off
425
0.2on
0.2off
0.2on
4.4off
425
Continuous
Brazil
425
0.25on
0.25 off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.05on
1.0off
425
0.1on
2.9off
Canada
480+
620
0.25on
0.25off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
440
0.3on
10.0off
480+
620
0.5on
0.5off
China
450
0.7on
0.7off
450
0.1on
0.1off
450
0.1on
0.9off
450
0.4 on
4.0off
450
0.2on
0.2off
0.2on
0.6off
Egypt
450
0.5on
0.5off
425*50
0.1on
0.1off
425*
50
0.1on
0.9off
425*50
0.1on
0.1off
0.1on
2.7off
450
0.5on
0.5off
10
France
440
0.25on
0.25off
440
0.1on
0.1off
440
0.1on
0.9off
440
0.3on
10.0off
440
0.1on
2.9off
11
Germany
425
0.24on
0.24off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.2on .2off
.2on 5.0off
425
0.1on
2.9off
12
Greece
425
0.15on
0.15off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.3on
10.0off
0.3on
10.0off
425
0.15on
0.25off
0.15on
1.45off
13
India1
400
0.25on
0.25off
400
1.0on
4.0off
400*
25
0.1on
0.9off
400
0.2on
0.1off
0.2on
7.5off
400
0.15on
4.85off
14
Indonesia
425
0.25on
0.25off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.15on
0.15off
0.15on
10.0off
425
0.1on
2.9off
15
Iran
425
0.25on
0.25off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.2on
0.2off
0.2on
10.0off
425
0.1on
2.9off
16
Iraq
400
0.25on
0.25off
400
0.1on
0.1off
400
0.1on
0.9off
400
0.1on
0.1off
0.1on
2.7off
400
0.1on
2.9off
17
Israel
400
0.25on
0.25off
400
0.17on
0.14off
0.34on
5.0off
400
0.1on
0.9off
400
0.5on
10.0off
400
0.1on
2.9off
1676
CPTG
Region Region
Code
Error Tone
Cadence
Freq.
(sec)
Confirmation Tone
Freq.
Feature Tone
Cadence
Freq.
(sec)
Cadence
(sec)
CCWT
Freq.
Cadence
(sec)
ICWT
Freq.
Cadence
(sec)
18
Italy1
425
0.2on
0.2off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.4on
0.1off
0.25on
0.1off
0.15on
5.0off
425
0.1on
2.9off
19
Japan
400
0.25on
0.25off
400
0.1on
0.1off
400
0.1on
0.9off
400*25
0.5on
2.0off
0.05on
0.45off
0.05on
3.45off
400*
25
0.1on
2.9off
20
Kenya
425
0.2on
0.6off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.1on
0.1off
0.1on
2.7off
425
0.1on
2.9off
21
Korea
480+
620
0.3on
0.2off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
350+440
0.25on
0.25off
0.25on
3.25off
350+
440
0.1on
2.9off
22
Malaysia
425
2.5on
0.5off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.2on
0.2off
0.2on
5.0off
425
0.1on
2.9off
23
Mexico
425
0.25on
0.25off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.1on
0.1off
0.1on
2.7off
425
0.1on
2.9off
24
New
Zealand
400
0.25on
0.25off
400
0.1on
0.1off
400
0.1on
0.9off
400
0.2on
3.0off
0.2on
5.0off
425
0.1on
2.9off
25
Phillippines
480+
620
0.25on
0.25off
425
0.1on
0.1off
425
0.1on
0.9off
440
0.3on
10.0off
440
0.1on
2.9off
26
Poland
425
0.5on
0.5off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.15on
0.15off
0.15on
4.0off
425
0.1on
2.9off
27
Portugal
450
0.33on
1.0off
425
1.0on
0.2off
425
0.1on
0.9off
425
0.2on
0.2off
0.2on
5.0off
425
0.2on
1.4off
28
Russia
425
0.25on
0.25off
425
0.1on
0.1off
425
0.1on
0.9off
950
0.333on
1.0off
425
0.1on
2.9off
29
Saudi
Arabia
425
0.25on
0.25off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.15on
0.2off
0.15on
10.0off
425
0.1on
2.9off
30
Singapore
425
0.25on
0.25off
425
0.125on
0.125off
425
0.1on
0.9off
425
0.3on
0.2off
0.3on
3.2off
425
0.25on
2.0off
31
South
Africa
400
0.25on
0.25off
400*33
0.1on
0.1off
400*
33
0.1on
0.9off
400*33
0.4on
4.0off
400
0.15on
0.25off
0.15on
1.45off
1677
CPTG
Region Region
Code
Error Tone
Cadence
Freq.
(sec)
Confirmation Tone
Freq.
Feature Tone
Cadence
Freq.
(sec)
CCWT
Cadence
(sec)
Cadence
(sec)
Freq.
ICWT
Freq.
Cadence
(sec)
32
Spain
425
0.25on
0.25off
425
0.1on
0.1off
425
0.1on
0.9off
425
0.175on
0.175off
0.175on
3.5off
425
0.1on
2.9off
33
Thailand
400
0.3on
0.3off
400*50
0.1on
0.1off
400*
50
0.1on
0.9off
400
0.1on
0.1off
0.1on
2.7off
400
0.1on
2.9off
34
Turkey
450
0.2on
0.2off
.6on .2off
450
0.04on
0.04off
450
0.1on
0.9off
450
.2on .6off
.2on 8.0off
450
0.1on
2.9off
35
UAE
400
0.4on
0.35off
0.225on
0.525off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
350+440
0.1on
0.1off
0.1on
2.7off
350+
440
0.1on
2.9off
36
UK
400
0.4on
0.35off
0.225on
0.525off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
350+440
0.1on
0.1off
0.1on
2.7off
400
0.2on
4.8off
37
USA
480+
620
0.25on
0.25off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
440
0.3on
10.0off
480+
620
0.5on
0.5off
38
Italy2
400
0.25on
0.25 off
400
0.1on
0.1off
400
1.75on
0.1off
400
0.2on
2.5off
400
0.2on
0.2off
0.2on
2.5off
39
Belgium
425
0.167on
0.167 off
425
0.1on
0.1off
425
0.1on
0.9off
1400
0.175on
0.175off
0.175on
3.5off
440
0.1on
2.9off
40
India2
400
0.25on
0.25 off
350+440
0.1on
0.1off
350+
440
0.1on
0.9off
350+440
0.1on
0.1off
0.1on
2.7off
350+
440
0.5on
0.5off
1.0on
5.0off
1678
1679
Exit SE mode.
For SE commands to change Interrupt Request Timer and the Barge-In Timer, Transfer-On Busy Timer,
refer the relevant topics: Interrupt Request (IR), Barge-In and Call Transfer.
How to use
It is important that users of ETERNITY also get acquainted with the different Call Progress Tones played by the
system, so that they understand the meaning of the terms used for various tones. Therefore, ETERNITY makes it
possible for users to listen to the various Call Progress Tones.
Demonstration of Tones
It is possible to demonstrate Call Progress Tones to users by dialing the SE commands from EON or an SLT.
By default, the system will play each tone as demonstration for 30 seconds. The duration of demonstration can be
changed by setting the 'Tone Demo Timer' to match user preference (see "Changing CPT-related Timers using a
Telephone" above).
241. Time for which the system demonstrates the tone/ring to the user.
1680
Exit SE mode.
1681
How it works
For this feature to work,
the IP Address based call traffic restriction flag must be enabled for the VoIP Ethernet Port, and
the White List IP Address Table, that is, a list of IP Addresses and their respective Subnet Masks from
where the traffic is to be allowed, must be configured for VoIP Ethernet Port.
With flag enabled and the table programmed, traffic coming from all IP Addresses, other than those
programmed in the White List, will be blocked.
If the flag "Call Restriction based on IP Address" is enabled, but the White List IP Address Table is
blank, all incoming traffic will be rejected and it will not be possible to make calls on such a VoIP Card.
Call Restriction will be applied also on all the SIP Trunks which are assigned to the VoIP Ethernet Port.
How to configure
Decide on which of the VoIP Ethernet Ports IP Address based call traffic restriction is to be applied.
For each VoIP Ethernet Port you want to create a White List Table, make a three column table on a sheet
of paper.
VoIP Ethernet Port 1
Index
IP Address
Subnet Mask
1
2
3
4
5
1682
Make a list of IP Addresses. You are allowed to program a maximum of 10 IP Addresses. For each IP
Address enter the corresponding Subnet Mask address.
With the White List Tables ready, you may program the tables using Jeeves or a telephone.
The White List IP Address page for VoIP Ethernet Port 1 will open.
Select the Enable IP Address based call traffic restriction check box.
Enter the IP Address and the respective Subnet Mask in the table. You may refer to the table you
prepared on paper.
To program the White List IP Address for another VoIP Ethernet Port, click the link of the desired VoIP
Ethernet Port.
Repeat the same steps as described above to program the White List IP Addresses and the corresponding
Subnet Masks.
You may logout of Jeeves after you have finished programming the White List IP Address for the desired
VoIP Ethernet Ports.
1683
1684
Exit SE mode.
Call Taping
What's this?
Call Taping allows extension users to record the telephone conversations they have with other extensions or
external numbers, without the opposite party coming to know about it.
Feature is useful for keeping records of important conversations. For this feature to work, the system must have a
VMS Card installed in it.
Call Taping can be done for:
Calls are taped in a common mailbox assigned to this feature. Extension users with access to the mailbox can
retrieve and listen to the recorded conversations. The tapped calls are stored along with the call details, that is, the
time and date of the call, the calling number and the called number.
To be able to record external incoming and outgoing calls, a list of phone numbers (both incoming and outgoing)
must have been programmed in the system.
Incoming calls without Calling Line Identification (CLI) can also be taped. For this, the flag 'Tape Calls Without CLI?'
must be enabled in the Station Advanced Feature Template.
To be able to record internal calls, the 'Call Taping for Internal Calls flag' must be enabled on the extension which
desires to use this feature.
Matrix Comsec is not responsible for any mis-/abuse of this feature by users.
How it works
A calls C
The system matches the dialed number with the numbers in the Number List - Outgoing Calls.
The system finds a match. When speech is established, the system starts recording the conversation
between A and C automatically in E's mailbox.
1685
D calls B
The system matches the incoming number with the numbers in the Number List-Incoming Calls.
On finding a match, system records the speech between D and B in E's mailbox.
Call Taping Beeps will be played to D and B only if this feature is enabled.
If an incoming call does not have any CLI, the system checks the flag 'Tape Calls without CLI' in the
Call Taping parameters.
A calls B
If the flag is enabled, the system records the speech between A and B in E's mailbox.
Call Taping Beeps will be played to A and B only if this feature is enabled.
If the flag is disabled, the speech between A and B will not be recorded.
The same is done when B calls A. The speech will be recorded in E's mailbox.
Feature Interaction:
Conversation Recording: If Call Taping and Conversation Recording both are enabled for an
extension, then priority is given to Conversation Recording.
How to configure
The functioning of this feature requires the following parameters to be programmed:
Call Tapping Mailbox Port: You must program the software port of the extension in whose mailbox the
calls are to be taped.
Taping Calls without CLI Flag: This flag must be enabled if you want calls without CLI to be taped.
Number Lists for Incoming and Outgoing Calls: The Call Taping Number List-Incoming Calls and Call
Taping Number List-Outgoing Calls are to be programmed so that the system can match the phone
numbers of the incoming and outgoing calls and initiate the recording of the speech.
On a sheet of paper, prepare the Call Taping List Incoming and Call Taping List Outgoing.
1686
You can add as many as 999 numbers to each list. Each entry on these Lists is stored in a serial order
against a 'Location Number'. So, draw three columns and enter the numbers against a location number
from 01 to 999.
Location
001
002
:
:
999
Use this table to program the Number lists. By default Number List 09 is assigned for numbers of incoming
calls, and Number List 10 is assigned to numbers of outgoing calls.
Call Taping Internal Flag: This flag is to be enabled in the Station Advanced Feature Template applied on
those extensions that are to be allowed Call Taping of internal calls, that is, calls made or received by them
to or from other extensions.
Call Taping Recording Beeps: This flag is to be enabled if Call Taping Beeps are to be played to the two
parties in speech. Enable Call Taping Beeps only when you want indication of speech recording to the two
parties in speech. By default, this flag is enabled.
Under Configuration.
1687
Enter the Access Code of any SLT, DKP, SIP Extension, ISDN Terminals, Department Group, General
Mailbox or Extension over QSIG, whose mailbox you want to assign for Call Taping.
1688
On the same System Parameters page, go to Play Beep when Call Taping/Conversation Recording
Starts and enable/disable beeps by selecting/clearing the check box.
By default, Station Advanced Feature Template 01 is assigned to all extensions of the ETERNITY. If you
want to assign Call Taping facility to all extensions, then program the Call Taping related flags and Number
Lists, in Template 01.
However, if only selected extensions are to be assigned this feature, then:
1. Prepare a separate Station Advanced Feature Template.
2. Set the Call Taping Parameters in this template.
3. Apply this new template to all SLT and DKP extensions that are to be allowed this feature.
Scroll with the horizontal bar to reach the Call Taping column of the Template Number assigned to the
extensions.
If you want calls without CLI to be taped, click the check box to enable the flag - Tape Calls coming
without CLI
To program the list of numbers of incoming calls, click the link Number List- Incoming Calls.
1689
Click the default number list 9-10 link assigned to Call Taping, then click the 001-255 link of the default.
The Number Lists 9 and 10 will open.
If the same incoming and outgoing numbers are to be programmed for all extensions, you may simply
program the default Number lists 09 and 10.
If different incoming and outgoing numbers are to be programmed for different extensions, then
prepare different number lists.
Enter the List of Incoming Numbers that the system should match in List No. 09.
Enter the List of Outgoing Numbers that the system should match in List No. 10.
You can program as many as 999 numbers in each list. Each entry on these Lists is stored in a serial order
against a 'Location Index, starting from 001-999'. There are 250 Location Index on each page on your
screen. To go to the next set of Location Index, for instance, 251-500, click the link under 09-10.
1690
Click Submit at the bottom of the page to save your number lists.
Follow the same steps to program a different Call Taping number list. But ensure that the different List
number you programmed is entered in the Station Advanced Feature Template applied to the extensions.
If you want calls between extensions to be taped, click the check box to enable the flag - Call Taping for
Internal Calls.
Click Submit at the bottom of the page to save changes to the template.
Now, apply the programmed template to DKP and SLT extensions to which you want to provide the Call
Taping facility. Refer the topic Station Advanced Feature Template for programming instructions.
Code
Flash (F)
#2
Pause (P)
#3
#4
#5
#6
#7
#8
Dot (.)
#9
##
**
1691
To enable Call Taping Internal Flag in a Station Advanced Feature Template, dial:
5602-1-Template Number-Feature Number-Code
Where,
Template Number is from 01 to 50.
Feature Number for Call Taping Internal Flag is 20.
Code is
0 for Disable
1 for Enable
To enable Tape calls coming without CLI Flag in a Station Advanced Feature Template, dial:
5602-1-Template Number-Feature Number-Code
Where,
Template Number is from 01 to 50.
Feature Number for Tape calls without CLI is 17.
Code is
0 for Disable
1 for Enable
Exit SE mode.
For SE commands for applying the programmed template to DKP and SLT extensions, refer the topic
Customizing Station Advanced Feature Template using a Telephone.
Also refer the topic Number Lists to know more.
How to use
This feature works automatically on extensions which have the related Call Taping parameters programmed in their
Station Advanced Feature Template.
Call Taping conversations are recorded in a single, common mailbox. These can be accessed directly by the
Mailbox Owner (user of the extension to which this common mailbox is assigned). Other extension users can also
access this common mailbox by calling the Voice Mail System.
Instructions for accessing the mailbox are provided separately for these two groups of users.
1692
If the common mailbox is password protected, make sure that you provide the password to all extension
users who are to be provided access to this mailbox.
The above instructions contain the default access codes. Check with your System Engineer, if these have
been changed and use the current access codes.
242. This is the default Voice mail Feature Access Code. Verify with you System Engineer if this has been changed and use the new
code.
243. Only if the mailbox is password protected, you will be prompted to enter the password.
244. Only if the mailbox is password protected, you will be prompted to enter the mailbox password.
1693
Call Toggle
What's this?
Call Toggle allows you to have two simultaneous telephone conversations, talking to two persons alternately.
Call Toggle is also referred to as Hold-Consult or Call Splitting,
You can toggle between:
How it works
1694
The party put on Consultation Hold during Call Toggle cannot hear the conversation between the other
two parties.
You can also toggle between an incoming internal/external call (indicated by call waiting tone) and an
internal/external call you are currently in speech with.
You can also answer an incoming 'Interrupt Request' call and toggle between the interrupting extension
and the extension you were in speech with.
You can convert a Call Toggle into a three-party conference by dialing Flash-*3.
You can transfer the call you are currently in speech with to another extension.
You can park the call you are currently in speech with.
How to configure
Call Toggle is a Class of Service (CoS) dependant feature.
In the default Station Basic Feature Template 01 assigned to all extensions of ETERNITY, Call Toggle is included in
the 'Basic Features' assigned to all CoS groups, including the default CoS group 01. So, all extensions of
ETERNITY can use this feature.
As Call Toggle is a part of the set of 'Basic Features', you cannot disable this feature selectively in the COS of
extensions, without disabling the entire set of features.
No specific programming is required for this feature, except for programming a DSS key for Call Toggle, if required.
Refer the topic DSS Keys Programming for instructions.
How to use
For EON and Extended IP Phone Users
Call Toggle between two internal calls:
1695
1696
Call Traffic
What is this?
Call traffic measurement feature of ETERNITY gives a graphical representation of the time duration for which
extensions and trunk ports remained Off-hook. The data is represented in a bar graph format for the SLT, DKP,
Magneto, ISDN Terminals, CO, E&M, T1E1, Mobile and LD ports as well as SIP Extensions.
In the graph, the duration is shown in Hours along the Y-axis while the extension(s)/ trunk(s) port names are shown
along the X-axis. The following call traffic measurement data (for each Extension and Trunk port separately) is
displayed:
Time for which each extension remained in speech for making/receiving Internal calls.
Time for which each extension remained in speech for making/receiving Trunk calls (Incoming +
Outgoing).
Time for which each trunk port remained in speech for Incoming calls.
Time for which each trunk port remained in speech for Outgoing calls.
This time is measured in terms of the number of hours, and the traffic is measured for last 24 hours.
You can view this traffic information in graphical format on the Jeeves; see the illustration below for Call Traffic
information generated for SLT extension.
The two-color bars distinguish Internal calls (green) and Trunk calls (red)
1697
You can also export this information in a database readable format like Microsoft Excel. The call traffic information
files can be saved on a local disk.
How to use
To view Call Traffic in graphical format, you need to log into the Jeeves as System Engineer.
Go to the configuration page of the desired trunk or extension port for which you want to view Call Traffic data.
On the page, click the Call Traffic button.
The call traffic will be generated as a graph.
If there is no call traffic usage data is available for the extension/ trunk you are currently viewing, the page
will not show any bars.
Click the Refresh button to get the latest 24hrs call traffic statistics. The application filters the SMDR records for
last 24hrs and graph is generated accordingly.
1698
Click the Export button to export the data you are currently viewing. The FTP server window 'Log On
As' will open.
Type the Password for FTP login from SE mode (Default: 1234).
On successful login a new window appears from where you can choose actions whether to open this
file or save it in .xls format. Select proper action as per your requirement.
1699
1700
Save this file on your local disk with proper name as shown in the following image.
If you open this file in MS Excel then it may look as shown in the following image.
The file contains the date and time when the call traffic data is calculated and the names of the extensions/
trunks for which it is calculated.
You can save this file on your local disk from Excel window also.
1701
Call Transfer
What is this?
Call Transfer enables you to relocate an existing call from an extension or trunk to another extension or to an
external number. Calls can be transferred after notifying the other extension/external number about the impending
transfer or can be transferred directly without notification.
The types of Call Transfer ETERNITY offers are:
Call Transfer - Screened: The Operator puts the caller on Consultation Hold, dials the desired party's
extension, and informs the desired party of the impending transfer. If the desired party chooses to accept
the call, the call is transferred over to them.
Call Transfer - While Ringing: The Operator puts the caller on Consultation Hold, dials the desired
party's number and transfers the call when the desired party's extension starts ringing.
This feature is used when there are several other calls to be attended and the Operator cannot wait for the
desired party to answer.
Call Transfer - On Busy: The Operator puts the caller on Consultation Hold, dials the desired party's
number and transfers the call even when the desired party is busy in speech with another person. The
busy extension gets intrusion tone and can choose to answer the intruding (transferred) call.
Call Transfer - Trunk-to-Trunk: An external call is transferred on to another trunk line. The Operator puts
the external caller on Consultation Hold, dials the desired party's external number, and transfers the call
after or without notifying the desired party of the impending transfer.
Trunk-to-Trunk call transfer may be used to transfer incoming calls for out-of-office extension users to their
cell phones, or to connect personnel at remote or distant locations. For instance: an out-of-office executive
who does not have long distance dialing permission can call the office and request the operator to connect
him to the desired party on a trunk line.
1702
Blind Transfer to VMS: The Operator puts the caller on Consultation Hold, dials the feature access code
for Blind Transfer to VMS, dials the desired party's number, and transfers the call. The call is transferred to
the mailbox assigned to the desired party. The caller may leave a message in the mailbox.
Call Transfer is not exclusively an Operator feature, though it is used mostly by Operators. Calls can be
transferred by any extension to another extension or external number, if "Basic Features" are allowed
in Class of Service of the transferring extension.
ETERNITY enables SIP extensions to resume a transferred call before it has been answered by the
transfer target (which may be an extension or an external number). For a list of IP phones on which this
feature has been tested, see ETERNITY Features tested on IP Phones of different Brandsin the
Appendix.
ETERNITY allows Semi-attended Transfer and Transfer on Conference Hangup on SIP Trunks. For a
list of IP phones on which this feature has been tested, see ETERNITY Features tested on IP Phones
of different Brandsin the Appendix.
How it works
A and B are extension users.
C is an external caller.
D is an external number.
If B does not accept the call, Operator may dial Flash to retrieve the call and speak to C.
The Operator can also abort call transfer while B's phone is ringing by dialing Flash. The Operator gets
connected to C.
3. Transfer On Busy:
1703
C and D are now in speech for the duration of the Trunk-to-Trunk Inactivity Timer245.
A warning tone is given at the end of the Trunk-to-Trunk Inactivity timer (programmable; default: 2
minutes). On expiry of this timer, the call is disconnected.
To extend the call, either C or D must dial any digit in tone (DTMF), except '##'.
If A does not have a mailbox assigned, the Operator will get an error tone while transferring the call.
The Operator may retrieve C's call by pressing Transfer Key/Flash /Call Appearance key.
Feature Interactions:
CLIP and Caller ID Presentation while Transfer: ETERNITY provides the flexibility to display either the
extension number that is transferring the call or the held party's number, that is, the number of the party
that is about to be transferred. Refer Calling Line Identification and Presentation (CLIP).
Privacy: Call Transfer-On Busy will not work if the busy extension has Call Privacy from intrusion Tone in
its Class of Service.
DND: Call Transfer will not work if the destination extension has set DND.
245. The process of Trunk-to-Trunk transfer takes place outside of the PBX. So, the PBX will not know which of the two trunks have
gone ON-Hook. Hence the call is automatically disconnected when the Trunk-to-Trunk Inactivity Timer expires.
1704
How to configure
To be able to use Call Transfer, this feature must be enabled in the Class of Service group of the extensions to be
allowed this feature. The default values of the related Timers may be changed, if required.
To be able to use Blind Transfer to VMS, the extensions must be assigned a mailbox in the VMS of the system. For
instructions, refer to Configuring Voice Mail System .
Transfer While Ringing Timer: This timer is related to Call Transfer - While Ringing. It is the time for
which the system rings the extension. By default it is set to 30 seconds. At the end of the timer the call is
returned to the transferring extension.
Transfer on Busy Timer: This timer is related to Call-Transfer on Busy. It is the time for which the system
waits for the busy extension to respond to the intrusion tone. By default the timer is set to 30 seconds. At
the end of the timer the call is returned to the transferring extension.
Trunk to Trunk Inactivity Timer: This is the time duration after which the system disconnects the call
transferred from one trunk line to another. By default it is set to 2 minutes. At the end of the timer the call is
disconnected, if either party does not dial digits to extend the call. This Timer is relevant for CO to CO and
CO to E&M calls only.
1705
Under Configuration, click System Timers and Counts to open the page.
Scroll to reach Other Features and change the values as required for Call Transfer related timers.
Enter SE mode.
To program Transfer While Ringing Timer:
Dial command 3806-Seconds
Where,
Seconds is from 001 to 255 seconds.
Default: 030
To program Transfer on Busy Timer:
Dial command 3807-Seconds
Where,
Seconds is from 001 to 255 seconds.
Default: 030
To program the Trunk-to-Trunk Inactivity Timer:
Dial command 3808-Minutes
Where,
Seconds is from 001 to 255 minutes.
Default: 2 minutes
1706
Exit SE mode.
How to use
For EON and Extended IP Phone Users
Extension to Extension:
Extension to Trunk:
246.Trunk Access Code: users worldwide may dial a code from 0, 5, 61, 62, 63, and 64. Users in USA may dial a code from 0, 9, 81, 82,
83, and 84.
1707
OR
247.Trunk Access Code: users worldwide may dial a code from 0, 5, 61, 62, 63, and 64. Users in USA may dial a code from 0, 9, 81, 82,
83, and 84.
1708
1709
How it works
When CLIP is enabled on a trunk,
It sends this information to the landing extension/Operator along with the ringing signal.
In case of, Internal calls the calling extension's name and number both are presented to the called
extension.
In the case of External calls, only the number will be displayed on the landing/Operator extension.
When the landing extension/Operator transfers the incoming call to an extension, putting the external
caller on hold, the system sends this information to the extension to which the call is transferred.
During the transfer, the number of the landing extension/Operator will be displayed on the transfer
destination extension.
On successful call transfer, the caller's number will be displayed on the transfer destination extension.
In the case of Call Transfer, the system also provides the option of displaying to the destination extension either the
number of the party that is put on hold to be transferred, that is, the Held Party OR the number of the Transferring
Party, while the call transfer is taking place. This feature is called Caller ID Presentation while Transfer.
It is also possible to remove and replace the '+' character received as CLI on telephones that do not support CLIP
starting with this character.
For example, the GSM network sends the calling party number with '+' as the prefix. If the telephone connected as
extension does not support this, it will not present the CLI of the caller. To overcome this, ETERNITY provides you
the option of replacing '+' with an appropriate number string which these telephones can display.
1710
Feature Interactions:
CLIR: CLIP and Caller ID Presentation while Transfer will work only if CLIR is not enabled on the
extension that has transferred the call. Refer the topic Calling Line Identification Restriction (CLIR).
Q-Sig: When two PBXs - PBX A and PBX B are networked using Q-Sig, and an extension of one PBX, for
instance, PBX A transfers a call to an extension of PBX B by putting the caller on hold, the CLI presented
on the extension of PBX B will be according to the type of Caller ID Presentation while Transfer set on the
transferring extension of PBX A.
How to configure
The functioning of this feature is controlled by two parameters: CLIP Type and Caller ID Presentation while
Transfer.
If you want to replace '+' characters received as CLI on telephones that do not support CLI prefixed with this
character, you must program the relevant flag and the desired number string in the 'System Parameters'.
All these parameters can be programmed using Jeeves and a Telephone.
CLIP Type
If SLTs supporting CLI are connected to the ETERNITY, the System Engineer must select a signaling protocol for
CLI in the SLT Hardware Template applied on the SLT extensions. By default SLT Hardware Template 01 is
assigned to all SLT extensions. The default CLIP Type in Template 01 is 'DTMF'.
There is no need to select a CLIP Type in the default Hardware Template 01, if all the SLTs support DTMF protocol.
If all SLTs support a different CLIP Type say FSK-Bellcore, you may simply select this CLIP Type in the default
Hardware Template applied on all SLT extensions.
However, if certain SLTs support a particular CLIP type and some support a different CLIP type, then create
separate SLT Hardware Templates with different CLIP types and apply them to the appropriate SLTs.
For example, you may select SLT Hardware Template 02 with FSK V.23 and SLT Hardware Template 03 with FSK
Bellcore, and Template 04 with DTMF as the CLIP Type and apply each template to the SLTs as per the CLIP Type
they support.
1711
1712
Exit SE mode.
Under Configuration, click Station Advanced Feature Templates to open the page.
In Caller ID Presentation while Transfer field, select the desired option: Held Party or Transferring
Party.
1713
Exit SE mode
Refer the topic Station Advanced Feature Template for instructions assigning Templates to extensions.
1714
Enable the Replace '+' from CLI flag by selecting the check box.
Enter the desired number string in the field Replace '+' from CLI with the number string.
Exit SE mode.
1715
How it works
Now,
Feature Interactions:
CLIP and Caller ID Presentation while Transfer: Both these features will not work if CLIR is enabled.
How to configure
CLIR and CLIR Override are Class-of-Service-dependant features. Extensions that are to be allowed these
features, must have them enabled in their Class of Service (CoS) group.
Decide which extensions should be allowed CLIR and which should be allowed CLIR Override.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
ETERNITY. Template 01 is assigned CoS group 01 in which both CLIR and CLIR Override are disabled. Thus,
none of the extensions of the ETERNITY can suppress their CLI or force any other extension to display its CLI.
If you want to enable both features on all extensions, simply enable CLIR and CLIR Override in the default CoS
group 01.
If you want to allow CLIR to all extensions, but not allow CLIR Override to any extension, simply enable CLIR in the
default CoS group 01.
1716
If you want to allow CLIR and/or CLIR Override to selected extensions, only, then follow these steps:
1. Define a new CoS group with CLIR/CLIR Override enabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
3. Assign this new Template to the extensions to which CLIR/CLIR Override is to be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions and
programming.
How to use
For EON and Extended IP Phone Users
To enable CLIR
To disable CLIR:
Lift handset.
Dial 103-1.
You get confirmation tone.
Replace handset.
To disable CLIR:
Lift handset.
Dial 103-0.
You get confirmation tone.
Replace handset.
248. System Engineer is recommended to assign a DSS Key with LED to this feature. When the assigned DSS key is pressed, it will
glow red indicating that CLIR is enabled.
249. If a DSS key with LED has been assigned, when you press the key again, the LED will be turned off indicating CLIR is now
canceled.
1717
Auto Answer
Auto Call Back
Auto Redial
Background Music
Call Forward
Do Not Disturb
Hot Line
Trunk Reservation
Walk-In Class of Service
How to use
The extension users can cancel all features set on their extension from their own extension or these can be
cancelled from SA Jeeves for any extension user.
Dial 1051.
You get confirmation tone and confirmatory message on your phone display.
Go idle or wait for dial tone.
1718
Lift handset.
Dial 1051.
You get confirmation tone.
Replace the handset.
Click Extension.
Now, Search Extension, by entering either extension number as Extension Number, or by entering the
name of the extension as Extension Name.
Click Submit.
Click the Cancel All Features button. The following features will get canceled if any is set on this
extension user:
Walk-in Class of Service
Do Not Disturb
Call Forward
Call Forward - Scheduled
Hotline
Auto Answer
Auto Redial
Auto Call Back
Trunk Reservation
Background music
1719
How it works
The list of all features allowed to an extension is referred to as 'CoS group'. There are 20 CoS groups numbered
from 01 to 20.
In each CoS group there are 56 features, which are identified by 2-digit numbers, from 01 to 56. These are referred
to as the 'CoS Feature Numbers'.
Each extension port of the PBX has an associated CoS group that indicates which features of the PBX the port is
allowed to access.
The CoS group of an extension port is defined in the Station Basic Feature Template applied to that extension
port. It is defined for each "Time Zone", namely, working hours, break hours, and non-working hours, in the
Template.
A feature can be allowed or denied to an extension by enabling or disabling it in the CoS group of the Station Basic
Feature Template applied to that extension.
The same CoS group uniformly to all extensions ports for all Time Zones. Doing so, all extensions can access the
same set of features in all time zones. For example: CoS group 03 is assigned to all extensions for Working, Break
and Non-Working hours.
A different CoS group for each Time Zone can be assigned to all extension ports. Doing so, all extensions can
access only those features allowed for the particular Time Zone.
For example: All extensions are assigned CoS group 03 for Working, CoS group 04 for Break hours and CoS group
05 for Non-Working Hours.
Different CoS groups can be assigned to different extension ports, for all or for different Time Zones. Doing so,
each extension can access a different set of features in each Time Zone.
1720
For example: extensions 3001 to 3010 are allowed CoS group 03 for all Time Zones, while extensions 3011 to 3015
are assigned CoS group 03 during Working Hours, and for the Non-Working and Break Hours, they are assigned
CoS group 04 and 05 respectively.
The following features can be set/cancelled on extensions from the SA mode, regardless of whether these
features are allowed or denied in the CoS assigned to the extensions:
Call Forward
DND
Dynamic Lock and Timer
Hotline
Basic Features
A set of features including Internal Call, Call Hold, Call Toggle, Call Transfer, Department Call, Operator Access,
Redial, and Call Mute defined as Basic Features and allowed in all CoS groups.
It is not possible to enable or disable selectively any of the features included in "Basic Features".
How to configure
The table below presents the CoS groups from 01 to 20 with the list of 01 to 58 features supported on the
extensions.
Account Code
System Admin. Mode Access
System Admin. Extension
Auto Call Back-Busy
Auto Call Back-No Reply
Auto Redial
Auto Redial Priority
Background Music
Barge-In
Call Forward
Call Park
Call Pickup
Change Room Clean Status
Global Directory Programming
CLIR
CLIR Override
Closed User Group (CUG)
Trunk Call Waiting
Conference
Continued Dialing
Conversation Recording
Decrement Dynamic Lock Timer
for Internal Calls
DISA
Do Not Disturb
1721
Feature
Feature Name
Number
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
a. Basic Features includes: Internal Call, Call Hold, Call Toggle, Call Transfer, Department Call, Operator Access,
Redial, Mute.
1722
CoS group number 01 is assigned for all Time Zones in the default Station Basic Feature Template 01
assigned to all extensions of the ETERNITY.
CoS group number 19 and 20 are assigned when the Hospitality Application of ETERNITY is used. See
ETERNITY Hospitality System Manual.
Against each extension name on the list, write the features needed for each Time Zone. You will notice
that the features needed by many extensions are identical.
List the common features to be allowed to and features to be denied to all extensions. Assign a CoS Group
Number to this list.
Are there any other features, in addition to those on the common list, which you want to allow to selected
extensions?
If yes, extend the common list you prepared by adding the features to be allowed to selected extensions.
Assign a CoS Group Number to this extended list.
You can prepare different CoS Groups for different Time Zones and assign a number to each group.
For example, you may end up creating five different CoS groups. The First group may contain none of the
features. The Second group may contain the most common features like Call Forward, Call Transfer,
Internal Dialing, etc. The Third group may contain more advanced features, and the Fourth group may
contain even more advanced features. The Fifth group may contain all the features.
When you are finished preparing the CoS groups you need, program the CoS groups using Jeeves or by
issuing SE commands from a Telephone (DKP or SLT). See below for instructions.
Now, the CoS groups to be assigned to extensions must be programmed in the Station Basic Feature
Template applied to the extensions. This can be done using Jeeves or by issuing SE commands from a
Telephone (DKP or SLT). See below for instructions.
1723
The default CoS groups from 01 to 20 appear. The check boxes selected under each CoS group column
indicate that the feature is enabled in that CoS group. The default CoS groups meet the requirements of
most extension users. Check the default CoS groups whether the features you want to allow are enabled
and features you want to deny are disabled.
To enable a feature in a CoS group, select corresponding check box in the CoS group. To disable a
feature simply clear the check box. For example: to enable DND-Override in CoS group 01, select the
check box against DND-Override in CoS group 01. To disable clear the check box again.
1724
Exit SE mode.
The default CoS group assigned to each time zone, that is, working hour (WH), non-working hour (NH) and
break hour (BH), appears under Class of Service in each Template.
To assign a CoS group to a Station Basic Feature Template, enter the CoS group number for each time
zone under Class of Service.
1725
extensions. For example: to assign CoS group 04 to all time zones in Template Number 01; enter 04 under
WH, NH and BH.
If all extensions are to be allowed a different set of features in each Time Zone, enter the CoS group for
each Time Zone. For example: to assign CoS group 03 in working hours, 04 in Break Hours and 05 in NonWorking hours in Template 01, enter 03, 04, 05 under WH, BH and NH respectively.
1726
If a set of features is to be allowed to select extensions only, assign the CoS group with these features
enabled to a separate Station Basic Feature Template. Apply this template to the select extensions which
are to be allowed this CoS.
For example: To assign all features to extensions, create a CoS group with all features enabled, CoS
group 07. Select a different Station Basic Feature Template, for example 05. Enter CoS Group 07 in all
Time Zones in Template 05. Apply Template 05 to the software ports of the extensions that are to be
assigned all features.
Remember to click Submit to save the changes you make on every page.
1727
Similarly, to program CoS Group 03 in working hours, 04 in Break Hours and 05 in Non-Working hours,
dial:
5502-1-06-03-03 to program CoS group 03 for Working Hours.
5502-1-06-04-04 to program CoS group 04 for Break Hours.
5502-1-06-05-05 to program CoS group 05 for Non-Working Hours.
Exit SE mode.
After you have programmed the CoS group in the Station Basic Feature Template, you must assign this
template to the stations. Refer the topic Station Basic Feature Template for instructions on applying
templates on SLT, DKP, ISDN Terminal and SIP extensions.
Finally, test the CoS programmed for each extension by invoking the features from each extension.
1728
How it works
A, B, C are extensions. D and E are external callers.
Calls made by D are to be landed on A.
Calls made by E are to be landed on B and C.
The CLI of D and E and their corresponding landing destinations should be entered in the CLI Based Routing
Table.
CLI Based Routing should be enabled on the desired trunks for each Time Zone (working hours, break hours and
non-working hours).
The system can match the incoming call CLI with the numbers configured in the CLI table in two ways, that is,
Match from last digit of CLI or Match from first digit of CLI.
If you select Match from last digit of CLI, this is how the call will be routed:
D calls on a trunk of ETERNITY.
The system checks if CLI Based Routing is enabled on the trunk for the current time zone.
If CLI Based Routing is enabled on the trunk, the system checks the numbers stored in the CLI Based
Routing table.
D's number is found in the CLI Based Routing Table.
The system checks the destination number stored against D's CLI.
A's number is found as the destination extension.
The system lands the call on A.
If you select Match from first digit of CLI, this is how the call will be routed:
D calls on a trunk of ETERNITY.
The system checks if CLI Based Routing is enabled on the trunk for the current time zone.
Then the system checks if the parameter Replace '+' from CLI, is enabled.
If enabled, the system will replace + sign received in the incoming CLI with the number string
configured in the Replace '+' from CLI with the number string. To know more, see System
Parameters.
Now the system checks if Incoming CLI Modification is enabled.
If enabled the system modifies the incoming number according to the parameters configured in the
Incoming CLI Modification. To know more, see Incoming CLI Modification in System Parameters.
The system matches the modified number string with the numbers stored in the CLI Based Routing
table.
D's number is found in the CLI Based Routing Table.
The system checks the destination number stored against D's CLI.
A's number is found as the destination extension.
1729
If D's number does not exist in the CLI Based Routing Table, the call will be routed according to the incoming
call management logic.
How to configure
For this feature to work, you must do the following:
enter the numbers of the calling parties and the numbers of the corresponding destination extensions in
the CLI Based Routing Table. You can store up to 2000 numbers in the CLI Routing Table.
enable CLI Based Routing on the desired trunks according to time zones in their Trunk Feature
Template.
On a sheet of paper, create a 5-column table, as illustrated below. Each calling party number in the CLI
table is stored a location index in the system. Enter the telephone numbers and names of the calling
parties and the corresponding landing destinations, that is, the Port Type and Port Number. The Port Type
may be SLT, DKP, ISDN Terminal, SIP Extension, a Routing Group or Virtual Extension or a Voice Mail
Auto Attendant Profile.
The 'Name' field is for identifying the entry. When placing a call on the destination extensions, both the
number and the 'Name' are presented in the CLI.
Determine the method which the system should use to match the incoming CLI with the numbers in the
table, that is, Match from last digit of CLI or Match from first digit of CLI Based Routing Table.
Index
Telephone Number
Name
Port Type
Port Number
2640459
MidasBiz
Routing Group
02
022281110001
Jet Set
SLT
004
:
10
1730
:
2640075
Bacchus
Routing Group
03
In Method for matching received CLI, select the method according to which you want the system to
match the received CLI with the numbers stored in the CLI table. You can select:
Match from last digit of CLI
Match from first digit of the CLI
The method you select will be applicable to all the numbers configured in the CLI Based Routing Table.
In the CLI table each number is to be stored at a Location Index numbered from 0001 to 2000.
There are 100 entries on each page. To go to the next 100 Index numbers, click the tabs 0101-0200, 02010300, 0301-0400 , 0301-0400.................................1901-2000.
At each Location Index, enter the information for the following parameters:
Calling Partys Number: enter the number of the calling party, not exceeding 16 digits. You can also
enter '+' in the number string.
1731
Calling Partys Name: enter the name of the calling party. You can enter a maximum of 8 characters
in this field.
Port Type: select the landing destination extension. It may be an SLT, a DKP, an ISDN Terminal, SIP
extension, a Routing Group, a Virtual Extension or a Voice Mail Auto Attendant Profile.
Port Number: enter the software port to which the landing destination SLT/DKP/ISDN Terminal/SIP
Extension is connected.
If you selected a Virtual Extension as the landing destination, enter the software port of landing
destination of the Virtual Extension.
If you have selected a Routing Group as the landing destination, enter the number of the Routing
Group (01 to 96) in this field.
If you select Voice Mail Auto Attendant Profile, enter the Profile (01 to 16) number in this field.
Enable CLI Based Routing for the desired Trunks in their Trunk Feature Template. Refer the topic
Customizing Trunk Feature Templates for instructions.
1732
Exit SE mode.
1733
Clock Synchronization
What's this?
When data is transmitted from the ETERNITY to external lines or when ETERNITY receives data from the external
lines, it is necessary that the transmitter and receiver be properly synchronized. If not clock slips can occur. A clock
slip can generate a loss or addition of data to the data stream.
How it works
This can be done in three ways viz. using the data clock or using the external clock (clock is sent by the
network on a dedicated cable pair) or using the internal clock. ETERNITY does not support external clock.
When the ETERNITY is connected to the PSTN, then it is recommended to extract the clock from the
incoming data whereas if the ETERNITY is used to form a private network, you are recommended to use
the internal clock. For example, if a private network is formed by connecting three ETERNITY systems,
then one system should be programmed as master clock whereas other two should be programmed in the
slave mode.
If two or more T1E1 Ports are connected to the PSTN (or a Private Network) then in such case, clock will
be extracted from the first T1E1 Software port whereas the transmit data on all other ports whether
connected to PSTN or private network will be clocked as per received.
How to configure
Configuring Clock Synchronization using Jeeves
1734
Clock Source - Priority 1to 4: In the Priority levels, Clock Source Priority-1 to Clock Source Priority-4,
select the Clock Source option as per your preference.
By Default: Priority 1 is T1E1-001, Priority 2 is T1E1-002, Priority 3 is T1E1-003 and Priority 4 is T1E1-004.
Clock Synchronization Frequency: Select Clock Synchronization Frequency in this field. You can select
from the following options:
8 KHz Derived
8 KHz
2.048 MHz
1.54 MHz
By Default, Clock Synchronization Frequency is set to 2.048 MHz for India and other countries except
USA. For USA, it is set to 1.54 MHz.
To syncronize the clock with 2.048 MHz as the source clock frequency, make sure you have the Switch
Card with firmware V5R10 or later and the T1E1 Card with firmware V3R10 or later, Design Version:
D-101-017-02-03; CPLD Version of CPLD1 and CPLD2: V2R2.
PLL Locking Mode: Depending upon the speed required for clock synchronization, select the speed for
PLL Locking Mode. You can set PLL Locking Mode to either fast or slow. By default, it is slow.
PLL TIE Control: You can enable or disable PLL TIE Control. By default, it is disabled.
1735
PLL Operating Mode: Select the PLL Operating Mode in this field. You can select from the following
options:
Normal
Hold Over
Free Run
By default, PLL Operating Mode is Normal.
You can program PLL TIE Control and PLL Operating Mode only using Jeeves.
Meaning
Port Offset
05
T1E1
1-8
04
BRI
01-32
00
Null
000
T1E1-1
T1E1-2
T1E1-3
T1E1-4
The system checks this table for a master lock. If none of the ports is synchronized out of this table, the system
gives priority to the internal clock. If any one port is synchronized, the system selects that port as a system clock
master. Here index is given priority, that is, if the second port of this table is selected as clock master and suddenly
first port is synchronized, then the system changes its master from 2nd port to first port. Now if first port has lost its
synchronization then in this case again, the second port is selected as master clock of the system.
1736
Meaning
8 KHz Derived
8 KHz
2.048 MHz
1.54 MHz
By Default, 'System Clock Synchronization' is 2.048 MHz for India and other countries except USA.
For USA, default 'System Clock Synchronization' is 1.54 MHz.
This command is applicable only for 'ETERNITY GE', when Software version/revision 'V8R6' is used.
Selecting option 1('8 KHz Derived'): If you are using, the Software version/revision 'V8R6' onwards,
with CPLD version/revision 'V1R2 or earlier', the System Clock Synchronization will be done only at '8
KHz Derived' option, irrespective of the selected 'System Clock Synchronization' option. If you are
using the BRI/T1E1 card with CPLD version/revision 'V1R3 and onwards', the clock synchronization
option will work as you have programmed the option, using this command. To know the CPLD version/
revision, open the cover of T1E1 card of your system and check the label on the CPLD device or
contact your dealer for more information.
Selecting option 2('8 KHz'): By default, Master Clock Synchronization port number is given as T1E1-1,
T1E1-2, T1E1-3 and T1E1-4. The options for Master clock synchronization allow selecting T1E1 or BRI
port or a combination thereof as required. If any BRI port is selected in Master Clock Synchronization
option (for any option from 1 to 4), SE should select the System Clock Synchronization option 2 = 8
KHz.
Selecting option 3('2.048 MHz'): Select 'System Clock Synchronization' option 3 = 2.048 MHz, only for
E1 T1E1 line. To syncronize the clock with 2.048 MHz as the source clock frequency, make sure you
have the Switch Card with firmware V5R10 or later and the T1E1 Card with firmware V3R10 or later,
Design Version: D-101-017-02-03; CPLD Version of CPLD1 and CPLD2: V2R2.
Selecting option 4('1.54 MHz'): Select 'System Clock Synchronization' option 4 = 1.54 MHz, only for T1
T1E1 line.
1737
Meaning
Fast
Slow
1738
PBX-A
PBX-B
T1
T1
E&M1
T2
T2
PSTN
E&M2
PSTN
E&M3
Tn
S1 S2
Tn
Sn
2001 2002
S1 S2
2003
3001 3002
Sn
3003
PBX-C
S1 S2
4001 4002
Sn
4003
In the above figure, 3 PBX systems are connected through E&M connectivity.
S1 to Sn are extensions.
E&M1 to E&M3 are E&M lines between the three PBX systems.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic License Management to know more.
How it works
For Closed User Group, you must
have unique extension number in all the systems, that is, one cannot have extension number 2001 in PBXA as well as in PBX-B or PBX-C.
enable Closed User Group (CUG) in the Class of Service assigned to the extension users.
1739
Few new words have been used to explain this application, each of these words have been explained below:
Closed User Group Table: This table has five parameters viz. Route Index, Route Code, OG Trunk
Bundle Group, Strip Digit Count, Self Route and Apply Toll Control flag. The closed user group
programming works according to this table.
Index
Route Code
OGTBG
Self Route
Apply Toll
Control
001
002
003
:
250
Route Code: Route code could be of maximum sixteen digits. Digits 0 to 9 are allowed. However, * and
# are not allowed. Generally route code will be a truncated number of the extension numbers. For
example in the figure given above, route code for PBX-B can be defined as 3 and that for PBX-C can be
defined as 4.
If PBX-B were having extension numbers from 3100 to 3199 and PBX-C were having extension numbers
from 3200 to 3299 then route code for PBX-B can be defined as 31 and that for PBX-C can be defined as
32. If PBX-B were having extension numbers from 301 to 399 and 401 to 499 then two route codes can
be defined for PBX-B viz. 3 and 4. Likewise for PBX-C.
OG Trunk Bundle Group: An OG Trunk Bundle Group (OGTBG) is assigned to each route code.
Whenever a call is to be made on that route, a free trunk from the OGTBG is selected and the extension
number is dialed on it. The same logic of Rotation On/Off for trunk selection from the OGTBG is used. If
rotation is OFF then always the first trunk in the OGTBG is selected. If it is busy then the next trunk in the
group is selected. This helps to select an alternate route. Whereas if Rotation is ON then the trunks in the
OGTBG are selected in round robin fashion.
Strip Digit Count: For the Closed User Group application, configure this parameter as 0. This parameter
is relevant when you are configuring Closed User Group-With Exchange ID.
Self-Route: For the Closed User Group application make sure this is disabled. This parameter is relevant
when you are configuring Closed User Group-With Exchange ID.
Dialed Digit Count: When digits are dialed on the trunk, the system waits for inter digit timer after the last
digit is dialed. In order to avoid this timer and number of digits dialed to be routed without further delay,
count for the number of digits to be programmed in this field. If the number of digits received are equal to
the parameters programmed then the number is dialed out immediately without waiting for the inter digit
timer. If the number of digits dialed by the user are not equal to the digits programmed, the number is
dialed after inter digit timer.
Apply Toll Control: When Self Route flag is disabled, system will check this parameter. By default, this
flag is enabled. The system will apply toll control to all the outgoing calls.
Disable this flag, if you do not want to apply toll control to the CUG numbers dialed by you.
1740
Route Code: Route code could be of maximum sixteen digits. Digits 0 to 9 are allowed. However, *
and # are not allowed. Generally route code will be a truncated number of the extension numbers. For
example in the figure given above, route code for PBX-B can be defined as 3 and that for PBX-C can
be defined as 4.
If PBX-B were having extension numbers from 3100 to 3199 and PBX-C were having extension
numbers from 3200 to 3299 then route code for PBX-B can be defined as 31 and that for PBX-C can
be defined as 32. If PBX-B were having extension numbers from 301 to 399 and 401 to 499 then two
route codes can be defined for PBX-B viz. 3 and 4. Likewise for PBX-C.
OG Trunk Bundle Group: An OG Trunk Bundle Group (OGTBG) is assigned to each route code.
Whenever a call is to be made on that route, a free trunk from the OGTBG is selected and the
1741
extension number is dialed on it. The same logic of Rotation On/Off for trunk selection from the OGTBG
is used. If rotation is OFF then always the first trunk in the OGTBG is selected. If it is busy then the next
trunk in the group is selected. This helps to select an alternate route. Whereas if Rotation is ON then
the trunks in the OGTBG are selected in round robin fashion.
Strip Digit Count: It has no significance for Closed User Group application. But it has to be
programmed as 0.
Self-Route: It has no significance for Closed User Group application. But it has to be programmed as
0.
Dialed Digit Count: When digits are dialed on the trunk, the system waits for inter digit timer after the
last digit is dialed. In order to avoid this timer and number of digits dialed to be routed without further
delay, count for the number of digits to be programmed in this field. If the number of digits received are
equal to the parameters programmed then the number is dialed out immediately without waiting for the
inter digit timer. If the number of digits dialed by the user are not equal to the digits programmed, the
number is dialed after inter digit timer.
Apply Toll Control: By default, this flag is enabled. The system will apply toll control to all the outgoing
calls. Disable this flag, if you do not want to apply toll control to the CUG numbers dialed by you.
Step 1
Use following command to configure route code:
4502-1-Route Index-Route Code-#*
4502-2-Route Index-Route Index-Route Code-#*
4502-*-Route Code-#*
Where,
Route Index is from 001 to 250.
Route Code is a sixteen digits string of numbers.
Use following command to clear a particular route code:
4502-1-Route Index-#*
4502-2-Route Index-Route Index-#*
4502-*-#*
Where,
Route Index is from 001 to 250.
By default, Program Route Code is Blank.
Step 2
Use following command to assign OG Trunk Bundle Group to the route code:
4503-1-Route Index-OG Trunk Bundle Group
1742
Meaning
Step 5
Use following command to configure maximum dialed digits to select router for a route code:
4506-1-Route Index-Dialed Digit Count
4506-2-Route Index-Route Index-Dialed Digit Count
4506-*-Dialed Digit Count
Where,
Route Index is from 001 to 250.
Maximum dialed digits is from 00 to 99.
Step 6
Use following command to clear an entry in a routing table:
4501-1-Route Index
4501-2-Route Index-Route Index
4501-*
Where,
Route Index is from 001 to 250.
1743
ETERNITY offers few features associated with closed user group. Each of these are discussed below:
Alternate Route:
This feature provides flexibility of accessing an extension of other exchange through alternate routes if
the normally used route (the shortest route) is not free. To achieve this, the trunks that offer shortest
routes should be programmed first in the OGTBG followed by the trunks that provide alternate routes.
Also the rotation within the OGTBG should be OFF. In figure the requirement is that if E&M1 is busy
then E&M2 should be used to call 3001 from PBX-A. In this case a OGTBG is to be so formed that it
has E&M1 as first trunk and E&M2 as second trunk and should be assigned to route code 3. However,
the rotation within the OGTBG should be disabled. Similarly, if the call is to be made to 4001 then
E&M2 should be used. Hence another OGTBG should be programmed with E&M2 as first trunk and
E&M1 as second trunk and it should be assigned to route code 4. Also the round robin option for the
OGTBG should be selected.
Transit Barring:
This feature helps to bar the Transit calls through the exchange. Consider figure 1. It is required that an
extension user in PBX-A can access extension 3001 using alternate route through PBX-C but an
extension user in PBX-B cannot access extensions 2001 to 2010 using alternate route through PBX-C.
This can be accomplished using Transit Barring. To achieve this, a denied list containing 10 numbers
viz. 2001 to 2010 should be assigned as Toll Control for the SLT port programmed for the E&M2. Doing
so when an extension user from PBX-B dials 2001, if E&M1 is busy, the system would try dialing
through E&M2 but since E&M2 does not have requisite toll control, it will give error tone to extension
user. Transit Barring adds value to the Alternate Route by allowing a selective access to the extension
with alternate route.
Each system in the network has a routing table. By default, the routing is blank and hence E&M works
as per normal E&M connectivity. After programming the routing table as per the requirement, when the
user dials an extension number, the system first searches for the dialed number in the same system by
considering Self Route flag. If it is not available in its own system, it checks for it in the routing table,
finds the best fit, selects a free E&M path and reaches the dialed port.
Please refer figure 1. When an extension in PBX-A dials an extension number 3001, the system
searches for this number in PBX-A. Since there is no extension with flexible number 3001 in PBX-A,
the system checks the E&M routing table. The system follows the routing table, identifies that the dialed
number is in PBX-B, selects a free E&M path and reaches the dialed port.
As shown in figure 1, the shortest path to reach extension 3001 from PBX-A is through E&M1. But if
E&M1 is busy then the network can be programmed to reach extension 3001 through E&M2 and
E&M3. For this both PBX-A and PBX-B have to be programmed to accomplish this.
Relevant Topics:
1. E&M Connectivity
1914
2. Forced Call Disconnection
1960
3. Closed User Group-With Exchange ID
4. OG Trunk Bundle Group
2077
1744
1745
To have private networks, few PBXs can be connected to each other using E&M, T1/E1, QSIG, etc. The
requirement demands that the PBXs connected to each other forming the network behave as a single
group. The users need not dial a separate code to access an extension user of other PBX. The entire
network should behave as a single unit. The extension users will not know whether they are dialing an
extension number of their own PBX or other PBX. This is called Closed User Group. However, it is
possible that the PBXs connected to form a network may have same extension numbers. Also, all the
exchanges within the network may have their own identity (called Exchange ID). In such cases, the routing
scheme (the routing table) has to be programmed keeping the Exchange ID (EID) in mind. This is known
as Closed User Group-With Exchange ID.
This facility is generally used in PLCC Applications wherein new power extensions (and hence PBXs) are
added in the network. It is not feasible to have unique extension numbers throughout the network. In such
cases, an Exchange ID is assigned to the newly added PBX and a routing table is programmed in the
exchange. Also, the routing tables of other exchanges are modified to include the newly added exchange
in the network.
S1 to Sn are extensions.
PBX-A
T1
PBX-B
21
22
E&M1
T2
T2
PSTN
E&M2
PSTN
E&M3
Tn
S1 S2
2001 2002
T1
Tn
S1 S2
Sn
2099
2001 2002
Sn
2099
PBX-C 23
S1 S2
2001 2002
Sn
2099
1745
How it works
In this application, it is possible to have same extension numbers in two or more PBXs of the network, but you must
enable Closed User Group (CUG) in the Class of Service assigned to the extension users.
Few new words have been used to explain Closed User Group-With Exchange ID application, each of these words
have been explained above.
Closed User Group Routing Table: This table has five parameters viz. Route Index, Route Code, OG
Trunk Bundle Group, Strip Digit Count and Self Route flag. The Closed User Group-With Exchange ID
programming works according to this table.
Index
Route
Code
OGTBG
Strip Digit
Count
Self Router
Flag
Dialed Digit
Count
Apply Toll
Control
001
:
250
1746
Route Code: Route code could be of maximum six digits (XXXXXX). Digits 0 to 9 are allowed. Generally,
route code will be a unique number. The route code should not clash with any of the extension numbers of
same PBX. For example in the figure given above, route code for PBX-A can be defined as 21, route code
for PBX-B can be defined as 22 and that for PBX-C can be defined as 23. This means that no extension
in PBX-A can start with 22 or 23. Similarly, no extension in PBX-B can start with 21 or 23 and no
extension in PBX-C start with 21 and 22.
OG Trunk Bundle Group: An OG Trunk Bundle Group (OGTBG) is assigned to each route code.
Whenever a call is to be made on that route, a free trunk from the OGTBG is selected and the extension
number is dialed on it. The same logic of rotation On/Off for trunk selection from the OGTBG is used. If
rotation is OFF then always the first trunk in the OGTBG is selected. If it is busy then the next trunk in the
group is selected. This helps to select an alternate route. Whereas if rotation is ON then the trunks in the
OGTBG are selected in round robin fashion.
Strip Digit Count: This count signifies the number of digits to be stripped off while dialing/decoding a
number. To elaborate: Consider figure 1. The requirement is that if extension 2001 of PBX-B dials 212002
and if E&M 1 is busy then the call should reach extension 2002 of PBX-A through alternate route. In this
case the strip digit count of PBX-A should be programmed as 2 and that of PBX-B and PBX-C should be
programmed as 0. Doing so, when extension 2001 of PBX-B dials 212002 and if E&M1 is busy then the
call is routed through PBX-C. In this case, PBX-B dials 212002 on E&M3, PBX-C receive this code and
dials out the same code, that is, 212002 on E&M2 without striping of any digit. On receiving 212002, PBXA strips of two digits as per the programming and routes the call to extension 2002.
Self-Route: This flag signifies that the digits being dialed are for the same PBX and are not to be dialed on
the E&M trunk.
Dialed Digit Count: When digits are dialed on the trunk, the system waits for inter digit timer after the last
digit is dialed. In order to avoid this timer and number of digits dialed to be routed without further delay,
count for the number of digits to be programmed in this field. If the number of digits received are equal to
the parameters programmed then the number is dialed out immediately without waiting for the inter digit
timer. If the number of digits dialed by the user are not equal to the digits programmed, the number is
dialed after inter digit timer.
Apply Toll Control: This parameter is not relevant as Self Route flag is enabled. This parameter is
relevant when you are configuring Closed User Group (CUG).
Please note that the ETERNITY has only one routing table. The same table is used for Closed User Group
and Closed User Group-With Exchange ID. Hence the table has to be programmed keeping the
application in mind.
How to configure
Please refer topic Closed User Group (CUG) for more details.
ETERNITY offers few features associated with Closed User Group-With Exchange ID. Each of these are
discussed below:
Alternate Route
Please refer Closed User Group (CUG) for more details.
Transit Barring
Please refer Closed User Group (CUG) for more details.
Strip Digit Count is of significance in a network in which few exchanges possess Exchange ID whereas
others do not. Refer figure 2. PBX-A and PBX-B are made to work as Closed User Group since they have
unique extensions. But since PBX-C possess extensions whose flexible number clashes with extensions
in PBX-A, it cannot be made a part of Closed User Group. In such case Closed User Group-with
Exchange ID can be used with the combination of PBX-A + PBX-B and PBX-C forming the network.
In this case, if 2001 of PBX-A wants to access 3001 through E&M1, he must dial 3001. But if E&M1 is busy
then system will allot him E&M2 and since there would not be any programming done, it will give error
tone to the caller. To avoid this condition, the extension user of PBX-A should be asked to call 3001 by
dialing 223001. For this, the strip digit count for the route index with route code 22 in the routing table of
PBX-B should be programmed as 2 and the strip digit count for the route index with route code 22 in the
routing table of PBX-C should be programmed as 0. Doing so, if the call to 3001 is made through E&M1,
then PBX-B would strip of the first two digits on receiving 223001 and make the caller reach 3001. If the
call to 3001 is routed through E&M2, the PBX-C will not strip of any digit and would dial out 223001 on
E&M3. On receiving 223001, PBX-B as per the programming would strip of 22 and make the call land on
3001.
Strip Digit Count is also of importance in following case:
For example in figure 1, if extension 2001 of PBX-A dials 212002 then also the call should go to 2002 of
PBX-A only. To accomplish this, the strip digit count should be programmed as 2 and Self Route flag
should be enabled. Doing so when 2001 dials 212002, the system strips off first two digits and checks for
the remaining digits, which in this case would be 2002 and thereby the user reaches 2002 of the same
PBX.
Forced Call Disconnection
Please refer Forced Call Disconnection for more details.
1747
This feature is programmed only for few Exchanges like ET1, Genesis, or BPL which does not send 0
when user dials a number to call an extension number of the exchange. For other Exchanges like ET2 it
is not required to be programmed as it sends 0 also if it is dialed.
Prefix String is a string of characters which is prefixed to the string, dialed by the user and then CUG
Routing-Table is applied.
Example:
Detailed application for Prefix String feature is explained by following steps:
If this feature is not programmed for Exchange of type: ET1 or BPL, and Prefix String is blank, then, only
2223 is dialed by the Exchange when 02223 is dialed by the user, to call extension 23.
Now when 2223 reaches Exchange B, since there is no entry in the CUG table, the feature-extension
flexible number table is checked. Now since a match starting with 22.. is found, the call is routed to
extension 22 instead of 23, because extension number 22 is present on the Extension.
To avoid this, an entry is made in the routing table containing 22 as the route code with strip digit count
2. But then since the system checks CUG Routing-table first, the first two digits out of 2223 always get
striped off and the call is not routed to extension 22, that is, the other extensions of Exchange B will never
be able to call extension 22.
To solve this problem Prefix String feature is programmed in the E&M Feature Template with 0 as prefix
string, so that string with prefix (022) is matched with the entries of CUG Routing Table (as shown in
figure). When user dials 02223, first 3-digits are stripped off and the required Extension 23 can be
called.
Relevant Topics:
1. E&M Connectivity
1914
2. Forced Call Disconnection
1960
3. Closed User Group (CUG)
1739
4. OG Trunk Bundle Group
2077
5. E&M Feature Template
895
1748
Communication Ports
What' this?
ETERNITY supports serial, asynchronous, RS232C Communication Ports.
The ETERNITY ME supports two communication ports, COM1 and COM2.
ETERNITY GE supports a single communication port.
ETERNITY PE6SP and PE3SP support a single communication port each. There is no communication port on
ETERNITY PE3SS.
A Communication Port is used for the following facilities:
A Communication Port is necessary for Programming ETERNITY using a PC, whereas for other above listed
facilities, Communication Port may or may not be used250.
How to configure
In order for each of the above listed facilities to work, a Communication Port must be assigned first as the
'Destination Port' and the attributes of the Communication Port of ETERNITY and the Communication Port of the
PC to which it is connected must programmed to match.
PMS Interface
Configuring using Serial COM Port
Station Message Detail Recording-Report
Station Message Detail Recording-Online
Station Message Detail Recording-Posting
System Activity Log
System Fault Log
250. PMS can be interfaced on the Ethernet Port. For System Activity and Fault Logs, SMDR Reports and Online Printer Port can be
used. For SMDR-Posting Ethernet Port can be used.
1749
Speed in bps.
Number of data bits.
Number of stop bits.
Parity
Flow Control
DSR Sensing
These attributes must be programmed keeping in mind the application for which the communication port is used
(for instance, Programming through PC, generating SMDR Reports, etc.)
When DSR Sensing is enabled, the system continuously monitors the physical connection between the
ports. Whenever the physical connection is detected to be inactive, the system stops data transfer. If DSR
Sensing is disabled, the physical connection between the ports is not monitored. Hence, even when the
physical connection is inactive, data transfer continues, and this data is lost.
The Communication Port attributes can be changed using Jeeves and dialing SE commands from a telephone.
1750
Set the desired values for COM Port 1 and COM Port 2:
Speed (spd)
Data Bits
Parity
Stop Bits
Flow Control
DSR Sensing
251. Please note that maximum speed of the Communication port allowed in two-way communication like programming through computer, programming through the Jeeves is 2400 bps only.
1751
1 for Odd
2 for Even
3 for Mark
4 for Space
By default, Parity is set as 'None'.
To set Stop Bits for a COM Port, dial:
3204-Port-Stop Bits
Where,
Port is
1 for COM Port 1
2 for COM Port 2
For ETERNITY GE/PE, dial '1' for COM Port.
Stop Bits are
0 for 1 stop bit
1 for 2 stop bit
By default, Stop Bits are 1.
To set Flow Control for a COM Port, dial:
3205-Port-Flow Control
Where,
Port is
1 for COM Port 1
2 for COM Port 2
For ETERNITY GE/PE, dial '1' for COM Port.
Flow Control is from 0 to 2
0 for None
1 for Hardware (RTS/CTS)
2 for Software (XON/XOFF)
By default, Flow Control is 'None'.
To enable DSR Sensing on a COM Port, dial:
3206-Port-DSR Sensing
Where,
Port is
1 for COM Port 1
2 for COM Port 2
For ETERNITY GE/PE, dial '1' for COM Port.
DSR Sensing
0 for Disable
1 for Enable
By default, DSR Sensing is disabled.
1752
Exit SE Mode.
1753
Stop Bits = 1.
Flow Control = None.
How to use
For ETERNITY to communicate with a PC through the Communication Ports (COM1 and COM2), it must be
connected with the Communication Port of the PC.
Signal Name
Ground (GND)
252. This cable is supplied as an optional item. Contact your Matrix Dealer or the company to obtain this cable.
1754
Conference-3 Party
What's this?
ETERNITY offers three types of conference calls: Conference-3 Party, Conference Dial-In, and ConferenceMultiparty.
Conference-3 Party (also referred to as Three-Way Calling) is a telephone call, in which the calling party can have
two other persons participate in the call.
A 3-Party Conference is initiated by dialing the number of the first person one wishes to talk to. The first person is
informed about the conference and put on Consultation Hold. The number of the second person one wishes to talk
to is dialed. When the second person answers, s/he is informed about the conference. Three-way speech is
established by pressing Flash-*3.
An already connected two-way speech can be converted into a conference by adding a second person, without
disconnecting the call with the first person.
Thus, a 3-Party Conference may be planned or conducted on the spur of the moment.
A 3-Party Conference can be conducted with extensions of ETERNITY and between extensions and external
numbers.
It is also possible to conduct an Unsupervised 3-Party Conference, wherein the operator connects two trunks
through the system and withdraws from the three-way speech.
The maximum number of simultaneous 3-Party Conferences supported by each model/variant of ETERNITY are
mentioned in the table below.
Max. number of simultaneous 3-Party
Conferences supported
Model
ETERNITY ME
ETERNITY GE
ETERNITY PE6S
How it works
A, B, C are extensions.
D and E are external numbers.
A is in speech with B.
A and B want to include C in their conversation.
A presses the Conference Key. B is put on Consultation Hold.
A gets feature tone. B gets on-hold music.
A dials C's extension number. A gets ring back tone.
1755
A is in speech with B.
A and B want to include D in their conversation.
A presses the Conference Key. B is put on Consultation Hold.
A gets feature tone. B gets on-hold music.
A grabs a Trunk and dials D's extension number. A gets ring back tone.
A is in speech with D. B cannot hear their conversation.
A presses the Conference Key to enable three-way speech.
A, B, and D are now in speech.
1756
A is in speech with D.
A and D want to include E in their conversation.
A presses the Conference Key. D is put on Consultation Hold.
A gets feature tone. D gets on-hold music.
A grabs a Trunk and dials E's number.
A is in speech with E. D cannot hear their conversation.
A presses the Conference Key to enable three-way speech.
A, D, and E are now in speech.
A, B and C are in speech. When A disconnects, either B and C are also disconnected or speech is
established between them depending on the option you select in If the Extension creating 3 party
conference, disconnects during Conference in the System Parameters.
The Conference can be broken only by the master DKP/Extended IP Phone that has initiated the
Conference.
If all the parties to the conference are SIP Extensions/Trunks and if the initiator of the Conference goes
on-Hook during the conference, the other parties will still remain in conversation. This is known as
Transfer on Conference Hangup.
How to configure
For this feature to work, the feature 'Conference' must be enabled in the Class of Service group of the extensions
that are to be allowed this feature.
If extension users at remote locations are to be allowed to initiate the 3-party conference, Direct Inward System
Access (DISA) or Auto Attendant must be enabled on the trunk on which their call lands.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions and
programming.
The feature 'Conference' in the Class of Service also includes Dial-In and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed all three types of
conferences - 3-Party, Dial-In and Multi-party Conference.
How to use
For EON and Extended IP Phone Users
If Party 2 is a trunk,
1757
If Party 2 is a trunk,
1758
Dial Trunk Access Code to grab a trunk. You get Trunk dial tone.
Dial telephone number of Party 2. You get ring back tone.
Speech with Party 2.
Dial Flash-*3.
Three-way speech is established.
Conference-Multiparty
Whats this?
Like the Dial-In Conference, a Multi-party conference allows speech between more than three participants.
The key difference between Dial-In and Multi-party conference is that in a Dial-In conference participants can
include themselves in the conference by dialing into it without assistance, whereas in a Multi-party Conference the
party initiating the conference must include the participants by dialing their numbers.
In a 3-party Conference, when you add the forth participant, a Multiparty Conference is initiated.
A Multiparty conference may be
between extensions
between extensions and trunks, that is, external numbers.
Any participant in a Multiparty Conference can Include a party, Remove a party, Leave a conference temporarily or
can Cancel a conference.
External callers can initiate multiparty conference using Direct Inward System Access (DISA).
There are 7 digital conferencing circuits in the system. The ETERNITY supports between 6 to 21 parties in a
conference depending on the model you are using.
Maximum conference
participants
Maximum simultaneous
conferences (If all the
conferences involve 3
parties)
ETERNITY ME
21
21
ETERNITY GE
15
15
ETERNITY PE 6S
15
15
ETERNITY PE 3SP
ETERNITY PE 3SS
Model
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic License Management to know more.
How it works
A, B, C, and D are extension users.
E and F are external numbers.
A decides to hold a teleconference with B, C, D, E and F.
1759
1760
S1
S2
ETERNITY S3
PSTN
C
D
S4
S5
S1
S2
B
C
S3
PSTN
S4
ETERNITY
S5
S6
S7
X
Y
S8
S9
S10
S11
S12
T1
PSTN
ETERNITY
S1
T2
S2
T3
S3
T4
S4
T5
S5
A
B
C
D
E
T6
S6
S7
F
G
H
S8
S9
1761
PSTN
T1
S1
T2
S2
T3
A
B
S3
T4
S4
T5
ETERNITY
C
D
S5
PSTN
T1
S1
T2
S2
T3
A
B
S3
T4
S4
T5
ETERNITY
C
D
S5
Simultaneous Multiparty Conference between a few trunks and extensions and between extensions.
PSTN
T1
S1
T2
S2
T3
S3
A
B
C
T4
T5
S4
T6
S5
E
D
S6
S7
ETERNITY
F
G
How to configure
To provide this feature to extensions,
You must enable the feature 'Conference' in theClass of Service (COS) of the extensions in their Station
Basic Feature Template. By default, this feature is enabled on all extensions, so all extensions can use
this feature.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Dial-In as well as 3Party and Multi-party Conference.
1762
If desired, you may also change default value of the Release Conference if Idle for more than (min.)
Timer. See System Timers and Counts.
If external parties are to be allowed to initiate or join the Conference, Direct Inward System Access
(DISA) must be enabled on the trunk on which they call.
You can program a DSS key for Terminating a Conference, Temporarily Leave/Rejoining a Conference, if
required. Refer the topic DSS Keys Programming for instructions.
How to use
For EON and Extended IP Phone Users
To initiate multiparty conference:
After the Multiparty Conference has been initiated. Press the Conference Key.
The Multiparty Conference menu appears on the LCD.
Select the option Include Party and dial the number.
When you are in speech with the party, press the Conference Key.
Repeat the above steps to add new participants in the conference (max. 21).
1763
Lift handset.
Dial number of party 1.
Speech with party 1.
Dial Flash.
Dial number of party 2.
Speech with party 2. Dial Flash-*3. A 3-way speech is established.
Dial Flash.
Dial number of party 3.
Speech with party 3. Dial Flash-*3. A multiparty conference is established.
Lift handset.
Dial number of party 1.
Speech with party 1.
Dial Flash.
Dial number of party 2.
Speech with party 2. Dial Flash-*3. A 3-way speech is established.
Dial Flash.
Dial number of party 3.
Speech with party 3. Dial Flash-*3.
Repeat the steps to include the desired number of parties (max. 21).
1764
To use Multiparty Conference from DISA mode, you may see the instructions Dial-In Conference using DISA
under Conference Dial-In.
1765
Conference Dial-In
Whats this?
Dial-In Conference is a multi-party conference held at a pre-defined time. Extension users can schedule a Dial-In
Conference and inform other participants to join in the conference at the scheduled time. Dial-In Conference can be
used to conduct client meetings or sales presentations, project meetings and updates, regular team meetings, and
to communicate with coworkers who operate in different locations. Thus, this feature helps to increase productivity
by saving time and cost of travel for out-of-office meetings.
The number of simultaneous Dial-In Conferences supported by ETERNITY varies by model.
Number of Simultaneous Dial-In
Conferences supported
ETERNITY ME
21
ETERNITY GE
15
ETERNITY PE 6S
15
ETERNITY PE 3SP
Model
How it works
A Dial-In Conference can be scheduled by dialing the access code for Dial-In conference followed by the
conference number and a password.
The Conference Password is a four digit number string. The default conference password is 1111 and
must be changed before using this feature.
Let us understand how Dial-In Conference works with the following example.
1766
Extension user A wants to schedule a Dial-In Conference at 4:30 p.m. with B, C, D, E and F.
B and C are extension users. C is an extension user who has been provided a DISA login to access an
extension of ETERNITY.
D, E and F are external parties.
Any extension user can initiate the conference, in this case A initiates the conference.
A informs B, C, D, E and F about the conference and provides them conference number, for example, '1'
and the password, '4040'.
If C wants to schedule a conference, C must log into his extension from DISA mode.
Any extension user can initiate the Dial-In conference by dialing the feature access code for Dial-In
conference followed by the conference number and the password, for instance: '1' and '4040'. In this case,
A initiates the conference.
1767
How to configure
To provide this feature to extensions,
You must enable the feature 'Conference' in theClass of Service (COS) of the extensions in their Station
Basic Feature Template. By default, this feature is enabled on all extensions, so all extensions can use
this feature.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Dial-In as well as 3Party and Multi-party Conference.
If you want beeps to be played when any one joins the conference, enable Play Beep when Raid/
Conference/Dial-in Conference begins. See System Parameters, for instructions.
If desired, you may also change default value of the Release Conference if Idle for more than (min.)
Timer. See System Timers and Counts.
If external parties are to be allowed to initiate or join the Conference, Direct Inward System Access
(DISA) must be enabled on the trunk on which they call.
You can program a DSS key for Dial-In Conference, Terminating a Conference, Temporarily Leave/
Rejoining a Conference, if required. Refer the topic DSS Keys Programming for instructions.
How to use
For EON and Extended IP Phone Users
To Schedule a Dial-In Conference:
Go OFF-Hook.
Press the DSS key assigned for Dial-In Conference.
OR
Dial *19.
The Dial-In Conference menu appears on the LCD.
1768
Select the option Schedule a Conf and press the Enter Key.
Enter Conference Number on the prompt.
Enter Conference Password on the prompt.
You get confirmation tone and the message 'Conf <number> Scheduled' on your phone's display.
Go ON-Hook.
Call all participants and inform them of the time of the Dial-In conference, the Conference Number and
Password.
1769
If you have configured a DSS key for Temporary Leave and you leave the Conference by pressing the
DSS key, to Rejoin the Conference press the DSS key again.
To permanently leave from the Dial-In conference:
While in Conference, go ON-Hook.
To cancel Dial-In conference:
While in Conference, press the Conference Key.
The Multiparty Conference menu appears on the LCD.
Select the option Terminate Conference and press the Enter Key.
OR
Press the DSS key assigned to Terminate Conference.
All the participants will get an Error Tone and the system resource occupied by the conference will be
freed.
To cancel a Dial-In Conference from the System Administrator (SA) Mode:
Enter SA mode from a DKP/SLT/Extended IP Phone.
Dial 1072-026-Conference Number.
You get confirmation, the conference is released.
You can also cancel a Dial-In Conference from System Administrator (SA) Mode using Jeeves. To do this,
1770
Open Jeeves.
Enter the conference number (1 to 7) which you want to cancel in the Cancel Dial-In Conference Number
field.
1771
When you enter DISA mode, you get beeps, dial digits before the DISA Inactivity Timer elapses.
Never dial 'Flash' when in DISA mode, you will get disconnected.
Keep dialing any digit to continue the conference.
See Direct Inward System Access (DISA) to know more.
1772
Conflict Dialing
What's this?
You may recall that Access Codes are dialed at different call phases. No two Access Codes must be the same in
the same call phase.
For example, the same access code cannot be used for two different features like Call Forward and Redial, since
both these features are invoked in the 'Dial' phase. Similarly, Station and Logical Group Codes too must be unique
and should not match with any of the features invoked in the 'Dial' phase.
However, ETERNITY allows overlaps within Feature Codes and Flexible Numbers (Station Codes). One Feature
Access Code can be a part of (subset) another code, for example, 4, 41, 412; Flexible Numbers of extensions can
be 201, 2011 etc.
So, when such overlapping access codes are dialed, the system matches the first digit. On finding more than one
Access code starting with the same digit, the system will not know how to interpret the instruction and act
accordingly.
Conflict Dialing feature resolves this confusion. When an access code that is a subset of any other access code is
dialed, the system waits for some time for the extension user to dial the next digit. If the user does not dial any digit
within that time, the system interprets it as the smaller Access Code, and invokes the associated feature.
The time for which the system waits for the next digit to be dialed before resolving the Access Codes is called
"Conflict Dialing Timer". This timer is set to 2 seconds and is programmable.
Refer the topics Access Codes to know more.
How it works
You may set,
If A does not dial any other digit before the Timer elapses, the system interprets the code as '41' and
invokes the Alarm feature.
1773
If such access codes exist, the system again waits for the duration of the Conflict Dialing Timer for another
digit to be dialed.
Thus, only when the conflict in the access codes is resolved will the system respond accordingly.
How to configure
The working of this feature is controlled by the Conflict Dialing Timer, which is set by default to 2 seconds and can
be changed as desired.
If the duration of the Conflict Dialing Timer is long, it may cause delay in the system's response to the
feature. If the duration is less, the system may misinterpret the access codes. Ensure that the value of the
Timer is programmed optimally (that is, at least the default value).
1774
Scroll to Other Features and set the time in seconds as desired for the Conflict Dialing Timer .
Exit SE mode.
1775
Conversation Recording
What's this?
Conversation Recording allows extension users to record their talk with other extension users or external parties,
after or without informing the opposite party.
This feature can be used to record verbal agreements, important discussions, instructions, interviews, client
requirements, take or place orders, etc.
Extensions must have a mailbox assigned to them for recording conversations. So, a VMS card must be installed in
the system for this feature to work.
Matrix Comsec is not responsible for any mis-/abuse of this feature by users.
On SIP extensions, ETERNITY supports Conversation Recording using INFO Message. For a list of IP
phones on which this feature has been tested, see ETERNITY Features tested on IP Phones of different
Brandsin the Appendix.
It is not possible to pause the Conversation Recording in on SIP Extension. So, when SIP Extension puts
the call on hold and then retrieves the call then for the hold duration, silence will be recorded in the
recorded file.
How it works
A and B are extensions. Both are assigned a mailbox each.
C and D are external parties.
A calls C.
C answers the call.
A presses the Transfer Key.
C is put on Consultation Hold.
A dials the command for Conversation Recording.
The system sends a string of digits to the Voice Mail System to initiate Conversation Recording.
A and C are in speech again.
The conversation recording starts in A's mailbox. The system plays beeps, if Conversation Recording
Beeps are enabled.
A or C disconnects the call.
Conversation recording ends.
A can listen to the recorded conversation by invoking the Voice mail feature.
The same is repeated when B calls A. As both have mailboxes assigned, both can record the conversation.
1776
How to configure
The functioning of this feature is controlled by three parameters: 'Class of Service', 'Mailbox', and 'Conversation
Recording Beeps'. These parameters can be programmed using Jeeves or by dialing SE Commands from a
telephone.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions and
programming.
Mailbox
Extensions that are to be allowed Conversation Recording must also have a mailbox. Refer Voice Mail for SLT
Extensions, Voice Mail for DKP Extensions, Voice Mail for ISDN Terminal, Configuring Voice Mail Settings for
SIP Extensions for more information and programming instructions.
1777
Go to Play Beep when Call Taping/Conversation Recording starts. Click the check box to enable or
clear the check box to disable this feature.
Exit SE mode.
How to use
For EON and Extended IP Phone Users
To record a conversation:
1778
253. This is the default Voice Mail Feature Access Code. Verify with you System Engineer if this has been changed and use the new
code.
1779
Customer Name
What's this?
Customer Name is the name of the organization/enterprise that has deployed ETERNITY. As the User, you can
enter the name of your company/organization in the system.
When Customer Name is assigned in the system, this name will appear as header on the various System Reports
generated and printed by the ETERNITY like SMDR Incoming, Outgoing and Internal Call Reports, T1E1PRI
Performance reports, Alarm Status reports, etc.
The Customer Name may consist of a maximum of 80 alphanumeric characters, including punctuation marks. So,
you can enter the organization's address along with the Customer Name.
How to configure
Customer Name can be programmed using Jeeves and dialing SE commands from a Telephone at the time of
installation, or any time thereafter. It can also be corrected or changed any time.
1780
Enter the name (and address, if desired) of the organization/enterprise in the field Customer Name. For
example: Prudent Investment, 701 Sunshine Boulevard, Bannerghatta, Bangalore.
Enter SE mode
To enter Customer Name, dial:
5401-Customer Name-#*
Exit SE mode
Use EON to assign Customer Name, as SLT does not support alphanumeric dialing.
The method of entering Customer Name from EON is similar to typing text messages from the mobile
phone.
1781
How to configure
Setting Day/Night Mode from SE mode using Jeeves
1782
1783
Exit SE mode.
1784
Dial 1072-018-Code
Where,
Code is from 1 to 4
1 is for Day Mode
2 is for Night Mode
3 is for Operate system as per Time Table.
4 is for Break Hours
Exit SA mode.
If you are setting Day/Night Mode from a DKP using a DSS key, refer the LED indication in table below.
EON48/EON310
Event
Color
Cadence
Blue
Continuous ON
Red
Continuous ON
Orange
Continuous ON
--
OFF
1785
How it works
The forward and backward adjustment of clocks can be Scheduled or Manual.
Scheduled DST Adjustment: The Real Time Clock of the ETERNITY is advanced and set backward
automatically according to the DST convention of the country/region where the ETERNITY is installed.
Scheduled DST Adjustment is useful in countries/regions where DST Time is fixed, such as in Europe,
USA and Canada, without yearly variations.
The table below gives describes the DST conventions followed in the different countries for which
ETERNITY will automatically adjust DST.
ETERNITY supports 18 DST Types for Scheduled DST Adjustment.
DST
Type
1786
DST Timings
Applicable in Countries
Start Time
End Time
01
02
03
04
Brazil
05
Canada
DST
Type
DST Timings
Applicable in Countries
Start Time
End Time
06
Chile
07
08
Finland
09
First APR
02:59 04:00
First OCT
03:59 03:00
Iraq
10
Kyrgyzstan
11
Egypt
12
Lebanon
13
Namibia
14
New Zealand
15
Norway
16
Paraguay
17
First APRIL
23:59 01:00
First OCT
23:59 23:00
Syria
18
First APRIL
23:59 01:00
Cuba
The DST Type is to be selected according to the country/region where the system is installed.
When DST Mode is set to 'Scheduled' and the DST Type is selected, the system will automatically adjust
DST at the preset dates and time for the country/region where the system is installed.
For example, if ETERNITY is installed Spain, the DST Type 01 applicable to this country should be
programmed as Scheduled DST. The system will automatically advance the clock on the last Sunday of
March at 01.59.03:00 am every year (the start date of DST) and set the clock backward on the last Sunday
of October at 02.59.02:00 am of the same year.
Manual DST Adjustment: The Real Time Clock of the ETERNITY is advanced and set backward
manually according to the DST convention of the country/region where the ETERNITY is installed.
Manual DST Adjustment is to be used in regions/countries that have no fixed DST Convention and where
yearly variations in DST practices are likely.
1787
When DST Mode is set as 'Manual', you must set the start and the end time, that is, the time at which the
clock is to be advanced and the time at which the clock is to be delayed.
There are two ways to adjust DST manually:
1. The 'Day of Month' method, which specifies a day of the month DST will start or end. For example:
starting on the 2nd Sunday of March and ending on 1st Sunday of November.
2. The 'Date and Month' method, which specifies a date of the month that DST will start or end. For
How to configure
Adjusting DST using Jeeves
1788
If you have selected Scheduled as DST mode, in the Region list, select the name of the country/region
where your system is installed.
Click Submit at the bottom of the page to save your DST setting.
If you do not find your region on this list, you are recommended to set DST Mode to 'Manual' and adjust
DST manually.
If you have selected Manual as DST Mode, set the Forward and Backward Time Adjustments.
Go to the option Forward Time Adjustment to advance the time when DST starts.
1789
Day-Month Wise to specify the day of the month DST will start.
OR
Date-Month Wise to specify the date of the month DST will start.
If you select 'Day-Month Wise' option, the 'Date-Month Wise' option will be disabled, and vice versa.
Day-Month Wise
1790
If you select the 'Day-Month Wise' option, you should now select the desired options in each of the
following:
Ordinal number: Select the Ordinal number of the day of the month, that is, the 1st, 2nd, 3rd, 4th, 5th
day, when DST begins.
Day: Select the day of the month - Sunday, Monday, Tuesday, Wednesday, Thursday, Friday, Saturday
- when DST begins.
Change Time From: Select the time when DST will begin to change. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
To: Select the time to which the DST is advanced. The time mode is 24 hours, with options from 00 to
23 hours and 00 to 59 minutes.
Date-Month Wise
If you select 'Date-Month Wise' option, you should now select the desired options in each of the following:
Change Time From: The time when DST will begin to change. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
To: The time to which the DST is advanced. The time mode is 24 hours, with options from 00 to 23
hours and 00 to 59 minutes.
Now, go to the option Backward Time Adjustments to set the time back (that is, end DST and begin
standard time).
Follow the same steps described above (step no. 4 to 6) to set the day/date, month, hours and minutes
except, here you must set these parameters according to the time when DST ends.
Click Submit at the bottom of the page to save your DST settings.
ETERNITY gives you the flexibility to set the 'Forward DST Adjustment' according to Date-Month, while
the Backward DST Adjustment according to Day-Month. Similarly, the reverse is also possible, that is,
Forward DST may be set according to Day-Month, while the Backward DST may be set as DateMonth. This is flexibility is particularly useful for setting DST of countries where the start of DST is
defined by date and month, like the First of April, but the end of DST is defined by Day and Month, such
as the last Sunday of October (as observed in Cuba).
1791
When the DST of a particular country starts or ends on the Last Sunday or any other day, for example,
the last Tuesday, last Friday of the month, always set the Ordinal Number as '5th'.
Wherever time adjustments are made at 00:00 hours, use the previous date and set DST start time
(that is, "from" time) at 23:59 hrs.
Enter SE mode.
To set the DST Mode, dial:
1010-DST Mode
Where,
DST Mode is
0 for Disabled
1 for Manual
2 for Scheduled
If DST Mode is selected as 'Manual'.
To set the start time of DST, dial:
1011-Date-Month-Current Time-Advance Time
Where,
Date is from 01 to 31. Use leading zero in case of single digit date (Default 01).
Month is from 01 to 12. Use leading zero in case of single digit month (Default 01).
Current Time254 is in HH:MM format,
HH from 00 to 23 (use leading zero)
MM from 00 to 59 (use leading zero) (Default 00:00).
Advance Time255 is in HH:MM format,
HH is from 00 to 23 (use leading zero)
MM from 00 to 59 (use leading zero) (Default 00:00).
Please note that the advance time will be greater than the current time.
To set the end time of DST, dial:
1012-Date-Month-Current Time-Delay Time
Where,
Date is from 01 to 31 (use leading zero for single digit date) (Default 01).
Month is from 01 to 12 (use leading zero for single digit month) (Default 01).
Current Time is in HH:MM format (use leading zero),
HH is from 00 to 23
MM is from 00 to 59 (Default 00:00).
Delay Time256 is in HH:MM format (use leading zero must)
HH is from 00 to 23
MM is from 00 to 59 (Default 00:00).
254. The current time is the time that is presently followed by the system.
255. This is the time to which the Real Time Clock should be advanced.
256. This is the time to which the Real Time Clock should be set back to.
1792
Please note that the delay time will be less than the current time.
If DST Mode is selected as 'Scheduled'
To select the DST Type, dial:
1013-DST Type
Where,
DST
Type
Applicable in Countries
01
02
03
04
Brazil
05
Canada
06
Chile
07
08
Finland
09
Iraq
10
Kyrgyzstan
11
Egypt
12
Lebanon
13
Namibia
14
New Zealand
15
Norway
16
Paraguay
17
Syria
18
Cuba
Exit SE mode.
1793
The DDI Routing Table is a set of general features that define the complete logic of identifying the flexible
numbers and DDI equivalent numbers when there is an incoming or outgoing call on a BRI/T1E1PRI and
SIP trunk. The ETERNITY offers 224 such tables each of which can be programmed as per the
requirement.
1794
DDI Routing Reference ID-This is the reference number acts as an identifier to the mapping logic
programmed in the DDI Routing Table. Any number of table can have the same reference number. A
Reference ID is assigned to both IC reference tables and OG reference tables of the trunks. For more
details on call resolving, please refer the topics Direct Dialing-In (DDI), IC Reference Table and OG
Reference Table.
Matrix ETERNITY System Manual
Start DDI Number-This is the First DDI Number for the ISDN Installation Number (MSN).
Total DDI Numbers-The total number of DDI numbers supported for an ISDN Installation Number.
Suppose an ISDN Trunk supports 200 DDI numbers, then the total DDI Number will be 200.
DDI Number of Digit-The number of digits in a DDI number. Suppose 200 DDI numbers are supported
on an ISDN Trunk, then the Number of Digits for that Trunk should be programmed as 3. Suppose 10
DDI numbers are supported on another ISDN Trunk, then the Number of Digits for that Trunk should be
programmed as 2.
Port Type-You can select BRI, T1E1, E&M, AOP, Department Group, Quick Dial, Routing Group,
Voice Mail Auto Attendant, Mobile, SIP Trunk, Virtual Extension or Flexible Number as the Port Type.
Port Number-The range of the Port Number depends on the Port Type you select.
Port
Port Type
Port Number
BRI
04
01-32
T1E1
05
01-08
E&M
06
001-128
AOP
08
001
Department Group
11
01-16
Quick Dial
12
001-999
Routing Group
20
01-96
Mobile
25
01-64
SIP Trunk
26
01-32
Virtual Extension
36
01-64
27
41
01-16
This field is not relevant if Flexible Number is selected as the port type.
Start DDI Flexible Number- If you have selected Flexible Number as the Port Type, you must enter
the extension number of the first DDI Number for the Index. Once this is programmed based on the
start DDI number the rest of the flexible numbers of extensions to which DDI Number is assigned is
calculated. 'Start Flexible Number' field is of significance only if the port type is 'Flexible Number'.
1795
Use the following command to configure the feature in a DDI Routing Table:
6322-1-DDI Routing Table ID-Parameter Number-Value
6322-2-DDI Routing Table ID-DDI Routing Table ID-Parameter Number-Value
6322-*-Parameter Number-Value
Where,
DDI Routing Table ID is from 001 to 224.
Parameter Value is from 01 to 06. Refer the DDI Routing Table for Parameter Values and their codes.
Code takes different values that vary from parameter to parameter. Please refer default table, which provides all the
values that can be assigned to various parameters.
Use following command to default a DDI Routing Table:
6321-1-DDI Routing Table ID
6321-2-DDI Routing Table ID-DDI Routing Table ID
6321-*
Where,
DDI Routing Table ID is from 001 to 224.
Following table shows default DDI Routing Table
01
02
03
04
Para. No./
Table ID
DDI
Routing
Ref. ID
Start
DDI
Numbers
Total
DDI
Numbers
DDI
Number
of Digit
Port Type
Port
Number
Start
Flexible
Number
001
00
000000
000000
None
Blank
Blank
002
00
000000
000000
None
Blank
Blank
003-223
224
05
06
Same as 224
00
000000
000000
None
Blank
Blank
00-99
000001999999
000000999999
0-4
None
--
Max. 6digits
Parameter Value:
Code
1796
00
04
BRI
01-32
05
T1E1PRI
06
E&M
001-128
08
AOP
11
Dept Grp
01-16
12
Quick Dial
100-999
20
Routing Grp
01-32
25
Mobile
01-64
26
SIP
01-32
27
Flex. No.
41
1-8
4-digits
01-16
Refer to above table. Whenever Flexible number is selected as the port type, the range concept of Start
flexible number becomes relevant. In all other cases, range concept is not relevant.
When a call is placed on SLT/DKP port, the calling party number is displayed on the terminal.
When the call is placed on BRI-NT or PRI-NT, the calling party number and the called party number both
are sent to the NT port. Doing so, the PBX connected to the NT port can resolve the DDI number and place
the call on the programmed extension.
Relevant Topics:
1. Direct Dialing-In (DDI)
1860
2. Configuring BRI Trunks
924
3. Configuring PRI Trunks
1029
4. IC Reference Table
1986
5. OG Reference Table
2070
1797
Department Call
What's this?
Department Call enables you to group together extensions of a particular department so that callers can reach
anyone in the department by dialing a common access code assigned to the department.
Calls made to such groups of extensions are called Department Calls and the access code used to make
department calls is called Department Number.
This feature is useful in situations where any member of a department may interact with callers, as for instance in a
information counter, a customer care cell, a technical support team, etc.
Callers can also reach individual extensions in a Department group by dialing the extension number.
ETERNITY supports the formation of 16 department groups. The member extensions of a department group may
be single line telephones (SLT), digital key phones (DKP), SIP Extensions, ISDN Terminals or Virtual Extensions.
Each Department Group can also be assigned a mailbox for voice mail, which any member extension can access.
Each Department Group can forward its calls to an extension or to its voice mail, or to another Department Group.
How it works
Extensions A, B, C, D are grouped as a Department with the access code 3901.
Internal Calls
External Calls
Department Calls can be made using Auto Attendant - Built-In Auto Attendant or VMS Auto Attendant. For
example, a company may use the Built-In Auto Attendant to have callers who want information only to dial the
Information Department instead of waiting for the Operator.
An external caller places a call to Department 3901 using the Built-In Auto Attendant.
The system checks if Rotation is enabled in the routing group assigned to the Department.
1798
As the Rotation flag is enabled, and the first call was landed on A, the system lands the call on the next
extension B.
Extension B rings for the duration of the Ring Timer (configurable; default: 15 seconds). If the Continuous
Ring flag is enabled for B, it will continue to ring, even as the system hunts for another extension in the
group to land the call.
A third call internal/external made to Department 3901.
The same process as described above will be repeated.
But the system will land the call on extension C first, because Rotation flag is enabled on this routing
group.
The subsequent incoming calls will land on the extension which is next to the one that received the last
call. So the next call to the Department will land on extension D, the one thereafter on A, and so forth.
Thus for each call, the system will hunt for a landing extension as per the Rotation set for the routing group. The
extensions will ring for the duration of the Ring Timer, either continuously or one-by-one (as per the Continuous
Flag configured), and according to the sequence in which the extensions in the group are arranged.
Rotation ensures equal distribution of call traffic. If Rotation is disabled, the fresh call will always land on first
extension of the Department group.
Department Group and ISDN Terminal: When more than two ISDN terminals connected to the same BRI
port are configured as members of a Department group, if a call is made to this group using the department
group access code, only two ISDN terminals connected to the BRI port will ring. This limitation is because
of the BRI protocol.
Voice Mail
A Department Group can be assigned a common mailbox for Voice Mail, called the Department Group Mailbox. For
this a Voice Mail System Card must be installed in the system. This common mailbox for the group is called
Department Group Mailbox. You can assign Department Group Mailbox to selected extensions or to all extensions
in the Department.
To take the example of Extensions A, B, C, D with the Department Access Code 3901 further,
Extensions A, B, C and D are all members of Department Group 1 with the Access Code 3901.
When there is a new message in the Group Mailbox, all four extensions - A, B, C, D - will get the Message
Wait Notification.
The message wait notification may be a Stuttered Dial Tone or a Voice Message when the extension user
goes OFF-Hook, or blinking of the LED Lamp on the extension, or a Ring.257
To the first extension that answers the notification call, for example, Extension A, the Voice Mail System
informs about the new message(s) waiting in the Department Group Mailbox and in the Personal Mailbox.
"You have <x> new Message in your Personal Mail Box. You have <y> new Messages in your Department
Group Mail Box."
If there is no new message in both mailboxes, the VMS will play the message: "You have Zero new
Message."
257. This will depend on the type of Message Wait Notification configured for the extension in its Voice Mail Settings.
1799
If there is a new message in the Department Group Mailbox, but none in the Personal Mailbox, the
VMS will play the message:"You have <x> new Message in your Department Group Mailbox."
If there is no new message in the Department Group Mailbox, but new message in the Personal
Mailbox, the VMS will play the message: "You have <x> new Message in your Personal Mailbox."
The VMS prompts Extension A to access the Group mailbox: "To go to Personal Mailbox, press 1. To go to
Department Group Mailbox, press 2."
The user of Extension A presses 2, and is taken to the Department Group Mailbox,
VMS prompts A: "Enter your mailbox password". (Enterpwd.wav). Enter your department group mailbox
password.
if 80% of the mailbox memory has been consumed, the VMS prompts the caller: Your Mailbox is 80%
Full. Please Delete few messages. (MB80Full.wav).
if 100% of the mailbox memory has been consumed, the VMS prompts the caller: Your Mailbox is Full.
Please Delete few messages. (MBFull.wav)
Extension A presses 1.
VMS plays the new messages.
After playing the new messages, the VMS cancels Message Wait Notification set for extensions B, C and
D.
Call Forward
Just as calls can be forwarded to a Department Group, a Department Group can also forward its calls to:
an extension
its own Department Group Mailbox
another Department Group
For Department Groups, ETERNITY does not support Call Forward to an external destination number.
You can set Call Forward for Department Group from the SA Mode only.
ETERNITY supports the following Call Forward options for Department Groups:
1800
Call Forward - unconditionally: calls are forwarded to the destination number, without checking the
status or waiting for a response from the Department Group.
Call Forward- if Busy: calls are placed on the Department Group as per the Rotation configured for it and
are forwarded to the set destination, only when all the member extensions of the Department Group are
found to be busy.
Call Forward- if No Reply: when a call is made to the Department group, ETERNITY will place the call as
per the Rotation configured for the Department Group for the duration of the 'Call Forward No Reply Timer
for Department (default: 30sec ). If none of the member extensions answers the call before the expiry of
this timer, the call is forwarded to the destination.
If you select this option, you may set the Call Forward No Reply Timer for Department to the desired value.
This Timer is commonly applied on all Department Groups which set Call Forward No Reply.
Call Forward - if Busy/No Reply: calls made to the Department Group will be routed to the destination, if
all members of the Department Group are busy or when none of the member extensions answered the call
within the Call Forward No Reply Timer for the Department.
Member extensions of a Department Group can set Call Forward on their extensions. However, Call
Forward set for the Department Group will have precedence over Call Forward set by individual
member extensions.
Call Forward set by member extensions in a routing group will be ignored by the system if, the Ignore
call forward set by member extension, when call is routed on Routing/Dept. Group flag is enabled. See
System Parameters for more information.
Call Forward for a Department Group can also be set from SA Mode. See Setting Call Forward for
Department Group.
Again, taking the above example of Department Group 1 further, heres how call forward will work:
Extensions A, B, C, D of Department Group 1 are allowed Call Forward Department Group in their Class of
Service.
Any of them can set Call Forward for Department Group 1. Extensions A, B, C and D can also set Call
Forward on their own extensions.
When any extension or an external caller (also using Auto Attendant or Direct Inward System Access)
dials the Access Code 3901 to call Department Group1, ETERNITY will check the Call Forward option set
for the Department Group and route the call accordingly.
If Call Forward - unconditionally is set, the call will be routed to the destination number, regardless of Call
Forward set by any of the member extensions.
If Call Forward - Busy is set, and the first extension in the Department Group is busy, the system will hunt
for the next free extension in the group. It will continue to hunt for a free extension. If all extensions in the
group are busy, the call will be forwarded to the destination number.
Call Forward unconditional, busy, or busy/No reply set by any member extension will not work.
If Call Forward - No Reply is set, the system will start the Call Forward No Reply Timer Department Group
and place the call as per the Rotation set for the Department Group. If the call is not answered by any of
the extensions before the timer expires, the call will be forwarded to the destination number.
1801
If a member extension that is offered the call has set Call Forward-Unconditional, and the Call Forward No
Reply Timer Department Group has not expired, the call forward set by the extension will be applied. If the
timer expires, the Call Forward No Reply set for the Department Group will be applied.
If a member extension that is offered the call has set Call Forward-No Reply, or No-Reply/Busy, the Call
Forward No Reply Timer (for individual extension) will start simultaneously with the Call Forward No-Reply
Timer Department Group. If the No Reply Timer for the extension expires first, the call will be forwarded to
the destination set for the extension. If the No Reply Timer of the Department Group expires first, before
the call is answered, the call will be forwarded to the destination set for the Department Group.
Call Forward-No Reply Timer can be set from the SE Mode only.
How to configure
The functioning of this feature requires you to do the following:
create Department Groups.
configure Routing Group (each routing group consisting of extensions related to a Department) and
assign the routing groups and appropriate access codes to the department groups.
If you want to provide voice mail facility to the Department Group, you must:
assign a Mailbox to the Department Group.
allow member extensions access to the Department Group Mailbox.
If you want to enable Call Forward to the Department Group, you must
enable 'Department Group Call Forward' in the Class of Service (CoS) of the member extensions.
change, if required the default value of the Call Forward No Reply Timer for Department Group.
1802
Decide the number of department groups you want to create, e.g.: 4 groups.
Group all the extensions you want to put in each department group. You cannot group more than 16
extensions in a single department group.
Decide in what sequence the extensions in each group should ring, that is, which extensions should ring
first, second, third, and so forth.
Decide the access code you want to assign to each department group.
Access
Code
to be
assigned
3901
SLT
DKP
SIP
Extension
2001, 2002,
2003
3010, 3011,
3012
3301, 3302
ISDN
Terminal
Department
Group
Index
Access
Code
to be
assigned
SLT
DKP
SIP
Extension
3902
2015, 2020,
2021
3015, 3020
3305
3916
2023, 2024,
2025
3025
3315
ISDN
Terminal
:
16
The access codes for the department groups and extensions in this table are default access codes.
Now, with this information ready, you may configure the department groups using Jeeves or dialing the
relevant SE commands from a Telephone.
1803
To create a Department group, assign an Access Code for the department group against the Index
Number.
By default, the Access Codes assigned to Department groups are from 3901 to 3916.
If you decide not to use the default access codes, ensure that the access code you assign to each
department group is unique and does not match with any SLT, DKP, ISDN, SIP access code or any feature
access code of the Dial Phase. Refer the topic Access Codes to know more.
To assign Station Access Codes according to your preference and requirment to a range of Department
Groups, see Assigning Access Codes to a Range of Extensions.
1804
You may also assign a Name to the department group to facilitate identification. This name will appear in
the Dial by Name directory along with the department group number. The Name can be a maximum of 18
characters.
Now, enter the Routing Group, that is, the number of the group you created for this department.
Where multiple departments exist, you must create separate routing groups for each department group.
This can be done in two ways:
create the routing groups first and simply enter the relevant routing group number against the
Department Group Index (to which you have assigned the access code).
OR
Click the link of the column Routing Groups to open the page.
Choose the Routing Group number (from 01 to 96) you want to assign to the department group. In each
routing group you can include a maximum 32 extensions as 'members'.
For each routing group you want to assign to a department group, configure the following parameters:
Name: You may assign a name to each Routing Group. Name can be of maximum 18 characters.
By default, it is blank. This name will not be relevant, if you are assigning this routing group for a
Department Group.
Rotation Flag: With this flag, you can enable or disable the rotation of calls in the routing group
which has multiple 'member' extensions. When enabled, each fresh call will land on the extension
which is next to the one that received the last call. This ensures equal distribution of incoming calls
to all the destinations within the routing group. The flag has no relevance if the routing group has
only one member extension.
Member Type: Select the 'Member Type' from the combo box. The extensions that you can add as
members of the group can be an SLT, DKP, ISDN Terminal, SIP Extension, Virtual Extension,
Outgoing Trunk Bundle Group (OTBG) or the Voice Mail Auto Attendant. Include only as many
extensions as you want in the routing group and set the remaining Member Types to 'None'. For
example: if you want to include only two extensions in the routing group, set the Member Type in
the remaining columns (Member 03-Member 32) to 'None.'
Port Number: Enter the software port number on which the SLT/DKP/SIP Extension/ISDN terminal/
Virtual Extension to be grouped is attached. If you select Voice Mail Auto Attendant Profile, enter
the Profile number here. If you have selected OTBG then enter the OTBG number here.
1805
Ring Timer(s): This timer defines the time for which the extension, on which the call lands, should
ring. By default, the ring timer is set to 015 seconds and can be changed.
Continuous Ring Flag: With this flag, you can set an extension to ring continuously until the call is
answered. The first extension will continue to ring even as the system hunts for other extensions in
the routing group to land the call. If the call still remains unanswered, the system will return the call
to the first extension once again. This flag is of no relevance, if there is only one member extension
in a routing group.
Repeat the above steps to include other extensions in the routing group.
For example:
The Customer Care Department of a company has four extensions: 201, 202, 203 and 204 (on software ports 001,
002, 003, and 004 respectively), which needs to be grouped for Department Calls.
Extensions 201 and 202 are SLTs. Extensions 203 and 204 are DKPs.
Requirement: The company wants the following:
1. Call traffic should be distributed equally on all four extensions.
2. Ring sequence should be 201, 202, 203, and 204 (first to last).
3. First 201 should ring for 20 seconds.
4. If no reply, 201 should continue to ring and 202 should ring for 10 seconds.
5. If still no reply, 201 should continue to ring and 203 should ring for 15 seconds.
6. If still no reply, 201 should continue to ring and 204 should ring for 20 seconds.
7. 51 should be the access code for this department group.
Solution: Select a routing group e.g. 03, and configure as follows:
1. 201 as member 01, with member type SLT, and Port number 001.
2. 202 as member 02, with member type SLT, and Port number 002.
3. 203 as member 03, with member type DKP, and Port number 003
4. 204 as member 04, with member type DKP, and Port number 004.
Set member type of members 5 to 32 in Routing Group 3 to 'None'.
1. Enable the 'Rotation Flag' on routing group number 03 to distribute call traffic.
2. Enable the 'Continuous Ring Flag' for member 01 (201) and set the 'Ring Timer' to '20 seconds.
3. Set the Ring Timer of member 02 (202) to 10 seconds. Disable 'Continuous Ring' flag.
4. Retain the Ring Timer of member 03 (203) as default 15 seconds. Disable 'Continuous Ring' flag.
5. Set the Ring Timer of member 04 (204) to 20 seconds. Disable 'Continuous Ring' flag.
6. Assign 51 as access code and routing group 03 to the Customer Care Department.
1806
Mailbox: Select this check box to assign a mailbox to a Department Group. By default, no mailbox is
provided.
Mailbox Size (min): Define the size of the mailbox to any desired value from 00001 to 60000 minutes.
The default Mailbox Size is 5 minutes.
In the Hotel Application of ETERNITY, the default Mailbox size is 999 seconds.
Maximum Message Length (sec): Define the length of each message (in seconds) callers are to be
allowed to record in the Department Group mailbox. Change the maximum message length to any
desired value between 0001 to 9999 seconds.
By default, maximum message length is 15 seconds. In the Hotel Application of ETERNITY, the default
maximum message length is 120 seconds.
New Message Delivery option in Mailbox Full condition: When the Department Group mailbox is
full, you may select one of the following options for delivery of new messages:
Do not offer to record a message: The VMS will not allow the caller to record a message by
declining delivery of the message.
Deliver new message to General mailbox: Select this option if you want the VMS to record the
message of the callers in the General mailbox. The General mailbox is a shared mailbox between
extension users.
Only extension users who have General Mailbox in their Class of Service (COS) are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the member extensions of the Department Groups. Refer Class of Service (COS) for instructions.
1807
Overwrite old messages: Select this option if you want the VMS to overwrite the existing
messages to record the new messages. The VMS starts overwriting the oldest message first.
By default, Deliver to General mailbox is selected.
Play message details after delivery of message: After the extension user has finished listening to a
message in the mailbox, you can also have the VMS play message details such as Date and Time
when the message was recorded, the callers number258, and the extension number dialed by the
caller259 to the extension user.
You may select one of the following options:
Never: The VMS will not play message details to the mailbox owner after playing the message.
Always: The VMS will play message details to the mailbox owner after playing each message.
On Demand: The VMS will play message details to the mailbox owner only when the mailbox
owner requests it. On completion of each message, the VMS will prompt the extension user to
press a digit for date and time stamp. When the mailbox owner presses the digit, the VMS will play
the message details.
Ask Password to access Mailbox: By default, access to the mailbox is not password protected.
Select this check box, if you want the VMS to ask for the password, whenever the mailbox owners
accesses the mailbox.
Any department group extension user can access the Department Group Mailbox by entering the
default password, 1111.
If required, to avoid unauthorized usage you can change the Password. In Mailbox Password enter
the new password. The password you assign may consist of a maximum of 4 digits. Valid Range: 0000
to 9999.
If you want to remove password protection, clear this check box.
Allow Mailbox Management: Mailbox Management allows the extension user to change mailbox
settingsrecord Extension Name for the mailbox, redirect messages from the mailbox, delete all old
messages from the mailbox, record greeting messages for the mailbox. By default, Mailbox
Management is disabled.
To allow the extension user to change the mailbox settings, in Allow Mailbox Management, select
Yes.
To know more about this feature, see Mailbox Settings
Auto Delete Messages: Select the type of messages you want the VMS to automatically delete from
your mailbox. You can select All or Old. Default: None.
Days for Auto Delete Messages: Select the number of days after which you want the VMS to
automatically delete the messages in your mailbox. Default: 90 days.
258. The number of person who left the message in the mailbox.
259. The number of the extension user for whom the message is intended.
1808
Voice Mail Auto Attendant Features: This parameter is applicable only if you are using the VMS Auto
Attendant.
Abbreviated Name: When the VMS is used as Auto Attendant, the callers can be prompted to Dial
by Name of the desired party instead of the extension number.
To allow callers to reach the Department group using Dial By Name, abbreviate the Department
Groups name to the first three letters and enter it in this field.
Announce Name: If you want the VMS to announce the Department groups name when
transferring the call to the extension, select the check box to enable Announce Name. By default,
Announce Name is disabled.
If you enable Announce Name, make sure you record the Department groups name on the VMS.
Refer Recording Extension Names for instructions.
Call Transfer: Select the desired method for transferring the call answered by the VMS Auto Attendant
to the SLT extension. You may select any of the following methods of call transfer for each time zone,
Working Hours (WH), Break Hours (BH) and Non-working Hours (NH):
None: When the caller dials the extension number, the VMS Auto Attendant will check if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
Blind: When the caller dials the extension number, the VMS Auto Attendant will transfer the call on
the extension without checking its status.
Wait for Ring: When the caller dials the extension number, the VMS Auto Attendant will wait for the
extension to start ringing and then transfer the call.
Wait for Answer: When the caller dials the extension number, the VMS Auto Attendant will transfer
the call when the extension answers (goes OFF-Hook).
Screen: The VMS Auto Attendant prompts the caller to record his/her name, puts the caller on hold
and places the call on the desired extension. If the extension is free and answers the call, the VMS
announces the callers name to the extension user and prompts the extension user to choose
whether or not to speak to the caller. If the extension user chooses to talk, the VMS transfers the
call. If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the
extension user and asks the caller to leave a message.
By default, Wait for Answer is selected as Call Transfer method for all time zones.
Message Wait Notification via Call: The message wait notification will be sent to a number
(destination number). This number can be an internal or an external number. To use this feature,
configure the following parameters:
Type: If you want the notifications to be sent as soon as a new message arrives in the mailbox of
the extension user, select Immediate.
If you want the notification to be sent at fixed time schedules, select Scheduled.
If do not want to set message wait notification via call, select None. Default: None.
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Profile: Assign the Profile according to which you want the system to send the notifications. The
Profile determines the time intervals during which the notifications must be sent to the destination
number.
Destination Number: Enter the number on which you want the system to send the notification
calls.
The destination number can be an internal or an external number. The destination number can be a
maximum of 16 digits. Valid digits are 0 to 9, # and *.
When the notification call is answered, the VMS informs the callee of the new message and allows
the callee to access it.
Refer the feature description Message Wait Notification via Call to know more.
Message Wait Notification via Email: The message wait notification will be sent to the email address
of the extension user. To use this feature, configure the following parameters:
Notification: If you want the message wait notification to be mailed to the extension user along with
the new voice message as attachment, select the option Send With Attachment.
If you want only the notification to be mailed, select the option Send Without Attachment.
If do not want to set message wait notification via email, select Do not send. Default: Do not send.
Email Address: Enter the email ID of the extension user to which the notification is to be sent.
Email ID may consist of up to 64 characters. Default: blank.
Extensions users will receive notifications only for the mailbox memory utilization, if you configure the E-mail
Address and select Do not sent as the Notification option.
Refer the feature description Email Based Notification to know more.
Assign the Department Group Mailbox to member extensions within the group. To do this, go to the Voice
Mail Settings page of the extension type.
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To assign group mailbox to SLT extensions in the group, open the Voice Mail Settings page under SLT
Configuration.
To assign group mailbox to DKP extensions in the group, open the Voice Mail Settings page under
DKP Configuration.
To assign group mailbox to ISDN extensions in the group, open the Voice Mail Settings page under
ISDN Terminal Configuration.
To assign group mailbox to SIP extensions in the group, open the Voice Mail Settings under VoIP
Configuration.
Enter the Number of the Department Group to whose group mailbox is to be accessed by the extension(s).
Click the desired Department Group Number tab for which you want to set Call Forward.
The color of the text indicating that Call Forward is set will change to red.
To cancel Call Forward,
The color of the text indicating that Call Forward is not set will change to black.
1811
Scroll with the vertical bar to Call Forward No-Reply Timer Department Group (sec).
Set the timer to the desired value. The range of this timer is 1 to 255 seconds.
1812
02 for DKP
28 for ISDN terminal
34 for SIP Extension
36 for Virtual Extension
16 for OGTBG
Port Number is the Software port number on which the member extension SLT, DKP, SIP Extension,
ISDN Terminal is attached.
Software port number of the SLT, from 001 to 512.
Software port number of the DKP, from 001 to 128.
Software port number of the ISDN Terminal, from 01 to 64.
Software port number of the SIP extension, from 001 to 999.
Virtual Extension is from 01 to 64.
OGTBG Number is from 01 to 32.
E.g.: To include the four extensions in the above example in Routing Group 03, dial:
To set the Ring Timer for each member extension in the routing group, dial:
6503-1-Routing Group-Destination Index-Ring Timer
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is number of the member extension in the routing group from 01 to 32.
Ring Timer is from 000 to 255 seconds. (Default: 015 seconds)
E.g.: To set the Ring Timer for the individual extensions of the Routing Group 03 in the above example,
dial:
To set the Continuous Ring Flag for extensions in the routing group, dial:
6504-1-Routing Group-Destination Index-Flag
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is the number of the member extension in the routing group from 01 to 32.
Continuous Ring Flag is:
0 for disable continuous ring (each member extension in the group will ring for the duration of the 'Ring
Timer' for the group).
1 for enable continuous ring (the first extension in the group will ring till the call is answered).
E.g.: To enable/disable the Continuous Ring Flag for the individual extensions of the Routing Group 03
in the above example, dial:
1813
1814
3163-2-Department Group Index-Department Group Index to default the access codes of a range of
department groups.
3163-* to default the access codes of all department groups.
Exit SE mode.
1815
1816
0 for Disabled
Default:Disabled.
To assign transfer type to the Department Group for Scheduled mode, dial:
2019-1-Department Group-Time Zone-Transfer Type-#* to assign transfer type to a single Department
Group.
2019-2-Department Group-Department Group-Time Zone-Transfer Type-#* to assign transfer type to a
range of Department Groups.
2019-*-Time Zone-Transfer Type-#* to assign transfer type to all Department Groups.
Where,
Department Group is from 01 to 16.
Time Zone: 1 - WH, 2 - BH, 3 - NWH
Transfer type is:
0 - None
1 - Blind
2 - Wait for Ring
3 - Wait for Answer
4 - Screen
Default: Wait for Answer.
Exit SE mode.
How to use
Making a Department Call
Making a department call is the same as calling another extension.
Lift handset
Dial the desired Department Group Number (default: 3901-3916).
You get Ring Back Tone as the call lands on an extension within the department group.
Talk when the call is answered.
Replace handset.
1817
Lift handset.
Dial 1179 (users worldwide). Users in the Philippines, dial 1108.
Dial the Department Group Number whose calls are to be forwarded.
1818
1819
Dial By Name
What's this?
Dial By Name enables extension users to call another extension or an external party by dialing the name of the
person, instead of dialing their telephone number.
This feature is accessible only to users of the proprietary digital key phones and the Extended IP phones of Matrix.
With Dial By Name users need not remember the desired party's telephone number or short codes, that is,
Abbreviated Dialing codes.
For each extension, the database for names used in Dial by Name is drawn from:
the Personal Directory, which is assigned to each extension, wherein up to 25 external party numbers
along with their names may stored. The system uses the Personal Directory to dial external parties by their
names. See Abbreviated Dialing to know more.
Global Directory, which is assigned to the extension in its Class of Service (COS). The Global Directory
is a system-wide list of external party numbers and names. Up to 999 numbers can be stored in this
directory, and parts of the Global Directory (Part 1, 2, 3) can be assigned to each extension in its Class of
Service. See Abbreviated Dialing to know more.
Names of Extensions, which are names of users/departments groups. Their names are assigned to SLT,
DKP and SIP extensions to identify the extension users. Names of Extensions are necessary for making
internal calls using the Dial By Name feature.
How it works
Extension user presses the DSS Key assigned to 'Dial By Name' feature.
On EON48/EON310 models and on SPARSH VP248, press the 'Names' key.
The prompt <Name:
> appears on the phone display.
OR
Instead of scrolling the entire list, the user enters more than one initial letter of the contact's name. The
search is narrowed down to more accurate matches. The phone displays the matching entries in the
directory.
The user must select the desired name by pressing 'Enter' Key.
The system dials out the number stored under the selected name. The name and number are displayed on
the user's phone.
261. The process of entering the names is the same as when writing text messages (SMS) from a cell phone. The keys must be
pressed multiple times in quick succession to enter the desired alphabet.
1820
How to configure
For this feature to work, the following must programmed:
1. DSS Key: A direct station selection (DSS) key must be programmed for the Dial by Name feature. Without
telephone numbers in the directory. Refer the topic Abbreviated Dialing for instructions on programming
the Global Directories.
3. Personal Directory: The names of the external parties must be programmed against their respective
telephone numbers in the Personal Directory. Refer the topic Abbreviated Dialing for instructions on
programming the Personal Directories.
4. Extension Names: Extensions may be SLTs, DKPs, ISDN Terminals or SIP extensions. Refer the topics
the following features, which must be enabled in the Class of Service of the DKP and SIP extension users:
Internal Calls - This is a part of the Basic Features. By default these are enabled.
Global Directory Part 1
Global Directory Part 2
Global Directory Part 3.
Global Directory Part 1 is assigned to the default CoS group 01 assigned to all extensions in the default
Station Basic Feature Template 01.
If you want the names to be drawn from Global Directory Part 2 and Part 3, provided these are
programmed, you must enable these two directories in the CoS of the DKP and SIP extensions.
Refer Abbreviated Dialing, for instructions on programming the Global Directory.
Refer Class of Service (COS) and Station Basic Feature Template for programming instructions on how
to enable a feature in the CoS and how to apply it on extensions.
The system will display the names exactly as they have been programmed in the Personal and Global
Directories and the SLT/DKP/ISDN Parameters. Refer the topic Configuring DKP Extensions.
How to use
For EON48 and Extended IP Phone Users
262. You may also refer the instructions provided under the topic Configuring Extensions: Configuring SLT Extensions, Configuring
DKP Extensions, Configuring ISDN Terminals.
1821
The number of matching entries that will appear at a time on your phone's display will vary according to
your phone's LCD display capacity.
Scroll with the Up/Down navigation keys to reach the desired contact's name on the list.
Press 'Enter' key to select the name.
The system displays the name and number being dialed out.
You get Ring Back Tone or Busy Tone.
1822
Go ON-Hook.
Go OFF-Hook.
Press the DSS Key labeled 'Names' again.
Enter the name/initial letters of the contact's name.
How it works
When a DKP extension user makes an outgoing external call, the number is stored in the Redial Number
List.
The list is updated using the First-In First-Out logic, whereby the earliest dialed number is replaced with
the most recently dialed number.
To use this feature, the DKP user must invoke the Last Number Redial feature.
Doing so, the Redial Number List will appear on the phone display.
The user may now navigate the list, select the number to be dialed out.
The system will dial out the selected number using the same Outgoing Trunk Bundle Group used to place
this call earlier.
If the number had been dialed earlier using Abbreviated Dialing, the system will check for Toll Control
when dialing out the number again from the dialed number directory263.
How to configure
No specific programming required.
How to use
For EON and Extended IP Phone Users Only!
263. Recall that the system does not check for Toll Control when Abbreviated Dialing is used.
1823
Dial 7
1824
Digest Authentication
What's this?
Digest Authentication is a challenge-based authentication service of SIP to authenticate the identity of the
originator of SIP request in the INVITE message. The recipient of the request can ascertain whether or not the
originator of the request is authorised to make the request. When the digest credentials of the originatorUser
Name and Passwordin the INVITE message are authenticated and accepted by the recipient, the originator and
the recipient are connected.
ETERNITY supports Digest Authentication. You may use Digest Authentication to
restrict access to ETERNITY to specific callers.
prevent unwanted or malicious calls.
How it works
The Digest Authentication feature works on the basis of the Digest Authentication Table, in which the credentials,
namely the User Name and Passwords of trusted/authorised calling party SIP devices are stored. You must
configure this table. The Digest Authentication Table is common for all SIP trunks on which this feature is enabled.
When you enable this feature on a SIP trunk, for all incoming calls (SIP requests),
ETERNITY will challenge the identity of the calling party (the SIP device initiating the request) to send its
digest credentials.
When the calling party sends its credentials, ETERNITY authenticates the credentials by matching it with
its Digest Authentication Table.
If a match is found, the calling party will be authenticated and the call will be allowed on the SIP trunk.
If no match is found, ETERNITY will consider it as invalid authentication information and reject the call.
How to configure
To use this feature on SIP Trunks, you must do the following:
Enable Digest Authentication on the SIP trunks you want to use this feature.
Configure the Digest Authentication Table.
You can configure the Digest Authentication Table using Jeeves and from a telephone.
1825
In the User ID field, enter the User ID to be authenticated. The User ID must be within 40 characters.
In the User Password field, enter the corresponding Password. The Password must be within 16
characters.
Now, enable Digest Authentication on the desired SIP trunks. For instructions, see Configuring SIP
Trunks.
1826
Default: Blank.
To configure the User Password for the User ID, dial:
4119-Index-User Password
Where,
Index is from 01 to 99. The User Password should be configured at the same index as the User ID.
User Password may have a maximum for 16 characters. If the User Password has fewer than 16
characters, terminate the command string with #* if using an SLT or press the 'Enter' key if using a DKP.
Default: Blank.
Exit SE mode.
For SE Command for enabling Digest Authentication on SIP trunks, see Configuring SIP Trunks.
1827
How it works
1828
The system waits for the duration of the Minimum Instigation Time (programmable; default: 01 second).
This is the time for which the instigation signal from the sensor should remain present on the DIP for it to
be identified as a genuine signal.
If the DIP is being used for an Automated Control Applications, the system will instigate the Digital
Output Port (DOP) which will turn ON or OFF the Digital Output Port and hence the gadget connected
to it on receiving the instigation.
If the DIP is being used for Security Alarm and Reporting, the system will trigger the alarm device hooter/siren - connected to the Digital Output Port or it will make a call to the external number or the
Routing Group programmed as destination for triggering Security Alarms.
It will also report the alarm call to the group of extensions programmed to receive Security Reporting
calls.
Do not connect devices that do not conform to the specifications of the DIP!
How to configure
Configuring DIP parameters using Jeeves
Whether you are using the DIP for an automated control application or for Security Alarm and Reporting,
you must configure the following parameters:
Instigation Signal: Select the appropriate instigation signal for the DIP: from High or Low, depending
on the application for which it is being used.
'High' state signifies that the DIP is normally open. DIP should be programmed as 'High' when the
sensor connected to the DIP keeps the Loop open and closes it to signal an event.
1829
'Low' state signifies that the DIP is normally closed. DIP should be programmed as 'Low' when the
sensor connected to the DIP normally keeps the Loop closed and opens/breaks it to signal an
event.
Minimum Instigation Time: This is the time for which the instigation signal from the sensor device
should remain present on the DIP to be recognized by the DIP as a genuine signal. The range of this
timer is from 01 to 99 seconds. By default the Minimum Instigation Time is set to 01 second. You may
set the 'Minimum Instigation Time' to the desired value.
1830
Exit SE Mode.
DKP Features
PBX Features
Listed below are the features of ETERNITY that require a Digital Key Phone:
Abbreviated Dialing
Auto Answer
Background Music
Call Chaining
Call Cost Display
Call Duration Display
Call Mute
Dialed Number Directory
Directory Dialing by Name
Dynamic Lock
Forced Answer
Keypad Lock
Live Call Screening
Message Paging
Off-Hook Alert
Room Monitor
Text Message Reply
Time Zone Display
User Status (Presence)
1831
EON48S
EON48P
EON48DS
EON48DP
EON310a
48
48
48
48
35
29
29
29
29
17
Capsense keys
Yes
Yes
Yes
Yes
No
Feature Keys
12
12
12
12
16
16
16
16
DSS keys
LCD display capacity
Yes
Yes
Yes
Yes
Yes
Headset Interface
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Speaker Phone
a.
Full duplex
EON48
EON48S/EON48D-S
2 lines and 24 characters LCD display, full duplex, capsense feature keys
1832
EON48P/EON48D-P
6 lines and 24 characters LCD display, full duplex, capsense feature keys.
LCD Display
The LCD display of EON48P/48D-P/48S/48D-S is backlit and can be tilted at a convenient angle for a clear view of
the text/characters displayed.
The LCD backlight can be turned on and off as well as adjusted for contrast and brightness from the "Phone
Settings" of the DKP Phone Menu.
Ringer LED
The Ringer LED will glow in Blue to indicate incoming internal and external calls.
Navigation Keys
These are 5 capsense keys. The functions of each are described briefly below.
Enter Key: To enter the Menu; when the phone is in the idle state (without any incoming or outgoing call
being made), if you tap the 'Enter' key, you will enter into the 'Menu'.
Enter key is also used to make a selection from the Menu/sub-menu options or to complete an action.
Back Key: To move backwards when dialing a number; to go back one level in the Menu.
1833
Feature Keys
These are 12 capsense keys assigned to important or frequently accessed features of ETERNITY. Refer to the
table given below:
Sr.No.
Description
LED
1.
Voice Mail
2.
Call Back
3.
Cancel
4.
Mute
5.
Conference
No
6.
Transfer
No
7.
Forward
8.
DND
9.
Names
No
10.
Redial
No
11.
Release
No
12.
Hold
No
No
Single Color - Blue
These keys are programmable. However, as you cannot change the labels avoid programming these keys.
Refer the DSS Keys Programming topic for instructions on programming these keys.
Status of Stations and Trunks: The LED of DSS keys assigned to Stations/Trunks glow in three colors to
indicate status of the call event on the Stations/Trunks and on the DKP.
Thus, the status of the DKP user's own Station as well as that of the other Stations and the status of Trunk
lines are indicated by the LED of the DSS keys assigned to those Stations and Trunks on the DKP.
1834
The following table shows the relationship between the color of the LED and various events:
LED
Color
LED Mode
Continuously ON
Slow Blink
Fast Blink
Blue
Red
Violet
Blue indicates the state of the station/trunk you access. For example, when you make a call to another
Station 203, the LED of the DSS key assigned to Station 203 blinks Blue to indicate ringing at the
Station. If you have successfully established speech with Station 203 the LED glows Blue continuously.
Red indicates the state of other Stations/Trunks. For example, if the LED of the DSS key assigned to
Station 201 is glowing Red continuously, it means Station 201 is busy with another Station or Trunk.
Violet indicates the state of the trunk you are in speech with. For example, when you are in speech on
an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
LED
Color
LED Mode
Continuously ON
Slow Blink
Fast Blink
Blue
Violet
Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
The LED of DSS keys assigned to Stations/Trunks glow in a single color - Red - to indicate status of the
call event on the Stations/Trunks and on the DKP.
Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
A feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the
DSS key to which the Auto Redial feature has been assigned will glow Red, when Auto-Redial is set,
and the LED is turned off when the feature is canceled.
1835
Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, and otherwise remain inactive for example, Raid, Interrupt Request, Barge-In, Last Caller
Recall.
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * and #. The dial pad is used for
dialing numbers of stations, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable,
you can assign any other feature/function on this key.
Volume Keys
"+" (plus): This is the increase key, to raise the volume of speech while talking and to decrease the Ringer
volume, when the phone is ringing.
"-" (minus): This is the decrease key, to lower the volume of speech while talking and to decrease the
Ringer volume when the phone is ringing.
Headset Connectivity
The EON48P/48D-P/48S/48D-S provides two Headset interfaces: A 2.5mm Audio Jack and an RJ11 connector at
the bottom of the phone body.
So you can use any stereo headset of standard make with a 2.5 mm single connector or a stereo headset with an
RJ11 connector.
You can also assign Headset key function to any of the DSS keys. Refer the topic DSS Keys Programming for
instructions.
1836
Key Maps
As EON48P/48D-P/48S/48D-S may be the extension of the Operator/s and Executives in an enterprise, and the
extension of the Front Desk Attendant and Guest in hotels, to meet the varied requirements of each user group,
ETERNITY provides Key Maps for Operator, Executive, Hotel Attendant, and Hotel Guest.
Operator/Executive
Hotel Attendant
17
18
Mute
19
3PConf
23
24
25
28
20
21
22
26
27
CallFwd
DND
Names
Redial
Release
01
09
02
10
03
11
04
12
05
CA 4
13
06
CA 3
14
07
CA 2
15
08
CA 1
16
Transfer
abc
Hold
def
4 ghi
jkl
7 pqrs
tuv
9 wxyz
mno
29
Guest
17
18
Mute
19
3PConf
23
24
25
28
20
21
22
26
27
CallFwd
DND
Names
Redial
Release
01
09
02
10
03
11
04
12
05
Retrv
Msg.
13
06
CA 3
14
07
CA 2
15
08
CA 1
16
1
4
Transfer
abc
Hold
def
jkl
7 pqrs
tuv
9 wxyz
ghi
mno
29
17
18
Mute
19
20
21
22
CallFwd
DND
Names
3PConf
24
25
26
27
28
Redial Release
01
09
02
10
03
11
04
12
05
DKP2
13
06
DKP1
14
07
CA 2
15
08
CA 1
16
Transfer
23
Hold
abc
def
jkl
mno
7 pqrs
tuv
9 wxyz
1
4
ghi
29
These key maps can be customized to match the exact requirement of individual users. Refer the topic DSS Keys
Programming for instructions on customizing the Key Maps.
Phone Menu
You can access the following PBX and phone features from the Menu of EON48P/48D-P/48S/48D-S:
Menu option
Description
Call Logs
To view call history of internal and external Missed, Answered and Dialed calls.
You can also edit numbers in the call logs and store them in the Personal Directory.
Contacts
To add, edit, delete names and numbers of contacts in the Global Directory Part 1.
Call Forward
To set and cancel Call Forward-Busy, Call-Forward No Reply, Call-ForwardUnconditional, and Follow Me.
Dynamic Lock
User Status
Keypad Lock
To lock the keypad of the phone (when the keypad is locked, the features Call Log,
Contact, Call Forward, Dynamic Lock, User Status, DND, Call Cost Display, Hotline,
Alarm, Background Music, Change User Password will not be accessible.)
Do Not Disturb
To set/cancel Do Not Disturb on the phone, that is, block incoming internal and external
calls.
Hotline
Alarm
Background Music
Change User
Password
To change User Password (required for using certain features like Call Follow Me,
Dynamic Lock, DISA, Walk-In Class of Service, User Absent/Present, Hot Desk, Voice
Mail) and for customizing Phone Settings.
1837
Menu option
Description
Phone Settings
To customize settings of the phone such as Speech and Ringer Controls, LCD Display
settings (Brightness and Contrast, Backlight ON/OFF), Headset Connectivity, Call
Answering Mode (manual/auto answer).
To exit menu,
Go ON-Hook.
Type of Terminal
EON 48D/EON48/
EON310
EON Soft
SPARSH VP248
SPARSH VP330
1838
NA
NA
NA
NA
Type of Terminal
EON 48D/EON310
EON 48
EON Soft
SPARSH VP248
SPARSH VP330
NA
Type of Terminal
EON 48D/EON310
EON 48
EON Soft
NA
1839
Type of Terminal
SPARSH VP248
SPARSH VP330
Operating EON48
Please refer the User Card for EON48 for instructions on operating the features of ETERNITY using EON.
EON310264
1840
2 lines and 24 characters LCD display, full duplex, fixed feature keys.
Ringer LED
3 0 0 3 R e c e p ti on
LCD
Navigation
Keys
Dial Pad
DSS Keys
(Programmable Keys)
DSS 5
DSS 6
Volume Keys
Speaker Key
(Feature Key)
Feature Keys
(Programmable Keys)
LCD Display
The LCD display of EON310 is backlit for a clear view of the text/characters displayed.
The LCD backlight can be turned on and off as well as adjusted for contrast and brightness from the "Phone
Settings" of the DKP Phone Menu.
Ringer LED
The Ringer LED will glow in Blue to indicate incoming internal and external calls.
Navigation Keys
These are 6 keys. The functions of each are described briefly below.
Enter Key: To enter the Menu; when the phone is in the idle state (without any incoming or outgoing call
being made), if you tap the 'Enter' key, you will enter into the 'Menu'.
Enter key is also used to make a selection from the Menu/sub-menu options or to complete an action.
Back Key: To move backwards when dialing a number; to go back one level in the Menu.
1841
Feature Keys
Sr.No.
Icon
Description
LED
1.
Transfer
No
2.
Hold
No
3.
Names
No
4.
Redial
No
5.
Voice Mail
6.
Forward
7.
Conference
8.
Mute
9.
Speaker
These keys are programmable. However, as you cannot change the labels avoid programming these keys.
Refer the topic DSS Keys Programming for instructions on programming these keys.
1842
Status of Stations and Trunks: The LED of DSS keys assigned to Stations/Trunks glow in three colors to
indicate status of the call event on the Stations/Trunks and on the DKP.
Thus, the status of the DKP user's own Station as well as that of the other Stations and the status of Trunk
lines are indicated by the LED of the DSS keys assigned to those Stations and Trunks on the DKP.
The following table shows the relationship between the color of the LED and various events:
LED
Color
LED Mode
Continuously ON
Slow Blink
Fast Blink
Blue
Red
Violet
Blue indicates the state of the station/trunk you access. For example, when you make a call to another
Station 203, the LED of the DSS key assigned to Station 203 blinks Blue to indicate ringing at the
Station. If you have successfully established speech with Station 203 the LED glows Blue continuously.
Red indicates the state of other Stations/Trunks. For example, if the LED of the DSS key assigned to
Station 201 is glowing Red continuously, it means Station 201 is busy with another Station or Trunk.
Violet indicates the state of the trunk you are in speech with. For example, when you are in speech on
an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
LED
Color
LED Mode
Continuously ON
Slow Blink
Fast Blink
Blue
Violet
Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
A feature like Call Forward requires an LED to show that it has been set or canceled. So, the LED of
the DSS key to which the Call Forward feature has been assigned will glow Red, when Call Forward is
set, and the LED is turned off when the feature is canceled.
1843
Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, and otherwise remain inactive for example, Raid, Interrupt Request, Barge-In, Last Caller
Recall.
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * and #. The dial pad is used for
dialing numbers of stations, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable,
you can assign any other feature/function on this key.
Volume Keys
"+" (plus): This is the increase key, to raise the volume of speech while talking and to decrease the Ringer
volume, when the phone is ringing.
"-" (minus): This is the decrease key, to lower the volume of speech while talking and to decrease the
Ringer volume when the phone is ringing.
Headset Connectivity
The EON310 provides two Headset interfaces: A 3.5mm Audio Jack and an RJ12 connector at the bottom of the
phone body.
So you can use any stereo headset of standard make with a 3.5 mm single connector or a stereo headset with an
RJ12 connector.
You can also assign Headset key function to any of the DSS keys. Refer the topic DSS Keys Programming for
instructions.
1844
Key Maps
As EON310 may be the extension of the Operator/s and Executives in an enterprise, and the extension of the Front
Desk Attendant and Guest in hotels, to meet the varied requirements of each user group, ETERNITY provides Key
Maps for Operator, Executive, Hotel Attendant, and Hotel Guest.
Operator/Executive
Hotel Attendant
Guest
01
01
01
02
02
02
03
03
03
04
04
04
CA 1
05
CA 1
05
CA 1
05
CA 2
06
CA 2
06
CA 2
06
These key maps can be customized to match the exact requirement of individual users. Refer the topic DSS Keys
Programming for instructions on customizing the Key Maps.
Phone Menu
You can access the following PBX and phone features from the Menu of EON310:
Menu option
Description
Call Logs
To view call history of internal and external Missed, Answered and Dialed calls.
You can also edit numbers in the call logs and store them in the Personal Directory.
Contacts
To add, edit, delete names and numbers of contacts in the Global Directory Part 1.
Call Forward
To set and cancel Call Forward-Busy, Call-Forward No Reply, Call-ForwardUnconditional, and Follow Me.
Dynamic Lock
User Status
Keypad Lock
Do Not Disturb
To set/cancel Do Not Disturb on the phone, that is, block incoming internal and external
calls.
Hotline
Alarm
Background Music
Change User
Password
To change User Password (required for using certain features like Call Follow Me,
Dynamic Lock, DISA, Walk-In Class of Service, User Absent/Present, Hot Desk, Voice
Mail) and for customizing Phone Settings.
1845
Menu option
Description
Phone Settings
To customize settings of the phone such as Speech and Ringer Controls, LCD Display
settings (Brightness and Contrast, Backlight ON/OFF), Headset Connectivity, Call
Answering Mode (manual/auto answer).
To exit menu,
Go ON-Hook.
Operating EON310
Please refer the User Card for EON310 for instructions on operating the features of ETERNITY using EON.
EONSOFT
The EONSOFT is a PC-based Digital Key Phone. Based on a graphic user Interface (GUI), the EONSOFT offers all
the features of EON48, making it a substitute for the Digital Key Phone. Its integration with the ETERNITY obviates
the need for a separate telephone instrument.
The EONSOFT can be installed on any personal computer with Windows or NT operating system.
Two PC-based DSS64 Consoles are available to be used with the EONSOFT. You can use either one or both
DSS64 Consoles.
1846
The EONSOFT occupies only a single port, even when both PC-based DSS64 Consoles are used. Thus it supports
all the features of the Digital Key Phone and the DSS64 Console on a single window.
DKP Port
The DKP port connects EONSOFT to the DKP port of ETERNITY's DKP card.
Handset Port:
The Handset port connects the Receiver of the phone, to be used for speech.
The EONSOFT has the provision for attaching a Handset. A handset with spring cord is supplied by Matrix and is to
be connected to the handset jack (RJ12) on the Dongle.
Headset connectivity
EONSOFT supports headset connectivity, providing a MIC and a Speaker interface. Any stereo Headset of
standard make, with dual connectors can be connected to the MIC and the Speaker on the Dongle.
COM Port
The COM port connects EONSOFT to a PC (COM Port).
After EONSOFT has been successfully installed on a PC and the DKP parameters have been configured, each
time you open EONSOFT, the display and keypad of the phone will appear on your PC screen.
1847
The illustration below shows EONSOFT with two DSS64 Consoles attached to it.
Phone Display
The EONSOFT has a 2-line and 24-character display. In the ON-Hook or idle condition, the first line displays the
Station Number and the Station name. The second line displays the Day, Date and Time.
When there is an incoming call, the calling party's number is displayed on Line 2 of the LCD265.
The LCD messages for various call events (dial, transfer, forward, hold, etc.), for prompts, alerts, confirmation,
errors, text messages, are displayed.
265. Only if the Station, to which EONSOFT is connected, has been allowed CLIP facility in its Class of Service.
1848
Status of Stations and Trunks: The LED of DSS keys assigned to Stations/Trunks may change in three
different colors to indicate status of the call event on the Stations/Trunks on EONSOFT.
Thus the status of the Stations and the Trunk lines are indicated by the LED of the DSS keys assigned to
those Stations and Trunks on EONSOFT.
The following table shows the relationship between the color of the LED and various events:
LED
Color
LED Mode
Continuously ON
Slow Blink
Fast Blink
Green
Red
Orange
Green indicates the state of the station/trunk you access. For example, when you make a call to
another Station 203, the LED of the DSS key assigned to Station 203 blinks Green to indicate ringing at
the Station. If you have successfully established speech with Station 203 the LED is Green
continuously.
Red indicates the state of other Stations/Trunks. For example, if the LED of the DSS key assigned to
Station 201 is Red continuously, it means Station 201 is busy with another Station or Trunk.
Orange indicates the state of the trunk you are in speech with. For example, when you are in speech
on an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
Table 2: LED pattern of Call Appearance Keys
LED
Color
LED Mode
Continuously ON
Slow Blink
Fast Blink
Green
Orange
Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
The LED of DSS keys assigned to Stations/Trunks turn Red to indicate status of the call event on the
Stations/Trunks.
1849
Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
A feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the
DSS key to which the Auto Redial feature has been assigned will turn Red, when Auto-Redial is set,
and the LED is turned off when the feature is canceled.
Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, otherwise it remains inactive.
The LEDs of DSS keys to which features like Raid, Interrupt Request, Barge-In, Last Caller Recall are
assigned, will not change color.
Dial Pad
The dial pad consists of 12 keys (non-programmable), which include the digit keys for 0, 1-9, and character keys for
* and #.
Function Keys
These are non-programmable keys on the keypad of EONSOFT which have fixed functions.
Redial: This key is used for redialing the last external number.
Func Key: This key is used for accessing the Phone menu.
: This key is used for accessing the Address Book. The EONSOFT provides the facility of an
Address Book that is integrated with the Standard Windows Address Book, for storing the numbers and
addresses of contacts. So, when a call is to be made, you can select and dial the desired number from the
directory.
Hold: This key is used for putting the caller on hold. This key is also used to make a selection in the Phone
Menu.
: This key is used for going OFF-Hook. It simulates lifting of the handset, pressing of the speaker
key to make or receive calls.
: This key is used for going ON-Hook. It simulates replacing of the handset, pressing of the speaker
key to disconnect.
1850
Navigation Keys
The following keys are used for navigating the phone menu:
'Func' key: This key is used for entering the Phone menu and to go back one level in the menu.
Up and Down keys: The and keys function as the Up and Down keys to scroll the Menu and submenu options. You can scroll up down the menu by clicking and scroll up the menu by clicking .
Speaker key
The 'Spk' key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable; you
can assign any other feature/function to this key.
Shortcut keys
You can use the Keyboard of the PC to operate EONSOFT, with the help of "shortcut keys'. The following table
describes the functions performed when shortcut keys on the keyboard are pressed:
Short Cut Key Label
Description
F1
Help
F2
F3
Spd
F4
Func
F5
F6
Alt+Enter - Hold
F7
Xfr
F8
Spk
Esc
ON-Hook
. (dot/period)
Flash
Ctrl+C
Ctrl+T
Tab (Tab
Backward)
Shift + Enter
1851
Description
Num Pad
Key Maps
EONSOFT can function as a Station for the Operator, Executive, and Hotel Attendant, also Guest (though unlikely
to be used by guests).
Phone Menu
The Phone menu is the same as EON48.
To exit menu,
Press the 'Func' key repeatedly to go back one level in the menu, till you reach 'Menu'.
Or
If you want to use the Keyboard, press the Shortcut key for the desired function.
Tool Tips
You can assign labels and tool tips for the DSS keys, which are displayed to the user on mouse over. You can
assign the function of each key as Tool Tip, to help user in intuitive operation of EONSOFT.
To assign a tool tip,
Call Indication
Incoming Calls are indicated by:
1852
In order for the EONSOFT window to pop up, you must have enabled the 'PopUp When Ring' option. When
this option is enabled and the EONSOFT window is minimized a new incoming call causes the window to pop
up to its full size notifying the user about the new call. When this option is enabled and the EONSOFT window
is maximized, a new incoming call is indicated by the flashing of the Title bar of the window.
When the 'PopUp When Ring' option is disabled and the EONSOFT window is minimized, a new incoming call
is indicated by the flashing of the EONSOFT Title at the bottom bar of the PC screen.
By default, the 'PopUp When Ring' option is enabled.
Operating EONSOFT
EONSOFT can be operated using the keyboard and the mouse.
Making calls
To make calls,
A headset must be connected and 'Headset Connectivity' must be enabled in the 'DKP Parameters'.
Refer DSS Keys Programming for instructions.
If you are using the keyboard instead of the mouse, press the appropriate Shortcut Keys listed above
and use the Number pad on the keyboard to dial digits.
Receiving calls
1853
EON74
EON74, also called Digital Turret or Dealer Board, is a special terminal for Trading Rooms of stock exchanges.
With two separate speech paths (two handsets) and 74 DSS keys, EON74, supports quick and efficient handling of
multiple calls by dealers/agents.
For detailed product description and operation instructions, refer to the EON74 User Guide.
1854
How it works
If the DOP is being used for an Automated Control Application, it will turn on the gadget connected to it
according to the "Gadget Operation Mode" programmed.
There are 9 different Operation Modes. Refer the topic Automated Control Applications for an overview of
the operation modes in which the DOP may be used.
The DOP can be turned ON/OFF by dialing the relevant Feature Command, which in turn switches ON/
OFF the gadget connected to it.
The DOP can be operated also from a remote location from the Direct Inward System Access (DISA)
mode.
ETERNITY remembers the state of DOP during power failure. For instance, a water pump is being
controlled using DOP. If a power failure occurs while the pump is running, the Operator need not turn on
the water pump again on power restoration. The ETERNITY will remember the last state (in this case
pump on) and switch ON the water pump when power is restored.
1855
How to configure
When you connect a gadget to the DOP, you must select the 'Contact Type' of the DOP, and enable the feature
'DOP Turn ON/Turn OFF' in the Class of Service (CoS) group assigned to the extensions which you want to allow
access to operation of the DOP.
In the default Station Basic Feature Template 01 assigned to all extensions of ETERNITY, the feature 'DOP Turn
ON/Turn OFF' is enabled in the default CoS group 01. Thus all extensions of ETERNITY can use switch ON/OFF
the DOP.
If you want to restrict access to DOP operation to selected extensions, simply disable this feature in the default CoS
group, and follow these steps:
Define a new CoS group with DOP Turn ON/Turn OFF enabled.
Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
Assign this new Template to the extensions to which DOP operation is to be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions on
programming.
If required you may also change the default Access Code '1174' assigned to the DOP, and assign a DSS key with
the function of DOP operation on the DKP extensions which are allowed access to the DOP. Refer the topic DSS
Keys Programming for instructions.
1856
Set the Normal Contact Type for DOP-1 for the DOP as appropriate: Normally Open/Normally Close. By
default, the contact type for the DOP is 'Normally Open'.
If you are using ETERNITY PE, you may configure the settings of DOP-1, DOP-2 or DOP-3, as desired.
Exit SE mode.
How to use
The DOP will turn ON/OFF the gadget connected to it on the basis of a Timer or a Schedule or on instigation from
the sensor device connected to the Digital Input Port.
Irrespective of the "Gadget Operation Mode" programmed, the DOP (and hence the gadget connected to it) can be
turned ON/OFF by dialing a Feature Command at any time from an extension of the ETERNITY. This feature
command overrides the Operation mode defined for the gadget.
The extension from which you dial this feature command must have the feature 'DOP turn ON/turn OFF' in its Class
of Service.
1857
DOP Number is the number of the DOP from 1 to 3 to which the gadget is connected.
If using ETERNITY ME/GE, dial 1 for DOP Number. If using ETERNITY PE, dial the number of the
DOP which you want to turn ON/OFF.
1858
1859
DDI is an ISDN Service which allows the caller to call the user on an ISDN compatible PBX or private
network directly without operator intervention.
Using the DDI feature of ISDN, the calls can be made to land directly on the desired extensions.
The T1E1PRI and BRI trunks must be assigned a IC ReferenceNumber and OG Reference Number which
in turn defines the translation logic to handle an IC/OG. For more details refer DDI Routing Table, IC
Reference Table and OG Reference Table corresponding topics.
Each ISDN Trunk is given an Installation Number by the SP. This is the combination of Main Number (MSN
No.) and the DDI Number. The Number is of max.16 digits. This is also known as ISDN Installation
Number.
The MSN number is given by the SP whereas the Directory Numbers can be selected by the User.
However the number of digits to be used for the Directory Number should be informed to the SP.
Please refer the topics DDI Routing Table, Configuring PRI Trunks and Configuring BRI Trunks for
details on programming.
DDI Routing is not supported on T1/E1 trunk line if you have selected E&M as the Signal Type.
How it works
Incoming Call
1860
When the call lands on the ISDN trunk of the PBX, the PBX checks if CLI based routing is enabled on the
trunk. If Yes, the call is routed accordingly. If CLI based routing is not enabled, then the PBX checks the IC
Reference Number assigned to the Trunk.
If the IC Reference Number assigned to the trunk is 00, then the system further follows the logic of
Incoming Call management. For more details please refer the topic Incoming and Outgoing Call Routing.
If any other number is assigned as IC Reference Number to the trunk, the PBX searches the different IC
reference tables for a match. When a match is found, the system matches channel number of the trunk
with the channels of the table. If it does not match then the next table with the same IC reference number
is searched.
When the channel number matches, the system uses the DDI Flexible Reference No. of the table to
identify the DDI Flexible Number table. This table helps the system to route the call to the target extension.
The system first compares the received DDI number (called party number) with the DDI numbers
programmed in the DDI Flexible Number table. If a match (DDI Number) is found, the system goes ahead
with further interpreting the translation logic in the IC reference table else the system searches the next
matching DDI Flexible Number table and repeats the above procedure.
If the number does not match with any number in the DDI Flexible Number tables assigned to the trunk, it
routes the call to the TLG assigned to the trunk.
Once a perfect match is found, the system checks for the DDI routing flag on first extension in the IC
reference table. If it is enabled, the system routes the call to the first extension in the DDI Flexible Number
Table else identifies the extension according to the DDI logic.
Once the extension is identified the system checks the DDI IC routing flag of the extension (Please refer
the topic Station Advanced Feature Template for more details) on which the call is to be routed. If the flag
is enabled the call lands on the extension else the call is routed to the TLG assigned to the Trunk.
When the call lands on the DDI extension, the caller gets the Ring Back Tone. The extension rings for
time=DDI Timer, If the call is not answered the system checks for the Route when No reply flag in the IC
reference Table. If it is enabled the call is routed to the TLG programmed in the IC Reference Table (Trunk
Template) else the call is disconnected.
When DDI extension being called is busy, the caller gets the busy tone. The system checks for the Route
when busy flag in the IC Reference Table. If it is enabled the call is routed to the TLG programmed else
the call is disconnected.
When the call is answered the system checks for DDI OG flag. If the flag is disabled the system does not
send the answering party number to the network. If the flag is enabled the system prepares the OG
number (Answering party number) and sends it to the ISDN Network.
Depending on the Channel Number and the port grabbed, the OG reference number assigned to the port
is identified.
After the OG reference number is identified, the ISDN installation no. is identified from the OG reference
table.
The DDI Flexible reference number is also identified from the OG reference table and this helps in
identifying the flexible number of the calling extension. The equivalent DDI number is found out from the
flexible number. The DDI number replaces the last digits (Number of DDI digits parameter in the DDI
Flexible Number table) of the ISDN installation number. This forms the answering/calling party number.
This is sent to the ISDN network.
When an OG is made by a DDI extension, the MSN Number + DDI number of the extension is sent in the
calling party field.
When an outgoing call is made by a Non-DDI extension, the MSN Number + the first DDI number of the
ISDN =Trunk is sent in the calling party field.
1861
DISA Variants
ETERNITY offers three types of DISA, each with a different method of authentication and level of access:
PIN Authentication-Multiple Calls
DISA with CLI Authentication-Multiple Calls
DISA with CLI Authentication-Single Calls
1862
For this type of DISA, the DISA CLI Authentication Table must be configured first.
How it works
For this feature to work, you must enable the desired DISA variant on the desired trunks: CO, Mobile, SIP,
T1E1PRI, BRI.
The system checks if a DISA variant is enabled on the trunk for the current time zone, that is, working
hours, break-hours and non-working hours.
If a DISA variant is enabled on the trunk, the system processes the call according to the DISA variant
enabled on the trunk.
The caller must dial the DISA Login Code consisting of:
the DISA Feature Access Code.
the number of the extension the caller wants to access.
the user password of the extension.
On successful login, the system starts the DISA Idle State Timer (configurable; default: 20 seconds).
The system waits for the caller to go Off-hook267.
266. If no voice message is recorded, the system plays music-on-hold to the caller.
267. If the caller does not go Off-hook within this timer, the system releases the call.
1863
When the caller goes Off-hook by dialing the Off-hook code #1, the system plays the internal dial tone
and waits for the caller to dial digits.
If the caller dials an external number using a CO trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes)268.
The system waits for the caller to dial digits within the DISA Inactivity Timer.
The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are received before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
The caller can make as many trunk calls and internal calls as the caller wants.
The caller can terminate the DISA login session either by disconnecting from the remote end or by
dialing the Termination Code #9.
The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table269.
The caller gets logged into the Auto Login extension and gets the dial tone of ETERNITY.
At the end of the call, the caller dials the On-hook code #0 to go On-hook. To make another call, the
caller dials Off-hook code #1 and dials the desired number. Thus the caller dials the On-hook and Offhook codes to make as many trunk and internal calls as desired.
If the caller dials an external number using a CO trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes)270. The system waits for the caller to dial digits within the DISA
Inactivity Timer.
The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are received before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table271.
268. DISA Inactivity Timer is not applicable for T1E1PRI lines, BRI lines, SIP and Mobile trunks.
269. If no match is found for the CLI of the caller in the Table, the call will be routed as per the Incoming Call Routing configured in
ETERNITY.
270. DISA Inactivity Timer is not applicable for T1E1PRI line, BRI, SIP and Mobile trunks.
1864
The caller gets logged into the Auto Login extension and gets dial tone of the outgoing trunks selected
for TAC-1 for the current Time Zone (working hours, break hours, non-working hours).
If the caller dials an external number using a CO trunk, the system starts the DISA Inactivity Timer
(configured; default: 2 minutes).
The system waits for the caller to dial digits within the DISA Inactivity Timer. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are received before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
After the external call is completed, that is, the caller disconnects from the remote end or the other
remote called party has disconnected, the caller is logged out.
To make another external call, the caller must call the DISA enabled trunk of ETERNITY again.
In all the variants of DISA, the caller can use all the features allowed in the Class of Service (COS) of the
extension the caller is logged in to (using PIN Authentication or CLI Authentication).
DISA calls in the SMDR report are marked as "O" in the remarks column. See Station Message Detail
Recording-Report.
If DISA is disabled, ETERNITY will route the call by Auto Attendant logic, if Auto Attendant is enabled.
If DISA and Auto Attendant both are disabled, the incoming call will be routed as per the incoming call
routing configured. To know more, see Auto Attendant.
WARNING! This feature allows access to system resources to remote users, and therefore has serious
implications for your system's security. There is a risk of fraudulent calls being made from your system, if a
third party comes to know the authentication PIN or the User Password of an extension number. The cost
of such fraudulent calls will have to be borne by the owner of ETERNITY.
So, protect your system from unauthorized access and misuse by putting strong authentication
mechanisms in place.
Keep PINs strictly confidential.
Change PINs regularly.
Choose PINs that are complex and difficult to guess.
Feature Interaction:
If both, Built-In Auto Attendant and DISA are enabled on the trunk, ETERNITY supports all types of DISA.
If both, VMS Auto Attendant and DISA are enabled on the trunk, ETERNITY supports only PIN
Authentication-Multiple Calls. To know how the VMS handles a DISA call, see VMS DISA Login272.
271. If no match is found for the CLI of the caller in the Table, the call will be routed as per the Incoming Call Routing configured in the
ETERNITY.
272. This feature is supported in Firmware Version V10R10 and later.
1865
How to configure
To provide DISA to remote users you need to do the following configuration:
Select the DISA variant for the Trunks on which you want to apply this feature in their Trunk Feature
Template.
Enable DISA in the Class of Service (COS) of the extensions which you want to allow callers to access
using DISA.
Change the User Password of the DISA extensions, if you selected DISA PIN Authentication-Multiple
Calls. If you selected DISA PIN Authentication-Multiple Calls on a trunk, the default User Password (1111)
will not work. See User Password and System Security - V10R11 and later more information and
instructions.
Configure the related timers, DISA Idle State Timer and DISA Inactivity Timer, if required. See System
Timers and Counts for instructions.
If you have selected the DISA CLI Authentication-Multiple Calls or CLI Authentication-One Call Answer
Signaling on a trunk, you must configure the CLI Authentication Table.
1866
Make a list of remote users and their numbers whom you want to allow DISA.
For each remote users number on your list, write the Extension number of the ETERNITY you want to
allow this extension user to log in.
Open Jeeves.
Under Configuration, click DISA - CLI Authentication. The CLI Authentication Table page opens.
You can configure as many as 999 numbers in this table, by clicking the tabs of the index on the top of the
table.
Refer to the list of remote user numbers and the corresponding ETERNITY extension numbers you made.
In the Calling Partys Number column, enter the number of the remote users whom you want to allow
access to DISA using CLI Authentication. The system will match the CLI of the callers with the numbers
you store here.
For each Calling Party Number, in the Auto Login as field, select the extension Port Type (SLT, DKP, SIP
Extension, ISDN Terminal, Virtual Extension) and Port Number you want to allow access to after the
Calling Party Number is authenticated.
1867
4111-Index-Calling Number-#*
Where,
Index is from 001 to 999.
Calling Number may contain a maximum of 16 digits.The allowed digits are 0-9, #, *, A, B, C, D, +. Use
following codes to enter these digits:
Special Digit
Code
#4
#5
#6
#7
#8
**
##
Meaning
00
000
None
01
001 to 512
SLT
02
001 to 128
DKP
28
01 to 64
ISDN Terminal
34
001 to 999
SIP Extension
36
01 to 64
Virtual Extension
Port Type
For example, to configure extension '3001', which is a DKP with port number 001, as auto login station in
Index 001 of the Table, you must dial 4112-001-02-001.
1868
Exit SE mode.
How to use
If you are a Remote user, to be able to use DISA, you must know:
the number of the Trunk on which DISA is enabled and the variant of DISA enabled on this trunk.
the number of the extension and the user password which you want to access, if using DISA with PIN
Authentication.
the duration of the DISA related Timers: The DISA Idle State Timer and the DISA Inactivity Timer, so that
you may dial digits accordingly, without delay.
If you are a Remote user, to be able to use Authority Codes via DISA, you must know:
the number of the Trunk on which DISA is enabled and the variant of DISA enabled on this trunk.
the duration of the DISA related Timers: The DISA Idle State Timer and the DISA Inactivity Timer, so that
you may dial digits accordingly, without delay.
Code to be dialed
on-hook
#0
off-hook
#1
Flash
#2
Pause
#3
#4
#5
#6
#7
#8
1869
Special Digit/activity
Code to be dialed
#9
##
End of String
#*
To program # when in
SE Mode
####
To use DISA,
Dial the number of the Trunk on which DISA is enabled for the current time zone, Working, Break, Nonworking hours.
ETERNITY answers the call. You will get music or Built-In Auto Attendant Voice Message, if configured.
The features listed below are not supported in the DISA mode.
Auto Call Back
Auto Redial
Call Park
Call Chaining
Self Ring Test
Trunk Reservation
Walk-In Class of Service
Live Call Supervision
1870
Two DSS Consoles can be attached to a single DKP. Each DSS Console occupies a Digital Key Phone Port. For
example, if you attach two DSS consoles to a single DKP, three DKP ports would be occupied.
The DSS Console can be attached with the ETERNITY in the same way as the DKP, EON, and is programmed as
an attachment of the DKP (Refer the sections Installing DSS Consoles and Programming DSS Console Keys for
instructions).
You can attach two DSS consoles to a single DKP. This may be necessary, if you want to access most or all of the
features/functions of the ETERNITY at a single touch of a key.
When a single DSS64 is attached with a DKP, the DSS keys of the DKP as well as all the 64 keys of the DSS64 can
be used. If the DSS72 is used, 72 keys can be used as DSS key, in addition to the DSS keys on the DKP. Similarly,
if two DSS64 are attached to a DKP, 128 additional keys are at your disposal to be used as DSS keys.
Each DSS Console that is attached to a DKP occupies a DKP port. Hence, the more DSS Consoles you attach to
DKPs, the lesser number of DKP ports will be available on the ETERNITY.
1871
DSS Keys
The keys on DSS64 and DSS72 are mapped as follows in the factory default settings:
DSS64: 16x4 (64 Keys)
01
17
33
49
01
25
49
02
18
34
50
02
26
50
03
19
35
51
03
27
51
04
20
36
52
04
28
52
05
21
37
53
05
29
53
06
22
38
54
06
30
54
07
23
39
55
07
31
55
08
24
40
56
08
32
56
09
25
41
57
09
33
57
58
10
34
58
11
35
59
12
36
60
13
37
61
14
38
62
15
39
63
16
40
64
17
41
65
18
42
66
19
43
67
20
44
68
21
45
69
22
46
70
23
47
71
24
48
72
10
26
42
11
27
43
59
12
28
44
60
13
29
45
61
14
30
46
62
15
31
47
63
16
32
48
64
You can assign Station numbers or features/functions to the keys on the DSS Console in the same way as you
would assign functions to the DSS keys of various models of EON, so that they can be accessed easily simply by
pressing a single key.
LEDs
Each DSS Console key is equipped with an LED which glows in single (Red) or in tri-color (Green, Red, Orange)
depending on the function assigned to it.
When a Station or Trunk is assigned to a DSS Console key, the LED functions as a tri-color LED to show the status
of the Station (whether ringing, busy, in speech, on hold). When a Feature is assigned a DSS Console key, it
functions as a single color LED to indicate whether the Feature has been accessed or activated (for example:
whether the feature is set or canceled).
The LED color and cadence of the DSS Console keys is the same as that of the DSS keys of EON48/EON310.
Refer the topic Digital Key Phone-Operation to know more.
Remember, as not all Features/Functions require an LED, the LED of the DSS keys function only if the LED is
relevant for the feature/function which is assigned to the keys.
1872
Distinctive Rings
What's this?
Distinctive Rings are ringing patterns used for distinguishing between different types of call events.
ETERNITY supports the following types of call events:
1. Internal Call
2. Priority Internal Call
3. External Call
4. Alarm Call
5. Auto Call Back Call
6. Auto Redial Call
7. Message Wait Call
8. SE Mode (Programming Ring)
9. Operator Alarm
10. Emergency
11. Self Ring
12. Call Supervision
13. Door Phone Call
14. Presence
15. Emergency Conference
16. Conference
With Distinctive Rings, it is possible to use ring cadence of user's choice for each of these call events. For instance,
Triple ring can be set for 'Priority Internal Calls' and long rings can be set for 'Alarm Calls'.
A set of ring types is called Distinctive Ring type. The default Distinctive Ring Types are:
Call Event
Ring Type - T1
Ring Type - T2
Ring Type- T3
Internal Call
Double
Double
Trunk Call
Double
Long Slow
Long Slow
Short Slow
Short Slow
Short Slow
Auto Redial
Self Alarm
Long Fast
Long Fast
Long Fast
Emergency
Long Fast
Long Fast
Long Fast
Operator Alarm
Long Fast
Long Fast
Long Fast
Message Wait
Call
Short Fast
Short Fast
Short Fast
Programming
Ring
Continuous
Continuous
Continuous
Self Ring
Short Slow
Short Slow
Short Slow
Priority
Triple
Triple
Triple
Call Supervision
Continuous
Continuous
Continuous
1873
Call Event
Ring Type - T1
Ring Type - T2
Ring Type- T3
Door Phone
Triple
Triple
Triple
Presence
Continuous
Continuous
Continuous
Emergency
Conference
Triple
Triple
Triple
Conference
Triple
Triple
Triple
Short Fast
750-750
Short Long
500-1500
Short Very
Slow
750-2250
Long Fast
1500-500
Long Slow
1000-4000
2000-4000
Double
400-200-400-2000
Triple
400-200-400-200-4002000
Ring Type - T1
Ring Text
Internal Call
internal
Trunk Call
Double
external
Short Slow
acb
autord
Self Alarm
Long Fast
selfalram
Emergency
Long Fast
emergency
Operator Alarm
Long Fast
opratoralarm
Message Wait
Short Fast
msgwait
Programming Ring
Continuous
prog
Ring Test
Short Slow
test
1874
Call Events
Ring Type - T1
Ring Text
Priority
Triple
priority
Call Supervision
Continuous
callsup
Door Phone
Triple
doorph
Presence
Continuous
presence
Emergency
Conference
Triple
emergencyconf
Conference
Triple
conf
The Ring Text is sent in the Alert-INFO field of the INVITE message and the corresponding Ring Type is played on
terminal registered as the SIP Extension, if the terminal supports Distinctive Rings.
The Ring Text is programmable. You can change the Ring Text, if required.
How to configure
At the time of installation, when you select the Region (as per the geographical location of the system), and set
the system to default, ETERNITY loads the country-specific Distinctive Ring Type defined for the selected Region.
Refer the topic Default Settings for the default Distinctive Ring Type applied to your country/region.
However, if required, you can change the default Ring Pattern and the Ring Text (for SIP Extensions) loaded by the
system.
When you change the Ring Texts for the Ring Types in ETERNITY, you must configure the same Ring Text in the
SIP phones.
1875
Select the desired Ring Pattern (Ring Type) for each call event that you want to customize.
You can also customise the Ring Text of each call event on SIP extensions. The text can be a maximum of
20 alphanumerical characters.
1876
Exit SE mode.
Demonstration of rings
It is possible to demonstrate Ring Types to users by dialing the SE commands on DKP and SLT extensions of
ETERNITY.
By default, the system will play each Ring Type as demonstration for 30 seconds.
Users of ETERNITY may be acquainted with the different Distinctive Rings played by the system so that they can
associate the terms used to describe the rings with the sound emitted by the system for each ring.
Enter SE mode.
1877
You get the prompt 'Go Idle for Ring' on your phone display.
Go Idle.
Exit SE mode.
Enter SE mode.
1878
Exit SE mode.
Extension users
Operator for extension users, referred to as DND-Remote.
Doing so, calls from other extensions will be barred. However, the extension user would continue to receive:
Alarm calls.
Reminder calls.
Auto Call Back calls.
DND has three supplementary features: DND-Override, DND Text Message and Voice Message for DND
Notification.
DND when set/cancelled from the SA mode, will not depend on the assigned CoS.
DND-Override
As the feature title suggests, 'DND-Override' allows the caller to land on the called extension, despite DND set on
the extension.
DND-Override will not work if the called extension has 'Privacy from DND-Override' enabled in its Class of Service.
1879
The ETERNITY supports 9 different DND Text Messages, which can be changed as per user requirement by the
System Engineer. User can select and set on their phones any of the DND messages programmed by the System
Engineer.
How it works
A, B and C are extension users.
B has EON, while C has an SLT.
B has DND-Override in his Class of Service, C does not have this feature.
DND Text messages as well as Voice Message Notification for DND have been programmed by the System
Engineer.
A has set DND on his extension with the DND Text message 'In Meeting'273.
B calls A.
As B has DND-Override, the Voice Message for DND Notification is played to B once, and the DND
message 'In Meeting' set by A appears on B's phone display. B gets routing Beeps.
To exercise DND-Override, B must dial '4' the feature access code for 'DND-Override' during either during
the Voice Message or during the routing Beeps.
B gets Ring Back Tone, if A's extension is free.
B gets Busy Tone, if A's extension is busy.
However, if A has Privacy from DND Override, B will get error tone and the DND message set by A
appears on B's phone.
If B fails to dial the DND-Override code before the end of the routing beeps, error tone will be played to
him.
C calls A.
As C has an SLT, C will get only the Error tone.
But as Voice Message for DND Notification is programmed in the system, C will be played the prerecorded message once.
273. While DND and DND Text Message can be set from any phone, DND Text Message can be viewed on EON only.
1880
Since C is not allowed 'DND-Override' in his Class of Service, he cannot exercise this feature during the
Voice Message.
At the end of the voice message, C will be played error tone.
How to configure
For this feature to work, 'DND', 'DND-Override' and 'Privacy from DND-Override' must be enabled in the Class of
Service group of the extensions which is to be allowed this feature.
Besides these, the System Engineer may program the DND Text Message and the Voice Message for DND
Notification, as per user requirements.
enabled.
Similarly, if 'DND-Override' is to be to be allowed to the Operator and a few other extensions, follow these
steps:
1. Define a CoS group with DND-Override enabled. If DND is also to be allowed, enable both DND-Override
Override' is to be enabled.
Repeat the same steps to allow 'Privacy from DND-Override' to selected extensions. For extensions that are to be
allowed 'DND' as well as 'Privacy from DND-Override', enable both features in the CoS group in the Station Basic
Feature Template applied on these extensions.
Similarly, for extensions that are to be allowed 'DND', 'DND-Override' and 'Privacy from DND-Override', enable all
three features in the CoS group that you prepare for these extensions.
1881
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed programming
instructions on how to prepare a CoS in the Station Basic Feature Template and how to apply this template on
extensions.
DND Message
Do Not Disturb
Unavailable
In a Meeting
In a Conference
Try on Mobile
On Vacation
On Business
Trip
Out of Office
With a Guest
You can use these default message options or program messages as per user preferences.
1882
All the default text messages appear in the DND message field. Change the DND text messages as
required. Click the field and enter your custom DND text message.
Enter SE mode
To reload default DND text messages, dial command 1501. The default DND Text Messages are given
above in the table.
1883
Exit SE Mode.
When you dial the command string for DND Text Messages, alphanumeric dialing will be automatically
enabled on the DKP - EON.
You can enter alphabets by pressing the relevant key repeatedly, just as you would do when you type
text messages on your mobile phone (SMS mode). For example, to enter the alphabet 'b' you must
press the digit '2' twice in quick succession.
Also, refer the topic Digital Key Phone-Operation for instructions on entering alphanumeric characters
using the keypad of EON.
How to use
DND set/canceled by Extension Users
For EON and Extended IP Phone Users
To set DND:
Dial 18
Scroll to select from any of the DND messages that appears on the phone's display:
1884
Do Not Disturb
Unavailable
In a Meeting
In a Conference
Try on Mobile
On Vacation
On Business Trip
Out of Office
With a Guest
Dial 18-0
You get a text message 'DND Cancelled' on the phone's display and confirmation tone.
Lift handset.
Dial 18-0
Replace handset.
DND-Remote
For EON and Extended IP Phone Users
To set DND for an extension user:
Do Not Disturb
Unavailable
In a Meeting
In a Conference
Try on Mobile
On Vacation
On Business Trip
Out of Office
With a Guest
1885
You get a text message 'DND Set on <Extension Number>' and confirmation tone.
Go Idle or you get dial tone after confirmation tone.
To cancel DND-Remote,
DND-Override
For EON and SLT Users
1886
Door Phone
What's this?
A Door Phone is typically used for monitoring an entrance door. It is installed in place of the Doorbell.
The door phone is similar to any ordinary phone; except it does not have a hook-switch or a dial pad. Usually, it is a
weather tight box, equipped with a button like a doorbell, which visitors press.
ETERNITY offers the Door Phone feature exclusively on its ETERNITY PE model and its variants.
When visitors press the Door Phone Call Button, the phone programmed to receive the call (landing destination)
rings. The user of the called phone can answer the door phone call by simply lifting the handset and talk to the
visitor at the door. The user of the called phone can have the door opened for the visitor by either physically
appearing at the door or operating a door lock release device.
The Door Phone feature of the ETERNITY PE allows users to operate the door phone from a remote location (offpremises) by having their calls routed to an external number.
The Door Phone feature of the ETERNITY PE is very convenient to have at:
Delivery entrances: It is not necessary to have company personnel monitor delivery entrances. They can
just answer the Door Phone instead.
Residences and Apartment entrances: The identity of the visitors can be screened before letting them
in. The occupants of the house can greet their guests/relatives and let them enter the house even in their
absence by answering the door phone from their current (remote) location and opening the door for them.
Matrix does not supply Door Phones, but only the Interface to connect Door Phones. Any standard 4-wire
Door Phone can be connected to the ETERNITY PE.
How it works
The Pre-requisites
A four-wire Door Phone connected to the Door Phone Port on the Door Phone Card of the ETERNITY PE.
ETERNITY PE supports 3 Door Phone Ports, so you can connect as many four-wire Door Phones. This
may be required in buildings with more than one entrance.
1887
A Door Lock Release device connected to the Digital Output Port (DOP) on the Door Phone Card of the
ETERNITY PE, if a Door Lock Release is to be used in conjunction with the Door Phone.
ETERNITY PE supports 3 Digital Output Ports, allowing you to connect as many Door Lock devices.
The Process
When a visitor arrives at the entrance and presses the door phone button, the system senses the doorbell
and places the call on the phone programmed as the landing destination. The destination phone may be
extensions of the ETERNITY or an external fixed line or mobile number.
The ETERNITY plays a distinct Ring Type on the extension, to indicate to the extension users that it is a
door phone call. The Ring Type is programmable; by default Triple Ring is set as ring type.
The visitor is played Ring Back Tone while the destination extension rings for the duration of the 'Door
Phone Ring Timer (programmable)274.
If the destination extension does not answer the call within this Timer, the Door Phone call is dropped and
the Door Phone goes idle.
When the destination extension answers the call, the Door Phone circuit is activated and two-way speech
is established between the visitor and the extension user.
The extension user may now either physically appear at the door to open it, or dial the Open Door Lock
Code (default: 1173) from the current extension.
the extension user puts the call on Consultation Hold, music-on-hold is played to the visitor.
the extension user dials the Feature Access Code to 'Open the Door' (default: 1173).
the Door Lock is opened for the duration of the Timer 'Open Door for Time, (programmable; default: 5
seconds)275,276.
as soon as this Access Code is dialed, speech is reestablished with the visitor.
the extension user invites the visitor to enter the building.
The Door Lock closes on the expiry of the "Open Door for" timer.
The Door Phone goes idle only when the extension user goes ON-Hook.
The extension user can open the Door Lock also by dialing the feature command to operate the DOP.
For this, however, the extension must have the facility "DOP Turn ON/Turn OFF' in its Class of Service
(COS). Refer the topic Digital Output Port (DOP) for operation instructions.
It is possible for the any extension of the ETERNITY PE to establish speech with the Door Phone even
when it is idle, by dialing the unique access code assigned to the Door Phone.
274. The "Door Phone Ring Timer" determines the time for which the landing destination shall ring for the door phone call. This Timer is
necessary because often visitors may press the door phone switch as they would do a door bell, for one or two seconds only,
whereas the call must remain present for a longer period of time for it to be answered.
275. This is the time for which the door will remain open.
276. An error tone will be played to the extension user for the duration of the error tone timer, if a DOP has not been assigned to the
Door phone port.
1888
When the visitor presses the Door Phone button, the ETERNITY makes a call to the pre-programmed
external number, which may be a fixed line or a mobile number.
The ETERNITY connects the Door Phone to the Trunk Port on the basis of the Outgoing Trunk Bundle
Group assigned to the Door Phone Port. When connected to the Trunk Port, the Door Phone receives Call
Progress Tones of the CO Network.
When the external number answers the call, speech is established between the visitor and the external
number.
The called party on the external number ascertains the identity of the visitor.
1173)277.
the Door Lock is opened for the duration of the Timer 'Open Door for Time, (programmable; default: 5
seconds)278,279.
as soon as this Access Code is dialed, speech is reestablished with the visitor.
the visitor is invited to enter the building.
the called party on the external number disconnects the call280; the Door Phone goes idle.
This way, it becomes possible to answer the door bell call from a remote location and open the door lock
from the remote location.
The Door Lock can be opened by the called party on the external number also by dialing the command
for operating the DOP to which the Door Lock is connected, instead of dialing the 'Open the Door'
access code. Refer the topic Digital Output Port (DOP)for operation instructions.
The called party on the external number can also call the Door Phone using Direct Inward System
Access (DISA).
It is also possible for the called party on the external number to make multiple calls, while putting the
visitor at the Door Phone on Consultation Hold. For this, the party must be logged in Direct Inward
System Access (DISA) mode.
The Door Phone feature of the ETERNITY PE offers the flexibility of selecting the Routing Mode for Door
Phone calls. The system supports two Door Phone Call Routing Modes:
Manual: Whenever you want to route the Door Phone Calls, you can alternately select the landing
destination, that is, select a routing group at one time, the external number (the programmed fixed line
or mobile number) the next time, as required. The extension user can change the Call Routing Mode
viz. routing group or external number, as and when required.
277. This access code is to be dialed only when in speech with the visitor. If this access code is dialed when there is no speech, an error
tone will be played to the extension user for the duration of the Error Tone Timer.
278. This is the time for which the door will remain open.
279. An error tone will be played to the extension user for the duration of the error tone timer, if a DOP has not been assigned to the
Door phone port.
280. If the trunk used for placing this call is a Two-Wire Trunk, dial #0 to disconnect the call.
1889
Scheduled: You can program the system to route Door Phone Calls automatically to the landing
destination phone according to the current Time Zones, that is, working hours, break hours and nonworking hours. For example, you can have Door Phone calls during working hours landed on the
extensions of the ETERNITY, while Door Phone calls during non-working hours can be landed on an
external number. It is also possible to assign different routing groups (extensions) for different Time
Zones.
For this, a Time Tables must be assigned to the Door Phone Port, defining the Time Zones, that is,
working hours, break hours and non-working hours. Routing Group must be defined for each Time
Zone. Similarly, if external number is selected as landing destination for any Time Zone, the number
must be programmed. The ETERNITY will follow the Time Table programmed and route the calls
according to the current Time Zone to the destination phone (routing group/external number)
programmed for the current Time Zone.
The landing destination - Routing Group and the External Number - can be programmed only by the
System Engineer (SE).
The Routing Mode for Door Phone Calls, 'Scheduled' or 'Manual' mode can be set by the SE as well as
the extension users (User Mode). However, extension users must have the feature "Door Phone
Settings" enabled in the 'Class of Service (COS) group assigned to their extension. With this feature
included in their Class of Service, the extension users can switch between scheduled and manual
modes by dialing the Feature Command.
The Time Table, the Routing Group, and the External Number for routing the Door Phone Calls can be
programmed by the System Engineer only.
The Outgoing Trunk Bundle Group assigned to the Door Phone Port used for routing door phone calls
to an external number will be common for both Scheduled and Manual modes.
How to configure
For the Door Phone feature to work, you must program the related set of Door Phone Parameters, allow 'Door
Phone Settings' in the Class of Service of extensions defined as the landing destination for Door Phone calls. If a
Door Lock is installed in conjunction with the Door Phone, you must also enable access to operate the Digital
Output Port in the Class of Service of extensions that are to be allowed to open the Door Lock.
1890
The parameters for Manual Door Phone Routing will appear on your screen.
Access Codes: Each Door Phone port can be given a unique access code. The Access Code allows the
extension user to call the related Door Phone. When an extension user dials the Door Phone Access
Code, the extension user will get connected to the related door phone and can speak to the visitor.
By default, access codes are not assigned to the Door Phone ports. You can assign Access Codes to the
Door Phone ports as per your preference. The Access Code can be a may be a maximum of 6 digits.
Refer the topics Access Codes and Conflict Dialing to know more.
Name: It is also possible to assign a 'Name' to each door phone port for easy identification. This is useful
where there are multiple entrances each having a door phone installed. The Name you assign to a Door
Phone may be a maximum of 18 characters. By default Door Phone Port1, 2 and 3 are named 'Door
Phone 1', Door Phone 2' and Door Phone 3' respectively.
Route Door Phone Calls to: Select the landing destination for Door Phone calls: Routing Group or
External Number.
Routing Groups: program this parameter if you have selected Routing Group as the landing destination.
By default Routing Group 01 is assigned.
If you want to assign a different Routing Group, click the 'Routing Group' link to open this page.
Create another routing group, for instance, 03, and click 'Submit' at the bottom of the page to save
changes.
Return the Door Phone Parameters page.
Assign the number of the Routing Group you created (03) in this field.
External Number: Program this parameter if you have selected External Number as the landing
destination. Enter the fixed line or mobile number to which door phone calls should be routed. The number
must not exceed 16 digits.
1891
OGTB Group: Define the Outgoing Trunk Bundle Group (OGTBG) through which the door phone calls to
the external number should be routed. By default OGTBG 01 is selected.
The Door Phone parameters page will open again, with additional parameters.
Scroll to the right with the horizontal scroll bar, and program the following parameters for Scheduled
routing.
Time Table: Each door phone port can be assigned a Time Table, with defined working hours, break
hours and non-working hours, so that calls landing on Door Phone can be routed to the destination phone
(routing group or external number) according to the current time zone. By default Time Table 1 is assigned
to all Door Phone ports. You may retain the default Time Table or assign a different Time Table.
If you want to retain the default Time Table, check if the time zones defined in the default Time Table 1
fulfill your requirement by opening the 'Time Table' link in this column.
The Time Table page will open, make the necessary changes, if required and click 'Submit' at the
bottom of the page to save your settings.
Return to 'Door Phone Parameters' page.
If you want to define a new Time Table for the door phone port, open the 'Time Table' page and define
the time zones in a new Time Table, for example, 2. Click 'Submit' at the bottom of the page.
Return to the 'Door Phone Parameters' page.
Enter the number of the new Time Table (2) you defined in the 'Time Table field of the Door Phone port
you want to assign it to.
Door Phone Call Routing Mode: Select 'Scheduled' as the mode of call routing for the door phone port.
When you select one of these options, only those parameter fields related to the option will be editable on
the page.
The Routing Mode can be selected also by the extension user.
1892
Route Door Phone Calls to: define the landing destination for the door phone calls. You can select a
different landing destination - a Routing Group or an External Number - for each Time Zone, that is,
Working Hours (WH), Break Hours (BH) and Non-Working Hours (NH).
You cannot program a different External Number for each Time Zone.
Routing Group: program this parameter if you have selected Routing Group as the landing destination for
any of the Time Zones. By default Routing Group 01 is assigned.
If you want to assign a different Routing Group, click the 'Routing Group' link to open the page.
Create another routing group, for example, 04, and click 'Submit' at the bottom of the page to save
changes.
Return the Door Phone Parameters page.
Assign the number of the Routing Group you created (04) to the related Time Zone.
External Number: Program this parameter if you have selected External Number as the landing
destination for any of the Time Zones. Enter the fixed line or mobile number to which door phone calls
should be routed. The number must not exceed 16 digits.
OGTB Group: Define the Outgoing Trunk Bundle Group (OGTB) through which the door phone calls to
the external number should be routed. By default OGTBG 01 is selected.
Door Opener: Select the number of the Digital Output Port (DOP) to which the Door Lock for the Door
Phone is connected. For example, if the Door Lock for Door Phone 1 is connected at DOP1, select this
port as Door Opener. By default, no Door Opener has been assigned.
Open Door For: This is the Time for which the Door Lock should remain open. Program this parameter if
using a Door Lock with the Door Phone. The range of this timer is from 01 to 99 seconds. By default the
Open Door Timer is set to 05 seconds.
Door Phone Ring Timer: This is the time for which the Door Phone will ring on the landing extension in
the Routing Group. The range of this timer is from 001-255 seconds. The duration of this Ring Timer is set
to 30 seconds by default. You may program the desired duration.
Click Submit at the bottom of the page to save your settings.
1893
1894
3221-*-Time Table- #* to assign the same Time Table to all door phone ports.
Where,
Door Phone is the software port of the Door Phone from 1 to 3
Time Table is from 1 to 8
Default: 1 for all Door Phones
1895
1896
Default: 30 sec.
Exit SE mode.
The option of selecting the Routing Mode for Door Phone calls, that is, 'Scheduled' or 'Manual'.
The facility to select the Call Routing Destination in the 'Manual Mode'.
In the default Station Basic Feature Template 01 assigned to all extensions of the ETERNITY PE, the default Class
of Service group 01 has the feature "Door Phone Settings" enabled. So, all extensions of ETERNITY PE are by
default allowed this feature.
There is no need to program this feature, if the default COS group 01 is assigned to the landing destination
extensions.
If a different COS group is assigned to the landing destination extensions, check if this feature is enabled in the
assigned COS group and enable this feature if not already included.
If you want to allow this COS feature exclusively to the landing destination extensions and deny this feature to all
other extensions, follow these steps:
1. Define a CoS group with Door Phone Settings enabled.
2. Prepare a Station Basic Template with this CoS group applicable in all the time zones.
3. Assign this newly prepared Station Basic Feature Template to the extensions on which 'Door Phone
Settings' is to be enabled.
Similarly, for extensions that are to be denied the 'Door Phone Settings' feature, follow the same steps, but
disable the 'Door Phone Settings' in the COS group and apply the Station Basic Feature Template on the
extensions which are to be denied this feature.
Refer the topic Class of Service (COS) and Station Basic Feature Template for instructions.
1897
However, if you want to allow this COS feature exclusively to the landing destination extensions and deny this
feature to all other extensions, follow these steps:
1. Define a CoS group with DOP Turn ON/OFF enabled.
2. Prepare a Station Basic Template with this CoS group applicable in all the time zones.
3. Assign this newly prepared Station Basic Feature Template to the extensions to which 'DOP Turn ON/OFF
is to be allowed.
For extensions that are to be denied the 'DOP Turn ON/OFF' feature, follow the same steps, but disable the
'Door Phone Settings' in the COS group and apply the Station Basic Feature Template on the extensions
which are to be denied this feature.
Refer the topics Class of Service (COS) and Station Basic Feature Template for instructions.
How to use
To select a Call Routing Mode281:
Users world wide
Dial 1171-Access Code of the Door Phone-1 for Scheduled Mode.
Dial 1171-Access Code of the Door Phone-2 for Manual Mode.
Users in the Philippines
Dial 1101-Access Code of the Door Phone-1 for Scheduled Mode.
Dial 1101-Access Code of the Door Phone-2 for Manual Mode.
To select a Call Routing Destination for Manual Mode:
Users world wide
Dial 1172-Access Code of the Door Phone-1 to route calls to an extension.
Dial 1172-Access Code of the Door Phone-2 to route calls to an external number.
281. Manually by the user of the extension programmed as the landing destination for the Door Phone calls.
1898
DSS Call Pick-Up-Station - internal or external calls ringing on any extension, can be picked-up by
pressing the DSS Key assigned to that extension on the users DKP/Extended IP Phone.
DSS Call Pick-Up-Trunk - incoming calls on any trunk for any extension can be picked-up by pressing the
key assigned to that trunk on the users DKP/Extended IP Phone.
DSS Call Pick-Up-Trunk is not applicable for SIP Trunks.
How it works
For this feature to work, you must:
enable the desired Call Pick-up in the COS of the extension user.
assign DSS Keys with LED to the desired extensions/trunks on their DKP/Extended IP Phone.
This is how DSSS Call Pick-Up works:
Extension user A has configured a DSS Keys for extension 2007 and CO trunk 1 on his/her DKP/Extended
IP Phone.
When a call lands on extension 2007 and it rings, the DSS Key assigned to 2007 blinks fast in Blue color
to indicate that the extension is ringing. A presses the DSS Key to pick-up the call ringing on extension
2007.
If DSS Call Pick-Up-Station is not enabled in the COS assigned to extension user A, the DSS Key blinks
fast in Red color to indicate that the extension is ringing. However, A will not be able to pick-up the call
ringing on extension 2007.
Similarly when there is an incoming call on CO trunk 1, the DSS Key assigned to CO trunk 1 blinks fast in
Violet color to indicate that there is an incoming call on the trunk. A presses the DSS Key to answer the
call on the trunk.
If DSS Call Pick-Up-Trunk is not enabled in the COS assigned to extension user A, the DSS Key will
remain steady-on in Red color to indicate that the there is an incoming ringing call. However, A will not be
able to pick-up the incoming ringing call on that trunk.
If DKP 2001 is a station in the Trunk Landing Group of CO Trunk1, 2001 can pick-up incoming calls on
CO1 when the LED glows in Red to indicate the incoming call, even if DSS Call Pick-Up-Trunk is disabled
in the COS of 2001.
1899
Feature Interactions:
Call States: DSS Call Pick-Up-Station and DSS Call Pick-Up-Trunk are possible only when the calls are in
ringing state.
Priority: If multiple calls are ringing on an extension, when you press the DSS Key assigned to that
extension, you will be connected to the ringing call with the highest priority. To know more, see Priority.
How to configure
To provide this feature to extension users, you must
enable these features in their Class of Service to be assigned to the users. For instructions, see Class of
Service (COS) and Station Basic Feature Template.
configure the DSS Keys for the desired extensions and trunks on their DKP/Extended IP Phone. To know
more about assigning DSS Keys, see DSS Keys Programming.
How to use
For EON & Extended IP Phone Users
To use DSS Call Pick-Up-Station:
When the DSS key assigned to the station blinks color fast in blue to indicate that the station is ringing,
press the DSS Key.
You are in speech with the calling party.
You may talk.
1900
When the DSS key assigned to the trunk blinks fast in violet color to indicate that there is an incoming call
on the trunk, press the DSS Key.
You are in speech with the calling party.
You may talk.
Dynamic Lock
What's this?
Dynamic Lock allows extension users to change the Toll Control Levels (Calling Permissions) of their extensions on
their own by dialing a code.
The System Administrator/Operator can also change the Toll Control Levels of extensions using Dynamic Lock.
With this feature, extension users can prevent misuse of outgoing call facility from their extensions, especially in
their absence.
Dynamic Lock also forms the basis of 'Call Privilege', which is feature of the Hotel Application of ETERNITY. Refer
the ETERNITY Hospitality System Manual to know more.
There are four types of Toll Control Levels, starting from Level 0 to Level 3 that can be set for extension phones.
For each Toll Control Level from 0 to 3, a 'Call Privilege282 is to be assigned and corresponding numbers strings to
be allowed and number strings to be denied for each Call Privilege are to be programmed.
Toll Control - Level 0 is Time Zone based, wherein the Call Privilege Type must be defined for each Time
Zone, that is, Working Hours, Break Hours and Non-Working Hours. For instance, you may define 'All
Calls' as Call Privilege for Working Hours, 'Local Calls' as Call Privilege for Break Hours and 'No Calls' as
Call Privilege for 'Non-Working' Hours.
By default, Call Privilege 'All Calls' is selected for all three Time Zones.
Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
The Call Privilege for each of the above Toll Control Levels can be redefined according to user
requirements. For example, Toll Control Level 3 can be programmed for allowing all types of calls by
selecting 'All Calls' as Call Privilege Type and Level 0 can be programmed to allow only Local Calls, by
programming the strings of 'Local Numbers'.
Refer the feature description for Toll Control to know more.
Extension users who are allowed the Dynamic Lock feature in their Class of Service, can set the Toll
Control Level in two ways:
Manually: the extension user changes the Toll Control Level of the extension whenever s/he wants by
dialing the feature access code.
282. The Call Privilege types are: No Calls, Local Calls, Regional Calls, National Calls, International Calls, All Calls and Limited Calls.
1901
For example, an extension user having Toll Control Level 2 (default: National calls) can restrict long
distance dialing on his/her extension by setting the Toll Control Level to 1 (default: Local calls) before
leaving the workplace. On return, the user can restore the previous Toll Control Level, by setting it back
to Level 2.
Thus the extension user sets Dynamic Lock s/he manually selects the desired Toll Control Level for
his/her extension and restores the original Toll Control Level assigned to the extension.
Automatically: the extension user changes the Toll Control Level of the extension using the Dynamic
Lock Timer. The user sets the Timer to the desired number of minutes. On the expiry of this Timer, the
system restores the original Toll Control Level assigned to the extension.
For example, an organization has defined Toll Control Level 0 as Local Calls, and Level 3 as All Calls.
An extension user of this organization is assigned Level 0. When this extension user wants to make
international calls, he sets the Dynamic Lock Timer and selects Toll Control Level 3. At the end of the
timer, Level 3 gets locked and Toll Control Level 0 is reapplied on the extension phone.
The changing of Toll Control level requires the user to dial the 4-digit User Password. The system will
not accept the default User Password (1111). The extension user must first change the default User
Password.
The Dynamic Lock Timer must be set to '00' when using Manual Dynamic Lock.
Dynamic Lock when set/cancelled from the SA mode, will not depend on the assigned CoS.
How it works
The Pre-requisites
The Toll Control Levels 0 to 3 are programmed in the Station Basic Feature Template applied on the
extension.
The Process
For Dynamic Lock - Manual
The user of extension A sets the Dynamic Lock manually by entering the User Password and selecting the
desired Toll Control Level.
OR
The Operator sets Dynamic Lock manually for an extension by entering the extension number and
selecting the Toll Control Level.
1902
The user of extension A sets the Dynamic Lock by entering the User Password, setting the Dynamic Lock
Timer, and selecting the desired Toll Control Level.
OR
The Operator sets Dynamic Lock for an extension by entering the extension number, setting the Dynamic
Lock Timer, and selecting the Toll Control Level.
Now, whenever a call is made from extension A, the system checks for Toll Control Level.
The system then checks the associated Lists of allowed and denied numbers.
If the Toll Control Level is 0, then Toll control is time zone based, that is, working hours, break hours
and non-working hours. The outgoing call is allowed/denied as per the Call Privilege and the
corresponding Allowed and Denied Number List programmed for that time of the day by the System
Engineer.
If the Toll Control Level is 1, 2, 3 the outgoing call is allowed/denied as per the Call Privilege and the
corresponding number list programmed for each level.
If Dynamic Lock - Automatic has been set by user/Operator, the system waits for the duration of the
Dynamic Lock Timer set for the extension. At the end of each outgoing call made during the period of this
Timer, the system will restart the Timer again. The system will change the Toll Control back to the pervious
Level when no call outgoing call is made till the expiry of this Timer.
If Dynamic Lock - Automatic has been set by user/Operator, and an internal call is made during the period
of the Dynamic Lock Timer, the system will check for the 'Decrement Dynamic Lock Timer Internal Calls'
feature in the Class of Service of allowed to the extension. If this feature is enabled, the system will start
the decrement of the Dynamic Lock Timer. The system will change the Toll Control back to the previous
level on the expiry of this Timer. However, if the 'Decrement Dynamic Lock Timer' feature is disabled in the
Class of Service, the system will reset the Toll Control as described in the previous step.
If Dynamic Lock - Manual has been set, the extension user/Operator must set the Toll Control Level back
to the previous Level.
Feature Interactions
Redial and Auto Redial: The system will check for Toll Control Level when an extension, on which
Dynamic Lock is set, attempts Redial or Auto Redial.
Emergency Number Dialing: All extensions will be able to dial Emergency numbers always, regardless
of the Toll Control set on them.
ETERNITY provides for separate programming of Emergency Numbers, which remain unaffected by
Dynamic Lock set on the phones. Refer the topic Emergency Dialing to know more about this feature.
How to configure
For this feature to work, it must be enabled in the Class of Service of the extensions; Toll Control Level must be
programmed in the Station Basic Feature Template of the extensions. The user must change the default User
Password.
1903
In the default COS group 01, 'Decrement Dynamic Lock Timer for Internal Calls' is disabled.
Retain the default template, if you want to allow this feature to all extensions and keep the Decrement Timer
disabled.
If you want to deny Dynamic Lock to all extension, simply disable this feature in the default COS group 01 of Station
Basic Feature Template 01.
If you want to allow Dynamic Lock and/or the Decrement Dynamic Lock Timer for Internal Calls only to selected
extensions, then follow these steps:
a. Define a new CoS group with Dynamic Lock and the Decrement Dynamic Lock Timer for Internal Calls
enabled.
b. Prepare a Station Basic Template with this CoS group applicable in all the time zones.
c. Assign this newly prepared Station Basic Feature Template to the extension on which 'Dynamic Lock' and
How to use
Dynamic Lock, Manual and Automatic, can be set by extension users as well as from their own extensions or from
the SA mode by the Operator.
The extension user/Operator must first set the Dynamic Lock Timer and then change the Dynamic Lock Level.
To set Dynamic Lock-Manual, the extension user/Operation must set the Dynamic Lock Timer to 00.
Recall that
1904
When the Dynamic Lock-Manual is set (Timer set to 00), the extension user/Operation must dial the
feature access code to restore the previous Toll Control Level.
When Dynamic Lock-Automatic is set (Timer set to desired number of minutes), the system will restore
the previous Toll Control Level at the end of the Timer.
The extension user must change the default User Password to be able to set the Dynamic Lock on his/
her extension. Refer the topic User Password for instructions on changing the password.
Dial 142.
OR
Dial 142.
OR
Dial 141.
OR
1905
Dial 1072-002283.
283. Ensure that the feature 'SA Extension' is enabled in the Class of Service (COS) allowed to the extension from which this code is
being dialed.
1906
OR
Dial 1072-002284.
OR
Dial 1072-002285.
OR
284. Ensure that the feature 'SA Extension' is enabled in the Class of Service (COS) allowed to the extension from which this code is
being dialed.
285. Ensure that the feature 'SA Extension' is enabled in the Class of Service (COS) allowed to the extension from which this code is
being dialed.
1907
1908
E1 Maintenance
Whats this?
The E1 Maintenance consists of Error Counts (Performance Statistics), Alarms and Loop Back Tests.
G.775 is also considered for detection of defect conditions like Loss of Signal (LOS), Loss of Frame (LOF),
Alarm Indication Signal (AIS), etc.
To elaborate, the Digital line can have transmission errors. All the errors will not generate an Alarm. Few
severe errors generate Alarms. However, all the errors are logged in the System Fault Log.
The SNIIC (Subscriber Network Interface Integrated Circuit), is used to interface E1 line to ETERNITY. It
supports error counters listed in the table given below.
Each error detected by the ETERNITY ME Card T1E1PRI/port is sent to the master in the form of an
event.
The master counts these errors and prepare a statistical record if the condition matches. For example,
Severely Errored Seconds Count is incremented when one OOF (Out of Frame) event reaches the master
or more than 320 framing errors reach the master.
Signaling on ISDN PRI trunks consists of messages transported over the D-Channel, which is channel 24
on T1 interface or channel 16 on E1 interface. This signaling can be provided by two methods:
FAS: The D channel can provide signaling for the other B channels on the same interface. This is
called Facility Associated Signaling (FAS).
NFAS: The D channel can provide signaling for the other B channels on more than one interface. This
is called No facility Associated Signaling (NFAS). The signaling arrangements, the capability is
supported to designate a D channel on one interface to be a backup to a D channel on another
interface in case of failure. This is called D channel backup.
E-bit
CRC-4 Error
This counter is incremented when the received frame has CRC-4 errors.
1909
Following parameters form the statistical record. This can be generated in the form of a report as shown below:
Performance Parameter
Seconds/Count
Error Seconds
000 to 255
000 to 255
000 to 255
000 to 255
Unavailable Seconds
000 to 255
000 to 255
000 to 255
000 to 255
00000 to 65535
000 to 255
00000 to 65535
Out Of Frame (OOF)-Out of Frame is the occurrence of a particular density of framing error events. OOF is
declared when three consecutive frame alignment signals have been received with an error. OOF ends when;
1910
Severely Errored Framing Seconds (SEFS)-It is a second with either one or more OOF defects or a detected AIS
defect.
Unavailable Seconds-It is defined as a second in which E1 service is unavailable. An unavailable state is declared
at the onset of 10 consecutive severely errored seconds and is cleared on onset of 10 consecutive seconds with no
severely errored seconds.
Positive Slip Seconds-It is defined as a second in which a frame is repeated to account for frequency drift
between ET2 and the network.
Negative Slip Seconds-It is defined as a second in which a frame is deleted to account for frequency drift between
ET2 and the network.
Loss of frame count-Loss of Frame is declared after 2.5 seconds of continuous loss of signal or OOF. LOF is
cleared after 10 seconds of continuous no loss of signal or OOF.
Line Errored Seconds-It is a second in which one or more than one line code violation error occurs.
Excessive Zeroes Error Count-This counter is incremented when excessive zeroes are received on the line or
when line code violation error occurs.
CRC-6 Error Count-This counter is incremented when a CRC-6 Error is detected.
Alarms
RED Alarm:
1911
The master logs this event in the System Fault Log as RED Alarm <Slot No.> <Port No.> at HH:MM:SS.
This alarm is cleared when the signal is acquired back and persists for 10 seconds.
The LED is turned OFF. The master logs this event in the System Fault Log as RED Alarm Cleared <Slot
No.> <Port No.> at HH:MM:SS.
YELLOW Alarm:
If equipment is connected in downstream (Drop and insert mode, that is, NT mode) then on receipt of Yellow
Alarm, a Blue Alarm will be sent on the port, which is configured in NT mode.
BLUE Alarm:
Card Status
Port Status
1912
Card Status
Port Status
L2 Green
L2 Yellow
Card
Status
Port Status
L2 Green Steady
L4 Green
L4 Yellow
1913
E&M Connectivity
Whats this?
E&M connectivity feature of ETERNITY offers seamless connectivity in PLCC network and also in between
various communication products like PBX, Router, Lease Line.
E&M interface is widely used interface to connect such diverse equipment. For example, in a PLCC
network, number of PLCC EPAX needs to be connected. As shown in the figure 1 of PLCC network,
number of EPAXs are connected with each other through E&M tie lines.
Say, an existing PBX capacity needs to be expanded beyond the configuration limit of a PBX. Installing
one more PBX and connecting both the PBXs through E&M interfaces can get us the desired expansion.
1914
PBX-A
PBX-B
T1
E&M1
E&M1
T1
T2
E&M2
E&M2
T2
E&M3
E&M3
PSTN
Tn
S1 S2
PSTN
Tn
Sn
S1 S2
Sn
S1 to Sn are stations.
1915
T2
PSTN
Tn
S1 S2
2001
E&M1
E&M2
E&M3
E&M4
E&M1
E&M2
E&M3
E&M4
PBX-A
T1 (8x8x8)
PBX-B
(16x8)
E&M5
E&M6
E&M7
E&M5
E&M6
E&M7
E&M8
E&M8
S8
S1
3001
T1
T2
PSTN
Tn
S2
S16
PBX-C
(24x8)
S1
4001
S2
S24
Figure 4: Two PBX systems located far from each other connected to each other using E&M connectivity.
PBX-A
Router
T1
PSTN
T2
E&M
E&M
T3
S1
Sn
2001
Lease
circuit,
VSAT
2099
PBX-B
T1
PSTN
T2
Router
E&M
E&M
T3
S1
Sn
2001
2099
How it works
E&M interface is achieved using an ETERNITY ME Card E&M. An E&M port of ETERNITY-PLCC EPAX has dual
personality: both of a station and a trunk. An E&M port works like a station interface for any incoming call to it and
works like a trunk interface when any station makes an outgoing call through it. However, please note that a trunk
line cannot be connected to an E&M port. Also a SLT or DKP cannot be connected to an E&M port.
1916
How to configure
Please refer Station Basic Feature Template, Station Advanced Feature Template, Trunk Feature Template
for more details.
Relevant Topics:
1. Station Basic Feature Template
726
2. Station Advanced Feature Template
738
3. Trunk Feature Template
874
1917
Emergency Conference
Whats this?
Emergency Conference enables you to establish a Conference between a pre-defined group of extensions using a
feature access code.
This feature can be used to call and consult with a group of people in emergency situations.
The number of parties that can be included in an Emergency Conference group depends on the Multiparty
Conference Capacity of your model of ETERNITY. The ETERNITY supports between 6 to 21 parties in a Multiparty
conference depending on the model you are using. For details, see Conference-Multiparty.
This feature is not aplicable for ETERNITY PE3S.
How it works
For this feature to work, you must do the following:
First decide the key persons in the organization who should be parties to the Emergency Conference.
Form a Department Group with the extensions of these key persons as members. A single Department
Group can have up to 32 extensions. For more information on forming Department Groups, see the topic
Department Call.
For example, you have formed a Department Group for Emergency Conference, with the extensions A to G as
members. The Access Code assigned to the Department Group is 3901. H is the initiator of the conference.
Another Emergency Conference is formed with Department Group 3902, having extensions J to P as members and
J as the initator of the Conference.
Now, extension H wants to initiate an Emergency Conference.
This is how the feature will work:
H dials the feature access code for Emergency Conference, followed by access code of the Department
Group (3901).
All extensions in the Department Group (extensions A to G) which are free will start ringing. The system
will play Emergency Conference ring (default: Triple Ring) on the Extensions. Extensions that are busy will
not be included in the call.
If there are DKP/Extended IP Phone extensions in the group, and these phones have a Call Appearance
free, the system will ring these extensions on the free Call Appearance, but will not wait for the extensions
to become free.
1918
The number of extensions that the system will ring will depend on the resource occupied in the
conferencing circuit in the system at the time of initiation of the Emergency Conference. For example,
ETERNITY GE supports up to 15 participants in a single Multi-party conference, and 5 simultaneous
conferences, if all conferences involve 3 parties. Now, if there are already three such simultaneous
Multiparty conferences in the system when Emergency Conference is initiated, the system will ring on the
first 6 extensions of the Emergency Conference group, even if the group has more extension members.
This is because the system supports 15 participants and 9 parties are already involved in the three
simultaneous conferences.
Extension A goes Off-Hook to answer the call first. A gets connected to the initiator of the conference,
extension H.
Two-way speech is established with extension A and H. All other extensions continue to ring.
When another extension, B goes Off-Hook to answer the call, A and H get a beep, and three-way speech
is established between A, B, H.
Thus, whenever a new member joins the conference, all other extensions already in conference will get a
beep, if the flag Play Beep when Conference/Dial-In Conference Starts is enabled in the System
Parameters.
If the conference initiator, extension H, goes idle, all other extensions in the conference will still be in
conversation.
Only the initiator of the conference, extension H, can merge the conference using the phone menu.
From the Multiparty Conference Menu, H selects Merge Conference to merge with another ongoing
Emergency Conference.
H selects 3902 from the list of ongoing Emergency Conferences. Both the Emergency Conferences
(3901 and 3902) are merged.
Only the initiator of the conference, extension H, can cancel the conference. The initiator of the conference
can cancel the conference at two stages:
When speech is established with one or more member extensions of the Emergency Conference
department group.
Or
During Ring Back Tone, as the system rings on the extensions of the group, after the initiator of the
conference has dialed the feature access code.
To cancel the Emergency Conference, extension H must dial the feature access code for Cancel
Conference, 190 (default).
1919
How to configure
To provide this feature to extensions,
You must enable the feature Emergency Conference in theClass of Service (COS) of the extensions in
their Station Basic Feature Template. By default, this feature is enabled on all extensions, so all
extensions can use this feature.
If the extension you are providing this feature is a DKP or an Extended IP phone, you may program a DSS
key on the phone with this feature.
You must also create a Department Group as Emergency Conference group. For instructions, see
Department Call.
By default, the system plays a beep when the Emergency Conference starts. If you do not want the beep
to be played, you must disable the flag Play Beep when Conference/Dial-In Conference Starts. For
instructions, see System Parameters.
This flag is common for other features like Conference-Multiparty, Conference Dial-In and Raid.
By default, the system plays Triple Ring as Ring Type for Emergency Conference. If necessary, you may
configure a different ring type. For more information and for instructions, see Distinctive Rings.
How to use
You can initiate an Emergency Conference also using Direct Inward System Access (DISA).
1920
Dial 1177
Dial Department Group Number.
1921
How it works
When an extension of ETERNITY makes an emergency call by dialing an Emergency Number,
the system hunts for a free trunk in the OGTBG selected for routing the emergency number, and dials out
the number from a free trunk.
simultaneously, the system informs the Operator by ringing on the Operator extensions for the duration of
the Emergency Reporting Call-Ring Timer (configurable; default: 10 minutes).
If Operator is a DKP or an Extended IP Phone, it will ring continuously, and an emergency message will be
displayed on the LCD.
The emergency message shows the number of the extension which has made the emergency call, in this
case, extension 2003.
To acknowledge the Emergency call the operator must press the enter key. The acknowledged Emergency
calls are logged into the System Activity Log.
If the Emergency Call is not acknowledged by the operator, the emergency call is logged into the
Emergency Alarms Log. To know more about the Emergency Alarms Log, see Emergency Alarms Log at
the end of the topic.
Also see the topics Configuring Emergency Number Dialing and Emergency Dialing.
1922
How to configure
To be able to use this feature, the Emergency Dialing Reporting flag must be enabled in the System Parameters.
See System Parameters for instructions. By default, this flag is enabled.
If required, you can change the Emergency Reporting Call-Ring Timer, see System Timers and Counts.
A list of the last 20 unacknowledged Emergency calls appears with the following details:
Date and Time when the Emergency call was initiated from that Extension.
Press the enter key to acknowledge the Emergency Call. The message "Emergency Acknowledged"
appears on the screen.
The system plays the Confirmation Tone followed by the Dial Tone.
The acknowledged Emergency call is removed from the Emergency Alarms Log and is logged into the
System Activity Log with the details of the extension that acknowledged the call.
1923
Emergency Dialing
What's this?
The ETERNITY supports dialing of Emergency number immediately without any blocking.
When an extension user dials an Emergency number, the system will hunt for a free trunk from the outgoing trunk
bundle group selected for the emergency number. See Configuring Emergency Number Dialing.
The system will not apply any of the following on the extension dialing the Emergency number:
Toll Control (Allowed Denied Numbers, Dynamic Lock)
Call Budget (even when call budget is consumed)
Call Duration Control
Automatic Number Translation
The system will allow the extension to dial the Emergency number even in the following conditions:
the extension is in Off-Hook state.
the extension is in Standby Mode.
the extension has grabbed the trunk line (using Trunk access code or selective access)
the call state is in any state: Ringing, Busy, Error, Confirmation.
SIM card is not present in the Mobile port.
Mobile port is not registered with the network.
SIM PIN is not valid.
the keypad of the extension phone is locked.
Emergency Numbers will always be out dialed through the OGTBG you have selected for the numbers, except
when you have grabbed a trunk using Selective trunk access code/Selective Trunk Access DSS key. In which case,
the number will be dialed only from the trunk you have grabbed.
Emergency Number will not be out dialed in the following cases:
If the trunk port from which number is to be routed (CO, Mobile, T1E1PRI, BRI, SIP) is disabled.
If the hardware related to dialing of Emergency number is not present.
If the Emergency number is dialed from the SE Programming mode.
Emergency dialing will not work if Mains Power to the ETERNITY fails.
How to configure
The Emergency numbers are fixed as per the Region where ETERNITY is installed, you can add emergency
numbers, as required. For instructions, see Configuring Emergency Number Dialing.
How to use
To dial an Emergency number,
Go Off-Hook
Dial the Emergency Number
OR
Dial Trunk Access Code-Emergency Number
1924
For example:
Dial 0-112
Wherever the Trunk Access Code conflicts with the Emergency Number, the emergency number
should be dialed after dialing the Trunk Access Code.
Let us take the example of Australia, where the emergency number is 000 and the trunk access code is
0. Now, when an extension user of ETERNITY located in Australia dials 0 of the emergency number,
the system will consider it as trunk access code and will apply the trunk access code logic.
Therefore, in such cases, the extension user must first dial the Trunk Access Code and then the
Emergency Number. In this case, the extension user must dial 0-000 for emergency number dialing, so
that the system will not wait for the Conflict Timer to apply the Trunk access code logic.
1925
SPARSH VP248S - the standard model, with a 2-line x 24-character LCD display.
SPARSH VP248P - the premium model, with a 6-line x 24-character LCD display.
It is a powerful extension, supporting a host of phone and ETERNITY features, as listed below.
IP Phone Features
ETERNITY Features
The SPARSH VP248 supports ETERNITY features. A few of these are listed below:
1926
Abbreviated Dialing
Auto Answer
Call Chaining
Call Cost Display
Call Duration Display
Call Mute
Dialed Number Directory
Directory Dialing by Name
Dynamic Lock
Forced Answer
Keypad Lock
Message Paging
Off-Hook Alert
Room Monitor
User Status (Presence)
M e nu
DND
Forward
S a t 0 1 0 5: 3 0
Names
Redial
Release
Hold
CLIR
CA4
Hotline CA3
12
SIP2
CA2
SIP1
CA1
11
10
Ringer LED
3
4
C u rs o r
Touch sense feature keys
Dial Pad
Programmable feature keys
11
12
10
abc
4 ghi
jkl
6 mno
7 pqrs
tuv
9 wxyz
def
Handset
4P4C Spring Cord
1927
Model
SPARSH VP248S
SPARSH VP248P
48
48
29
29
Capsense keys
Yes
Yes
2 lines x 24
characters
6 lines x 24
characters
Touch Keys
No
No
No
No
Yes
Yes
Headset Interface
Yes
Yes
Yes
Yes
Full duplex
Full duplex
Speaker Phone
SPARSH VP248S
2 lines and 24 characters LCD display, full duplex, capsense feature keys
1928
SPARSH VP248P
6 lines and 24 characters LCD display, full duplex, capsense feature keys.
LCD Display
The LCD display of SPARSH VP248 is backlit and can be tilted at a convenient angle for a clear view of the text/
characters displayed.
The LCD backlight can be turned on and off as well as adjusted for contrast and brightness from the "Phone
Settings" of the SPARSH VP248 Phone Menu.
Ringer LED
The Ringer LED indicates incoming internal and external calls. The LED Cadence will match with the Ring
Cadence of the incoming internal/external call.
The Ringer LED changes colour according to the type of call, as described in the table below.
Type of Call
Internal Call
Red
External Call
Blue
Alarm
Blue
Red
Priority
Red
Programming mode
Red
1929
Navigation Keys
The phone has 5 touch sense navigation keys to be used to move the cursor and scroll through Menu options.
is the Enter key, used to make a selection or to complete an action.
is the Up key, used to scroll upwards while navigating the 'Menu'.
is the Down key, used to scroll downwards while navigating the 'Menu'.
is the Forward key, used to move the cursor.
is the Back key, used to move the cursor, return from the Sub-menu to the Main Menu.
Feature Keys
These are 12 capsense keys assigned to important or frequently accessed features of ETERNITY. Refer to the
table given below:
Sr.No.
Description
LED
1.
Voice Mail
2.
Call Back
3.
Cancel
4.
Mute
5.
Conference
No
6.
Transfer
No
7.
Forward
8.
DND
9.
Names
No
10.
Redial
No
11.
Release
No
12.
Hold
No
No
Single Color - Blue
These keys are programmable. However, as you cannot change the labels avoid programming these keys.
For instructions on programming these keys, see Phone Key Settings under Configuring Matrix SPARSH VP248
- Extended IP Phone.
1930
Status of Extensions and Trunks: The LED of DSS keys assigned to Extensions/Trunks glow in three
colours to indicate status of the call event on the Extensions/Trunks and on the Extended IP Phone.
Thus, the status of the Extended IP Phone user's own Extension as well as that of the other Extensions
(i.e. Extended IP Phones and SLTs) and the status of Trunk lines are indicated by the LED of the DSS
keys assigned to those Extensions and Trunks on the Extended IP Phone.
The following table shows the relationship between the colour of the LED and various events:
LED
Colour
LED Mode
Continuously ON
Slow Blink
Fast Blink
Blue
Red
Violet
Blue indicates the state of the extension/trunk you access. For example, when you make a call to
another Extension 203, the LED of the DSS key assigned to Extension 203 blinks Blue to indicate
ringing at the Extension. If you have successfully established speech with Extension 203 the LED
glows Blue continuously.
Red indicates the state of other Extensions/Trunks. For example, if the LED of the DSS key assigned
to Extension 201 is glowing Red continuously, it means Extension 201 is busy with another Extension
or Trunk.
Violet indicates the state of the trunk you are in speech with. For example, when you are in speech on
an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
LED
Colour
LED Mode
Continuously ON
Slow Blink
Fast Blink
Blue
Violet
1931
Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
The LED of DSS keys assigned to Extensions/Trunks glow in a single colour - Red - to indicate status
of the call event on the Extensions/Trunks and on the Extended IP Phone.
Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
A feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the
DSS key to which the Auto Redial feature has been assigned will glow Red, when Auto-Redial is set,
and the LED is turned off when the feature is canceled.
Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, and otherwise remain inactive for example, Raid, Interrupt Request, Barge-In, Last Caller
Recall.
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * and #. The dial pad is used for
dialing numbers of extensions, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable,
you can program any other feature/function on this key.
Volume Keys
"+" (plus): This is the increase key, to raise the volume of speech while talking and to decrease the Ringer
volume, when the phone is ringing.
"-" (minus): This is the decrease key, to lower the volume of speech while talking and to decrease the
Ringer volume when the phone is ringing.
Headset Connectivity
The SPARSH VP248 provides two Headset interfaces: a 2.5mm Audio Jack and an RJ11 connector at the bottom
of the phone body.
So you can use any stereo headset of standard make with a 2.5 mm single connector or a stereo headset with an
RJ11 connector.
1932
You can also program any of the DSS keys to function as the Headset key. For instructions on programming the
key, see Phone Key Settings under Configuring Matrix SPARSH VP248 - Extended IP Phone.
Key Maps
As SPARSH VP248 may be the extension of the Operators and Executives in an enterprise to meet the varied
requirements of each user group, these key maps can be customized to match the exact requirement of individual
users. For instructions on customizing the Key Maps, see For instructions on programming these keys, see Phone
Key Settings under Configuring Matrix SPARSH VP248 - Extended IP Phone.
By using Key Templates you can prepare and assign common key maps to all or as many Extended IP Phones as
you want, at one go.
ETERNITY also offers the flexibility to personalize the Key Maps of each Extended IP Phone, instead of using the
Key Templates. For example, if you have assigned a common Executive Key Template to 12 Extended IP Phones,
but you want to reassign some of the keys on two of these Extended IP Phones, ETERNITY allows you to
selectively personalize the key maps of these two Extended IP Phones.
Phone Menu
You can access the following ETERNITY and phone features from the Menu of SPARSH VP248:
Menu option
Description
Call Logs
To view call history of internal and external Missed, Answered and Dialed calls.
You can also edit numbers in the call logs and store them in the Personal Directory.
Call Forward
To set and cancel Call Forward-Busy, Call-Forward No Reply, Call-ForwardUnconditional, and Follow Me.
Dynamic Lock
User Status
Keypad Lock
To lock the keypad of the phone. (when the keypad is locked, the features Call Log,
Contact, Call Forward, Dynamic Lock, User Status, DND, Call Cost Display, Hotline,
Alarm, Background Music, Change User Password will not be accessible.)
Do Not Disturb
To set/cancel Do Not Disturb on the phone, i.e. block incoming internal and external
calls.
1933
Menu option
Description
Hotline
Alarm
Change User
Password
To change User Password (required for using certain features like Call Follow Me,
Dynamic Lock, DISA, Walk-In Class of Service, User Absent/Present, Hot Desk) and for
customizing Phone Settings.
Phone Settings
To customize settings of the phone such as Speech and Ringer Controls, LCD Display
settings (Brightness and Contrast, Backlight ON/OFF), Headset Connectivity, Call
Answering Mode (manual/auto answer).
To exit menu,
Go ON-Hook.
Type of Terminal
SPARSH VP248
1934
Type of Terminal
SPARSH VP248
Type of Terminal
SPARSH VP248
1935
Key Features
1936
Capacitive Touch Screen: 4.3 inch Capacitive Touch Screen LCD (Liquid Crystal Display) that delivers
easy access to advanced features and a unique experience beyond traditional desk phones. Supports
adjustable Brightness controls from the touch screen to suit your customized LCD requirement.
Enhanced Call Management: Dedicated one-touch feature keys and intuitive user interface provides
quick access to full range of PBX call management features including Call Hold, Call Park, Call Transfer,
Conference and Voicemail.
Easy-to-Use Navigation Cluster and Hard Keys: Four-way navigation cluster provides choice and pace
at work by allowing easy scrolling for features and function selection. Supports following hard keys and
LEDs 12 Alphanumeric Digit Keys (Dial pad Keys), 4 Navigation Keys and 2 Control Keys.
1 Speaker Key.
14 Programmable Feature Keys/ DSS (Direct Station Selection) keys.
1 Ringer LED.
Improved Audio: High Definition (HD) Audio output that delivers crystal clear voice and life like
conversations over HD handset and hands-free speaker.
Access to Corporate Directory: Easy integration with the enterprise's Corporate Directory (Global
Directory) which allows you to easily locate and dial corporate contacts at one click.
Easy Access to Voice Mails: On-screen voice mail icon and dedicated feature key to access Corporate
Voice Mailbox ensures that you do not miss a single opportunity and stay connected to the business.
Presence: Provides intuitive Presence status display and supports changing your Presence status which
is viewable to other extension users.
Full-duplex Hands-free Speaker: High quality speaker with acoustic echo cancellation to deliver natural
and clear speech even for hands free operation without any distortion.
Plug & Play: Integrated Plug & Play feature that enables to power up the phone and start using it. On the
other hand, it helps in mass deployment of the phones in your organization without requiring a lot of
manual intervention to configure each of the phones separately.
High Speed Ethernet connectivity: Dual switched 10/100 Base-T auto-sensing Ethernet LAN
connectivity allows unconstrained bandwidth from the network to the phone. LAN port is used to connect
the phone to a Switch or a Hub or a Router or an xDSL Modem. You can connect your PC to the PC port
of the phone.
Power over Ethernet (PoE) : Integrated IEEE 802.3f Power over Ethernet allows easy deployment with
centralized powering without a need for external power adapter.
Wi-Fi Support: Supports Wi-Fi (WLAN) connectivity using which the phone provides seamless
connectivity to the enterprise Wi-Fi network and offers flexibility to work from anywhere in the office. If your
installation setup does not meet the requirements of suitable wired Ethernet connectivity due to any
reason, then you can connect the compatible Wi-Fi adapter supplied by Matrix to the Wi-Fi USB port to
register and use the phone.
Multiple Language Support: The phone can be operated in six different languages including English,
French, German, Spanish, Portuguese and Italian.
Front View
Ringer LED
LCD Screen
Navigation Keys
10
Speaker Key
11
12
Handset
1937
Bottom View
Wi-Fi Port
PC Port
LAN Port
Power Port
1938
Feature Assigned
SLT 001
SLT 002
DKP 001
DKP 002
CA 01
CA 02
Here, SLT = Single Line Telephone, DKP = Matrix Proprietary Digital Key Phone, CA = Call Appearance.
1939
Keys numbered from 7 to 14 (as shown in the image above) have characteristic feature icons imprinted on them.
Default features assigned to these keys are as follows.
DSS Key
Feature Assigned
Transfer
Hold
Contacts
10
Redial
11
Voicemail
12
Call Forward
13
Conference
14
Mute
1940
Make sure the phone in which you install Matrix SPARSH MS, runs on Android V2.2 or later.
Key Features
PBX Extension: Matrix SPARSH MS becomes a mobile extension of the ETERNITY. As an increased
number of business professionals are using the collaborative tools found in smartphone devices to aid in
their work activities, this application offers businesses an easy way to integrate their enterprises' voice
solutions within the Android OS family.
Advanced Call Capabilities: Access to features such as Callback, Dial-in Conference, Conversation
Recording and many more.
Mobility: Matrix SPARSH MS provides you the mobility that you need in today's highly competitive
business environment; with the ability to access ETERNITY features easily once you are connected to
either Wi-Fi or 3G network. Considering the case of roaming users, one can register Matrix SPARSH MS
with the ETERNITY using the enterprise Wi-Fi network when working within the office (that is within the
organization's dedicated Wi-Fi coverage area). While working out of the office (where Wi-Fi network may
not be available), one can register the application using the Mobile Data (3G) network.
Dial by Extension: Flexibility to reach to office users with direct extension number dialing.
Presence: Supports changing your Presence status viewable to other extension users.
Corporate Directory Access: Enhance business collaboration with one-touch access to the Corporate
Directory contacts using ETERNITY's Global Directory.
Voicemail Access: Access to the corporate Voicemail system from any location ensures no opportunity is
lost.
Multiple Call Support: With multiple call support, you can easily handle multiple incoming calls, merge
and split calls apart, and place users on hold with a simple tap. With this Android application, it's like taking
your deskphone on the road.
Single Number Reach: Retains the identity of the corporate phone system while working away from the
office; so enhances business collaboration and lowers communication delays.
Better Voice Quality: Using customized codec settings, enhanced voice output is available. If you are
aware of the bandwidth and the network criteria of your location, you can select the proper codec from the
application to get high quality voice output.
Standard Telephone Features: Provides intuitive access to Keypad, Contacts, Call Logs and more,
based on the native Android smartphone designs. One-touch access to call feature options during VoIP
(Voice over IP) calls including Adding a New Call, Mute, Hold, Transfer and Speakerphone. Also provides
DTMF support to enter numbers using an auto attendant.
Cost Effective Calling: If you are using the enterprise Wi-Fi network to register Matrix SPARSH MS with
the ETERNITY; calls made from the application will be almost free. Even if you are using the application
via 3G network during roaming, external calls can be made using the ETERNITY trunks and thus reducing
mobile calling and roaming charges.
Multiple Language Support: The application can be viewed in six different languages including English,
French, German, Spanish, Portuguese and Italian.
1941
Application diagnostics: Supports logging and sending of log files to concerned recipients by e-mail.
These logs are used by the system engineer and/ or Matrix support engineers for troubleshooting .
Installing SPARSH MS
For detailed instruction to install SPARSH MS, refer to the Matrix SPARSH MS Android Application User Guide.
Configuring SPARSH MS
For detailed instructions on how to configure SPARSH MS, see Configuring Matrix SPARSH MS Android/iPhone
Application.
Operating SPARSH MS
Refer to Matrix SPARSH MS Android Application User Guide for instructions on operating the features of
ETERNITY.
Key Features
PBX Extension: Matrix SPARSH MS becomes a mobile extension of the ETERNITY. As an increased
number of business professionals are using the collaborative tools found in smartphone devices to aid in
their work activities, this application offers businesses an easy way to integrate their enterprises voice
solutions within the iOS family.
Advanced Call Capabilities: Access to features such as Callback, Dial-in Conference, Conversation
Recording and many more.
Mobility: Matrix SPARSH MS provides you the mobility that you need in today's highly competitive
business environment; with the ability to access ETERNITY features easily once you are connected to
either Wi-Fi or 3G network. Considering the case of roaming users, one can register Matrix SPARSH MS
with the ETERNITY using the enterprise Wi-Fi network when working within the office (that is within the
286.ETERNITY, the hybrid IP -PBX of Matrix which offers universal connectivity (since it supports multiple type of interfaces like PSTN,
ISDN, VoIP, GSM etc) and seamless mobility. Matrix SPARSH MS can work only as a registered SIP extension of the ETERNITY. All
necessary pre-configurations required to register your phone with the ETERNITY must be performed by the system administrator. For
more details about the ETERNITY, refer to the ETERNITY System Manual. To access the manual, contact your system administrator.
1942
organization's dedicated Wi-Fi coverage area). While working out of the office (where Wi-Fi network may
not be available), one can register the application using the Mobile Data (3G) network.
Dial by Extension: Flexibility to reach to office users with direct extension number dialing.
Presence: Supports changing your Presence status viewable to other extension users.
Corporate Directory Access: Enhance business collaboration with one-touch access to the Corporate
Directory contacts using ETERNITY's Global Directory.
Voicemail Access: Access to the corporate Voicemail system from any location ensures no opportunity is
lost.
Multiple Call Support: With multiple call support, you can easily handle multiple incoming calls, merge
and split calls apart, and place users on hold with a simple tap. With this iPhone application, its like taking
your deskphone on the road.
Single Number Reach: Retains the identity of the corporate phone system while working away from the
office; so enhances business collaboration and lowers communication delays.
Better Voice Quality: Using customized codec settings, enhanced voice output is available. If you are
aware of the bandwidth and the network criteria of your location, you can select the proper codec from the
application to get high quality voice output.
Standard Telephone Features: Provides intuitive access to Keypad, Contacts, Call Logs and more,
based on the native iPhone design. One-touch access to call feature options during VoIP (Voice over IP)
calls including Adding a New Call, Mute, Hold, Transfer and Speakerphone. Also provides DTMF support
to enter numbers using an auto attendant.
Cost Effective Calling: If you are using the enterprise Wi-Fi network to register Matrix SPARSH MS with
the ETERNITY; calls made from the application will be almost free. Even if you are using the application
via 3G network during roaming, external calls can be made using the ETERNITY trunks and thus reducing
mobile calling and roaming charges.
Multiple Language Support: The application can be viewed in six different languages including English,
French, German, Spanish, Portuguese and Italian.
Application diagnostics: Supports logging and sending of log files to concerned recipients by e-mail.
These logs are used by the system engineer and/ or Matrix support engineers for troubleshooting.
Installing SPARSH MS
For detailed instruction to install SPARSH MS, refer to the Matrix SPARSH MS iPhone Application User Guide.
Configuring SPARSH MS
For detailed instructions on how to configure SPARSH MS, see Configuring Matrix SPARSH MS Android/iPhone
Application.
Operating SPARSH MS
Refer to Matrix SPARSH MS iPhone Application User Guide for instructions on operating the features of
ETERNITY.
1943
External Music
Whats this?
ETERNITY provides a facility to play external music to extension users and external callers by connecting an
external music source to the Analog Input Port. For more details, please refer Music on Hold (MOH).
Relevant Topic:
1. Music on Hold (MOH)
1944
2057
Flash Timer
What is Flash?
Pulse dialing is a type of signaling in which codes (digits) are dialed in pulses. A hook switch or a Flash key is
generally used to dial this code. Technically, Flash is breaking the loop current for 200 milliseconds to 900 ms.
Please note that since this code is not simulated in standard DTMF convention, one cannot dial it in DTMF mode.
Flash timer signifies the time period for which the loop current breaks. Flash timer is programmable. Flash timer
ranges from 083 ms to 999 ms. By default, Flash Timer is 600 ms.
Where is it used?
Extensions dial flash to use few PBX features and also to use few PSTN features. Flash is used in following cases:
Now a days, more number of basic service providers and different types of advanced electronic telephone
exchanges are prevailing. It is possible that one service provider interprets breaking of loop current for 300 ms
as flash, other service provider interprets breaking of loop current for 900 ms as flash and system interprets
breaking of loop current for 600 ms as flash. Hence if the system engineer sets the flash timer to 600 ms then
he might not be able to use features provided by the service provider interpreting 900 ms for flash.
To take care of this situation, ETERNITY offers Flexibility to program different flash timers for both extensions and
trunks.
How to configure
Step 1
Please refer the topic SLT Hardware Template for more details on assigning flash timer to a SLT.
Step 2
Please refer the topic CO Hardware Template for more details on assigning Flash Timer to CO Trunks.
Step 3
Please refer E&M Feature Template for more details on assigning Flash Timer to an E&M.
Many times it happens that while transferring the call, the call either gets disconnected or is not
transferred. This happens due to mismatch of time for which the hook switch is pressed, if used for
transferring the call hence it is advisable to use Flash Key of the telephone instrument, instead of hook
switch.
This problem may occur with Flash Key also, if the timer for the Flash Key on the telephone instrument
and the flash timer of the system are not set properly.
1945
Few telephone instruments have flash timers set to 800 ms. In such case the call does not get
transferred because the flash timer of all extensions is set at 600 ms by default. In such cases the flash
timer of the extension where the phone is connected should be increased to 800 ms.
Relevant Topics:
1. SLT Hardware Template
717
2. CO Hardware Template
853
3. Digital Key Phone-Operation
1831
4. E&M Feature Template
895
1946
Flashing on Trunks
Whats this?
Trunk exchanges support many advanced features like call waiting, call forward. To use these features it is
required to dial codes during speech. The dialing of codes during speech do not create problem when you are
dialing on the trunk directly. But with the PBX connected between the user and the central office, the central office
codes clash with PBX codes. This leads to difficulty in accessing CO features while in speech. However,
ETERNITY supports dialing codes on trunk when in speech from any extension. But it is required to inform the
PBX, prior to dialing some code on the trunk.
How to use
For EON and Extended IP Phone Users
While in speech on trunk,
Press Transfer Key, dial * and the Desired Service Provider Code.
Or
Press DSS Key assigned to Flashing on Trunks (if programmed).
Dial the Desired Service Provider Code.
Press Flash.
Dial *
Dial the Desired Service Provider Code.
Example:
To use Call Waiting facility of service provider exchange from an SLT extension, perform following steps:
1
Dial Flash-*.
Dial Flash-1.
Dial Flash-*.
Dial Flash-1.
Dial Flash-*.
Dial Flash-1.
How to configure
Please refer Class of Service (COS) for details on how to allow Flashing on Trunk to a user.
Relevant Topics:
1. Class of Service (COS)
2. Flash Timer
1945
1720
1947
Flexible Numbers
Whats this?
ETERNITY offers Flexibility to assign a code of your choice to access an extension. This code is called
Flexible number. For example, to access first SLT having software port 001, one has to dial 2001. It is
possible to change this code to any other number of your choice.
ETERNITY offers the following types of extensions, namely SLT, DKP, ISDN Terminals, SIP Extensions,
Radio Extensions, Virtual Extensions and Department Groups. The system loads default access codes to
all extensions on first power ON. Later on the extensions can be assigned default Flexible numbers using
a command.
The Default Access Codes for the Extensions are given below:
Software Port
001 to 512
001 to 128
01 to 64
001 to 999
01 to 16
01 to 64
01 to 16
ETERNITY also allows you to assign a Flexible Number to each extension individually or to a range of
extensions simultaneously.
1948
It is possible to clear the flexible number of a extension, range of extension and all extensions.
Flexible numbers are the codes dialed from dial phase to call another extension. These flexible
numbers should be unique and should not match with either other SLT extensions or DKP extensions
or any of the features available from the dial phase.
Flexible number having common digits can be assigned to another extension. Please refer Conflict
Dialing for more details.
Use flexible numbers for all the features used from User mode and SA mode. Software port numbers
are to be used only from the SE mode.
When the access code of a extension is cleared; its flexible number becomes null or void.
If access code of a extension is cleared, one cannot call that extension. However the extension with
NULL flexible number can make calls as usual.
Extension Type: Select the Extension Type. You can select SLT ,DKP, SIP, ISDN Terminal, Magneto,
Radio, Department Group or Virtual Extension.
1949
Start S/W Port Number: Enter the Software Port Number from which you want the system to start
assigning the desired extension numbers.
Define the range of Station Access Codes/Extension Numbers (Flexible Number/Access Code) that
you wish to assign to the Extension Type you selected in Start Extension Number and End
Extension Number.
For the given range of extension numbers, the system will assign extension numbers from the software
port number specified in Start S/W Port Number for this particular entry. The range of extension
numbers will be assigned in ascending order of the Software Port Number.
For example:
Extension Typeyou selected is STL
Start Software Port Number is 1
Start Extension Number is 2001
End Extension Number is 2100
The system will assign Extension Number 2001 to Software Port Number 1, 2002 to Software Port
Number 2 and so on. The system will assign the last Extension Number 2100 to Software Sotware Port
Number 100.
Click Submit.
1950
Click the Extension Type, in this case, under Configuration click SLT Configuration.
Similarly you can assign the Access Codes (Flexible Numbers) to DKP, SIP, ISDN Terminal, Magneto,
Radio, Department Group or Virtual Extension.
Floor Service
What's this?
The Floor Service feature allows you to provide a common access code to extension users which they can dial to
call floor service.
Essentially a hospitality feature, Floor Service is also useful in offices. Floor service can be any administration or
service department in the building, such as a stationery room, back office, backroom, photocopy/ mail room,
secretarial assistance, concierge/janitor, Storeroom.
Just as all extension users can reach the Operator by dialing the common access code '9', they can reach the floor
service by dialing a common access code, '38'. This is the default Floor Service access code, for all geographical
regions where ETERNITY is installed.
This feature can be used in:
Multi-storied buildings, which have floor service (pantry, mail sorting, house keeping, janitor, coffee room,
refreshment area) for each floor. The ETERNITY can be programmed to land calls made by extension
users dialing the common access code '38' on the floor service extensions of their respective floors.
Offices that have a centralized floor service, instead of one on each floor. The ETERNITY can be
programmed to land calls made from all extension phones by dialing '38' on the common floor service
extensions.
This feature requires a license. To use this feature you must purchase the license for Hospitality. Refer the
topic License Management to know more.
How it works
For example, Midas Towers houses different departments on each floor. Each floor has Floor Service.
Extensions 2001 to 2010 are on the first floor, 2011 to 2020 on the second floor, and 3001 to 3010 on the third floor.
The floor service extensions are numbered as 2012 on the first floor, 2022 on the second floor and 3012 on the
third floor.
With the Floor Service programmed for each floor, when the extension user 2001 dials '38', the call will land on the
service extension 2012, assigned to room service on the first floor. Similarly, when the extension user 3008 on the
third floor dials '38', the call will land on the service extension 3012 on the third floor.
If Midas Towers had a single floor service extension 2012 for all floors, with Floor service programmed, calls made
from all extensions by dialing '38' would land on extension 2012 only.
How to configure
Programming the Floor Service feature involves the following steps:
1951
1. Creating a routing group for each floor. Include Floor service extensions of a floor in a routing group
Template. Prepare a different Station Advance Feature Template for each floor.
3. Applying the Station Advanced Feature Template (with the Floor service group programmed) to the
extension. This will assign the extensions to the routing group programmed in the Template.
If the Enterprise/Building has centralized floor service, you only need to create a single Routing Group with
service extensions, as required. This routing group number can be programmed on a common Station
Advanced Feature Template which will be applied to all extensions.
Floor Service parameters can be programmed using Jeeves and a Telephone.
Choose the Routing Group number (01-96) you want to use as floor service group.
You can program different routing groups for different floors. In each routing group you can program
maximum 32 service extensions as 'members'.
For routing group to be used as floor service, program the following parameters:
Rotation Flag: With this flag, you can enable or disable the rotation of calls in the routing group which
has multiple 'member' extensions. When enabled, each fresh call will land on the extension which is
next to the one that received the last call. This ensures equal distribution of incoming calls to all the
destinations within the routing group. The flag has no relevance if the routing group has only one
member extension.
Member Type: Select the 'Member Type'. You can select SLT, DKP, SIP, Virtual Extensions, ISDN
Terminal, OGTBG or the Voice Mail Auto Attendant.
Program only as many extensions as you want in the routing group and set the remaining Member
Types to 'None'.
1952
For example: if you want to program only one extension in the routing group, set the Member Type in
the remaining columns (Member 02-Member 32) to 'None.'
Port Number: Enter the software port number on which the SLT/DKP/SIP Extension/ISDN Terminal/
Virtual Extension floor service extension is attached. If you select Voice Mail Auto Attendant Profile,
enter the Profile number here. If you have selected OGTBG then enter the OGTBG number here.
Ring Timer(s): This timer defines the time for which the extension, on which the call lands, should ring.
By default, the ring timer is set to 015 seconds and can be changed.
Continuous Ring Flag: With this flag, you can set an extension to ring continuously until the call is
answered. The first extension will continue to ring even as the system hunts for other extensions in the
routing group to land the call. If the call still remains unanswered, the system will return the call to the
first extension once again. This flag is of no relevance, if there is only one member extension in a
routing group.
Repeat the above steps to include other floor service extensions in the routing group.
Select a Station Advance Feature Template number to be assigned to the user extensions of a floor. For
example: Template number 02 for extensions 2001 to 2010 are on the first floor, Template number 03 for
extensions, 2011 to 2020 on the second floor, and Template number 04 for extensions 3001 to 3010 on the
third floor.
Scroll with the horizontal bar to reach the column Floor Service of the selected Templates. Enter the
Routing Group number you want to use as floor service group for that particular Station Advance Feature
Template.
1953
Now, apply the Station Advanced Feature Templates (with floor service routing groups programmed) to the
extensions of the respective floors. For example: Template number 02 on extensions 2001 to 2010 are on
the first floor, Template number 03 on extensions, 2011 to 2020 on the second floor, and Template number
04 on extensions 3001 to 3010 on the third floor.
If extensions are SLT, assign the Template on the SLT Parameters page.
If extensions are DKP, assign the Template on the DKP Parameters page.
If extensions are ISDN Terminals, assign the Template on the ISDN Parameters page.
If extensions are Virtual Extensions, assign the Template on the Virtual Extensions page.
Refer the topic Station Advanced Feature Template for instructions.
Enter SE mode.
To program a routing group with member extensions, dial:
6502-1-Routing Group-Destination Index-Port Type-Port Number
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is from 01 to 32
Port Type is the 'Member type':
00 for None
01 for SLT
02 for DKP
28 for ISDN terminal
34 for SIP Extension
36 for Virtual Extension
Port Number is the Software port number287 on which the floor service member extension SLT, DKP,
ISDN Terminal is attached.
Software port number of the SLT, from 001 to 512.
Software port number of the DKP, from 001 to 128.
Software port number of the ISDN Terminal, from 01 to 64.
Software port number of the SIP Extension, from 001 to 999.
Software port number of the Virtual Extension, from 01 to 64.
To program the Ring Timer for the routing group, dial:
6503-1-Routing Group-Destination Index-Ring Timer
Where,
Routing Group is the number of the Routing Group 01 to 96
Destination Index is from 01 to 32
Ring Timer is from 000 to 255 seconds (default: 015 seconds).
To program the Continuous Ring Flag for the routing group, dial:
6504-1-Routing Group-Destination Index-Flag
287. Refer the topic 'Software Port and Hardware ID' in the ETERNITY System Manual.
1954
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is from 01 to 32
Continuous Ring Flag is
0 for disable continuous ring (each member extension in the group will ring for the programmed 'Ring
Timer' for the group)
1 for enable continuous ring (the first extension in the group will ring till the call is answered)
To program the routing group in a Station Advanced Feature Template, dial:
5602-1-Template Number-11-Routing Group
Where,
Template Number is from 01 to 50
11 is the feature code for Floor Service
Routing Group is from 01 to 96288
To apply the Station Advanced Feature Template now programmed with the Routing Group to SLTs,
DKPs, ISDN Terminals, SIP extensions, refer the topic Customizing Station Advanced Feature
Template using a Telephone.
Exit SE mode.
How to use
To be able to use floor service, extension users may dial the default Access Code defined for Floor Service: 38.
Check with your System Engineer if this access code has been changed and dial the new access code
obtained from the System Engineer.
Go OFF-Hook.
Press DSS Key assigned to Floor Service (if programmed)
OR
Lift handset
Dial, Access Code '38'
Talk
Replace Handset.
288. Enter the number of the routing group you programmed as Floor Service group.
1955
Follow Me
Whats this?
Using this feature, extension users can make your calls follow you wherever you go. Extension users can receive
their calls on another extension, whenever they want.
How it works
The extensions dial tone changes to feature tone if its calls are forwarded.
Follow Me can be overwritten. Extension A sets Follow-Me on extension B. After a period of time; goes
to extension C. A can receive calls on extension C by setting Follow Me on extension C. Follow Me set
by A on extension B will be cancelled.
Follow Me cannot be chained. If extension A sets Follow Me to extension B. And extension B sets
Follow Me on extension C, Follow Me of extension A is automatically cancelled.
Also see Call Forward, Class of Service (COS) and Do Not Disturb (DND)
How to configure
To be able to use Follow Me, extension users must have Call Forward feature enabled in their Class of Service for
the time zone. For instructions, see Class of Service (COS) and Station Basic Feature Template.
How to use
For EON & Extended IP Phone Users
To set Follow Me from another extension,
1956
1957
Forced Answer
Whats this?
Extension users can force other extension users to answer their calls when there is no response from the called
extensions.
How it works
Forced Answer can be requested by the calling extension. The calling extension may be an SLT, a DKP or an
Extended IP Phone. However, the called extension (being forced to answer) must be either a DKP or an Extended
IP Phone.
How to configure
To be able to use Forced Answer, extension users must have this feature enabled in their Class of Service for the
time zone. For instructions, see Class of Service (COS) and Station Basic Feature Template.
How to use
For EON & Extended IP Phone Users
To use forced answer on an extension:
Dial the desired extension number.
Press the DSS Key assigned to Forced Answer on Ring Back tone.
OR
Dial 5 on Ring Back Tone.
The Ring Back Tone stops.
The called extensions speaker is turned on.
You are in speech with the called extension.
You may talk.
1958
You can also dial 5, the feature code for Forced Answer, immediately after dialing the desired extension
number, instead of dialing it during the Ring Back Tone. This way, you can talk to the desired extension
user without waiting for the called extension user to answer your call.
1959
How it works
Forced Call Disconnection of an Extension:
A, B and C are extensions.
A and B are in speech.
C calls B and finds it busy.
C uses Forced Call Disconnection by dialing the feature command.
C gets confirmation tone, while A and B get error tone.
Forced Call Disconnection of a Trunk:
A and B are extensions. C is the external party.
A is in speech with C on Trunk 1.
B grabs Trunk 1 using Selective Port Access, but gets busy tone.
B uses Forced Call Disconnection by dialing the feature command.
B gets confirmation tone. A gets disconnected and gets error tone.
B must grab Trunk 1 again to get the dial tone of the network.
To be able to use Forced Call Disconnection, the extension user must have a higher Priority than the
extension user whom he/she tries to forcibly disconnect.
To be able to use Forced Call Disconnection on a busy trunk, the extension user must have grabbed that
trunk using Selective Port Access. If the extension user has grabbed the trunk using a Trunk Access
Code, the feature code to dial Forced Call Disconnection will not work.
In PLCC applications, Forced Call Disconnection can be used in a chain to reach the last Exchange
through many tandem exchanges in between.
.
You are advised to restrict access to this feature only to important extension users. Extension Users who
are allowed this feature are advised to use it judiciously.
How to configure
To be able to use Forced Call Disconnection, the extension must have:
1960
Forced Release feature enabled in the Class of Service. For instructions see Class of Service (COS)
and Station Basic Feature Template.
As Forced Call Disconnection on a busy trunk is possible only if the extension user has grabbed that trunk
using Selective Port Access, this feature must be enabled in the Class of Service of the extension. For
instructions see Class of Service (COS) and Station Basic Feature Template.
Higher Priority assigned than other extensions. See Priority for instructions.
How to use
For EON & Extended IP Phone Users
To forcibly disconnect a busy extension/trunk289:
Press the DSS Key assigned to Forced Call Disconnection on Busy tone.
OR
289. Only if you have grabbed this trunk using Selective Port Access.
1961
Gain Settings
What's this?
To avoid noise or echo during speech, you must set the speech volume levels on the ports. ETERNITY allows you
to set the speech volume levels for the following port typesCO, Mobile, SIP and SLT.
The speech volume levels can be adjusted by increasing or decreasing the Gain Settings provided on each port
type.
How it works
A call received on the CO port can be placed on any of the following portsDKP, SLT, Mobile, T1E1, SIP, BRI,
E&M. The speech volume levels differ according to the port type. Hence, on the CO port you must set the speech
volume levels for each of these port types.
In this case, let us assume that the call on the CO Trunk is to be placed on the SLT Port.
Before placing the call on the SLT Port, the system applies the CO to SLT Gain Setting (Receive and Transmit Gain
settings) configured on the CO Trunk to adjust the speech volume level.
When the call is placed on the SLT Port, the SLT to CO Gain Settings (Receive and Transmit Gain settings) on the
SLT Port are applied to adjust the speech volume level.
Hence, you can set different speech volume levels for each port type and the system automatically detects and
applies these gain settings for each port type.
How to configure
Select the CO Gain Settings Template Number you want to assign in the CO Hardware Template and
configure the following Transmit and Receive Gain Settings:
1962
CO - System (Tx-Gain and Rx-Gain): Configure the Gain Settings that you want the system to apply
on the CO port with respect to the system (for example DID, DISA, Auto Answer, Voice Message,
MOH, VMS etc.). These will not be applicable for any other port type. Valid Range for Tx gain: +10dB to
-15dB and Rx Gain +10dB to -15dB.
CO - SLT (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on the
CO port when the CO port is connected to any FXS Port of the system during an incoming or outgoing
call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
CO - CO (Tx-Gain and Rx-Gain): Configure the Gain setting that you want the system to apply on the
CO port when the CO port is connected to another CO Port of the system during an incoming or
outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
CO - DKP (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on
the CO port when the CO port is connected to any DKP Port of the system during an incoming or
outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
CO - SIP(Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on the
CO port when the CO port is connected to any SIP Trunk or SIP Extension of the system during an
incoming or outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
CO - Mobile (Tx-Gain and Rx-Gain):Configure the Gain Setting that you want the system to apply on
the CO port when the CO port is connected to any Mobile Port of the system during an incoming or
outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
CO - T1E1/BRI (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply
on the CO port when the CO port is connected to any T1E1/BRI Port of the system during an incoming
or outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
CO - E&M (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on
the CO port when the CO port is connected to any E&M Port of the system during an incoming or
outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
Similarly, to configure the Gain settings for the Mobile Port, click Mobile Gain Settings under Mobile
Configuration.
For SIP Trunk / Extension, click SIP Gain Settings under VoIP Configuration.
For SLT Port, click SLT Gain Settings under SLT Configuration.
Assign the CO Gain Settings Template you configured in the CO Hardware Template, see CO Hardware
Template.
Assign the Mobile Gain Settings Template you configured in the Mobile Port Parameters, see Mobile Port
Parameters.
Assign the SIP Gain Settings Template you configured in the SIP Hardware Template, see SIP Hardware
Template.
Assign the SLT Gain Settings Template you configured in the SLT Hardware Template, see SLT
Hardware Template.
1963
When ETERNITY acts as a Gateway, this feature is used to convey the call maturity information on source
port, when call made using destination port gets matured.
This information can be useful for billing equipment connected at calling party side.
How it works
This feature of answer signaling is applicable only for DISA option selected as 'CLI Authentication-one callAns. Sig.' in the DISA-CLI Authentication Table. Refer chapter Direct Inward System Access (DISA).
The ETERNITY works as gateway for routing the call to the destination.
If the calling party's number is programmed in Table - 'DISA-CLI Authentication', the ETERNITY will
consider the calling party as successfully logged in as station which is programmed as "Auto Login
station". The call will get answered by the ETERNITY. The caller will get dial tone.
When called party answers the call (that is, call on destination port gets matured), the Answer Signaling
will be done on source port, if enabled.
Answer Signaling will be done in form of DTMF digit string as programmed on the source port.
If Answer Signaling is disabled on the source port, the DTMF digit string will not be dialed out, even if it is
programmed.
On receiving these digits, Billing equipment/PBX with which the calling party is connected, can consider
the call is matured and start billing.
How to configure
To configure 'Gateway Application-Answer Signaling' flag and 'DTMF String' on desired trunk refer following
chapters:
1964
Application:
Mumbai
Mobile
GFX11
Delhi
FXS
PSTN
N/w
Mobile
DS1-1
ETERNITY
DS1-2
GFX11
FXS
PSTN
N/w
Kolkata
Mobile
GFX11
Chennai
FXS
Mobile
GFX11
FXS
GFX11s are configured for multi stage dialing, in which when ever caller dials the number using pay
phone, GFX11 will store the number, it will first make a call to ETERNITY's T1E1 line on which DISA - 'CLI
Auth- One Call - Ans. Sig.' is enabled.
GFX11's SIM Numbers are programmed in 'DISA - CLI Auth.' table in ETERNITY.
ETERNITY compares the Calling Number received on T1E1 Port with DISA-CLI Auth. Table.
As the number is programmed in the table, ETERNITY answers the call and offers the trunk assigned in
Station Basic Feature Template of the station used as 'Auto Login'.
When ETERNITY answers the call, the GFX11 sends the stored called number (which is actually dialed by
the caller) in DTMF digits.
ETERNITY routes the call on this number using the offered trunk.
When called party answers the call, ETERNITY will send 'DTMF Digit Strings' programmed on the T1E1
Port (Source Port, on which call originated) as a Gateway Answer Signaling'.
The Pay Phone connected with FXS Port of the GFX11 is also configured to understand the same DTMF
string as call maturity.
Pay Phone will start billing only on receipt of the desired DTMF digits.
1965
Remarks:
Source Port: Port of the ETERNITY on which call originates.
Destination Port: Port of the ETERNITY on which call terminates.
Relevant Topics:
1. Direct Inward System Access (DISA)
2. CO Hardware Template
853
3. E&M Feature Template
895
4. Configuring Mobile Trunks
1086
5. Configuring PRI Trunks
1029
6. Configuring BRI Trunks
924
7. Configuring SIP Trunks
1140
1966
1862
GPAX Application
Whats this?
GPAX application is one of the applications provided by ETERNITY and used in commercial establishment/Society/
Organization, etc. Group PBX are installed, operated and maintained by organizations/agencies. The owner will be
treated as the main hirer. The private exchange (PSTN)/service provider will provide junctions to the PBX and the
owner will pay the rental of the junctions, and the call charges. The number of junctions provided to the PBX should
be adequate to carry the traffic. The owner/agencies of the PBXs will be responsible for the payment of all charges
to the private exchange/service provider.
In this application say a PBX A is connected to PSTN. PBX A is given an exchange ID (say 2837). A number of
stations (say 001-100) can be connected to PBX A. When a station 015 dials 2837025, PBX A interpret this
number to be dialed for the same system. However, for dialing a station number belonging to same system, it is not
necessary to prefix the station number with exchange ID. When a station 020 dials 2834537, PBX A does not
interpret this to be dialed for the same system and hence will dial the digits on the trunk.
When a station user picks up the handset and dials any digit except the one programmed in the routing table will be
dialed on the trunk. If the user dials digit that is programmed in the routing table with Self-flag enabled, the system
will not dial the digits on the trunk since it would interpret these to be dialed for the same system.
For dialing the digits on the trunk it is required to program the routing table carefully. Route code should be
specified in route code column of routing table in association with Self Route Flag disable. This will make the call to
be routed on trunk which is specified in OG Trunk Bundle Group. Regarding this please refer Closed User Group
(CUG) and Closed User Group-With Exchange ID.
T1
PBX A
T2
2837
PSTN
Tn
S1 S2
001
002
Sn
100
In order to make internal calls, user is advised to program Routing Table such that one of the entries in routing table
should have route code as #, Strip digit count as 1 and Self route flag as 1 (enable). Doing so, when user picks
up handset and dials the required number with prefix #, the system interpret this number as an activity for the
internal users and waits for relevant access code.
Because of Strip digit count=1, the first digit dialed by the user (that is, # in this case) will be ignored and next digit
will be processed which could be a feature code or a station number.
1967
In order to use system features, the feature codes should be prefixed by #. Whenever the user dials # on picking
the handset, the PBX assumes it to be an activity for the internal users and waits for relevant access code.
In GPAX application, Answer Signaling on SLT and Answer Supervision on trunk shall e programmed properly. Also
the Disconnect Supervision parameter shall be programmed as appropriate for proper billing of calls.
Following figures shows how call is established between users using PSTN and ETERNITY-GPAX.
PSTN
Customer Site
ETERNITY-GPAX
Makarpura
Alkapuri
Subscriber
Card
FXO
(Trunk)
Subscriber
Card
A
Inter Exchange
Trunk interface
card
FXS
(SLT)
Inter Exchange
Trunk interface
card
Inter
Exchange
Trunk card
PSTN
Customer Site
ETERNITY-GPAX
Trunk
Subscriber
Card
SLT
Card
Inter
Exchange
Trunk
interface card
MFC
Operator
1968
Trunk Offer
PSTN
ETERNITY-GPAX
Trunk
Subscriber
Card
SLT
Card
Inter
Exchange
Trunk
interface
card
MFC
Operator
Relevant Topic:
1. Closed User Group (CUG)
1739
1969
GPAX Billing
Whats this?
In commercial establishments/large societies/organizations etc. staff make calls from their rooms. It is required that
the cost of these calls is calculated so that amount of calls made by a staff member can be paid. GPAX provides a
facility which if enabled can calculate cost of each call if programmed properly.
To calculate the total cost of a call please refer following topic for more details:
Call Cost Calculation (CCC), Call Duration Control (CDC), Call Budget, and GPAX charge Internal Calls in
Station Advanced Feature Template.
In order to make billing for internal calls between GPAX users, GPAX charge internal calls flag to be set to enable
(please refer Station Advanced Feature Template for more details).
If GPAX charge internal calls flag is enabled, this call will be recorded in the Station message detail recordingoutgoing buffer. If GPAX charge internal calls flag is set to disable, the call made to an internal station will not be
billed and will be recorded in the SMDR-Internal buffer as normal internal call.
1970
GPAX has a dedicated memory space (commonly called buffer) to store details of each call.
These calls are retained in the buffer even during power failure.
Various reports can be routed either on the printer or on the computer from this buffer.
Once the buffer is 100% full, the new call overwrites the oldest one.
It is recommended that printing of various reports should be regularised on fixed dates. This should be
done regardless of whether the buffer is full or not.
This will prevent spilling and subsequent loss of data. You can enable or disable call logging for individual
trunk.
This also prevents frequent spilling of the buffer, as new local calls will not be recorded.
In office/organization, it is required that the flexible numbers are given in such a manner that it identifies
the departments/blocks, that is, say subscribers for marketing department may start with 5, for technical
support with 6 and so on.
ETERNITY offers Flexibility to the user to assign a code of your choice to access a station. This code is
called Flexible number. For example, to access first SLT having software port 001, one has to dial 2001. It
is possible to change this code to any other number of your choice.
ETERNITY offers two types of stations viz. SLT and DKP. The system loads default access codes to all the
SLT and DKP stations on first power ON. Later on the stations can be assigned default Flexible numbers
using a command.
Default table of SLT station numbers looks like:
Software Port
Access Codes
001
2001
002
2002
003
2003
512
2512
Access Codes
001
3001
002
3002
003
3003
128
3128
How to configure
Please refer topic Flexible Numbers for more details.
1971
While using the ETERNITY in the GPAX application, SE should program different toll control levels as
per the user's requirement for making the calls which can be different than office user.
For example, SE can program the External number list for Allowed / Denied numbers for home user, as
per following Table:
Dynamic Toll Control Level
Relevant Topics:
1. Conflict Dialing
2. Access Codes
1440
3. Flexible Numbers
1948
4. Dynamic Lock
1901
1972
Help Desk
Whats this?
An organization may have a Centralised Information Office which provides information related to different
departments such as HR, IT, or General information. For each department in the organization, an extension number
can be defined as a Help Desk.
How it works
Extension 2002 is defined as Help Desk for HR policies and general rules.
Extension 2016 calls the Help Desk extension 2002.
If the Help Desk extension is busy, an Auto Callback request is set automatically on the Help Desk
extension.
As soon as the Help Desk extension is free, the system will serve the auto callback request.
The Help Desk extension calls back extension 2016.
How to configure
You can define an extension as Help Desk by enabling the Help Desk flag in its Station Advanced Feature
Template.
1973
Holiday Table
Whats this?
The Holiday Table feature of ETERNITY enables you to configure incoming call management for holidays. Using
the Holiday Table feature of ETERNITY you can,
define the landing destination for incoming calls on trunks on holidays.
greet callers with customized holiday messages.
determine the way extensions must work on the holidays.
You can configure a list of holidays in a single table.
How it works
In the Holiday Table, you need define the following:
The Start and the End Dates and the Time to be considered as Holiday.
The Time Zone to be considered for operating the time-zone based trunk and extension features290 during
the Date and Time configured as Holiday. The Time Zone for Holiday can be defined as Non-Working
Hours, Working Hours, or As per Time Table.
If you have Voice Mail System and are using the Voice Mail Auto Attendant as the landing destination for calls, on
holidays, you can play customized greeting messages to callers. For each holiday, you can play a different
message.
For example, A company, ABC Ltd., has the following requirements:
December 23 to December 31, all the employees will be on a holiday. The callers must be greeted with a
holiday message.
January 1 to January 4, few employees will be attending the office.
In this case you must define the following in the Holiday Table:
At Index 1,
In Start, enter the starting date and time of the holiday in DD-MMM-HH-MM format, that is 23 DEC, 00:00
and in End enter the last date and time of the holiday, that is 31 DEC, 23:59.
In Holiday Message select the customised holiday message number. The system will greet the callers
with this message.
At Index 2,
In Start, enter the starting date and time of the holiday in DD-MMM-HH-MM format, that is 1 JAN, 00:00
and in End enter the last date and time of the holiday, that is 4 JAN 23:59.
290. Trunk Landing Group, Auto-Attendant, DISA are time-zone based features of Trunks configured in the Trunk Feature Template
assigned to trunks. Class of Service, Toll Control, and OG Trunk Bundle Group, are time-zone based features of extensions that
are configured in the Station Basic Feature Template assigned to extensions.
1974
Index
Holiday
Start
(DD-MMM-HH-MM)
Name
Time
Zone for
Holiday
Holiday
Message
End
(DD-MMM-HH-MM)
23
DEC
00
00
31
DEC
23
59
Christmas
Holidays
NonWorking
Hours
31
DEC
00
00
04
JAN
23
59
Christmas
Holidays
As per
Time
Table
01
After you have defined the above parameters, this is how the feature Holiday Table works,
On the set date and time, when ETERNITY detects a day as a holiday, it checks the configured Time Zone
for Holiday and whether Holiday Message is configured.
Similarly, if the Time Zone for Holiday is defined as Working Hours, the system will operate the time-zone based
features of the trunks and extensions according to Working Hours.
Feature Interaction:
Day/Night Mode: You can set ETERNITY in Day/Night Mode, even when you have configured the Holiday Table.
In that case, the mode you select, Day (Working Hours) or Night (Non-working Hours) will override the Time Zone
you have selected for Holiday.
In order for the Time Zone for Holiday to come into effect, you must set the Day/Night Mode in System Parameters
to Operate System as per Timetable Assignment.
See Day Night Mode and System Parameters to know more about this feature.
1975
How to configure
You can configure the Holiday Table from the SE as well as the SA mode.
In Holiday configure the Start Date, Month and Time of the holiday and End Date, Month and Time of
the holiday.
Default, Start and End Date is Blank. Valid Range is from 01 to 31.
Start and End Month is Blank. Valid Range is from January to December.
Start and End Time is 00:00 (Hours:Minutes). Valid Range is from 00:00 to 23:59
1976
You can assign a Name to each time period you have defined as Holiday. For example, Christmas,
Independence Day, Thanksgiving. Default: Blank
As the Time Zone for Holiday select the time zone for the period you have defined as Holiday:
Working Hours, Non-working Hours or As Per Time Table. Default: As Per Time Table.
If the incoming calls are routed to the VMS Auto Attendant, select the Holiday Message number that
you want the system to play to the callers. Default: 01.
Click Submit to save changes.
You can also configure the Holiday Table from the SA mode also. To do this,
Follow the same steps as given above to configure the Holiday Table.
Ensure that Voice Mail Auto Attendant is selected as the Auto Attendant in the Trunk Feature Template.
For detailed information, see Trunk Feature Template.
Make sure you disable the parameter Directly Route to Root Node in the VMS Auto Attendant Profile you
assign to the Trunks and Extensions. For more details, see Voice Mail Auto Attendant Profile.
You can either use the default Holiday messages or customize your messages by recording messages of
your preference. To know more about the default Holiday Messages and how to record customized
Holiday Messages, see Recording Voice Messages.
1977
Hot Desking
Whats this?
Hot Desking enables extension users to use all the properties of their own extension from another extension.
Hot Desking is useful for people who are often away from their own desks and must work from another. Hot
Desking allows them to use all the features and facilities of their own extension from another.
How it works
This feature is supported on DKP and SLT extensions only.
Hot Desking is possible only between extensions of the same type: SLT to SLT and DKP to DKP extensions.
The User Password of both extensions involved in Hot Desking must not be 1111.
Hot Desking can be performed only when both the extensions are idle.
To perform Hot Desking two extensions are required:
The Host Extension - the extension whose user performs the Hot Desking.
The Hot Desk Extension - the extension on which Hot Desking is performed.
When Hot Desk is performed from the Hot Desking extension, all the properties of the Host Extension are
copied to the Hot Desk Extension.
On the Host Extension, the user cannot perform any activity except Cancel Hot Desking.
You must cancel Hot Desk from both the Hot Desk Extension and the Host Extension.
After cancelling Hot Desk, the Host Extension and the Hot Desk Extension acquire their original properties.
How to configure
For this feature to work, the feature 'Hot Desk' must be enabled in the Class of Service of the Host Extension and
the Hot Desk Extension. See Class of Service (COS) and Station Basic Feature Template for instructions.
How to use
For EON users
To perform Hot Desk:
Go to the EON extension (Hot Desk Extension) with which you want to swap your EON extension (Host
Extension) properties.
Press DSS Key assigned to Hot Desk.
OR
1978
Dial 1091
Enter Host Extension number
Enter Host Extension User Password
Go ON-Hook.
The User password of both the extensions involved cannot be default password.
Go to the SLT extension (Hot Desk Extension) with which you want to swap your SLT extension (Host
Extension) properties.
Lift the handset of the Hot Desk extension.
Dial 1091
Dial Host Extension number
Dial Host Extension User Password
Replace handset.
1979
Hotline
Whats this?
The Hotline feature connects the extension user immediately to a particular number or trunk, whenever the
extension user goes OFF-Hook.
You can set Hotline to connect immediately to another extension, to a Department Group, to an external number or
to an outgoing trunk.
Hotline set for external numbers and outgoing trunks is referred to as Hot Outward Dailing.
ETERNITY offers two types of Hotline/Hot Outward Dialing:
Immediate: As soon as the extension user goes Off-Hook, the user gets connected to the desired hotline
extension number, department group, external number, or outgoing trunk. For this the Hotline Timer must
be set to 00 seconds (default: 3 seconds).
Delayed: When the extension user goes OFF-Hook, the system plays Dial Tone to the extension user and
waits for the Hotline Timer (default: 3 seconds). On the expiry of this timer, it connects the extension user
to the desired hotline extension number, department group, external number or outgoing trunk.
How it works
Hotline/Hot Outward Dialing can be set from an SLT, DKP or Extended IP Phone extension, if the
extension has Hotline in its Class of Service.
To be able to use Hotline/Hot Outward Dialing, extension users must do the following:
Select the type of Hotline they want to set on their extension; whether to an internal Extension Number,
a Department Group, or an External Number or Outgoing Trunk.
Configure the Hotline Timer. For Immediate Hotline, extension users must set the Hotline Timer to 00
seconds. For Delayed Hotline, extension users can set the Timer as per their requirement.
1980
If A had set the Hotline Timer to 00 seconds (Immediate Hotline), A would be connected to B as soon
as A goes Off-Hook.
If A sets delayed Hot Outward Dialing for a Trunk or an External Number, the system will play dial tone
to A and wait for the duration of the Hotline Timer for A to dial digits. If A does not dial any digits within
this timer, the system connects A to the Trunk/External Number.
If A sets immediate Hot Outward Dialing (Hotline Timer set to 00 seconds), A will be connected to the
Trunk/External number as soon as A goes Off-Hook.
Delayed Hotline/Hot Outward Dialing allows extension users to dial out other numbers or grab another
trunk, without having to cancel the Hotline/Hot Outward Dialing they have set for a particular number or
trunk.
How to configure
To be able to use Hotline, extension users must have this feature enabled in their Class of Service (COS) for the
time zone, as required.
How to use
Hotline can be set/canceled by users for their own extension, or for any other extension from the SA mode.
Hotline when set/cancelled from the SA mode, will not depend on the assigned CoS.
Click Extension.
Now, Search Extension, by entering either extension number as Extension Number, or by entering the
name of the extension as Extension Name.
Click Submit.
1981
Select the type of Hotline you want to set for the extension user from the following:
To set Hotline for an Extension or Department Group, select the radio button Hotline to Station or
Department Group. Enter the Extension number or the Department Group Number in the
corresponding box. Default: Blank.
To set Hotline for a group of trunks, select the radio button Hot Outward Dialing to Group of Trunks
using TAC and select the Trunk Access Code from the corresponding drop down list.
Each Trunk Access Code has a group of trunks for which Hotline will be set.
To set Hotline for an external number, select the radio button Hot Outward Dialing to External
Number. Enter the external number in the corresponding box and in Using TAC select a TAC from the
drop down list. Using a free trunk from this TAC the external number will be dialled out by the system.
To set Delayed Hotline, in Hotline Timer enter the desired time in seconds.
On the expiry of this timer, it connects the extension user to the desired hotline extension number,
department group, external number or outgoing trunk.
1982
Meaning
Validity
Cancel Hotline
Not Applicable
Exit SA mode.
1983
OR
Dial 152-TAC
You cannot set Hotline and Hot Outward Dialing on the same extension at the same time.
The cancellation code must be dialed from the dial tone. You have to be very quick in dialing the
cancellation code, if the delay in the Hotline Timer is set to 1 or 2 seconds.
1984
When you set the Hotline Timer to 00 seconds (for immediate Hotline), you will not be able to dial any
digits, not even the feature code to Cancel Hotline.
If you have set Immediate Hot Outward Dialing for a Trunk or External Number, you will not be allowed to dial any
feature code, not even the feature code to cancel Hot Outward Dialing. However, if you need to cancel, you must
follow the steps described below.
Go OFF-Hook.
You get the CO network Dial Tone.
Dial by Digit.
You will hear Pause/Silence.
Press Flash.
You will hear the Feature Tone.
Dial the code to change the Hot Outward Dialing Timer (154) and change the duration of the timer
or
Dial the access code to cancel the Hot Outward Dialing (150).
You get Confirmation Tone.
Go ON-hook.
You get the return ring of the trunk.
Go OFF-Hook again.
You get connected to the held trunk.
Go ON-Hook.
1985
IC Reference Table
Whats this?
The IC Reference Table is a set of general features that define the logic of resolving an incoming call and placing
on the target DDI station. An IC Reference Table is assigned to every port. This table in conjunction with DDI
Routing Table identifies the target station. The ETERNITY offers 64 such table each of which can be programmed
as per the requirement.
How it works
The IC Reference Table consists of the following parameters:
IC Reference ID-This is the reference number acts as an identifier to the translation logic programmed in
the IC Reference Table. Any number of table can have the same reference number. An IC Reference ID is
assigned to ISDN and SIP trunks. When a call lands on the ISDN trunk, the system checks the IC
Reference Number assigned to it and identifies the corresponding DDI Routing Table for call resolving. For
more details on the complete Translation Logic please refer the topics Direct Dialing-In (DDI) and DDI
Routing Table.
Start Channel Number-This is the First Channel Number for the trunk to which the logic is applicable.
Total Channel Count-The Total number of channels of the trunk to which the IC Reference Table is
applicable.
DDI Routing Reference ID-This is the DDI Routing Table's Reference number used by the IC Reference
table for mapping the received DDI number to a flexible number. It is a link parameter.
Route on First Destination-This flag can be enabled or disabled. Once the station is identified, the
system checks the DDI IC routing flag of the station. If the flag is enabled, the call always lands on first
station of the MSN number. If the flag is disabled, the call is routed to the identified target station.
Ring Timer-This timer signifies the time for which the station on which the incoming call is received rings.
On expiry of this timer if the Call is not answered the call is routed as per the programmed logic. Route to
TLG-When No Reply-When the DDI station does not answer the call, the system checks for this flag. If it is
enabled, then the system routes the call to the TLG assigned to the Trunk. It is in seconds.
When No Reply-If the DDI station does not reply, the system checks parameter for processing the call
further. As per the option selected, the system does one of the following:
Disconnect the call
Route the call to Trunk Landing Group
Answer the call automatically, greet the caller with a voice message, and on completion of message
disconnect the call.
Answer the call, greet the caller with a voice message, and on completion of the message route the call
to Trunk Landing Group.
Route the call to Voice Mail; to the DDI stations Mail Box.
For this, a mail box must be assigned to the station. If the station is not assigned mail box, the caller will
hear the welcome message of the VMS, but will not be able to access the mail box.
1986
When Busy-If the DDI station cannot take the call because it is busy, the system checks this parameter for
processing the call further. As per the option selected, the system does one of the following:
Disconnect the call.
Route the call to Trunk Landing Group.
Answer the call automatically, greet the caller with a voice message, and on completion of message
disconnect the call.
Answer the call, greet the caller with a voice message, and on completion of the message route the call
to Trunk Landing Group.
Route the call to Voice Mail; to the DDI stations Mail Box.
Trunk Feature Template - A Trunk feature template is assigned to each DDI Routing Table. This enables
the user to allow Auto Answer Time Zone wise, allow Auto Attendant on a few numbers according to the
time zone. For more details refer the topic Trunk Feature Template.
How to configure
Configuring IC Reference Table using Jeeves
1987
Exit SE mode.
01
02
03
04
05
06
07
08
09
Feature
Name/Table
Index
IC
Ref.
ID
Start
Channel
No.
Total
Channel
Count
DDI
Routing
Ref. ID
Route on
First Dest.
DDI Ring
Timer
(Sec.)
When
No-Reply
When
Busy
Trunk
Feature
Template
01
00
01
00
00
Disable
045
Disconnect
Disconnect
01
02
00
01
00
00
Disable
045
Disconnect
Disconnect
01
64
00
01
00
00
Disable
045
Disconnect
Disconnect
01
01-30
00-30
00-99
Disable
001-255
Disconnect
Disconnect
01-50
Route to
TLG
Route to
TLG
Greet and
Disconnect
Greet and
Disconnect
Greet and
Route to
TLG
Greet and
Route to
TLG
Route to
Voice Mail
Route to
Voice Mail
Parameters Value:
Code
0
1
1988
0099
Enable
How it works
Incoming CLI Modification parameters must be programmed in the system considering the dialing pattern
supported by the local public network.
Accordingly, ETERNITY matches the CLI received with the programmed parameters.
It detects whether it is an international, national or local number.
It modifies the CLI according as per the Modification parameters programmed.
It presents the modified CLI to the extension; stores the modified CLI in the SMDR and in the Call Logs of
the extension, provided it is a digital key phone.
When the received CLI is dialed out by the extension user from Call Log, ETERNITY dials out the same
number.
How to configure
For this feature to work, the parameter Incoming CLI Modification must be programmed in the System
Parameters of ETERNITY. This can be done from Jeeves or by dialing SE commands from a Telephone.
1989
Enable Incoming CLI Modification: Enable this flag if you want to use the Incoming CLI Modification
feature. By default, this flag is disabled.
If you receive CLI in dialable format, there is no need to use this feature. In such case, keep the flag
disabled. You do not need to program any of the CLI modification parameters.
1990
Country Code: Enter the Country Code of the country where ETERNITY is installed. The Country Code
helps ETERNITY detect whether the Incoming CLI received is a national or an international number. Do
not enter any prefix for the Country Code. For example, if your ETERNITY is installed in USA, enter only 1
as the Country Code. Do not enter + or 00 as prefix to the country code 1. By default the Country Code
is 91 (India).
Area Code: Enter the Area Code of the place where the ETERNITY is installed. The Area Code helps
ETERNITY detect whether the Incoming CLI received is a local number. Do not enter any prefix for the
Area Code. For example, if you want to enter Area Code for Mumbai, enter only 22. Do not enter the
prefix 0 to the area code. By default, Area Code is 265 (Vadodara city).
International Prefix: Enter the digits that are required as Prefix for dialing International Numbers. The
prefix may be up to 5 digits, with numbers from 00000 to 99999. By default, 00 is set as the prefix for
dialing International numbers.
National Prefix: Enter the digits that are required as Prefix for dialing long distance, National (within the
country) numbers. The prefix may be up to 5 digits, with numbers from 00000 to 99999. By default, 0 is
set as prefix for dialing national numbers.
Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose from the following options:
No (Area Code not required): select this option if your public telephone network does not require the
dialing of Area Code for local numbers.
Yes (Area Code is required): select this option if your public telephone network requires you to dial
the Area Code for local numbers.
Yes (Area Code with Prefix required): select this option if you public telephone network requires you
to dial Area Code with a particular Prefix for local numbers. If you select this option, you must also
program the Prefix digits for the Area Code.
By default, the option, No (Area Code not required) is selected.
Prefix Area Code: If you have enabled the Area Code required to make local calls flag in the previous
parameter, enter the prefix digits for the area code for local calls in this field.
1991
Default: 00
To program National Call Prefix, dial:
5371-National Call Prefix
Where,
National Call Prefix is a number string up to 5 digits. The number string may consist of numbers from
00000 and 99999.
Default: 0
To enable/disable Area Code required to make local calls?, dial:
5372-Code
Where,
Code is
1 for No (Area Code is not required)
2 for Yes (Area Code is required)
3 for Yes (Area Code with Prefix required)
Default: 1
To program Prefix Digits to Area Code for Local Calls, dial:
5373-Prefix Digits
Where,
Prefix Digits to Area Code is a number string up to 5 digits. The number string may consist of numbers
from 00000 and 99999.
Default: Blank
1992
Exit SE mode.
Intercom
What is this?
The Intercom feature of ETERNITY enables extension users to connect quickly with any desired extension, without
waiting for the called extension to answer.
On SIP extensions, ETERNITY supports Intercom using Call-INFO / Alert-INFO Message. For a list of
IP phones on which this feature has been tested, see ETERNITY Features tested on IP Phones of
different Brands in the Appendix.
How it works
As extension number is 3001 with Priority Level 7 and Intercom feature enabled in the Class of Service.
A wants to quickly connect to B. A dials the Intercom feature code *5 followed by Bs number, 3003.
Bs extension is idle at the time of the call, and the speaker of Bs phone goes OFF-Hook, creating a
speech path between A and B.
Feature Interactions
Do Not Disturb (DND): If the called extension has set DND, ETERNITY will not place the intercom call on
the called extension.
Privacy from DND Override: If DND as well as Privacy from DND Override is enabled in the Class of
Service of the called extension, ETERNITY will reject the Intercom call.
Call Forward-Unconditional: If the called extension has set Call Forward-Unconditional, ETERNITY will
forward the intercom call to the forwarded destination number. The call placed on the forwarded
destination will not be an Intercom call.
Call Forward-No-Reply: If the called extension has set Call Forward-No-Reply, ETERNITY will not
forward the intercom call to the forwarded destination number on the expiry of the No-Reply Timer.
1993
Call Forward-Busy: If the called extension has set Call Forward-Busy, ETERNITY will place the call on
the forwarded destination number. However, this call (placed on the forwarded destination) will not be an
Intercom call.
User Absent/Present: ETERNITY will place the Intercom call on the called extension only if the status of
the called extension is Present.
Auto Call Back: When the Intercom call is generated and the called extension is busy, the calling
extension can set Auto Call Back on the called extension. When the called extension is free, ETERNITY
will serve the Auto Call Back request set by the calling extension. The ACB call placed on the called
extension will be a normal call.
Priority: The calling extension must have a higher Priority level than the called extension.
When the Intercom call is generated on SIP Extension having multiple call appearance and already a call
is present on the SIP Extension then the Eternity will place the Intercom call as normal call on the SIP
Extension.
How to configure
To provide this feature to extension users, you must enable this feature in their Class of Service. For instructions,
see Class of Service (COS) and Station Basic Feature Template.
How to use
For EON & Extended IP Phone Users
To use intercom to call an extension:
1994
Using this feature, the operator will be able to allow/restrict internal calls. The operator has flexibility to
allow calls during day time and restrict calls during night time. In ETERNITY this feature is implemented by
following ways:
Privacy of the user is very important in Enterprises and hospitals. In general, most of the users need to
communicate only with reception, operator, pantry and such other service stations. He is not required to
dial other user numbers. For example, in hospitals the patients need to call the nurse and not the doctor or
other patients.
Sometimes, a group of users occupy multiple rooms in the Enterprise. In such cases members of the
group would like to communicate only among themselves and service stations.
Certain service stations should be able to dial any other service station or any guest (for example Nurses,
Operator).
Certain stations should be able to dial only service stations (for example: patients, single user in the office).
A group of station users need to dial amongst each other and the service stations (for example: doctors).
Solo Group (for example patients, single user of the station). This is classified in group 00.
Universal Group (Nurses and other services staff for example). This is classified in group 99.
Friend's Group (for example, Group of visitors). Such a group of station users can be assigned to any
group number from 01 to 98.
Universal Group
99
Friends Group
00
Friend Group
A
Friend Group
B
Universal Group
99
Friends Group
00
Friend Group
A
Friend Group
B
1995
How to use
From SA mode:
Dial, 1072-904-Flexible Number-Guest Group
OR
Press DSS key for 'Guest Group'
Where,
Flexible Number is the code given to access a station.
Guest Group is from 00 to 99.
By default, the guest group assigned is 99.
The user can call any body in his 'guest group' number '99'.
The station users in one 'guest group', '99' can call each other within the group as well as the station users
in other groups.
How to use
From SA mode:
Dial, 1072-045-Code
OR
Press DSS key for 'Enable/Disable Internal Call Block'.
Where,
Code
1996
Meaning
When this feature is enabled, the LED on the DSS key assigned to 'Call Block' will glow Red and when
disabled, the LED will be turned Off.
When the operator issues "Call Block- Enable command, all the station users will be assigned guest
group = 00.
When the operator issues "Call Block- Disable command, all the station users will be assigned guest
group = 99.
Even after issuing SA command for 'Call Block', the operator can use the SA command 1072-904 to
change the guest group of any station user.
Please refer separate manual for more details about Hotel Applications for this feature.
How it works
C calls A.
C gets Ring Back tone (RBT) and A gets beeps indicating a new call.
To answer Cs call, A must dial Flash before the expiry of the Interrupt Request Timer. A will be in speech
with C. B will be put on hold and will get music on hold.
If A does not dial Flash before expiry of the Interrupt Request Timer, Cs call will be disconnected.
After the conversation between C and A is over, when C goes on-hook, speech between B and A will be
re-established.
Feature Interactions
Call States:
Interrupt Request works only if the dialed extension is busy. The dialed extension may be busy with
another extension or trunk (external number).
Interrupt Request works only if the user about to be interrupted in is in a two-way normal speech with
another user or external party.
It will not work if the busy signal is due to the user being Off-hook, or in the middle of dialing, or
accessing a feature of the ETERNITY.
Call Toggle: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
1997
Privacy against Interrupt Request: If the feature 'Privacy against Interrupt Request is enabled for an
extension, it cannot be interrupted. See Privacy.
Priority: No Interaction with Interrupt Request. If 'A' has lower priority than 'B' but has Interrupt Request
enabled; A can interrupt B.
How to configure
To be able to use Interrupt Request, extension users must have this feature enabled in their Class of Service
(COS) in their Station Basic Feature Template.
For instructions on configuring the different extension port types:
Configuring SLT Extensions
Configuring DKP Extensions
Configuring ISDN Terminals
Configuring SIP Extensions
If required you may also change the default value of the Interrupt Request Timer. For instructions see System
Timers and Counts.
How to use
For EON & Extended IP Phone Users
When dialed extension is busy,
1998
How it works
When the called extension answers, speech is established between A and the called extension user.
On SIP extensions, ETERNITY supports Last Caller Recall using text (lcr) as access code.For a list of IP
phones on which this feature has been tested, see ETERNITY Features tested on IP Phones of different
Brandsin the Appendix.
How to use
For EON & Extended IP Phone Users
1999
How it works
If Extension A is an SLT, the system dials the last external number dialed from Extension A using the same
trunk access code used for dialing that number.
If Extension A is a DKP or an Extended IP Phone, all external numbers dialed by Extension A are
displayed on the phones LCD.
Extension A may select the number to be dialed out. The system will dial out this number using the same
trunk access code used for dialing this number.
If Extension A has Dynamic Lock set and uses Redial feature, the system will check for Toll Control as
per the Lock Level set for Extension A before dialing out the number.
How to configure
No particular configuration is required for this feature to work. Redial is included in the Basic Features allowed to all
extensions by default in their Class of Service (COS). So, all extensions can use the Redial feature.
How to use
For EON & Extended IP Phone Users
2000
Lift Handset
Dial 7
The system dials out the external number last dialed from your extension.
Subscriber
PBX
Service
Provider A
Service
Provider C
Service
Provider D
A subscriber of Service Provider A must grab trunk lines of Service Provider A to call other subscribers in the local
area.
However, when the subscriber of Service Provider A wants to make a long distance call, the subscriber must dial a
prefix to select the a carrier (trunk) of the desired long distance, Service Provider B, C and D. Thus, the subscriber
accesses a secondary service provider by dialing a short code or prefix for long distance calling.
This feature works on the basis of Automatic Number Translation. Using Automatic Number Translation,
ETERNITY adds the code of the appropriate secondary Service Provider to the number string dialed by the
extension user to route the call to the desired secondary Service Provider.
How to configure
To use this feature, you must do the following:
Configure the Automatic Number Translation Table. In the Automatic Number Translation Table, in the
Dialed Number String column, enter the long distance numbers that extension users will dial. In the Add
Prefix column, enter the digits which are to be added as prefix to the Dialed Number string by the system
before dialing it out and in the Strip Digits column enter the number of digit(s) to be stripped off by the
system from the Dialed Number string before dialing it out. For example, the code 961 for Service
Provider B must be prefixed to the number 2630555 dialed by extension users, you must enter 2630555
in the Dialed Number String column, 961 in the Add Prefix column and 0 in the Strip Digits column.
2001
2002
Create OG Trunk Bundle Group that includes the OG Trunk Bundle you created with Automatic Number
Translation as member.
Assign the OG Trunk Bundle Group (containing the OG Trunk Bundle with ANT) to the extensions for the
Time Zones, that is, Working Hours, Non-Working Hours, Break Hours, in the Station Basic Feature
Template of the extensions.
License Management
What's this?
Certain features of ETERNITY require the purchase of a license. When you buy ETERNITY, you get a unique
license number for your product and a set of 'license-free' features are loaded in the system. Described below are
the features that require license.
PLCC
This functional module contains a set of special features supported by the ETERNITY when it is deployed in a
Power Line Carrier Communication Network of electric utilities. When you buy the license for this module, the
following features will be enabled:
Express Signaling.
Hospitality
This functional module contains a set of special telephone and guest/patient management features for hospitality
and accommodation establishments like hotels and hospitals, which ETERNITY supports when it is deployed in a
hotel or hospital. When you buy the license for this module, the following features will be activated:
Room Shift
Check-In, Check-Out
Floor Service
Q-Sig
When ETERNITY is networked with another ETERNITY or with any other ISDN-PBX, Q-Sig or Q-Signaling is
supported to facilitate feature transparency between the PBXs in the network. Q-Signaling will be activated when
you buy a license.
2003
Dealer Board
If you want to connect and use EON74, the proprietary Digital Turret for ETERNITY, you need to buy a Dealer
Board license.
Gateway
When ETERNITY is used as a Universal Gateway, a license is required to activate the Gateway functionality.
SIP Extensions
ETERNITY supports up to 999 SIP Extensions, depending on your model. With a license, you can register SIPenabled devices with the VoIP Card of the ETERNITY. Without a license you cannot register any SIP Extension.
Multi-Party Conference
SMDR Buffer (to increase capacity from 200 each of Incoming, Outgoing and Internal Calls to 5000
Incoming, 6000 Outgoing and 1000 Internal Calls)
SMDR Posting
Multi-stage Dialing
BCCH Selection
291. RCOC will work if you have any one license, the Business Suite or the Mobility Suite.
2004
Mobility Extension (DISA CLI Authentication, Call Forward-Scheduled, Call Forward-When Not Registered
- Scheduled)
SMS Gateway
With the SMS Gateway license, you can send/receive messages to/from individuals, selective groups or masses
using the Mobile Port of ETERNITY.
ETERNITY allows you to register multiple SMPP Clients (Software Applications used for sending/receiving
messages) with ETERNITY. ETERNITY functions as an SMPP Server. These Clients can send/receive messages
using the Mobile port/s of ETERNITY.
SMS Server
With the SMS Server license, you can:
Send/ receive SMS to/from individuals or groups using the Mobile Port of ETERNITY.
Forward SMS received on Mobile Port as Emails to users through the Email Client.
Forward Email of the users as SMS to the Mobile users through the Mobile Port.
The SMS Server application works as an intermediary between the GSM Short Message Service and the
ETERNITY. The Server supports multipart, 7 bit text messages as well as UNICODE messages.
The Server functions as an SMTP Client to send emails and as a POP3 Client to receive emails. SMS Server
supports three types of EmailsPlain Text, HTML and MIME from its mail clients.
Demo Provision
Demo provision for licensed features is useful: when the customers system having licensed features cannot be
repaired on-site and a standby system needs to be installed Or when end users demand to use certian licensed
features on trial basis before actually purchasing the license.
This feature is supported in Software Version V10R12 onwards.
Demo Provision of Licensed features enables users to use all licensed features of ETERNITY free of cost for a
period of 60 days.
To avail this facility,
Open Jeeves.
2005
Under Configuration, click License Management link. The License Management page opens.
Click the Demo Period Start button. The demo period for using licensed feature in your system starts and
the Start button will change to Pause. The demo period is of 60 days.
You may Pause the Demo Period, if required. When you pause the demo period, all licensed features will
work as per the license key installed in the system.
If you want to use licensed feature after the expiry of the demo period, you must purchase the license key
and activate it in your system.
When you default the ETERNTIY, the demo period will not reset.
A valid, unique User ID and Password from the Matrix License Support Centre.
Access to Internet.
2006
Open Jeeves.
Keep your Current License Key and the License Voucher (paper or PDF) ready.
2007
Enter your User Name and Password provided by Matrix and click the Login button.
2008
In the Current License Key field, type the current product license key you noted or paste the key you
copied from the License Management page of Jeeves.
2009
The page will show the current License Profile on ETERNITY. Click the Next button to continue.
The License Activation page opens.
In the License PIN field on this page, enter the 16-digit License PIN from the Voucher.
2010
Click Details. The details appear in the fields Product Family, Product Name, Product Variant.
Click the Next button. Your Current License Profile and your New License Profile will appear on this
page.
Click the Activate button and wait for a few seconds, as the activation is initiated.
2011
On successful activation, the confirmation message will appear on your screen along with the activation
date and time.
You will also be sent a confirmation mail to your e-mail ID (registered with Matrix).
You may Save, Print, or Email this information for your records, by clicking the relevant button.
2012
Note down or copy the New License Key generated on this page.
Go back to the Jeeves window (or log in as System Engineer again, if your session has ended).
In Enter License Key, paste or enter the new License Key generated.
2013
Open Jeeves.
Under Configuration, click License Management. The License Management page opens.
2014
Send your Current License Key and the License PIN (on the Voucher) to the Matrix License Support
Centre.
In Enter License Key, enter the New License Key you obtained from Matrix.
2015
How it works
To be able to use this feature,
the extension users who are to be provided this feature must have a Personal Mailbox assigned, and have
this feature allowed to them in their Class of Service.
the extension users who want to use this feature must set Call Forward to Voice Mail System on their
extension.
the extension on which this feature is used must be a DKP or an Extended IP Phone.
With the above pre-requisites fulfilled, this is how Live Call Screening will work:
B calls A, and is transferred to As mailbox (as Call Forward to Voice Mail System is set).
The VMS Auto Attendant offers B the option to leave a message in the mailbox of extension A.
As B starts to record a message in As mailbox, the speaker of the DKP/Extended IP Phone of A gets
turned ON for the duration of the Live Call Screening Timer (configurable; default: 10 seconds).
If A wants to answer the call, A can go Off-Hook. A gets connected to the caller and the system stops
recording the message in the mailbox.
If A does not answer the call, the speaker is turned Off automatically on the expiry of the Live Call
Screening Timer.
Also, after listening to some part of the message, if A finds that the call is not important, A can ignore the
caller by dialing any digit. When A dials any digit, the speaker of the DKP/Extended IP Phone is turned
OFF, but the message recording in the mailbox continues.
2016
How it configure
To be able to use Live Call Screening, extension users must have:
Live Call Screening enabled in the Class of Service (COS) for the time zone in their Station Basic
Feature Template, as required.
If required, you may also change the duration of the Live Call Screening Timer. See System Timers and Counts
for instructions.
How to use
For EON & Extended IP Phone Users only
To activate Live Call Screening,
2017
How it works
A is the supervisor of B.
When A requests Live Call Supervision for Bs extension, the system retrieves the last external number
dialed by B and presents it on the display of As phone.
If the last number dialed by B is an internal number, A will get error tone, as the system supports live call
supervision of external calls only.
Live Call Supervision can be used also when the extension being supervised is in speech with an external
party.
How to configure
To be able to use Live Call Supervision, extension users must have this feature enabled in their Class of Service
(COS) in their Station Basic Feature Template for the required time zones.
How to use
For EON & Extended IP Phone Users
2018
Logical Partition
What's this?
Logical Partitioning is used to restrict the flow of call traffic between PSTN and Private Networks as well as
between PSTN and VoIP networks.
This feature may be used in countries where such restrictions are mandated by telecom regulations. For example,
in certain countries, calls from VoIP to Public Networks (PSTN, Public Land Mobile Network) are not allowed.
Thus, local telecom regulations may either disallow termination of lines from both networks on the same equipment
or may allow lines from both networks to be terminated on the same equipment, provided the equipment is
designed to restrict flow of call traffic from these networks. For example, the Telecom Regulatory Authority of India
allows termination of lines the PSTN and VoIP Networks in the same equipment, only if these lines are logically
partitioned. Termination of lines from both these networks in the same equipment without a logical partition
constitutes an offence.
ETERNITY supports Logical Partitioning for this purpose.
How it works
Logical Partition is applied on the Trunk ports. Trunk ports are assigned to any of the following categories according
to their installation scenario:
Category 1: Trunk ports interfaced with PSTN /PLMN (Public Land Mobile Network) are assigned this
category.
Category 2: Leased lines terminated in the trunk ports are assigned this category.
Category 3: Trunk ports used to interconnect two PBXs are assigned this category. For example, QSIG
used on T1E1 port or the CO of ETERNITY is interfaced with FXS of other PBX for expanding
configuration.
These are default Categories. You have the flexibility to define each category according to suit your
preference. For example, you may, if you so prefer, define Leased lines to Category 3 and Trunk ports
connecting two PBXs to Category 1.
As calls from VoIP to Public network trunks are always be restricted, VoIP trunks cannot be assigned to
any of the three categories of the trunks as explained above and therefore do not require any
classification.
For each of the above categories, including VoIP trunks, the System Engineer can program the calling
permission, that is, whether to 'allow' or 'restrict' the calls across and within these categories and with the
VoIP trunks.
If you select India as the Region, by default, calls are restricted for all categories, except within VoIP, which
includes both SIP trunks and SIP extensions.
If you select any other Region, by default, the calls are allowed for all categories.
2019
You can change the settings as per the regulations in your country.
Depending on the calling permission programmed between the trunk categories, the system will allow or
deny the calls on the trunks.
How to configure
To able to use this feature you must first assign the trunk port - CO, ISDN BRI, T1E1PRI, Mobile, VoIP to the
appropriate Category and then define the calling permission - allow or restrict calls - in Category 1 to 3 and VoIP
trunks (as described above).
Now, assign the categories to the different Trunk Type, by programming the parameter 'Category (Logical
Partition)' in the Trunk Port Parameters of the respective trunk types: CO, BRI, T1E1, E&M and Mobile
Port.
2020
Open the desired Trunk Port Parameters page by clicking the link. For example, to assign a Category to
CO ports, open the CO Hardware Template page.
Scroll with the horizontal bar to reach the column 'Category (Logical Partition).
2021
Now, define the call permission for each category, that is, 'allow' or 'restrict' calls. For example, allow/
restrict calls from Category 1 to Category 1, from Category 1 to Category 2, from Category 1 to Category 3
and so forth.
First, program the parameter 'Category (Logical Partition)' in the Trunk Port Parameters of the respective
trunk types. For SE Commands, refer the topics:
CO Hardware Template
E&M Feature Template
Mobile Port Parameters
Configuring E1 Trunks and Configuring T1 Trunks
BRI Parameters in Configuring BRI Trunks
2022
Flag is
0 for restrict calls
1 for allow calls
Default: 0 (that is, restrict calls) for Category 1 to 3 and 1 (allow calls) for VoIP.
Exit SE mode.
By default for countries other than UK and USA calls within and between all categories of trunks are
restricted. In other words, trunk to trunk calls are not allowed.
In UK and USA the calls within all categories of trunks, including VoIP are allowed.
When call permission is restricted between two categories of trunks and/or between any category of
trunk to VoIP, following feature interactions will apply:
Call Transfer: Trunk to Trunk Transfer between restricted categories of trunk will not be allowed. If
the user attempts trunk-to-trunk transfer between restricted trunks, Error Tone will be played.
Raid: If a user using DISA attempts to Raid a conversation of an extension with a trunk to which call
permission is restricted, the Raid attempt will fail and the user will get an Error Tone.
Conference: An extension user will not be able to include restricted trunks in a 3 party or multiparty conference. An Error Tone will be played when s/he attempts it.
Dial-In Conference: Participation in a Dial-In Conference from trunks with restricted call permission
is not allowed.
External Call Forward: In the case of Auto Attendant, DISA or when transferring a trunk call to an
extension, if the extension has set call forward to an external number, the system will allow the call
only if the call permission between the source and destination trunk is allowed. Otherwise, an Error
Tone will be played to the user.
Hotline: When a user has logged into DISA and the extension being used for the DISA login has
the Hotline - Trunk or Hot outward dialing (HOD) feature enabled, the system will allow the call
between the source trunk (from where the DISA login is made) and the destination trunk (which is
used as Hotline Trunk) only if calling is permitted between them. Otherwise an Error Tone will be
played to the DISA caller on the expiry of the Hotline Timer.
2023
Far End Loop Back Test: These are of two type viz.
Line Loop Back test
Payload Loop Back test
These tests are conducted when the T1E1 port is in NT mode and is connected to other PBX and wants to
test the line between itself and the far end. In this mode, ETERNITY acts as a network.
2024
The protocol doesn't support the facility that the remote end can close/open the loop at the T1E1 port side
automatically.
Hence when the remote end wants to perform the loop back test, the SE must be informed, to form the
desired type of loop back (that is, Line Loop Back or Payload Loop Back) on the T1E1 port.
When the SE forms the loop back at T1E1 port side (by issuing appropriate SE Command), the remote
end can start the test.
On completion of the testing, the remote end must inform the SE to release/open the loop back formed on
the T1E1 port side.
On receipt of the request from the remote end, the SE will issue SE command to open the loops on the
T1E1 port.
The protocol supports loop back Activation and deactivation message, whereby the remote end can send
the loop activation code to the T1E1 port and the T1E1 port decodes the message and forms the loop back
automatically.
On completion of the testing, the remote end can send the loop deactivation code and the T1E1 port can
open the already formed loop back.
So in this case the SE's intervention is not required to form and release the loop back.
Incase the remote end doesn't support the facility to automatically form/release the loop back for the T1E1
port though the carrier is T1, the SE can use the commands (6141-1-T1E1-Type of Loop Back) on request
of the remote end to form and release the loop backs.
It will inform the ETERNITY ME Card T1E1 about the received command ETERNITY ME Card T1E1 will
release all the calls supported by the T1E1 Port under test.
The ETERNITY ME Card T1E1 will form the required type of loop back.
System will put the T1E1 port in maintenance mode. It will release all active calls supported by T1E1 port
and restrict the usage of T1E1 port for IC/OG calls.
When the loop back activation code is received from far end on the T1E1 port, (incase of T1 carrier):
ETERNITY ME Card T1E1 will inform the system about the received information and the T1E1 port
number.
ETERNITY ME Card T1E1 will release all the calls supported by the T1E1 Port under test.
System will put the T1E1 port in maintenance mode. It will release all active calls supported by T1E1 port
and restrict the usage of T1E1 port for IC/OG calls.
2025
The protocol doesn't support the facility that the ETERNITY can perform the loop back tests automatically.
The SE of the ETERNITY will inform the far end connected with the T1E1 port, to form the loop back as
required.
Once the far end has formed the desired loop back the SE can issue command to start the Far end loop
back test.
2026
On receiving the command to start Far End loop back test (either the Line Loop Back or Payload Loop
Back test).
The system will put this T1E1 port in maintenance mode and inform the ETERNITY ME Card T1E1 about
the test to be performed on the T1E1 port.
ETERNITY ME Card T1E1 will release all the calls supported by the T1E1 Port under test.
The ETERNITY ME Card T1E1 will start the PRBS generator and counter.
ETERNITY ME Card T1E1 will send the PRBS count every 1 second to the system.
ETERNITY ME Card T1E1 will increment PRBS Counter for every error encountered during the test every
one second.
ETERNITY ME Card T1E1 will reset the PRBS counter to zero, after sending the PRBS Counter to the
system, every second.
The system will store the received PRBS count (received every second) in Performance Report, which can
be captured from Serial port/Printer port/Ethernet port.
When the SE wants to end the loop back test he will issue the command to end the Far End Loop Back
Test, for the T1E1 port under the test.
SE will inform the other end's person that loop back test is finished and now the remote end can open the
loop formed.
On receiving the command to end loop back test for the T1E1 port, the system will take the T1E1 port out
of maintenance state and inform the T1E1 port about the received command.
When the line type is configured as "ISDN_T1_PRI" or "ISDN_T1_RBS" on the T1E1 port,
The activation methods are different for the D4 and ESF framing.
D4 framing supports only Line loop back
ESF framing supports both Line loop back and Payload loop back.
The test will start for the T1E1 port when SE issues the command to start the far end line loop back
test. When SE command (6142-1-T1E1-1) is issued.
The system will inform the ETERNITY ME Card T1E1 about the command and put this T1E1 port in
maintenance state.
The ETERNITY ME Card T1E1 will release all calls supported by T1E1 port under test.
The ETERNITY ME Card T1E1 will send the line loop back "Activation Code" and will start the PRBS
generator and counter.
ETERNITY ME Card T1E1 will send the PRBS count every 1 second to Master.
2027
ETERNITY ME Card T1E1 will increment PRBS Counter for every error encountered during the test
every one second.
ETERNITY ME Card T1E1 will reset the PRBS counter to zero, after sending the PRBS Counter to the
system, every second.
The system will store the received PRBS count (received every second) in Performance report, which
can be captured from Serial port/Printer port/Ethernet port.
When SE command (6142-T1E1-2) or (6142-T1E1-5) to end the Far end loop back test is issued.
The system will inform the ETERNITY ME Card T1E1 about the received command and will take out
the T1E1 port from the maintenance mode.
Depending upon the SE command issued to start the type of Loop Back (Line or Payload) test.
The system will put the T1E1 port under test in Maintenance mode and will inform the T1E1 Port about
the received command.
ETERNITY ME Card T1E1 will release all the calls supported by the T1E1 port.
ETERNITY ME Card T1E1 will send the Loop back "Activation Message" for the line/payload loop back
as informed by the system.
The ETERNITY ME Card T1E1 will start the PRBS generator and counter.
ETERNITY ME Card T1E1 will send the PRBS count every 1 second to Master.
ETERNITY ME Card T1E1 will increment PRBS Counter for every error encountered during the test
every one second.
ETERNITY ME Card T1E1 will reset the PRBS counter to zero, after sending the PRBS Counter to
Master, every second.
The system will store the received PRBS count (received every second) in Performance report, which
can be captured from Serial port/Printer port/Ethernet port.
2028
When SE issues command (6142-T1E1-2) or (6142-T1E1-4) or (6142-T1E1-5) to end the Far end loop
back test.
The system will inform the ETERNITY ME Card T1E1 about the received command and will remove
the T1E1 port from the maintenance mode.
ETERNITY ME Card T1E1 will send the "Deactivation Message" for the line/payload loop back test as
required.
Performance Report
A Performance Report of the tests can be generated. It contains the log of all the errors. The Performance Reports
stores upto 50 entries on FIFO basis.
From the Activity list, you can select to Activate or Release the loop back test.
2029
2030
Select the Communication Port/Ethernet Port/Printer Port to be assigned as destination port for Online
and Report logs.
If you select Ethernet Port as the destination, enter the IP Address of the Ethernet Port and the IP Port
(Listening Port for the packets) for the Online and Report logs respectively. Valid port range is:1025 to
65535.
From the Activity list, you can select to Start, End or Release the loop back test.
Meaning
Meaning
Performance Reports
Online Performance Report Printing
Use following command to assign the port for online Performance Report Printing:
6143-Port
Where,
Port
0
Meaning
None
2031
Port
Meaning
COM 1
COM 2
Printer
Ethernet
Default, None.
Offline Performance Report Printing
Use following command to assign the port for offline Performance Report Printing:
6144-Port
Where,
Port
Meaning
None
COM 1
COM 2
Printer
Ethernet
Default, None.
Meaning
Disable
Enable
By default, Flag is 0.
Offline Performance Reports
User the following command to start/abort offline printing of T1E1 performance report:
1072-031-Flag
Where,
Flag
Meaning
Disable
Enable
By default, Flag is 0.
2032
During loop back test the PRBS counter may be greater than zero at initial stage of the loop back
stage, but it will be zero afterwards consistently for the healthy condition.
PRBS counter = greater than zero, indicates the 'faulty' condition for the loop back test.
2033
Macros
Whats this?
Extension users often have to dial access codes for specific functions like dialing a feature code, making an internal
call, making an external call, etc.
ETERNITY supports Macros, using which, you can abbreviate long number strings for regularly used functions in to
macros and assign them to a DSS key on a DKP/Extended IP Phone extension.
You can also assign Macros on SLTs that have special keys.
How it works
Extension 2001, frequently sets Call Forward-All Calls to an external number 26550333.
To do this, each time, Extension 2001 must dial 131-Trunk Access Code-26550333-#*.
Instead of having to dial this lengthy number string, a Macro can be created for Call Forward-All Calls to
External number.
If Extension 2001 is an Extended SIP Phone or a DKP, the Macro can be assigned to a DSS key on the
phone.
Instead of dialing this number string, the user of Extension 2001 can simply press the DSS key on which
this Macro is assigned.
Thus when the DSS key on which a Macro is assigned is pressed, the corresponding access code is
executed.
ETERNITY also supports Macros for SLT which have special keys. When each of these keys is pressed, a special
number string, which you can program is dialed.
For example, an SLT instrument has 5 special purpose keys. When these keys are pressed, the strings *50, *51,
*52, *53, *54 programmed on these keys are dialed out.
You can create Macros for the strings dialed out using the special keys, whereby the string dialed by each of these
special keys is associated with a particular function. For example, the special key for dialing *50 is associated with
Call Forward -All Calls to an external number. So, when the extension user presses *50, the system receives this
string and takes appropriate action, that is, interprets it as call forward to the external number, and sets call forward.
Thus, each time the extension user presses the special key *50, the system considers that the extension user has
dialed 131-Trunk Access Code-26550333-#*.
2034
How to configure
You can create as many as 25 Macros using Jeeves and by dialing system commands from a telephone connected
to the ETERNITY.
The Macros page opens. Each macro is stored against an index number. By default the Macros Number
String and Access Code are blank when the system is operated in the Enterprise Mode.
In the Number String field, enter the strings the system should consider as command when the DSS
Key on the DKP/Extended IP Phone.
2035
When ETERNITY is operated in the Hotel Mode (see Customer Profile under System Parameters),
Number Strings and Access Codes are assigned for features such as Front Desk, Room Service, Voice
Guided Alarm, Reservation Desk, Voice Mail System, Retrieve Message. These Macros will appear on
this page. To know more, see the topic Customer Profile in the ETERNITY Hospitality System Manual.
In the Number String field, enter the strings the system should consider as command when the special
key on the SLT is pressed.
In the Access Codes field, enter the strings sent by the SLT on pressing the special function key.
For example, if the SLT sends the string *53 to the ETERNITY when the function key for Alarms is
pressed, enter the string 163 (the feature access code for Voice-guided Alarms) in the Number String
field, and enter the string *53 in the corresponding Access code field.
The Access Code that you assign here in Macros must not conflict with any other Access Codes in the
Dial Phase. See Access Codes.
2036
You can program a special digit for 'Pause' in the Key Board Macro Number string using the code #3.
To assign an Access Code for the Macro, dial:
3115-1-Macro Index-Access Code
Where,
Macro Index is from 01 to 25.
Access Code is a string of 4-digits.
Access Code is maximum of 6 digits. If it is less than 6 digits, terminate it with #*.
To clear Access Code assigned to a Macro, dial:
3115-1-Macro Index
Where,
Macro Index is from 01 to 25.
To default the access code for the Macro, dial:
3165-1-Macro Index
Where,
Macro Index is from 01 to 25.
When ETERNITY is operated in Enterprise mode, the default Macros Number String and Access Codes
are blank. So, when you dial this command, the values in the Macros will become blank.
When ETERNITY is operated in the Hotel Mode, the Number Strings and Access Codes assigned by
default will be restored when you dial this command. See the topic Customer Profile in the ETERNITY
Hospitality System Manual to know more.
Exit SE mode.
2037
Meet Me Paging
Whats this?
While a Paging announcement is being made, any extension user of ETERNITY can get connected to the Paging
extension, by dialing the Meet Me Paging feature code and the number of the Paging extension.
This feature is useful to Operators. Using this feature they can locate extension users who are away from their
desks and get connected to them at their current location.
How it works
Paging is an announcement made to a group of extensions within a Page zone. Extension users,
(including those who are outside the Page zone) who want to use Meet Me Paging to answer the
Paging call, will need to know the extension number they must call. Therefore, extension users who are
paging are advised to announce their extension number.
Meet Me Paging can be used only if the Paging call is active. Therefore, extension users who are
paging must keep their call active, if they want their call to be answered using Meet Me Paging.
How to configure
No configuration is required for Meet Me Paging. However, extension users who are using Paging to make their
announcement, must have the feature Paging allowed in their Class of Service.
How to use
You can use Meet Me Paging to answer a Paging call from any extension, if you know the number of the Paging
extension, and if the call is still active.
2038
2039
Message Wait
Whats this?
The Message Wait feature of ETERNITY enables extension users/Operator to set Message Wait on other
extensions to deliver important messages.
If the extension user has a mailbox assigned, the Message Wait feature indicates to the extension user, the arrival
of new messages in the users mailbox.
Thus, Message Wait can be set by extension users as well as by the Voice Mail System.
How it works
Message Wait set by Extensions/Operator
The Operator/any extension user can set Message Wait on another extension.
2040
The Operator has an important message to communicate. So, the Operator sets Message Wait on
Extension A, using the Message Wait key (if configured) or by dialing the feature access code.
Extension B tries to reach Extension A, and sets Message Wait on Extension A, using the Message Wait
key (if configured) or by dialing the feature access code.
Message Wait will be indicated to Extension A according to the Type of Message Wait Notification set for
Extension A. This may be in the form of a Stuttered Dial Tone, a Voice Message, Ring, or LED Lamp.
If Extension A is a DKP or an Extended IP Phone and has DSS key assigned for Retrieve New Message,
the LED of this key will glow to indicate new message wait.
Now, Extension A can dial the feature access code to retrieve Message Wait, or press the Retrieve
Message Wait Key, if assigned.
The system will call the extension that first set Message Wait on Extension A. In this case, the Operator. If
the Operator is busy, the system will place the call on Extension B. The system will try to call the
extensions that set Message Wait until the call is answered.
The extension that set Message Wait on A gets the CLI of A as Message Wait. A can now deliver the
message.
The LED of the Retrieve Message Wait key, if assigned, on Extension A will be turned off after all message
wait set by other extensions on Extension A have been served.
There is a new message in As Mailbox. The VMS indicates this to Extension A as per the Type of
Message Wait Notification set for Extension A. This may be in the form of a Stuttered Dial Tone, a Voice
Message, Ring or LED Lamp.
If Extension A is a DKP or an Extended IP Phone, the Voice Mail key on the phone will also glow to
indicate the arrival of a new message.
If the Retrieve Message Wait key is assigned on the DKP/Extended IP Phone of Extension A, the LED of
this key will also glow simultaneously to indicate arrival of the new voice mail.
SIP Trunk1 is registered with service provider on you have subscribed for Message Wait Indication.
When a Message Wait Indication is received on SIP Trunk1, the notification is sent to extension 3001 by
ETERNITY.
The new message indication on Extension 3001 is as per the Type of Message Wait Notification set for the
extension. This may be in the form of a Stuttered Dial Tone, a Voice Message, Ring or LED Lamp.
ETERNITY checks for a free call appearance of SIP Trunk1 and then automatically dials 5656.
2041
Message Wait Indication on SIP Trunk has a priority over Voice Mail and Voice Mail has priority over
extension Message Wait set by extensions.
If an extension has all three, Message Wait Indication on SIP Trunk, Message Wait and new Voice
mail, and when the extension user presses the Retrieve Message Wait key or dials the feature access
code to Retrieve New Message, the call will be placed first to the Mailbox of the SIP Trunk. When the
extension user presses the Retrieve Message Wait key again, the call will be placed to the Voice Mail
System. When the extension user presses the Retrieve Message Wait key for the third time, the call will be
placed to the extension that set Message Wait.
For detailed information and configuration instructions, see Message Wait Indication on SIP Trunks
When the extension user goes OFF-Hook, the user will hear a voice message, if a pre-recorded Voice
Module has been assigned for Message Wait Notification. If no voice module is recorded and assigned,
the extension user will hear a stuttered dial tone instead.
If you want voice message to be played as message wait notification, record and assign a Voice Module.
Refer Voice Message Applications for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait
Notification will not be played if there are already 4 being played simultaneously. In which case, Stuttered
Dial Tone will be played for Message Wait Notification, when the extension user goes OFF-Hook.
LED Lamp
When the extension user has an SLT with 'Message Wait' lamp, you can set this type of Message Wait
Notification. The lamp will blink continuously when Message Wait is set. The lamp will be turned off when
the extension user has retrieved all the waiting messages.
Ring
2042
When a new Message Wait is set on the extension, the system will play Message Wait Ring (Short, Fast)
on the extension. See Distinctive Rings.
The extension will ring for the duration of the Message Wait Ring Timer (configurable; default: 30
seconds). If the call is not answered within this timer, the system will ring on the extension again for as
many times as the Message Wait Ring Count (configurable; default: 10 times), and at the interval set as
the Message Wait Ring Timer Interval (configurable; default: 30 minutes).
When the extension user answers the call, the user gets connected to the VMS or the extension that set
Message Wait.
Value
Peak Voltage
82-85 V
ON Time
100ms
OFF Time
150ms
Frequency
4Hz
DC Offset
48V
How to configure
To provide this feature to extensions, you must do the following configuration on the extensions:
Enable the Message Wait feature in the Class of Service (COS) of the Station Basic Feature Template
of the extensions. This allows the extensions to set and cancel Message Wait on other extensions. Only
those extensions that have this feature in their COS can set or cancel Message Wait on other extensions.
By default, this feature is enabled in the COS of all extension types for all the time zones.
Select the desired Message Wait Notification Type in the Voice Mail Settings of the SLT, DKP, ISDN
Terminal, SIP extensions and Department Groups.
If you selected Voice Message as Message Wait Notification Type for an extension, you must also record
the desired Voice Message in a Voice Module and assign it to the Message Wait application. See Voice
Message Applications for instructions.
If you selected Ring as Voice mail/Message Wait Notification Type for an extension, you may configure
the following Ring Parameters:
Message Wait Ring Timer (default: 30 seconds)
Message Wait Ring Count (default: 10 attempts)
Message Wait Ring Timer Interval (default: 30 minutes).
See System Timers and Counts for instructions.
You may also assign DSS Keys to Message Wait and Retrieve Message Wait on DKP and Extended IP
Phone extensions.
For instructions on assigning these features to DSS keys of a DKP, see DSS Keys Programming.
For instructions on assigning these features to DSS keys of the Extended IP Phone, see Configuring SIP
Extensions.
2043
How to use
For EON & Extended IP Phones
To set Message Wait:
Press DSS Key assigned to 'Retrieve Message Wait' or the Voice mail Key when the LED glows.
OR
Dial 1077
For SLT
To set Message Wait:
2044
How it works
To be able to use Message Wait Indication (MWI) for the voice mail service of the ITSP, you must do the following
configuration on the SIP trunk,
When you subscribe for Message Wait Indication for a SIP Trunk, you get an indication about the status of your
mailbox from the ITSP.
ETERNITY passes the status of the message wait to the extension (SLT, DKP, ISDN Terminal, SIP Extension) you
have configured as the destination for receiving Message Wait Indication on the SIP trunk.
ETERNITY gives Message Wait Indication to the destination extension according to the type of Message Wait
Notification selected for the destination extension293, The types of Message Wait Notification offered by ETERNITY
on extensions are: Stuttered Dial Tone/Voice Message, LED Lamp and Ring. For more information, under the
chapter Features and Facilities, see Message Wait, Types of Message Wait Notification.
The extension user can now retrieve messages by dialing the Retrieve Message Wait feature code (1077).
When the extension user dials the Retrieve Message Wait feature code, ETERNITY checks if the Message Wait is
set by extensions, the Voice Mail System, or the extension is configured as the destination for Message Wait
Notification for a SIP trunk.
On finding the extension as the destination for Message Wait Notification for a SIP Trunk, ETERNITY dials out the
Message Wait Retrieval Number and connects the extension to the Voice Mail server of the ITSP.
The extension user can follow the Voice Mail prompts and listen to messages.
2045
How to configure
To use Message Wait Indication feature on a SIP trunk, you must configure the following parameters on that SIP
trunk:
Subscribe for MWI: This flag must be enabled to subscribe for Message Wait Indication on the SIP trunk.
Message Retrieval Number: This is the number provided to you by your ITSP for retrieving messages.
Send MWI notification on: This is the destination extension on which MWI notification is to be sent.
For instructions, see Configuring SIP Trunks.
2046
Mobility Extension
What's this?
ETERNITY offers mobility to its extension users whose nature of work keeps them from their desks frequently and
for longer durations.
Using mobility extensions, the extension users of ETERNITY can make and receive their calls from their current
(remote) location, placing calls through the system, and can access the system just as any other normal extension
of the ETERNITY.
How it works
ETERNITY supports two types of users:
Station Users: they are extension users of ETERNITY to whom a dedicated physical station is assigned
on their desk.
Virtual Users: they are extension users of ETERNITY who share a physical station, or may not have any
physical station allotted to them.
The facility of Mobility Extension is provided to both virtual and normal extension users using the features
Direct Inward System Access (DISA) and Call Forward.
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic License Management to know more.
How to configure
To provide Mobility Extension to users, the follow the steps described below.
List out the Station Users and Virtual Users and configure them first. The software ports of SLT, DKP and
ISDN Terminals which are not assigned a physical slot - port, can be used as Virtual stations.
Make sure that stations which are to be provided Mobility Extensions have the features Call Forward and
DISA enabled in their Class of Service (COS)294. Class of Service is to be programmed in the Station
Basic Feature Template assigned to the Mobility Extensions.
Make a list of External numbers to which the Mobility Station users will forward their calls. Program these
numbers in the 'Allowed List' of Local, Regional, National and International Numbers, as appropriate.
The Toll Control assigned to the station will be applied when a call is forwarded to an external number.
Make sure that the stations which are to be provided Mobility Extension have the required Toll Control
level for Call Forward to the External Numbers (the numbers you have programmed in the Allowed List).
Toll Control level is to be programmed in the Station Basic Feature Template assigned to the Mobility
Extensions.
294. It is possible to map the virtual user's flexible number (extension number) to a physical station. With this mapping, whenever a virtual user's number is dialed, the call is placed to the assigned physical station. The physical station can be assigned by defining it
as the 'Landing Destination for Virtual Users' in the Station Advanced Feature Template assigned to the virtual user.
2047
Program the parameter 'External Call Forward for' in the Station Advanced Feature Template assigned to
the Mobility Extensions. This parameter defines the types of call for which the External Call Forwarding is
to be applied. Select any one of the options, Internal Calls Only, Trunk Calls Only, Internal + Trunk Calls as
required.
Program the parameter 'DISA' in the Trunk Feature Template of the trunk lines which Mobility Extensions
users are to be provided access to. Make sure you select the option 'CLI Auth. Multiple Calls' in the 'DISA'
parameter of the 'Trunk Feature Template'.
Make a list of numbers which the Mobility Extension users will use to access the ETERNITY from DISA
mode. Program this list of numbers in the "DISA-CLI Authentication Table".
Program this list of numbers in the 'Calling Number' field of the Authentication Table. Program the 'Port
Type' and 'Port Number' of the Station assigned to Mobility Extension Users in the 'Auto Login As' field for
the respective 'Calling Number' field. Refer the topic Direct Inward System Access (DISA) to know more.
How to use
The Mobility Extension Users of ETERNITY can use the features of ETERNITY from a remote location as
described below.
Receiving calls
To receive calls, the Mobility Extension User must set Call Forward on his station with an external number (mobile
number, landline number, etc.) as the destination number.
To make calls ring on the station and the external number simultaneously, the Mobility Extension User must
activate the Call Forward-Dual Ring feature on his station.
The Mobility Extension User can also choose where he wants to receive the calls during a particular time of the day.
For example, he can receive calls during a particular time of the day, that is, Time Zone on his external number and
have his calls received by his Voice mail or the Operator or any other number during another Time Zone. To do
this, he must set Call Forward-Scheduled" on his station. Dual Ring can also be set for Call Forward-Scheduled.
Making calls
The Mobility Extension User should make a call on the DISA enabled trunk of ETERNITY from the external number
and the system will provide the dial tone to the user after authenticating the external number with the help of the
DISA-CLI Authentication table.
On getting the dial tone, the Mobility Extension User can make internal as well as external calls as per the Toll
Control and Class of Service (COS) assigned to his Station.
The Mobility Extension User can also dial codes of the Personal directory and Global directory numbers to use the
feature Abbreviated Dialing.
Accessing Features
The Mobility Extension User can access the system features by dialing specific codes after making calls on the
DISA enabled trunk of ETERNITY, or after answering the calls received on his external number.
2048
Code to be dialed
On-Hook
#0
Off-Hook
#1
Flash
#2
Pause
#3
#9
Described below are instructions for Mobility Extension users on using different call management features.
Call Hold
To put a call on hold,
Call Transfer
To conduct a screened Call Transfer,
2049
Call Pick Up
To pick up the call of same Call Pick Up-Group,
2050
3-Party Conference
To conduct a 3-Party Conference,
Multiparty Conference
To create a Multiparty Conference,
2051
Call Forward-Unconditional
To set Call Forward-Unconditionally,
Call Forward-Busy
To set Call Forward-Busy,
2052
Call Forward-Scheduled
To set/cancel Call Forward-Scheduled,
Mobility Extension users can have Call Forward and Call Forward-Scheduled set on their extension by
the Operator or by another extension user.
Using Call Forward-Remote and by setting Call Forward-Scheduled from the SA mode (using SA
commands or Jeeves), the extension user/Operator can set Call Forward and Call Forward-Scheduled
for any Mobility extension user. Refer the respective topics to know more.
2053
Multi-Stage Dialing
What's this?
The Multi-Stage Dialing feature of ETERNITY is typically used in applications like Calling Card, where extension
users are required to dial digits in stages when making a call using the calling card.
The Multi-Stage feature enables extension users to directly dial the number they want to call, and the system dials
out the number at different stages of the call by suitably modifying the number.
This feature requires a license. To use this feature you must purchase the license for the Mobility Feature
Suite. Refer the topic License Management to know more.
How it works
A typical example of Multi-Stage Dialing is the use of prepaid Calling Cards. Here, the person using a calling card
must dial a fixed number string before dialing the actual number. When using a calling card,
Users must first dial the number of the Calling Card server, for example: 1602233 (7 digits).
After the call is answered by the Calling Card server, users must dial the PIN provided by the calling card
service provider, for example 1132121234.
After dialing the PIN number, users can dial the number they want to call, for example 0014125126508.
Thus, when using a calling card, users must dial a very lengthy number string, each time they need to make a call
using the calling card.
The use of Multi-Stage Dialing saves the time and effort of dialing out lengthy digits in stages.
The Multi-Stage Dialing makes use of the Automatic Number Translation table. This table must be configured on
the trunk from which extension users will make calls using Calling Cards.
To take the above example further,
Say, extension users are allowed to make international calls using Calling Card from the trunk CO1.
Automatic Number Translation table must be configured on CO1 trunk.
The Automatic Number Translation table consists of Dialed Number Strings, Strip Digit and Add Prefix
Number strings.
In Dialed Number, you must configure 00, the prefix for international numbers.
In Add Prefix you must configure the Calling Card server number and the PIN Number.
As the system must wait for the Calling Card server to answer before dialing the PIN, you must configure
Wait for Answer between the Calling Card server number and the PIN number.
You must insert a delay by configuring the Pause Timer after the PIN number and the destination number.
2054
Strip Digits
Add Prefix
00
00
1602233W1132121234P
2
3
4
5
6
:
32
When the Automatic Number Translation table is configured, the Extension user can simply dial the Trunk
Access code and the destination number: 0/5/61/62/63/64 - 0014125126508.
The system matches the dialed number with the Dialed Number String of the ANT table, the number
matches with the entry 00 stored in the table.
The system dials the Add Prefix Number string 1602233 (number of the calling card server). It waits for the
calling card server to answer the call.
When the call is matured, i.e. the calling card server has answered the call, the system dials the PIN
number 1132121234 and waits for the Pause Timer before dialing the destination.
Thus, the extension user directly dials the desired destination number and the system substitutes this number
by adding the Calling Card server number and PIN number and dials these numbers in two stages.
How to configure
Configuring Multi-Stage Dialing using Jeeves
To be able to use Multi-Stage Dialing, you must configure the following:
Automatic Number Translation table on the trunk you want to use this feature. For instructions on
configuring ANT on different Trunk types, see Automatic Number Translation and to assign the ANT
Table number see Outgoing Trunk Bundle.
Configure the DTMF Out Dial on the Trunk. Set the DTMF ON Time and the DTMF Inter Digit Pause
Timer and Pause Timer to the required values.
For instructions on configuring DTMF Out Dial on different Trunk types, see Configuring CO Trunks,
Configuring E1 Trunks, Configuring T1 Trunks, Configuring BRI Trunks, Configuring Mobile Trunks,
Configuring SIP Trunks, Configuring E&M Lines.
If required you may configure the Call Proceeding Tone for Multi-stage Dialing as Network tone,
Pseudo Tone, or Silent. For instructions, see System Parameters.
2055
It is recommended to SE, to not to program Pause character "P" before "W" character for the number
string to be out dialed from Mobile port. Else, the GSM Module may get restart while out dialing the DTMF
digits without call maturity signal.
Flash (F)
#2
Pause (P)
#3
#4
#5
#6
#7
#8
. (dot)
#9
##
**
*1
Enable 'ANT' for the trunk port using command 6702. Refer chapter Outgoing Trunk Bundle for more details.
Assign the ANT Table to the trunk port using command 6702. Refer chapter Outgoing Trunk Bundle for more
details.
Refer chapter CO Hardware Template for command '5902' to program Pause Timer on CO Trunk.
Refer chapter Configuring PRI Trunks for command '6109' to program Pause Timer on T1E1 Trunk.
Refer chapter Configuring BRI Trunks for command '6209' to program Pause Timer on BRI port.
Refer chapter Configuring Mobile Trunks for command '8014' to program Pause Timer on Mobile port.
Refer chapter Configuring SIP Trunks for command '7720' to program Pause Timer on SIP trunk.
Refer chapter E&M Feature Template for command '6002' to program feature Pause Timer on E&M trunk.
Refer chapter CO Hardware Template for command '5902' to program DTMF ON Time on CO Trunk
Refer chapter Configuring PRI Trunks for command '6117' to program DTMF ON Time on T1E1 Trunk.
Refer chapter Configuring BRI Trunks for command '6210' to program DTMF ON Time on BRI port.
Refer chapter Configuring Mobile Trunks for command '8015' to program DTMF ON Time on Mobile port.
Refer chapter Configuring SIP Trunks for command '7725' to program DTMF ON Time on SIP trunk.
Refer chapter CO Hardware Template for command '5902' to program Inter Digit Pause Time on CO Trunk
Refer chapter Configuring PRI Trunks for command '6118' to program Inter Digit Pause Timer on T1E1 Trunk.
Refer chapter Configuring BRI Trunks for command '6211' to program Inter Digit Pause Timer on BRI port.
Refer chapter Configuring SIP Trunks for command '7726' to program Inter Digit Pause Timer on SIP trunk.
Refer System Parameters for command '5311' for programming of Call Proceeding Tone.
2056
How it works
ETERNITY supports Music on Hold from an Internal Music Source as well as an External Music Source.
If the option Routing Group is selected as the Alarm Notification Type for an extension, when the
extension goes Off-hook to answer an alarm call, and the extensions in the Routing Group for Alarm
Notification are busy, Music-on-Hold will also be played to the extension answering the alarm call.
If your DKP/SIP Extension supports multiple call appearance and you want the system to play MOH to
internal callers, when your extension is busy, enable the Play MOH to Queued Internal Calls on DKP/
Extended IP Phone check box in System Parameters. As soon as your extension is free, Ring Back
Tone will be played to the caller.
2057
When connecting an External Music Source to ETERNITY GE, you must change the position of the
Jumper for AIP Music Source selection on the CPU Card. For Jumper details, see The CPU Card or The
CPU Card - V1R7 and Later, under Installing ETERNITY GE.
On ETERNITY ME and ETERNITY PE, you can select the recording source by dialing System Command
from an extension phone or from Jeeves. Under Voice Message Applications, see Recording Voice
Messages
For instructions on recording voice modules, seeVoice Message Applications.
How to configure
Use the following command to demonstrate Music on Hold:
3551-Code
Where,
Code
Meaning
Voice Module 01
By default, code is 1.
Use following command to select the type of music to be played when stations are kept on hold:
3552-Code
Where,
Code
Meaning
Voice Module 01
By default, code is 1.
Use following command to select the type of music to be played when trunks are kept on hold:
3553-Code
Where,
Code
Meaning
Voice Module 01
By default, code is 1.
2058
If all the extensions of the Routing Group you selected for Alarm Notification type are busy, the
extension user will be played MoH (MoH can be Voice Module 01 or through AIP).
Example:
Program the system such that when stations are kept on hold, they get Voice Module 01 on Hold and when trunks
are kept on hold, the caller gets music from AIP.
3552-2
3553-2
Relevant Topics:
Background Music (BGM)
Voice Message Applications
2059
Number Lists
Whats this?
A Number List is a group of number strings. ETERNITY uses Number Lists to support different features as Toll
Control, Call Duration Control, Call Taping, Call Back on Trunk Ports, Station Message Detail Recording (SMDR).
ETERNITY supports 16 Number Lists. Each Number List can contain upto 999 number strings. Each number string
consist of a maximum of 16 characters.
The number strings are stored against Location Index numbers in the Number List. The Location Index numbers
start from 001 to 999.
The default values of the Number Lists in the system are shown below:
List Number
01
02
03
04
05
06
07
08
09
10
11
12
13
14
001
00
00
002
003
004
005
006
007
008
009
010
011
012
013
014
015
15
16
016
:
999
2060
Call Taping
Call Taping allows you to record conversations of incoming and outgoing internal and external calls. When you set
Call Taping on an extension for external calls, the ETERNITY uses two Lists: Number List - Incoming Calls: this list
has the list of numbers of external callers whose conversation is to be recorded.
Number List - Outgoing Calls: this list has phone numbers of external called parties whose conversation is to be
recorded. The system matches the incoming and outgoing numbers with the respective lists to apply Call Taping.
By default, Number List 09 is assigned to Incoming Calls Number List, and Number List 10 is assigned to Outgoing
Calls Number List.
Refer the feature description Call Taping to know more.
Toll Control
The Toll Control Call Privilege Type Limited Calls allows and restricts dialing of telephone numbers starting with a
particular digit or a particular area code or certain telephone numbers only. To apply Call Privilege type Limited
Calls you must program an Allowed Number List with numbers that are to be allowed, and a Denied List with
numbers that are to be restricted.
By default, Number List 01 is assigned as Allowed List as well as Denied List.
Refer the feature description for Toll Control to know more.
2061
How to configure
Take a pen and a paper. Decide which of the above-mentioned seven features are to be used. Number List
according to the feature for which it is to be used.
Number Lists can be programmed from Jeeves or by dialing SE commands from a telephone.
2062
2063
Select the desired List Number. For example, Number List 03-04. Now click 001-250' of Number list 03-04.
The location Index 001 to 250 will appear for both lists on the page.
Enter each number string against a Location Index (refer to the list you prepared).
Enter SE mode.
To program a number in a Number List, dial:
4302-Number List-Location Index-Number-#*
Where,
Where
List is from 01 to 16.
Location Index is from 001 to 999.
2064
Number is a number string of maximum 16 digits. The digits allowed are: 0-9, #, *, A, B, C, D, F, P, +.
Refer the table below for codes for dialing the special digits. Terminate the number string with #*
Special Digit
Code
Flash (F)
#2
Pause (P)
#3
#4
#5
#6
#7
#8
Dot (.)
#9
##
**
Exit SE mode.
2065
Mute
Whats this?
This feature helps the extension user to disconnect the speech transmission path in the middle of a conversation.
The extension user can still listen to the opposite party because the receiving path remains connected. Mute is
useful when you want to consult someone in the middle of a conversation, but do not want the opposite party to
listen to your discussion. You can Mute a call before making a call or during speech.
How it works
A is in speech with B.
A wants to consult to C in the room, but does not want B to hear their conversation.
A presses the Mute Key.
The transmit speech path from A to B is disconnected. The receive path remains connected.
So, A will be able to hear B, but B will not be able to hear the conversation between A and C.
When A has finished consulting C, to resume speech with B, A presses the Mute key again.
The transmit speech path from A to B is restored. A and B are in speech again.
How to use
For EON & Extended IP Phone Users
To mute a call before making the call:
Press the Mute Key
The LED of the key glows.
Dial a number on confirmation tone.
OR
Dial 1052
Dial desired number.
To mute a call during speech:
Press the Mute Key to silence outgoing speech.
OR
Press Transfer Key.
Dial 1052
To resume outgoing speech:
Press the Mute Key.
The LED of the key is turned off.
OR
Press Transfer Key.
Dial 1052
2067
OFF-Hook Alert
What's this?
When the handset of an extension is not placed correctly, it will not be possible for the Operator or any other
extension to call the extension. Also, incoming calls will not reach the extension, Alarms and Reminders cannot be
placed on that extension.
To avoid this inconvenience, the ETERNITY supports the feature 'OFF-Hook Alert', whereby the system detects
and informs the Operator of the extension phone that remains OFF-Hook accidentally.
How it works
To give the Operator an OFF-Hook Alert,
When the Operator answers the call, s/he is played a confirmation tone, the text message Hangup
<extension number> Properly is displayed.
The Operator can send someone to inform the extension user to place the handset of the phone correctly.
If the extension phone is an SLT, OFF-Hook Alert will be given to the Operator phone only. The Operator
phone can be an EON or a SLT with CLI support.
If the extension phone that is OFF-Hook is EON, the ETERNITY will activate 'OFF-Hook Alert' on the
extension phone, by playing the Error Tone continuously, on speaker to draw the attention of the extension
user.
While the Error Tone for OFF-Hook Alert is being played on the extension phone, if the user presses the
Speaker Key, the Error Tone will continue to be played on the handset until it is replaced correctly.
How to configure
For this feature to work,
the 'OFF-Hook Alert to Operator' flag must be enabled by the System Engineer in the 'System General
Parameters'.
the Operator phone can be EON or SLT with CLI support, the extension phones may be EON or SLT.
2068
Go to the parameter Give Off-hook Alert to Operator, select the check box to enable the flag.
2069
OG Reference Table
Whats this?
The OG Reference Table is a set of general features that define the logic of building the DDI Number for a station
placing a call and sending it to the network. An OG Reference Table is assigned to SIP/T1E1PRI/BRI ports. This
table in conjunction with DDI Routing Reference Table builds the DDI Number. The ETERNITY offers 64 such
Tables each of which can be programmed as per the requirement.
2070
OG Reference ID-This is the reference number acts as an identifier to the translation logic
programmed in the OG Reference Table. Any number of table can have the same reference number.
An OG Reference ID can be assigned to ISDN, SIP, T1E1PRI and BRI trunks. For more details on the
complete translation logic please refer the topics Direct Dialing-In (DDI), DDI Routing Table.
Start Channel Number-This is the first channel number for the trunk to which the logic is applicable.
Channel Count-The Total number of channels from the Start Channel Number of the port (Trunk) to
which the OG Reference Table is applicable.
ISDN Number-Each ISDN Trunk is given an Installation Number by the Service Provider. This is the
combination of Main Number (MSN Number) and the first DDI Number. The Number is of maximum 16
digits. This is also known as ISDN Installation Number or just ISDN Number. The MSN number is given
by the service provider whereas the Directory Numbers can be selected by the user. However the
number of digits to be used for the Directory Number should be informed to the service provider.
DDI Routing Reference ID-The DDI Routing Reference ID programmed in the OG reference table,
provides mapping with the DDI Routing Reference ID programmed in the DDI Routing Table. Using
the mapped entry of DDI Routing Table, the DDI number gets created from the flexible number. This
DDI number is sent as Calling Party Number while making OG call.
Parameter Name/
Index
OG Ref.
ID
Channel Count
ISDN
Number
01
00
01
00
Blank
000
2071
Parameter No.
Parameter Name/
Index
OG Ref.
ID
Channel Count
ISDN
Number
02
00
01
00
Blank
000
03-63
64
Same as 02
00
01
00
Blank
000
01-99
01-02
01-30
16 digits
001-128
Parameter Value:
Code
01-30
Relevant Topics:
1. Direct Dialing-In (DDI)
2. Configuring BRI Trunks
3. Configuring PRI Trunks
4. DDI Routing Table
2072
Trunk Type: Specific trunk bundle consists of some of these ports. The different types of ports are CO,
BRI, T1E1PRI, E&M, Mobile, LD and SIP.
Trunk Number: Once the type of port is identified in the bundle, it is required to identify the number of that
port, because there can be more than one port for the given type.
2073
Total Trunk Count: This is the number of trunks to be kept in the same bundle. For CO and E&M this
could be 128. This value for the BRI and T1E1 Trunks is counted from the start channel. For example, if
the port type is CO, port number is 002 and channel count is 025, then CO channels 002 to 027 would be
grouped together. Consider a second example where channels 15 to 25 of T1E1PRI port are to be
programmed in one channel group, then the port type will be T1E1PRI, port number would be 5, start
channel number will be 15 and channel count will be 11.
Rotation Type: This parameter shows which channel should be selected when the next call lands on that
port. For example, if ascending order is selected, the system checks 001-128, for first free channel and if
descending order is selected the system checks from 128-001.
Ascending Order:
001 to 128 (for CO/E&M)
01 to 30 (for T1E1PRI)
01 to 02 (for BRI)
01 to 32 (for SIP/LD)
Descending Order:
128-001 (for CO/E&M)
30 to 01 (for T1E1PRI)
02 to 01 (for BRI)
32 to 01 (for SIP/LD)
Cyclic:
Always the next channel is picked for a new OG call.
ANT Apply: Select from enable/disable as required. To use ANT feature, it should be enabled for the trunk
port from which the number is to be dialed out.
ANT Table No.: This is a Table number in which Dialed Number Strings and corresponding Substitute
number strings are programmed at specific Index. This table number is assigned to the specific trunk port
from which the number is to be dialed. Refer chapter Automatic Number Translation for more details.
2074
Bundle No.
Trunk Port
Start
Total
Channel
Trunk
6
Alternate Number
Rotation
Type
Number
No.
Count
001
CO
001
01
008
002
MOB
001
01
003
BRI
001
004
T1E1
005
Translation (ANT)
Type
Apply
Ascending
No
032
Ascending
No
01
002
Ascending
No
001
01
032
Ascending
No
SIP
001
01
032
Ascending
No
006
None
000
01
017
Cyclic
No
007
None
000
01
017
Cyclic
No
None
000
01
017
Cyclic
No
060
None
000
01
017
Cyclic
No
061
CO
001
01
008
Ascending
No
062
BRI
001
01
002
Ascending
No
063
T1E1
001
01
030
Ascending
No
064
MOB
001
01
032
Ascending
No
065
None
000
01
017
Cyclic
No
None
000
01
017
Cyclic
No
128
None
000
01
017
Cyclic
No
Parameter Values:
Code
Port
Type
None
None
000
CO
03
001-128
BRI
04
001-032
Descending
T1E1
05
001-008
Cyclic
E&M
06
001-128
Mobile
25
001-064
SIP
Trunks
26
001-032
LD
35
001-032
No
01-30
001-128
Ascending
Yes
1-8
2075
Relevant Topics:
Automated Control Applications
Outgoing Trunk Bundle
OG Trunk Bundle Group
Automatic Number Translation
2076
All the trunks connected to the system can be bunched in different groups called OG Trunk Bundle Group.
Maximum 8 Trunk Bundles can be put in one OG Trunk Bundle Group and 32 such OG Trunk Bundle
Groups can be formed.
An Extension can be allotted different OG Trunk Bundle Group during different timings of the day.
How it works
System uses two methods while selecting a trunk from the OG Trunk Bundle Group: Remember last trunk and
Dont Remember last trunk.
In Remember last trunk method, the system remembers the last trunk used and allots next trunk in the group to
the extension.
In Dont remember last trunk method, the system searches for a first free trunk from the group.
2077
Following flow chart depicts the chronology of events when a extension grabs a trunk.
Start
Which
LCR type is
selected ?
Is the
cheapest
trunk free
?
No
Mixed LCR
A
Yes
Select next
cheapest trunk
in this group
SP-SP LCR
C
System finds
cheapest trunk for
this number
End
Yes
System allots the
trunk to the station
End
Are other
trunk available
in this group
No LCR
No
Is cheapest
trunk free?
Yes
System dials out
the number
End
No
Are other
trunks available
in this group?
Yes
Select next cheap
trunk in this group
No
End
2078
Yes
Is the cheapest
trunk free ?
Are
other trunks
available with OG
trunk bundle
group ?
No
System gives busy
tone to the station
End
2079
2080
Meaning
Rotation OFF
Rotation ON
2081
61
62
63
64
Use following command to program the desirable access code for a trunk access index (TAC):
3112-1-OGTBG Index-Access Code-#*/Press <Hold>
3112-2-OGTBG Index-OGTBG Index-Access Code-#*/Press <Hold>
3112-*-Access Code-#*/Press <Hold>
Where,
OGTBG Index is from 1 to 6.
Access Code is maximum 6 digits (Generally access code for trunk is of two digits).
Use following command to clear the access code for a OGTBG index:
3112-1-OGTBG Index-#*
3112-2-OGTBG Index-OGTBG Index-#*
3112-*-#*
Use following command to assign default access code for a OGTBG index:
3162-1-OGTBG Index
3162-2-OGTBG Index-OGTBG Index
3162-*
How to use
1
Dial tone
2082
OGTBG has Rotation flag, and each OG trunk bundle has Rotation type Cyclic/Descending/Ascending.
These two flags dont have any relation with each other, and so, will work in isolation.
The first call will get routed using the OG Trunk bundle Member1. The trunk port from the OG Trunk
Bundle member 1 will get selected using the rotation type programmed for the OG trunk bundle
programmed as member 1.
When there is a second call, it will get routed using the OG trunk bundle Member 2, and the trunk port
from the OG trunk bundle member 2 will get selected according to the rotation type programmed in the OG
Trunk Bundle programmed as member2.
Accordingly, the member of the OGTBG will be accessible to the station user accessing the OGTBG in
sequence.
The calls will always get routed from the OG trunk Bundle member 1 if any trunk/channel in it is free. If all
the trunks/channels of the OG trunk bundle member 1 are busy, then the call will get routed using the
OG trunk bundle member 2.
Now the trunk/channel to route the call will get selected as per the rotation type programmed for the OG
trunk bundle used as member 2. When the trunk ports of OG trunk bundle programmed in member 1 and
member 2 all are busy, the OG trunk bundle member 3 will be used to route the call.
Thus the Rotation flag of OGTBG will be used to select the OG trunk bundle member1 to member 8 as per
call basis while the rotation type flag associated with the OG trunk bundle will decide the rotation
mechanism to select the trunk port from the particular OG trunk bundle.
Group
No.
Rotation
LCR
OGTB
Member
OGTB
Member
OGTB
Member
OGTB
Member
OGTB
Member
OGTB
Member
OGTB
Member
OGTB
Member
01
None
01
02
03
04
05
00
00
00
02
None
01
00
00
00
00
00
00
00
03
None
01
00
00
00
00
00
00
00
04
None
01
00
00
00
00
00
00
00
05
None
01
00
00
00
00
00
00
00
::
None
01
00
00
00
00
00
00
00
21
None
01
00
00
00
00
00
00
00
22
None
01
00
00
00
00
00
00
00
23
None
01
00
00
00
00
00
00
00
24
None
01
00
00
00
00
00
00
00
25
None
01
00
00
00
00
00
00
00
26
None
01
00
00
00
00
00
00
00
2083
27
None
01
00
00
00
00
00
00
00
28
None
01
00
00
00
00
00
00
00
29
None
01
00
00
00
00
00
00
00
30
None
61
62
63
00
00
00
00
00
31
None
64
00
00
00
00
00
00
00
32
None
61
62
63
64
00
00
00
00
Relevant Topics:
1. Outgoing Trunk Bundle
2073
2. Configuring LCR
1291
3. Time Tables
2372
4. Class of Service (COS)
1720
5. Station Basic Feature Template
726
6. Trunk Access Group (TAG)
2392
2084
Paging
What's this?
Paging allows you to make announcements to groups of extension users and to make public announcements over
a public address system. You can deliver a message to a mass of people at once by just lifting the handset of your
phone and dialing a code.
This feature is useful when you want to call several people at once; for example, to inform them about a meeting
you have scheduled. If the persons you want to call have Digital Key Phones (DKP) or the Matrix Extended IP
Phones, Radio devices or Open SIP Phones as their extensions, you can use paging instead of calling them up one
by one.
This feature is of great use in factories, offices, where it is not feasible to provide individual extensions in every
place, or when announcements are to be made in a hall, a lobby or a shop floor. In such situations, if you need to
call or inform someone, you can simply make an announcement over a Public Address System (PAS).
ETERNITY supports two types of paging:
Internal Paging: Announcements are made on DKP/SIP extensions /Radio Ports of the ETERNITY.
External Paging: Announcements are made on a Public Address System (PAS) connected to the Analog
Output Port of the ETERNITY.
For both types of Paging, the extensions which are to be paged and the Analog Output Port to which the PAS is
connected must be included in 'Page Zones'.
ETERNITY PE3SS does not support External Paging as there is no Analog Output Port.
You can start paging from an SLT, a DKP or any SIP Extension. However, the paged extensions must
be DKPs, Extended IP Phones, Radio device or Open SIP Phones. The Open SIP Phones on which
you are paging must support Call-Info or Alert-Info header for Paging.
When the Paging call is generated in SIP Extension having multiple call appearance and already a call
is present on the SIP Extension then the Eternity will place the Paging call as normal call on the SIP
Extension (as headers required in INVITE for paging call and intercom call are same).
Paging is a one-way communication. As the mic of the paged extensions is muted during Paging, the
users of the paged extensions cannot speak to the paging extension.
How it works
The Pre-requisites
Page Zones must be created. Each Page Zone accommodates up to 16 DKP/SIP extensions/Radio
devices and the Analog Output Port. You can create 12 different Page Zones of 16 DKP/SIP extensions/
Radio devices each.
Paging must be enabled in the Class of Service allowed to the DKP/SIP/SLT/Radio extension from which
this feature is to be used.
2085
For External Paging, a Public Address System must be connected to the Analog Output Port of the
ETERNITY. For instructions, refer the topic 'Connecting a Public Address System' under Installing
ETERNITY ME, Installing ETERNITY GE and Installing ETERNITY PE295.
The Process
A user of a DKP/SLT/SIP Extension having Paging in its Class of Service, dials the Access code for Paging
and the Number of the Page Zone to which the user wants to make the announcement.
The system activates the speakers of the DKP/SIP Extensions programmed in the Page Zone number.
The system activates the speakers only of those DKP/SIP/Radio extensions in the Page Zone that are
free.
If the Analog Output Port is included in the Page Zone, the PAS connected to this port will be activated.
All DKPs/SIP/Radio Extensions in the Page Zone can hear the announcement. But as the mic of their
phones is muted, their speech will not be heard by the calling DKP/SLT/SIP extension user.
If the Analog Output Port is included in this Page Zone, the announcement will be heard by the public.
To answer the Paging call the desired extension user must use Meet Me Paging while the Paging call is
active. For details refer, Meet Me Paging.
If no reply is received via Meet Me Paging, the calling DKP/SLT/SIP extension goes ON-Hook after the
announcement.
The system deactivates the speakers of the DKPs/SIP/Radio Extensions and the PAS connected to the
Analog Output Port.
How to configure
For this feature to work, you must create Page Zones and enable this feature in the Class of Service of the
extensions which are to be allowed this feature.
2086
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions and
programming.
Member Number
Type of Extension
DKP/SIP
Extension Port
Number
DKP
002
DKP
003
DKP
008
SIP
009
No
No
Member Number
Type of Extension
DKP/SIP
Extension Port
Number
SIP
007
SIP
010
SIP
012
DKP
013
Yes
Yes
:
:
16
Include Analog Output Port in Page Zone 1?
Page Zone 2
:
:
16
Include Analog Output Port in Page Zone 2?
2087
If you want to include only the Analog Output Port in a Page Zone, you do not need to assign any DKP/SIP/Radio
extensions to that Page Zone.
Make a note of the Page Zone for which you want to use only the Analog Output Port.
Now, program Page Zones using Jeeves or by dialing SE commands from a Telephone.
Refer to the Page Zones you created on paper and program the following parameters in each Page Zone:
Page on AOP: Select this check box if you want to include AOP in the Page Zone.
If you want to include only the AOP in a Page Zone, do not program any of the DKPs/SIP/Radio
Extensions in the Page Zone.
2088
Member Type: Select the type of extension you want to include in the Page Zone: DKP, Radio or SIP
extension.
Port Number: Enter the number of the Software Port of the extension you selected in the Port Type.
Click Submit at the bottom of the page to save your Page Zone settings.
To go to other Page Zones, you may click the hyperlinked page zone numbers on the top of the Jeeves
screen.
If you have completed programming Page Zones, you may log out of Jeeves.
Exit SE mode.
2089
How to use
It is possible to page from a DKP, a SIP Phone and an SLT.
2090
Peer-to-Peer Calling
Whats this?
Making an IP call without the intervention of a proxy server is called Peer-to-Peer Calling. As Peer-to-Peer calling
does not require a proxy server, voice communication using this application can be done virtually free of cost. The
major cost savings offered by this application makes it a very attractive mode of inter-branch or intra-office voice
communication.
How it works
Let us understand how to use Peer-to-Peer Calling with the following illustration:
There are three branch offices of the a company located at different places and they are connected to each other
over IP Network as shown in the figure below:
Mumbai
Baroda
PSTN
265
2637223
192.168.1.5
192.168.1.1
Sumer@pulver.com
32
PBX-A
2630555
PSTN
22
PBX-E
2654515
28111263
2001
2002
3001
28121234
3002
IP Network
123@abc.com
192.167.100.1
192.168 .1.3
Delhi
PSTN
11
31
PBX-C
9898001122
2001
6545351
2002
Peer-to-Peer calls can be made between the locations with suitable configurations of the PBX.
2091
Select a SIP trunk to be used for this application and enable it. For example, SIP Trunk 1.
Scenario 1
Extension 2001 of PBX-A (Baroda) wants to call extension 3001 of PBX-E (Mumbai).
In PBX-A, to route the calls configure the following:
The CUG table
The Peer-to-Peer table.
To configure the CUG table in PBX-A,
In the Route Code field of the CUG table, enter the Number that will be dialed to call the users of PBX-E.
In this case, 3001.
For the number you entered, in the Dialed Digit Count field, enter the 4.
Select the OG Trunk Bundle Group. Configure SIP Trunk1 as the only member in this group. The calls will be
routed through this SIP Trunk only.
The CUG table you configure in PBX-A would look like this:
Index
Route Code
OG Trunk
Bundle
Group
3001
01
Strip Digit
Count
Dialed Digit
Count
Self Route
As the system uses the best match logic to match number strings in the CUG table, you may configure only
the prefix of the number to be dialed. Instead of configuring the complete number string, you may configure
only the prefix as follows, the system will place all calls that start with '3' to the IP Address 192.168.1.1
2092
Index
Route Code
OG Trunk
Bundle Group
01
Strip Digit
Count
Self Route
Dialed Digit
Count
4
In the Number field of the Peer-to-Peer table, enter the numbers of the extension users of PBX-E. In this
case, 3001.
For the number you entered, in the Domain Address field, enter the Domain Name/IP Address of PBX- E.
In this case, 192.168.1.1
The SIP messages can be transported using UDP, TCP or TLS. Select the Default Transport for
Outgoing Message as per your requirement for each index.
The Peer-to-Peer table you configure in PBX-A would look like this:
Index
Number
No Match
Found
3001
Domain
Address
Name
192.168.1.1
PBX-E
UDP
As the system uses the best match logic to match number strings in the Peer-to-Peer table, you may
configure only the prefix of the number to be dialed. Instead of configuring the complete number string, you
may configure only the prefix as follows, the system will place all calls that start with '3' to the IP Address
192.168.1.1
Index
Number
No Match Found
Domain Address
Name
192.168.1.1
PBX- E
UDP
When 2001 from PBX-A dials 3001, the system compares it with the CUG table configured in PBX-A.
When a match is found in the CUG table, the system uses SIP Trunk1 to route the call.
SIP Trunk1 is configured as a Peer-to-Peer trunk, hence the system will check the Peer-to-Peer table.
As 3001 is configured in the Peer-to-Peer table, the system fetches Destination address and transports the
SIP messages using the protocol select as the Default Transport for Outgoing Message.
When there is an incoming call on PBX-E, the system checks the CUG table first and as 3001 is not
programmed in the CUG table, it checks the flexible number of the extensions.
2093
As 3001 is found in the flexible number list, the call is routed to the extension 3001.
Scenario 2:
Extension 2001 of PBX-A wants to call extension 2001 of PBX-C. In this case, both the offices have the same
extension numbers. Thus, when the extension users of PBX-A want to reach users of PBX-C, they must dial 312001, that is, the PBX Exchange ID (31) along with the extension number (2001).
In PBX-A, to route the calls configure the following,
The CUG Table
The Peer-to-Peer Table
To configure the CUG table in PBX-A,
In the Route Code field of the CUG table, enter the Number that will be dialed to call the users of PBX-C
(Delhi). In this case, 2001.
For the number you entered, in the Dialed Digit Count field enter the 6.
Select the OG Trunk Bundle Group. Configure SIP Trunk1 as the only member in this group. The calls will be
routed through this SIP Trunk only.
The CUG table you configure in PBX-A would look like this:
Index
Route Code
OG Trunk
Bundle
Group
312001
01
Strip Digit
Count
Dialed Digit
Count
Self Route
As the system uses the best match logic to match number strings in the CUG table, you may configure only
the prefix of the number to be dialed. Instead of configuring the complete number string, you may configure
only the prefix as follows, the system will place all calls that start with '3' to the IP Address 192.168.1.3
Index
Route Code
OG Trunk
Bundle Group
31
01
Strip Digit
Count
Self Route
Dialed Digit
Count
6
2094
In the Number field of the Peer-to-Peer table, enter the Number of the extensions of PBX-C. In this case,
2001.
For the number you entered, in the Domain Address field, enter the Domain Name/IP Address of PBX-C.
In this case, 192.168.1.3
The SIP messages can be transported using UDP, TCP or TLS. Select the Default Transport for
Outgoing Message as per your requirement for each index.
The Peer-to-Peer table you configure in PBX-A would look like this:
Number
Destination
Address
Name
Default Transport
for Outgoing
Message
192.168.1.3
PBX-C
UDP
No Match Found
312001
As the system uses the best match logic to match number strings in the Peer-to-Peer table, you may
configure only the prefix of the number to be dialed. Instead of configuring the complete number string, you
may configure only the prefix as follows, the system will place all calls that start with '3' to the IP Address
192.168.1.3
Number
Destination Address
Name
192.168.1.3
PBX-C
UDP
No Match Found
31
In the Route Code field of the CUG table, enter the extension numbers along with the PBX Exchange ID.
In this case, 312001.
Select the OG Trunk Bundle Group. Configure SIP Trunk1 as the only member in this group. The calls will be
routed through this SIP Trunk only.
2095
The CUG table you configure in PBX-C would look like this:
Index
Route Code
OG Trunk
Bundle
Group
Strip Digit
Count
Self Route
Dialed Digit
Count
01
312001
01
enabled
As the system uses the best match logic to match number strings in the CUG table, you may configure only
the prefix of the number to be dialed. Instead of configuring the complete number string, you may configure
only the prefix as follows, the system will place all calls that start with '3' to the IP Address 192.168.1.3
Index
Route Code
OG Trunk
Bundle Group
Strip Digit
Count
Self Route
Dialed Digit
Count
01
31
01
enabled
When 2001 from PBX-A dials 31-2001, the system compares it with the CUG table configured in PBX-A.
When a match is found in the CUG table, the system routes the call using SIP Trunk1.
SIP Trunk1 is configured as a Peer-to-Peer trunk, hence the system will check the Peer-to-Peer table
configured in PBX-A.
As a match is found in the Peer-to-Peer table, the system fetches the Destination address and transports
the SIP messages using the Default Transport for Outgoing Message.
When there is an incoming call on PBX-C, the system checks the CUG table configured in PBX-C and as
312001 is found in the CUG table with self route flag enabled and strip digit count 2, the system strips the
first two digits and the call is routed to the station 2001 of PBX-C.
2096
Under VoIP Configuration, click Peer-to-Peer Table. The Peer-to-Peer table opens.
In the Number field, enter the peer-to-peer number stringprefix or entire numberthat will be dialed.
The number string must not exceed 8 digits. Default: Blank.
If the number to be dialed out is <dialednumber@destination address>, for example, 1234@abc.com,
you must enter 1234 in this field.
In the Domain Address field, enter the domain name or IP Address to where the call is to be placed.
The Domain Address may consist of up to 40 characters (maximum). Default: 192.168.1.112
For example, if the peer-to-peer number to be dialed out is 1234@abc.com, enter abc.com as
Destination Address. If the number is 1234@ 192.168.1.197, enter 192.168.1.197 as the Destination
Address. The Destination Address can also be in the form of Address: Port number.
In the Name field, enter a name to identify the number string you configured. It may be the name of
your contact or any name you wish to assign to the number string. The name may consist of 24
characters (maximum). Default: Blank.
The name you configure here will not be used in SIP signaling.
.In the Default for Outgoing Message field, select the option for transporting outgoing SIP messages.
You can select UDP, TCP or TLS.
The paramerter Default for Outgoing Message can be configured through Jeeves only.
2097
2098
PLCC-An Introduction
Whats this?
ETERNITY-PLCC EPAX is a digital PBX. It uses a digital switch and hence is a 100% non-blocking
system. In PLCC network, number of PLCC EPAX needs to be connected.
Refer to diagram showing cluster of exchanges in a PLCC network. Each exchange in PLCC network is
assigned with Exchange Identity (SID) in order to get identified by other exchanges in the network. Thus
each exchange is identified by Exchange Identity (SID). As shown, an exchange is connected to the other
exchange through an E&M tie line. Also, it is possible that an exchange may not be directly connected to
all other exchanges in a PLCC network through E&M tie lines.
For example, SID-52 exchange is connected to SID-51, SID-34, SID-71 and SID-60 exchanges through
direct E&M tie lines. However, SID-52 exchange is not directly connected to SID-61, SID-62 and SID-72
exchanges through E&M tie lines.
Consider, an example where subscriber 20 of SID-52 needs to call subscriber 30 of SID-51. After
dialing trunk access code, dial 51 (SID number of the exchange) and then dial 30 (subscriber number
of SID-51). This is how a call is completed, as both exchanges; SID-51 and 52 are connected through a
direct E&M tie line.
However, consider another example, where subscriber 20 of SID-52 needs to call subscriber 10 of SID61. Even though, here both exchanges are not connected through direct E&M tie line, it is possible by
dialing trunk access code with SID number of the exchange (here 61) followed by subscriber number
(here 10). However, the call will proceed through SID-60 and then reach SID-61. This is called Transit
Call. Here, the subscriber will not be able to know that call has proceeded to the required exchange
through transit facility.
PLCC Network
SID-62
Sub. 10
SID-61
Sub. 11
Sub. 12
SID-34
SID-60
Sub. 20
SID-52
Sub. 21
Sub. 22
SID-51
SID-71
SID-72
Sub. 30
Sub. 31
2099
E1
En
E2
PLCC-EPAX (SID-61)
E&M1 E&M2 E&M3
E&Mn
PLCC-EPAX
(SID-70)
E&M1
PLCC-EPAX
(SID-34)
E&M1
E&M1
E&M2
E&M2
PLCC-EPAX E1
(SID-52)
E&M3
E&M3
E&Mn
E&Mn
E1
E2
En
E&Mn
E2
En
E&Mn
PLCC-EPAX (SID-51)
E1
E2
En
PLCC-Priority
All callers do not have same hierarchical position in an organization. It is not advisable to keep a call
waiting ofsome important person just because there is already one unimportant call pending at the
destination. The important caller should be allowed to jump the queue and be attended first ahead of other
earlier pending calls.
The ETERNITY supports flexible priority assignment for different users. Each port can be assigned a
priority level between 1 to 9. Higher the priority level, more important the caller is. Accordingly 1 has the
least priority and 9 has the highest priority.
This feature enables a station user to free the system resources (station or a trunk) for him.
The PLCC functional module requires a license. You must purchase a license to activate CCS Signaling
when End Point and Transit, Express Signaling, and Seizure Pulse and Release Pulse Signaling. Refer the
topic License Management.
2100
How to use
1
Dial tone
Busy tone
Dial #*.
The called station/trunk gets disconnected. You get dial tone after the
confirmation tone.
How it works
Suppose, Station A and Station B are talking to each other. Station C calls Station B and finds it to be busy.
If he uses Priority, he gets connected to Station B. Station A gets disconnected and gets error tone.
However, for this to happen the priority of Station C should be higher than that of Station B and Station A
else he wont be able to use this feature.
Likewise suppose Station A is talking to an external party through trunk 1. Station B tries to grab Trunk 1.
On finding it busy, he uses Priority. Doing so, he gets connected to trunk 1 and gets P&T dial tone whereas
Station A gets disconnected and gets error tone. However, for this to happen priority of Station B should
be higher than Station A. In this case, priority of the trunk is not considered.
Suppose station A is talking to an external party through trunk1. Station B calls station A and finds it busy.
He uses Priority. Doing so, station B gets dial tone whereas, the trunk 1 gets disconnected.
SID-61
In conversation
SID-52
SID-34
Sub. 20
Sub. 21
Sub. 22
(With priority access)
How to use
Please refer above diagram. If any two subscribers, say 20 and 21 of an exchange with SID-52 (without priority
access) are in conversation and a third subscriber, say 22 (of same exchange) with priority access and wants to
talk with subscriber 21. But, when subscriber 22 dials for subscriber 21, it gets busy tone. Subscriber 22 can use its
priority by dialing code #* in such case and can come in conference with subscribers 20 and 21.
2101
SID-61
Sub. 10
(With priority access)
SID-52
SID-34
In conversation
Sub. 20
(Without priority access)
Sub. 21
(Without priority access)
How to use
Please refer above diagram. If any two subscribers, say 20 and 21 of an exchange with SID-52 are in conversation
and a subscriber, say 10 of an exchange with SID-61 (from the same network with priority access) want to talk with
subscriber 20 of SID-52. After dialing the required subscriber 20 of SID-52, he gets busy tone. Subscriber 10 of
SID-61 can now press #* on his telephone and can get into conference with both subscriber 20 and 21 of SID-52.
SID-61
SID-52
Sub. 10
(With priority access)
Sub. 11
(Without priority access)
Sub. 20
Sub. 21
How to use
Please refer diagram, if two subscribers, say 20 of SID-52 (without priority access) and 11 of SID-61 (without
priority access) are in conversation on an E&M tie line. Both this exchanges SID-52 and SID-61 are connected with
each other through a single E&M tie line. Now, if subscriber, say 10 (with priority access) of SID-61 wants to talk
with subscriber 21 of SID-52, then he will get busy tone. Subscriber 10 of SID-61 can now use its priority by
2102
pressing #* on his telephone and can get into conference with both subscriber 11 of SID-61 and subscriber 20 of
SID-52. After termination of the call by subscribers 11 and 20, subscriber 10 can get dial tone and make a call on an
E&M tie line to subscriber 21 of SID-52.
PLCC-Routing Table
In PLCC network, number of PLCC-EPAX needs to be connected. The entire network should behave as a
single unit or one group. It is not feasible to have unique station numbers throughout the network. In such
cases, an Exchange ID is assigned to the PBX and a routing table is programmed in the exchange. In case
of newly added exchange in the network, the routing tables of other exchanges required to be modified.
In the above figure, 3 PBX systems are connected through E&M connectivity.
How it works
In this application, it is possible to have same station number in two or more PBXs of the network. Few new words
have been used to explain PLCC routing table, each of these words have been explained below:
Routing Table: This table has five parameters viz. Route Index, Route Code, OG Trunk Bundle Group, Strip Digit
Count and Self Route flag. The PLCC routing table programming works according to this table.
Route
Index
Route
Code
OGTBG
Strip Digit
Count
Self Route
Flag
Maximum Dialed
Digits
001
:
:
250
2103
2104
How to configure
For more details on above steps, please refer topic Closed User Group (CUG) and Closed User Group-With
Exchange ID.
How to configure
PLCC Express Line Communication System is one of the applications of the ETERNITY. We recommend the user
to read relevant topics before programming it for PLCC Express Line application.
Step 1
Refer chapter E&M Feature Template to program the feature in an E&M Feature Template.
Step 2
Refer chapter E&M Feature Template to assign default values to an E&M Feature Template.
Step 3
Refer chapter E&M Feature Template to assign an E&M Feature Template to an E&M.
Step 4
Refer chapter Configuring DKP Extensions to program a name for E&M trunk.
Step 5
Hardware ID is an attribute of a software port. Hardware ID of a software port decides where the port is physically
located. To derive hardware ID of a software port, we need slot number and port number of the card. Hence, all the
programming is done for the software port and not for the hardware ID. Accordingly, the software port number is
used for all the programming. Please refer Software Port and Hardware ID for more details, to assign hardware ID
to an E&M software port:
Step 6
Refer chapter Station Advanced Feature Template to assign a Station Advanced Feature Template to an E&M.
Step 7
Refer chapter Station Advanced Feature Template to assign a Station Basic Feature Template to an E&M.
Step 8
Please refer Time Tables chapter to program the time zone of a trunk.
Step 9
To program a feature in a Trunk Feature Template, please refer Trunk Feature Template for more details.
Step 10
To assign a Trunk Feature Template to an E&M, please refer Trunk Feature Template topic.
Matrix ETERNITY System Manual
2105
Step 11
Refer chapter DSS Keys Programming to assign a function to a key of DSS-64 assigned to a DKP. In this
commands function type and function number can be programmed as explained below.
Function:
A function should be assigned to these Keys, which they should perform. There are 2 different types of functions,
which can be assigned to a key for PLCC express line application. Each function that can be assigned to a key is
given unique number. Following table list the function types available:
Function Type
Meaning
Function Number/Port
00
Null
--
06
Access a trunk
001 to 128
Step 12
Refer chapter DSS Keys Programming to assign a function to a DKP key. In this command refer above
explanation to program Function Type.
Step 13
Define dialing property of E&M port. The dialing properties are based on the applications. If the dialing type for a
E&M is programmed as pulse type then Pulse-Dialing ratio should be defined. The E&M support six different Pulse
Dialing Ratios.
Value
Meaning/Ratio
10PPS, 1:2
10PPS, 2:3
10PPS, 1:1
20PPS, 1:2
20PPS, 2:3
20PPS, 1:1
For assigning Dial Type and Pulse Dial Ratio, please refer E&M Feature Template topic.
Relevant Topics:
1. Trunk Feature Template
874
2. Time Tables
2372
3. E&M Feature Template
895
4. Closed User Group (CUG)
1739
5. Closed User Group-With Exchange ID
2106
1745
Presence
What's this?
Extension users may want to indicate their availability to callers from other extensions.
For example, an extension user may want to leave his desk for an indefinite period, but does not want to use Call
Forward or set Do Not Disturb. He wants to indicate to callers about his absence. Similarly, extension users who
are present at their desk may want to hide their presence from other users; or they may want to show their current
activity to the other extension users like they are Busy, or are away from their desks, or on the phone with someone
on another call, etc.
With the Presence feature of ETERNITY, extension users, including the Operator, can 'publish' their presence to
callers from other extensions. By doing so, they can indicate to the other extensions about their availability.
In the same way, the Presence feature allows extension users to view the 'Presence' status (availability) of the
extensions that they want to call, before making the call or when their call is not answered.
How it works
Publishing Presence
Any SLT, DKP, ISDN Terminal, and SIP Extension User can 'publish' their presence by setting any of the messages
listed in the following on their phone, by dialing the access code for this feature.
SIP Extension users who want to publish their presence have two options:
Using the PUBLISH feature supported by the SIP Client.
Using the feature access code for Publish Presence supported by ETERNITY.
The first option requires the parameter 'PUBLISH' to be enabled in the SIP Extension Settings. Refer
Configuring SIP Extensions. By default, this parameter is disabled.
2107
When an SLT extension user calls the extension which has set 'Absent', an error tone will be played.
However, it is possible for the SLT extension user to find out the presence status of the called
extension. Refer Viewing Presence later in this topic.
External callers who call the extension, on which 'Absent' is set, will get an error tone only.
Outgoing calls can be made from the extension which has set 'Absent'. Only incoming calls are
restricted.
If more than one extension is configured as "Operator" (routing group), incoming calls will be blocked
only on the Operator extension which has set User Absent.
1. Present:
When an extension user sets 'Present', all incoming calls will be received as normal on this extension.
If previously set as 'Absent', when a DKP extension user sets 'Present' the letter 'A' will disappear from the
phone's display.
When any other DKP extension user calls this extension, the name of the extension user will be displayed
on the caller's phone display, when the called extension is ringing.
2. Auto Detect: When an extension user sets 'Auto Detect', the system will detect the state of the phone;
depending on the call state, it will publish the presence message to the other extensions. Three types
Publish Presence messages are possible, with Auto Detect:
a. Idle: When the system detects the extension phone to be ON-Hook, it indicates the status of the phone
or if it detects an incoming call placed on the phone, it will indicate to the other extensions that this
extension user is 'On the Phone' with another party.
c. DND Text message: When the system detects that the extension phone has Do Not Disturb (DND) set
on it with a DND Text message, it will display to the calling extension, the DND message set by the
called party (this may be the default DND message or the DND Text message set by the called
extension).
3. Away: When an extension user sets 'Away', the system will display this message to the other extensions.
4. On the Phone: When an extension user sets 'On the Phone', the system will display this message to the
other extensions.
5. Do Not Disturb: The extension user can set this message to be published to other extensions, if s/he
2108
7. In Meeting: The extension user can set this message to be displayed to the callers, if s/he is busy in a
discussion or meeting.
8. Out for Meal: The extension user can set this message to be displayed to other extensions when going on
a lunch break.
9. Out of Office: The extension user can set this message to be displayed to the callers when s/he leaves
Viewing Presence
Extension users can know the status of another extension user before calling or when the extension user
does not answer the call.
Generally, when DKP extension users call another extension, the name of the called extension is
displayed on the calling DKP extension. Now, if the flag 'Display User Status during Call' is enabled in the
System Parameters, when DKP extension users call another extension, the calling DKP extensions will be
displayed the 'presence' status message published by the called extension296.
SLT extension users, whose phone is equipped with a CLI display, can see the status of another extension
by dialing a feature access code, then going ON-Hook. The system will ring back the SLT and send the
Presence status of the desired extension as CLI.
SIP extension users can use the Presence feature of ETERNITY to view the presence status of other
extensions. For this, they must dial the feature access code and the number of the desired extension.
SIP extension users who want to view the status of other extensions using the feature supported by their
SIP Client, must have 'Presence Subscription' enabled in their SIP Extension Settings. Refer Configuring
SIP Extensions.
How to configure
This feature involves the programming of the following parameters:
'Display User Status during Call' flag: DKP extension users will be able to view the presence status for
the called extension only if this flag is enabled in the System Parameters.
PUBLISH: SIP extension users who want to publish their presence using the feature supported by their
SIP client will be able to publish their presence status only if this feature is enabled in their SIP Extension
Settings. This parameter is not necessary, if they want to publish presence using the feature of ETERNITY.
296. DKP users can also dial a feature access code and the number of the extension to see the status of that extension on their DKP.
But this would not be required, if the 'Display User Status during Call' flag is enabled in the System Parameters.
2109
Presence Subscription: SIP extension users who want to view the presence of other extensions using
the feature supported by their SIP client must have this feature enabled in their SIP Extension Settings.
This parameter is not necessary, if they want to view presence using the feature of ETERNITY.
Publish Messages: It is possible to customize the Publish Messages listed above from 6 to 9 viz.: 'I am
Mobile', 'In Meeting', 'Out for Meal', 'Out of Office'.
The above parameters, with the exception of 'Publish Messages', can be programmed using both Jeeves and
a Telephone. You can program 'Publish Messages' using Jeeves only.
2110
Go to Display Presence Status during Call on DKP. Click to enable the flag.
You can change message number 6 to 9 as desired. The string may consist of a maximum of 16
characters. All ASCII characters except < > and (double quote) are allowed.
Now, go to the desired SIP Extension number for which you want to enable the features PUBLISH and
Presence Subscription. By default both features are disabled. Click the respective check boxes to enable
the features.
2111
1 for Enable
By default, flag is disabled.
For SE commands to enable PUBLISH and Presence Subscription on SIP Extensions, see
Configuring SIP Extensions using a Telephone.
Exit SE mode.
How to use
This feature requires you to dial your User Password. The default User Password 1111 is not accepted. Please
change the User Password first.
Publish Presence can be set for an extension also from the System Administrator mode.
Dial 104
Enter User Password on the prompt.
Scroll to the desired Publish message from the menu:
Absent
Present
Auto Detect
Away
On the Phone
Do Not Disturb
I am Mobile
In Meeting
Out for Meal
Out of Office
Press Enter key to select message.
You get the confirmatory tone.
2112
Dial 1072-014
Enter Destination Number, that is, the number of the extension Publish Presence is to be set.
Scroll to the desired Publish message from the menu:
Absent
Present
Auto Detect
Away
On the Phone
Matrix ETERNITY System Manual
Do Not Disturb
I am Mobile
In Meeting
Out for Meal
Out of Office
Press Enter key to select message.
You get the confirmatory tone.
Dial 1097.
Enter Extension number
The status of the extension number you dialed will be displayed on your phone's LCD.
Go ON-Hook.
Meaning
Absent
Present
Auto Detect
Away
On the Phone
Do Not Disturb
I am Mobile
In Meeting
Out of Office
Replace handset.
Meaning
Absent
2113
Index No.
Meaning
Present
Auto Detect
Away
On the Phone
Do Not Disturb
I am Mobile
In Meeting
Out of Office
Replace handset.
2114
Lift handset.
Dial 1097-Extension Number.
You get confirmation tone.
Go ON-Hook during confirmation tone.
Your phone will ring and the status of the extension number you dialed will be displayed on your phone as
CLI.
E&M MFCR2 Signaling is currently supported on ETERNITY GE E&M Card only. This feature is currently
available on ETERNITY GE only.
How it works
The extensions which are to be allowed Priority Call feature must be defined as 'Priority Subscriber' in their
Station Advanced Feature Templates.
The extensions on which Priority Calls should be allowed to land must be 'Non-Priority Subscribers' (that
is, defined as 'Ordinary Subscriber' in the Station Advanced Feature Template).
In the following illustration, ETERNITY is networked with PBX-A over E&M Lines with MFCR2 Signaling.
Priority Call from ETERNITY to Networked PBX
2001
ETERNITY
2002
Rx
Rx
Tx
Tx
SA
SA
SB
SB
3001
PBX-A
3002
Extension 2001 of ETERNITY is a 'Priority Subscriber'. Extension 2002 is an 'Ordinary Subscriber' (NonPriority).
Extension 3001 of PBX-A is a 'Non-Priority Subscriber'. Extension 3002 of PBX-A is also a 'Non-Priority
Subscriber'.
2115
When Extension 2001 calls Extension 3001, depending upon its implementation, PBX-A understands it as
a priority call.
If Extension 3001 is busy with another extension, 3002, and since both are non-priority subscribers, PBXA will automatically intrude the conversation between 3001 and 3002 and establish 3-way speech between
Extensions 2001, 3001 and 3002.
Now, if Extension 2001 (the Priority caller) does not want Extension 3002 to be part of the conversation, s/
he can ask 3002 to go idle OR s/he can use the feature Forced Release Order to disconnect 3002 from the
conversation. 2001 can use Forced Release Order only if it is allowed to it in its Class of Service (COS).
When Extension 2001 dials the Feature Access Code for Forced Release Order, Extension 3002 will be
disconnected and two-way speech will be established between 2001 and 3001.
Priority Call from Networked PBX to ETERNITY
3001
PBX-A
3002
Rx
Rx
Tx
Tx
SA
SA
SB
SB
2001
ETERNITY
2002
Extension 3002 of PBX-A is a Priority Subscriber. Extension 3001 of PBX-A is Non-Priority Subscriber.
Extension 2002 is busy with Extension 2001. ETERNITY checks the priority of both extensions. Since both
are non-priority subscribers, ETERNITY gives priority to Extension 3002 and plays beeps to 2002 and
2001 before establishing 3-way speech.
2116
If the busy call is itself a priority call, that is, at least one of the two parties in the busy call is a priority
subscriber.
If any party involved in a matured call is Priority Extension or Trunk, the call will be considered as
priority call and hence no incoming priority call will be allowed to intrude this priority call.
ETERNITY allows non-priority subscribers to intrude on a busy call using the feature Manual Priority
Intrusion. For this, the intruding extension must have the feature 'Manual Priority Intrusion' in its Class of
Service.
When a non-priority extension requests Manual Priority Intrusion, ETERNITY receives an 80msec pulse
signal on the E wire of the E&M port, on detecting this signal ETERNITY will treat it as a Priority Call and
check whether the called extension that is busy can be intruded or not.
If intrusion is possible ETERNITY creates a conference call, after playing beeps to notify both parties of the
3-way conference call. This beep will be played only if Conference beeps are enabled.
How to configure
For this feature to work, the extensions of the networked PBXs which are to be allowed Priority Call feature must be
defined as 'Priority Subscribers'.
Extensions on which Priority Calls must be allowed to land may be defined as 'Non-Priority Subscribers'.
If non-priority subscribers are to be allowed to Priority Calls when necessary, Manual Priority Intrusion feature must
be enabled in their Class of Service.
If the Priority Subscriber extension users are to be allowed to disconnect the second party (other than the desired
party) from the conversation during a Priority Call, the feature Forced Release Order must be enabled in their Class
of Service.
For Stations on which data terminals are connected (like Fax machine), it is recommended that the 'Station
Category' of these stations be programmed as 'Data Transmission' (in their Station Advanced Feature
Template).
Programming Manual Priority Intrusion and Forced Release Order in Class of Service
In the default the default CoS group 01 in Station Basic Feature Template Number 01 assigned by default to all
extensions of ETERNITY, 'Manual Priority Intrusion' and 'Forced Release Order' are disabled.
If you want to allow all extensions these features, simply enable them in the default CoS group 01.
However, if either or both these features are to be allowed only to select extensions, follow these steps:
1. Define a CoS group with Manual Priority Intrusion/Forced Release Order enabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
2117
3. Assign this new Template to the extensions to which Manual Priority Intrusion/Forced Release Order is to
be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for detailed instructions.
How to use
Manual Priority Intrusion
For EON and Extended IP Phone Users
2118
Priority
Whats this?
Priority is the precedence given to certain trunks and extensions over others in being answered by the destination
extension.
When Priority is assigned to trunks, whenever there are incoming calls on multiple trunks at the same time, the call
on the trunk with highest priority will be answered by the landing destination extension/Operator first.
When Priority is assigned to Extensions, calls from extensions with highest priority will have precedence in landing
on the destination extension.
You can set priority levels from 1 to 9 as given in the table below.
Priority
Level
Meaning
None
Lowest
Lower
Low
Normal
Medium
High
Higher
Highest
Highest Priority can be assigned to Extensions of important or higher ranking persons in an organization; for
example, calls from senior managers or top executives in an organization can be allowed to be answered first by
the destination extension.
Highest Priority can be assigned to particular Trunks, such as special or private trunk lines, trunk lines dedicated as
help lines or emergency trunks, or trunks designated as hotlines, so that when there are incoming calls on different
trunks at the same time, the call on these trunks gets answered first by the destination extension.
Priority can be assigned to all Trunk types (CO, Mobile, SIP, T1E1PRI, BRI) and Extension types (SLT, DKP, ISDN
terminal, SIP, E&M as Station, Radio, Magneto port).
2119
M e nu
Priority: 9
S at 0 1 0 5:3 0
TWT Trunk 1
Priority: 9
Incoming Call
at 10:00:10
TWT1
Redi al
ETERNITY
CO
Voi e
c M
a li Nam es
C A4
S IP 2
C A2
S IP 1
C A1
Rej ect
Hol d
ab c
3 de
j kl
6 mno
7 pq r s 8
t uv
9 w xyz
4
C LI R
gh i
H ot il ne C A3
Priority: 5
(Calls Operator at 10:00:00)
2001
Priority: 5
(Calls Operator at 10:00:05)
2002
Mobile Trunk 1
GSM
Priority: 7
Incoming Call
at 10:00:12
Priority: 7
(Calls Operator at 10:00:15)
Mobile 1
Me nu
DN D
rn c
e Transfer
M ute Conf ee
Voi c
e Ma li Names
Redi al
C A4
C A3
S IP 2
C A2
S IP 1
C A1
Rej ect
ab c
j kl
7 pq r s 8
tuv
4
C LI R
H ot il ne
Here,
S at 0 1 0 5: 3 0
gh i
Hol d
de f
6 m no
9
w xyz
There are two incoming calls, one on the Analog Trunk, CO 1 and the Mobile Trunk 1 at the same time.
Three extensions, 2001, 2002 and 3001 are calling the Operator. Extension 3001 has priority 7, while
extensions 2001 and 2002 have the same priority, 5.
Now, on the Operator extension, which is the landing destination, the incoming calls from the trunks and
the extensions will land in the following chronological order:
2120
Caller
Priority
SLT 2001
10:00:00
SLT 2002
10:00:05
CO Trunk
1
10:00:10
Mobile
Trunk 1
10:00:12
DKP 3001
10:00:15
These incoming calls, however, will appear on the Display of Operators phone (DKP or Extended IP
Phone) in the order of priority:
CO 1
Mobile Trunk 1
DKP 3001
SLT 2001
SLT 2002
Now, when the Operator goes Off-hook (pressing speaker key or picking up the handset), the call on CO 1
will be answered first, as CO Trunk 1 has the highest priority.
The Operator goes On-hook and then Off-hook, the call on Mobile Trunk 1 will be answered. Though
Mobile Trunk 1 and DKP 3001 have the same priority, 7, Mobile Trunk 1 will be answered first, following
the chronological order.
When the Operator goes On-hook after answering the call on Mobile Trunk 1, the call from DKP 3001 will
be placed on the Operator phone with a Priority Ring (configurable; default: Triple Ring).
When the Operator goes Off-hook, the call from DKP 3001 is answered.
When the Operator goes On-hook and then Off-hook after answering the call from DKP 3001, the call from
SLT 2001 will get answered first, though both 2001 and 2002 have the same priority, 5. In this case,
Priority Ring will not be played.
Thus, calls from trunks and extensions are answered by the landing destination in the order of priority.
Where priority is the same, calls are answered in chronological order. Calls from extensions with higher
priority are indicated by a Priority Ring on the landing destination.
Priority is relevant only when there is more than one call on the destination.
You can assign Priority to SLT extensions. However, Priority is not relevant when the SLT is a landing
destination, because there cannot be more than one call ringing on an SLT extension at a time.
An Intercom call will be placed at the destination only if the caller has a higher priority.
How to configure
To assign Priority to Trunks, you must set the priority in their Trunk Feature Template. See Configuring Trunks.
To assign Priority to Extensions, see instructions for configuring the respective Extension port type:
Configuring SLT Extensions
Configuring DKP Extensions
Configuring ISDN Terminals
Configuring SIP Extensions
Configuring E&M Lines
Configuring Radio Interface
Configuring Magneto Interface
Virtual Extension
If required, you may change the Ring Pattern of the Priority Ring. See Distinctive Rings for instructions.
2121
Privacy
Whats this?
Extensions of ETERNITY can be protected from the intrusions by other extensions or from trunk calls by activating
Privacy.
How it works
Intrusions can occur on an extension when another extension invokes the following features:
DND-Override
Interrupt Request (IR)
Barge-In
Raid
Intrusions can also occur,
When an external caller uses Auto Attendant to reach an extension.
When there is a call from another trunk line when you are in speech.
To prevent such intrusions, ETERNITY enables you to set the following types of Privacy:
Privacy from Interrupt Request, Barge-In, DND Override: This type of Privacy protects an extension
from intrusions by other extensions using Interrupt Request, Barge-In or DND Override.
For example: Extension A has Privacy from Interrupt Request, Barge-In and DND Override.
Extension A and B are in speech, Extension C attempts to intrude the conversation Interrupt Request or
Barge-In. Extension Cs call will be blocked and C will get error tone.
Now, Extension A has set DND and Extension B attempts to override it using DND Override. Since A has
Privacy from DND Override, Bs call will be blocked and B will get error tone.
Privacy from Raid: This type of Privacy protects an extension from intrusions by other extensions using
Raid.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, Extension C
uses Raid to intrude the conversation. Extension Cs call will be blocked and C will get error tone.
Privacy from Trunk call intrusion: This type of Privacy prevents the extensions in the Trunk Landing
Groups that are busy from being intruded by another waiting call. For this type of Privacy to work, the
feature Trunk Call Waitingmust be disabled on the extension.
For example: Extension A is the first trunk landing extension for calls on Trunk 1. Extension A and B are in
speech. A new call lands on Trunk 1. If A has Trunk Call Waiting beeps disabled, A will not hear the
intrusion beeps. The system will land the call on the next extension in the trunk landing group for calls on
Trunk 1.
2122
Privacy from Built-In Auto Attendant call: This type of Privacy protects the extension from being
accessed by external callers using Auto Attendant.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, external caller
C uses Built-In Auto Attendant to call extension A. Cs call will be blocked and C will get error tone.
How to configure
To provide Interrupt Request to extension users, you must enable this feature in the Class of Service (COS)
assigned to them for the time zones in their Station Basic Feature Template.
By default, Privacy from Raid is enabled in the Class of Service of all Extension types: SLT, DKP, SIP, ISDN. So,
none of the extensions can raid the other. You may disable this feature in the Class of Service of extensions, which
you want to protect from Raid.
By default, Privacy from Interrupt Request, Barge-In and DND Override are disabled in the Class of Service of all
Extension types. You may enable this feature on extensions which you want to protect from intrusions using any of
these features.
By default, Privacy from Built-In Auto Attendant is disabled on all Extension types. You may enable this feature on
extensions which you do not want external callers to reach.
By default, Trunk Call Waiting is disabled on all Extension types. You may keep this feature disabled on extensions
which you want to provide Privacy from Trunk Call intrusion beeps.
For instructions, see Class of Service (COS), Station Basic Feature Template. Also see,
Configuring SLT Extensions
Configuring DKP Extensions
Configuring ISDN Terminals
Configuring SIP Extensions
2123
QSIG
What's this?
Q-Signaling (QSIG) is an ISDN based protocol for signaling between two PBXs. QSIG is a protocol based
on internationally agreed Standards for ISDN.
You can network two or more ETERNITY using QSIG. This is known as 'Interoperability'.
The basic call procedure in the QSIG is implemented as per the ECMA-143.
The generic functional protocol for the Supplementary Services is implemented as per ECMA-165. Refer
relevant chapter for more details for the features explained in this chapter.
QSIG support is a licensed feature of ETERNITY. To use this feature you must purchase a License Key.
Refer the topic License Management to know more.
2124
Identification:
The Calling Line Identification Presentation (CLIP)
Calling Line Identification Restriction (CLIR)
The Connected Line Identification Presentation (COLP)
Connected Line Identification Restriction (COLR)
Name Identification:
The Calling Name Identification Presentation (CNIP)
The Connected Name Identification Presentation (CONP)
Call Diversion:
Call Forward Unconditional (CFU)
Call Forward On Busy (CFB)
Call Forward on No Reply (CFNR)
Call Completion:
Call Completion on Busy Subscriber (CCBS)
Call Completion on No Reply (CCNR)
Recall (RE)
The implementation of these features in QSIG is as per specific ECMA standards as described below.
Advice on Charge:
Using this feature, the caller can know the cost of the call made to public N/W using networked trunk. The
cost of the call will be determined by using Call Cost Calculation. Refer chapter Call Cost Calculation
(CCC)
The ETERNITY supports called 'AOC-E' End of the Call Charge Information. ETERNITY supports
charging for calls made to public network. The Cost of the call is calculated by the end PBX from which the
call is terminated to public Network.
CLIP/CLIR/COLP/COLR/CNIP/CONP
Refer chapters Calling Line Identification and Presentation (CLIP) and Calling Line Identification
Restriction (CLIR).
Call Diversion
Call Diversion is implemented as per ECMA-173 and ECMA-174. Refer chapters Call Forward and Call
Forward-Remote.
Call Completion (CC) features CCBS and CCNR are implemented as per standards ECMA-185 and
ECMA-186.
The CCBS is implemented as the feature Auto Call Back on Busy and the CCNR is implemented as the
feature Auto Call Back on No Reply. Refer chapter Auto Call Back (ACB).
DND is implemented as per standards ECMA-193 and ECMA-194. Refer chapters Do Not Disturb (DND).
2125
DND is implemented as per standards ECMA-193 and ECMA-194. See the topic Do Not Disturb (DND).
The Call Intrusion (CI) is implemented as Raid feature. However the extension on which intrusion is made,
will not get 'beeps'. On successful intrusion, the 'conference 3-party' type call will be established. See the
topic Raid and Conference-3 Party.
Call Transfer
Call Transfer is implemented as per standards ECMA-177 and ECMA-178. Refer chapter Call Transfer.
If 'CLIP-Hold' flag is enabled on the Transferring Station, the ETERNITY will send the held party's number
as calling number while placing call on QSIG.
If 'CLIP-Hold' flag is disabled on the Transferring Station, the ETERNITY will send the calling station's
number as calling number. Refer chapter Calling Line Identification and Presentation (CLIP) and Station
Advanced Feature Template for more details of CLIP-Hold.
Note 1: It is recommended to enable CLIP Hold flag for the operator extension, so that when operator
transfers the call to another PBX using QSIG, the terminating PBX (ETERNITY) can identify it as call
from Public network and treat as incoming call.
Note 2: As a Terminating PBX, the ETERNITY will consider the call as internal or from public network
depending upon the length of the digits for the calling number received.
Note 3: If Calling number is not received because of any reason, and CLIP- Hold flag is enabled, the
ETERNITY will send the 'Trunk Name' programmed for the trunk port as calling line identification.
Recall (RE)
Recall (RE) is implemented as per standards ECMA-213 and ECMA-214 and ECMA-143.
The ETERNITY supports Recall-Busy and Recall-No Answer features like ETERNITY features, Call
Transfer-On Busy and Call Transfer-While Ringing.
2126
Message Wait (MWI) is supported as per standards ECMA- 241 and ECMA-242.
Call Offer
TAC-0
Q-Sig
DS1-1
2103
2102
DS1-1
PBX-1
PSTN/ISDN
GSM/VoIP etc.
3103
PBX-2
3102
2101
3101
For simple application as shown in above figure, make settings as below for PBX 1 and PBX 2.
Now open Web Jeeves and make below settings in PBX 1 and PBX 2.
Settings at PBX 1:
Following basic settings are required in ETERNITY PBX 1.
Route Code
OGTBG
Strip Digit
Count
Self Router
Flag
Max. Dialed
Digits
001
21
01
Enable
04
002
31
01
Disable
04
003
01
Disable
04
2127
Rotation Flag
LCR Type
OGTB
Member 1
OGTB
Member 2
OGTB
Member 3
-------
OGTB
Member 8
01
Disable
None
01
00
00
-------
00
OG Trunk Bundle
Trunk Port
Template
Number
01
Port
Type
Port
No.
T1E1PRI
001
Start
Channel
Number
Total
Trunk
Count
Rotation
Type
01
030
Cyclic
Automatic Number
Translation (ANT)
Apply
Dialed
No. List
Substitute
No. List
Disable
05
06
Table ID
OG Ref. ID
Channel Count
ISDN Number
01
01
01
30
2100
01
Reference
ID
Start DDI
No.
Total DDI
Numbers
DDI No.
Digit Count
Port
Type
Port
Number
Start DDI
Flexible No.
001
01
2100
0100
FLEXNUM
000
2100
Settings at PBX 2:
Following basic settings are required in ETERNITY PBX 2.
2128
Route Code
OGTBG
001
21
02 Note
Disable
04
002
31
02 Note
Enable
04
Rotation
Flag
LCR
Type
OGTB
Member 1
OGTB
Member 2
OGTB
Member 3
-------
OGTB
Member 8
01
Disable
None
01
00
00
-------
00
02
Disable
None
02
00
00
-------
00
OG Trunk Bundle
Template
Number
Trunk Port
Start
Channel
no.
Total
Trunk
Count
Rotation
Type
Port
Type
Port
No.
01
Mobile
001
01
008
02
T1E1PRI
001
01
030
Automatic Number
Translation (ANT)
Apply
Dialed
No. List
Substitute
No. List
Cyclic
Disable
05
06
Cyclic
Disable
05
06
Table ID
OG Ref. ID
Channel Count
ISDN Number
01
01
01
30
3100
01
Reference
ID
Start DDI
No.
Total DDI
Numbers
DDI No.
Digit Count
Port
Type
Port
Number
Start DDI
Flexible No.
001
01
3100
0100
FLEXNUM
000
3100
Basic Call:
Below examples show various calls between both PBX.
Example 1:
To make call from station of PBX 1 to any station of PBX 2 or vice versa.
From station 2101 of PBX 1, dial station number 3101 or from station 3101 of PBX 2, dial station number
2101.
Example 2:
2129
From station 2101 of PBX 1, dial 0297 (TAC at PBX 2 for Trunk) and then dial 02652630555 (External Party
number).
Example 3:
Now External Party make IC call on Mobile port. IC call is answered by PBX 2. External Party gets Auto
Attendant Music. Now External Party dials station number.
Case 1: If External Party dials station number 3101 then call lands on 3101 of PBX 2.
Case 2: If External Party dials station number 2101 then call lands on 2101 of PBX 1.
Supplementary Services:
Working of Supplementary Services of QSIG in ETERNITY is explained as below.
Identification:
From station 2101 of PBX 1, dial station number 3101.
CLIP/CNIP:
Name and Number of station 2101 is displayed as CLI on station 3101.
COLP/CONP:
When 3101 of PBX 2 answers the call then Name and Number of station 3101 is displayed on station 2101
using reverse DDI method.
CLIR:
To restrict CLI at station 3101, below settings required for station 2101.
Here, Name and Number of station 2101 is not displayed on station 3101.
If "CLI Restriction (CLIR) Override" is enabled in Class of Service for station 3101, then Name and Number
of station 2101 is displayed on station 3101.
COLR:
To restrict CLI when IC call answered by 3101, below settings required for station 3101.
297. Here 2101 of PBX 1 does not get dial tone after dialing 0 (TAC at PBX 2 for Trunk) because 0 is programmed in CUG table for PBX
1.
298. Refer "c. Configuring for-IC Call using DDI Routing Over QSIG topic for how to route Incoming Call on T1E1PRI/SIP over QSIG
using DDI Routing.
2130
Here, Name and Number of station 3101 is not displayed on station 2101 when station 3101 answers the
IC call.
If "CLI Restriction (CLIR) Override" is enabled in Class of Service for station 2101, then Name and Number
of station 3101 is displayed on station 2101.
Call Diversion:
Station 2101 of PBX 1 wants to set Call Diversion (Call Forward) to station number 3101 of PBX 2.
Call Forward-Unconditional:
From station 2101 of PBX 1, dial 131-3101-#*. Now all IC calls on 2101 gets forwarded to 3101 of PBX 2.
Call Completion:
Station 2101 of PBX 1 wants to set CCBS/CCNR (ACB) to station number 3101 of PBX 2.
CCBS/CCNR:
From station 2101 of PBX 1, dial station number 3101 of PBX 2. Now dial Access code of ACB (Dial 2) to
set CCBS/CCNR on station 3101 of PBX 2.
Call Intrusion:
Station 2101 of PBX 1 and station 3101 of PBX 2 are in speech. Now another station wants to make Call
Intrusion on station 2101 of PBX 1.
Case 1: Station 2102 of PBX 1 wants to make Call Intrusion on station 2101 of PBX 1.
Set "Priority" level of station 2102 higher than "Priority" level of station 2101.
2131
Disable "Privacy from Raid" in Class of Service for station 2101 of PBX 1.
Also disable "Privacy from Raid" in Class of Service for station 3101 of PBX 2.
Now from station 2102, make call on 2101. After getting busy tone, dial Access code of Raid (Dial 5).
Speech will be established between all three parties 2101, 2102 and 3101.
If "Privacy from Raid" is enabled in Class of Service for station 2101 of PBX 1 or 3101 of PBX 2, then Raid
not possible in above case.
Case 2: Station 3102 of PBX 2 wants to make Call Intrusion on station 2101 of PBX 1.
Disable "Privacy from Raid" in Class of Service for station 2101 of PBX 1.
Also disable "Privacy from Raid" in Class of Service for station 3101 of PBX 2.
Now from station 3102 of PBX 2, make call on station number 2101 of PBX 1. After getting busy tone, dial
Access code of Raid (Dial 5). Speech will established between all three parties 2101, 3101 and 3102.
If "Privacy from Raid" is enabled in Class of Service for station 2101 of PBX 1 or 3101 of PBX 2, then Raid
not possible in above case.
Call Transfer:
Station 2101 and station 2102 of PBX 1 are in speech. Now station 2101 of PBX 1 wants to transfer the
call to any station number or external number of PBX 2.
From station 2101, dial Flash and then dial 3101 (Station number of PBX 2) or dial 0-02652630555
(External Number).
Now from station 2101, goes ON hook or press 'Transfer' key to transfer the call.
User can transfer the call after making speech with second party or when second party is 'Ringing' or
'Busy'.
Call Recall:
Station 2101 of PBX 1 transfers the call to station 3101 of PBX 2 when station 3101 of PBX 2 is 'Ringing' or
'Busy'. Now station 3101 does not answer the call.
Call gets return to station 2101 after expiry of 'Transfer on Busy Timer' for transfer- on busy case and
'Transfer while Ringing Timer' for Transfer- while ringing case.
Call Offer:
Station 2101 of PBX 1 and station 3101 of PBX 2 are in speech. Now station 2102 of PBX 1 wants to give
Call Offer to station 3101 of PBX 2.
2132
Allow "Interrupt Request" feature in Class of Service for station 2102 of PBX 1.
Now from 2102 of PBX 1, make call on station number 3101 of PBX 2. After getting busy tone, dial
Access code of Interrupt Request (Dial 3).
Station 3101 of PBX 2 gets beeps.
If "Privacy from Interrupt Request and Barge-In" is enabled in class of Service for station 3101 of PBX 2,
then Call Offer not possible from station 2102 of PBX 1.
Allow "Do Not Disturb" in Class of Service for station 2101 of PBX 1.
Now from station 2101, activate Do Not Disturb by dialing 181.
Now station 3101 of PBX 2 will not be able to make call on station number 2101 of PBX 1.
Allow "Do Not Disturb- Override" in Class of Service for station 3101 of PBX 2.
Now station 3101 of PBX 2 can make call on station number 2101 of PBX 1.
If "Privacy from DND Override" is enabled in Class of Service for station 2101, then station 3101 of PBX 2
can not make call on station number 2101 of PBX 1.
After completion of call, PBX 2 gives Cost of the call using AOC.
Cost of call in AOC is calculated at PBX 2 as per parameters set in Web pages for 'Call Cost Calculation' at
PBX 2.
If PBX 2 does not give any charge in AOC then no charge applies on station 2101 of PBX 1.
All calls between two stations of both PBX are free of charge.
Cost of the call is given as per "Call Toggle Flag" and "Originating Flag" set in SMDR-Outgoing Calls at
PBX 2.
If "Call Toggle Flag" is set to 'Split' at PBX 2, then charge before the call transfer is applied to station 2101
and charge after the call transfer is applied to station 2102.
2133
If "Split Flag" is set to 'Don't Split' at PBX 2, then charge is applied as per "Originating Flag".
If "Originating Flag" is set to 'Originating' then all charges before and after the call transfer is applied on
station 2101 who has transferred the call.
If "Originating Flag" is set to 'Terminating' then all charges before and after the call transfer is applied on
station 2102.
Open "Station Advanced Feature Template" page from Jeeves of PINX2 and program "MW Notification
Type" as Stuttered Dial Tone".
From station 2101, dial Access Code "1076 - 3101-1", to set Message wait on station 3101 of PINX2.
Now Message wait is set on station 3101. When user goes off hook from station 3101, he will get stuttered
dial tone.
Same way Message wait can also be canceled for 3101 by dialing access code "1076-3101-0".
0265-304XXXX
Q-Sig
2199
DS1-1
ISDN/VoIP
DS1-1
PBX-1
PBX-2
2100
3199
3100
2134
T1E1PRI trunk at PBX 2 has ISDN Number with MSN number 0265304 and DDI numbers are 2100-2199
and 3100-3199.
PBX 1 and PBX 2 are connected with each other using QSIG.
If Calling Party number is received as 0265304-2100 to 0265304-2199 then it should route on stations of
PBX 1. If Calling Party number is received as 0265304-3100 to 0265304-3199 then it should route on
stations of PBX 2.
Settings at PBX 1:
Settings for PBX 1 are given in Configuration of "Basic Call" and "Supplementary Services". No need to
change in these settings.
Settings at PBX 2:
Only make following settings in ETERNITY PBX 2.
OG Trunk Bundle
Trunk Port
Template
No.
02
Port
Type
Port
No.
T1E1PRI
001
Start
Channel
no.
Total
Trunk
Count
Rotation
Type
01
030
Cyclic
Automatic Number
Translation (ANT)
Apply
Dialed
No. List
Substitute
No. List
Enable
05
06
001
026530421
21
Program Incoming (IC) Reference ID for all Time Zones as "01" in T1E1PRI Port Parameters.
Table
ID
IC
Ref.
ID
Start
Channel
No
Total
Channel
Count
DDI
Routing
Ref. ID
Route
on First
Destination
Ring
Timer
(Sec)
Route
on TLG
when
No
Reply
01
01
01
30
01
No
045
No
No
01
02
01
01
30
02
No
045
No
No
01
Route
on TLG
when
Busy
Trunk
Feature
Templates
Reference
ID
Start
DDI No.
Total DDI
Numbers
DDI No.
Digit
Count
Port
Type
Port
Number
Start DDI
Flexible No.
001
01
3100
0100
FLEXNUM
000
3100
002
02
2100
0100
ROUTGRP
009
Routing Group
Routing Group
Rotation Flag
Member 01
Member 02
Member 03
Member 04-32
09
Enable
OTBG-02
None
None
None
2135
Basic Call:
Below examples show various calls using DDI Routing.
Example 1: External Party makes call on 0265304-2101
Call lands directly on station 2101 of PBX 1 through QSIG.
Example 2: External Party makes call on 0265304-3101
Call lands on station 3101 of PBX 2.
For the extensions of the PBX you wish to assign voice mail, configure the following:
Extension Number: Enter the number assigned to the extensions of the remote PBX. The number
must be entered along with the Exchange ID (if applicable).
The Extension Number can be a maximum of 6 digits max. Valid Digits are 0-9, * ,# (*, # can be used
as fist digit only).
2136
Name: Assign a 'Name' to the extension. The name may be of the person using the extension number
of the remote PBX or it can be the name of a department.
You can program a name of a maximum of 18 alphanumeric characters. Default: Blank.
Personal Mailbox: Select the check box to assign Personal Mailbox to the extension user of the
remote PBX. By default, Personal Mailbox is not assigned to the extension in Enterprise mode.
In the Hotel mode, Personal mailbox is assigned to all the extension phones, by default.
Mailbox Size (min): You may increase or decrease the size of the personal mailbox assigned to the
extension of the remote PBX, by changing the default Mailbox size of 5 minutes. You may change the
mailbox size to any desired value from 00001 to 60000 minutes. Default: Enterprise mode: 5 minutes;
Hotel mode: 999 minutes.
Maximum Message Length (sec): You can define the length of each message (in seconds) callers
are to be allowed to record in the mailbox. You may change the maximum message length to any
desired value from 0001 to 9999 seconds. Default: Enterprise mode: 15 seconds; Hotel mode: 120
seconds.
The VMS card will stop recording the message of the callers if it exceeds the maximum message
length, and will store only that part of the message recorded within the maximum message length limit.
New Message Delivery option in Mailbox Full condition: When the personal mailbox is full, you may
select one of the following options for delivery of new messages:
Do not offer to record a message: The VMS will not allow the caller to record a message by
declining delivery of the message.
Deliver new message to General mailbox: The VMS will record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users of ETERNITY as well as
the extensions users of the remote PBX.
Overwrite old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS starts overwriting the oldest message first.
By default, Deliver to General mailbox is selected.
Play message details after delivery of message: After the extension user of the remote PBX has
finished listening to a message in the mailbox, you can also have the VMS play message details such
as Date and Time when the message was recorded, the callers number299, and the extension number
dialed by the caller300 to the extension user.
You may select from one of the following options to Play message details:
Never: The VMS will not play message details to the mailbox owner after playing the message.
Always: The VMS will play message details to the mailbox owner after playing each message.
299. The number of person who left the message in the mailbox.
300. The number of the extension user for whom the message is intended.
2137
On Demand: The VMS will play message details to the mailbox owner only when the mailbox
owner requests it. On completion of each message, the VMS will prompt the extension user to
press a digit for date and time stamp. When the mailbox owner presses the digit, the VMS will play
the message details.
Ask Password to access Mailbox: By default, access to the mailbox is password protected. The
User Password is required to access the mailbox. Whenever the mailbox owner accesses the
mailbox, the VMS will ask for the (user) password.
If you want to remove password protection, clear this check box.
Since a Mailbox can be accessed using the default User Password, 1111, extension users who are assigned
a mailbox are recommended to change their User Password to a unique 4 digit number to prevent
unauthorized access to their mailbox.
Allow Mailbox Management: Mailbox Management allows the extension user of the remote PBX to
change mailbox settingsrecord Extension Name for the mailbox, redirect messages from the
mailbox, delete all old messages from the mailbox, record greeting messages for the mailbox. By
default, Mailbox Management is enabled.
If you do not want to allow the extension user to change the mailbox settings, in Allow Mailbox
Management, select No.
To know more about this feature, see Mailbox Settings
Auto Delete Messages: Select the type of messages you want the VMS to automatically delete from
your mailbox. You can select All or Old. Default: None.
Days for Auto Delete Messages: Select the number of days after which you want the VMS to
automatically delete the messages in your mailbox. Default: 90 days.
Voice Mail Auto Attendant Features: This parameter is applicable only if you are using the VMS Auto
Attendant.
Abbreviated Name: When the VMS is used as Auto Attendant, the callers can be prompted to Dial
by Name the desired extension users instead of their extension numbers.
To allow callers to reach the extension users of the remote PBX using Dial By Name, abbreviate the
extension users name to three letters and enter it in this field. Default: Blank
Announce Name: If you want the VMS to announce the extension users name to the caller when
transferring the call to the extension, select the check box to enable Announce Name. By default,
Announce Name is disabled.
If you enable Announce Name, make sure you record the remote PBX extension users name on
the VMS. Refer Recording Extension Names for instructions.
2138
Call Transfer: Select the desired method for transferring the call answered by the VMS Auto Attendant
to the extension user of the remote PBX. You may select any of the following methods of call transfer
for each time zone, Working Hours (WH), Break Hours (BH) and Non-working Hours (NH):
None: When the caller dials the extension number, the VMS Auto Attendant will check if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
Blind: When the caller dials the extension number, the VMS Auto Attendant will transfer the call on
the extension without checking whether it is busy or free.
Wait for Ring: When the caller dials the extension number, the VMS Auto Attendant will wait for the
extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the extension,
if assigned, or take the caller back to the home node.
Wait for Answer: When the caller dials the extension number, the VMS Auto Attendant will transfer
the call when the extension answers (goes OFF-Hook).
If the extension does not answer301, the VMS Auto Attendant will transfer the call to the mailbox of
the extension, if assigned, or take the caller back to the home node.
Screen: The VMS Auto Attendant prompts the caller to record his/her name. It puts the caller on
hold and places the call on the desired extension. If the extension is free and answers the call, the
VMS announces the callers name to the extension user and prompts the extension user to choose
whether or not to speak to the caller. If the extension user chooses to talk, the VMS transfers the
call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the extension
user, if assigned, and asks the caller to leave a message.
By default, Wait for Answer is selected as Call Transfer method for all time zones.
Message Wait Notification via Call: The message wait notification will be sent to a number
(destination number). This number can be an internal or an external number. To use this feature,
configure the following parameters:
Type: If you want the notifications to be sent as soon as a new message arrives in the mailbox of
the remote PBX extension user, select Immediate.
If you want the notification to be sent at fixed time schedules, select Scheduled.
If you do not want to set message wait notification via call, select None. Default: None.
Profile: Assign the Profile according to which you want the system to send the notifications. The
Message Wait Notification Profile determines how notification calls are to be made to the
destination numbers.
Destination Number: Enter the number on which you want the system to send the notification
calls.
The destination number can be an internal or an external number. The destination number can be a
maximum of 16 digits. Valid digits are 0 to 9, # and *.
301. The VMS will wait for the duration of the Built-In Auto Attendant Ring Timer (default: 30 seconds; the timer is configurable). If the
call is not answered before this timer expires, it is treated as No Reply.
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When the notification call is answered, the VMS informs the callee of the new message and allows
the callee to access it.
Refer the feature description Message Wait Notification via Call to know more.
Message Wait Notification via E-mail: The message wait notification will be sent to the e-mail
address of the extension user. To use this feature, configure the following parameters:
Notification: If you want the message wait notification to be mailed to the extension user along with
the new voice message as attachment, select the option Send With Attachment.
If you want only the notification to be mailed, select the option Send Without Attachment.
If do not want to set message wait notification via e-mail, select Do not send. Default: Do not send.
E-mail Address: Enter the e-mail ID of the extension user to which the notification is to be sent. Email ID may consist of up to 64 characters. Default: blank.
Extensions users will receive notifications only for the mailbox memory utilization, if you configure the E-mail
Address and select Do not sent as the Notification option.
Refer the feature description Email Based Notification to know more.
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Quick Dial
Whats this?
Quick Dial provides DKP and Extended IP phone users the facility of One-touch dialing of numbers stored in their
Personal Directory and the Global Directory.
How it works
Quick Dial is based on Abbreviated Dialing.
To be able to Quick Dial a number,
the number must exist in the Personal or Global Directory assigned to the extension.
Personal and Global Directory dialing must be allowed in the Class of Service of the extension.
On the DKP and Extended IP Phones, DSS keys must be configured with the Short Codes or Abbreviated
Numbers that are to be dialed out. These short codes are derived from the Index numbers of the Personal
Directory and the Memory Location Index of the Global Directory.
You can Quick Dial a number simply by pressing the DSS key.
The system locates the number to be dialed out in the Personal/Global Directory on the basis of the Index
Number/Memory Location Index configured on the DSS Key.
How to configure
See Abbreviated Dialing for instructions on configuring and assigning the Personal and Global Directories.
To assign the Short Codes or Abbreviated Numbers to be used for Quick Dial on DSS keys, for each DKP/
Extended IP Phone extension,
List down the numbers from the Personal Directory and Global Directory to be used for Quick Dial.
If the number is from the Personal Directory assigned to the extension, note the Index number at which it is
stored in the Personal Directory: 001 to 025.
If the number is from the Global Directory assigned to the extension, note the Memory Location Index at
which it is stored in the Global Directory: 100 to 999.
Now, configure the Quick Dial numbers on the DSS keys of the DKP and Extended IP Phone.
For detailed instructions on configuring DSS Keys on a Digital Key phone, see DSS Keys Programming.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see Configuring SIP Extension
Settings as per the Extended Phone Type under Configuring SIP Extensions.
2141
As Offset, select the Index Number against which the number is stored in the Personal/Global Directory.
How to use
For EON and Extended IP Phone Users only
2142
Raid
Whats this?
Raid allows you to interrupt a telephone conversation between two extension users, turning the conversation into a
three-way call.
You can use Raid to land in a conversation between two extension users, and between an extension user and an
external caller, with a warning beep to the extension user. The extension user will hear a beep when you raid and
you will enter in to three-way speech with both parties.
You may also Raid a conversation without any warning by disabling the beep.
How it works
C calls A.
If any of these three parties disconnects, two-way speech is established between the remaining parties.
Feature Interactions
Raid works only if the dialed extension is busy in two-way speech. The two-way speech may be with
another extension or with an external number on a trunk.
You cannot Raid on Trunks, that is, the external number which is in two-way speech with an extension. In
this case, C can raid the conversation between A and B, but not between A and another external number.
Raid will not work if Privacy against Raid is enabled in the Class of Service of the extension being raided.
In this case, if Extension A has Privacy against Raid in its Class of Service, C will not be able to Raid the
conversation between A and B. To know more about this feature, see Privacy.
The extension using Raid must have higher Priority assigned to it than the extension being raided. In this
case, C must have higher Priority than A to be able to invoke Raid.
Raid is a sensitive feature. You are advised to restrict access to this feature to select extension users.
2143
How to configure
To be able to use Raid, extension users must have this feature enabled in the Class of Service (COS) assigned to
them for the time zones in their Station Basic Feature Template.
By default, beep is played as a warning to the extension being raided. If required, you may disable the beep played
during Raid, by clearing the Play Beep when Raid/Conference/Dial-In Conference begins check box in the
System Parameters. For instructions, see System Parameters.
How to use
For EON & Extended IP Phone Users
When dialed extension is busy,
2144
How it works
The Prerequisites
RCOC is enabled on the desired Trunk/s - BRI, T1E1PRI, Mobile and SIP.
RCOC is enabled in the Class of Service group assigned to the extension.
The Process
When an extension having RCOC feature in its Class of Service makes an out going call, the system
checks if RCOC is enabled on the trunk through which the outgoing call is routed.
If RCOC is enabled on the trunk, the system stores the record of the outgoing call in an internal database
referred to as the RCOC Table.
The system sets RCOC for the outgoing call in the following conditions, according to the Destination
Port302:
2145
If the Destination Port is a BRI or T1E1PRI (DS1) Port or a SIP Trunk, RCOC is set when:
If the Originating Port is either BRI-NT or T1E1-NT, the system checks the Class of Service allowed
to the trunk port.
If the Originating Port is a trunk (T1E1-TE, BRI-TE, CO, MOBILE, SIP), RCOC will be set, if it is
enabled on Destination Port
RCOC shall be set only if the Calling Party's Number is available. If calling party number is missing,
then RCOC shall not be set.
Whenever there is an incoming call on any trunk, the system matches the CLI of the incoming all with the
RCOC Table.
If a matching record entry is found, the system routes the call to the original caller and clears the record
entry from the RCOC Table.
The return call rings on the original caller's extension for the period of the Ring Back Tone Timer
(programmable; default 45 seconds). If the original caller does not answer the call within this Timer, the call
is routed to the Trunk Landing Group programmed for that trunk.
If no match is found in the RCOC Table or the extension or the original caller is busy, the call will be routed
according to the incoming call logic programmed (as programmed in the assigned Trunk Feature
Template) in the system.
When DISA CLI Authentication (Multiple Calls or One Call) is enabled on a trunk, whenever there is an
incoming call on the trunk, the system will first check the DISA CLI Authentication Table.
If a matching entry is found in the DISA CLI Authentication table, the system will give dial tone to the caller.
The caller can now invoke RCOC feature by dialing ** (pressing Star key twice).
OR
The caller can make calls to a station or an external number or use a feature as required.
If the caller invokes RCOC feature by dialing ** (pressing Star key twice), the system will check the RCOC
Table.
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If a matching record entry is found, the system routes the call to the original caller and clears the record
entry from the RCOC Table.
As RCOC is a Class of Service (COS) based feature, extensions that are not allowed this feature in their
COS cannot have their calls returned; even if this feature is enabled on the Trunk they used to make the
call.
The record of the outgoing call is stored in the RCOC Table, only if the same record does not already exist
in the database.
Each entry is kept for the duration of the RCOC Record Delete Timer (programmable; default: 999
minutes). Whenever a record is stored in the RCOC database, the Record Delete Timer for that entry is
activated. On the expiry of the Timer, the entry is deleted by the system.
Each record is deleted from the database either after the call is returned or on expiry of the Record Delete
Timer.
In case of Call Transfer, RCOC will be set for the extension which made the call on the trunk.
The Ring Back Tone Timer is common to all internal calls; calls made from one extension will ring on the
destination extension till the end of this timer. Change in the Ring Back Tone Timer for RCOC returned
calls on original caller's extension will also be applied on Ring Back Tone Timer for all internal calls. So,
program this Timer taking this into consideration.
Persons using DISA must be informed about RCOC feature access code ** and how to use this feature
when in DISA mode.
How to configure
For this feature to work, it must be enabled on the Trunk and in the Class of Service of the extensions. If desired,
the related Timers, that is, the RCOC Record Delete Timer and the Ring Back Tone Timer may also be changed.
RCOC on Trunk
Enabling RCOC on Trunk using Jeeves
Click the trunk parameters of the trunk type on which you want to enable this feature, namely:
SIP Parameters
BRI Parameters
2147
Select the 'Return Call to Original Caller (RCOC)' check box on the page to enable this feature on the
desired trunk port.
2148
Exit SE mode.
disabled.
Refer the topic Class of Service (COS) and Station Basic Feature Template for instructions.
2149
How to Configure
Configuring RTC using Jeeves
2150
The current Day will appear as per the date you set.
Exit SE mode.
Time Zones
Index
Time Zone
001
Afghanistan (GMT+04:30)
002
Algeria (GMT+01:00)
003
004
Argentina (GMT-03:00)
005
006
007
2151
Index
Time Zone
008
Austria (GMT+01:00)
009
Bahamas (GMT-05:00)
010
Bahrain (GMT+03:00)
011
Bangladesh (GMT+06:00)
012
Belarus (GMT+02:00)
013
Belgium (GMT+01:00)
014
Bhutan (GMT+06:00)
015
Bolivia (GMT-04:00)
016
017
Botswana (GMT+02:00)
018
Brunei (GMT+08:00)
019
020
021
022
023
Bulgaria (GMT+02:00)
024
Cambodia (GMT+07:00)
025
Cameroon (GMT+01:00)
026
027
028
029
030
031
032
Chile (GMT-04:00)
033
China (GMT+08:00)
034
Colombia (GMT-05:00)
035
036
Croatia (GMT+01:00)
037
Cuba (GMT-05:00)
038
Cyprus (GMT+02:00)
039
040
Denmark (GMT+01:00)
041
Egypt(GMT+02:00)
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Index
Time Zone
042
Fiji(GMT+12:00)
043
Finland(GMT+02:00)
044
France(GMT+01:00)
045
Germany(GMT+01:00)
046
Greece(GMT+02:00)
047
Guyana(GMT-04:00)
048
Hong Kong(GMT+08:00)
049
Hungary(GMT+02:00)
050
India(GMT+05:30)
051
Indonesia(GMT+07:00)
052
Iran(GMT+03:30)
053
Iraq(GMT+03:00)
054
Ireland(GMT)
055
Israel (GMT+02:00)
056
Italy(GMT+01:00)
057
Japan(GMT+09:00)
058
Jordan(GMT+02:00)
059
Kazakhstan(GMT+06:00)
060
Kenya(GMT+03:00)
061
Korea - North(GMT+09:00)
062
Korea - South(GMT+09:00)
063
Kuwait(GMT+03:00)
064
Kyrgyzstan(GMT+06:00)
065
Lebanon(GMT+02:00)
066
Libya(GMT+02:00)
067
Malaysia(GMT+08:00)
068
Maldives(GMT+05:00)
069
Mauritius(GMT+04:00)
070
071
072
073
Mongolia(GMT+08:00)
074
Mozambique(GMT+02:00)
075
Myanmar(GMT+06:30)
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Index
Time Zone
076
Namibia(GMT+01:00)
077
Nepal(GMT+05:45)
078
Netherlands(GMT+01:00)
079
New Zealand(GMT+12:00)
080
Nigeria(GMT+01:00)
081
Norway(GMT+01:00)
082
Oman(GMT+04:00)
083
Pakistan(GMT+05:00)
084
Paraguay(GMT-04:00)
085
Peru(GMT-05:00)
086
Philippines(GMT+08:00)
087
Poland(GMT+01:00)
088
Portugal(GMT)
089
Qatar(GMT+03:00)
090
Romania(GMT+02:00)
091
092
093
094
Singapore(GMT+08:00)
095
Slovakia(GMT+01:00)
096
South Africa(GMT+02:00)
097
Spain(GMT+01:00)
098
099
Sudan(GMT+03:00)
100
Sweden(GMT+01:00)
101
Switzerland(GMT+01:00)
102
Syria(GMT+02:00)
103
Taiwan(GMT+08:00)
104
Tajikistan(GMT+05:00)
105
Thailand(GMT+07:00)
106
Turkey(GMT+02:00)
107
Uganda(GMT+03:00)
108
Ukraine(GMT+02:00)
109
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Index
Time Zone
110
United Kingdom(GMT)
111
112
United States (Chicago, Dallas, Des Moines, Memphis, Minneapolis, New Orleans,
Oklahoma, Omaha, St. Louis) (GMT-06:00)
113
United States (Albuquerque, Boise, Cheyenne, Denver, Salt Lake City) (GMT-07:00)
114
United States (Las Vegas, Los Angeles, Phoenix, San Francisco, Seattle) (GMT08:00)
115
116
117
Uzbekistan(GMT+05:00)
118
Venezuela(GMT-04:30)
119
Vietnam(GMT+07:00)
120
Yemen(GMT+03:00)
121
Yugoslavia(GMT+02:00)
122
Zambia(GMT+02:00)
123
Zimbabwe(GMT+02:00)
124
2155
Reminder
What's this?
Reminders are a variation of the Alarms feature, requiring the Date and Time to be set for each Reminder call.
Reminder calls are useful for extension users who wish to be reminded of important tasks or appointments.
For Reminder calls, date and time are set in the following format:
Date is set, according to Date Format you selected in the Real Time Clock (RTC) parameters, as:
Day-Month-Year (DD:MM:YYYY)
Or
Month-Date-Year (MM:DD:YYYY).
Multiple Reminders can be set for an extension by the Operator and/or by the extension user.
The mechanism for serving Reminders calls can be configured as 'Personalized' or 'Automated'.
Reminders can be voice-guided, if the ETERNITY has a Voice Mail System (VMS) installed in it.
ETERNITY can register as many as 48 Reminders set by the Operator and extension users.
How it works
Personalized Reminder
When the Reminder call serving mechanism is configured as 'Personalized',
The Operator Phone rings first303, displaying the number of the extension to which the reminder call is to
be served.
When the Operator answers this call, a call is placed on the extension on which the reminder call is set.
The extension phone rings for the duration of the Alarm Ring Timer.
When the extension user answers the call, the Operator greets the extension user with the reminder
message.
303. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the ETERNITY will
make two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.
2156
If the extension user does not answer the call till the Alarm Ring Timer has elapsed, the Operator phone
will display a text message notifying 'No Reply' from the extension. The Reminder is now considered as
served.
If the extension is busy304, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
inform the extension user about the Reminder in person or send someone to do it.
OR
Personal Reminders will work even if the extension user has set DND or Call Forward.
Automated Reminder
When the Alarm serving mechanism is configured as 'Automated',
The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is a
DKP or the Matrix Extended IP Phone, Reminder message will appear on its display.
When the extension user answers the call, the user may be played music-on-hold, or a pre-recorded voice
message, or be connected to the Voice Mail, or routed to the Operator, depending upon the Alarm
Notification Type you have configured for the extension.
If the extension user does not answer the reminder call, the ETERNITY makes two more attempts (in all, 3
attempts) at an interval of 5 minutes between each attempt, to call the extension.
If all Reminder call attempts go unanswered, the ETERNITY places the call on the Operator Phone. The
Operator Phone rings till the end of the Alarm Ring Timer. The Operator Phone displays the number of the
extension with the message 'No Reply'. The Reminder call is now considered as served.
If the extension phone is busy, the ETERNITY will continue to make the Reminder call Attempts at the
Alarm Interval programmed. When all Alarm Attempts go unanswered, ETERNITY will place a call on the
Operator phone. The Operator Phone will display the number of the extension phone with the message
'Busy'.
304. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'.
2157
Snooze
The Snooze function can be added to Automated Reminders to ensure that the extension user answers the call.
Snooze is a system-wide feature; when set, this function will be added to all Automated Reminder calls.
When Snooze is activated,
The extension phone rings for the Number of Alarm Attempt configured, at the set Alarm Attempt
Interval.
The extension stops ringing when the user answers the call and dials 0 to acknowledge the Reminder
call. This reminder call Acknowledgement Code 0 is non-configurable.
Reminder settings will be retained in the system during power down and system upgrades.
When multiple reminder requests have been set by an extension user, the extension user cannot
selectively cancel a particular reminder request. Only the Operator can selectively cancel Reminders
set for an extension user from the System Administrator pages of Jeeves.
It is not possible to modifychange the date and timeof a reminder request. So, you may cancel the
Reminder request and set a new one.
How to configure
The configuration of Reminders is the same as Alarms.
To configure Reminders feature, do the following:
Configure, as required, the Alarm Call related parameters: Alarm Ring Timer, Number of Attempts,
Alarm Attempt Interval, Configurable Alarm Type and Configurable Alarm Category, and Snooze.
Configure Macros, if the SLT extension has special function keys, and you want to a function key for the
Reminder feature.
For instructions, see the topic How to configure under Alarms.
2158
How to use
Reminders can be set by the extension users by themselves. The extension users can also ask the Operator to set
the Reminder for them.
If the Voice Mail System (VMS) is installed in the ETERNITY, it can offer voice-guided Reminders to extension
users and the Operator.
Voice-guided reminders lead users through a menu, helping them set the alarm in a step-by-step manner.
For SLT
If the SLT of the extension user has a special Reminder function key, the extension user can set the alarm using
this key.
Press 'Reminders' key. (The label on the SLT key may differ from model to model)
Follow the Voice Mail System prompts to set/cancel reminders.
2159
SLTs with special function keys will work only if the corresponding Macros are programmed by the
System Engineer.
Without the Voice Mail System installed, the extension user having SLT with the special Reminder
function key will not be able to set/cancel Reminders. This extension user can set/cancel Reminders
only by dialing the feature access code for voice-guided Reminders.
For SLT
To set Reminder for an extension,
2160
Lift handset.
Dial 1072-033
Dial Extension Number.
Dial Date and Time in the format:
DDMMYYYYHHMM
OR
MMDDYYYYHHMM (users in USA)
Dial 1 for Personalized, Dial 2 for Automated.
You get confirmation tone.
Replace handset.
Lift handset.
Dial 1072-033
Dial Extension Number.
Dial #
You get confirmation tone.
Replace handset.
To cancel reminder calls selectively, go to 'Reminder Status' page from the System Administrator of
Jeeves.
To cancel Reminder:
Press 'Reminder' Key again.
OR
Dial 162
Select 'Cancel All'.
Press Enter Key.
Lift handset.
Dial 162
Dial Date and Time in the format
DDMMYYYYHHMM
OR
MMDDYYYYHHMM (users in USA)
You get confirmation tone.
Replace handset.
2161
To cancel Reminder,
2162
Open Jeeves.
Select the Cancel Reminder check box of the extension number for which you want to cancel the
reminder.
Click the Cancel Selected Reminders button at the bottom of the page.
2163
Reminder Report
AS ON 10-10-2013(Thu) AT 23:50
------------------------------------------------------------------------------Room#
Phone# Reminder
P
Room#
Phone# Reminder
P
------------------------------------------------------------------------------3001
09-10-2013 12:25
3001
09-10-2013 12:14
3001
10-10-2013 17:11
------------------------------------------------------------------------------+ indicates Personal Reminder
Page : 1
---End of Report---
2164
If the date format of ETERNITY is set as MM-DD-YYYY or the Region 'USA' is selected, then the
reminder report will be printed according to this date format. To know more, see Real Time Clock
(RTC).
Remote Programming
Whats this?
One can program ETERNITY from any remote location. Direct inward system Access (DISA) facility of the
system allows a remote user to login and use most of the functions of the system. Programming is one of
such functions allowed to the remote user. The remote user can program the system using the same
commands as used by the normal local station to program the system. The user can login into the SA
programming or SE programming mode.
How to use
Login as DISA user. Get the DISA login beeps. Enter the SE/SA mode. Get programming/Dial Tone.
In case none of the trunk lines have DISA facility and it is required to do remote programming, then follow
following steps:
SE Mode
Dial 1#91-SE Password program the system. Dial 00 exit from the programming mode (SE Mode).
When you have logged into SE Mode from DISA, to program #, you need to press # four times.
SA Mode
Dial 1#92-SA Password program the system. Dial 1#92 exit from SA mode.
Once the user is out of SE/SA Mode the user gets DISA beeps.
2165
How it works
Following flow chart depicts the process:
Start
Is the
password
correct ?
No
Error Tone
Yes
Yes
Confirmation tone
Is the
command dialed
correct and accepted
by the system ?
Exit
Program
Mode
System gives dial tone
No
Error tone
Wait for error
tone to get over
End
Relevant Topic:
1. Direct Inward System Access (DISA)
2166
1862
Room Monitor
Whats this?
This feature enables the DKP and Extended IP Phone Extension users to listen to the conversations taking place in
another location where a DKP/Extended IP Phone is present.
Room Monitor can be used to monitor activities on the Shop Floors / Manufacturing areas from another location.
How it works
As room in on the second floor. The manufacturing area is on the ground floor.
To keep track of the activities in the plant on the ground floor, there must be a DKP or an Extended IP
Phone at the place where the activities are to be monitored, and As extension must have higher Priority
than the extension at the monitored location.
If there is a DKP or an Extended IP Phone at the desired location, A can activate Room Monitor.
A can activate Room Monitor only if the DKP/Extended IP Phone at the desired location is idle.
When A activates Room Monitor, the microphone of the DKP/Extended IP Phone on the ground floor goes
Off-hook. A can now hear all the sounds taking place on the ground floor, without anyone present there
coming to know that they are being monitored.
Room monitoring will be terminated on the DKP/Extended IP Phone on the ground floor, if someone lifts
the handset of this phone or if there is a call on this phone from another extension.
You can activate Room Monitor from any extension port type, but the extension being monitored must
be a DKP or an Extended IP Phone.
How to configure
To be able to use Room Monitor, extension users must have this feature enabled in the Class of Service (COS) in
the Station Basic Feature Template assigned to their extensions.
2167
How to use
For EON & Extended IP Phone Users
To enable Room Monitor on an extension,
2168
Routing Group
Whats this?
Routing Group is a group of extensions used for landing incoming calls as a Trunk Landing Group, as Alarm
Notification Group, as Floor Service Group and as Department Group.
How it works
ETERNITY supports the formation of 96 Routing Groups. In each group you can have upto 32 members.
The member of a Routing Group can be Single Line Telephones (SLT), Digital Key Phones (DKP), SIP Extensions,
ISDN Terminals, Virtual Extensions, Voice Mail Auto Attendant Profile and Outgoing Trunk Bundle Group.
These groups can be used :
as Trunk Landing Groups to route incoming calls.
as Alarm Notification Groups to server Alarm Notifications.
as Floor Service Groups to provide Floor Service.
as Department Groups to route incoming calls to a particular department.
This is how a Routing Group works,
Routing Group 1 is assigned as the Trunk Landing Group for CO1. The Routing Group has DKP 2001,
2002 and 2003 as landing destinations.
By default incoming calls will be placed on the members in rotation, that is first call on DKP 2001, second
call on 2002 and so on.
If you want incoming calls to be placed on DKP 2001 always, you must disable Rotation.
By default, an incoming call will be placed on 2001. 2001 rings for the duration of the Ring Timer, if the call
is unanswered the system re-directs the call to 2002 and so on, till the call is answered.
If you want all the extensions to ring continuously till the call is answered by any member, you must enable
Continuous Ring. 2001 will continue to ring even as the system hunts for other extensions in the routing
group to land the call. If the call still remains unanswered, the system will return the call to 2001 once
again.
In this way the system places the incoming calls on the member extensions in a Routing Group till the call
is answered.
If you have selected Voice Mail Auto Attendant Profile or OGTBG as members in a Routing Group, the
parameters Ring Timer and Continuous Ring are not applicable.
How to configure
Configuring Routing Groups using Jeeves
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Choose the Routing Group number (01-96) you want to use as routing group. In each routing group you
can program maximum 32 'members'.
Now program the following parameters for the selected routing group:
Name: You can assign a Name to the department group to facilitate identification. This name will
appear in the Dial by Name directory along with the department group number. The Name can be a
maximum of 18 characters.
Rotation: Select the Rotation check box to enable rotation of calls in the routing group which has
multiple 'member' extensions. When enabled, each fresh call will land on the extension which is next to
the one that received the last call. This ensures equal distribution of incoming calls to all the
destinations within the routing group. The flag has no relevance if the routing group has only one
member extension.
Member Type: Select the 'Member Type'. You can select SLT, DKP, SIP, Virtual Extensions, ISDN
Terminal, OGTBG or the Voice Mail Auto Attendant.
Configure only as many extensions as you want in the routing group and set the remaining Member
Types to 'None'.
For example: if you want to program only one extension in the routing group, set the Member Type in
the remaining columns (Member 02-Member 32) to 'None.'
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Port Number: Enter the software port number on which the SLT/DKP/SIP/Virtual Extension/ISDN
Terminal is connected. If you select Voice Mail Auto Attendant Profile, enter the Profile number here. If
you have selected OTBG then enter the OTBG number here.
Ring Timer(s): This timer defines the time for which the extension, on which the call lands, should ring.
By default, the ring timer is set to 015 seconds and can be changed.
Continuous Ring: Select the Continuous Ring check box, if you want an extension to ring
continuously until the call is answered. The first extension will continue to ring even as the system
hunts for other extensions in the routing group to land the call. If the call still remains unanswered, the
system will return the call to the first extension once again. This parameter is not relevant, if there is
only one member extension in a routing group.
Matrix ETERNITY System Manual
To route incoming calls on a trunk, you must assign a Routing Group in the Trunk Landing Group in a
Trunk Feature Template assigned to the trunk.
To assign a Routing Group as a Trunk Landing Group, under Configuring Trunks, see Trunk Feature
Template.
To assign a Routing Group as a Alarm Notification Group, under Alarms, see How to configure.
Meaning
Port Number
00
Nonea
000
01
SLT
001-512
02
DKP
001-128
16
001-025
28
ISDN Terminal
01-64
34
SIP Extension
001-500
36
Virtual Extension
01-64
41
01-16
2171
Meaning
Meaning
Fresh call lands on the first station within the group (disable continuous)
Port Type
Port Number
Continuous Ring
Rotation
01
DKP
001
015
OFF
ON
02
SLT
001
015
OFF
ON
03
SLT
002
015
OFF
ON
04-32
None
000
015
OFF
ON
2172
Exit SE mode.
Security Alarm
Security Alarm makes use of the Digital Input Port (DIP) to function.
A panic switch, a smoke detector or a break-in detector can activate the DIP. The system will sense the event and
will place a call to the numbers programmed to receive security alarm calls. These numbers may be:
The system will play a pre-recorded voice message to inform the called parties (external numbers or group of
extensions) of the emergency or activate the siren connected on the DOP as programmed.
It is possible to select a different destination for Security Alarm calls according to Time Zones, that is, working
hours, break hours, and non-working hours.
For example, factory manager can have the extension of Security Personnel programmed for Security Alarm during
working hours and break hours, and can have an external number, such as the factory managers residence phone
number, programmed for Security Alarm for non-working hours.
Security Reporting
It is possible to program the system to 'report' Security Alarm calls made to external numbers simultaneously also
to a group of extensions. Hence, the feature name Security Reporting.
The system will display the DIP Port number with the Text message 'Emergency' to the extensions on which it
lands the call. The extension user can know the location of the emergency if s/he knows the location of the sensor/
panic switch connected to the DIP Port Number displayed on his/her phone.
How it works
The Pre-requisites
An emergency switch/sensor, output of a Smoke detector, Glass break detector, Fire Alarm, etc., is
connected as instigator to the Digital Input Port (DIP) of ETERNITY.
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The Digital Input Port parameters306 are programmed. Refer the topic Digital Input Port (DIP) for
programming instructions.
A Time Tables defining the working hours, break and non-working hours is assigned as the Time Table
for Security Alarm.
For each Time Zone, the destination for Security Alarm calls is selected as DOP or External Numbers, or
Routing Group and the corresponding DOP/External Number/Routing Group of extensions is assigned.
The Security Reporting flag is enabled in the Security Alarm Parameters, to report the emergency with the
prerecorded voice message to a group of extensions.
Security Alarm - Delay Response Timer: the time of initiating the dialing after the Security Alarm is
triggered by the DIP.
Call Attempt Interval for External Number: the time gap between each attempt to call an external
number.
Number of Attempts for each External Number: the number of attempts the system should make to
dial each external number you have programmed (in case no acknowledgement is received from the
number).
The Process
The system waits for the duration of the 'Minimum Instigation Time' you programmed (default: 01 sec) to
respond to the instigation.
On expiry of this timer, the system waits for the Security Alarm - Delay Response Timer (programmable;
default: 15 seconds), while checking the 'Trigger Security Alarm on' destination programmed for the
current Time Zone.
On detecting 'External Number' as the destination for Trigger Security Alarm on, the system checks for the
first external number programmed in the for the current Time Zone.
At the end of the Delay Response Timer, it places a call to the first external number.
When Security Reporting flag is enabled it also simultaneously places a call to the first extension number
in the Security Reporting-Routing Group.
306.Enable the port, set the Instigation Signal and the Minimum Instigation Time as required.
307.The pre-recorded message provided by Matrix (on the Product CD) is: "This is an emergency call. Please dial '0' to acknowledge".
Refer the topic Voice Message Applications.
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If the Reporting group extension is a digital key phone, the system will display the text message <<DIP-1
Emergency>> on the phone display.
When the external called party answers the call, the ETERNITY delivers a pre-recorded Emergency
message (recorded in the Voice Module).
The system repeats the voice message until the external and the internal called parties dial '0' to
acknowledge the call.
If the internal called party number is as a digital key phone, the system will display the text message for
Security Alarm Acknowledgement.
When there is no response from the first external number, the ETERNITY dials the second external
number programmed for the current Time Zone.
If there is no response from the second external number or if the number is busy, the system tries the third
external number programmed for the current Time Zone and tries to deliver the message.
Thus, the system tries each number, one after the other, as many times as programmed in the Number of
Attempts (default: 5 attempts) with a time interval programmed in the Call Attempt Interval (default: 15
seconds).
The system stops dialing each external number only when the called party acknowledges the call by
dialing '0' or when the Number of Attempts is over.
On detecting 'Routing Group' as the destination for Trigger Security Alarm on, the system places the call
on the first extension number in the Routing Group programmed for the current Time Zone.
When Security Reporting flag is enabled, the system also simultaneously places a call to the first
extension number in the Security Reporting-Routing Group.
If the Reporting group extension is a digital key phone, the system will display the text message <<DIP-1
Emergency>> on the phone display.
When any of the Security Alarm-Routing Group extensions answers the call, the ETERNITY delivers the
pre-recorded Security Alarm message (recorded in the Voice Module).
It repeats the voice message until the called extension dials '0' to acknowledge the call.
If there is no response from the first Security Reporting-Routing Group extension number, the system will
follow the Routing Group Logic to try other extensions in the group to land the Security Alarm call.
The system stops attempting the Security Alarm call, when any of the extensions in the Security AlarmRouting Group answers the call and acknowledges the call by dialing '0'.
If the Security Alarm-Routing Group extension is a digital key phone, the system will display the text
message for Security Alarm Acknowledgement.
The Security Alarm 'Number of Attempts' count and the 'Call Attempt Interval' are not applicable for
routing Security Alarm calls to internal numbers (Routing Group).
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Time Zones are not applicable for Security Alarm Reporting. Regardless of the current Time Zone, the
Security Alarm call will be reported to the programmed Reporting-Routing Group.
How to configure
For this feature to work, you must program the Security Alarm parameters and the Digital Input Port.
Before you set the Security Alarm Parameters, ensure that you have the following programmed and ready:
The sensor device connected to the Digital Input Port and the port parameters programmed. Refer the
topic Digital Input Port (DIP) for instructions.
A voice message containing an Emergency Message is recorded in the Voice Module for Security and
Emergency. Refer the topic Voice Message Applications for instructions on recording voice module.
Routing Groups of stations, that is, Security Alarm-Routing Group, on which Security Alarm should be
triggered, if you have selected as 'Routing Group' for any of the Time Zones.
A Routing Group of stations, that is, the Security Reporting-Routing Group, on which security reporting
should initiated.
The Time Table for Security Alarm, defining the working hours, non-working hours and break hours for the
Time Zones of the Response Type selected. Refer the topic Time Tables for programming instructions.
Now, program the Security Alarm Parameters using Jeeves or by dialing SE commands from a Telephone.
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Use DIP to trigger Security Alarm: Enable this flag to use the Security Alarm feature. If this flag is
disabled, no alarm will be triggered. By default the flag is disabled.
You may program the Digital Input Port, if not done already. To do this,
Set the Instigation Signal type as 'High308' or 'Low309' according to sensor device connected to the
DIP. By default, 'High' is selected.
Set the Minimum Instigation Time310 to the desired value. By default, it is set to 01 second.
308.'High' state signifies that the DIP is normally open. DIP should be programmed as 'High' when the sensor connected to the DIP
keeps the Loop open and closes it to signal an event.
309.'Low' state signifies that the DIP is normally closed. DIP should be programmed as 'Low' when the sensor connected to the DIP normally keeps the Loop closed and opens/breaks it to signal an event.
310.This is the time for which the instigation signal from the sensor device should remain present on the DIP to be recognized by the DIP
as a genuine signal.
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Whether you are using the DIP for an automated control application or for Security Dialing and Reporting, you must
program the following parameters:
Time Table for Security Alarm: Select the Time Table you want to apply for Security Alarm. By default,
Time Table 1 is selected.
You may program the time table, if not done already. To do this,
Click the link Time Table in this parameter.
The Time Table page will open.
Configure the default Time Table, or another Time Table.
Click Submit at the bottom of the page to save changes.
Return to the Security Alarm page.
Select the Time Table you programmed for this parameter.
Any changes you make in the Time Table will affect the features/applications which use the same Time
Table. Program Time Tables taking this into account. Program different Time Tables for features/
applications so that they remain unaffected.
Security Alarm during Working Hours on: Select the desired destination - DOP, External Number,
Routing Group - on which Security Alarm should be initiated during the Time Zone - Working Hours.
Depending on the destination you selected, program the following parameters:
DOP number: If you selected DOP, select the DOP number. ETERNITY PE supports 3 DOPs,
whereas ETERNITY ME and GE support only a single DOP.
Routing Group number: If you selected Routing Group as destination, select the number of the
Routing Group you programmed for Security Alarm. For example, if you have programmed Routing
Group number 03 for Security Alarm for Working Hours, select 03. By default, Routing Group 01 is
selected.
External Number 1: If you selected External Number as the destination, enter the first external number
you want the system to dial for Security Alarm.
External Number 2: Enter the second external number you want the system to dial for Security Alarm.
External Number 3: Enter the third external number you want the system to dial for Security Alarm.
The External Numbers may be numbers of Emergency Services like the Police, Ambulance, or of any
key decision makers.
The External Number can be of a maximum of 16 digits.
2178
Security Alarm during Break Hours on: Like the previous parameter, select the desired destination DOP, External Number, Routing Group - on which Security Alarm should be initiated during the Time Zone
- Break Hours. Depending on the destination you selected, program the DOP Number/Routing Group/
External Numbers, as described above.
Security Alarm during Non-Working Hours on: Repeat the instructions given above to select the
desired destination for Security Alarm during the Time Zone - Non-Working Hours and program the DOP
Number/Routing Group/External Numbers.
Security Reporting
Security Reporting: Select the check box to enable this flag, if you want Security Reporting to be
initiated for Security Alarm calls. Security Reporting enables notification of emergency situations to
Routing Group member extensions.
Security Reporting-Routing Group: If you enabled Security Reporting, select the number of the
Routing Group you programmed for Security Reporting. For example, if you have programmed Routing
Group number 04 for Security Reporting, select 04. By default, Routing Group 01 is selected.
Security Alarm - Delay Response Timer (sec): enter the time for which the system should wait
before dialing out the first external number after receiving signal about the emergency from the DIP. By
default the Delay Response Timer is set to 15 seconds.
Call Attempt Interval for External Number (sec): If you programmed External Numbers as Security
Alarm destination for any of the three Time Zones, you may program this parameter. This parameter
defines the time gap between each attempt to call the external number. When no acknowledgement is
received from one external number you programmed, the system will attempt to call the other external
number you programmed after the Call Attempt Interval. By default, the interval is set to 15 seconds.
Number of Attempts for each External Number: If you programmed External Numbers as Security
Alarm destination for any of the three Time Zones, you may program this parameter. This parameter
defines the number of attempts the system should make to dial each external number you have
programmed. When no acknowledgement is received from an external number you programmed, the
system will repeatedly call the number for the number of Attempts you programmed. By default, the
number of attempts is set to 5.
OG Trunk Bundle Group to Dial External Number: If you programmed External Numbers as
Security Alarm destination for any of the three Time Zones, select the Outgoing Trunk Bundle Group
from which the system should dial the external number. By default, OG trunk bundle group 01 is
selected.
If you have finished programming the Security Alarm parameters, click 'Submit' to save your settings.
2179
To select the Security Alarm 'Trigger on' destination for a Time Zone, dial:
5204-TimeZone-Trigger on
Where,
Time Zone is
1 for Working Hours
2 for Break Hours
3 for Non-Working Hours
Trigger on destination is
1 for DOP
2 for Routing Group
3 for External Number.
To program DOP for Security Alarm for a Time Zone, dial:
5205-TimeZone-DOP
Where,
Time Zone is
1 for Working Hours
2 for Break Hours
3 for Non-Working Hours
DOP is from 0 to 3.
By default, DOP number is 0 for all Time Zones.
To program Routing Group for Security Alarm for a Time Zone, dial:
5206-TimeZone-Routing Group
Where,
Time Zone is
1 for Working Hours
2 for Break Hours
3 for Non-Working Hours
Routing Group is from 01 to 96.
By default, Routing Group 01 is selected for all Time Zones.
To program External Numbers for Security Alarm for a Time Zone, dial:
5207-TimeZone-Index-External Number-#*
Where,
Time Zone is
1 for Working Hours
2 for Break Hours
3 for Non-Working Hours
Index is from 1 to 3.
External Number is a number string of a maximum 16 digits. Terminate the command with #* if the
number of digits in the External number is fewer than 16.
To program OG Trunk Bundle Group for External Numbers, dial:
5208-OG Trunk Bundle Group
Where,
OG Trunk Bundle Group is from 01 to 32.
By default, OG Trunk Bundle Group 01 is selected.
To program Delay Response Timer for Security Alarm, dial:
5209-Delay Response Timer
Where,
Delay Response Timer is from 000 to 255 seconds.
2180
Exit SE mode.
How to use
This is an automatic application. The system automatically dials the programmed numbers on receiving the signal
from the DIP. Human intervention is required only for acknowledging the emergency call placed by Security Alarm.
To acknowledge the Security Alarm call, press '0'.
It is unlikely that external called parties would know that '0' must be pressed to acknowledge the
emergency call. You are recommended to include this information in the Voice Module you record for
Security Alarm prompting the called party to dial '0'.
It may happen that Security Alarm is initiated mistakenly. In which case, it must be terminated from the SA mode.
To terminate an erroneously initiated Security Alarm,
2181
2182
Dial 1#92.
You get Dial tone again.
How it works
Extension user A wants to access a particular Mobile port, Mobile Port 1 to make a call. Extension A must
dial the Selective Port Access Feature Code, followed by the Port Type Code for Mobile ports and then dial
the Port Number.
Port Numbers
SLT
01
001 to 512
DKP
02
001 to 128
CO
03
001 to 128
BRI
04
01 to 32
T1E1
05
1 to 8
E&M
06
001 to 128
Mobile
25
01 to 64
SIP Trunk*
26
01 to 32
ISDN Terminal
28
01 to 64
SIP Extension*
34
001 to 999
Magneto
29
001 to 128
35
01 to 32
Virtual Extension
36
01 to 64
Radio Ports
40
01 to 16
Here, Extension A must dial 69-25-01, where 69 is the feature code for Selective Trunk Access, 25 is the
port access code for the Mobile Port, and 01 is the number of the Mobile Port which A wants to access.
Similarly, if Extension A wants to call SIP Extension 10, A can dial 69-34-010.
How to configure
To be able to use Selective Port Access, extension users must have this feature enabled in their Class of Service
(COS).
2183
How to use
For EON & Extended IP Phone Users
To enable Selective Port Access on an extension,
Press DSS key assigned to Selective Port Access code
From the menu select the Port Type
Enter the Port Number of the selected Port Type
OR
Dial 69 / 89 (for users in USA)
Dial the Port Type - Port Number
2184
How to use
For EON & Extended IP Phone Users
Go OFF-Hook.
Press DSS Key assigned to Self Ring Test.
OR
Dial 1057.
Go ON-Hook.
Your phone rings.
Go OFF-Hook to stop the ring.
Go ON-Hook.
2185
Meaning
Idle
Seized
When the call appearance is been seized from any User binding using the lineseize subscription.
Progressing
When the User has generated a call using the call appearance and the called
destination is ringing.
Ringing
Active
When the call of the User at the call appearance is in matured state.
Held
When the call at the call appearance of the User has been put on public hold
from the User binding.
Held-private
When the call at the call appearance of the User has been put on private hold
from the User binding.
Open Standard IP Phones may differ in the type of indication (LED color and cadence, text message
display) they provide for the Call States. Refer to the manufacturers documentation for SCA indication
supported on the phones.
Calls put on 'Consultation Hold' from Location -1 of SIP Extension (binding -1), the SIP Extension
registered at Location-2/3 (another binding) can not retrieve that call.
How it works
ETERNITY supports up to 10 call appearances on SIP extensions. The number of call appearances that will be
shared by the SIP Phones will depend on the number of call appearances you have configured for the SIP
extension.
To provide SCA to the Open Standard IP Phones registered with the same ID, the Shared Call Appearance flag
must be enabled in the SIP Extension Settings of ETERNITY.
On the Open Standard IP Phones, make sure you have configured as many call appearances as allowed on the
SIP extension by ETERNITY and configure the corresponding number of CA keys.
2186
A, B and C are Open Standard SIP phones registered with ETERNITY at three different locations with the
same SIP ID, 602.
Two Call Appearances are configured for A, B and C and Shared Call Appearance is enabled for SIP ID
602.
Two keys are assigned for the two Call Appearances on A, B and C.
ETERNITY presents the incoming call on a free call appearance, Call Appearance1, as Ringing.
A, B and C get the same alert, Ringing simultaneously on the same call appearance, Call Appearance 1.
A, B and C get indication of the current call state as Active on the same call appearance. B and C will not
be able to make or receive any new call from this busy call appearance. However, they can make or
receive a new call from the other free call appearance, Call Appearance 2.
When A makes an outgoing call using Call Appearance 2, ETERNITY presents the state of the same call
appearance on A, B and C as Seized, then as Progressing when the destination number starts ringing,
and then as Active, when the call is answered.
B and C will not be able to make or receive a call from Call Appearance 2.
A can put an Active call on public Hold or on private Hold. When A puts an Active call on public hold,
ETERNITY presents the state of this call as Held to A, B and C. Now, B or C can retrieve the call by
pressing the corresponding call appearance key.
When A puts an Active call on private Hold, ETERNITY presents the state of this call as private-Held to
A, B and C. Only A can retrieve the call. Thus, if a call is put on private hold (Held-private), it can be
retrieved only from the IP Phone that put it on hold.
How to configure
You can provide this feature only to Open Standard IP Phones you have registered with ETERNITY. To provide this
feature,
On ETERNITY, you must enable the Shared Call Appearance check box on the SIP Extension Settings.
For instructions, see Configuring SIP Extensions.
2187
How to use
To be able to use this feature, first collect the following information from your Network Operator:
Balance Inquiry Number: This is the number provided by the Network Operator to the subscribers to
check Balance. Different Network Operators have different numbers. For example, the Balance Inquiry
number of Vodafone is *141#.
Recharging Service Number: This is the number provided by the Network Operators to their subscribers
for Recharging Service. Different Network Operators have different numbers for Recharging Service. For
example, the Recharging Service Number of Vodafone is *140*.
311. ETERNITY supports Unstructured Supplementary Service Data (USSD), the standard for transmitting information over CSM signaling channels and a commonly used method to query the available balance and other similar information in pre-paid GSM services.
2188
All the mobile ports configured in the system will appear on this page, by their Names you programmed
when configuring the mobile trunk ports.
If you have not programmed any name for a port, the Name field for that port will appear blank.
Balance Inquiry
Click the Request option button under Balance Inquiry, for all those Mobile Ports for which you want
to request Balance Inquiry.
Enter the Balance Inquiry Number provided by the Network Operator whose SIM Card you have
installed in the Mobile Port.
A maximum of 16 digits are allowed. The valid digits for Balance Inquiry number are any digits from 0 to
9 and the characters * and #
Click Submit.
Click the Request option button under Recharge, for all those Mobile Ports for which you want to make
a recharge request .
2189
Number: Enter the Recharging Service Number provided by the Network Operator in this field.
A maximum of 16 digits are allowed. The valid digits for Recharging Service number are any digits from
0 to 9 and the characters * and #
PIN: Enter the PIN number which is printed on the Recharge Voucher/Coupon. Your Recharge PIN
number may consists of a maximum of 20 digits.
The valid digits for PIN number are any digits from 0 to 9 and the characters * and #.
Make sure you enter the digits and characters of the Recharge PIN number exactly as given on the
Recharge Voucher/Coupon.
Click Submit.
In USSD Reply the response received from the GSM network (including possible error messages) will
be displayed. When the USSD Reply is received from the network, it will appear with the Date and
Time stamp of ETERNITY in this field.
For each port that you send a Balance Inquiry/Recharge Request, you will get this USSD-Reply: "Please
wait, processing the request. Refresh the page to see the current status."
The response received from the GSM network (including possible error messages) will be displayed under
USSD-Reply. When the USSD Reply is received from the network, it will appear with the Date and Time
stamp of ETERNITY in this field.
For each Mobile Port (SIM Card) at a time you can either request Balance Inquiry or Recharge the SIM
Card.
However, you can send Balance Inquiry/Recharge request for all the Mobile Ports available in the
system.
2190
During Balance Inquiry/Recharge-Request, the status of the Mobile port will be 'busy'. It will become
idle only after the USSD response is received from the GSM network.
The ETERNITY will clear the USSD Reply after system restart. So each time you open the 'SIM
Balance and Recharge' page after system restart, the USSD Reply box will be blank.
SMS Gateway
What's this?
The SMS Gateway feature of ETERNITY enables you to send/receive messages to/from individuals, selective
groups or masses using the Mobile Port of ETERNITY.
ETERNITY allows you to register multiple SMPP Clients (Software Applications used for sending/receiving
messages) with ETERNITY. ETERNITY functions as an SMPP Server. These Clients can send/receive messages
using the Mobile port/s of ETERNITY.
The messages are sent using the Short Message Peer to Peer Protocol (SMPP Version 3.4). Using this encoding,
it is possible to send up to 160 7-bit characters in one message, in the GSM network.
To use this feature you must purchase the SMS Gateway License. Refer to the topic, License
Management to know more.
How it works
For this feature to work,
you must have the SMS Gateway license.
you must have the SMPP Clients installed on a computer connected in the same LAN as ETERNITY.
you must configure the required parameters in the SMPP Clients to register itself with ETERNITY
you must define the Mobile port through which the messages are to be sent/received in the SMPP Clients.
you must configure the SMPP Client parameters in ETERNITY.
This is how the ETERNITY SMS Gateway works,
On successful registration of the SMPP Client with ETERNITY, a binding is established between the SMPP
Client and ETERNITY.
The SMPP Client can bind itself with ETERNITY as a Receiver, Transmitter or Transceiver.
As a Transmitter, the SMPP Client will only be able to send messages using the Mobile port of
ETERNITY.
As a Receiver, the SMPP Client will only be able to receive messages from the Mobile port of
ETERNITY.
As a Transceiver, the SMPP Client will be able to send and receive messages from the Mobile port of
ETERNITY.
The SMPP Client sends all the information required to send an SMS the message content, the
destination mobile number and the mobile port through which the message is to be sent to ETERNITY in
the Protocol Data Unit (PDU) format.
The message will be sent to the destination number through the Mobile port, if it is idle or in speech. The
message will be sent using the SMS Center Number configured for that port.
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If the Mobile Port is any other state the SMS will not be sent.
ETERNITY will accept a new SMS sent by the SMPP Client for the same Mobile port only after the
preceeding SMS is successfully sent.
All incoming SMS on the Mobile port will be sent to the SMPP Client.
After the SMS is sent to the SMPP Client the same will be deleted from the SIM card of the Mobile port.
How to Configure
For the SMS Gateway you need to configure the following parameters,
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SMPP Server: By default, the SMPP Server is disabled. To use the SMS Gateway feature, select
Enable.
SMPP Server Port: Enter the SMPP Servers listening port. The valid range is 1025 to 65535. Default:
SMPP Server Port is 2775.
Enquiry-Link Timeout (seconds): The SMPP Client sends the Enquiy-Link Requests to the SMPP
Server at regular intervals to refresh its binding with the Server. The system re-loads this timer
(default:120 seconds) every time it receives the request from the Client. If no response is received from
the SMPP Client before the expiry of this timer, the Server considers the Client as disconnected. The
valid range of the timer is 005 to 999. Default: 120 seconds.
SMPP Server Debug: To monitor the events and processes of the SMPP Server, for trouble shooting
and identifying faults and errors, select the SMPP Server Debug check box. The debug messages are
sent to the remote Syslog Server. For detailed instructions on how to configure the Destination Port,
Syslog Server IP Address and Port, see System Debug.
You can register multiple SMPP Clients with ETERNITY. Against each SMPP Client configure the
following parameters:
SMPP Client System ID: Enter the ID you want the Server to use to authenticate the Client. A
maximum of 16 characters, including all ASCII characters are allowed.
SMPP Client Password: Enter the password you want the Server to use to authenticate the Client.
A maximum of 9 characters, including all ASCII characters are allowed.
Do not assign the same SMPP Client System ID and Password to multiple SMPP Clients.
Configure the same SMPP Client System ID and Password in the SMPP Client. This ID is sent to the
Server in the connection request made by the Client.
Mobile Port: Select the Mobile Port using which the SMS is to be sent from/to the SMPP Client.
The ETERNITY supports a maximum of 64 Mobile ports312.
Make sure you do not assign the same Mobile Port to multiple SMPP Clients.
Debug: If you want to monitor the events and processes of the SMPP Client, for trouble shooting
and identifying faults and errors, select the Debug check box. The debug messages are sent to the
remote Syslog Server. For detailed instructions, for configuring the Destination Port, Syslog Server
IP Address and Port, see System Debug.
312. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.
2193
Click Status.
The following parameters are displayed for each SMPP Client registered with ETERNITY:
System ID: This is the ID received in the SMPP Clients connection request.
Description
Not Connected
Connected
2194
Binding Type: This field displays the type of SMPP Client Binding, that is Transmitter, Receiver or
Transceiver.
Address: This field displays the SMPP Clients IP Address and Port.
Port
Station
Port
Virtual
Port
SLT
Auxillary
Port
Trunk
Port
DKP
ISDN
Terminal
SIP
Extension
(VoIP)
CO (2 Wire
Analog Trunk) Port
AIP (External
Music Port)
E&M Port
DIP
BRI Port
DOP
T1/E1 Port
SIP Port (VoIP)
Mobile Port
The ETERNITY treats a port as an entity and processes it on the basis of port type and its programmed attributes.
Type of Port
SLT
DKP
Trunk
DOP
DIP etc.
Software
Port
Attributes of Port
Access Code
COS Group
Toll Control Group
DID, etc.
Hardware Port
Slot Number
Port Number
2195
Software Port
The ETERNITY takes a software port as fundamental entity. It processes the software port. Hardware ID and the
access code are just two attributes of a software port and hence they are not used anywhere in processing and
programming.
Software ports are always numbered from 1 to the maximum number of ports supported for each port type. The
following table lists this range for each port type.
Port Type
SLT
512
001-512
DKP
128
001-128
Trunk
128
001-128
E&M
128
001-128
BRI
32
01-32
T1E1PRI
1-8
Mobile (GSM)
64
01-64
SIP Trunks
32
01-32
Magneto
128
001-128
Each type of software port has different attributes. The System Engineer (SE) programs these attributes using the
corresponding template at the time of installing the ETERNITY.
Hardware ID is an attribute of a software port. Hence, all the programming is done for the software port and not for
the hardware ID. Accordingly, the software port number is used for all programming tasks.
An access code is just a Flexible number assigned to the software port. Programming is for the software port and
not for the access code (Flexible number). Accordingly, the software port number is used for all the programming.
The System Engineer allocates software port numbers to different users. This allocation is Flexible and any
software port number can be assumed for any user. Hardware ID is not relevant at this stage. Hardware ID can be
programmed for a software port any time. Further, it can be changed any time in case of hardware failure of a port.
Following example elaborates this point.
Name
Position
Port Type
Anil Sharma
Managing Director
DKP
001
Nikhil Rao
VP (Marketing)
DKP
002
Revathi Thyagarajan
VP (Finance)
DKP
003
Anand Chakraborty
Manager (MD)
SLT
001
Pankaj Shah
Accountant
SLT
003
Ravi Tandon
Sales Executive
SLT
002
Conference Room
--
DKP
005
Canteen
--
SLT
004
Leased Line
--
Trunk
003
2196
Name
Position
STD Line
--
Port Type
Trunk
001
Software port numbers start from 001 for all different port types.
Order is not important while allocating software port numbers.
Hardware ID
The hardware ID of a software port denotes where the port is physically located. To derive hardware ID of a
software port, we need:
2197
2198
Relevant Topics:
1. Paging
2085
2. Music on Hold (MOH)
2057
3. DSS Keys Programming
808
2199
How it works
For example, two Local Area Networks, Network A and Network B, are connected through Frame Relay/ Multi
Protocol Label Switching (MPLS) network to give access to local resources and also to make Peer-to-Peer calls.
59.162.252.82
SIP Proxy
192.168.1.0/24
192.168.2.0/24
Public
IP
Frame Relay/MPLS
192.168.1.1
ETERNITY
ETERNITY
2200
The Static Routing Table defines the appropriate Gateway Address (or Routers LAN Address) where the IP
packets are to be sent.
In the Static Routing Table, you must configure:
The address of the final Destination where the packets are to be sent.
The Subnet Mask to be applied on the final destination address.
The Gateway Address where the IP packets are to be sent.
When ETERNITY sends packets, if the final destination IP Address and ETERNITY are not in the same Subnet, the
system will check the Static Routing Table.
If a perfect match is found, ETERNITY will start sending the IP packets to the corresponding Gateway Address
configured in the table.
If no match is found, ETERNITY will send the IP Packets to the Default Gateway Address (Network Connection
Type) you configured in the Master Ethernet Port Parameters. See Configuring Master Ethernet Port Parameters.
The Static Routing Table is common for all VoIP Ports and the Master Ethernet Port.
How to configure
The Static Routing Table must be configured at each location where ETERNITY is installed. You may configure the
Static Routing Table using Jeeves or by dialing the system commands from a telephone connected to the
ETERNITY.
2201
The Static Routing Table allows you to configure up to 8 entries. Each entry is stored against an Index
number.
For each entry, you must configure the following fields:
Destination Address: This is the address of the final destination where the call is to be made. This
can be a device IP Address or Network Address.
Gateway Address: This is the IP address of the node where the IP packets are to be sent. Generally,
it is the IP address of the LAN interface of the Router.
The Gateway Address must be in the same subnet as ETERNITY.
Destination Address
Subnet Mask
Gateway Address
192.168.2.0
255.255.255.0
192.168.1.1
2
:
8
The Gateway Address 192.168.1.1 specifies the LAN address of the Router A which connects location
A and location B.
The IP address of the LAN interface of the router which connects Location A to the public internet
should be configured as Default Gateway in the Network Parameters of ETERNITY in location A.
With the Static Routing Table configured thus, all calls made by ETERNITY to 192.168.2.0/ 24 will be
routed through the router which connects Location A to Location B. Whereas, all calls made by
ETERNITY to addresses other than 192.168.2.0/ 24 will be routed through the Default Gateway.
Similarly, configure the Static Routing Table in ETERNITY at location B to enable calling from Location
B to Location A.
2202
7811-1-Index-Destination Address
Where,
Index is 1 to 8.
Destination Address is of 15 digits maximum. Enter each octet in full. For example, to program
192.168.10.10, dial 192168010010.
Exit SE mode.
2203
SMDR Storage: These parameters are programmed to enable the storing of the IC, OG and Internal calls.
To know more, see Station Message Detail Recording-Storage.
SMDR Report: These parameters are programmed to assign destination port for getting report of IC, OG
and Internal calls and to get offline report. To know more, see Station Message Detail Recording-Report.
SMDR Online:These parameters enable you to obtain Online report of Incoming, Outgoing and Internal
calls. With Online SMDR you can obtain details of each call immediately after the call has been made or
received. You can also set the call record format you want for Incoming calls when ETERNITY is
interfaced with a third party call accounting software (CAS). To know more see, Station Message Detail
Recording-Online.
SMDR Posting: These parameters enable you to interface third-party call accounting software (CAS) with
ETERNITY for call cost calculation. You can select the protocol supported by the call accounting software
and further customize the handshaking parameters and call record formats. To know more see, Station
Message Detail Recording-Posting.
Refer separate Manual for Hotel Applications for more details about PMS.
2204
This feature requires a license. To use this feature you must purchase the license for the Business Feature
Suite. Refer the topic License Management to know more.
2205
How to configure
To get the Online report you must do the following:
Enable SMDR Storage in the SMDR buffer. See Station Message Detail Recording-Storage.
Select the Destination port for Incoming, Outgoing and Internal calls. If you select Ethernet as the
destination port, you must configure the Destination IP Address. The Online report is sent to this address
as soon as the incoming call is completed.
You may also change the default format for the SMDR Online report for incoming calls, like column
position and field length for calling number, speech duration, type of call etc., as required. For this you
need to configure the settings of SMDR Incoming Online Record Format.
In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
2206
In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
To configure Call Record Format for Incoming Calls, click SMDR Incoming Online Call Record Format
to expand.
Serial Number: This is the serial number generated for each call record. Serial numbers are generated
from 001 to 999. When serial number '999' is reached, the numbers roll over to 001.
When this field rolls over, it increments the increment counter.
Increment Counter: It increments when the serial number counter rolls over. The Increment counter
starts from A, ending at Z, and then roll over back to A. (in eternity it is numeric)
Property Code: This is the property code, if required. This may be an abbreviation of the property
name.
Extension Number: This is the extension number that answered the call. You can define the column
position and the field length for the extension number.
2207
Trunk Number: This is the number of the trunk on which the call was received.
Date: The date on which the call was received. The date fill flag is to be enabled.
Filler Character field is applicable for Date, Month and Year, i.e. whether the single digit date is to be
printed as space-X or 0-X. For example, date = 1 is to be displayed as '1' or '01'.
Where leading zeroes are not required, the date, month and year sub-fields are right aligned and the
spaces are filled with character 'space'.
The Date field is not linked to the global flag of Date Format. The global Flag of Date format is used,
while using features or in configuration reports but not for SMDR Online. This is because the date
format used by the CAS is not the same as used by the users of the system.
Time: The time when the call was received. The format of the time field and the time fill flag are to be
programmed.
Filler Character field is applicable for Hours, Minutes and Seconds i.e. whether the single digit hour is
to be printed as space-X or 0-X. For example, hour = 1 is to be displayed as '1' or '01'.
In case leading zeroes are not required, Date, Month and Year sub-fields are right aligned and the
spaces are filled with character 'space'.
Answer Duration: The time after which the call was answered. Program the duration unit and the
duration fill flag.
Hold Duration: The time for which the call was put on hold.
Speech Duration: The time for which the call was in speech with the extension.
When Duration Unit = Minutes, the rounding off to the nearest whole number is done. For seconds <= 30,
Minute is not incremented. For seconds > 30, minute is incremented.
2208
Called Number: This is applicable only for calls received on SIP trunks. The number dialed by the
caller is referred to as Called Number.
Digits dialed in Built-In Auto Attendant: This is the number dialed by the caller using Built-In Auto
Attendant.
Remarks: You may use this for indicating the Type of Call, for example, Built-In Auto Attendant.
Reset Serial Number to 001: The Serial number counter can be reset to 001 after 24 hours (from
00:00 HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.
Reset Increment Counter: The Increment Counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.
Meaning
None
COM1
COM2
Printer Port
Ethernet Port
Meaning
None
COM1
COM2
2209
Code
Meaning
Printer
Ethernet Port
By default, the port assigned is None. This means the On-line printing is disabled.
To assign an IP Address, if you select Ethernet Port as the destination port, dial:
2732-IP Address
By default, IP Address is 192.168.1.104
To assign the Port, dial:
2733-IP Port
Where,
Port is from 514 and 1025-65535
By default, IP Port is 514.
To Start/Abort report generation, dial:
1072-101-Flag
Where,
Flag
Meaning
Abort
Start
By default, Flag is 0.
The ETERNITY provides a facility to abort the report generation in midway (1072-101-0). Once the
report generation is aborted, then it has to be explicitly started with command (1072-101-1). This
command is issued from the SA mode.
Incoming Calls
To assign destination port for Online SMDR-IC Call Record, dial:
2930-Code
Where,
Code
Meaning
None
COM1
COM2
Printer Port
Ethernet Port
By default, the port assigned is None. This means the Online printing is disabled.
2210
To assign an IP Address, if you select Ethernet Port as the destination port, dial
2932-IP Address
By default, IP Address is 192.168.1.104.
To assign the Port:
2933-Port
Where,
Port is from 514 and 1025-65535
By default, IP Port is 514.
To start/abort report generation:
1072-151-Flag
Where,
Flag
Meaning
Abort
Start
By default, Flag is 0.
The ETERNITY provides a facility to abort the report generation midway (1072-151-0). Once the report
generation is aborted, then it has to be explicitly started with (1072-151-1). This command is issued
from the SA mode.
Serial Number
To program column position for serial number, dial:
8200-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 01.
To program field length for serial number, dial:
8201-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 04.
To program alignment for serial number, dial:
8202-Alignment
Where,
Alignment
Meaning
Left Alignment
Right Alignment
2211
By default, Alignment is 2.
To program fill character for serial number, dial:
8203-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is Space.
To program the reset for serial number, dial:
8204-Reset
Where,
Reset
Meaning
No Compulsory Reset
By default, Reset is 1.
Serial Number starts from 1 and not 0.
When this field rolls over, it increments the increment counter.
Increment Counter
To program column position for increment counter, dial:
8205-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 00 (This field is not available by default).
By default, Field Length is 1, which is fixed.
To program the reset for increment counter, dial:
8206-Reset
Where,
Reset
Meaning
No Compulsory Reset
By default, Reset is 1.
Increment Counter starts from A to Z and then rolls over back to A.
Increment Counter increments when Serial Number Counter rolls over.
2212
Property Code
To program column position for property code, dial:
8207-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 00 (This field is not available by default).
To program field length for property code, dial:
8208-Field Length
Where,
Field Length is from 00 to 78.
To program property code string for property code, dial:
8209-Property Code String
This code is required by the Property Management System (PMS) when it is catering to more than one
PMS interfaces.
Refer separate Manual for Hotel/Motel Applications for more details about PMS and Hotel applications
for this feature.
Extension Number
To program column position for extension number, dial:
8210-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 29.
To program field length for extension number, dial:
8211-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 04.
To program alignment for extension number, dial:
8212-Alignment
Where,
Alignment
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
2213
Trunk Number
To program column position for trunk number, dial:
8214-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 23.
To program format type for trunk number, dial:
8215-Format Type
Where,
Format Type
Meaning
Matrix Format
Check-In Format
First Character in Check In Format is X (Fixed). Remaining three characters show the software port
number. However, this will not specify whether the call is made through CO 125 or E&M 125. Also the
channel number will not be specified in case of call made through T1E1PRI port or BRI port.
Date
To program column position for date field, dial:
8216-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 34.
To program field length for date field, dial:
8217-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 10.
To program alignment for date field, dial:
8218-Alignment
Where,
Alignment
1
2214
Meaning
Left Alignment
Alignment
2
Meaning
Right Alignment
By default, Alignment is 2.
To program fill character for date field, dial:
8219-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is Zero.
To program date format for date field, dial:
8220-Date Format
Where,
Date Format
Meaning
01
DD-MM-YY
02
DD/MM/YY
03
DD.MM.YY
04
DD MM YY
05
DDMMYY
06
DD-MM-YYYY
07
DD/MM/YYYY
08
DD.MM.YYYY
09
DD MM YYYY
10
DDMMYYYY
11
MM-DD-YY
12
MM/DD/YY
13
MM.DD.YY
14
MM DD YY
15
MMDDYY
16
YY-MM-DD
17
YY/MM/DD
18
YY.MM.DD
19
YY MM DD
20
YYMMDD
21
YYYY-MM-DD
22
YYYY/MM/DD
23
YYYY.MM.DD
24
YYYY MM DD
2215
Date Format
Meaning
25
YYYYMMDD
26
MM-DD
27
MM/DD
28
MM.DD
29
MM DD
30
MMDD
31
DD-MM
32
DD/MM
33
DD.MM
34
DD MM
35
DDMM
Meaning
Disable
Enable
By default, Date Fill Flag is 1 (that is, single digit in Date, Month and year is printed with prefix 0).
Leading Zeros field is applicable for Date, Month and Year, that is, whether the single digit date is to be
printed as space-X or 0-X. For example: date = 1 is to be displayed as 1 or 01. In case when leading
zeroes are not required, the date, month and year sub-fields are right aligned and the spaces are filled
with character space.
This Date field is not linked to the global flag of Date Format. The global Flag of Date format is used
while using features or in configuration reports but not in PMS. This is because the date format used by
the PMS is not the same as used by the users of the system.
Time
To program column position for time field, dial:
8222-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 45.
To program field length for time field, dial:
8223-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 08.
2216
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for time field, dial:
8225-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is Zero.
To program time format for time field, dial:
8226-Time Format
Where,
Time Format
Meaning
Disable
Enable
Meaning
Disable
Enable
Answer Duration
To program column position for answer duration field, dial:
8227-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 54.
2217
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for answer duration field, dial:
8230-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is Space.
To enable/disable the Filler character flag for Answer Duration, dial:
8259-Filler Character Flag for Answer Duration
Where,
Filler Character Flag for
Answer Duration
Meaning
Disable
Enable
Default = Enable, (that is, the Filler Character will used as programmed)
To program duration unit for answer duration field:
8231-Duration Unit
Where,
Duration Unit
Meaning
HH:MM:SS
HHMMSS
Minutes
Seconds
2218
Hold Duration
To program column position for hold duration field, dial:
8232-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 58.
To program field length for hold duration field, dial:
8233-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 03.
To program alignment for hold duration field, dial:
8234-Alignment
Where,
Alignment
Meaning
Left Alignment
Right Alignment
Meaning
Disable
Enable
Speech Duration
To program column position for speech duration field, dial:
8237-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 62.
2219
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for speech duration field, dial:
8240-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is Space.
To enable/disable the Filler character flag for Speech Duration, dial:
8261-Filler Character Flag for Speech Duration
Where,
Filler Character for Speech Duration
Meaning
Disable
Enable
Called Number
To program column position for called number field, dial:
8242-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 00. (This field is not available by default)
To program field length for called number field, dial:
8243-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 16.
To program alignment for called number field, dial:
8244-Alignment
2220
Where,
Alignment
Meaning
Left Alignment
Right Alignment
By default, Alignment is 1.
To program number format for called number field, dial:
8245-Number Format
Where,
Number Format
Meaning
Continuous
Separated
Calling Number
To program column position for calling number field, dial:
8246-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 06.
To program field length for calling number field, dial:
8247-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 16.
To program alignment for calling number field, dial:
8248-Alignment
Where,
Alignment
Meaning
Left Alignment
Right Alignment
By default, Alignment is 1.
2221
Meaning
Continuous
Separated
Meaning
Left Alignment
Right Alignment
Remarks
To program column position for remarks field, dial:
8253-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 68.
To program field length for remarks field, dial:
8254-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 02.
2222
Meaning
Left Alignment
Right Alignment
By default, Alignment is 1.
To assign default IC SMDR format, dial:
8256
How to use
You can start and stop SMDR Online report from the System Administrator mode using Jeeves or dialing SA
Commands from an extension phone.
To start/stop Online report using Jeeves,
Open Jeeves.
2223
To start SMDR Online for Outgoing Calls, Incoming Calls and Internal Calls, click the Start button.
To stop SMDR Online for any of these call types, click Abort button.
2224
Exit SA mode.
424@192.168.50.2
00919724302156
389@192.168.50.2
300@192.168.50.2
301@192.168.50.2
+919898337166
+919898337166
389@192.168.50.2
V001
P001
V001
V001
V001
M001
M001
V001
738
548
707
542
594
593
735
723
08-10-13
08-10-13
08-10-13
08-10-13
08-10-13
08-10-13
08-10-13
08-10-13
17:33:40
17:30:26
17:24:42
17:38:39
17:39:17
17:39:59
17:40:33
17:43:10
0
3
0
0
0
6
0
0
0
0
0
0
0
15
0
0
9
246
694
20
3
14
48
48
N
N
N
N
N
N
N
N
638
611
621
639
606
624
634
605
722
735
548
626
634
624
09-10-2013
09-10-2013
09-10-2013
09-10-2013
09-10-2013
09-10-2013
09-10-2013
15:23:44
15:22:53
15:24:08
15:23:56
15:24:29
15:27:57
15:29:08
14
76
16
32
46
16
36
732
555
591
738
717
000
000
000
000
000
M001
M001
M002
M001
M001
9898337166
9426774529
9979109107
9879529041
9898991823
08-10-13
08-10-13
08-10-13
08-10-13
08-10-13
17:36:56
17:37:26
17:40:07
17:42:10
17:49:51
14
7
81
24
48
1
1
1
1
1
3.10
3.10
3.10
3.10
3.10
I
I
I
I
I
2225
SMDR-Posting Protocols
The ETERNITY supports different SMDR posting protocols from the system to CAS. The flow of messages
between the ETERNITY and the protocols of CAS Interface (Matrix and Blind Send) are described below:
Matrix
CAS to ETERNITY
CAS to ETERNITY
2226
ETERNITY to CAS
CAS to ETERNITY
NAK
CAS to ETERNITY
2227
The ETERNITY will make 5 attempts (default value of Data Transfer Retry Count - on Negative Response) to
send the message after a regular interval of 3 seconds ( default value of Data Transfer Retry Timer - on
Negative Response). If the ACK is still not received from the CAS, the ETERNITY will proceed to the next
message.
CAS to ETERNITY
Blind Send
If you select this protocol as the SMDR-OG Posting Protocol, ETERNITY sends the call details without waiting for
any response from the CAS. Each record is sent with the End of Packet Character.
Customised
If you select this protocol as the SMDR-OG Posting Protocol, ETERNITY provides you the flexibility to set the
values for the OG Handshaking Protocol and the OG Online Call Record Format as per your requirement.
2228
Matrix
Alignment
Filler Char.
Required?
Filler Char.
Decimal
Value
Fixed
Right
Yes
032
Every 6 hours it is
cleared to 001.
(Starting from
mid-night
00:00:00)
01
Fixed
Left
NA
NA
Every 6 hours it is
cleared to A.
(Starting from
mid-night
00:00:00)
00
04
Fixed
Left
Yes
032
As per the
Programmed
String
Extension
Number
06
05
Fixed
Right
Yes
032
Trunk Number
12
05
Matrix
Format
Left
Yes
032
Date
37
10
DDMMYYYY
Right
Yes
032
Time
48
08
HH:MM:
SS
Right
Yes
032
Duration
057
005
Second
s
Right
Yes
032
Units
063
004
Fixed
Right
Yes
032
Amount
068
007
Currenc
y with
Decimal
Point
Right
Yes
032
Format is
DDD.CC
Currency
000
001
Fixed
Right
Space
032
Country Specific
Start
Column
Number
Field
Length
Format
Serial Number
01
04
Increment
Counter
00
Property
Code
Parameter
Remarks
2229
Start
Column
Number
Field
Length
Format
Alignment
Filler Char.
Required?
Filler Char.
Decimal
Value
Call Type
Indicator
000
001
Fixed
Right
NA
NA
Location
000
005
Fixed
Right
NA
NA
Called
Number
18
19
Continu
ous
Left
Space
NA
Account Code
00
04
Fixed
Right
Yes
032
Remarks
76
02
Fixed
Left
Space
NA
Parameter
Reset Serial
Number to
001
Staring
Character Increment
Counter
Reset
Increment
Counter
Do not Reset
No
Property
Code
AAA
2230
Do not Reset
Prefix String
Required
Currency
Symbol (Enter
Decimal
Value)
Remarks
Blind Send
Start
Column
Number
Field
Length
Format
Alignment
Filler Char.
Required?
Filler Char.
Decimal
Value
Serial
Number
01
04
Fixed
Right
Yes
032
Increment
Counter
00
01
Fixed
Left
NA
NA
Property
Code
00
04
Fixed
Left
Yes
032
Extension
Number
06
05
Fixed
Right
Yes
032
Trunk
Number
12
05
Matrix
Format
Left
Yes
032
Date
37
10
DD-MMYYYY
Right
Yes
032
Time
48
08
HH:MM:SS
Right
Yes
032
Duration
057
005
Seconds
Right
Yes
032
Units
063
004
Fixed
Right
Yes
032
Amount
068
007
Currency
with Decimal
Point
Right
Yes
032
Format is
DDD.CC
Currency
000
001
Fixed
Right
Space
032
Country Specific
Call Type
Indicator
000
001
Fixed
Right
NA
NA
Location
000
005
Fixed
Right
NA
NA
Called
Number
18
19
Continuous
Left
Space
NA
Account
Code
00
04
Fixed
Right
Yes
032
Remarks
76
02
Fixed
Left
Space
NA
Parameter
Remarks
As per the
Programmed
String
2231
Parameter
Reset
Serial
Number to
001
Staring
Character
Increment
Counter
Reset
Increment
Counter
Start
Column
Number
Format
Alignment
Filler Char.
Required?
Remarks
Do not Reset
Do not Reset
Prefix
String
Required
No
Property
Code
AAA
Currency
Symbol
(Enter
Decimal
Value)
Field
Length
Filler Char.
Decimal
Value
To setup the CAS Interface on COM Port (RS232)313, the following functional components are required to
make the interface work:
313. There are two Communication Ports (COM1 and COM2) in the ETERNITY ME, and a single Communication Port in the ETERNITY GE and PE.
2232
Now, connect the COM port of the PC with the COM port the ETERNITY using the communication cable
supplied by Matrix314.
To setup the CAS Interface on the Ethernet Port (TCP/IP), the following functional components are
required to make the interface work:
A PC with a spare Ethernet port (not supplied by Matrix) Or any free Ethernet Port of the LAN Switch on
which the CAS server application software is running.
Now, connect the Ethernet port of the Master/CPU card of the ETERNITY with the Ethernet Port of the PC
(on which CAS server application is running) or to one of the Ethernet ports of the LAN Switch, if the CAS
server is in the same LAN.
How to configure
Configuring the SMDR-Posting feature involves the following steps:
Enabling storage of Outgoing (OG) SMDR. By default, OG SMDR storage is enabled. Refer Station
Message Detail Recording-Storage.
If SMDR-Posting is through RS232 (that is, the CAS Interface is to be set up on the COM Port),
program the attributes of the COM port. Refer the chapter Communication Ports to set attributes of
the COM port.
If SMDR-Posting is through TCP/IP (that is, the CAS Interface is to be set up on the Ethernet port),
program the destination IP address and Port.
2233
In the SMDR-OG Posting Protocol (Handshaking and OG Call Record Format) drop down list, select the
appropriate protocol to be used. Default: Matrix.
Select the Destination Port on which the SMDR Posting is set up. You can select Comm. 1, Comm. 2,
Printer or Ethernet. Default: None.
In Destination IP Address: Port, enter the IP Address and port of the PC on which the CAS server
application software is running, that is, where ETERNITY should post SMDR. Valid port range is: 1025 to
65535.
In Listening Port (of ETERNITY) for Posting, enter the Port number at which ETERNITY must listen for
SMDR Posting response. Default: 6000. Valid port range is: 1025 to 65535.
2234
Response to ENQ Timeout (sec): The time for which the sender waits for a response to ENQ from the
receiver.
ENQ Retry Count - on No Response: The number of times the sender should send ENQ before
dropping the process, in case response is not received for the last message sent.
ENQ Retry Timer (sec) - on No Response: The time after which the sender should sent the ENQ
again, in case the response is not received for the last message sent.
ENQ Retry Count - on Negative Response: The number of times the sender should send ENQ
before dropping the process, in case of a negative response received for the last message sent.
ENQ Retry Timer (sec) - on Negative Response: The time after which the sender should sent the
ENQ again.
Response to Data Timeout (sec): The time for which the sender waits for a response to data from the
receiver.
2235
Data Transfer Retry Count - on No Response: The number of times the sender should send ENQ
before dropping the process. This parameter is used when ACK is received against ENQ and there is
some problem while sending the data.
Data Transfer Retry Timer (sec) - on No Response: The time after which the sender should send the
ENQ again before dropping the process. This parameter is used when ACK is received against ENQ
and there is some problem in sending the data.
Data Transfer Retry Count - on Negative Response: The number of times the sender should send
ENQ before dropping the process. This parameter is used when ACK is received against ENQ and
there is some problem in sending the data.
Data Transfer Retry Time (sec) - on Negative Response: The time after which the sender should
sent the ENQ again before dropping the process. This parameter is used when ACK is received
against ENQ and there is some problem in sending the data.
Use ENQ Character: This is enabled if the protocol uses ENQUIRE (ENQ) Signal.
ENQ Character: This parameter signifies the ASCII character (Single Character) used to send
ENQUIRE (ENQ) signal to the receiver.
Acknowledgement Character: This parameter signifies the ASCII character (Single Character) used
by the receiver to acknowledge the receipt of the Link Control Character/Message Data.
No Acknowledgement Character: This parameter signifies the ASCII character (Single Character)
used by the receiver to dis-acknowledge the receipt of the Link Control Character/Message Data.
Start of Packet Character: A string of four ASCII characters used by the receiver to indicate Start of
Packet. Each ASCII character is from 000 to 252. Start of Packet may be of one character only, in
which case the string should be completed by programming remaining three characters with ASCII Null
Character (000).
End of Packet Character: A string of four ASCII characters used by the receiver to indicate End of
Packet. Each ASCII character is from 000 to 252. End of Packet may be of one character only, in which
case, the string should be completed by programming the remaining three characters should be
programmed as ASCII Null (000).
Use Byte Code Check (BCC): This flag is to be enabled when the protocol uses BCC Signal.
2236
This may be required if you have selected a 'customized' protocol. To refine Call Record Format,
Serial Number: This is the serial number generated for each call record. Serial numbers are generated
from 000 to 999. When serial number '999' is reached, the numbers roll over to 000.
Serial Number starts from 1 and not 0.
When this field rolls over, it increments the increment counter.
Increment Counter: It increments when the serial number counter rolls over. The Increment counter starts
from A, ending at Z, and then roll over back to A.
Property Code: This is the property code required by the CAS used in the organization. It is a string of
alphanumeric characters and is to be terminated with #*. This field has a maximum of 128 alphanumeric
characters.
You must program this string keeping in mind the field length used by the selected/customized posting
protocol.
The default value of the default Property Code String has been set as 'AAA', as at least two known
protocols use this field. You can set a different value here and the new value will appear in the CDR
record, irrespective of the protocol type selected.
2237
If Bell Hobic or Hilton has been selected, you should program this field as 'AAA'. If Xiox protocol has
been selected, you should program this field as HTL. These values are not protocol dependent, but can
be configured by you.
Extension Number: This is the extension number from which the call was made. You can define the
column position and the field length of the Extension number in the Call Detail Record.
Trunk Number: This is the number of the trunk from which the call was made.
The First Character in the Check-Inn Format is X (Fixed). The remaining three characters show the
software port number. However, this does not specify whether the call is made through CO 125 or E&M
125. Also, the channel number is not specified in case of call made through T1E1PRI port or BRI port.
Date: The date on which the call was made. The date fill flag is to be enabled.
Filler Character field is applicable for Date, Month and Year, that is, whether the single digit date is to
be printed as space-X or 0-X. For example, date = 1 is to be displayed as '1' or '01'.
Where leading zeroes are not required, the date, month and year sub-fields are right aligned and the
spaces are filled with character 'space'.
The Date field is not linked to the global flag of Date Format. The global Flag of Date format is used,
while using features or in configuration reports but not in CAS. This is because the date format used by
the CAS is not the same as used by the users of the system.
Time: The time when the call was made. The format of the time field and the time fill flag are to be
programmed.
Filler Character field is applicable for Hours, Minutes and Seconds, that is, whether the single digit hour
is to be printed as space-X or 0-X. For example, hour = 1 is to be displayed as '1' or '01'.
In case when leading zeroes are not required, Date, Month and Year sub-fields are right aligned and
the spaces are filled with character 'space'.
Duration: The duration of each call. Program the duration unit and the duration fill flag.
When Duration Unit = Minutes, the rounding off to the nearest whole number is done. For seconds <= 30,
Minute is not incremented. For seconds > 30, minute is incremented.
Units: The duration of the call interpreted in terms of units. The number of units depends on the Pulse
Rate. The number of units is derived from the Call Unit = Call duration in seconds/Pulse rate in seconds.
Amount: This is the Amount of the call. Program the amount format and the fill flag.
2238
Filler Character field is applicable for both the sub fields of Amount viz. Rupees/Paisa, that is, whether
the single digit Rupee is to be printed as space-X or 0-X. For example, Rupee = 1 is to be displayed as
'1' or '01'. Where leading zeroes are not required, the Rupee and Paisa are right aligned and the
spaces are filled with character 'space'.
When Amount Format = Higher Currency, rounding to nearest whole number is done. For Lower
Currency <= 50, Higher Currency is not incremented and for Lower currency > 50, Higher Currency is
incremented.
Currency: This is the symbol of the currency in which the Amount is charged. A maximum of 8 ASCII
Characters are allowed.
Generally, Currency Symbol field prefixes to Amount field. Hence, to comply with various CDR formats,
it is recommended that the column position of Currency Symbol and Amount field should be
programmed properly.
You can change the Currency Symbol used in the OG-SMDR Format. However, this change will not be
reflected in the Front Desk User Wizard.
Call Type Indicator: This indicates the type of call made, that is, whether local, international, information,
etc.
Your must program the Number String, the Text String and its Meaning as explained in following table:
Number Index
Number String
Text String
Meaning
01
LD
Long Distance
02
95
IC
Inter Circle
03
197
INFO
Information
04
INTL
International
36
:
Local
The Text String is a string of Alphanumeric characters. Number String is of a maximum 4-digits.
The Number Index is kept as '36' as one of the SMDR-OG Posting protocols, INN-FORM XL supports 24
different types of calls.
By default, all the entries in this table are blank.
You are advised to program the first 10 entries of this table as below if the selected posting protocol is Bell
Hobic or XIOX.
Number Index
Number String
Text String
01
02
03
04
05
06
07
08
Meaning
2239
Number Index
Number String
Text String
09
10
36
Blank
Blank
Meaning
Blank
You are advised to program the first 11 entries of this table as below, if the selected posting protocol is
Holidex or Hobic.
2240
Number Index
Number String
Text String
01
02
03
04
05
06
07
08
09
10
11
36
Blank
Blank
Meaning
International
Blank
You are advised to use default (that is, Blank) table, if the selected protocol is Hilton, as Hilton uses
blank entries in this field which is 12 bytes long.
The Text String should preferably be same as Field Length. If not, the remaining spaces will be filled
with character 'Space'. If the Field length is less than the Text string characters, then the number of text
characters equal to the Field length will be printed.
Location: This column indicates the location of the external number to which the call was made.
The system detects the location from the called location programmed in the Area and Country Code
Tables.
Called Location is programmed as one of the parameters of the Area Code Table and Country Code
Table. Depending upon the prefix dialed, the Location string is picked up from either Country Code
table or Area Code table.
The Called Location parameter in the Country Code table and Area Code table is of 8 Characters.
If the number of characters in the field Called Location is more than Field length then the remaining
characters will not be printed (overlapped by next field).
If the number of characters in the field Called Location is less than Field length then the remaining
characters in the field Called Location will be filled by spaces.
Called Number: This is the external number to which the call was made.
One way to separate the called party number is by Area Code, Exchange code and Subscriber
Number. This is difficult in an Open numbering system, in which the field size of area code, exchange
code are not standard but vary from two digits to four digits (for example, the Area code for 'Mumbai' is
of 2 digits, whereas that of 'Vadodara' is 3 digits).
In the Closed numbering system, the Area Code, Exchange Code and the Subscriber number are of
fixed length. In such case, including '-' in the called party number is not difficult. Hence, '-' is put in the
called party number. The called party number is assumed to be of 10 digits. The first '-' is placed after
four digits, counting from the right. The second '-'is placed after seven digits, counting from the right. If
the dialed number is a local number of 7 digits then the second '-'is not placed. Also, the remaining
three digits are not placed, but filled with character 'space'.
In this case, even if the call is made to a geographical area where open numbering system is followed,
'-' is placed in the same way.
Account Code: This is the Account Code (Refer Note4) using which the call was made.
Remarks: This column indicates the details of the call; whether it was a DISA call, DOSA call, Auto Redial
Call, type of call maturity.
Fixed Characters are used to indicate the type of call, call details, etc. The notations for the Remarks field
are:
D
DISA Call
CPD
12KHz/16KHz
Reversal
Delay
Connect
Reset Serial Number to 001: The Serial number counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial number
counter will not be automatically reset.
Starting Character - Increment Counter: Specify the starting character of the increment counter as the
serial number rolls over, in this field.
Reset Increment Counter: The Increment Counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial number
counter will not be automatically reset.
2241
Prefix String Required: This flag is to be programmed if the prefix string 0ac1 is to be sent when
interfacing with OG-SMDR Posting Protocol.
2242
In the Dialed Number String column, enter the number strings for each Call Type. You can enter the
prefix, e.g. 0 for long distance calls, 2 for local numbers, etc.
For each Dialed Number String, define a Call Type Indicator, this is an abbreviation of the Call Type,
e.g.: LD for long distance, INTL for International, etc.
Number String is of a maximum 4-digits. The Text String is a string of 4 alphanumeric characters. Your
entries may look like these:
Number Index
Dialed
Number String
01
LD
Long Distance
02
95
IC
Inter Circle
03
197
INFO
Information
04
00
INTL
International
36
Meaning
:
Local
You may enter as many Call Types as supported by the Posting Protocol you have selected.
Meaning
None
COM1
COM2
Printer Port
Ethernet Port
2243
Protocol Name
01
Blind Send
02
Matrix
03
Holidex
04
HOBIS A
05
HOBIS B
06
HOBIC
07
BELL HOBIC
08
MICROS A
09
MICROS B
10
Hilton
11
Xiox
12
Comm One
13
Call-Inn
14
RSI-CMS
15
Customized
2244
8333-Code
Where,
Code
Meaning
Abort
Start
Exit SE mode.
Handshaking Parameters
Enter SE mode.
To set ENQ no response timer, dial:
8302-ENQ No Response Timer (Response to ENQ Timeout)
Where,
ENQ No Response Timer is from 01-99 Seconds.
To set ENQ no response retry count, dial:
8303-ENQ Retry Count
Where,
ENQ Retry Count is from 01-99.
To set ENQ no response Retry Timer, dial:
8304-ENQ No Response Retry Timer
Where,
ENQ No Response Retry Timer is from 01-99 Seconds.
To set ENQ Retry Count, dial:
8305-ENQ Retry Count
Where,
ENQ Retry Count is from 01-99.
To set ENQ Retry Time, dial:
8306-ENQ Retry Timer
Where,
ENQ Retry Time is from 01-99 Seconds.
To Response to Data Timeout, dial:
8307-Response to Data Timeout
Where,
Response to Data Timeout is from 01-99 Seconds.
2245
Meaning
Disable
Enable
2246
BCC Flag
Meaning
Disable
Enable
Exit SE mode.
Enter SE mode.
To program column position for serial number, dial:
8100-Column Position
Where,
Column Position is from 000 to 128.
By default, Column Position is 001.
To program field length for serial number, dial:
8101-Field Length
Where,
Field Length is from 000 to 128.
By default, Field Length is 004.
To program alignment for serial number, dial:
8102-Alignment
Where,
Alignment
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for serial number, dial:
8103-Fill Character
Where,
Fill Character is 3 digit ASCII value ranging from 032 to 254.
By default, Fill Character is 'Zero'.
2247
Meaning
No Compulsory Reset
Meaning
No Compulsory Reset
2248
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for extension number, dial:
8113-Fill Character
Where,
Fill Character is 3 digit ASCII value and ranging from 032 to 254.
By default, Fill Character is 'Space'.
To program column position for trunk number, dial:
8114-Column Position
Where,
Column Position is from 000 to 128.
By default, Column Position is 012.
To program format type for trunk number, dial:
8115-Format Type
Where,
Format Type
Meaning
Matrix Format
Check-Inn Format
2249
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for date field, dial:
8119-Fill Character
Where,
Fill Character is 3 digit ASCII value from 032 to 254.
By default, Fill Character is 'Space'.
To program date format for date field, dial:
8120-Date Format
Where,
Date Format
2250
Meaning
01
DD-MM-YY
02
DD/MM/YY
03
DD.MM.YY
04
DD MM YY
05
DDMMYY
06
DD-MM-YYYY
07
DD/MM/YYYY
08
DD.MM.YYYY
09
DD MM YYYY
10
DDMMYYYY
11
MM-DD-YY
12
MM/DD/YY
Date Format
Meaning
13
MM.DD.YY
14
MM DD YY
15
MMDDYY
16
YY-MM-DD
17
YY/MM/DD
18
YY.MM.DD
19
YY MM DD
20
YYMMDD
21
YYYY-MM-DD
22
YYYY/MM/DD
23
YYYY.MM.DD
24
YYYY MM DD
25
YYYYMMDD
26
MM-DD
27
MM/DD
28
MM.DD
29
MM DD
30
MMDD
31
DD-MM
32
DD/MM
33
DD.MM
34
DD MM
35
DDMM
By default, the date format depends upon the Posting Protocol selected.
Meaning
Disable
Enable
2251
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for time field, dial:
8125-Fill Character
Where,
Fill Character is 3 digit ASCII value ranging from 032 to 254.
By default, Fill Character is 'Space'.
To program time format for time field, dial:
8126-Time Format
Where,
Time Format
Meaning
HH:MM:SS
HH:MM
Meaning
Disable
Enable
2252
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for duration field, dial:
8130-Fill Character
Where,
Fill Character is 3 digit ASCII value ranging from 032 to 254.
By default, Fill Character is 'Space'.
To program duration unit for duration field, dial:
8131-Duration Unit
Where,
Duration Unit
Meaning
HH:MM:SS
HHMMSS
Minutes
Seconds
Meaning
Disable
Enable
2253
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
To program fill character for units field, dial:
8135-Fill Character
Where,
Fill Character is 3 digit ASCII value ranging from 032 to 254.
By default, Fill Character is 'Space'.
To program column position for amount field, dial:
8136-Column Position
Where,
Column Position is from 000 to 128.
By default, Column Position is 068.
To program field length for amount field, dial:
8137-Field Length
Where,
Field Length is from 000 to 128.
By default, Field Length is 007.
To program alignment for amounts field, dial:
8138-Alignment
Where,
Alignment
Meaning
Left Alignment
Right Alignment
By default, Alignment is 2.
2254
Meaning
Higher Currency
Lower Currency
Meaning
Disable
Enable
Meaning
Left Alignment
Right Alignment
2255
Decimal
000
032
033
034
035
036
037
038
039
040
041
042
043
044
045
046
047
048
049
050
051
052
053
054
055
056
057
058
059
060
061
062
063
064
065
066
067
068
069
070
071
072
073
074
075
2256
ASCII
Null
Space
!
"
#
$
%
&
'
(
)
*
+
,
.
/
0
1
2
3
4
5
6
7
8
9
:
;
<
=
>
?
@
A
B
C
D
E
F
G
H
I
J
K
Decimal
076
077
078
079
080
081
082
083
084
085
086
087
088
089
090
091
092
093
094
095
096
097
098
099
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
ASCII
L
M
N
O
P
Q
R
S
T
U
V
W
X
Y
Z
[
\
]
^
_
`
a
b
c
d
e
f
g
h
i
j
k
l
m
n
o
p
q
r
s
t
u
v
w
x
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
Decimal
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
ASCII
Decimal
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
ASCII
Meaning
Left Alignment
Right Alignment
2257
Meaning
Left Alignment
Right Alignment
Meaning
Left Alignment
Right Alignment
Meaning
Continuous
Separated
2258
Meaning
Left Alignment
Right Alignment
Meaning
No
Yes
Meaning
Left Alignment
2259
Alignment
2
Meaning
Right Alignment
Country Code
001
041
002
051
200
096
2260
Index
Country Name
001
India
002
Kenya
200
UAE
Online: as and when a call is made or received (see Station Message Detail Recording-Online)
Or
Offline: whenever required, the records of calls stored in the buffer can be printed.
ETERNITY allows you to set a variety of filters for printing SMDR Reports.
ETERNITY supports SMDR-Reports on Serial RS232 Communication Port, Printer as well as on TCP/IP Ethernet
Port.
How to configure
To be able to generate SMDR -Report, you must do the following:
Enable SMDR Storage in the SMDR buffer. See Station Message Detail Recording-Storage.
Assign the Destination port for Incoming, Outgoing and Internal calls.
2261
2262
If you select Ethernet as the Destination Port, in the Destination IP Address: Port field, enter the IP
Address and the port of the remote Syslog Server.
If you select Ethernet as the Destination Port, in the Destination IP Address: Port field, enter the IP
Address and the port of the remote Syslog Server.
If you select Ethernet as the Destination Port, in the Destination IP Address: Port field, enter the IP
Address and the port of the remote Syslog Server.
Meaning
None
COM1
COM2
Printer Port
Ethernet Port
If you assigned Ethernet Port as destination port, to assign the IP Address to the Ethernet Port, dial:
2934-IP Address
By default, IP Address is 192.168.1.104
To assign the Port, dial:
2935- Port
Where,
Port is from 514 and 1025-65535
By default, IP Port is 514.
Meaning
None
COM1
COM2
Printer Port
Ethernet Port
If you assigned Ethernet Port as destination port, to assign the IP Address to the Ethernet Port, dial:
2834-IP Address
By default, IP Address is 192.168.1.104
To assign the IP Port, dial:
2835-IP Port
2263
Where,
IP Port is from 514 and 1025-65535
By default, IP Port is 514.
OG Calls Report
Destination Port
To assign a destination port for SMDR-OG Report, dial:
2731-Code
Where,
Code
Meaning
None
COM1
COM2
Printer Port
Ethernet Port
If you assigned Ethernet Port as destination port, to assign the IP Address to the Ethernet Port, dial:
2734-IP Address
By default, IP Address is 192.168.1.104
To assign the IP Port, dial:
2735-IP Port
Where,
IP Port is from 514 and 1025-65535
By default, IP Port is 514.
How to use
You can print SMDR Report whenever you want or schedule printing of the report from the System Administrator
mode using Jeeves or dialing SA Commands from an extension phone.
2264
Open Jeeves.
Outgoing Calls
2265
Set the following filters as desired. You can print records of outgoing calls Originating and outgoing calls
Terminating on specific Extension numbers, and trunks: CO, E&M, T1E1, BRI, Mobile, SIP and LD.
You can also print outgoing calls made using Account Code, Authority Codes and Calls for
Department Billing Groups.
2266
Enter the desired numbers in the Number List of your choice and select the same number list in Calls
made on called numbers matching with Number List.
To print outgoing calls made on a certain date or between a certain time period, set the filter Calls made
between. To print calls made on a particular date, select the same Date, Month and Year in both fields.
2267
To print outgoing calls made at a particular time, set the Hours and Minutes in 24-hour format calI in the
filter Calls made between 00: 00 and 23:59.
If you want outgoing calls that exceed as certain duration to be printed, set the filter Calls with duration
more than (sec) to the desired duration. All outgoing calls with duration greater than this value, will be
printed.
If you want outgoing calls exceeding certain metering units to be printed, set the filter Calls with units
more than (units) to the desired value. All outgoing calls that have metering units greater than this value
will be stored.
2268
To generate SMDR Report of outgoing calls on a particular day, day of the week, or day of the month, set
Scheduled Report Generation, as required.
You can print the SMDR Report of outgoing calls any time you want. To do this, click the Print Report
button.
Incoming Calls
2269
Set the filters as desired. You can print records of incoming calls received on a specific extension or a
range of extensions and trunk ports: CO, E&M, T1E1, BRI, Mobile, SIP and LD.
2270
You can also print incoming calls of different Call Types: Normal calls, calls received using Built-In Auto
Attendant, calls that remained Unanswered, Unanswered calls on Built-In Auto Attendant, and calls
made using DISA.
To print outgoing calls received from certain numbers, enter the CLI of these numbers in the Number
List of your choice and select the same number list in Calls received from calling numbers
matching with Number List.
To print incoming calls received on a certain date or between a certain time period, set the filter Calls
received between. To print calls made on a particular date, select the same Date, Month and Year in
both fields.
To print incoming calls received at a particular time, set the Hours and Minutes in 24-hour format calI in
the filter Calls received between 00: 00 and 23:59.
To print incoming calls that remained unanswered for more than a certain duration, set the filter Calls
remain unanswered for duration more than (sec).
2271
To print calls that were kept on hold for more than a certain duration, set the filter Calls kept on hold
with duration more than (seconds) to the desired value.
To print calls with speech duration of a certain duration, set the filter Calls with speech duration more
than (seconds)
2272
To generate SMDR Report of incoming calls on a particular day, day of the week, or day of the month, set
Scheduled Report Generation, as required.
You can print the SMDR Report of incoming calls any time you want. To do this, click the Print Report
button.
Internal Calls
To print Internal call Report with filters, click Internal Call - Print Filters link.
2273
To print calls made by a particular extension or a range of extensions, set the filter Calls made by
Stations, by entering the extension numbers in the From and To fields.
If you want to print calls made by a particular extension only, enter the same extension number in both
From and To fields.
You can also print calls made and calls received by these extensions by selecting the Call Type.
Select Both as Call Type, to print calls made and received by the extensions.
Select Caller as Call Type, to print only those calls that were made by the extension.
Select Receiver as Call Type, to print only those calls that were received by the extension.
Select None, if you do not want to use the Call Type filter.
2274
To print calls with speech duration of a certain duration, set the filter Calls with speech duration more
than (seconds)
To generate SMDR Report of internal calls on a particular day, day of the week, or day of the month, set
Scheduled Report Generation, as required.
2275
You can print the SMDR Report of internal calls any time you want. To do this, click the Print Report
button.
Commands
Values
1072-152-Flag
Flag:
0 = Disable
1 = Enable
1072-153-Flag
Flag:
0 = Disable
1 = Enable
1072-154-Flag
Flag:
0 = Disable
1 = Enable
2276
Filter
Commands
Values
1072-155-Flag
Flag:
0 = Disable
1 = Enable
1072-156-Flag
Flag:
0 = Disable
1 = Enable
1072-157-Seconds
Seconds:000-255
1072-158-Seconds
Seconds:000-255
1072-159-Seconds
Seconds:000-255
1072-161-CO-CO
CO: 000-128
1072-162-BRI-BRI
BRI: 00-32
1072-163-T1E1PRI-T1E1PRI
T1E1PRI: 0-8
1072-164-E&M-E&M
E&M: 000-128
1072-165-Mobile-Mobile
Mobile: 00-64
1072-166-SIP-SIP
SIP: 00-32
1072-167-DD-MM-YYYY-DDMM-YYYY
Date: 01-31
Month: 01-12
Year: 0000-9999
1072-168-HH-MM-HH-MM
HH: 00-23
MM: 00-59
1072-169-Number List
1072-116-LD-LD
LD: 00 - 32
1072-117-LD-LD
LD: 00 - 32
1072-118-LD-LD
LD: 00 - 32
Note: If you do not want to print calls for a particular type, set the start and end range as 0.
2277
Meaning
Abort
Start
Meaning
2278
Daily Reports
To program the time for daily scheduled reports, dial:
1072-173-HH-MM
Where,
HH and MM is in 24 hour format.
By default, HH:MM is 18:00.
Weekly Reports
To program the day and time for weekly scheduled reports, dial:
1072-174-Day-HH-MM
Where,
HH and MM is in 24 hour format.
Day is from 1 to 7 (1 is Sunday, 2 is Monday and 7 is Saturday).
By default, HH:MM is 10:00 and Day is 2.
Monthly Reports
To program the date and time for monthly scheduled reports, dial:
1072-175-Date-HH-MM
Where,
HH and MM is in 24 hour format.
Date is from 01 to 31.
By default, HH:MM is 10:00 and Date is 01.
ETERNITY provides a facility to Abort the scheduled report generation midway (1072-171-0). This aborts
the current report generation but does not affect any other scheduled report generation. The abortion of a
report does not affect its consecutive schedule. That is if a daily report is aborted on Monday, it does affect
the report generation schedule of Tuesday.
Command
Value
1072-138-Seconds
Seconds: 000-999
2279
Meaning
Abort
Start
By default, flag is 0.
Once the start command is issued, the report generation stops only after the complete report based on
the filters set is generated. ETERNITY provides a facility to abort the report generation in midway
(1072-141-0). Once the report generation is aborted, then it has to be explicitly started (1072-141-1) if
the report is required again.
Meaning
Daily Reports
To program the time for daily scheduled reports, dial:
1072-143-HH-MM
Where,
HH and MM is in 24 hour format.
By default, HH:MM is 18:00
2280
Weekly Reports:
To program the day and time for weekly scheduled reports, dial:
1072-144-Day-HH-MM
Where,
HH and MM is in 24 hour format.
Day is from 1 to 7 (1 is Sunday, 2 is Monday and 7 is Saturday).
By default, HH:MM is 10:00 and Day is 2.
Monthly Reports:
To program the date and time for monthly scheduled reports, dial:
1072-145-Date-HH-MM
Where,
HH and MM is in 24 hour format.
Date is from 01 to 31.
By Default, HH:MM is 10:00 and Date is 01.
ETERNITY provides a facility to abort the scheduled report generation in midway (1072-141-0). This
aborts the current report generation but does not affect any other scheduled report generation. The
abortion of a report does not affect its consecutive schedule. That is if a daily report is aborted on Monday,
it does affect the report generation schedule of Tuesday.
Commands
Values
1072-103-CO-CO
CO: 001-128
1072-104-BRI-BRI
BRI: 01-32
T1E1PRI : 1-8
1072-106-E&M-E&M
E&M: 001-128
1072-107-Mobile-Mobile
Mobile: 00-64
1072-108-SIP-SIP
SIP: 00-32
Dept Group: 00 to 99
1072-110-DD-MM-YYYY-DD-MMYYYY
Date: 01-31.
Month: 01-12.
Year: 0000-9999.
1072-111-HH-MM-HH-MM
HH: 00-23
MM: 00-59
2281
Filter
Commands
Values
1072-113-Seconds
Seconds = 000-999.
1072-114-Units
Unit = 0000-9999
1072-115-Account Code-Account
Code
1072-102-Flexible No.-Flexible
No.
1072 - 183 - CO - CO
CO = 000-128
BRI = 00-32
T1E1PRI = 0-8
E&M = 000-128
Mobile = 00-64
SIP = 00-32
1072-189
Meaning
Abort
Start
By default, Flag is 0.
Once the start command is issued, the report generation stops only after the complete report based on
the filters set is generated. ETERNITY provides a facility to abort the report generation in midway
(1072-121-0). Once the report generation is aborted, then it has to be explicitly started (1072-121-1) if
the report is required again.
Meaning
Daily Reports
To program the time for daily scheduled reports, dial:
1072-123-HH-MM
Where,
HH and MM is in 24 hour format.
By default, HH:MM is 18:00.
Weekly Reports
To program the day and time for weekly scheduled reports, dial:
1072-124-Day-HH-MM
Matrix ETERNITY System Manual
2283
Where,
HH and MM is in 24 hour format.
Day is from 1 to 7 (1 is Sunday, 2 is Monday and 7 is Saturday).
By default, HH:MM is 10:00 and Day is 2.
Monthly Reports
To program the date and time for monthly scheduled reports, dial:
1072-125-Date-HH-MM
Where,
HH and MM is in 24 hour format.
Date is from 01 to 31.
By Default, HH:MM is 10:00 and Date is 01.
ETERNITY provides a facility to abort the scheduled report generation in midway (1072-121-0). This
aborts the current report generation but does not affect any other scheduled report generation. The
abortion of a report does not affect its consecutive schedule. That is if a daily report is aborted on Monday,
it does affect the report generation schedule of Tuesday.
The Offline report for Incoming calls looks like shown below:
2284
21
415@192.168.50.2 V001
738 08-10-13
12:36:52
0
0
76 N
22
+918128680289
M001
593 08-10-13
12:42:36
5
2
10 N
23
C001
08-10-13
12:43:35
9
0
0 U
24
+918128680289
M001
542 08-10-13
12:42:52
0
0
409 N
25
C001
08-10-13
12:50:58 21
0
0 U
26
347@192.168.50.2 V001
729 08-10-13
12:50:59
0
0
25 N
27
430@192.168.50.2 V001
642 08-10-13
12:50:53
0
0
96 N
28
427@192.168.50.2 V001
599 08-10-13
12:52:06
0
0
25 N
29
C001
08-10-13
12:53:35 21
0
0 U
30
C001
08-10-13
12:54:45 12
0
0 U
31
C001
08-10-13
12:55:35 12
0
0 U
32
391@192.168.50.2 V001
707 08-10-13
12:57:19
0
0
1 N
33
303@192.168.50.2 V001
707 08-10-13
12:58:50
0
0
18 N
34
391@192.168.50.2 V001
707 08-10-13
13:08:38
0
0
31 N
35
300@192.168.50.2 V001
593 08-10-13
13:35:00
0
0
75 N
36
8128698957
P001
593 08-10-13
14:07:13
4
0
85 N
37
C001
08-10-13
14:15:54 12
0
0 U
38
314@192.168.50.2 V001
542 08-10-13
14:04:01
0
0
937 N
------------------------------------------------------------------------------TOTAL CALLS
: 875
TOTAL ANS DUR : 2417
TOTAL HOLD DUR : 244
TOTAL SPCH DUR : 72073
------------------------------------------------------------------------------D=Built-In Auto Attendant DU=Built-In Auto Attendant UnAnswered
R(CALL TYPE):N=Normal, U=UnAnswered, I=DISA, G = Gateway, Q=Qsig, T = Transfer
Additional Ports: X=MODEM, O=Door Phone, A=DIP
Extension: S=SLT, D=DKP, G=MAG, I=SIP Extn, R=Virtual Extn, N=ISDN Extn
Trunk
: C=CO, B=BRI, P=T1E1, E=E&M, M=MOB, V=SIP, L=LD
------------------------------------------------------------------------------Eternity V11R2
Page : 1
The Offline report for Internal calls looks like shown below:
2285
17
530
541
08-10-13 15:35:59
67
18
606
626
08-10-13 15:36:25
45
19
530
540
08-10-13 15:37:20
28
20
518
516
08-10-13 15:38:08
63
21
621
734
08-10-13 15:32:36
413
22
606
629
08-10-13 15:41:22
43
23
618
605
08-10-13 15:43:48
12
24
734
621
08-10-13 15:41:18
176
25
728
799
08-10-13 15:45:52
7
26
530
*3931
08-10-13 15:46:51
14
27
544
543
08-10-13 15:46:21
50
28
737
551
08-10-13 15:46:20
284
29
546
545
08-10-13 15:51:11
28
30
744
634
08-10-13 15:50:50
79
31
532
542
08-10-13 15:51:02
119
32
593
703
08-10-13 15:52:44
18
33
631
613
08-10-13 15:53:42
19
34
634
744
08-10-13 15:53:12
76
35
518
525
08-10-13 15:50:01
268
36
503
594
08-10-13 15:54:43
10
37
503
714
08-10-13 15:55:01
34
38
547
522
08-10-13 15:55:24
12
39
561
518
08-10-13 15:55:06
48
40
532
525
08-10-13 15:57:11
63
41
723
799
08-10-13 15:58:53
11
42
714
503
08-10-13 15:58:41
37
43
737
553
08-10-13 15:58:31
60
44
725
731
08-10-13 16:00:11
12
45
714
718
08-10-13 16:01:12
51
46
523
513
08-10-13 16:02:15
25
47
513
530
08-10-13 16:05:05
12
------------------------------------------------------------------------------Eternity V11R2
Page : 1
The Off line report for Outgoing calls looks like shown below:
Authrty Cod:
Accout No :
Units
:
Dur - Sec :
Department :
Number List:
Time
:
Date
:
Extension :
000 To 000
000 To 000
0000
000
00 To 00
02
00:00 To 23:59
01-05-2005 To 31-12-2037
000000 To 999999
LD
SIP
MOB
E&M
T1E1
BRI
CO
:001
:001
:001
:001
:001
:001
:001
To
To
To
To
To
To
To
032
032
064
128
008
032
128
Terminated on
LD
SIP
MOB
E&M
T1E1
BRI
CO
:001
:001
:001
:001
:001
:001
:001
To
To
To
To
To
To
To
032
032
064
128
008
032
128
2286
2
632 000 P001 8401514337
08-10-13 09:59:03
37
1
3.10 I
3
799 000 V001 342
08-10-13 09:59:44
49
1
3.10 I
4
593 000 V001 315
08-10-13 09:59:00
101
1
3.10 I
5
725 000 V001 389
08-10-13 10:03:35
47
1
3.10 I
6
713 000 P001 09687672493
08-10-13 10:13:09
9
1
3.10 I
7
713 000 P001 09687672483
08-10-13 10:13:57
12
1
3.10 I
8
713 000 P001 09687672482
08-10-13 10:14:39
10
1
3.10 I
9
713 000 P001 09687603954
08-10-13 10:15:24
8
1
3.10 I
10
632 000 V001 429
08-10-13 10:15:51
56
1
3.10 I
11
583 000 C002 8128680291
08-10-13 10:08:39
509
3
5.30 D
12
717 000 M002 9898572368
08-10-13 10:16:47
55
1
3.10 I
13
583 000 C002 3044782
08-10-13 10:17:41
20
1
3.10 D
14
639 000 V001 392
08-10-13 10:18:38
31
1
3.10 I
15
583 000 C002 3044782
08-10-13 10:18:19
80
1
3.10 D
16
583 000 C002 3044782
08-10-13 10:19:45
62
1
3.10 D
17
583 000 C002 3044782
08-10-13 10:21:00
107
1
3.10 D
18
591 000 V001 302
08-10-13 10:26:49
16
1
3.10 I
19
583 000 C002 3044782
08-10-13 10:26:53
154
1
3.10 D
20
583 000 C002 812860291
08-10-13 10:29:51
3
1
3.10 D
21
583 000 C002 8128680291
08-10-13 10:30:17
20
1
3.10 D
22
583 000 C002 8128680291
08-10-13 10:30:54
5
1
3.10 D
23
741 000 M002 9558874302
08-10-13 10:30:43
9
1
3.10 I
24
583 000 C002 8128680291
08-10-13 10:31:06
5
1
3.10 D
25
583 000 C002 8128680291
08-10-13 10:31:21
6
1
3.10 D
26
707 000 V001 389
08-10-13 10:31:25
12
1
3.10 I
27
593 000 M001 9601258612
08-10-13 10:30:56
14
1
3.10 I
28
583 000 C002 8128680291
08-10-13 10:31:56
256
2
4.20 D
29
707 000 V001 391
08-10-13 10:38:03
19
1
3.10 I
30
508 000 V001 371
08-10-13 10:38:50
39
1
3.10 I
31
583 000 M002 8128680291
08-10-13 10:37:03
140
1
3.10 I
32
583 000 C002 9726009016
08-10-13 10:39:53
18
1
3.10 D
33
591 000 V001 389
08-10-13 10:41:18
31
1
3.10 I
34
717 000 V001 324
08-10-13 10:40:33
130
1
3.10 I
35
782 000 M002 02653044583
08-10-13 10:43:11
4
1
3.10 I
36
583 000 M002 02653044782
08-10-13 10:44:34
202
2
4.20 I
------------------------------------------------------------------------------Total Calls
: 5997
Total Dur-Sec : 662320
Total Units
: 7723
Total Amount : 20489.30
------------------------------------------------------------------------------AuC = Authority Code
MATURITY: C = CPD, R = Reversal, D = Delay, I = Connect
R(CALL TYPE):O=DISA,A=Auto Redial,E=External Forwarded,G=Gateway,T = Transfer
Additional Ports: X=MODEM, O=Door Phone, A=DIP
Extension: S=SLT, D=DKP, G=MAG, I=SIP Extn, R=Virtual Extn, N=ISDN Extn
Trunk : C=CO, B=BRI, P=T1E1, E=E&M, M=MOB, V=SIP, L=LD
------------------------------------------------------------------------------Eternity V11R2
Page : 1
2287
How to configure
To enable storage of SMDR of Outgoing, Incoming and Internal Calls, you must enable this feature in the system
and set the storage filters as per your requirement.
2288
To configure storage filters for outgoing calls, click SMDR Storage - Outgoing Calls to expand,
Select the Store Outgoing Calls check box to enable storage of outgoing calls as per the filters you
set. If outgoing call storage is disabled, no outgoing call will be stored.
2289
You can limit the storage of calls to certain numbers. Select a number list and enter the desired called
party numbers in the selected list. In Store Calls of Called Number matching with Number List,
select the same list number.
If you want outgoing calls that exceed as certain duration to be stored, set the filter Store Calls with
speech duration more than (sec) to the desired duration. All outgoing calls with duration greater than
this value, will be stored.
If you want outgoing calls exceeding certain metering units to be stored, set the filter Store Calls with
metering units more than (units) to the desired value. All outgoing calls that have metering units
greater than this value will be stored.
Outgoing calls made by an extension user can be transferred to another extension. In such cases, you
may enable Call Splitting if you want to charge the amount to each extension according to the duration
of speech that each extension was involved in the call.
If Call Splitting is disabled, you have the option of charging the call amount either to the extension that
originally made the call, i.e. the Originating Extension, or to the extension that was last in speech on the
call, i.e. the Terminating Extension.
In the When Call Splitting is OFF, charge calls to field, select the desired extension you want to
charge the call to as Originating Extension or Terminating Extension.
2290
To configure storage filters for incoming calls, click SMDR Storage - Incoming Calls to expand,
Select the Store Incoming Calls check box to enable storage of incoming calls as per the filters you
set. If incoming call storage is disabled, no incoming call will be stored.
If you want incoming calls that exceed as certain duration to be stored, set the filter Store Calls with
speech duration more than (sec) to the desired duration. All calls with duration greater than this
value, will be stored.
If you want incoming calls that remain unanswered for certain duration to be stored, set the filter Store
Calls remaining un-answered for more than (sec) to the desired duration. All calls with duration
greater than this value, will be stored.
If you want incoming calls that were kept on hold for a certain duration to be stored, set the filter Store
Calls kept on hold for more than (sec) to the desired duration.
If you want all calls, except calls received using Auto Attendant to be stored, select the Store Normal
Calls check box.
If you want calls received on Auto Attendant to be stored, select the Store calls received on Built-In
Auto Attendant check box.
If you want all calls, that remained unanswered to be stored, select the Store Unanswered Calls
check box.
If you want all calls received using Auto Attendant that remained unanswered to be stored, select the
Store Unanswered calls from Built-In Auto Attendant check box.
If you want calls made using DISA, select the Store DISA Calls check box.
To configure storage filters for internal calls, click SMDR Storage - Internal Calls to expand,
Select the Store Internal Calls check box to enable storage of internal calls as per the filters you set.
2291
To store internal calls that exceed as certain duration, set the filter Store Calls with speech duration
more than (sec) to the desired duration. All internal calls with duration greater than this value, will be
stored.
Meaning
By default, all the calls are stored as per the filters set.
Commands
Values
2902-Flag
Flag:
0 = Dont Store
1 = Store
2903-Flag
Flag:
0 = Dont Store
1 = Store
2904-Flag
Flag:
0 = Dont Store
1 = Store
2905-Flag
Flag:
0 = Dont Store
1 = Store
2906-Flag
Flag:
0 = Dont Store
1 = Store
2907-Seconds
Seconds:000-999
2908-Seconds
Seconds:000-999
2909-Seconds
Seconds:000-999
2292
Meaning
By default, all the calls are stored as per the filters set.
Command
Value
2802-Seconds
Seconds: 000999
Meaning
2293
Destination Wise
It is possible to store outgoing calls selectively depending on the destination numbers. The ETERNITY supports
this feature in association with Number Lists. An outgoing call will be stored only if the number matches with an
entry in the Number List assigned.
To assign a Number list containing numbers for call storage, dial:
2702-Number List
Where,
Number List is 01-16.
By default, Number List assigned is 02.
Duration wise
Sometimes it is required to filter out the calls of small durations. System will not store the calls with duration less
than duration, programmed.
To set the filter of call duration, dial:
2703-Seconds
Where,
Seconds is from 000 to 999.
By default, Seconds is 000.
Unit wise
Sometimes it is required to filter out the calls based on Call units. System will not store the calls with units less than
the units programmed:
To set the filter for call Units, dial:
2704-Units
Where,
Units are from 0000 to 9999.
By default, Units is 0000.
Call Toggle
To set the Call Toggle flag, dial:
2716-Toggle Flag
Where,
Toggle Flag
2294
Meaning
Toggle OFF
Toggle ON
Meaning
OFF
ON
How to use
The SMDR stored in the buffer can be cleared at any time from the System Administrator mode, using Jeeves or by
dialing SA commands from an extension phone.
To delete SMDR records using Jeeves,
Under SMDR Management, click SMDR - Delete Call Record to open the page.
To delete all records in the internal SMDR buffer, select Delete All Internal Calls check box.
To delete all records in the Incoming SMDR buffer, select the Delete All Incoming Calls check box.
If you want to delete all records in the Outgoing SMDR buffer, select the Delete All OG Calls radio button.
You can also delete records of outgoing calls selectively, i.e. delete only records of outgoing calls made by
a particular extension or a range of extensions, or calls made between a certain period.
To delete records of outgoing calls selectively, select the Delete Selective OG Calls radio button.
2295
To delete calls made by a particular extension or a range of extensions, select the Delete OG Calls
made by Stations button.
In the first edit box, enter the number of the first extension in the range. In the second box, enter the
number of the last extension in the range. If you want to delete the records of a particular extension,
enter the same extension number in both fields.
To delete the records of outgoing calls made on a particular date or during a certain period, select the
Delete calls made between radio button. Select the start and end Date, Month and Year for this
period. If you want to delete the records of a particular date, enter the same date as start and end.
The SMDR buffer will be cleared according to the settings you enabled on this page.
1072-132-DD-MM-YYYY-DD-MM-YYYY (The format of the date depends on the Date Format of the
system)
Exit SA mode.
2296
Report (Offline): The activity report is printed/downloaded whenever desired. In the Offline mode, the last
500 activities recorded by the system are printed/downloaded.
The System Administrator can print/download System Activity Log, online or offline using any serial device
connected to the COM Port of ETERNITY.
ETERNITY also supports Syslog Client for System Activity Logs. The Syslog Client enables the system to send
activity logs in syslog format to the remote Syslog Server. You can view the logs on the remote server.
You may use Syslog for System Activity Log, if you have no spare COM Port on your ETERNITY or your system
does not have a COM Port, as ETERNITY PE3SS.
How it works
A destination port, serial or ethernet, must be assigned for activity logs. The system will send the activity
log to this port.
If the System Administrator extension is a DKP or an Extended IP Phone, a DSS Key can be assigned for
System Activity Log.
Whenever an activity is recorded by the system, the DSS key, if assigned for this feature on the System
Administrators DKP/Extended IP Phone extension, is turned ON.
The System Administrator can view the activity log by pressing the DSS key (if assigned). The DKP/
Extended IP Phone of the System Administrator will display the activity in this format:
DD-MM HH:MM <Activity Index>
The format of the Date will be DD-MM or MM-DD as per Date Format selected in the Real Time Clock
settings of the system.
2297
2298
Activity
Description
10
The date and time when the SMDR of the internal calls
buffer was deleted.
11
12
13
14
15
16
17
18
Event
Index
Activity
Description
19
20
21
Reserved
22
Reserved
23
24
25
26
27
28
29
30
Reserved
31
32
33
Reserved
34
Reserved
35
Reserved
36
37
38
39
40
41
2299
Event
Index
Activity
Description
42
43
44
45
46
Reserved
47
Reserved
48
Reserved
49
Reserved
50
51
SS-Slot Number, PP -Port Number, TT -Trunk Number, XXXXXX - Station Number, v-Version for VvRr, rRevision for VvRr, ZZZZ = Card Type, nnn = Authority Code.
For Firmware Version 10 Revision 8 refer to the following table for the Index of the Type of Activities
recorded in the System Activity Log
Event
Index
Activity
Description
28
29
30
31
When installed in the Hotel mode, the ETERNITY captures Hotel-Motel Activity Log. To know more, see the
ETERNITY Hospitality System Manual.
How to configure
The two functional parts of system activity log are: Storage and Report Generation in the Online or Report modes.
To be able to use this feature, you must enable storage of Activity Logs, and assign the Syslog Server address as
Destination Port for the logs.
2300
The destination port may be a COM Port, Ethernet Port or Printer Port315.
If you want to use Syslog Server, you must assign the IP Address of the remote Syslog server as the Destination
Port for the System Activity Log.
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System
Activity Log.
For instructions on configuring a DSS key on a DKP, see DSS Keys Programming.
For instructions on configuring a DSS key on an Extended IP Phone, see Configuring SIP Extensions.
Open Jeeves
By default, System Activity Log Storage is Enabled. Select Disable, if you do not want the system to
keep a record of all the system activities.
To generate System Activity Log - Online, that is, as and when the activity occurs, select Destination Port
for Online SAL from the following options:
Comm. 1 or Comm. 2: Select a communication port if you want to use a serial device to capture the
logs. Make sure the device is connected to the COM port.
Printer: Printer port is available on ETERNITY ME only. Select this option if you want the logs to be
sent to the printer port of ETERNITY ME.
2301
Ethernet: Select Ethernet port if you want to use the remote syslog server for the logs.
If you select Ethernet, in the Destination IP Address: Port - Online SAL field, enter the IP
Address and the port of the remote Syslog Server. Valid port range is: 1025 to 65535.
To generate System Activity Log - Report, that is, offline, whenever desired, select Destination Port for
SAL Report from the options Comm. 1 or Comm. 2, Printer, Ethernet. Default: None.
If you selected Ethernet as the Destination port, in the Destination IP Address: Port - SAL Report
field, enter the IP Address and the port of the remote Syslog Server. Valid port range is: 1025 to
65535.
0 for None
1 for COM Port 1
2 for COM Port 2
3 for Printer port
4 for Ethernet Port
Default: None
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port, dial:
6406-IP Address
To assign Port for the IP Address, dial:
6407-Port
Where,
Port is from 514 and 1025-65535
Default: 514.
To restore default values of the system activity log parameters, dial:
6410
Exit SE Mode.
How to use
You can start and stop System Activity Log - Online and Report from the System Administrator mode using Jeeves
or dialing SA Commands from an extension phone.
To start/stop report generation using Jeeves,
Open Jeeves.
2303
To clear System Activity Logs from the buffer, click the Clear System Activity Log button.
2304
Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
The Online System Activity Log report would look like this:
26-04-2013 16:51:48 DKP Normal
: 3009
, Slot=07, Port=05
The Offline System Activity Log report would look like this:
2305
How to use
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Activity Log Display.
For instructions on configuring DSS Keys on a Digital Key phone, see DSS Keys Programming.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see Configuring SIP Extensions.
Go Off-hook.
Press the DSS key assigned to System Activity Log Display.
OR
Dial 1072-009
The last recorded Activity log appears on your phones display in this following format: Date-Time-Activity
Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 23.
See System Activity Log Activity Index table in System Activity Log.
The date and month format will be DD-MM or MM-DD as per date format set in the system. See Real Time
Clock (RTC) for setting the date format.
2306
Report (Offline): The faulty report is printed/downloaded whenever desired. In the Report (Offline) mode,
the last 100 faults recorded by the system are printed/downloaded.
The System Administrator can print/download System Fault Log, Online or Report using any serial device
connected to the COM Port of ETERNITY.
Matrix ETERNITY also supports Syslog Client for System Fault Logs. The Syslog Client enables the system to
send fault logs in syslog format to the remote Syslog Server. You can view the logs on the remote server.
You may use Syslog for System Fault Log, if you have no spare COM Port on your ETERNITY or your system does
not have a COM Port, as in ETERNITY PE3SS.
How it works
A destination port for sending the report must be selected to which the system can send the log.
If the System Administrator extension is a DKP or an Extended IP Phone, a DSS Key can be assigned for
System Fault Log.
Whenever a fault is detected, the LED of the Fault Log DSS key, if assigned, is turned ON. The buzzer of
the Master Card of ETERNITY ME starts sounding.
If more than one DKP/Extended IP extension is assigned Fault Log DSS Key, the LED of all keys will be
turned ON.
The System Administrator must acknowledge the Fault indication by pressing the Fault Log key or by
dialing the Fault Log access code. The LED of the Fault Log key is turned OFF. The buzzer of the Master
Card of ETERNITY ME stops sounding.
The different fault events that are logged are summarized in this table:
Event
ID
1
Supported in
Event
Description
ETERNITY
ME
ETERNITY
GE
ETERNITY
PE
For all
cards
For VoIP
card and
VMS card
For VoIP
card and
VMS card
2307
Supported in
Event
ID
Description
Event
DKP Absent :
Reserved
ETERNITY
ME
ETERNITY
GE
ETERNITY
PE
For all
DKP types
For all
DKP types
For BRI
ports
For BRI
ports
For BRI
ports
For T1E1
ports
For T1E1
ports
For T1E1
ports
For CPU
Card
, Slot=SS, Port=PP
For Master
Card
Communication fails
between the
communication
manager and any
slave card.
For Master
Card
For VoIP
ports
For VoIP
ports
For VoIP
ports
Causea=CC
Reserved
10
NA
NA
11
For VoIP
ports
For VoIP
ports
For VoIP
ports
12
SIP Trunk
registration failed
due to expiry of the
Registration Timer.
For VoIP
ports
For VoIP
ports
For VoIP
ports
13
Reserved
14
Reserved
15
2308
NA
For all
cards
(except
VoIP)
NA
Supported in
Event
ID
16
Event
Description
ETERNITY
ME
NA
17
18
19
20
Reserved
21
Reserved
22
ETERNITY
GE
ETERNITY
PE
For all
cards
(except
VoIP)
NA
VMS Card
VMS Card
VMS Card
VoIP Card
VoIP Card
VoIP Card
VoIP Card
VoIP Card
VoIP Card
For T1E1
ports
For T1E1
ports
For T1E1
ports
23
For BRI
ports
For BRI
ports
For BRI
ports
24
SMS Server
Scheduled backup
file is deleted
a. For Comm Manager Failures, refer to Comm Manager Failure Cause List, at the end of this topic.
b. For SMTP Error Code details see SMTP Errors at the end of this topic.
SS-Slot Number, PP -Port Number, TT -Trunk Number, CC-Cause Number, XX-Code Number, NNNNNNFlexible Number (Access Code).
VOPP Fail: If VoPP fail message is logged, the ETERNITY VoIP card will not be functional.
Registration Timer Fail: The system may fail to load either the Re-registration Timer or the Registration
Retry Timer. In such a case the Proxy SIP trunk will remain un-registered and will not be functional.
The system will decode the registration status message received from the VoIP module and, if it is found to
be a problem caused by Registration Timer Failure, this will be logged in the System Fault Log.
This can happen to one or more SIP trunks, while the other SIP Trunks are functioning normally. You need
to restart the system to resolve the problem.
2309
For Firmware Version 10 Revision 8 refer to the following table for the events that are logged
Supported in
Event
ID
13
14
Event
Description
ETERNITY
ME
ETERNITY
GE
ETERNITY
PE
NA
For all
cards
(except
VoIP)
NA
NA
For all
cards
(except
VoIP)
NA
How to configure
To be able to use this feature, you must enable storage of Fault Logs, and assign a Destination Port for the Fault
Logs. The destination port may be a COM Port, Ethernet Port, or Printer port (available on ETERNTIY ME only).
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System Fault
Log.
For instructions on configuring DSS Keys on a Digital Key phone, see DSS Keys Programming.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see Configuring SIP Extensions.
You may configure the System Fault Log settings using Jeeves and by dialing system commands from a telephone
connected to the ETERNITY.
2310
Open Jeeves
By default, System Fault Log Storage is Enabled. Select Disable, if you do not want the system to keep
a record of all the system faults.
To generate System Fault Log - online, that is, as and when the fault occurs, select the Destination Port
for Online SFL from the following options:
Comm. 1 or Comm. 2: Select a communication port if you want to use a serial device to capture the
logs. Make sure the device is connected to the COM port.
Printer: The printer port is available only on ETERNITY ME. Select this option if you want the logs to
be sent to the printer port of ETERNITY ME.
Ethernet: Select Ethernet port if you want to use the remote syslog server for the logs.
If you select Ethernet, in the Destination IP Address: Port - Online SFL field, enter the IP Address
and the port of the remote Syslog Server. Valid port range is: 1025 to 65535.
To generate System fault Log - Report, that is, offline, whenever desired, select Destination Port for SFL
Report from the options: Comm. 1 or Comm. 2, Printer, Ethernet. Default: None.
If you selected Ethernet as Destination Port, in the Destination IP Address: Port - SFL Report field,
enter the IP Address and the port of the remote Syslog Server. Valid port range is: 1025 to 65535.
2311
2312
Exit SE Mode.
How to use
You can start and stop System Fault Log - Online and Report from the System Administrator mode using Jeeves or
dialing SA Commands from an extension phone.
To start/stop fault report generation using Jeeves,
Open Jeeves.
To clear System Fault Logs from the buffer, click the Clear System Fault Log button.
2313
Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
Column Position
Blank
00
01
Time (HH:MM:SS)
12
21
Event Description
24
: 3009
, Slot=07, Port=05
SMTP Errors
The VMS may fail to send emails. These email failures are logged into the System Fault Log with a specific code.
The table below describes the meaning of the codes.
Error Code
Error Description
SMTP Client Mail Failure Errors
2314
01
02
03
04
05
06
07
08
Error Code
Error Description
09
10
11
12
13
14
15
16
17
18
19
20
51
The calling process does not have write permission on the message
queue, and does not have the CAP_IPC_OWNER capability
52
The message can't be sent due to the msg_qbytes limit for the queue and
IPC_NOWAIT was specified in msgflg
53
54
55
56
57
The system does not have enough memory to make a copy of the
message pointed to by msgp
Error Description
01
02
03
04
05
06
07
08
2315
How it works
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Fault Log.
When a fault occurs, the LED of the DSS Key assigned for the System Fault Log, glows.
The buzzer of the ETERNITY ME Master Card starts sounding.
The System Administrator may press the DSS key or dial the System Fault Log feature access code to
acknowledge it.
On pressing the DSS Key or dialing of the acknowledgment command, the LED of the Fault Log key is
turned OFF.
The buzzer of the ETERNITY ME Master Card stops sounding.
How to use
To view the System Fault Log from System Administrator Mode,
Go Off-hook.
The Fault log appears on your phones display in this format: Date-Time-Fault Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 12.
See System Fault Log Activity Index table in System Fault Log.
2316
System Parameters
System Parameters are general parameters, related to features and facilities that are applied system-wide, such as
customer name, Day-Night mode, storage of call logs, end of dialing, alarms, Built-In Auto Attendant call disconnect
options, Presence, and DND messages. Each of these is described briefly along with the instructions for
configuring them using the Jeeves and from a telephone.
How to configure
Configuring System Parameters using the Jeeves
Under Configuration, click System Parameters. The System Parameters page opens.
As Customer Name, you may enter the name of the organization that is using ETERNITY. The Customer
Name may contain up to 80 characters. You may enter the address of organization along with the name.
The Customer Name you assign will appear on the various System Reports generated and printed by the
ETERNITY. Default: Blank.
You can assign Customer Name also on the Configuring System Pre-requisites page. If you have
entered the Customer Name on this page, the same Name will appear on the System Parameters page.
2317
System Parameters
Click System Parameters to expand. To view more parameters, use the vertical scroll bar on your right.
ETERNITY two major applications: Enterprise application to meet requirements of businesses, and
Hospitality application to meet the specific requirements of Hotels and Hospitals.
You must select Customer Profile as Enterprise or Hotel according to the application you are using.
When you select the Customer Profile, all the features and facilities specific to the application
Enterprise/Hotel along with their default settings are loaded. By default, the Customer Profile of
ETERNITY is defined as 'Enterprise'.
If you want the system to present for configuration only those trunks and extension ports that are
detected by it at Power-On for configuration, select the On-site Configuration check box. Default:
Disabled.
When you enable On-site Configuration, the Jeeves, will show the pages for only those trunks and
extension port types that are on board the system, that is, detected by the system at Power-On.
2318
Station Name Pattern: The Station Name Pattern is the format in which the names of extensions will
be stored on the extension phones and displayed to other extensions. You can store names by First
Names only, First names and Last Names. You can also add Titles indicating gender, designation,
rank, social standing, like Mr. Mrs. Ms., Dr., Prof. Cmdr., Rev., to Names of extensions.
ETERNITY supports the following Station Name patterns.
Option
Meaning
Title<space>First Name<space>Name
Name only
First Name<space>Name
Title<space>Name
Station Name Pattern must be configured for the Guest Name and Title feature of the ETERNITY
Hospitality module. To know more, refer the feature description in the ETERNITY Hospitality System
Manual.
By default, Name Only is selected as the Station Name Pattern when ETERNITY is operated in the
Enterprise mode, and Title<space>Name is selected as the Station Name Pattern when ETERNITY is
operated in the Hotel Mode.
Default Call Hold Type: This parameter is related to the Call Hold feature of ETERNITY. You can
select Global Hold or Exclusive Hold. Default: Exclusive Hold.
To have ETERNITY store also internal calls in the Missed Call Log, select the Store Internal Calls in
Missed Call Log check box. To know more about this feature, see Call Logs. Default: Enabled.
You may have ETERNITY store internal calls in the Dialed Call Log by selecting the Store Internal
Calls in Dialed Call Log check box. To know more about this feature, see Call Logs. Default:
Enabled.
To have the system store internal calls in the Answered Call Log, select the Store Internal Calls in
Answered Call Log check box. To know more about this feature, see Call Logs. Default: Enabled.
In the MoH Source when Station kept on Hold box, keep the option Internal (VM-01). To know more,
read the feature description for Music on Hold (MOH). ETERNITY will play the Music-On-Hold
recorded in the Voice Module Number 01 to the extension that is put on hold. Default: Internal (VM-01).
In the MoH Source when Trunk kept on Hold box, keep the option Internal (VM-01). To know more,
read the feature description for Music on Hold (MOH). ETERNITY will play the Music-On-Hold
recorded in the Voice Module Number 01 to the external callers who are put on hold. Default: Internal
(VM-01).
The DKP/SIP Extension users can set multiple call appearances on their phones. If you want the
system to play MOH to all the queued internal calls when the user extension is busy, select the Play
MOH to Queued Internal Calls on DKP/Extended IP Phone check box. To know more, read the
feature description for Music on Hold (MOH).
End of Dialing Digit is a single digit, on receipt of which, the system considers the number string
dialed by the extension users as the complete string. The system does not wait for further digits to be
2319
dialed, and dials out the number. The digits * (star) or # (hash/pound) are used to indicate end of dialing
to the system, as these are unique and distinguishable from the digits generally dialed by extensions
(0, 1, 2... to 9).
In the End of Dialing Digit box, enter the digit you want the system to consider as end of dialing.
Default: #
If you select the Give Off-hook Alert to Operator check box, the system will detect extensions that are
off-hook and ring on the Operator extension to alert the Operator about the state of the phone. This
alert is useful for detecting whether the handset of extension phones are placed correctly. Read the
feature description for OFF-Hook Alert to know more.
You can set the Time Zone of the system as Working-Hours or Break Hours or Non-Working hours any
time you want by setting the Day/Night Mode. You can set the system in the Day Mode or the Break
Hours Mode or the Night Mode, or let the system Operate as per the Time Table assignment316.
For more details see Day Night Mode and Time Tables. Default: Operate System as per Time Table
assignment.
To have the system detect the extension that has made an emergency call, select the Emergency
Dialing Reporting check box.
When this flag is enabled, the system detects the extension that has made the emergency call and
reports it to the Operator extension. Thus the Operator can know which extension has made an
emergency call. To know more about this feature, see Emergency Detection and Reporting.
If you want to remove the + prefix in the CLI of the calling party (presented by the Mobile Network) and
replace it with another string, select the Replace '+' from CLI check box. Default: Disabled.
If you want to program the number string with which the + prefix is to be replaced, in the Replace '+'
from CLI with the number string field, enter the desired number string.
If you keep the number string field blank, ETERNITY will remove '+' sign from the CLI of calling party
and present the remaining digits on the CLI of the Called Party.
For example:
The number string +919327237228 is received as CLI.
If 00 is configured as the replace string, the CLI number would become 00919327237228
If no replacement string is configured (that is, left blank), the CLI number would be presented as
919327237228.
To disconnect Built-In Auto Attendant Calls when the landing extensions are busy, select the
Disconnect Built-In Auto Attendant Call, when Dialed Number is Busy check box. When you
enable this flag, the call will not be routed to the Operator. Instead, it will be disconnected. Default:
Disabled.
To disconnect Built-In Auto Attendant Calls when there is no reply from the landing destination
extensions, select the Disconnect Built-In Auto Attendant Call, when Dialed Number is not
Responding check box.
316. Certain features of the ETERNITY require extensions and trunks to behave differently according to the working hours, break hours
and non-working hours, which are referred to as Time Zones. The Time Zones, are defined for the entire week in a Time Table.
Time Table is assigned to trunks, extensions and other time-zone dependant features.
2320
When this flag is enabled, the system disconnects the call if there is no reply from the landing
destination extensions. The call will not be routed to the Operator. Default: Disabled.
To disconnect Built-In Auto Attendant Calls if the caller fails to dial a digit, select the Disconnect BuiltIn Auto Attendant Call, when Caller Does not Dial any Digit check box.
When this flag is enabled, the system will disconnect the call if the caller fails to dial a digit within the
First Digit Wait Timer. The call will not be routed to the Operator. Default: Disabled.
If the Extension creating 3 party conference, disconnects during Conference, you can select
either to Transfer the call or Disconnect other parties.
If you select to Transfer the call, the 3-party conference is converted into a two-way speech
between the other two parties.
If you select to Disconnect other parties, all the parties involved in the 3-party conference are
disconnected.
To play a beep to participants of a conference to indicate inclusion of a new participant, select the Play
Beep when Conference/Dial-in Conference begins check box selected. Default: Enabled.
This flag is common for the features Conference-Multiparty, Conference Dial-In, Emergency
Conference, and Raid. When this flag is enabled,
the system plays a warning beep to the extension which is being raided by another extension,
before establishing three-way speech.
the system plays beeps to the other participants in a Dial-In Conference when a new participant
joins in (dials into an on-going Dial-In Conference)
the system plays beeps to the other participants connected in a Multi-Party Conference and an
Emergency Conference, when a new participant is included.
If you disable this flag, no warning Beep will be played in Raid, the existing participants in a Dial-In or
Multi-party conference will not hear any beep tone indicating the addition of a new participant.
Play Beep when Raid/Call Taping/Conversation Recording Starts is a common flag for the features
Call Taping and Conversation Recording. When this flag is enabled, the system plays Beeps to the
extensions/calling party and extension before it starts taping the call in the common mailbox or
recording the conversation in the extensions mailbox.
When this flag is disabled, no indication will be given to the opposite party when the call is being taped/
conversation is being recorded. Default: Enabled.
In Play Feature Tone in place of Dial Tone when Call Forward is Set, you can select whether you
want the system to play Feature Tone instead of Dial Tone to the extensions when Call Forward is set
on these extensions. When this flag is disabled, the system will play dial tone to the extension on which
Call Forward is set, whenever the extension goes Off-hook. Default: Enabled.
2321
Select the Ignore call forward set by member extension, when call is routed on Routing/Dept.
Group flag, if you want the system to place calls on member extensions in a Routing Group even if
they have set Call Forward.
Select Call Proceeding Tone for Multistage Dialing. Default: Network Tone.
This parameter is used in Multistage Dialing where you need to configure Pause and Wait for Answer in
the Substitute Number string for the number string dialed by the extension users.
When an extension user makes a call using a Calling Card, and the system dials out the number in
stages, the extension user will get Ring Back Tone twice; first after the system has dialed the Calling
Card Number, and again after the system has dialed out the destination number (called party number).
Thus the extension user will get Ring Back Tone, twice. To avoid this, you may configure the 'Call
Proceeding Tone' to be played by the system when using Multi-Stage Dialing.
You must configure the type of 'Call Proceeding Tone', according to your requirement; whether the
extension user should be connected to the speech path when the Calling Card number is out dialed or
when the called party number is out dialed. You can select any of the following Call Proceeding Tone
options, as per your requirement:
Network Tone: If this option is selected, the extension user will get Ring Back Tone after dialing the
calling card number and again, after the system has dialed the called party number (when the
system is dialing out the number with Pause and Wait for Answer configured in the substitute
number string).
Pseudo Tone: If this option is selected, the extension user will get Feature Tone when the user has
completed dialing all the digits. At the end of the tone, the extension user gets connected to the
called party (destination number).
Silent: If this option is selected, the extension user will get Silence (no tone), after the extension
user has completed dialing all digits. After dialing out the called party number in DTMF, the system
will connect the caller to the called party number (destination number).
You may select the Companding Algorithm according to the Regulatory Requirement of the country
where ETERNITY is installed. Default: A-law
The companding Algorithm A law or lawis automatically selected when you select Region for
ETERNITY. However, if necessary, you may change the default companding Algorithm that appears in
this field.
If you change the Companding Algorithm, the terminals - EON 48D, EON 310 and Turret will restart.
You can select the Language of SE, SA and Front Desk User Web Interface as per your
requirement. The GUI of ETERNITY supports the languages English, Italian, Spanish, French,
German, and Portuguese. When you select Region for ETERNITY, one of these languages will be
applied as appropriate for the region you selected. For instance, if you selected India, English will be
applied. If you selected Spain, Spanish will be applied. If you selected a country where none of these
languages are the local language, English will be applied.
The language set by the system on Region selection will be applied on the pages of the GUI for every
login session. You can change the default language set on Region selection, by configuring this
parameter.
2322
To print each system report on a separate page, keep the Form Feed in Report Printing check box
enabled. Default: Enabled.
To be able to distinguish between incoming calls from the Public Network and those from the Private
Network on the basis of the number digits received in the CLI, enter the desired number of digits in the
Minimum No. of digits received in CLI to consider the call is from Public N/w box.
This parameter is applicable to calls originating on the E1-PRI ports (Tie-Line) configured for the QSignaling. By default, CLI number with 8 or more digits will be considered as call from Public Network.
ETERNITY will check this parameter, whenever the incoming call is to be analyzed as call from PISN
(Private Integrated Subscriber Network) or non-PISN (Public Network Number).
If you want to enable the Operator Console to view the presence status of the extension they are
calling, select the Display Presence Status during Call on DKP/Extended IP Phone check box.
Default: Disabled.
To configure ETERNITY via its Serial COM port connected to a computer using communication
Software, select the Enable Programming through Comm Port check box.
If you have enabled Programming through Comm Port, in the Communication Port for Programming
box, select the COM Port to which you have connected the computer you will use for system
configuration.
ETERNITY supports two COM Ports, you may select Comm 1 or Comm 2.
To use the Watch dog function supported by ETERNITY, select the Enable Watch Dog check box.
ETERNITY supports the Watchdog function to detect and restart the system, whenever the system
hangs. Default: Disabled.
When Watch Dog function is disabled, you must manually restart the system when it hangs.
You may set the duration of the Master Buzzer. ETERNITY has a buzzer on the master card which
sounds whenever the system detects a fault. To know more, see System Fault Log.
You can set the duration of time for which the buzzer should sound by setting the Master Buzzer On
Time and OFF Time as Master Buzzer On Timer and Master Buzzer Off Timer respectively.
The range of the both timers is between 000 to 9999 milliseconds.
When there is an incoming call on the CO Trunk and if total 5 digits are dailed out, before expiry of
Trunk Inter Digit Wait Timer, then the system will treat this call as an outgoing fraudulent call. To drop
this call, select the Detect Possible toll bypass attempt by Extn. during IC Call from CO Line &
Drop the Call check box.
2323
Magneto
If you want calls from Magneto ports to be disconnected automatically, when silence is detected, select the
Enable Silence Detection on Magneto check box. Default: Enabled.
You may change the duration for the Magneto Silence Detection Timer, to the desired duration. The
range of this timer is from 01 to 32 seconds. Default: 30 seconds.
Set the Magneto VAD Threshold Level to the desired level. Default: -71(dBm).
Radio
2324
DTMF Detection - Minimum Level (dB): This parameter signifies the minimum level (dB) of the
DTMF digit to be considered as valid. By default, Minimum levels set to -10.5dB.
DTMF Detection - Minimum ON Time (msec): This parameter signifies the minimum time period for
which the DTMF signal should be present in order to be detected. The valid range of this time is 17 to
204 milliseconds. By default, Minimum ON Time is set to 17 milliseconds.
DTMF Detection - Minimum OFF Time (msec): This parameter signifies the minimum time period
between successive DTMF digits. The valid range of this time is 17 to 204 milliseconds. By default,
Minimum OFF Time is set to 17 milliseconds.
Matrix ETERNITY System Manual
DTMF Generation - DTMF ON Time (msec): It is the width of DTMF digit to be dialed out by the Radio
port. By default the ON Time is set to 102 milliseconds.
DTMF Generation - Inter Digit Pause Time (msec): When the Radio port dials out the DTMF digits, it
waits for the Inter Digit Pause Timer, while dialing the DTMF digits. By default the timer is set to 102
milliseconds.
The DTMF Detection and Generation parameters are applicable only when there are Dial Pads connected
to the Radio devices.
Alarm
Use Alarm with snooze: Enable this flag if you want to use the Snooze function for the Alarm Call.
Alarm Ring Timer (Sec.): You may change the time for which the Alarm Call will ring on the extension
phone and the time for which the Operator phone will ring to notify an unanswered Alarm Call.
Number of Alarm Attempts: You may increase or decrease the number of attempts the system
should make to serve an Alarm call.
Alarm Attempt Interval: You may increase or decrease the time gap between each attempt the
system makes to serve an Alarm call.
Configurable Alarm Type flag: Disable this flag, if you do not what the system to provide the Operator
and the extension users the option of setting 'Once Only' or 'Daily' Alarms. When this flag is disabled,
the system will allow only 'Once Only' alarms to be set.
Configurable Alarm Category: Disable this flag, if you do not want the system to provide the Operator
the option of setting 'Personalized' or 'Automated' Alarm calls. When this flag is disabled, the system
2325
will follow the 'Automated' Alarm call serving mechanism. The Operator will not be prompted to choose
between 'Automated' and 'Personalized' Alarm calls when setting Alarm calls for an extension phone.
Voice Guided Alarm Verification: Enable this flag if you want to enable extension users to confirm
the Time they have set for an alarm or the Date and Time they have set for a reminder. Default:
Disabled.
Distinctive Rings
Distinctive Rings are ringing patterns used for distinguishing between different types of call events, like
Internal Calls, Trunk Calls, Auto Call Back, Auto Redial, Alarm, Emergency call, Priority, etc. If you want to
customise the Ringing pattern, for a call event, select the desired Ring Type. For more details see
Distinctive Rings.
2326
Incoming CLI Modification is useful in countries where the Calling Line Identification (CLI) received by the
PBX extension users must be suitably modified before it can be used to dial out the number. To know
more, see Incoming CLI Modification.
If you receive CLI in dialable format, there is no need to use this feature. In such case, keep the flag
disabled. You do not need to program any of the CLI Modification parameters.
To apply Incoming CLI Modification, select the Enable Incoming CLI Modification check box.
Country Code: This is the Country Code of the country where ETERNITY is installed. The Country
Code helps ETERNITY detect whether the Incoming CLI received is a national or an international
number. Do not enter any prefix such as + or 00 for the Country Code. Default: 91 (India).
Area Code: This is the Area Code of the place where the ETERNITY is installed. The Area Code helps
ETERNITY detect whether the Incoming CLI received is a local number. Do not enter any prefix for the
Area Code. For example, if you want to enter Area Code for Mumbai, enter only 22. Do not enter the
prefix 0 to the area code. Default: 265 (Vadodara city).
International Prefix: These are digits required as Prefix for dialing International Numbers. The prefix
may be up to 5 digits, with numbers from 00000 to 99999. Default: 00.
National Prefix: These are digits required as Prefix for dialing long distance, National (within the
country) numbers. The prefix may be up to 5 digits, with numbers from 00000 to 99999. Default: 0.
Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose:
No (Area Code not required), if your public telephone network does not require the dialing of Area
Code for local numbers.
2327
Yes (Area Code is required), if your public telephone network requires you to dial the Area Code for
local numbers.
Yes, with Prefix Digit, if your public telephone network requires you to dial Area Code with a
particular Prefix for local numbers. If you select this option, you must also enter the prefix digits for
the area code for local calls in the Prefix Area Code field.
Clock Synchronization
The ETERNITY supports four clock sources for Clock Synchronization for the E1-PRI and BRI ports. To
know more, see Clock Synchronization.
You must select the Clock Source in the order of Priority from 1 to 4. By default, the Clock Source priority
is selected as follows:
Clock Source - Priority 1 - E1 - 001
Clock Source - Priority 2 - E1 - 002
Clock Source - Priority 3 - E1 - 003
Clock Source - Priority 4 - E1 - 004
Also set the Clock Synchronization Frequency as: 8KHz Derived, 8KHz, 2.048MHz, 1.54 MHz
2328
ETERNITY allows you to program DND different Text Messages, which extension users can select when
setting DND on their extensions. The DND text message they select is displayed to the calling extensions.
See Do Not Disturb (DND) to know more.
The default DND Text messages appear on your screen. You may customize these messages, according
to your requirement. Text Message can be of maximum 16 alphanumeric characters. All ASCII characters
except < > and (double quote) are allowed.
Message No.
Message Text
Do Not Disturb
Unavailable
In Meeting
In Conference
Try on Mobile
On Vacation
On Business Trip
Out of Office
With a Guest
Publish Message
Matrix ETERNITY System Manual
2329
ETERNITY offers 10 different text Messages to Publish Message, as listed in the table below. You can
customize message 6 to 9 to match your requirement. Text Message can be of maximum 16 alphanumeric
characters. All ASCII characters except < > and (double quote) are allowed.
Message No.
Message Text
Absent
Present
Auto Detect
Away
On the Phone
Do Not Disturb
I am Mobile
In Meeting
Out of Office
2330
When Auto Attendant is enabled on trunks, the greeting messages are played to the callers according to
the time of the day, morning, afternoon, evening.
If Built-In Auto Attendant is enabled on trunks, the system answers the call and plays the greeting
message as per the voice modules.
If Voice Mail Auto Attendant is enabled on trunks, the Voice Mail System answers the call and plays
the greeting message.
To know more about this feature see Auto Attendant.
You can set the desired Start Time for Morning, Afternoon and Evening greetings.
In Start Morning Greeting at set the start time for the Morning Greeting Message. Similarly, in Start
Afternoon Greeting at and Start Evening Greeting at set the start time for the Afternoon and Evening
Greeting Message respectively.
The time must be in HH:MM format. The valid range for Hours (HH) is 00 to 23 and for Minutes (MM) is 00
to 59. By default the time in Start Morning Greeting at is set to 00:00, Start Afternoon Greeting at is set
to 12:00 and Start Evening Greeting at is set to 16:00.
The system plays the Morning Greeting Message between the Morning and Afternoon Greeting Start time,
the Afternoon Greeting Message between the Afternoon and Evening Greeting Start time and the Evening
Greeting Message between the Evening and Morning Greeting Start time.
As the system plays the Evening Greeting Message between the Evening and Morning Greeting Start time,
to prevent the Evening Greeting Message from being played after midnight, you are recommended to set
the Morning Greeting Start time to 00:00 hrs.
2331
ETERNITY supports a Digital Output Port . A DC contactor (60VDC max.) can be connected to the DOP.
Any external relay based device, like a Door Lock opener, a siren, or a hooter can be interfaced with the
DOP via this DC contactor.
Set the Normal Contact Type for DOP-1 for the DOP as appropriate: Normally Open/Normally Close. By
default, the contact type for the DOP is 'Normally Open'.
The DOP is used for connecting a door lock relay device to be used in conjunction with the Door Phone
connected to the ETERNITY. When you connect a Door Lock release device, you must set the Normal
Contact Type of the DOP to Normally Open. Default: Normally Open. See Digital Output Port (DOP).
2332
If you have connected a sensor device to the DIP for an automated control application or for Security
Alarm and Reporting, you must configure the following port parameters
Instigation Signal: Depending on the application for which you are using the DIP, select the
appropriate instigation signal for the DIP as High or Low state.
'High' state signifies that the DIP is normally open. DIP instigation signal should be set as 'High'
when the sensor connected to the DIP keeps the Loop open and closes it to signal an event.
'Low' state signifies that the DIP is normally closed. DIP instigation signal should be set as 'Low'
when the sensor connected to the DIP normally keeps the Loop closed and opens/breaks it to
signal an event.
Minimum Instigation Time: This is the time for which the instigation signal from the sensor device
should remain present on the DIP to be recognized by the DIP as a genuine signal. The range of this
timer is from 01 to 99 seconds. By default the Minimum Instigation Time is set to 01 second. You may
set the 'Minimum Instigation Time' to the desired value.
To know more about the usage of the DIP, see Digital Input Port (DIP) and Automated Control
Applications.
2333
Customer Name
System Parameters
2334
To select the type of music to be played when stations are kept on hold, dial:
3552-Code
Where,
Code is
1 for Voice Module 01
2 for Analog Input Port
Default: 1.
To select the type of music to be played when trunks are kept on hold, dial:
3553-Code
Where,
Code is
1 for Voice Module 01
2 for Analog Input Port
Default: 1.
If all the extensions of the Routing Group you selected for Alarm Notification type are busy, the extension
user will be played MoH (MoH can be Voice Module 01 or through Analog Input Port).
2335
To enable/disable the Disconnect when Caller Doesn't Dial a Digit' flag, dial:
5338-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable.
To enable/disable the Disconnect Built-In Auto Attendant call, when Dialed Number Busy, dial:
5336-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable.
To enable/disable Disconnect Built-In Auto Attendant Call, when Dialed Number does Not Reply, dial:
5337-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable
To enable/disable Beep when Conference/Dial-In Conference, Raid starts, dial:
5331-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Enable
2336
To enable/disable Beep when Call Taping and Conversation Recording starts, dial:
5332-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Enable
To disable/enable Feature Tone in place of Dial Tone when Call Forward is set, dial:
5312-Feature Tone Flag
Where,
Feature Tone Flag is
0 for Disable
1 for Enable
Default: Enable
To select a Language for SE, SA and Front Desk User Web Interface, dial:
5319-Language
Where,
Language is
1 for English
2 for French
3 for German
4 for Spanish
5 for Portuguese
6 for Italian
Default: English (country specific)
2337
2338
To program minimum number of digits received in CLI to consider as call from the Public Network, dial:
5314-Minimum Caller ID Digits
Where,
Minimum Caller ID Digits is from 01 to 16.
Default:8
To set the ON/OFF time of the ETERNITY ME Master Card buzzer, dial:
5308-On Timer-Off Timer
Where,
ON Timer and OFF Timer are 0000 to 9999 milliseconds.
Magneto
To enable or disable the "Enable Silence Detection on Magneto?" flag, dial:
5357-Flag
Where,
Flag is
1 for Enable
0 for Disable
Default: Enabled.
To set Magneto-Silence Detection Timer, dial:
5356-Silence Detection Timer
Where,
Silence detection timer value range is from 001 to 255 seconds.
Default: 60 seconds.
To set the Magneto Threshold Level, dial:
5358 - Magneto VAD Threshold Level
Where,
The value of the Level is from 0 to -96.
Default: -25 dBm
2339
2340
Default: 0
Clock Synchronization
Meaning
Port Offset
05
T1E1
1-8
04
BRI
01-32
00
Null
000
T1E1-1
T1E1-2
T1E1-3
T1E1-4
2341
Meaning
8 KHz Derived
8 KHz
2.048 MHz
1.54 MHz
Default: 2.048 MHz for India and other countries except USA. For USA: 1.54 MHz.
DND Message
Do Not Disturb
Unavailable
In Meeting
In Conference
Try on Mobile
On Vacation
On Business
Trip
Out of Office
With a Guest
2342
Publish Message
You can customize Publish Messages using Jeeves only.
Exit SE mode.
2343
Auto Redial
Name
Description
Range
Default
0 to 255
60 seconds
0 to 255
45 seconds
0 to 255
45 seconds
0 to 255
5 tries
0 to 255
10 seconds
0 to 255
20 attempts
Description
Range
Default
0 to 255
60 seconds
0 to 255
5 seconds
0 to 255
5seconds
0 to 255
10 seconds
2344
Name
Description
Range
Default
0 to 255
30 seconds
0 to 255
15 seconds
0 to 255
5 seconds
Description
Range
Default
The time for which the system plays the Dial tone.
2 to 255
7 seconds
The time for which the system plays the Ring Back
Tone.
1 to 255
45 seconds
The time for which the system plays the Busy Tone.
1 to 255
7 seconds
The time for which the system plays the Error Tone.
1 to 255
30 seconds
Feature Confirmation
Tone Timer (sec.)
1 to 255
7 seconds
Programming Error
Tone Timer (sec.)
The time for which the system plays the Error Tone
when you have entered an invalid command string
while configuring a feature from a phone.
1 to 255
3 seconds
Programming
Confirmation Tone
Timer (sec.)
1 to 255
3 seconds
1 to 255
30 seconds
1 to 255
30 seconds
Description
Range
Default
0 to 255
20 seconds
2345
Name
Description
Range
Default
0 to 255
2 minutes
Description
Range
Default
Message Notification
Retry Count
0 to 15
Message Notification
Ring Timer (sec)
0 to 255
45 seconds
Message Notification
Interval (min)
1 to 255
5 minutes
Description
Range
Default
1 to 255
30 seconds
1 to 255
45 seconds
1 to 255
10 seconds
Trunk Reservation
Timer (min.)
1 to 255
10 minutes
1 to 255
30 seconds
1 to 255
30 seconds
1 to 255
2 minutes
Message Notification
Name
Other Features
Name
2346
Name
Description
Range
Default
2 to 255
2 minutes
1 to 255
3 minutes
1 to 255
10 seconds
0 to 255
10 attempts
1 to 255
30 seconds
1 to 255
30 minutes
1 to 255
2 seconds
2 to 255
7 seconds
1 to 255
25 seconds
1 to 255
3 seconds
1 to 999
60 seconds
2347
Name
Description
Range
Default
1 to 255
2 minutes
1 to 999
999 seconds
Release Conference if
Idle for more than (min.)
This is the time for which the system will wait for
participants of a Dial-In Conference to withdraw or
release themselves from the conference, one-byone. On the expiry of this timer, the system releases
the Dial-In Conference and frees the resource
occupied by this conference in the conferencing
circuit.
1 to 255
2 minutes
1 to 255
60 seconds
1 to 255
45 seconds
0-9
3 attempts
Emergency Reporting
Call - Ring Timer (min)
1 to 255
10 minutes
1 to 255
5 minutes
How to configure
Configuring Timers and Counts using Jeeves
2348
The Timers and Counts on this page are arranged by the name of the feature or function these are related
to.
Change the value of the Timer or Count by entering the desired duration or count in the respective fields.
Auto Redial
To set Auto Redial Normal - Timer, dial:
1704-Seconds
Where,
Seconds is from 000 to 255.
To set Auto Redial Normal - Count, dial:
1705-Count
2349
Where,
Count is from 000 to 255.
To set Auto Redial - Priority, dial:
1706-Seconds
Where,
Seconds is from 000 to 255.
To set Auto Redial - Count, dial:
1707-Count
Where,
Count is from 000 to 255.
To change Auto Redial RBT Wait Timer, dial:
1702-Seconds
Where,
Seconds is from 000 to 255 seconds.
Do not set this timer to less than 2 seconds.
317. Time for which the system demonstrates the tone/ring to the user.
2351
2415-Seconds
Where,
Seconds is from 001 to 255 seconds.
Default: 030 seconds.
DISA
To set DISA Idle State Timer, dial:
2420-Seconds
Where,
Seconds is from 001 to 255 seconds.
Default: 020 seconds.
To set DISA Inactivity Timer, dial:
2421-Minutes
Where,
Minutes is from 001 to 255 minutes.
Default: 002 minutes.
Other Features
To set Auto Call Back Ring Timer, dial:
3801-Seconds
Where,
Seconds is from 001 to 255 seconds.
Default: 030 seconds.
To set Interrupt Request Timer, dial:
3802-Seconds
Where,
Seconds is 001 to 255 seconds.
Default: 045 seconds.
To set Barge-In Timer, dial:
3803-Seconds
Where,
Seconds is 001 to 255 seconds.
Default: 010 seconds.
To set Trunk Reservation Timer, dial:
3804-Minutes
Where,
2352
2353
2354
Exit SE mode.
2355
For Firmware Version V10R10 and earlier, refer System Security - V10R10 and earlier.
The SE password is stored in the Master Card/CPU Card. If you forget the SE password, the only way
to restore the default SE password is to change the Jumper settings of the Master Card/CPU card.
You are advised to record and store the SE password at a safe place, where it can be accessed by you
(the System Engineer) to avoid the inconvenience of restoring the default SE password.
2356
Enter New Password. The new password can be a minimum of 4 characters to a maximum of 12
characters. All ASCII characters except '%', '#', '=', '+', '&', '/', '<', '>', 'Double Quote (")' and 'Space' and
digits 0 to 9 are allowed.
2357
Enter New Password. The new password can be a minimum of 4 digits to a maximum of 12 digits. The
valid digits are from 0 to 9. The default SE Password is 1234.
ETERNITY ME
2358
ETERNITY GE
ETERNITY PE
The default SE password will be restored to 1234. You can now enter the programming mode by dialing 1#911234 (the default password). You can also change the SE password again using Jeeves or by dialing a
command as described above.
If you change the default SE password (1234) again after you have reset SE Password (by changing the
Jumper on the Master Card/CPU card to AB position), the system will not store the new password, until
you change the Jumper back to the default BC position. So, make sure that you have recorded the new SE
password in a safe place, from where it can be retrieved. Change the Jumper back to the default BC
position.
2359
The SA Password for accessing Jeeves can only be changed using Jeeves.
2360
User Password
Extension Users can secure their respective stations/extensions from unauthorized use with a password unique to
each extension. The User password too is a combination of any four digits, from 0 to 9. The default User Password
is 1111, which each user can change from their respective extensions. Refer the topic User Password to know
more.
2361
Enter the new User Password in the field Change User Password to. The User Password may be a
combination of 4 digits. Valid digits: 0 to 9.
Click Submit button to save your new SA password.
You may log out of Jeeves.
2362
Extension users can also set their status as 'Absent' or 'Present' from their respective extension
phones. Refer User Absent/Present).
DKP extension users can also lock the keypad of their phones from the DKP Phone Menu. Refer
Digital Key Phone-Operation for instructions on navigating the phone menu.
It is also possible to lock/unlock the DKP keypad and set the user extension status as 'Absent'/'Present'
from a remote location using Direct Inward System Access (DISA).
2363
T1 Maintenance
Whats this?
The T1 system format logs which are useful for maintenance purpose of the PBX, consist of:
Error Counts (Performance Statistics or T1-Statistics)
Alarms
The ETERNITY supports T1-Statistics and Alarm-logs. This chapter explains both Statistics and Alarms
and Loop Back Tests for T1.
The T1 Maintenance consists of Error Counts (Performance Statistics), Alarms and Loop Back Test. This
is as per standards like G.704, G.706 and G.732. G.775 is also considered for detection of defect
conditions like LOS, LOF, AIS, etc. (Loss of Signal, Loss of Frames, Alarm Indication Signal).
Digital line can have transmission errors. All the errors will not generate an Alarm. Few severe errors will
generate Alarms. However, all the errors will be logged in the System Fault Log.
The SNIIC (subscriber Network interface integrated circuit) is used to interface T1 line to ETERNITY. It
supports error counters listed in the table given below. Each error detected by the ETERNITY ME Card
T1E1PRI/port will be sent to the system in form of an event.
The system will count these errors and prepare a statistical record if the condition matches.
For Example:
Severely Errored Seconds Count is incremented when one OOF event reaches the master or more than 320
framing errors reach the master. This statistical record is updated and maintained by the master.
Performance Statistics
Error Counters supported by SNIIC
2364
Framing Bit Error Counter: This counter is incremented on receipt of any error in the framing pattern. In
D4, FS errors are counted. (FS errors are counted if enabled). In ESF, any error in the 001011 framing
pattern increments this counter.
Out of Frame Counter: Out of Frame is the occurrence of a particular density of framing error events. For
D4 framing, OOF is declared when the receiver detects two or more framing errors within 0.75ms or two or
more errors out of five or fewer consecutive framing bits. It ends when there are fewer than two frame bit
errors within 0.75ms period. For ECF framing, OOF is declared when the receiver detects two or more
errors out of five or fewer consecutive framing bits. It ends when there are fewer than two frame bit errors
within 3ms period.
CRC-6 Error Counter: This counter is incremented when the received frame has CRC-6 errors. This is
applicable for ECF framing only.
Line Code Violation Error Counter: This counter is incremented when a bipolar violation error occurs or
when excessive zeroes event occurs.
Excessive Zeroes Error Event: For AMI-coded signal, it is an occurrence of seven contiguous zeroes.
Positive Slip Counter: This counter is incremented every time a positive slip occurs.
Negative Slip Counter: This counter is incremented every time a negative slip occurs.
No. of Seconds/Count
Errored Seconds
Bursty Errored Seconds
Severely Errored Seconds
Severely Framing Seconds
Unavailable Seconds
Positive Slip Seconds
Negative Slip Seconds
Loss of Frame Count
Line Errored Seconds
Excessive Zeroes Error Count
CRC-6 Error Count
Errored Seconds
For D4, it is defined as a second with one of the following:
It is defined as a second in which more than one but less than 320 framing errors (Frame sync. error in D4
and CRC error in ESF) have occurred but with:
2365
For ESF signals-It is a count of one second interval with one of the following:
Slip defects are not counted in SES. Also this is not incremented during an Unavailable second. It is defined
as a second in which either an OOF has occurred or 320 or more framing errors have occurred.
It is a second with either one or more OOF defects or a detected AIS defect.
Unavailable Seconds
It is defined as a second for which T1 service is unavailable. An unavailable state is declared at the onset
of 10 consecutive severely errored seconds and is cleared on onset of 10 consecutive seconds with no
severely errored seconds.
It is defined as a second for which a frame is repeated to account for frequency drift between ETERNITY
and the network.
It is defined as a second for which a frame is deleted to account for frequency drift between ETERNITY
and the network.
Loss of Frame is declared after 2.5 seconds of continuous loss of signal or OOF. LOF is cleared after 10
seconds of continuous no loss of signal or OOF.
A second in which one or more Line Code Violation error events were detected.
2366
FDL is used for communicating general maintenance information or for transmitting user defined
information within the T1 link. General maintenance information is in the form of Performance Message
Report which is generated by the ETERNITY ME Card T1E1PRI and depending upon the T1 FDL
Protocol, the Performance Message Report is sent every second or on request.
How to configure
The commands explained below should be referred as:
To program a single port: XXXX-1
To program a range of ports: XXXX-2
To program all the ports: XXXX-*
T1 FDL
T1 FDL can be enabled/disabled. This parameter is applicable only if Framing = ESF. If the Network (Public or
Private) to which the ETERNITY is connected does not support FDL then T1 FDL will be disabled.
Use following command to enable/disable T1 FDL on a T1E1PRI port:
6164-1-T1E1PRI-T1 FDL
6164-2-T1E1PRI-T1E1PRI-T1 FDL
6164-*-T1 FDL
Where,
T1E1PRI is from 1 to 8.
T1 FDL
Meaning
Disable
Enable
T1 FDL Protocol
ETERNITY will support both the protocols of reporting the performance monitoring. This parameter is applicable
only if T1 FDL is enabled and Framing = ESF. This parameter will match the protocol expected by the other end of
the link.
Use following command to program the T1 FDL Protocol for a T1E1PRI port:
6165-1-T1E1PRI-T1 FDL Protocol
6165-2-T1E1PRI-T1E1PRI-T1 FDL Protocol
6165-*-T1 FDL Protocol
Where,
2367
T1E1PRI is from 1 to 8.
T1 FDL Protocol
Meaning
Disable
AT&T 54016
ANSI T1.403
ANSI T1.403
As per this standard, the receiving equipment transmits a performance report message (PRM) each
second over the FDL. This PRM is not sent to any specific remote location, but is broadcast so that any
PRM receiving device on the T1 line can intercept the message.
The PRM contains error information pertaining to only the previous 4 seconds.
It is the responsibility of the PRM receiver to accumulate the information and store it for 24 hours or the
time desired. This method allows performance monitoring points at different locations along the T1
network so that error localization is determined.
AT&T 54016
As per this standard, the receiving equipment collects the data but does not transmit it on its own based on
time as done by ANSI. Instead, the transmitting end sends a request to the receiving end to transmit the
performance data.
Explanation of Alarms
Alarms are indicated on the LEDs of the ETERNITY ME Card T1E1PRI. T1E1PRI card has four LEDs viz.
L1 to L4. L1 and L2 indicate alarms for T1E1PRI-1 whereas L3 and L4 indicate alarms for T1E1PRI-2.
During normal conditions, the LED blinks green (1 sec. ON, 1 sec. OFF).
RED Alarm
This alarm is generated if Loss of Signal persists for 2.5 seconds. This is indicated by flashing the LED
Red (500ms ON, 500ms OFF). The master logs this event in the System Fault Log. It is logged as System
Fault event 11 as RED Alarm <Slot No.> <Port No.> at HH:MM:SS.
It is cleared:
When the signal is acquired back and persists for 10 seconds. The LED is turned OFF. The system logs
this event in the System Fault Log. For example, it is logged as a System Fault event viz. Fault index 12 as
RED Alarm Cleared <Slot No.> <Port No.> at HH:MM:SS.
It is declared if:
2368
The interface SNIIC has settings for 32 consecutive zeroes or 192 consecutive zeroes. Hard program it to
32 consecutive zeroes.
When RED Alarm is declared, Yellow Alarm is sent to the far end within 12ms of detection of LOS.
YELLOW Alarm
This Alarm is also known as Remote Alarm Indication. This Alarm is generated when Yellow Alarm is sent
by the far end (Yellow Alarm is sent by the far end to indicate that it has lost the incoming signal).
It is declared:
When the signal corresponding to Yellow Alarm persists for 0.5 seconds. This is indicated by flashing the
LED Orange (500ms ON, 500ms OFF). The system logs this event in the System Fault Log. For example
it is logged as a System Fault event 13 as YELLOW Alarm <Slot No.> <Port No.> at HH:MM:SS.
It is cleared:
When No Yellow Alarm signal persists for 0.5 seconds. The LED is turned OFF. The master logs this event
in the System Fault Log. For example it is logged as a System Fault event 14 as YELLOW Alarm Cleared
<Slot No.> <Port No.> at HH:MM: SS.
Remarks:
Yellow Alarm in D4 is declared if:
More than 285 zeroes are received in bit position 2 of incoming DS0 channels during an integration period
of 1.5ms.
More than 3 ones are detected in bit position 2 of incoming DS0 channels during an integration period of
1.5ms.
Yellow Alarm signal pattern 0000000011111111 does not occur in 10 contiguous 16-bit signal pattern
intervals.
2369
When clearance of AIS is detected for continuous 10 seconds. The LED is turned OFF. The master logs
this event in the System Fault Log. For example it is logged as a System Fault event viz. Fault index 10 as
BLUE Alarm Cleared <Slot No.> <Port No.> at HH:MM:SS.
If less than six zeroes are received on the incoming line data during a 3 ms interval. AIS is cleared if the
above condition does not exist for 3 ms. This interval of 3 ms could be upto maximum of 75ms.
When BLUE Alarm is declared, Yellow Alarm is sent to the far end.
LED Indication
LED L1 and L2 are assigned to port 1 and LED L3 and L4 are assigned to port 2.
LED L1 is used for Card Heart Bit as well as status of the PORT1.
Port Status
L1 Green Steady
L1 Red Steady
L1 Yellow
L2 Green
L2 Yellow
L2 Green Flashing
@500msec.
L2 Yellow Flashing
@500msec.
2370
Port 1 is disabled
Port Status
L3 Green Steady
L3 Red Steady
L3 Yellow Steady
L4 Green
L4 Yellow
Port 2 is disabled
LED L1 is used for both Port 1 status and System Heart Bit. So in case LED is flashing, it will flash for 1
second and will OFF for 1 second.
2371
Time Tables
What's this?
Certain features of the ETERNITY like Operator, Class of Service, Toll Control, Outgoing Trunk Access, among
others, require stations and trunks to behave differently according to the time of the day, which is referred to as
Time Zone.
For example, incoming calls are to be routed to the security personnel extension, instead of the Operator when the
office is closed, or certain features in the Class of Service are to be allowed only during working hours, or access to
outgoing long distance calls are to be denied during non-working hours, or the station must play a different greeting
message to the callers during break hours and holidays.
Time Tables can be assigned to stations and trunks to define their behavior according to the time of the day, that is,
Time Zone.
Time Zones
A day can be divided into three time zones: Working hours, Break hours and Non-working hours. The default Time
Zones defined for each day are:
Working, Break and Non-Working hours are set to 00:00 for Sunday.
You can define a different Time Zone for your organization. Further, you can program each day of a week with
different time zones. For example, you may define the Working hours from Monday to Friday as 09:30 to 18:30, and
for Saturday, from 09:30 to 15:00. If you have a 24x7 business, you may set Working Hours also for Sunday.
Time Tables
A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours, Non-Working hours, for
the entire week.
A Time Table is assigned to stations defining the Time Zones for the entire week, so that the system can execute
the Time Zone-dependent features and facilities according to the Time Table.
2372
By default, the Time Table 1 is assigned to all stations and trunks in their Station Basic Feature Template and
Trunk Feature Template respectively. In Time Table 1, six days of the week - Monday to Saturday -have working
hours from 9:00-18:00, break hours from 13:00-14:00 hours and non-working hours from 18:00 to 09:00. Sunday is
a holiday, with all three Time Zones set to 00:00 hours.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the stations
and trunks.
ETERNITY offers the facility to switch the system manually into "Day/Night mode", at any point in time, by
issuing a command. When you set the system in Day/Night Mode, the system overrides the Time Tables
assigned to Trunks, Stations and Operator. According to the mode you selected, it applies Working Hours/
Non-Working Hours to run all the Time-Zone dependent features of the system.
Refer the topic Day Night Mode to know more.
How to configure
A Trunk port can be assigned Time Table in the Trunk Feature Template assigned to it. A Station port can be
assigned a time table in the Station Basic Feature Template assigned to it.
The default Time Table 1 is assigned to both stations and trunks of ETERNITY. Check if this time table matches the
working hours of the organization, and the Time Zone requirements of the individual stations and trunks.
The following station parameters can be programmed differently for different Time Zones:
Class of Service.
Toll Control.
The following trunk parameters can be programmed differently for different Time Zones:
Auto Attendant
DISA
2373
The following features can be programmed differently for different Time Zones:
Security Alarms
You may retain the default Time Table 1 or customize it to suit your requirements. Or you may customize different
Time Tables and assign them to different stations and trunks.
You can customize a Time Table using Jeeves or by dialing commands from a Telephone.
2374
Select the desired Time Table number and define the Time Zones, that is, working hours, break hours and
non-working hours.
Now, assign the Time Table you program to the desired Station/Trunk.
To assign Time Table to Stations, go to Station Basic Feature Template. Refer the topic Customizing
Station Basic Feature Template using a Telephone for instructions.
To assign Time Table to Trunks, go to Trunk Feature Template. Refer the topic Customizing Trunk
Feature Template using a Telephone for details a how to assign timetable to trunks.
Exit SE mode.
2375
M on 20 D EC 16:58
3003 R eception 2
During Non-working hours, in the idle state, the phone display will look like this:
M on 20 D EC 16:58
3003 R eception 2
During Non-working hours, in the idle state, if the extension user has set User Absent and activated Keypad Lock
on the phone, the phone display will look like this:
2376
N AL
Toll Control
What's this?
Toll Control (or Toll Restriction) is an expense control feature of ETERNITY. It enables you to program the system
so that each extension has a designated calling permission referred to as 'Call Privilege'.
Each type Call Privilege allows the extension to call certain areas and restricts it from calling others. The extension
can also be restricted from the dialing of specific telephone numbers.
The ETERNITY supports five types of Call Privileges, these are:
No Calls: Dialing of all external numbers is restricted. Only internal (extension-to-extension) calls are
allowed.
Only the numbers programmed in Global Directory Part I will be allowed to be dialed out, if the directory is
allowed in the Class of Service of the extension.
Local Calls: Dialing of outgoing calls to Local area numbers, in addition to internal calls, is allowed. It is
possible to restrict calls to certain local numbers. To apply this Call Privilege, you must configure the 'Local
Numbers' list.
Regional Calls: Dialing of outgoing calls to regional numbers is allowed, in addition to internal and local
calls. It is possible to restrict calls to certain regions. To apply this Call Privilege type, you must configure
the 'Regional Numbers' list.
National Calls: Dialing of domestic, long-distance numbers within the country is allowed, in addition to
internal and regional calls. You can also restrict calls to certain parts of the country. To apply this Call
Privilege type, you must configure the 'National Numbers' list.
International Calls: Dialing of international numbers is allowed, in addition to local area, long distance
and internal numbers. You can also restrict calls to certain countries. To apply this Call Privilege type, you
must configure the International Numbers list.
All Calls: Dialing of all types of numbers - local, regional, national, international- is allowed, without any
restriction.
Limited Calls: Dialing of only specific Telephone numbers (local, regional, national or international) is
allowed. By applying this Call Privilege type, you can allow and restrict dialing of telephone numbers
starting with a particular digit, or a particular area code, or certain telephone numbers only. To apply this
Call Privilege type, you must program a list of the Limited Numbers that are to be allowed and numbers
that are to be restricted. You can configure three such Limited Numbers lists.
Toll Control forms the basis of the features Dynamic Lock and Call Budget.
Using Dynamic Lock, extension users can change the Toll Control (Call Privilege) of their extensions on their own.
The Operator or System Administrator can also change the Toll Control of the extension using Dynamic Lock. To
support this feature, ETERNITY offers fours levels of Toll Control, from 0 to 3.
2377
When the Call Budget feature is used on extensions, it becomes necessary to define the calling permission for
extensions that have consumed their allotted budget. To support this feature, ETERNITY offers Toll Control-Call
Budget Consumed.
Toll Control - Level 0 is Time Zone based, wherein you must define the Call Privilege Type for each Time
Zone, that is, Working Hours, Break Hours and Non-Working Hours. For instance, you may define 'All
Calls' as Call Privilege for Working Hours, 'Local Calls' as Call Privilege for Break Hours and 'No Calls' as
Call Privilege for 'Non-Working' Hours.
By default, Call Privilege 'All Calls' is selected for all three Time Zones.
Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
Toll Control - Call Budget Consumed is applied only if the Call Budget feature is enabled on the
extension.
ETERNITY offers you the flexibility to redefine the Call Privilege for each of the above Toll Control Levels according
to user requirements.
How it works
2378
When a call is made, the ETERNITY checks the Toll Control Level assigned to the extension making the
call.
The system checks the 'Call Privilege' programmed in the Toll Control Level of the extension.
For each call privilege type detected, the system will check the following to determine if call is to be
allowed or denied, as summarized in the table below:
Type Call Privilege detected
Local calls
Regional calls
National calls
Internationals calls
Limited Calls
The Local, Regional, National, International and Limited Calls Number Lists consist of Allowed Numbers
and Denied Numbers.
The system compares the each digit of the dialed number string with the number strings programmed in
the Allowed and Denied Number Lists of the Local/Regional/National/International/Limited Number Lists,
using the following logic:
Result
match found
match found
Call allowed
match found
no match found
Call allowed
no match found
no match found
Call allowed
no match found
match found
Call denied
matches with Allowed Number list and the Denied Number list.
matches with Allowed Number list, but not with the Denied Number list.
matches with neither the Allowed List nor the Denied List.
The call is restricted, if the dialed number matches with the Denied Number list, but not with the Allowed
Number list.
How to configure
Decide the type of Call Privilege you wish to assign to each extension port type: SLT, DKP, SIP, ISDN Terminals.
For Toll Control to work, you must first program the lists of Local Numbers, Regional Numbers, National Numbers,
International and Limited Numbers, according to the type of Call Privilege you want to assign to the extensions. To
do this,
Make a two-column tables each for Local, Regional, National, International and Limited Call Numbers on paper or
using a computer.
In one column of each list, write down the numbers you want to permit as Allowed Numbers. In the other column
write down the numbers you want to restrict as Denied Numbers.Your Table may look like these:
List of Local Numbers for Call Privilege - Local Calls
Sr. No.
Allowed List
Denied List
1
2
:
:
999
2379
Allowed List
Denied List
1
2
:
:
999
List of Regional Numbers for Call Privilege - International Calls
Sr. No.
Allowed List
Denied List
1
2
:
:
999
List of Limited Numbers for Call Privilege - Limited Calls 1
Sr. No.
Allowed List
Denied List
1
2
:
:
999
2380
Local Numbers
Enter the local area numbers that are permitted to be dialed in the Allowed Numbers list and the numbers
that are to be restricted in the Denied Numbers list. You may enter as many as 999 numbers in each list.
2381
Regional Numbers
2382
Enter the regional area numbers that are permitted to be dialed in the Allowed Numbers list and the
numbers that are to be restricted in the Denied Numbers list.
Repeat the entries you made in the Local Numbers list also in the Regional Numbers list.
National Numbers
Enter the long distance numbers within the country that are to be permitted in the Allowed Numbers list
and the numbers that are to be restricted in the Denied Numbers list.
Repeat the entries you made in the Local Numbers and Regional Numbers lists in this list.
2383
International Numbers
2384
Enter the overseas numbers that are to be permitted in the in the Allowed Numbers list and the numbers
that are to be restricted in the Denied Numbers list.
Repeat the entries you made in the Local Numbers, Regional Numbers and National Numbers lists in
this list.
Limited Numbers
Enter the specific numbers or digits that are to be allowed to be dialed in the Allowed Numbers list.
Enter the specific numbers or digits that are to be restricted from being dialed in the Denied Numbers this
list.
By default all stations of ETERNITY are assigned Template 01. You may customize this template or select
another template.
2385
Select the desired Call Privilege Type for each Time Zone - Working Hours, Break Hours, Non-Working
hours.
Similarly, select the Call Privilege type for other Toll Control Levels 1, 2 and 3.
For the type of call privilege you select, the respective number list - Local, Regional, National, International
or Limited- you configured will be automatically assigned.
Local Numbers
To program Local Numbers Allowed List, dial:
4303-Index-Number String-#*
Where,
Index is the location at which the number should be stored in the list, from 001 to 999.
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
2386
Refer the following table for codes for dialing special digits: 0-9, #, *, A, B, C, D, Flash (F), Pause (P), +,
Dot (.).
Special Digit
Code
Flash (F)
#2
Pause (P)
#3
#4
#5
#6
#7
#8
Dot (.)
#9
##
**
Regional Numbers
To program Regional Numbers Allowed List, dial:
4305-Index-Number String-#*
Where,
Index is the location at which the number should be stored in the list, from 001 to 999.
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
For keying in special digits, refer the above table.
To program Regional Numbers Denied List, dial:
4306-Index-Number String-#*
To default the Regional Numbers List, dial:
4312-Reverse SE Password
National Numbers
To program National Numbers Allowed List, dial:
4307-Index-Number String-#*
Where,
Index is the location at which the number should be stored in the list, from 001 to 999.
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
2387
Code
Flash (F)
#2
Pause (P)
#3
#4
#5
#6
#7
#8
Dot (.)
#9
##
**
1 for No Calls
2 for Local Calls
3 for Regional Calls
4 for National Calls
5 for International Calls
6 for All Calls
7 for Limited Calls 1
8 for Limited Calls 2
9 for Limited Calls 3
To program Toll Control - Level 0 (WH) Allowed List, dial:
5502-1-Template-08-Number List
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
08 is the feature number on the template for Toll Control Level 0 (WH) Allowed List for Limited Calls.
Number List is the number of the list prepared as Allowed List for Limited Calls, from 01 to 16.
To program Toll Control - Level 0 (WH) Denied List, dial:
5502-1-Template-09-Number List
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
09 is the feature number on the template for Toll Control Level 0 (WH) Denied List for Limited Calls.
Number List is the number of the list prepared as Denied List for Limited Calls, from 01 to 16.
To program Toll Control - Level 0 (BH), dial:
5502-1-Template-10-Call Privilege Type
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
10 is the feature number for Toll Control Level 0 (BH) on the template.
Call Privilege Type is
1 for No Calls
2 for Local Calls
3 for Regional Calls
4 for National Calls
5 for International Calls
6 for All Calls
7 for Limited Calls 1
8 for Limited Calls 2
9 for Limited Calls 3
To program Toll Control - Level 0 (BH) Allowed List, dial:
5502-1-Template-11-Number List
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
11 is the feature number on the template for Toll Control Level 0 (BH) Allowed List for Limited Calls.
Number List is the number of the list prepared as Allowed List for Limited Calls, from 01 to 16.
To program Toll Control - Level 0 (BH) Denied List, dial:
5502-1-Template-12-Number List
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
09 is the feature number on the template for Toll Control Level 0 (BH) Denied List for Limited Calls.
Number List is the number of the list prepared as Denied List for Limited Calls, from 01 to 16.
2389
2390
1 for No Calls
2 for Local Calls
3 for Regional Calls
4 for National Calls
5 for International Calls
6 for All Calls
7 for Limited Calls 1
8 for Limited Calls 2
9 for Limited Calls 3
To program Toll Control - Level 3, dial:
5502-1-Template-18-Call Privilege Type
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
18 is the feature number for Toll Control Level 3 on the template.
Call Privilege Type is
1 for No Calls
2 for Local Calls
3 for Regional Calls
4 for National Calls
5 for International Calls
6 for All Calls
7 for Limited Calls 1
8 for Limited Calls 2
9 for Limited Calls 3
To program Toll Control - Call Budget Consumed, dial:
5502-1-Template-19-Call Privilege Type
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
19 is the number for this feature on the template.
Call Privilege Type is
1 for No Calls
2 for Local Calls
3 for Regional Calls
4 for National Calls
5 for International Calls
6 for All Calls
7 for Limited Calls 1
8 for Limited Calls 2
9 for Limited Calls 3
Exit SE mode.
Also refer the topics Configuring Extensions, Station Basic Feature Template, Number Lists.
2391
2392
For all Calls: the system answers all incoming calls landing on the trunk line.
When Busy: the system answers incoming calls on the trunk, only if the landing destinations are busy.
Delayed: the system first routes the calls to the Trunk Landing Group. If not answered by any extension,
the call is answered by the system.
How it works
ETERNITY handles incoming calls on the trunk according to the type of Trunk Auto Answer selected for the trunk:
For all Calls or When Busy or Delayed.
When Trunk Auto AnswerFor all Calls is enabled on a Trunk, for each incoming call on the trunk,
The System answers the call with a Greeting message, known as the Trunk Auto Answer Greeting, and
rings the landing destination selected for the time of the day.
The system starts the Built-In Auto Attendant Inactivity Timer (default: 60 seconds). The Trunk Auto
Greeting message is played once. You may assign a Trunk Auto Answer Greeting of your preference.
If the landing destination does not answer before the Trunk Auto Answer Greeting message ends, the
system plays Trunk Auto Answer Ring Back Tone message to the caller.
The Ring Back Tone message is played repeatedly for the duration of the Built-In Auto Attendant Inactivity
Timer.
However, if no Trunk Auto Answer Ring Back Tone message is assigned, the system will plays Ring Back
Tone to the caller for the duration of this timer.
If any of the landing destinations answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system stops the Built-In Auto Attendant Inactivity Timer and the Ring Back Tone message, and
connects the caller to the extension that answered the call.
If none of the landing extensions answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system plays the Trunk Auto Answer Busy Bye message and releases the trunk port.
2393
If no Trunk Auto Answer Busy Bye message is assigned, the system plays the Busy Tone for the duration
of the Busy Tone Timer and releases the trunk port.
When Trunk Auto AnswerWhen Busy is enabled on a Trunk, for each incoming call on the trunk,
The System answers the with the Trunk Auto Answer Greeting message and loads the Built-In Auto
Attendant Inactivity Timer.
The Greeting message is played once.
The System waits for any of the landing destinations selected for the time of the day to be free.
If no landing destination is free at the end of the Trunk Auto Answer Greeting message, the system plays
Ring Back Tone or Trunk Auto Answer Ring Back Tone message, if assigned, to the caller for the duration
of the Built-In Auto Attendant Inactivity Timer.
If any of the landing destinations is free before the expiry of the Built-In Auto Attendant Inactivity Timer, the
system places the call on that destination.
If none of the landing destinations is free at the end of the Built-In Auto Attendant Inactivity Timer, the
system plays the Trunk Auto Answer Busy Bye message, if assigned, and releases the trunk port.
If the Busy Bye message is not assigned, the system will play the Busy Tone to the caller for the duration
of the Busy Tone Timer.
When Trunk Auto AnswerDelayed is enabled on a Trunk, for each incoming call on the trunk,
The system first routes the incoming calls to the Trunk Landing Group. It waits for the duration of the
Delayed Trunk Auto Answer Timer (programmable; default:10 seconds) for any of the extensions in the
Trunk Landing Group to answer the call.
If the call is not answered by any of the extensions, the System answers the call with a Greeting message,
known as the Trunk Auto Answer Greeting, and rings the landing destination selected for the time of the
day.
The system starts the Built-In Auto Attendant Inactivity Timer (default: 60 seconds). The Trunk Auto
Greeting message is played once. You may assign a Trunk Auto Answer Greeting of your preference.
If the landing destination does not answer before the Trunk Auto Answer Greeting message ends, the
system plays Trunk Auto Answer Ring Back Tone message to the caller.
The Ring Back Tone message is played repeatedly for the duration of the Built-In Auto Attendant Inactivity
Timer.
However, if no Trunk Auto Answer Ring Back Tone message is assigned, the system will plays Ring Back
Tone to the caller for the duration of this timer.
2394
If any of the landing destinations answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system stops the Built-In Auto Attendant Inactivity Timer and the Ring Back Tone message, and
connects the caller to the extension that answered the call.
If none of the landing extensions answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system plays the Trunk Auto Answer Busy Bye message and releases the trunk port.
If no Trunk Auto Answer Busy Bye message is assigned, the system plays the Busy Tone for the duration
of the Busy Tone Timer and releases the trunk port.
How to configure
For this feature to work, you must do the following:
Enable Trunk Auto Answer on the Trunk Feature Template of the desired trunks. To know more and for
instructions, see Trunk Feature Template.
Select the Trunk Auto Answer Greeting message, the Trunk Auto Answer Ring Back Tone Message,
and the Trunk Auto Answer Busy Bye Message for the Working Hours, Break Hours and Non-Working
Hours.
You may select different Greeting, Ring Back Tone and Busy Bye Message for each time zone. You can
also select Music-On-Hold instead of Ring Back Tone Message for the time zones.
Configure the Delayed Trunk Auto Answer timer, if required.
Configure the Trunk Auto Answer related Timers, if required. The following Timers are of relevance to the
Trunk Auto Answer Feature:
The Built-In Auto Attendant Inactivity Timer (default: 60 seconds)
The Ring Back Tone Timer (default: 45 seconds)
The Busy Tone Timer (default: 7 seconds)
You may change the duration of these timers from theSystem Timers and Counts page.
The Ring Back Timer and the Busy Tone Timer are also applicable for the Ring Back Tone and the Busy
Tone played for internal calls.
Record and assign Voice Modules for the following Voice Messages related to this feature:
Trunk Auto Answer Greeting Message
Trunk Auto Answer Ring Back Tone Message
Trunk Auto Answer Busy Bye Message
For each of these messages, you can record four different messages.
See the topic Voice Message Applications for instructions on recording and assigning voice modules to
greeting messages.
2395
How it works
This feature is not supported on SIP Extensions.
How to configure
For Trunk Call Waiting to work on an extension, it must be enabled in the Class of Service allowed to that
extension. This can be done using Jeeves as well as a Telephone.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
ETERNITY. Template 01 is assigned CoS group 01. Trunk Call Waiting is disabled in the CoS. So, none of the
extensions of ETERNITY are provided call waiting indication for incoming trunk calls.
If you want to allow Trunk Call Waiting uniformly to all extensions of ETERNITY, simply enable this feature in the
default CoS group (01) in the default Template (01).
However, if you want to allow this feature to only selected extensions,
1. Define a CoS group with Trunk Call Waiting enabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the Time Zones.
3. Assign this new Template to the extensions to which Trunk Call Waiting is to be allowed.
Refer the topics Class of Service (COS) and Station Basic Feature Template for programming instructions.
2396
How it works
You can configure as many as 95 TLGs. Each group is numbered from 01 to 95.
A maximum of 32 stationsSLT, DKP, SIP, ISDN, OGTB or Virtual Extensionscan included in each
Trunk Landing Group.
To use the Gateway Application of ETERNITY, select OGTB as the station.
For each group that you create, you can do the following:
set the Sequence in which the stations in the group should ring, by selecting the member stations in a
sequence from 1 to 32.
set the Time for which each station in the group should ring, by setting the Ring Timer (default: 15
seconds).
set each station to ring continuously till the call matures by enabling Continuous Ring (default:
disabled).
When Continuous Ring is enabled, once a station receives a ring, it rings continuously till the call
matures. The station continues to ring even as other stations in the group are hunted.
If the call is not answered even after the last station in the group has been hunted, the system will loop
back and start hunting from the first station, all over again.
have a number of stations in the group ring simultaneously by enabling Continuous Ring on these
stations and setting the Ring Timer for these stations to 00 seconds.
set equal distribution of incoming calls on all stations in the group, by enabling Rotation for the entire
group (default: disabled).
When Rotation is enabled on a TLG, for each new call on a trunk, the system will land the call on the
extension next to the one that received the last call.
When Rotation is disabled in a TLG, for each new call on a trunk, the system will land the call on the
first free station of the TLG.
2397
To each Trunk, you must assign a TLG for the Time Zones, working hours, break hours and non-working
hours. You may assign the same TLG for all three Time Zones, or a different TLG for each Time Zone.
How to configure
According to the number of trunks interfaced with your ETERNITY and the number of extensions you
have, identify the trunks to which you want to assign TLG for each Time Zone. This will help you decide the
number of TLGs to be formed, the type and number of extensions in each group, and their sequence.
Configure each TLG as a Routing Group. See Routing Group for instructions.
Assign the TLGs you formed for each trunk for the three Time Zones in its Trunk Feature Template. for
instructions, see Trunk Feature Template.
Example:
Two CO Lines (configured on software ports 001 and 002) are interfaced with ETERNITY.
Incoming calls on CO 1 during working hours should land on SLT stations 2001, 2003, 2005 (configured on
software ports 008, 010, 012 respectively).
Incoming calls on CO 1during break hours and non-working hours should land on SLT stations 2002, 2004
(configured on software ports 009, 011 respectively).
Incoming calls on CO 2 should land always on DKP stations 3001, 3002, 3003, 3009, 3010 (configured on
software ports 013, 014, 015, 016, 017).
Incoming call on CO 1 should ring for 10 seconds on each station in the TLG.
Incoming call on CO 2 should ring for 20 seconds on each station in the TLG.
The stations of the TLGs of CO 1 and CO2 should ring for the set time only.
In this example, for CO 1, you would need to form 2 TLGs; one TLG for working hours and one for break hours and
non-working hours. So, form two Routing Groups. For example, Routing Group 10 for working hours and Routing
Group 11 for break hours and non-working hours.
In Routing Group 10, select the member SLTs in this sequence: 2001, 2003 and 2005. Set the Ring Timer for each
member SLT to 10 seconds. Disable Continuous Ring for the SLT, as each station in the group must ring for the set
time.
In Routing Group 11, select the member SLT in this sequence: 2002 and 2004, and set the Ring Timer for each SLT
to 10 seconds. Disable Continuous Ring for the SLT, as each station in the group is required to ring for the set time
only.
Enable Rotation for Routing Group 10 and Routing Group 11.
2398
For CO 2 you would need to form a common TLG for working hours, break hours and non-working hours. So,
configure as single routing group, for example, Routing Group 13 for CO 2. In Routing Group 13, select the
member stations in this sequence: 3001, 3002, 3003, 3009, 3010. Set the Ring Timer for each DKP member station
to 20 seconds. Disable Continuous ring for each DKP in the group. Enable Rotation for Routing Group 13.
Select a Trunk Feature Template number for CO 1 and CO 2. For example, feature template 04 for CO 1 and
template 05 for CO 2.
In the Feature Template of CO 1, assign TLG. For working hours, assign Routing Group 10, for break and nonworking hours, assign Routing Group 11.
In the feature Template of CO 2, assign Routing group 13 as TLG for all three time zones.
2399
Trunk Reservation
Whats this?
This feature enables any extension user to reserve a trunk for exclusive use, for a specific time period.
Trunk Reservation can be requested from an SLT extension, a DKP extension and from an Extended IP Phone
extension.
How it works
Let us understand this feature with the help of an example:
In an organization there are four CO trunk lines, CO 1, 2, 3 and 4, but all of these have full traffic throughout the
day.
Extension user A is a sales executive. To complete the sales target, A needs to make long-distance calls to
customers. Since there this full traffic on all the four trunks throughout the day, and these trunks are constantly
busy, A would need a dedicated trunk line to save time and complete the target.
So, A can reserve one of the four trunk lines for the desired duration. To do this,
2400
Extension A dials the feature access code for Trunk Reservation for the busy trunk.
A answers the call and gets connected to the trunk, and gets dial tone.
The trunk remains reserved for the duration of the Trunk Reservation Timer. This timer is configurable, and
by default is set to 10 minutes. A can have this Timer configured to the desired duration.
All other extension users who try to access this trunk get error tone, even if this trunk is free.
If A is finished with the calls before the expiry of the Trunk Reservation Timer, A has two options:
a) release the trunk manually, by cancelling Trunk Reservation.
or
b) wait for the expiry of Trunk Reservation Timer.
Only when the trunk is released (by A or at the end of the Timer) will other users be able to access it.
How to configure
To be able to use Trunk Reservation, extension users must have this feature enabled in their Class of Service for
the time zone. For configuration instructions, see Class of Service (COS), and Station Basic Feature Template.
You may increase or decrease the duration of the Trunk Reservation Timer. See System Timers and Counts for
instructions.
How to use
For EON & Extended IP Phone Users
To set Trunk Reservation, when Trunk you access is busy,
To release a reserved Trunk, wait for the Timer to expire, or cancel Trunk Reservation, manually.
To cancel Trunk Reservation Manually,
Dial 102
Lift handset.
Dial 6 on Busy Tone.
Replace handset.
2401
User Absent/Present
What's this?
Extension users may sometimes want to leave their desks, and expecting to return soon, they may not have
forwarded their calls or set Do Not Disturb on their extensions. In such cases, incoming calls will continue to land on
the extension and go unanswered. The callers have no way of knowing that the extension user is not present at the
extension and may try the extension number repeatedly.
With the User Absent/Present feature of ETERNITY, extension users, including the Operator, can set 'User Absent'
when they leave their desks. By doing so, they can block all incoming external as well as internal calls from landing
on their extension. When they return to their desks, they can set 'User Present' and receive incoming calls again.
While Do Not Disturb blocks only internal calls, User Absent can be set when you want to block all
incoming calls (external as well as internal). Thus User Absent can be used as an extension of DND to also
block external calls.
There are more options for indicating availability to other extensions. Refer the topic Presence to know
more.
How it works
When an extension user of EON sets 'User Absent', the letter 'A' appears on the phone's display:
EON48P
Fri29JAN16:58A
617Himanshu
The letter 'A' disappears when the extension user sets 'User Present'.
When an extension user of EON calls the extension which has set 'User Absent', the text message 'User Absent'
will appear on the caller's phone display.
When an SLT extension user calls the extension which has set 'User Absent', callers who dial this extension will get
an error tone.
External callers who call the extension, on which 'User Absent' is set, will get an error tone only.
2402
Outgoing calls can be made from the extension which has set 'User Absent'. Only incoming calls are
restricted.
If more than one extension is configured as "Operator' (routing group), incoming calls will be blocked
only on the Operator extension which has set User Absent.
User Password is required for this feature. The default User Password, 1111, will not work. Change the
User Password first.
How to use
For Extension Users
To set User Absent on your extension:
Dial 104-User Password-0
To set User Present on your extension:
Dial 104-User Password-1
2403
User Password
Whats this?
The User Password is a 4-digit code for extension users to protect their extension phones from unauthorized use.
The default User Password is 1111. It can be changed by the extension users from their phones to any desired
value, not exceeding 4 digits.
In case the extension user forgets the password, it can be cleared and restored to the default value 1111 by the
System Engineer (SE) or the System Administrator (SA). Refer the topic System Security - V10R11 and later for
instructions.
The User Password is also required to access and use certain features of ETERNITY, which are listed below.
Call Follow Me
Dynamic Lock
Direct Inward System Access (DISA)
Walk-In Class of Service
User Absent/Present
Hot Desk
Phone Settings of the DKP
Mailbox of Voice mail
The extension user must change the default password for all the above listed features except: Phone Settings,
Mailbox of Voice mail. Both these features allow the extension user to use the default User Password, whereas in
the case of others, the system will not allow feature access without changing the User Password.
In the case of Hot Desking, the default password will work only for one extension involved.
The User Password for an extension can be changed only from that extension phone only.
Since the Mailbox can be accessed using the default User Password, extension users who are assigned a
mailbox are recommended to change their User Password to prevent unauthorized access to their mailbox.
How to use
For EON and Extended IP Phone Users
2404
2405
Video Call
What's this?
Video calling has become an increasingly important tool in todays business world. It offers people the power of
face-to-face communication, at reasonable cost without incurring the expense of traveling. ETERNITY allows you
to make and receive video calls by connecting video capable user terminals.
ETERNITY supports video calling over:
T1E1 PRI Trunks and ISDN Terminals
SIP Trunks and SIP Extensions
How it Works
Video Calling over T1E1 PRI Trunks and ISDN Terminals
For video calling over T1E1 PRI Trunks, you require the following:
Connect the T1E1 PRI Trunks to ETERNITY.
Make sure you have a BRI card installed in the system and set the Orientation of the BRI Port as Network.
Connect ISDN Terminals with video capability to the BRI ports.
The minimum bandwidth required for a video stream to flow is 128 kbps. Hence, 2 BRI channels will be consumed.
When this video stream flows through the PRI line it will consume 2 PRI channels too. However, for higher
resolution you can connect multiple BRI ports, if supported by the ISDN Terminal.
To conduct a video call,
From the ISDN Terminal dial the number of the desired party (other ISDN Terminal). This is a one to one
call.
You can also make a video call from one ISDN Terminal of ETERNITY to another ISDN Terminal of
ETERNITY. In this case 4 BRI channels will be consumed.
How to configure
To conduct a video call you must configure the following:
2406
None of the proprietary Extended SIP Phones of Matrix support video calling.
Feature Interactions
Call Hold: You can put a video call on hold using Flash and then dial the second number to initiate another
video call.
Call Toggle: You can toggle between two video calls using Flash-1.
Conference: When you already have two video calls and want to merge the calls into a conference (using
Conference access code), the conference call will be converted into an audio call.
Blind Transfer: When you blind transfer a video call using the access code, the other two parties are
connected but the call is converted into an audio call.
Attended Transfer: When you make an attended transfer of a video call using the access code, the other
two parties are connected but the call is converted into an audio call.
Auto Call Back (ACB) - No Reply: You cannot access ACB - No Reply when you try to initiate a video call
and the called party is ringing.
Forced Answer: You cannot access Forced Answer when you try to initiate a video call and the called
party is ringing.
318. Current industry standard video codecs include H.261, H.263, H.263p, H.264, MPEG4 etc. For more details, refer the documentation of the corresponding video terminals you are using for video calling.
319. Eternity PE supports maximum 16 VoIP channels hence, 8 video calls are possible.
2407
2408
You can use #2, if your IP Phone does not support Flash dialing.
Virtual Extension
Whats this?
The Virtual Extension feature of ETERNITY enables multiple users to share one telephone instrument as their
extension, yet be considered as individual extensions by the system, with distinct extension properties and class of
service.
Such shared extensions are called Virtual Extensions, as their users do not have individual phones for their use.
ETERNITY supports
Virtual Extensions are useful in laboratories, common rooms, dormitories, shop floors, and wherever it is not
feasible to provide dedicated telephone instruments to individual extension users. Virtual extensions allow you
make optimum use of the existing phones without investing in new ones.
How it works
The shared telephone instrument is called the Master Extension. A Master Extension can be an SLT, a DKP or the
Matrix Extended IP Phone.
Virtual Extensions are assigned to the Master Extension. A Master Extension can have multiple Virtual
Extensions, but a Virtual Extension can have only one Master Extension.
The Virtual Extension functions as any other extension of ETERNITY. It can be assigned all features and
facilities, like Class of Service, Toll Control, Call Forward, just like any other physical extension of
ETERNITY. You can assign Station Basic Feature Template and Advanced Feature Template to the
Virtual Extension.
All incoming, outgoing, internal and external calls of the Virtual Extensions are recorded in the Station
Message Detail Records.
To make outgoing calls, the Virtual Extension user must use the feature Walk-In Class of Service.
The Virtual Extension user is logged out of the Master Extension according to the Walk Out mode
assigned to it: Walk out single call or Walk out multiple calls.
How to configure
For this feature to work, you must do the following:
Make a list of the number of Virtual Extensions required by you along with their names and numbers.
Decide the Master Extension (landing destination) that is, the Port Type and Port Number. The Port Type
may be SLT, DKP, ISDN Terminal, or SIP Extension. Port Number is the number of the software port to
which the landing destination extension is connected.
2409
If required, you can customise the Station Basic Feature Template and Station Advanced Feature
Template you want to assign to the Virtual Extensions. For making outgoing calls users of Virtual
Extensions must have Walk-In Class of Service enabled in their COS.
Decide the Priority you want to assign to the Virtual Extensions. When an outgoing call is made from any
Virtual Extension, the ring type played to the called party will be as per the set priority.
Access Code: Assign Station Access Codes to the Virtual Extensions. Station Access Codes are
commonly referred to as Extension Numbers. These may be a combination of 1, 2, 3 , 4, 5 and 6 digits,
which are dialed to call the Virtual Extension to which they are assigned.
To assign Station Access Codes according to your preference and requirment to a range of Virtual
Extensions, see Assigning Access Codes to a Range of Extensions.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics Access Codes and Conflict Dialing to know
more.
2410
Name: Assign a 'Name' to the Virtual Extension. The name may be of the person who will use the
extension. This name will be displayed on the LCD of the remote user's phone, if it is equipped with
Caller ID.
You can program a name of a maximum of 18 alphanumeric characters.
Login Destination: Configure the Port Type and Port Number you want the Virtual Extension user to
log into to make outgoing calls.
Landing Destination: Configure the Port Type and Port Number on which you want the Virtual
Extension user to receive incoming calls.
Click the Advanced button at the bottom of the page and configure the following parameters:
Mobile Number: Enter the Mobile Number of the extension user you wish to store. The Number can be
a maximum of 16 digits.
Email ID: Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum
of 64 characters.
Group: You can assign the extension user to a Group. The system clubs together extension users
assigned the same Group. The Group can be a maximum of 16 characters. Default: Blank.
If you have completed configuring the parameters, click Submit at the bottom of the page to save your
settings.
It is possible to default all the parameters by clicking the Default button. You can also restore default
values of the parameters of a single Virtual Extension by clicking the Default One button and specifying
the Virtual Extension Number you want to set to default.
2411
Access Code is a number string of any combination of 1, 2, 3 or 4 digits. Terminate the command with
#* if the number string has fewer than 4 digits.
To clear the access codes for all the Virtual Extension, dial:
3109-*
2412
3001-1-Virtual Extension-Port Type-Port Number to program landing destination for a single Virtual
Extension.
3001-2-Virtual Extension- Virtual Extension -Port Access Code - Port Number to program landing
destination for a range of Virtual Extensions.
3001-*-Virtual Extension-Port Access Code-Port Number to program landing destination for all the
Virtual Extensions.
Where,
Virtual Extension is from 01 to 64.
Port Type and Port Number is:
Port Type
Port Number
SLT
01
001-512
DKP
02
001-128
ISDN Terminal
28
01-64
SIP Extension
34
001-500
2413
Voice Help
Whats this?
The Voice Help feature of ETERNITY allows you to record and play short (16 second duration) voice messages to
provide quick help to extension users. Voice Help messages may contain important instructions, or frequently
accessed feature codes, or important phone numbers, etc.
For example, the Access Codes of the frequently used features can be recorded in the Voice Help message.
Extension users may simply dial the Voice Help feature code and listen to the voice message.
How to configure
To be able to use Voice Help, you must first record a voice module with the contents you wish to provide as help.
Record the voice module considering the maximum duration of the voice module, so that the message is not
truncated.
The voice module must be assigned to the Voice Help application. Refer topic Voice Message Applications for
instructions on recording the voice module and assigning it to Voice Help.
How to use
For EON and Extended IP Phone Users
2414
Once-Only: the message is played only once from its start to its end.
Continuous: the message is played repeatedly from the start to the end.
When the recorded voice modules are assigned to the features/applications, they are played to the callers/
extension users whenever the feature/application is activated.
As many as five different voice messages can be played simultaneously to a caller/extension user.
Voice messages can be used for different applications or situations as described in the following.
Built-In Auto Attendant - Welcome Greeting message for working hours: "Welcome to Cotton
Software".
Built-In Auto Attendant - Welcome Greeting message for lunch hours: "Welcome to Cotton Software.
This is lunchtime. Please call after 2.00 pm".
Built-In Auto Attendant - Welcome Greeting Message for non-working hours: "Welcome to Cotton
Software. We are closed for the day. Please call later".
2415
Built-In Auto Attendant - Dial Message: "Please dial the desired station number".
Built-In Auto Attendant - Dial Message: "Please dial the desired Station Number".
Built-In Auto Attendant - Wrong Dial Message: "Sorry you have dialed an invalid Station Number.
Please dial a valid Station Number".
Built-In Auto Attendant - Ring Back Tone (RBT) Message: "You have successfully dialed the desired
Station".
Built-In Auto Attendant - Busy Message: "The station you have dialed is busy. Please hold the line or try
any other Station Number".
Built-In Auto Attendant - No-Reply Message: "The Station you have dialed is not responding. Please try
another Station".
Built-In Auto Attendant - No-Dial Message: "Sorry you have not dialed any Station Number. Please wait
while your call is transferred to the Operator".
Built-In Auto Attendant - Conference Number320 Message: "Please dial the Conference Number".
Built-In Auto Attendant - Conference Password Message: "Please dial the Conference Password".
Built-In Auto Attendant - Call Transfer Message: "Transferring the call to the Operator". This message
is played to the caller, when he does not dial any number and the call is transferred to the Operator.
Alarms: A pre-recorded voice message is played to the extension on which Alarm is set when the wakeup call is served. This feature is very useful in hotels where wake-up alarms are to be set for guests at the
oddest hours. With the Voice Message for alarms, guests can be greeted when they pick up the handset to
answer their phone.
For example, "Good Morning. This is a wake up call. You may please call room service for any assistance.
Thank you. Have a nice day".
You are recommended to record the message "Please press 0 to acknowledge the Alarm and Reminder"
as a voice module for the Alarm/Reminder message, so that extension users can acknowledge Snooze
calls. Refer the topics Alarms and Reminder to know more.
2416
Security Alarm: A pre-recorded voice message is played to the external number and to the extension
which answers the Security Alarm call. The default message played to the called parties is: "This is an
emergency call. Please dial '0' to acknowledge.
Voice Help: You can record and play voice message to provide quick help to extension users. The Voice
Help message may contain important instructions, or frequently accessed feature codes, or important
phone numbers, etc.
For example, the Access Codes of the frequently used features can be recorded in the Voice Help
message. Extension users may simply dial the Voice Help feature code and listen to the voice message.
Help message must be recorded considering the maximum duration of the Voice Module (16 seconds), so
that voice messages are not truncated.
Music-on-Hold: Callers who are put on hold are usually played music from an internal/external source as
they wait. You can play a voice message instead of music to the callers. The message may contain any
promotional information about your company or services provided by your organization, etc.
For example, "Welcome to Progressive Bearings. We are glad to announce that we are now an ISO 9001
company."
Message Waiting: Whenever there is a new message in the mailbox of the extension user and if the VMS
informs the ETERNITY about the new message, the ETERNITY changes the dial tone of the extension to
a stuttered dial tone. The ETERNITY also offers the facility to playback a message instead of the stuttered
dial tone to indicate the waiting message. An appropriate voice message can be played back to the
extension user when he lifts the handset.
For example, "You have a new message in your Mailbox. Please access your mailbox".
2417
2418
How to configure
To be able to play Voice Messages, you must first record them in voice modules. Once you have recorded the
voice messages, you must assign the voice module to the appropriate Voice Message Application.
Pre-recorded voice messages are provided in WAV format on the documentation CD shipped with the ETERNITY.
You may either use them when recording the voice modules or you may record messages of your choice.
An External Music Source (PC, Music System) connected to the Analog Input Port (AIP) of the
ETERNITY. For instructions, refer the topic Connecting External Music Source under Installing ETERNITY
ME, Installing ETERNITY GE, Installing ETERNITY PE as relevant to your model of ETERNITY.
When recording from an External Music Source on ETERNITY GE, you must change the position of the
Jumper J5 on the CPU Card to AB position. On ETERNITY ME and ETERNITY PE, you can select the
recording source from Jeeves and by dialing System Command from an extension phone.
If you want to use the pre-recorded voice messages provided with the system, first copy the WAV files on
to the PC to which ETERNITY is connected. The contents of the audio files are indicated by their file
names. Select the audio files (containing the messages) you wish to use, and copy them on to the PC.
Play the files when you record the messages. Recording instructions are provided below.
2419
2503-Voice Module
Where,
Voice Module is from 01 to 16.
In this case, dial the number of the Voice Module you just recorded.
If the audibility of the recorded message is not satisfactory, you may repeat this procedure again.
2420
Recording Source: Select the source of recording - Telephone Instrument or Music System. By
default, Telephone instrument is selected.
Recording Format: Select the format of the voice message recorded, A-Law, Mu-Law or Linear, as
appropriate. By default Mu-Law is selected.
By default, the following Voice Messages Applications have been assigned to Voice Modules 01 to 13
(see table below). If the default Message recorded in the module suits your purpose, simply assign the
Voice Module number to the relevant Voice Application. The system will automatically set the duration
of the Voice Message Application.
If you want to change the duration of a recorded Voice Module, you can do so, by changing the
duration manually for the particular recorded message.
For example, if you want to use Morning, Afternoon and Evening Greetings. You may simply assign the
default Voice Modules 02, 03, and 04.
2421
Refer the following table for Voice Message Applications assigned by default to the Voice Modules:
Default Voice Message Applications assigned to Voice Modules in the Enterprise Mode
Voice Module
Number
01
Music-On-Hold
02
Voice Message
Good Morning!
Greeting
03
Good Afternoon!
Greeting
04
Good Evening!
Greeting
05
Welcome!
message
09
message
10
Busy message
11
Reply message
13
to Operator message
Default Voice Message Applications assigned to Voice Modules in the Hotel Mode
Voice Module
Number
01
Music-On-Hold
02
Voice Message
Good Morning!
Greeting
03
Good Afternoon!
Greeting
04
Good Evening!
Greeting
2422
Voice Module
Number
05
Voice Message
Welcome!
08
message
09
message
10
Busy message
11
Ringing message
(Ring Back Tone)
12
Reply message
13
to Operator message
14
Alarm
15
16
Blank
It is possible to assign the same Voice Module to more than one Voice Message Application.
If you have already recorded Voice Module 08, ETERNITY will automatically detect and display the
duration of the Voice Module you recorded. So you need not define the duration of the Voice Module.
You may define the duration of the Voice Module, only if you want the recorded voice message to be
played for a specific duration. For example, the message you recorded in the voice module is 15
seconds long, but you want to play only the message contents of the first 8 seconds, you can define the
duration of the message as 8 seconds.
Voice Module 01 is reserved for Music-on-Hold by default. You are advised not to assign this module to
any other Voice Message Application.
2423
Enter SE mode.
To assign a voice message application to a voice module:
Dial 2505-Voice Message Application Number-Voice Module
Where,
Voice Message Application Number is from 01 to 45. See table below.
Voice Module is the voice module number from 1 to 16.
By default, Voice Module assigned is 00.
Voice Message
Application Number
Meaning
Once/Continuous
(Not Programmable)
01
Once
02
Once
03
Once
04
Once
Once
Once
Once
08
Continuous
Message
09
Once
10
Once
Message
11
Once
Message
12
Once
13
Once
14
Once
Password
15
Once
Operator
2424
16
Once
17
Once
18
Once
19
Once
20
Once
21
Continuous
22
Continuous
Voice Message
Application Number
Meaning
Once/Continuous
(Not Programmable)
23
Continuous
24
Continuous
25
Once
26
Once
27
Once
28
Once
29
Once
Message
30
Once
Message
31
Once
Message
32
Alarm
Continuous
33
Once
34
Once
35
Continuous
36
Once
37
DND Notification
Once
38
Security/Emergency Message
Continuous
39
Dial Tone
Once
40
Continuous
41
Busy Tone
Once
42
Error Tone
Once
43
Confirmation Tone
Once
44
Voice Help
Continuous
45
Once
46
Once
To use the default Voice Modules, refer the table Default Voice Message Applications assigned to Voice
Modules in the Enterprise Mode, and assign the desired Voice Module to a Voice Message Application.
To define voice message duration for voice modules321:
Dial 2504-Voice Module-Duration
Where,
Voice Module is 01 to 16.
Duration is from 00 to 16 seconds.
By default, Duration of Voice Modules is 16 seconds.
321. Necessary only if the duration is less than 16 seconds.
2425
Exit SE mode.
2426
One call: The extension user is automatically logged out from the extension into which the user has
walked-in, after one call.
Multiple Calls: The extension user can make as many calls as desired, and remains 'walked-in' until the
user dials the feature code to 'Walk-Out', or until another extension user walks into the same extension.
To allow One call or Multiple Calls to an extension, you need to set the 'Walk-Out Mode in the Station Advanced
Feature Template of the extensions.
Walk-In Class of Service is a password-protected facility and the default User Password 1111 will not be accepted.
To be able to walk into another extension, extension users must first change their User Password to another value.
How it works
With the help of this illustration, let us understand how Walk-in Class of Service works.
In this illustration, Extension user A has a DKP with the number 3001, with long distance calling facility (toll control:
All calls). Extension user B has an SLT with the number 2001, without long distance calling (toll control: local calls).
Here,
3001 is the Source Extension, whose CoS and Toll Control are used from another extension (2001) by
performing Walk-In.
2001 is the Destination Extension on which Walk-In is performed.
Now, extension user A is at B's desk and needs to make a long distance call. Bs extension does not have
long distance calling.
2427
On successful Walk-In, ETERNITY applies the Class of Service and Toll Control of the Source Extension
3001 on the Destination Extension 2001.
Extension user A can make external and internal call from extension B.
If Extension 2001 has 'One Call' selected as the 'Walk-Out Mode' for the extension, A will be 'Walked-Out'
when the current call ends or if A goes ON-Hook at any time after walking into extension 2001.
If Extension 2001 has 'Multiple Calls' selected as the 'Walk-Out Mode' for the extension, extension user A
must manually walk out by dialing the feature code for 'Walk-Out'.
If Extension user A does not 'Walk-Out', the system will perform a walk out for A only when another
extension user walks into extension 2001.
At a time, only one extension user can walk in to another extension.
Calls made after walk-in will be charged to the Source Extension. Here, calls made by Extension user A
from extension 2001 using Walk-In will be calculated and charged to extension 3001 only.
Call record details of calls made after walk-in will be recorded for the Source Extension. Here, call record
details of calls made by Extension user A from extension 2001, using Walk-In, will be recorded in the
Station Message Detail Record of extension 3001 only.
Extension user A with number 3001 has Call Forward in the Class of Service.
Extension user B with number 2001 does not have Call Forward in the Class of Service.
Extension user A is currently at Bs desk. A needs to forward calls of own extension to an external number.
To do this,
As User Password322
On successful Walk-In, the system applies the Class of Service and Toll Control of the Source Extension
3001 on the Destination Extension 2001.
From extension 2001, extension user A sets Call Forward for 3001, to an external number.
Call Forward can be canceled from the Source Extension or from the Destination Extension. To cancel
Call Forward, extension user A can go back to 3001 and cancel Call Forward from 3001, or can cancel Call
Forward from 2001, if user A is still walked-in on 2001.
322. The default password 1111 will not be accepted, it must be changed first.
2428
CAUTION! The Destination Extension user can access their Class of Service or Toll Control only after the
Source Extension user has walked out from their extension. For example, user B cannot set or cancel Call
forward on extension 2001, until user A has walked out from 2001.
Incoming calls on the Destination Extension (2001) will work according to the setting of the Destination
Extension only, whereas outgoing calls on the Destination Extension will work according the settings of the
Source Extension (3001).
The following set of features of the Destination Extension will remain unaffected:
Key map
Language
Priority
Call Pick-Up group
Personal Directory
CAUTION! There is a risk of fraudulent calls being made from your extension, if a third party comes to
know the User Password of your extension. The cost of such fraudulent calls will have to be borne by the
owner of ETERNITY.
So, protect your system from unauthorized access and misuse by putting strong authentication
mechanisms in place.
Keep Passwords strictly confidential.
Change Passwords regularly.
Choose Passwords that are complex and difficult to guess.
How to configure
This feature is available to all extensions of ETERNITY. All you need to do is, select of the 'Walk-Out Mode' for the
extensions in their Station Advanced Feature Template.
By default, One call is selected as the Walk -Out mode in the default Station Advanced Feature Template 01
assigned to all extensions.
If you want to allow different walk-out modes to different extensions, use different Templates, but make sure other
features on these templates are also configured according to requirement.
For detailed instructions refer the topics Customizing Station Advanced Feature Template using Jeeves and
Customizing Station Advanced Feature Template using a Telephone.
How to use
For EON and Extended IP Phone Users
To perform a Walk-In, on the Destination Extension,
2429
You need to perform a Walk-Out only if the Source Extension as Multiple Calls set as Walk-Out mode.
If the Source Extension has 'One Call set as the Walk-Out mode, you will be walked out of the Destination
Extension when you go ON-Hook after making a call or accessing a feature.
If the extension you are walking in has 'One Call as the Walk-Out mode, and you go ON-Hook before you
make the call, you will be 'Walked Out'. You must Walk-In again.
2430
CHAPTER 14
Dial 3931.
Dial 3931.
Dial 0 to reach the Home Code Node option. While the system plays the Welcome Message, dial 8.
2431
If CLI Based Authentication is enabled, the system prompts you to dial desired extension number.
If PIN Authentication is enabled, the system prompts you to dial your Extension Number and Password.
The VMS answers the call. While the system plays the Welcome Message, dial 8.
2432
How it works
Voice-guided Alarm and Reminder requests are served as per the date and time set by extension users. The
different ways in which Alarm or Reminder requests will be served are described in the following:
When user press '0', VMS prompts: "Your Alarm is Acknowledged." (Acknowledge.wav)
When user press '0', VMS prompts: "Your Alarm is Acknowledged." (Acknowledge.wav)
How to configure
The VMS allows you to enable/disable the Alarm Verification for alarms and reminders, allowing extension users
who want to use alarms and reminders to confirm
Time set for an alarm
Date and time set as a reminder.
When Alarm Verification is disabled, the VMS will not confirm the alarm and reminder set by the extension user.
2433
Select the Voice Guided Alarm Verification check box to enable extension users to confirm the Time
they have set for an alarm or the Date and Time they have set for a reminder. Default: Enabled.
How to use
Alarm set by Extension Users
Pick up the handset of your telephone and dial 163 VMS prompts: "Enter the time, HH MM in twenty four
hour format323. To cancel all alarms, press '# (pound/hash)'." (EntertimeI.wav)
If no time is entered, VMS prompts: "You have not entered any input" (NoInput.wav)
If invalid time is entered, VMS prompts: "You have entered invalid input." (InvalidInput.wav)
323.The Date and time format depends on the Region/Country selected for the system.
2434
To set alarm, dial valid time VMS prompts: "To set once, Press '1', To set Daily Press '2'."
(SetOnceDaily.wav)
Once Only
Dial 1 VMS responds with: "You have set Wake up Alarm at ." (WakeupVeri.wav) followed by the
If alarm is not set, the VMS responds with: "Sorry! Your Wake Up Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav). VMS further responds with: "Thanks for using this
Service." (Thankservice.wav)
Daily Alarm
Dial 2 VMS responds with: "You have set Daily Wake up Alarm at ." (DailyWakeupVeri.wav) followed
by the prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
Dial 1 to confirm the time set for alarm VMS responds with: "Your Daily Wake up Alarm is set."
(DailyWakeupSet.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
If no alarm is set, the VMS responds with: "Sorry! Your Wake Up Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav). The VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
Dial # (pound/hash) to cancel all alarms VMS responds with: "Your all Wake up Alarm are cancelled."
(WakeupCancel.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
If no alarms are set, the VMS responds with: "Sorry! There is no Alarm to cancel." (Alarmnocancel.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav).
Dial 034 the VMS prompts: "Enter the Extension number for which you have to set or cancel Wake Up
Alarm." (RemoteExt.wav)
Dial 1 to select the extension user for which the Alarm is to be set. VMS responds with: "Enter the time, HH
MM in twenty-four hour format. To cancel all alarms, press '# (pound/hash)'." (EntertimeI.wav)
If no time is entered, the VMS prompts: "You have not entered any input" (NoInput.wav)
If invalid time is entered, the VMS prompts: "You have entered invalid input." (InvalidInput.wav)
To set alarm, dial valid time VMS prompts: "To set once, Press '1', To set Daily Press '2'."
(SetOnceDaily.wav)
324. This option will not be played if Alarm Verification is disabled in the System Parameters.
2435
Once Only
Dial 1 the VMS responds with: "To set it as Personal, Press 1. To set it as Automated, Press 2."
(Alarmmode.wav)
Dial 1 VMS responds with: "You have set Personal Wake up alarm at." (PerWakeupVeri.wav)
followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
Dial 1 VMS responds with: "Your Personal Wake up Alarm is set." (PerWakeupSet.wav) followed by
the prompt: "Thanks for using this Service." (Thankservice.wav)
If alarm is not set, the VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using this
Service." (Thankservice.wav)
OR
Dial 2 the VMS responds with: "You have set Automated Wake up alarm at."
(AutoWakeupVeri.wav) followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2."
(AlarmConf.wav)
Dial 1 the VMS responds with: "Your Automated Wake up Alarm is set." (PerWakeupSet.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
If alarm is not set, VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
Daily Alarm
Dial 2 VMS responds with: "To set it as Personal, Press 1. To set it as Automated, Press 2."
(Alarmmode.wav)
Dial 1 VMS responds with: "You have set Daily Personal Wake up alarm at."
(DailyPerWakeupVeri.wav) followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2."
(AlarmConf.wav)
Dial 1 VMS responds with: "Your Daily Personal Wake up Alarm is set." (PerWakeupSet.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
If the alarm is not set, the VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please
call Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for
using this Service." (Thankservice.wav)
OR
2436
Dial 2 the VMS responds with: "You have set Daily Automated Wake up alarm at."
(DailyAutoWakeupVeri.wav) followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2."
(AlarmConf.wav)
Dial 1 VMS responds with: "Your Daily Automated Wake up Alarm is set." (DailyPerWakeupSet.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
If alarm is not set, VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using this
Service." (Thankservice.wav)
format325. To Cancel all Reminders, Press '# (pound/hash)'. For example, To enter Date 17th March
2008, Dial One Seven Zero Three Two Zero Zero Eight." (AlarmDateI.wav)
If no date is entered then VMS prompts: "You have not entered any input" (NoInput.wav)
If invalid date is entered then VMS prompts: "You have entered invalid input." (InvalidInput.wav)
Dial valid Date the VMS prompts: "Enter the time, HH MM in twenty four hour format." (Entertime2I.wav)
If no time is entered, the VMS prompts: "You have not entered any input" (NoInput.wav)
If invalid time is entered, the VMS prompts: "You have entered invalid input." (InvalidInput.wav)
Dial valid time VMS prompts: "You have set Reminder for..." (ReminderVeri.wav) followed by the
prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
Dial 1 to confirm the date and time set for Reminder the VMS responds with: "Your Reminder is set."
(ReminderSet.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
If Reminder is not set, the VMS responds with: "Sorry! Your Reminder cannot be set. Please call
Operator for further assistance." (ReminderNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
Dial # (pound/hash) to cancel Reminder the VMS responds with: "Your Reminder is cancelled."
(ReminderCancel.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
If no reminder is set, the VMS responds with: "Sorry! There is no Reminder to cancel."
(Remindernocancel.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
Dial 1072 to enter SA mode followed by 035 VMS prompts: "Enter the Extension number for which you
have to set or cancel Reminder." (RemoteExtRem.wav)
Dial 1 to select the station user for which the Reminder is to be set. VMS responds with: "Enter the Date in
DD MM YYYY format. To Cancel all Reminders, Press '# (pound/hash)'. For example, To enter Date 17th
March 2008, Dial One Seven Zero Three Two Zero Zero Eight." (AlarmDateI.wav)
If no date is entered, the VMS prompts: "You have not entered any input" (NoInput.wav)
If invalid date is entered then VMS prompts: "You have entered invalid input." (InvalidInput.wav)
325.The date format in the prompt will be MM DD YYYY, if you selected USA as the Region/Country for your system.
2437
Dial valid date VMS prompts: "Enter the time, HH MM in twenty-four hour format." (Entertime2I.wav)
If no time is entered then VMS prompts: "You have not entered any input" (NoInput.wav)
If invalid time is entered then VMS prompts: "You have entered invalid input." (InvalidInput.wav)
Dial valid time VMS prompts: "To set it as Personal, Press 1. To set it as Automated, Press 2."
(Alarmmode.wav)
Dial 1 VMS responds with: "You have set Personal Reminder for." (PerReminderVeri.wav)
followed by the prompt: "To Confirm326, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
Dial 1 VMS responds with: "Your Personal Reminder is set." (PerReminderSet.wav) followed by the
prompt: Thanks for using this Service." (Thankservice.wav)
If Reminder is not set, the VMS responds with: "Sorry! Your Reminder cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
OR
Dial 2 VMS responds with: "You have set Automated Reminder for." (AutoReminderVeri.wav)
followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
Dial 1 VMS responds with: "Your Automated Reminder is set." (PerReminderSet.wav) followed by
the prompt: "Thanks for using this Service." (Thankservice.wav)
If Reminder is not set, VMS responds with: "Sorry! Your Reminder cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
Dial # to cancel Reminder VMS responds with: "Your Reminder is cancelled." (ReminderCancel.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
326.This option will not be played, if Alarm Verification is disabled in the System parameters.
2438
How it works
For this feature to work,
you must enable DISA-PIN Authentication-Multiple Calls on the desired trunk: CO, Mobile, SIP,
T1E1PRI, BRI.
you must select Voice Mail Auto Attendant as the Auto Attendant option on the desired trunks: CO,
Mobile, SIP, T1E1PRI, BRI.
you must enable DISA in the Class of Service (COS) of the extension the caller is allowed to log into
(using PIN Authentication).
you must change the default User Password (1111) of the extension the caller is allowed to log into.
The incoming call on the trunk is answered by the VMS Auto Attendant. By default, the VMS greets the
caller with the Greeting Message followed by the Welcome Message: "Welcome! Please dial the extension
number Or to dial by name press 7. To leave message, press 6. To go to operator, press 9. For more
options, press 0. To disconnect, press #."
The caller must dial the DISA Login Code, 1079. The VMS checks if DISA-PIN Authentication-Multiple
Calls is enabled on the trunk for the current time zone, that is, working hours, break-hours and nonworking hours.
The VMS finds DISA is enabled on the trunk and prompts the caller: Please enter the Extension Number.
(Entextn.wav), and starts the First Digit Wait Timer (programmable; default: 25 seconds).
The caller must dial the DISA Extension Number before the expiry of this timer.
2439
DISA is enabled in the CoS of the dialed Extension Number, the VMS prompts: Please enter your
password. (EntPwd.wav) and waits to receive digits till the expiry of the First Digit Wait Timer.
The Password must be dialed by the caller before the expiry of this timer.
The Password is valid and the DISA Login is successful. The caller is logged into the dialed Extension
Number.
When the caller goes Off-hook by dialing the Off-hook code #1, the system plays the internal dial tone and
waits for the caller to dial digits.
If the caller dials an external number using a CO trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes)327.
The system waits for the caller to dial digits within the DISA Inactivity Timer.
The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any digit
within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed; 15
seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA session.
If digits are received before the end of the Warning Beeps, the system reloads the DISA Inactivity Timer.
The caller can make as many trunk calls and internal calls as the caller wants.
The caller can terminate the DISA login session either by disconnecting from the remote end or by dialing
the Termination Code #9.
The VMS plays the default Greeting Message, Welcome Greeting and DISA prompts to the callers. You can
customize them as per your requirement, if required.
How to configure
For instructions to enable DISA-PIN Authentication-Multiple Calls and Voice Mail Auto Attendant on the
desired trunks, see the topic Trunk Feature Template in Configuring Trunks.
For instructions to enable DISA in the CoS of the extensions which you want to allow callers to access
using DISA, see Class of Service (COS).
To change the default User Password (1111) of the extensions which you want to allow callers to access
using DISA, see User Password and System Security - V10R11 and later.
To set the DISA Timers as per your requirement, see System Timers and Counts.
To customize the DISA VMS Prompts as per your requirement, see Recording Voice Messages.
327. DISA Inactivity Timer is not applicable for T1E1PRI lines, BRI lines, SIP and Mobile trunks.
2440
Sending Messages
Whats this?
The VMS enables extension users to send messages to other extensions that have a mailbox. An extension user
can send a message to as many as 10 destinations at a time. The extension user can send the message either to a
specific mailbox or to a Distribution List.
VMS also gives facility to the Sender of the message to request a read receipt of the message sent. When the
Recipient has read the message, the VMS generates a file containing the first 5 seconds of the message that was
sent and delivers it to the Sender's mailbox in the form of a new message with the Date and Time stamp (if enabled
in Play Message Details) and the prompt: "This message was read by <Extension Name> <5 seconds of message
sent>". If the Sender does not request read receipt, no such message is delivered to the Sender.
If the sent message is not delivered to the Recipient, the VMS generates a file containing the first 5 seconds of the
message that was sent by the Sender and delivers it to the Sender's mailbox in the form of a new message with the
prompt: "This message was not delivered to <Extension Name><5 seconds of the message>".
How to use
VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
Enter your mailbox password VMS prompts: "You have n new/no new messages."
VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
Dial valid extension numbers/distribution list number VMS prompts: "Record your message after the
beep and press # (hash/pound) to end".
Speak to record the message and press # (hash/pound) to end. VMS plays back the recorded message
and prompts: To re-record press 1, to confirm press 2.
After you press 2 to confirm, VMS prompts "To request read receipt press '1', to ignore read receipt press
'2'."
If Message Verification flag is disabled then VMS will not playback the recorded message.
If the message could not be delivered to the destination you dialed, the VMS responds: "Sorry the
message cannot be sent." (Pending.wav)
2441
Extension users must be careful in dialing destination numbers. If invalid destination is entered then the
VMS will clear all the entries and will ask the mailbox owner to re-enter all the destinations again.
Once a valid destination number is entered and no more extensions are selected, the VMS
understands it to be the end of list and sends the message.
2442
Redirecting Messages
Whats this?
The VMS offers extension users to re-direct the messages in their mailbox to another mailbox. The feature can be
used by employees who are out of office or unable to access their mailbox. Using Redirect Messages, they can
ensure that important messages are attended to by their colleagues in their absence.
To be able to use this feature, extension users must have the option Allow Mailbox Management enabled
on their extension. By default, Allow Mailbox Management is disabled. For instructions to provide this
feature to extension users, see the Voice Mail Settings of the respective extension typesDKP, SLT, ISDN
Terminal, SIP Extension and Department Call.
How to use
VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
Enter your mailbox password VMS prompts: "You have n new/no new messages."
VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
Dial 4 The VMS responds with: For Mailbox Name, Press '1', For message redirection, Press '2', To
delete all old messages of your mailbox, press '3', For Mailbox Greetings, press 4, For Assistance
Number Programming, press 5 For Personal Number Programming, press 6, To go to previous menu
press '0'. (Chgmbset.wav)
Dial 2 VMS prompts: "To set message redirection press '1', to cancel message redirection press '2', to
go to previous menu press '0'." (MbMsgRd.wav)
Dial 1 to set message redirection VMS prompts: "Enter the destination extension." (MsgRdHdl.wav)
Dial valid destination extension VMS responds: "Command has been executed." (Okcmd.wav)
Dial 2 to cancel message redirection VMS responds: "Command has been executed." (Okcmd.wav)
2443
2444
Click Extension.
Now, Search Extension, by entering either extension number as Extension Number, or by entering the
name of the extension as Extension Name.
Click Submit.
In Redirect Messages to Extension, enter the extension number on which you want the messages to be
redirected.
Click the desired Department Group Number tab for which you want to set message redirection.
In Redirect Messages to Extension, enter the extension number on which you want the messages to be
redirected.
Exit SA mode.
2445
To be able to use this feature, extension users must have the option Allow Mailbox Management enabled
on their extension. For instructions to provide this feature to extension users, see the Voice Mail Settings of
the respective extension typesDKP, SLT, ISDN Terminal, SIP Extension and Department Call.
How to use
2446
VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
Enter your mailbox password and the VMS prompts: "You have n new/no new messages."
VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
Dial 4. The VMS responds with: For Mailbox Name, Press '1', For message redirection, Press '2', To
delete all old messages of your mailbox, press '3', For Mailbox Greetings, press 4, For Assistance
Number Programming, press 5 For Personal Number Programming, press 6, To go to previous menu
press '0'. (Chgmbset.wav)
Dial 4. The VMS prompts: For Personal Greetings, press 1, For Conditional Greetings, press 2, To go to
previous menu press '0'. (MbGrt.wav)
Dial 1. The VMS prompts: "For working hours greeting, press '1', for break hours greetings, press '2', for
non-working hours greeting, press 3. (MbGrtZone.wav)
Dial the digit of the desired time zone. The VMS prompts: "To record greeting, press 1, to playback
greeting, press 2, to go to the previous menu, press 0. (MbGrtPlay.wav)
To record the message, press 1. You can record a greeting message of 60 seconds. If the message
duration is less than 60 seconds, press # after you have completed the message.
The VMS prompts: "To record greeting, press 1, to playback greeting, press 2, to go to the previous
menu, press 0. (MbGrtPlay.wav)
On completion of play back of the greeting message, the VMS prompts: "To record greeting, press 1, to
playback greeting, press 2, to go to the previous menu, press 0. (MbGrtPlay.wav)
The recorded greetings can be deleted, if required. To do so, see Delete Voice Mail Messages and Greetings in
Voice Mail Memory Status.
2447
To be able to use this feature, extension users must have the option Allow Mailbox Management enabled
on their extension. For instructions to provide this feature to extension users, see the Voice Mail Settings of
the respective extension typesDKP, SLT, ISDN Terminal, SIP Extension and Department Call.
How to use
2448
VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
Enter your mailbox password and the VMS prompts: "You have n new/no new messages."
VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
Dial 4. The VMS responds with: For Mailbox Name, Press '1', For message redirection, Press '2', To
delete all old messages of your mailbox, press '3', For Mailbox Greetings, press 4, For Assistance
Number Programming, press 5 For Personal Number Programming, press 6, To go to previous menu
press '0'. (Chgmbset.wav)
Dial 4. The VMS prompts: For Personal Greetings, press 1, For Conditional Greetings, press 2, To go to
previous menu press '0'. (MbGrt.wav)
Dial 2. The VMS prompts: "For Busy, press '1', for No Reply, press '2', for Unconditional, press 3, To go to
previous menu press '0'. (MbGrtCond.wav)
Dial the desired digit. The VMS prompts: "To record greeting, press 1, to playback greeting, press 2, to
go to the previous menu, press 0. (MbGrtPlay.wav)
To record the message, press 1. You can record a greeting message of 60 seconds. If the message
duration is less than 60 seconds, press # after you have completed the message.
The VMS prompts: "To record greeting, press 1, to playback greeting, press 2, to go to the previous
menu, press 0. (MbGrtPlay.wav)
On completion of play back of the greeting message, the VMS prompts: "To record greeting, press 1, to
playback greeting, press 2, to go to the previous menu, press 0. (MbGrtPlay.wav)
The recorded greetings can be deleted, if required. To do so, see Delete Voice Mail Messages and Greetings in
Voice Mail Memory Status.
2449
Message Verification
Whats this?
Message Verification enables extension users to check the message they have recorded before sending it to
someone.
Message Verification allows callers to check the message they have recorded in the mailbox of an extension user.
Thus Message Verification is used in the VMS features Leaving a Message, Sending Messages, and Broadcast
Message.
How it works
With Message Verification enabled, each time a caller or an extension user records a message, the VMS
offers to the caller/extension user the option verify the recorded message and re-record the message, if
they want.
When the caller/extension user uses the option to verify and re-record the message, the VMS sends the
message to the mailbox of the receiver.
How to configure
By default, Message Verification is enabled in the VMS. So, callers and all extension users with voice mail facility
can verify the message they have recorded. If you want to disable this feature,
2450
Open Jeeves.
2451
2452
By default, Message Verification is enabled. To disable Message Verification, clear the check box.
Message Notification
Whats this?
The Voice Mail System (VMS) sets Message Wait on the extension, whenever a new message arrives in its
personal Mailbox of the extension. The VMS indicates the new message to the extension as per the Type of
Message Wait Notification set for the extension. This may be in the form of a Stuttered Dial Tone, a Voice
Message, Ring or LED Lamp. See Message Wait to know more.
The Message Notification feature of the VMS is an extension of the Message Wait feature. When Message Wait
Notification Type for an extension is set as Ring, the VMS makes a Message Notification call to the extension.
How it works
Whenever there is a new message in As mailbox, the system will play the Message Wait Ring (Short,
Fast) on extension A. See Distinctive Rings.
Extension A will ring for the duration of the Message Wait Ring Timer (configurable; default: 30 seconds).
When Extension A answers the call within this timer, Extension A gets connected to the VMS.
The VMS answers the call and allows access to Extension As mailbox.
If Extension A does not answer the Message Notification call within the Message Wait Timer, the system
will ring on the extension again for as many times as the Message Wait Ring Count (configurable; default:
10 times), and at the interval set as the Message Wait Ring Timer Interval (configurable; default: 30
minutes).
How to configure
To provide Message Notification Call to extensions, you must configure Ring as Message Wait Notification Type
for the extension in its Voice Mail Settings.
You may also configure the related Message Wait Timer, the Message Wait Ring Count and the Message Wait
Ring Interval, if required. See System Timers and Counts for instructions.
How to use
Go Off-hook, when your extension rings to indicate Message Wait (short, fast ring),
The VMS greets with the message:This is your message notification call.
The VMS plays the message: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
2453
Mailbox Settings
Whats this?
The VMS allows extension users to change the settings of the following facilities of their mailbox:
Record the Extension Name for their mailbox.
Redirect Messages from their mailbox.
Delete all old messages
Record Personal and Conditional Greetings for their mailbox.
Configure the Personal (Mobile/Alternate) Number and the Assistance Number.
To be able to change their mailbox settings, extension users must have the option Allow Mailbox
Management enabled on their extension. By default, Allow Mailbox Management is disabled. For
instructions to provide this feature to extension users, see the Voice Mail Settings of the respective
extension typesDKP, SLT, ISDN Terminal, SIP Extension and Department Call.
How to use
VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
VMS prompts: "To listen to new messages press '1', to listen old message press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
The VMS responds with: For Mailbox Name, Press '1', For message redirection, Press '2', To delete all
old messages of your mailbox, press '3', For Mailbox Greetings, press 4, For Assistance Number
Programming, press 5 For Personal Number Programming, press 6, To go to previous menu press '0'.
(Chgmbset.wav)
Dial 1 VMS prompts: "To record Name, press '1'. To play Name, press '2'. To go to Previous menu,
press '0'." (MailboxName.wav).
Dial 1 to record name for mailbox VMS prompts: "Record your name after the beep and press # (hash/
pound) to end." (Recname.wav)
Names for extensions can also be recorded from the System Administrator mode. See Recording
Extension Names.
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Dial 3 VMS prompts: You are about to delete all old messages of your mailbox. To proceed press 1, to
cancel press 2. (DelAllCnf.wav)
Dial 1 to delete all old messages in your mailbox VMS responds with: Your old messages have been
deleted. (DelAllDone.wav)
The SE or SA can Delete Messages for a range of Extensions, see Delete Voice Mail Messages and
Greetings in Voice Mail Memory Status
Dial 5 VMS prompts: Enter the Destination Number and dial hash (#)/ pound (#) to end.
(MsgPhonel.wav)
Make sure the Assistance Number is an extension number.
Dial 6 VMS prompts: Enter the Destination Number and dial hash (#)/ pound (#) to end.
(MsgPhonel.wav)
Any external number can be configured as the Personal (Mobile/Alternative) Number.
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Listening to Messages
Whats this?
The callers/extension user leave messages in the mailbox of extension users, when they are inaccessible or the
user has forwarded his calls to the mailbox. Messages may also be received by the extensions users as
notifications for certain events. User should access their mailboxes to listen to the messages.
VMS offers two options:
To listen to old messages.
To listen to new messages.
Once the message is heard by the mailbox owner, VMS treats it as an old message and places it in the old
message list. The VMS also offers you the option of saving the message you have heard, as a new message.
How to use
2456
VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
if 80% of the mailbox memory has been consumed, the VMS prompts the caller: Your Mailbox is 80%
Full. Please Delete few messages. (MB80Full.wav).
if 100% of the mailbox memory has been consumed, the VMS prompts the caller: Your Mailbox is Full.
Please Delete few messages. (MBFull.wav)
VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav).
On the completion of a message, the VMS plays the message: To replay the message press '1', for
Message Details (date and time stamp) press '2', to delete the message press '3', to play the next
message press '4', to forward the message press '5', to save the message as new press '6', to go to
previous menu press '0'. (Mmsmo.wav).
Leaving a Message
Whats this?
The VMS allows,
To leave a message, the called extension must have a mailbox. The length of message recorded by the callers/
extension users must not exceed the message length set for the called extensions mailbox. If the message
recorded by the callers/extension users exceeds the message length set for the called extensions mailbox, the
VMS will stop recording the message after the time set and save the partially recorded message.
How to use
External Callers
The VMS Auto Attendant answers the call. The VMS greets the caller with the Greeting message followed
by the Welcome Message: "Welcome! Please dial the extension number Or to dial by name press 7. To
leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #."
The caller dials 6 to leave message VMS prompts: "Enter the Extension number for which you wish to
leave message." (LeavemsgE.wav)
The caller dials the desired extension number VMS prompts: "Record your message after the beep and
press # (hash/pound) to end. (Recmsg.wav)
If no mailbox is assigned to the dialed extension number, the VMS prompts: "Mailbox not assigned. To
disconnect, press 1. To go to Home Position, Press 0." (NoMailbox.wav)
The caller speaks to record the message and presses # to end recording. The VMS plays back the recorded
message and prompts: To re-record the message press 1, to confirm press 2." (RecAgain.wav)
If Message Verification flag is disabled, the VMS does not playback the recorded message and prompts:
"Your message has been recorded" press # to end message recording.
Extension Users
The VMS plays the Welcome Message: "Welcome! Please dial the extension number Or to dial by name
press 7. To leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect,
press #."
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Dial 6 to leave message VMS prompts: "Enter the Extension number for which you wish to leave
message." (LeavemsgE.wav)
Dial the desired extension number VMS prompts: "Record your message after the beep and press #
(hash/pound) to end. (Recmsg.wav)
If no mailbox is assigned to the dialed extension number, the VMS prompts: "Mailbox not assigned. To
disconnect, press 1. To go to Home Position, Press 0." (NoMailbox.wav)
Record your message and press # to end recording. The VMS plays back the recorded message and
prompts: To re-record the message press 1, to confirm press 2." (RecAgain.wav)
If Message Verification is disabled, the VMS does not playback the recorded message and prompts:
"Your message has been recorded", press # (hash/pound) to end.
If the extension you called is busy, you may leave a message for another extension:
It is mandatory for the caller/extension user to terminate the recording by dialing # (or the digit configured).
If recording of the message is terminated simply by going on-hook, the VMS will not terminate the recording
and the call will be disconnected only after time-out, that is, the Maximum Message Length configured for
the extension.
2458
How it works
The VMS offers the following options to extensions when their mailbox is full:
Not offer the caller to record a message.
Overwrite the existing messages in the mail box with the new message.
Deliver the new message to the General Mailbox.
If you configure Delivery of new messages to General Mailbox on an extension, whenever the mailbox of the
extension is full, the VMS will offer the caller to record a message. This message will be recorded in the General
Mailbox.
The extension user, whose mail box is full, can listen to the new message by accessing the General Mailbox.
The extension user can access the General Mailbox, if this feature is enabled in the Class of Service of the
extension and the Password assigned to the General Mailbox (if configured) is known to the extension user. For
details see Configuring General Mailbox Settings.
How to configure
To offer extension users the facility of the General Mailbox when their mailbox is full, you must do the following:
Configure the option When Mailbox is full in the Voice Mail Settings of the extension.
For the option When Mailbox is full, select Deliver New Message in General Mailbox.
For instructions on configuring these parameters on the different extension types, see Voice Mail Settings
of the extensions.
Enable the feature General Mailbox in the Class of Service (COS) of the extension.
If required, assign a Password for accessing the General Mailbox, see Configuring General Mailbox
Parameters using Jeeves.
How to use
For EON & Extended IP Phone Users
2459
2460
Lift handset.
Dial 1176.
Follow VMS prompts.
Forwarding Messages
Whats this?
The VMS enables extension users to forward messages of their mailbox to other mailboxes.
How it works
The Forwarding Messages feature of the VMS offers to extension users the following options:
forward messages after adding a comment.
forward messages without adding comment.
forwarding messages with Message Read Receipt request.
Before forwarding a message, the VMS asks the Sender, if the Sender needs a confirmation that the message has
been read by the Recipient.
If the Sender requests for 'Message Read Receipt', the VMS remembers stores this request. When the Recipient
reads the message, the VMS generates a file containing the first 5 seconds of the message that was sent by the
Sender and delivers it to the Sender's mailbox in the form of a new message with a prompt: "This message was
read by <Extension Name><5 seconds of the message sent>" with the Date and Time (if this is enabled in Play
Message Details) at which the message was read.
In case the message was not delivered to the Recipient, the VMS generates a file containing the first 5 seconds of
the message that was sent by the Sender and delivers it to the Sender's mailbox in the form of new message with
the prompt: "This message was not delivered to <Extension Name><5 seconds of the message>" with the Date
and Time (if enabled in Play Message Details).
How to use
VMS responds with: "You have <n> new messages" followed by "Enter your mailbox password"
(Enterpwd.wav)
Enter your mailbox password VMS responds with: "You have 0/n new messages". (Nonewmsg.wav/
Newmsg.wav)
The VMS prompts: "To listen to new messages press '1', to listen to old message press '2', to send a
message press '3', to change your mailbox settings press '4', to go to home position press '0'".
(Mmmm.wav)
Dial 1 or Dial 2 and listen to the messages VMS prompts: "To replay the message press '1', for Message
Details (Date and Time stamp) press '2', to delete the message press '3', to play the next message press
'4', to forward the message press '5', to save the message as new press '6', to go to previous menu press
'0'." (Mmsmo.wav)
Dial 5 VMS prompts: "To forward the message without comment press '1', to forward the message with
comment press '2', to go to previous menu press '0'." (Fwdhow.wav)
2461
To forward message without comment dial '1'. The VMS prompts: "Enter the Destinations". (Msgdest.wav).
OR
To forward message with comment dial '2'. The VMS prompts: "Enter the Destinations". (Msgdest.wav)
The VMS prompts: "Record your message after the beep and press hash(#) to end." (Recmsg.wav)
Speak to record your comment and press # to end recording. VMS prompts: "To re-record the message
press '1', to confirm press '2'." (RecAgain.wav) Dial 2 to confirm.
The VMS prompts: "To request read receipt press '1', to ignore read receipt press '2'." (Askcfrm.wav)
Dial 1 or 2. VMS responds with "Your message has been sent". (Msgsent.wav)
Extension users must be careful in dialing destination numbers. If invalid destination is entered then the
VMS will clear all the entries and will ask the mailbox owner to re-enter all the destinations again.
A message can be forwarded to maximum 10 destinations. A destination can be an extension or a a
distribution list.
2462
For Email Notification to function, you must configure the SMTP Settings .
How to configure
To be able to use this feature, extension users must have Message Wait Notification via Email enabled in their
Voice Mail Settings.
You can select the desired notification option: Send without attachment or Send with attachment and specify
the E-mail Address of the extension user to which the notifications are to be sent.
For the General Mailbox you only need to specify the E-mail Address to which the notifications are to be sent.
If users want to receive Notifications for their mailbox memory utilization only, configure the E-mail Address and
select Do not sent as the Notification option. For details see Voice Mail for SLT Extensions, Voice Mail for DKP
Extensions, Configuring Voice Mail Settings for SIP Extensions, Voice Mail for ISDN Terminal , Voice Mail for
Department Group and Configuring General Mailbox Settings.
2463
Immediate: Users will receive notifications as soon as a new message arrives in their mail box
Or
Scheduled: Extension users will receive notification at specified time intervals.
You can set the preferred time slots in a day during which notification calls should be made to extension users. In
addition to the time slot preference, you can also choose to receive notification calls on a Holiday.
How it works
For this feature to work, you must do the following configuration for the extension:
Select the type of Notification call.
Define the preferred time slots by configuring Time Zones. You can configure four different Time Zones,
defining the Start Time and End Time for each Time Zone.
Configure the phone number to which the notification call is to be made. If the number is an external
number, configure the Trunk Access Code to be used for making the calls.
When Immediate is selected as the Type of notification,
The system checks the preferred start and end time of the time zones configured for the extension. If the
message has arrived within the preferred time slot (Start and End Time) it immediately makes the
notification call on the number configured for the user.
If the number is an external number, the system dials out the number using the Trunk Access Code (TAC)
assigned for making notification calls.
2464
When the call is answered, the extension user gets connected to the VMS and can listen to the message.
If the notification call is not answered, by default, the system makes three attempts (Message Notification
Retry Count; programable) at an interval of 5 minutes (Message Notification Interval; programable)
between each attempt.
If the notification call remains unanswered after the third attempt, the system will not make any more
attempts to place this notification call. The next notification call will be made only when another new
message arrives in the mailbox of the user between the start and end time of the configured time zone.
The system checks the start time of the time zone(s) configured and the notification call will be made on
the number at the subsequent start time.
If the number is an external number the system dials out the number through the Trunk Access Code
(TAC) assigned for making notification calls.
When the call is answered the extension user gets connected to the VMS and can listen to the message.
If the notification call is not answered, the system makes three attempts (Message Notification Retry
Count; programable) at an interval of 5 minutes (Message Notification Interval; programable) for each time
zone. The system will continue to make attempts to place the notification call till the call is answered.
Thus, when Notification type is Immediate, notification call is made for each message that is received within the
start and end time configured in the time zone.
When Notification type is Scheduled, notification call is made for all messages received before the start time
configured in the time zone. Where multiple time zones are configured, notification call will be made at the start
time of the next time zone.
How to configure
For Message Wait Notification via Call, you need to configure:
Message Wait Notification via Call parameters in the Voice Mail Settings of the desired extensions. For
instructions, see Voice Mail for SLT Extensions, Voice Mail for DKP Extensions, Voice Mail for ISDN
Terminal, Configuring Voice Mail Settings for SIP Extensions, Voice Mail for Department Group and
Voice Mail for Extensions over QSIG.
the preferred time zones in the Message Notification Profile. For instructions, see Configuring Message
Wait Notification Profile.
the Trunk Access Code (TAC) for notification calls to be made to external numbers. See Configuring VMS
General Parameters.
if required, the Message Notification Retry Count, Message Notification Interval and Message Notification
Ring. See System Timers and Counts.
2465
Select the Profile Number which you want to assign to Message Wait Notification via Call.
The Message Wait Notification Profile determines how notification calls are to be made to the desired
numbers. You can configure upto 16 different profiles. In each profile, you can set different time zones
according to the user preferences.
Configure the following parameters against the Profile Number you select:
2466
For Each Time Zone, Time Zone 1 to 4, configure the Start Time and End Time. The valid range is
00:00 to 23:59.
If you want to receive notifications on a holiday, select the Notify on Holiday check box.
Now, assign the profile numbers to the desired extensions. Configure the Message Wait Notification via
Call parameters in the Voice Mail Settings of these extensions.
Dial By Name
Whats this?
The VMS Auto Attendant allows external callers and extension users to reach the desired person in an organization
by dialing the name of that person. This feature useful when caller/extension user cannot recall the extension
number of the person they want to speak to.
How to configure
For this feature to work, each extension users name must be abbreviated and configured on the extension. It is
recommended that the extension users names be abbreviated to the first three letters of the name. As far as
possible, abbreviate names such that no two names are the same.
You must configure the Abbreviated Name in the Voice Mail Auto Attendant settings of the SLT, DKP, ISDN
Terminal, SIP extensions, and Department Groups.
To configure Abbreviated Name for SLT extensions, see Voice Mail for SLT Extensions under
Configuring SLT Extensions for instructions.
To configure Abbreviated Name for DKP extensions, see Voice Mail for DKP Extensions under
Configuring DKP Extensions for instructions.
To configure Abbreviated Name for SIP extensions, see Configuring SIP Extensions for instructions.
To configure Abbreviated Name for ISDN Terminals, see Voice Mail for ISDN Terminal under Configuring
ISDN Terminals for instructions.
To configure Abbreviated Name for Department Groups, see Voice Mail for Department Group under
Department Call for instructions.
How to use
External Callers
The VMS Auto Attendant answers the call. The VMS greets the caller with the Greeting message followed
by the Welcome Message: "Welcome! Please dial the extension number Or to dial by name press 7. To
leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #."
The caller dial 7 VMS prompts: "Please enter first three letters of the name." (DialName.wav)
The caller dials valid digits VMS prompts: "To confirm press '1', to cancel press '2'." (Xfromot.wav)
The caller dials 1 VMS transfers the call as per the transfer type of the selected station. Talk.
If the caller dials invalid digits, the VMS prompts: "Sorry no match found." (Empty.wav) followed by the
prompt: "Welcome! Please dial the extension number Or to dial by name press 7. To leave message,
press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #".
2467
Extension Users
VMS responds with a greeting followed by Welcome Message: "Welcome! Please dial the extension
number Or to dial by name press 7. To leave message, press 6. To go to operator, press 9. For more
options, press 0. To disconnect, press #."
Dial 7 VMS prompts: "Please enter first three letters of the name." (DialName.wav)
Dial valid digits VMS prompts: "To confirm press '1', to cancel press '2'." (Xfromot.wav)
Dial 1 VMS transfers the call as per the transfer type of the selected extension. Talk.
If the digits you dialed are invalid, VMS prompts: "Sorry no match found." (Empty.wav) followed by the
prompt: "Welcome! Please dial the extension number Or to dial by name press 7. To leave message,
press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #".
Extension users can use Dial by Name also to reach another extension, if the called extension is busy or
does not reply.
2468
How to use
External Callers
The VMS greets the caller with the Greeting message followed by the Welcome Message: "Welcome!
Please dial the extension number Or to dial by name press 7. To leave message, press 6. To go to
operator, press 9. For more options, press 0. To disconnect, press #."
The caller dials valid extension number VMS transfers the call as per the Call Transfer Type selected for
the extension.
If the extension number dialed by the caller is invalid, the VMS prompts: "The number is not valid."
(Invalno.wav) followed by the prompt: "Please dial the extension number or to dial by name press 7. To
leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #".
Extension Users
VMS responds with a greeting followed by Welcome Message: "Welcome! Please dial the extension
number Or to dial by name press 7. To leave message, press 6. To go to operator, press 9. For more
options, press 0. To disconnect, press #.
Dial valid extension number VMS transfers the call as per the Call Transfer type selected for the dialed
extension number. Talk.
If the extension number you dialed is invalid, the VMS prompts: "The number is not valid." (Invalno.wav)
followed by the prompt: "Please dial the extension number or to dial by name press 7. To leave
message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #" .
2469
Transfer to Mailbox: When the caller dials the extension number, the VMS Auto Attendant checks if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
Transfer immediately: When the caller dials the extension number, the VMS Auto Attendant transfers the
call on the extension without checking whether it is busy or free.
Transfer when extension rings: When the caller dials the extension number, the VMS Auto Attendant
waits for the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant transfers the call to the mailbox of the extension, if
assigned. If no mailbox is assigned, the VMS Auto Attendant takes the caller back to the Home Node.
Transfer when extension answers: When the caller dials the extension number, the VMS Auto Attendant
transfers the call when the extension answers (goes OFF-Hook).
If the extension does not answer328, the VMS Auto Attendant transfers the call to the mailbox of the
extension. If no mailbox is assigned to the extension, the VMS Auto Attendant takes the caller back to the
Home Node.
Transfer when extension permits: The VMS Auto Attendant prompts the caller to record his/her name. It
puts the caller on hold and places the call on the desired extension. If the extension is free and answers
the call, the VMS announces the callers name to the extension user and prompts the extension user to
choose whether or not to speak to the caller. If the extension user chooses to talk, the VMS transfers the
call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the extension user
and asks the caller to leave a message.
If no mailbox is assigned to this extension user, the VMS Auto Attendant takes the caller back to the Home
Node.
328. The VMS will wait for the duration of the Built-In Auto Attendant Ring Timer (default: 30 seconds; the timer is configurable). If the
call is not answered before this timer expires, it is treated as No Reply.
2470
How to configure
Call Transfer Type must be configured in the Voice Mail Auto Attendant parameters on each extension.
To select Call Transfer Type in the Voice Mail Auto Attendant parameter on the extensions and Department
Groups, for instructions:
See Voice Mail for SLT Extensions under Configuring SLT Extensions.
See Voice Mail for DKP Extensions under Configuring DKP Extensions.
See Voice Mail for ISDN Terminal under Configuring ISDN Terminals.
2471
Broadcast Message
Whats this?
Broadcasting Message allows you to send the same message to all extension users having voice mail, at the same
time. You can use Broadcast Message to make general announcements like hosting of an event, an unplanned day
off, and other such activities or events.
How to use
To Broadcast Message,
Enter SA Mode.
Dial 1072-301
VMS prompts: "Record your message after the beep and press any digit to end". (Recmsg.wav)
Speak to record your message after the beep, and press # (hash/pound) to end the message.
VMS prompts: "To re-record the message press 1, to confirm press 2". (RecAgain.wav)
Dial 2 to confirm VMS responds with "Your message has been sent." (Msgsent.wav)
If you press '1', the VMS prompts: "Record your message after the beep and press any digit to end"
(Recmsg.wav). Follow the prompts.
If Message Verification is disabled, the VMS will not offer to verify and re-record your message. Your
message will be sent to all mailboxes and the VMS will respond with: Your message has been sent
(Msgsent.wav).
The length of the message you want to broadcast must be equal to or less than the minimum of message
length programmed for the mailboxes, or your message will be truncated. For instance, if the Maximum
message length for a mailbox is configured as 15 seconds, maximum length of the message to be
broadcast must be less than or equal to 15 seconds. If the broadcast message exceeds this limit, the
system will play the first 15 seconds and truncate the remaining part of the message.
2472
CHAPTER 15
System Maintenance
Configuration Upload
ETERNITY provides you the facility to upgrade the system software at the click of a button. ETERNITY supports an
embedded FTP329 server which can be used for Uploading and Downloading Configuration Files.
Click My Computer.
Enter the IP Address of the Master Ethernet Port of ETERNITY in the Address Bar as ftp://192.168.1.101
329.FTP or File Transfer Protocol is a standard Internet Protocol that is used to exchange files between computers on the IP network.
2473
2474
In the Password field, enter the SE password (default 1234) and click the Log On button.
Open the config folder. All the configuration files will be displayed (the extension of these files is cfg)
You may either delete the existing files or copy these files to another location as backup.
Restart the system after uploading the files. The new configuration will be applicable only after the system
restarts.
2475
2476
Right click the first file, select the option Open link in FireFTP
The folders and files on the computer appear on the left. The FireFTP pane on the right displays the
existing files in the config folder. Delete the existing file in the config folder.
Select the config files from the computer pane. Click the upload arrow to copy the file in config folder in the
FireFTP pane.
Restart the system after uploading the files. The new configurations will be applicable only after the system
restarts.
2477
Follow the same steps as above. When the FireFTP window opens, click the Download button.
The selected configuration files will appear in the selected folder in the computer pane.
Before you upgrade the system firmware, you can save the existing application/configuration/driver/ web
pages folder and files as backup for retrieval, later. When required, you can copy and paste the same
folder back on the FTP server of ETERNITY.
You can archive the Back-up of configuration files by tagging the back-up folders on the PC by date.
2478
Firmware Upgrade
ETERNITY provides you the facility to upgrade the system software at the click of a button. ETERNITY supports an
embedded FTP330 server which can be used for Uploading and Downloading System files.
Click My Computer.
Type the IP Address of the Master Ethernet Port of ETERNITY in the Address bar.
In the Password field, enter the SE password (default 1234) and click the Log On button.
330.FTP or File Transfer Protocol is a standard Internet Protocol that is used to exchange files between computers on the IP network.
2479
2480
Open the system folder. The main application file, eternity will be displayed. This is an executable file.
You may either delete the existing files/folder or copy these files/folder to another location (as backup).
Restart the system after uploading the files.The new firmware will be applicable only after the system
restarts.
2481
2482
Right click on the first file, select the Open link in FireFTP option
The folders and files on the computer appear on the left. The FireFTP pane on the right displays the
existing files in the system folder. Delete the existing file in the system folder.
Select the system files from the computer pane. Click the upload arrow to copy the file in system folder in
the FireFTP pane.
2483
The FireFTP pane on the right displays all the files you copied into the system folder.
Restart the system after uploading the files. The new firmware will be applicable only after the system
restarts.
After the system restarts, login to Jeeves and click the Firmware Upgrade link again. Click on the View
button to see the list of parameters that have been set to default values.
Before you upgrade the system firmware, you can save the existing application/configuration/driver/ web
pages folder and files as backup for retrieval, later. When required, you can copy and paste the same
folder back on the FTP server of ETERNITY.
2484
Auto Update
ETERNITY provides you the facility to selectively update the firmware of the cards installed in the system. In Auto
Update, the system automatically downloads the firmware files stored on its embedded FTP server.
This feature is supported in ETERNITY ME only. All cards of ETERNITY ME, except the VoIP and VMS
Card, can be auto updated.
If the CO Card is being updated, you will not be able to conduct the AC Impedance Test.
To auto update the firmware of cards,
Slot No.: These are the number of universal slots in the system. These will differ according the variant of
ETERNITY you have.
Card Name: The type of card installed in the slots. For example, SLT16, SLT8, Switch.
Firmware: The current firmware version and revision of the card, for example V05R10.
2485
Firmware Available on the FTP: The firmware version and revision of the card available on the
embedded FTP server of the ETERNITY.
To update the current firmware of any of the cards that appear on this page, with the firmware available on
the FTP for the card, select the Firmware Update check box.
.
For example, in slot 8 the T1E1 Dual Card is installed. The Firmware field displays the current version and
revision of the card, V03R09 and the Firmware available on the FTP field displays V03R12. If you select
the Firmware Update check box of this card, ETERNITY will update the firmware of the T1E1 Dual Card
to V03R12.
2486
If you have selected the Firmware Update check box for multiple cards, ETERNITY will update only
two cards simultaneously.
If the Switch Card is being updated, no other card can be updated simultaneously.
Backup-System Configuration
ETERNITY stores the current settings of hardware and software features in the System Configuration data files.
System Configuration data may be lost when:
Therefore it is advisable to Back-Up System Configuration files to restore original system configuration.
System Configuration files can be stored on a computer. The ETERNITY provides an embedded FTP server to
transfer Configuration files on to a computer.
System Configuration files can be transferred to a computer over the FTP.
You can access the FTP using Windows FTP or the Configuration Upload link in the Jeeves.
Follow the instructions given for Configuration Upload to access the Windows FTP and the Upload Configuration
link, to access the configuration files. Copy the configuration folder to the local disk as back-up.
You can archive the Back-up of configuration files by tagging the back-up folders on the computer by date.
2487
Backup-System Software
The ETERNITY System Software may be accidentally deleted or corrupted during maintenance and upgrade
procedures. To prevent data loss, it is advised to back-up System Software.
This can be done by storing the System Software files on a computer, using the embedded FTP server provided by
the ETERNITY.
System Software files can be transferred to a computer over the FTP using the GUI.
You can access the FTP using Windows FTP or the Firmware Upgrade link in the Jeeves.
Follow the instructions given for Firmware Upload to access the Windows FTP and the Firmware Upgrade link, to
access the system folder. Copy the system folder to the local disk as back-up.
2488
Backup-SMDR
What's this?
Station Message Detail Records (SMDR), that is, records of internal, incoming and outgoing calls made to/from
extensions of the ETERNITY are stored331 by the system in the 'SMDR Buffer'. The SMDR Incoming Call buffer
has a capacity of storing a maximum of 5000 incoming call records. The SMDR Internal Call buffer can store up to
1000 internal call records, while a maximum of 6000 outgoing call records can be stored in the SMDR Outgoing
Call buffer.
SMDR buffer data can be cleared by the SE or SA manually or the system clears the data automatically when the
SMDR buffer is full, by replacing the oldest call record with the latest (First In First Out logic).
While the SMDR buffer data is maintained even during power failures, accidental data loss is not an uncommon
occurrence.
SMDR data may be lost when
Therefore it is advisable to Back-Up SMDR records to restore accidentally deleted, lost or corrupted files.
Back-up of SMDR records can be stored on a PC for retrieval later. The ETERNITY provides an embedded FTP
server332 to transfer SMDR call records on a PC.
You can access the embedded FTP Server using Windows FTP or the Back up SMDR link under Station
Message Detail Recording, under Configuration.
To access the Windows FTP and FireFTP (when using the Backup SMDR link), follow the instructions given under
Configuration Upload. Copy the smdr folder on the FTP to the local disk as back-up.
ETERNITY stores call records (SMDR) in the text format so that the files are readable when downloaded.
While uploading SMDR files on to ETERNITY, first, remove the current files in the system. Copy the new
files from computer (backup source) on to the system.
You can tag the back-up folders on the computer by date to store the records as archives.
331. ETERNITY will store SMDR only if the SMDR-Storage flag has been enabled. The call records are stored according to the Storage
filters set.
332. File Transfer Protocol (FTP), is a standard network protocol, used to exchange and manipulate files over a TCP computer network
such as the Internet. FTP is commonly used to transfer Web server for everyone on the Internet. It is also commonly used to download program and other files to your computer from other servers.
2489
2490
Select the file you would like to store and right click on it, select the Open link in FireFTP option.
The folders and files on the computer appear on the left. The FireFTP pane on the right displays the
existing files in the SMDR folder.
Select the path where you want to store the back-up files on the PC. Click the OK button.
2491
Default Settings
What's this?
ETERNITY is supplied with preset values for system and feature settings, as which may be altered and customized
by users to match their requirements and preferences. The factory-set values for system and feature settings that
are automatically assigned by the system are referred to as Default Settings or standard settings.
Every configurable parameter in the system has factory-set default values, which may be changed or customized to
match user requirements and preferences.
How it works
The default settings are to be loaded or restored in the following situations:
1. Installing the ETERNITY in a country other than India.
2492
Default
Time
Zone
Default
DST
Mode
Default
DST
Schedule
Type
CPTGa
Default
DKP
Language
001
Afghanistan
GMT+04:30
English
002
Algeria
GMT+01:00
English
003
Antigua and
Barbuda
GMT-04:00
English
Distinctive
Ringb
Opr
TAC
004
Argentina
GMT-03:00
04
Spanish
005
Australia
(Perth)
GMT+08:00 Scheduled
05
English
006
Australia
(Note2)
(Adelaide)
GMT+09:30 Scheduled
05
English
Abbr.
Dialing
Country Country
Code Name
Default
Time
Zone
Default
DST
Mode
Default
DST
Schedule
Type
GMT+10:00
007
Australia
(Brisbane,
Canberra,
Melbourne,
Sydney)
008
Austria
GMT+01:00 Scheduled
009
Bahamas
GMT-05:00
010
Bahrain
GMT+03:00 Scheduled
011
Bangladesh
GMT+06:00
CPTGa
Default
DKP
Language
05
1
3
Distinctive
Opr
TAC
English
German
English
English
Ringb
Abbr.
Dialing
English
012
Belarus
GMT+02:00
013
Belgium
GMT+01:00 Scheduled
English
014
Bhutan
GMT+06:00
English
015
Bolivia
GMT-04:00
Spanish
016
Bosnia and
Herzegovina
GMT+01:00
English
017
Botswana
GMT+02:00
English
018
Brunei
GMT+08:00
019
GMT-02:00
Brazil
(Fernando De
Noronha)
020
Brazil
(Brasilia, Rio
de Janeiro,
Sao Paulo)
GMT-03:00
021
Brazil
(Manaus)
French
English
06
Portuguese
06
Portuguese
GMT-04:00
06
Portuguese
06
Portuguese
Scheduled
022
Brazil (Acre)
GMT-05:00
023
Bulgaria
GMT+02:00
English
024
Cambodia
GMT+07:00
English
025
Cameroon
GMT+01:00
026
Canada (St.
John's)
GMT-03:30
Scheduled
03
English
T3
027
Canada
(Halifax)
GMT-04:00
Scheduled
03
English
T3
028
Canada
(Montreal,
Ottawa,
Toronto)
GMT-05:00
Scheduled
03
English
T3
029
Canada
(Winnipeg)
GMT-06:00
Scheduled
03
English
T3
030
Canada
(Calgary)
GMT-07:00
Scheduled
03
English
T3
031
Canada
(Vancouver)
GMT-08:00
Scheduled
03
English
T3
032
Chile
GMT-04:00
Scheduled
033
China
GMT+08:00
English
08
Spanish
English
034
Colombia
GMT-05:00
Spanish
035
Costa Rica
GMT-06:00
Spanish
036
Croatia
GMT+01:00
037
Cuba
GMT-05:00
038
Cyprus
GMT+02:00
English
039
Czech
Republic
GMT+01:00
English
English
Scheduled
18
040
Denmark
GMT+01:00 Scheduled
041
Egypt
GMT+02:00 Scheduled
11
042
Fiji
GMT+12:00
043
Finland
GMT+02:00 Scheduled
Spanish
09
English
English
English
8
English
2493
Country Country
Code Name
2494
Default
Time
Zone
Default
DST
Mode
Default
DST
Schedule
Type
CPTGa
Default
DKP
Language
Distinctive
Ringb
Opr
TAC
044
France
GMT+01:00 Scheduled
10
French
045
Germany
GMT+01:00 Scheduled
11
German
046
Greece
GMT+02:00 Scheduled
12
English
047
Guyana
GMT-04:00
048
Hong Kong
GMT+08:00
049
Hungary
GMT+02:00 Scheduled
Abbr.
Dialing
English
2
English
English
050
India
GMT+05:30
01
English
051
Indonesia
GMT+07:00
14
English
T1
15
English
16
English
English
052
Iran
GMT+03:30
053
Iraq
GMT+03:00 Scheduled
054
Ireland
GMT
055
Israel
GMT+02:00
056
Italy
GMT+01:00 Scheduled
057
Japan
GMT+09:00
058
Jordan
GMT+02:00
English
059
Kazakhstan
GMT+06:00
English
060
Kenya
GMT+03:00
20
English
061
21
English
062
063
Kuwait
GMT+03:00
064
Kyrgyzstan
GMT+06:00 Scheduled
10
English
065
Lebanon
GMT+02:00 Scheduled
12
English
066
Libya
GMT+02:00
067
Malaysia
(Note1)
GMT+08:00
068
Maldives
GMT+05:00
069
Mauritius
GMT+04:00
070
Mexico
(Mexico City)
GMT-06:00
Scheduled
03
Spanish
T3
071
Mexico
(Chihuahua)
GMT-07:00
Scheduled
03
Spanish
T3
072
Mexico
(Tijuana)
GMT-08:00
Scheduled
03
Spanish
T3
073
Mongolia
GMT+08:00
English
074
Mozambique
GMT+02:00
Portuguese
075
Myanmar
GMT+06:30
076
Namibia
GMT+01:00 Scheduled
T3
077
Nepal
GMT+05:45
078
Netherlands
GMT+01:00
079
080
Nigeria
081
Norway
GMT+01:00 Scheduled
082
Oman
GMT+04:00
083
Pakistan
GMT+05:00
084
Paraguay
GMT-04:00
085
Peru
GMT-05:00
086
Philippines
GMT+08:00
087
Poland
GMT+01:00 Scheduled
26
English
088
Portugal
GMT
27
Portuguese
089
Qatar
GMT+03:00
English
090
Romania
GMT+02:00
English
091
Russia
(Moscow, St.
Petersburg)
GMT+03:00 Scheduled
Scheduled
17
2
English
18
Italian
19
English
T1
English
English
English
22
English
English
English
English
13
03
English
English
English
14
24
GMT+01:00
English
English
15
English
English
English
Scheduled
16
Spanish
Spanish
25
Scheduled
28
English
English
Country Country
Code Name
Default
Time
Zone
Default
DST
Mode
Default
DST
Schedule
Type
CPTGa
Default
DKP
Language
Distinctive
Ringb
Opr
TAC
092
Russia
(Novosibirsk)
GMT+06:00 Scheduled
28
English
093
Russia
(Vladivostok)
GMT+10:00 Scheduled
28
English
094
Singapore
GMT+08:00
30
English
095
Slovakia
GMT+01:00
096
South Africa
GMT+02:00
097
Spain
GMT+01:00 Scheduled
098
Sri Lanka
GMT+05:30
Abbr.
Dialing
English
1
31
English
32
Spanish
English
099
Sudan
GMT+03:00
100
Sweden
GMT+01:00 Scheduled
English
English
9
101
Switzerland
GMT+01:00 Scheduled
German
102
Syria
GMT+02:00 Scheduled
17
English
103
Taiwan
GMT+08:00
English
104
Tajikistan
GMT+05:00
English
105
Thailand
GMT+07:00
33
English
106
Turkey
GMT+02:00
34
English
107
Uganda
GMT+03:00
108
Ukraine
GMT+02:00
109
United Arab
Emirates
GMT+04:00
110
United
Kingdom
GMT
111
English
English
35
English
Scheduled
02
English
T2
Scheduled
03
English
T3
112
Scheduled
03
English
T3
113
Scheduled
03
English
T3
114
Scheduled
03
English
T3
115
Scheduled
03
English
T3
116
03
English
T3
2495
Country Country
Code Name
Default
Time
Zone
Default
DST
Mode
Default
DST
Schedule
Type
CPTGa
Default
DKP
Language
117
Uzbekistan
GMT+05:00
English
118
Venezuela
GMT-04:30
Spanish
English
119
Vietnam
GMT+07:00
120
Yemen
GMT+03:00
English
121
Yugoslavia
GMT+02:00
English
122
Zambia
GMT+02:00
English
123
Zimbabwe
GMT+02:00
English
124
Saudi Arabia
GMT +3.00
English
Distinctive
Ringb
Opr
TAC
T1
Abbr.
Dialing
In addition to the common set of PBX features, there is a distinct set of in-built features for each of these
applications. When ETERNITY is to be installed in any of the two application scenarios, the 'Customer
Profile' - whether the user is an Enterprise or a Hotel - is to be defined at the time of installation.
When the Customer Profile is defined, all features specific to the application Enterprise/Hotel, along with
their default settings are loaded. By default the Customer Profile of ETERNITY is defined as 'Enterprise'.
Refer the ETERNITY Hospitality System Manual to know more.
3. Malfunctioning of the System.
When there is a system malfunction, possibly caused by a programming error that you are unable to
diagnose, you may restore default settings.
2496
Voice Module
Clock Synchronization Parameters
Source Port
Clock Type
PLL TIE Control
PLL Operating Mode
Locking Mode
System Activity Log
System Fault Log
SMDR
SIM PIN of Mobile
Master Ethernet Parameters
Wizard Parameters
Click OK.
2497
The SE password you enter must be the current password. For Example: if it is 1234, enter 4321 and click
OK.
The system will restart and load the default settings. It takes 50 seconds to load default settings. The
system count down in the seconds will appear on your screen as Loading Default Settings...Please Wait
for ____ seconds.
2498
The names programmed for the DKP and SLT will disappear and the default flexible numbers, that is,
extension numbers assigned to the DKP and SLT stations will appear.
Without the SE Password, you cannot restore default values via the software. If you forget the SE
Password, you must resort to hardware default of the SE-Programming Password first. Refer the topic
System Security - V10R11 and later for instructions on restoring the default SE Password.
You cannot default Region; you can only select a region to load the country-specific default settings and
default the system.
2499
PCAP Trace
What's this?
PCAP or packet capture consists of intercepting and logging the traffic passing over a digital network or a part of a
network. PCAP intercepts each packet in the data streams that flow across the network, and can decode and
analyze its contents.
PCAP can be used, among others, to monitor the network, detect and analyze network problems, debug client/
server communications, debug network protocol implementations.
ETERNITY supports PCAP Trace for the Master Ethernet Port, the VoIP Port of ETERNITY, and the Matrix
Extended IP Phones.
Packets traveling over a network are captured and saved in the system. You can save these trace files (packets
captured by the system) on a PC and open these trace files using a graphical packet capture and protocol analysis
tool such as Wireshark or Ethereal.
ETERNITY also supports Filters and 'Promiscuous' mode for capturing packets, which you can use to specify the
types of data packets to be captured.
How to use
Using PCAP Trace for Master Ethernet Port of ETERNITY
PCAP Trace for the Master Ethernet Port allows a maximum of 1 MB of packets can be captured and stored in the
ETERNITY.
To use PCAP Trace for the Master Ethernet Port of ETERNITY,
2500
Under Maintenance, click CPU PCAP Trace link to open the page.
Decide the type of packets to be captured and set the Filter accordingly. The Filter Settings parameter
should be maximum 60 characters in length; all ASCII characters are allowed. By default, this field is
blank. So all packets will be captured.
Refer the following examples to know how to set the Filters.
Examples of Filter settings:
To capture packets which are transmitted from the system, from IP address 192.168.1.191:
Filter Settings = src 192.168.1.191
To capture packets which are received for the system, to IP address 92.168.1.191:
Filter Settings = dst 192.168.1.191
2501
To capture only packets which are transmitted from the system and received to the system, IP address
192.168.1.191:
To capture packets which are transmitted from the system for particular port number only, from IP
address 192.168.1.191 and port number 161
Filter Settings = src 192.168.1.191 and port 161
If you do not enter a valid filter, you will get the message: 'Invalid filter! Please enter valid filter'.
It is not mandatory to set Filters. When the Filter Settings field is left blank, the system will capture all
packets.
You can set the Enable Promiscuous Mode? to Yes, if you want.
When you enable Promiscuous mode, the ETERNITY will capture all network traffic. However, this will
work only in a non-switched environment.
When Promiscuous Mode flag is disabled, the system will capture only traffic that is directly related to it.
Only traffic to, from or routed through the ETERNITY will be picked up by the PCAP Trace.
'Filter Settings' and 'Promiscuous Mode' (enabled) will not be cleared during power down.
Wait for the system to stop packet capturing. The system stops packet capturing once the maximum
allotted memory of 1 MB (RAM) is utilized.
Number of Packets and bytes captured as per the filter setting will be displayed as Packets Captured and
Total Bytes respectively.
Capturing of packets will not stop if you open any other page of Jeeves. So, you may continue using
Jeeves for any other purpose while PCAP Trace is being used.
When the packet capturing is stopped (by you or the system), click the Save Trace File button to save the
files on the systems FTP server.
The FTP Login page will open. Enter the SE password. Save your trace file on you local disk.
The current packets captured will not be deleted after you have saved the trace file. The current packets
will be deleted when you start the PCAP capture again.
2502
Now, you can open the downloaded trace file using Wireshark or Ethereal or any other similar software
which supports opening of trace files.
Click the VoIP Port tab for which you want to carry out PCAP Trace. For example, VoIP Port-1.
Set the Filter for the type of packets to be captured. The Filter Settings parameter should be maximum 60
characters in length; all ASCII characters are allowed. By default, this field is blank. So all packets will be
captured.
You may refer to the following examples for setting the Filters.
2503
Filter Setting
Logic
dst port 80
port 5060
host ip address
host 192.168.1.176
If you enter an invalid filter, you will get an error message: 'Invalid filter! Please enter valid filter' if you do
not enter a valid filter.
It is not mandatory to set Filters. When the Filter Settings field is left blank, the system will capture all
packets.
You can set the flag Enable Promiscuous Mode? to Yes, if you want.
When you enable Promiscuous mode, the ETERNITY will capture all network traffic. When Promiscuous
Mode flag is disabled, the system will capture only traffic that is directly related to it. Only traffic to, from or
routed through the ETERNITY will be picked up by the PCAP Trace.
'Filter Settings' and 'Promiscuous Mode' (enabled) will not be cleared during power down.
Wait for the system to stop packet capturing. The system stops packet capturing once the maximum
allotted memory of 2 MB (RAM) is utilized.
Capturing of packets will not be interrupted if you open any other page of Jeeves. So, you may continue
using Jeeves for any other purpose while PCAP Trace is being used.
When the packet capturing is stopped (by you or the system), you can download the captured file from the
FTP server of the VoIP Card.
The current packets captured will not be deleted after you have saved the trace file. The current packets
will be deleted when you start the PCAP capture again.
2504
If you want to run PCAP Trace for another port, click the desired VoIP Port Number, and repeat the steps
described above.
You may open the downloaded trace file using Wireshark or Ethereal or any other similar software which
supports opening of trace files.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt i on
DN D
Names
Local Menu
CA04
CA03
Redial Release
Hold
abc
3 def
4 ghi
jkl
6 mno
7 pqrs
tuv
9 wxyz
CA02
CA01
LOCAL ME NU
N et wo r k P a r a m e t er s
N et wo r k S t a t u s
2505
N E T W O R K PA R A M E T E R S
M A C :00 :1 b:09 :00 :9a :a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
Sub net Ma sk
G at e w a y A d d r e s s
N E T W O R K PA R A M E T E R S
P P P o E S e rv i c e N a m e
S e r v er A d d re ss
S e r v er P o r t
VLAN S etti ng
PCAP
PCAP T RACE
Sta rt PCAP
PCAP TRACE
Sta rted !
To stop PCAP,
2506
Enter the Local Menu of the phone again by pressing the DSS key.
PCAP T RACE
Stop PCAP
Sto ppe d!
PCAP T RACE
Go idle.
You can download the Trace file from the embedded FTP server of the Extended IP Phone. To access the FTP
server using Windows FTP, do the following:
Go to My Computer.
Type the current IP Address of the Extended IP Phone in the Address bar. For example: ftp://
192.168.201.134
In Password, enter the current User Password set for the phone.
On successful login, the FTP window will open. You will see the different Configuration folders in this
window.
In the ramdisk folder, right click the file trace.pcap and copy it on to your local disk.
Open the trace.pcap file using Wireshark or Ethereal or any other similar software which supports
opening of trace files.
You may also use FireFTP, if you are using Mozilla Fire Fox. Make sure your browser has the FireFTP
Add-on installed.
2507
2508
The page displays the Slot numbers and the card installed in the slot.
To restart a card, select the Restart check box of the respective card.
Click Submit.
Restart System
To restart ETERNITY,
You will get the message, This will restart the system. Do you want to continue?.
Click OK.
The SE password you enter must be the current password. For Example: if it is 1234, enter 4321 and click
OK.
2509
System Debug
You can monitor the state of software ports and IO operations for trouble shooting and identifying faults and errors
using System Debug. ETERNITY supports Syslog Client for debugging. Debug messages are sent to the remote
Syslog Server.
Configuring System Debug using Jeeves,
In Debug port, select the destination port for sending debug messages as either Comm. 1, Comm. 2,
Printer or Ethernet.
If you are using the Syslog Server for debugging, select Ethernet, and configure Syslog Server IP
Address and Port. Valid port range is: 1025 to 65535;514.
2510
Now, select the respective check box of the events/processes you want to debug:
Port State
Card IO
Matrix ETERNITY System Manual
PMS
Parameter Initialization on Power-ON
Error
Communication Manager
File Open/Read/Write
VMS
ACB
Maturity/AOC
Auto-upgradation / RCOC
WebJeeves
You can enable the Debug for the SMS Server Parameters. Select the check box of the respective
parameter you want to debug:
SMTP Client
POP3 Client
SMS Record
SMS Sender
SMS Receiver
For the SMS Server debug, configure the Syslog Server IP Address and Port. Valid port range is:
1025 to 65535;514.
You can also debug a particular Port, by configuring the Port Debug settings.
You can also debug a card installed in a particular slot, by configuring the Slot Debug settings.
Click Submit.
2511
2104-Value
Where, Value is a 4-digits string for each process type, as in this table:
Process Type
Value
Port State
0001
Card IO
0002
PMS
0004
0008
Error
0016
Communication Manager
0032
File Open/Read/Write
0064
VMS
0128
ACB
0256
Maturity/AOC
0512
Auto-upgradation/RCOC
1024
If you need to enable a combination of the above process types, use the sum of their values in the
command.
This command will be saved in the configuration and not changed even in the case of power failure.
Example:
To enable debug for 'VMS' and 'PMS' use values = 0004 + 0128 = 0132 and give command 2104 - 0132.
To disable debug for all process use command: 2104-0000.
2512
Meaning
01
SLT
02
DKP
03
CO
04
BRI
05
T1E1
06
E&M
10
DOP
25
MOBILE
26
SIP TRUNK
29
MAGNETO
Meaning
001
002
004
007
000
Default is 000.
To enable ETERNITY-GE/PE DSP Para 1 Debug, dial:
2184-1-Code
Where,
Code is from 000 to 255.
Code
Meaning
000
001
002
Default is 000.
It is not possible to enable all three debugs at a time, only one of the three can be enabled at a time.
Meaning
001
002
Enable CO Debug
004
08
016
2513
Code
Meaning
032
064
128
255
000
Default is 000.
To enable ETERNITY GE PCM Capture Debug, dial:
2172-Slot Number-Hardware Port Offset- Code
Where,
Code is
Slot Number is 01 to 24
0 for Rx and Tx PCM Capture Disable
1 for Rx PCM Capture Enable
2 for Tx PCM Capture Enable
3 for Rx and Tx PCM Capture Enable
To initiate the debug of IO operations, dial:
2199-Slot Number-1-Port Number-Code
2199-Slot Number-2-Port Number-Port Number-Code
2199-Slot Number-*-Code
Where,
Slot Number is 01 to 24,
Port Number is the Port offset on the card in 2 digit,
Code is 1 for Enable, 0 for Disable
If the Slot No. and the Port Number are programmed as 99 the debug of all slots and ports is generated.
Exit SE mode.
When you upgrade the Firmware of ETERNITY, you can view the names of the configuration files that have
been set to default values by dialing the command 2194
On dialing this command, the file names will appear on the System Debug screen of Jeeves.
You are advised to backup System Configuration before you upgrade Firmware.
Relevant Topics:
1. VoIP Ethernet Port Debug
2515
2. Communication Ports
1749
2514
2515
As per the level selected, debug log will be generated. For example: if debug log of Call is required,
enable 'CALL' level and disable all other debug levels.
Click Submit.
2516
Index
Meaning
01
System
Index
Meaning
02
Serial
03
SIP
04
Call
05
Registered User
06
Registered Trunk
07
BLF/MWI
08
Media
09
VoPP
10
Call Advance
11
12
13
BLF/MWI Advance
14
Media Advance
15
Presence
16
IM
2517
System Details
To view the System Details,
2518
System Usage
To view the active channels and their activities,
2519
2520
CHAPTER 16
VMS Maintenance
The VMS Card of ETERNITY supports an embedded FTP334 server which can be used for:
Upgrading System Files
Downloading extension users mailbox messages
Uploading customized System Greetings
Uploading customized Welcome Greetings
Uploading customized Voice Prompts
Uploading customized Graph and Node messages
To be able to access the FTP server of the VMS Card, connect the Ethernet Port of the VMS Card of ETERNITY
with a stand-alone PC or a PC in the LAN.
Refer the installation instructions for connecting a standalone/LAN as applicable to your model of ETERNITY
(Installing ETERNITY ME, Installing ETERNITY GE, and Installing ETERNITY PE).
When the Ethernet Port of the VMS Card is connected to a standalone/LAN, change the IP Address and Subnet
Mask so that the IP Address of the PC and the Ethernet Port of VMS Cards of ETERNITY do not clash and the PC
and the Ethernet Port of the VMS Card of ETERNITY are in the same Subnet.
Refer the instructions provided under the sub-topic VMS Ethernet Port Parameters.
VMS Configuration
To view the VMS configuration files,
Click My Computer.
334.FTP or File Transfer Protocol is a standard Internet Protocol that is used to exchange files between computers on the IP network.
2521
2522
Enter the current IP Address of the Ethernet Port of the VMS Card of ETERNITY in the Address Bar. The
default IP Address is 192.168.1.131.
In the Password field, enter the current SE password (default 1234). See System Security - V10R11 and
later for more details.
Click the VmsCard folder, the System_Data and the System_Firmware folders will be displayed.
Click the System_Data folder and then click the System_Configuration folder. All the configuration files
will be displayed (the extension of these files is cfg).
2523
This folder stores the files sent by the Master Card to the VMS Card at Power-on.
After power-on, when you update/change any VMS configurations using Jeeves, these changes/updations
are also sent by the Master Card to the VMS Card and stored in this folder.
If you need to replace the VMS Card, make sure you insert the Pen Drive of the old card in the new card,
to avoid loss of existing mailbox messages
to avoid loss of time, as the Master Card will have to re-send all the configuration files to the VMS Card
2524
Click the System_Firmware folder. The BIn folder contains all the firmware files.
You may either delete the existing files/folder or copy these files/folder to another location (as backup).
Restart the system after uploading the files.The new firmware will be applicable only after the system
restarts.
Before you upgrade the system firmware, you can save the existing folder as backup for retrieval, later.
When required, you can copy and paste the same folder back on the FTP server.
2525
2526
Click the desired folder - SLT, DKP, ISDN Terminal, SIP Extension, Department Group. For example,
DKP001_250
Each folder contains sub folders according to the Extension Numbers assigned to the users.
2527
Type of Messages
Messages
Call Type
Call forward
Extension Users
Messages
Abbreviation
CF
002-03072010-1005-CF-616.wav
Call Taping
CT
005-15072010-1048-CT-616629.wav
Conversation
Recording
CR
010-11072010-1356-CR2652630555.wav
Broadcast
Message
BM
009-03212012-1005-BM-2001.wav
Transfer to
Mailbox
TM
010-03212012-0626-TM-629.wav
Leave
Message
LM
007-03212012-0626-LM-2001.wav
Send Message SM
004-03212012-0822-SM-2001.wav
Call forward
with LCS No
reply, all
LS
005-03212012-0626-LM-2001.wav
Redirect
Message
RM
002-03072010-1005-RM-3001.wav
Message
forward
FM
002-03072010-1005-FM-2001.wav
Greeting Messages
for Working
hours(wh), NonWorking hours (nh)
and Break hours (bh)
Station Name
grt-wh.wav
grt-nh.wav
grt-bh.wav
Stn.Name.wav
You may copy these files/folder to another location on the local disk or paste the customized Mailbox
Greetings and the Station Name.
For detailed instructions on recording custom Mailbox Greetings and Station Name, see Recording Voice
Messages and Recording Extension Names.
To listen to the messages, you can use the Windows Media Player.
2528
The existing audio files, with their unique names and .wav extension appear.
Go to the location (CD, Pen Drive or Local disk) where you have stored the custom voice messages.
Click the desired folder. The Thank you message is a response, so open the System Prompts and
Responses folder.
Right click, and paste the file you copied in the System Prompts and Responses folder.
2529
As you are pasting a file with the same file name, you will be prompted if you want to replace the existing
file. Click OK.
2530
VMS Debug
What's this?
The VMS supports debug for the VMS Application and SMTP. You can view debug messages on HyperTerminal or
on Syslog Server335.
To be able to use HyperTerminal for debug, you will need to:
connect a PC to the serial communication port COM1 of the VMS Card. (For instructions, refer Installing
the VMS Card).
configure the communication port attributes for HyperTerminal. (For instructions, refer the Configuring
ETERNITY using COM Port and Communication Ports.)
To be able to use Syslog for debug, you will need to:
connect a PC to the VMS Ethernet Port. (For instructions, refer Installing the VMS Card.)
select the VMS Ethernet Port as the Destination Port.
configure the Syslog Server Address and the Server Port on which Syslog will listen for debug messages.
Destination Port: Select the desired destination port for debug. By default, no destination port is
selected.
335. The VMS supports Syslog Client, which enables the VMS to send debug messages in syslog format to the remote 'Syslog Server'
on the IP network. You can view the system debug messages on the remote Syslog server or any other application which can capture the Syslog debug messages.
2531
Server Port: Enter the address of the Listening Port of the Syslog Server. Valid port range is: 1025
to 65535; 514. By default the remote server port address is '514'.
VMS Application: Select this option if you want to debug the VMS Application. By default, debug of
VMS Application is disabled.
SMTP: Select this option if you want to debug SMTP. By default, debug of SMTP is disabled.
Configuration Transfer: Select this option if you want to debug Configuration Transfer parameter.
2532
Exit SE mode.
Appendix
Technical Specifications
Built-In Resources
System Resources
Description
ETERNITY ME
ETERNITY GE
ETERNITY PE
10S
16S
6S
12S
3SS
3SP
DKP Ports
SMDR/PMS/CAS
Interfaces
USB Port
6SP
1#
1#
1#
3#
3#
3#
Conference
Number of Conference
Participants
21
21
15
15
15
Conference
Maximum No. of
Simultaneous 3-Party
Conference
16
16
16
16
Ethernet Port
Web based
configuration, PMS,
SMDR, System Log
# Optional Connectivity.
Matrix ETERNITY System Manual
2533
System Resources
System Resources
Description
Maximum Capacity
ETERNITY ME
ETERNITY GE
ETERNITY PE
10S
16S
3S
6S
12S
3SS
3SP
6SP
16
19
14
10
16
12
324
516
60
120
240
24
24
48
System Capacity
Maximum Ports
System
Resources
Description
ETERNITY ME
ETERNITY GE
ETERNITY PE
10S
16S
3S
6S
12S
3SS
3SP
6SP
SLT Ports
320
320 or
512*
60
120
240
24
24
48
DKP/DSS
Ports
To connect Proprietary
Digital Key Phones or
DSS Consoles
64 or 128*
64 or 128*
48
48
48
24
24
48
CO (TWT)
Ports
To connect Two-Wire
Trunk Lines
128
128
48
96
128
16
BRI Ports
32
32
12
24
32
12
T1/E1/PRI
Ports
To connect to T1 or E1
PRI Network or ISDN
PRI Compatible Devices
GSM/3G
Ports
To connect to GSM/3G
Network
24 or 64*
24 or 64*
12
24
40
VoIP
Channels per
Card
Simultaneous VoIP
Calls per Card
16/32
16/32
32
16/32
16/32
8/16
8/16
8/16
VoIP Calls
Simultaneous VoIP
Calls participants
320
512
192
384
48
48
80
VoIP Trunks
32
32
16
16
16
VoIP Users
Registration of VoIP
Phones, Softphones
and Mobile SIP Clients
999
999
500
500
500
50
50
50
VoIP Cards
VoIP Phones,
Softphones, Mobile SIP
Clients
10
16
12
2534
Maximum Ports
System
Resources
Description
ETERNITY ME
ETERNITY GE
ETERNITY PE
10S
16S
3S
6S
12S
3SS
3SP
6SP
E&M Ports
To connect E&M
Network
80
128
12
24
48
Voice Mail
System
(Number of
Ports)
16
16
16
16
16
16
16
16
Magneto
Phone Ports
To connect Magneto
Phones
80
128
Redundancy
To provide Redundancy
in case of Primary
Hardware Failure for DC
Power Supply and
Control Cards
Yes
Hot-Swap
Yes
Yes
Door Phone
Ports
To connect Four-Wire
Door Phones
* With PS48VDC-500W
The maximum number of ports supported by the GSM, SLT and DKP Cards may vary according to the type
of Power Supply used. Refer the following table for maximum ports supported with Universal Power
(PSUNI) and DC Power Supply in ETERNITY ME.
GSM
SLT
DKP
96 ports maximum
Analog SLT ports supported for Short Loop with Loop Current
programmed
Loop Current Programmed
20mA
25mA
30mA
35mA
40mA
250
200
175
150
128
2535
GSM
SLT
DKP
128 ports
GSM
SLT
DKP
96 ports maximum
For details of the cards supported by each variant, see the ETERNITY Cards List.
Optional Items
EON48
EONSOFT
DSS64, DSS72
EON74
SPARSH VP248
Executive IP-Phone
Technology
Type of Switching
Processor
32-bit RISC
Architecture
Distributed Processing
Slots
Universal
Loop Start
Dialing
600/900/Complex
2536
39mA max
Loop Limit
-48V nominal
DTMF Detection
ITU-T Q.24
Return Loss
>18dB
Longitudinal Balance
>50dB
Ringing
REN
CLI Reception
Protection
Physical Connector
RJ45
Interface
Loop Limit
100
Speech Level
Protection
Physical Connector
RJ45
EON310
Signaling
Interface
Loop Limit
100
Speech Level
Protection
Physical Connector
RJ45
Display
DSS Keys
Headset
Speaker
2537
Display
Handset
Handset Interface
RJ12
Headset Interface
Dimension
Mounting Angle
35
Enclosure
Power Supply
12VDC, 1.25A
Weight
SPARSH VP248
VoIP
VoIP Protocols
Network Protocol
SIP
NAT
Voice CODECS
Voice
Quality of Service
Data Network
Security
Power Supply
Input
Power Consumption
5W (Typical)
Mechanical
2538
Dimensions (WxHxD)
Material
ABS Plastic
Installation Mounting
Environmental
Operating Temperature
Extended IP Phone
VoIP
Network Protocol
NAT
Voice CODECS
Voice
Quality of Service
Data Network
LCD Display
Security
Power Supply
Input
Power Consumption
5W (Typical)
Mechanical
Dimensions (WxHxD)
Material
ABS Plastic
Installation Mounting
Environmental
Operating Temperature
Loop Start
Loop Limit
1200
2539
600/900/Complex
Pulse Dialing
10/20 PPS
Return Loss
>18dB
Longitudinal Balance
>50dB
CLI Reception
Call Maturity
Protection
Physical Connector
RJ45
ISDN BRI
Channels
2B+D
Personality
Switch Variant
AT&T 4ESS, DMS-100, ETSI NET3, ITU-T Q.921, ITU-T Q.931, NTT
INS64, US NI1 (National ISDN1), France VNx
Protection
Physical
Connector
RJ45 (120)
ISDN PRI
Channels
Personality
Line Coding
Framing
Switch Variant
AT&T 4ESS, AT&T 5ESS, DMS-100, ETSI NET5, ITU-T Q.921, ITUT Q.931, NTT INS64, US NI2 (National ISDN 2), QSIC ECMA,
France VN
Protection
Physical Connector
Supplementary
Services
QSIG ECMA
E1 CAS
Bit Rate
2540
Line Coding
HDB3
Framing
Line Signaling
Register Signaling
MFC-R2
Alarms
Protection
Physical Connector
T1 RBS
Bit Rate
Line Coding
Line Signaling
FXS Loop Start, FXO Loop Start, FXS Ground Start, FXO Ground Start,
E&M (Immediate, Wink Start, Wink Start FGD)
Framing
D4, ESF
Digit Dialing
DTMF
Alarms
ANSI T1.231
Performance
Protection
Physical
Connector
GSM Trunks
GSM Band (MHz)
Compliant
SIM Card
SIM Interface
1.8V, 3V
Transmission
Power
RF Sensitivity
Protocol
AT Command Interface
External Antenna
2541
3G
GSM Band
(MHz)
Compliant
SIM Card
SIM Interface
1.8V, 3V
Transmission
Power
Output Power
Class 4 (2W) at GSM850/GSM900
Class 1 (1W) at DCS1800/PCS1900
Class 3 (0.25W) at UMTS 850/900/1900/2100
RF Sensitivity
Protocol
AT Command Interface
External Antenna
VoIP
VoIP Protocols
Network Protocol
IPv4, TCP, UDP, SNTP, STUN, ARP, ICMP, PPPoE, DHCP, DNS,
SMTP
SIP
NAT/Firewall
Support
Voice Codecs
Line Echo
Cancellation
Voice
Fax
Data Network
Quality of Service
Security
LED Indications
2542
E&M Trunks
Signalling
Speech Interface
2W/4W
Address Signalling
Return Loss
20dB
Transhybrid Loss
Transmit Gain
+/- 1.0 dB
Receive Gain
+/- 1.0 dB
Physical Connector
RJ45
2543
Power Supply
Input
ETERNITY PE
ETERNITY GE
ETERNITY ME
ETERNITY PE3SS-25W
ETERNITY GE6S-30W
ETERNITY ME10S-70W
ETERNITY PE3SP-25W
ETERNITY GE12S-50W
ETERNITY ME16S-100W
Power
Consumption
ETERNITY PE6SP-40W
LED Indications
Mechanical
Dimensions (W x H x D)
Unit Weight
2544
ETERNITY ME10S
ETERNITY ME16S
ETERNITY GE3S
ETERNITY GE6S
ETERNITY GE12S
ETERNITY PE3SS
ETERNITY PE3SP
ETERNITY PE6SP
ETERNITY ME10S
ETERNITY ME16S
ETERNITY GE3S
ETERNITY GE6S
ETERNITY GE12S
ETERNITY PE3SS
ETERNITY PE3SP
ETERNITY PE6SP
Installation
ETERNITY ME10S
ETERNITY ME16S
ETERNITY GE3S
ETERNITY GE6S
ETERNITY GE12S
ETERNITY PE3SS
ETERNITY PE3SP
ETERNITY PE6SP
Environment
ETERNITY PE
ETERNITY GE
ETERNITY ME
Operating Temperature
-10oC to +50oC
(14oF to +122oF)
-10oC to +40oC
-10oC to +50oC
(14oF to +104oF)
(14oF to +122oF)
Operating Humidity
5-95% RH,
Non-Condensing
5-95% RH,
Non-Condensing
5-95% RH,
Non-Condensing
Storage Temperature
-40oC to +85oC
(40oF to +185oF)
-40oC to +85oC
(40oF to +185oF)
-40oC to +85oC
(40oF to +185oF)
Storage Humidity
0-95%, RH,
Non-Condensing
0-95%, RH,
Non-Condensing
0-95%, RH,
Non-Condensing
Compliance
EMI/EMC
Conducted Emission
CISPR 22
Radiated Emission
CISPR 22
IEC 61000-3-2
Voltage Flicker
IEC 61000-3-3
Electro-static Discharge
IEC 61000-4-2
Radiated Susceptibility
IEC 61000-4-3
IEC 61000-4-4
Surge
IEC 61000-4-5
Conducted Immunity
IEC 61000-4-6
IEC 61000-4-8
IEC 61000-4-11
FCC
Conducted Emission:
Radiated Emission:
2545
FCC68
US: MTXMF01BETERNITY
TEC
IR/IPX-01/03 APR2005
EC Directives
R&TTE 1999/5/EC
LVD 73/23/EEC
EMC 89/336/EEC
Safety
Environment Test
Mechanical
2546
Master Card
Switch Card
CO Cards
BRI Cards
E&M Cards
Hardware Version
P-101-007-03-03
P-101-008-02-04
Et.2 PSUNI V1R2
P-101-006-02-01
N.A.
Et.2 PRI V2R1
Et.2 PRI V2R1
P-101-019-02-03
P-101-024-01-02
P-101-024-01-02
P-101-023-01-01
P-101-023-01-01
P-101-023-01-01
P-101-018-01-02
P-101-018-01-02
P-101-023-01-01
P-101-023-01-01
P-101-023-01-01
P-101-023-01-01
P-101-023-01-01
P-101-023-01-01
P-101-023-01-01
P-101-023-01-01
P-101-014-02-01
P-101-016-02-01
P-101-016-02-01
Software Version
V 10 R 7
V5R6
N.A.
N.A.
V2R2
V3R7
V3R7
V 4 R 15
V1R4
V1R4
V3R1
V3R1
V3R1
V 1 R 11
V 1 R 11
V3R5
V3R5
V3R5
V3R5
V3R5
V3R4
V3R4
V3R4
V2R1
V4R2
V4R2
2547
SLT Cards
Hardware Version
P-100-002-01-07
P-100-003-01-03
P-100-015-01-01
P-100-021-01-01
P-100-021-01-01
P-100-019-01-03
P-100-008-01-03
P-100-020-02-02
P-100-020-02-02
P-100-007-02-01
P-100-010-02-01
P-100-019-01-03
P-100-017-01-03
P-100-009-01-03
P-100-013-01-04
P-100-017-01-03
P-100-017-01-03
P-100-017-01-03
P-100-019-01-03
P-100-019-01-03
P-100-019-01-03
P-100-017-01-03
P-100-017-01-03
P-100-019-01-03
Software Version
V 10 R 7
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
V1R4
V1R4
N.A
N. A.
N. A.
N. A.
N. A.
V 4 R 15
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
CO Cards
VMS Card
VoIP Cards
2548
Hardware Version
P-138-006-01-02
P-138-006-01-02
P-138-001-01-03
P-138-007-01-01
P-138-010-02-02
P-138-003-01-02
P-138-003-01-02
P-138-003-01-02
P-138-008-01-02
P-138-008-01-02
P-138-002-01-02
P-138-003-01-02
P-138-003-01-02
P-138-009-01-01
P-138-003-01-02
P-138-003-01-02
P-138-003-01-02
P-138-003-01-02
P-138-003-01-02
P-138-004-01-02
P-138-013-01-02
P-138-013-01-02
Software Version
V 10 R 7
V 10 R 7
V 10 R 7
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
N. A.
V 4 R 15
V1R4
V1R4
2549
Packing List
Verify contents of the package shipped to you with the contents listed below. If any of the items is missing or
damaged, contact your Dealer/Reseller.
ETERNITY ME 16S
Sr. No
1
Item Name
Qty
ETERNITY ME16SAC IN a
Warranty Cards
User Card
10
Quick Start
11
12
13
14
Mounting Templates
a. Factory fitted with the Power Card, Master Card and Switch Card.
Sr. No
2550
Item Name
Qty
ETERNITY ME16SDC
Sr. No
Item Name
Qty
Warranty Cards
User Card
10
Quick Start
11
12
13
14
Mounting Templates
ETERNITY ME 10S
Sr. No
Item Name
Qty
ETERNITY ME10SAC IN
Warranty Cards
User Card
10
Quick Start
11
12
13
14
Mounting Templates
Sr. No
Item Name
Qty
ETERNITY ME10SDC
2551
Sr. No
Item Name
Qty
Warranty Cards
User Card
10
Quick Start
11
12
13
14
Mounting Templates
If your model is ETERNITY 10SR with the Redundancy Option, there will be 2 Power Cards, 2 Master
Cards and 2 Switch Cards factory fitted.
ETERNITY ME Cards
ETERNITY ME PS48VDC -200W (Power Supply)
Sr. No.
Item
Quantity
1.
ETERNITY ME CARD
PS48VDC-200 Watts
2.
Item
Quantity
1.
ETERNITY ME CARD
PS48VDC-500 Watts
2.
Item
Quantity
1.
2.
2552
ETERNITY ME Switch
Sr. No.
Item
Quantity
1.
2.
ETERNITY ME Master
Sr. No.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
2553
Item Name
Qty
ETERNITY ME CARD
DKP16
Item Name
Qty
Item Name
Qty
Item Name
Qty
Item Name
Qty
2554
Item Name
Qty
ETERNITY ME CARD
CO8+SLT24
Item Name
Qty
Mini jumper
16
Item Name
Qty
Mini jumper
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
2555
Item
Quantity
1.
2.
Item
Quantity
1.
ETERNITY ME Card
VoIP16
2.
Item
Quantity
1.
ETERNITY ME Card
VoIP24
2.
Item
Quantity
1.
ETERNITY ME Card
VoIP32
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
2556
Item
Quantity
2.
3.
User Card
a. Factory fitted.
Item
Quantity
1.
2.
ETERNITY GE
Sr. No.
Item
Quantity
ETERNITY GEa
3.
4.
Mounting Template
5.
User Card
6.
Quick Start
8.
ETERNITY3S
Sr.No.
Item Name
Qty
ETERNITY GE3SAC IN
2557
Sr.No.
Item Name
Qty
Warranty Cards
Users Card
Quick Start
Sr.No.
Item Name
Qty
ETERNITY GE3SDC
Warranty Cards
Users Card
Quick Start
ETERNITY GE6S
Sr.No.
2558
Item Name
Qty
ETERNITY GE6SAC IN
Warranty Cards
User Card
Quick Start
10
11
Mounting Templates
Sr.No.
Item Name
Qty
ETERNITY GE6SDC
Warranty Cards
User Card
Quick Start
10
11
Mounting Templates
ETERNITY GE12S
Sr.No.
Item Name
Qty
ETERNITY GE12SAC IN
Side Clamp
Warranty Cards
User Card
10
Quick Start
11
12
Mounting Templates
13
Sr.No.
Item Name
Qty
ETERNITY GE12SDC
2559
Sr.No.
Item Name
Qty
Side Clamp
Warranty Cards
10
User Card
11
Quick Start
12
13
Mounting Templates
ETERNITY GE Cards
ETERNITY GE Card PS48VDC
Sr. No.
Item
Quantity
1.
ETERNITY GE CARD
PS48VDC
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
2560
Item
Quantity
1.
2.
Item
Quantity
1.
ETERNITY GE Card
SLT16
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
2561
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
2562
Item
ETERNITY GE Card CO16
Quantity
1
Sr. No.
2.
Item
Quantity
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
ETERNITY GE Card
VoIP16
Quantity
1
2563
Sr. No.
2.
Item
Cable with RJ45
connectors on both ends
Quantity
1
Item
Quantity
1.
ETERNITY GE Card
VoIP24
2.
Item
Quantity
1.
ETERNITY GE Card
VoIP32
2.
Item
Quantity
1.
2.
Item
Quantity
2.
3.
User Card
a. Factory fitted.
Item
Quantity
1.
2.
2564
ETERNITY PE
Sr. No.
Item
Quantity
ETERNITY PEa
4.
5.
Mounting Template
6.
User Card
7.
Quick Start
9.
ETERNITY PE3SP
Sr.No.
Item Name
Qty
ETERNITY PE 3SP IN
Quick Start
User Card
Warranty Card
10
Mounting Template
ETERNITY 3SS
Sr.No.
Item Name
Qty
ETERNITY PE 3SS IN
2565
Sr.No.
Item Name
Qty
Quick Start
User Card
Warranty Card
10
Mounting Template
ETERNITY6SP
Sr.No.
Item Name
Qty
ETERNITY PE 6SP IN
Quick Start
User Card
Warranty Card
10
Mounting Template
11
12
Side Clamp
ETERNITY PE Cards
ETERNITY PE Card 3SS CPU
Sr. No.
1.
2566
Item
ETERNITY PE Card 3SS CPU
Quantity
1
Item
Quantity
Item
Quantity
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
ETERNITY PE Card CO8
Quantity
1
2567
Sr. No.
2.
Item
Quantity
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
Item Name
Qty
Mini jumper
Item
Quantity
1.
2.
2568
Item
ETERNITY PE Card GSM4
Quantity
1
Sr. No.
2.
Item
Quantity
Item
Quantity
1.
2.
Item
Quantity
1.
2.
Item
Quantity
1.
ETERNITY PE Card
VoIP16
2.
Item
Quantity
2.
3.
User Card
a. Factory fitted.
Item
Quantity
1.
2.
2569
ETERNITY PE PSUNI
Sr. No.
Item
Quantity
1.
2.
ETERNITY MEX12S
Sr. No
Item Name
Qty
Side Clamps
Screw Grips
10
Mounting Template
11
ETERNITY CD
12
13
Item
Quantity
1.
2.
Item
Quantity
1.
CPU Card
2.
2570
Sr. No.
3.
Item
Quantity
Cable with RJ45 connector at one end and the free loose wires on the
other end. Length 2 m
Item
Quantity
1.
E1FO Card
2.
1 (There are 2
connectors)
3.
1 (there are 2
connectors)
Item
Quantity
1.
BRU Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End, Lenght 2 meter
Item
DATA Card
Cable (Non Shielded) with RJ45 Connectors at Both End, Length 2000 mm
Quantity
1
4 (for Data 8
card and
Data 4 card)
Item
Quantity
1.
DATA Card
2.
Cable (Non Shielded) with RJ45 Connectors at Both End, Length 2000 mm
Item
Quantity
1.
COMBO Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End L= 2 meter
2571
Item
Quantity
1.
SLT Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End L= 2 meter
Item
Quantity
1.
CO4+DKP4+SLT8 Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End L= 2 meter
Item
Quantity
1.
CO8+SLT8 Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End L= 2 meter
Item
Quantity
1.
DKP8+SLT8 Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End L= 2 meter
Item
Quantity
1.
DKP Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End L= 2 meter
Item
Quantity
1.
CO Card
2.
Cable with 50 Pin Centronic Male Connector at One End and Stripped
Tinned Loose Wires on the Other End L= 2 meter
2572
Item
Quantity
1.
GSM Card
2.
3.
Item
Quantity
1.
GSM Card
2.
3.
Item
Quantity
1.
VMS Card
2.
3.
Item
Quantity
1.
VMS Card
2.
Item
Quantity
1.
VMS Card
2.
Item
Quantity
1.
VMS Card
2.
2573
EON48
Sr. No.
Item
Quantity
1.
EON48 Unit
2.
Handset
3.
4.
5.
Foot Stand
6.
7.
EON310
Sr. No.
Item
Quantity
1.
EON310 Unit
2.
Handset
3.
4.
Foot Stand
5.
6.
SPARSH VP248
Sr. No.
Item
Quantity
1.
2.
Handset
3.
4.
5.
Foot Stand
6.
7.
8.
9.
10.
2574
SPARSH VP330
Sr. No.
Item
Quantity
1.
2.
Handset
3.
4.
5.
Foot Stand
6.
7.
8.
DSS64
Sr. No.
Item
Quantity
1.
DSS64 Console
2.
Cable RJ45
DSS72
Sr. No.
Item
Quantity
1.
DSS72 Console
2.
Cable RJ45
3.
Foot stand
EONSOFT
Sr. No.
Item
Quantity
1.
EONSOFT Dongle
2.
3.
Communication Cable
4.
Screw m7/30
5.
Screw Grip
6.
Mounting Template
7.
EONSOFT CD
2575
EON74
Sr. No.
Item
Quantity
1.
EON74 Unit
2.
Handsets
3.
Foot Stand
4.
Spring Cords
5.
6.
PPM4
Sr. No.
Item
Quantity
1.
Matrix PPM4
2.
3.
4.
Mounting Template
2576
P
A.C. Input
Vs
Vc
Z1
N
Enclosure of the Gadget
Z2
Earth
CO Line
CO Line
Lightning
Protectors
System
Protective Earth
Terminal
Telecom Earth
2577
G.1 PIPE
4 Inch
BUS BAR
2578
Get a copper plate of size 1.5 feet x 1.5 feet x 0.25 feet.
Connect a copper strip of size 1-inch wide, 3 mm thick and 6 feet length at the center of the copper plate
by welding or nuts and bolts.
Insert a G.I. pipe onto the copper strip till it reaches the copper plate.
Place this set up into the pit. Make sure that at least 4 inches of the G.I. pipe is above the ground level.
Fill the bottom of the pit with a 1-inch layer of charcoal and salt in the proportion of 3:1 (3 parts charcoal, 1
part salt) and then cover with the soil.
Connect a bare 14 SWG copper wire (double) on the top of the copper strip and run it to the exchange
room and connect it on the bus bar.
The Bus bar is a copper strip, 4 inches long with 6 screws and nuts mounted on it. It has to be fixed on the
wall in the exchange room.
The earth wire of the Primary Protection Modules (PPM) should be connected to this Bus bar.
Follow the instructions given below to connect the PBX to the Earth. The instructions are given with reference to the
above diagram:
1. Loosen the screw.
2. Insert an earthing wire into the lug.
3. Crimp the earth wire with an appropriate tool.
4. Tighten the screw.
5. Connect the earthing wire to the earth.
Make sure you use an earthing wire that has a conductor with a cross-sectional area greater than 1.0
mm2 or AWG less than or equal to 16 AWG with Green and Yellow insulation.
2579
2580
Make sure you comply with all applicable laws, regulations and guidelines.
Proper earthing is very important to protect the PBX from external noise and to reduce the risk of
electrocution in the event of a lightning strike.
The AC cable's earthing pin may not be enough to protect the PBX from external noise and lightning
strikes. A permanent connection must be made between earth and the earth terminal of the main unit.
The SE password is stored in the Master Card/CPU Card. If you forget the SE password, the only way
to restore the default SE password is to change the Jumper settings of the Master Card/CPU card.
You are advised to record and store the SE password at a safe place, where it can be accessed by you
(the System Engineer) to avoid the inconvenience of restoring the default SE password.
Enter Current Password. If you have not changed the default SE password 1234, enter this code.
2581
ETERNITY ME
ETERNITY GE
ETERNITY PE
The default SE password will be restored to 1234. You can now enter the programming mode by dialing 1#911234 (the default password). You can also change the SE password again using Jeeves or by dialing a
command as described above.
2582
If you change the default SE password (1234) again after you have reset SE Password (by changing the
Jumper on the Master Card/CPU card to AB position), the system will not store the new password, until
you change the Jumper back to the default BC position. So, make sure that you have recorded the new SE
password in a safe place, from where it can be retrieved. Change the Jumper back to the default BC
position.
:
To issue a new SA Password using a Telephone,
2583
User Password
Extension Users can secure their respective stations/extensions from unauthorized use with a password unique to
each extension. The User password too is a combination of any four digits, from 0 to 9. The default User Password
is 1111, which each user can change from their respective extensions. Refer the topic User Absent/Present to
know more.
2584
Extension users can also set their status as 'Absent' or 'Present' from their respective extension
phones. Refer User Absent/Present.
DKP extension users can also lock the keypad of their phones from the DKP Phone Menu. Refer
Digital Key Phone-Operation for instructions on navigating the phone menu.
It is also possible to lock/unlock the DKP keypad and set the user extension status as 'Absent'/'Present'
from a remote location using Direct Inward System Access (DISA).
2585
VMS Prompts
No.
Filename
Prompts/Responses
000
Extra0.wav
Blank
001
Num0.wav
Zero
002
Num1.wav
One
003
Num2.wav
Two
004
Num3.wav
Three
005
Num4.wav
Four
006
Num5.wav
Five
007
Num6.wav
Six
008
Num7.wav
Seven
009
Num8.wav
Eight
010
Num9.wav
Nine
011
Num10.wav
Ten
012
Num11.wav
Eleven
013
Num12.wav
Twelve
014
Num13.wav
Thirteen
015
Num14.wav
Fourteen
016
Num15.wav
Fifteen
017
Num16.wav
Sixteen
018
Num17.wav
Seventeen
019
Num18.wav
Eighteen
020
Num19.wav
Nineteen
021
Num20.wav
Twenty
022
Num21.wav
Twenty One
023
Num22.wav
Twenty Two
024
Num23.wav
Twenty Three
025
Num24.wav
Twenty Four
026
Num25.wav
Twenty Five
027
Num26.wav
Twenty Six
028
Num27.wav
Twenty Seven
029
Num28.wav
Twenty Eight
030
Num29.wav
Twenty Nine
031
Num30.wav
Thirty
032
Num31.wav
Thirty One
2586
No.
Filename
Prompts/Responses
033
Num40.wav
Forty
034
Num50.wav
Fifty
035
Num60.wav
Sixty
036
Num70.wav
Seventy
037
Num80.wav
Eighty
038
Num90.wav
Ninety
039
Num100.wav
Hundred
040
Month1.wav
January
041
Month2.wav
February
042
Month3.wav
March
043
Month4.wav
April
044
Month5.wav
May
045
Month6.wav
June
046
Month7.wav
July
047
Month8.wav
August
048
Month9.wav
September
049
Month10.wav
October
050
Month11.wav
November
051
Month12.wav
December
052
N.wav
053
Invalno.wav
054
Extra1.wav
Blank
055
Invalpwd.wav
Invalid password
056
PreName.wav
"More than one match found. Matching names will be played one by one.
Press '1' after the name you wish to select., To skip to the next name
press 2, to go to home position press 0".
057
Empty.wav
058
Plscont.wav
Please continue
059
DialAgn.wav
060
Noextn.wav
061
Recmsg.wav
Record your Message after the beep and press any digit to end
062
PlayMsg.wav
063
RecAgain.wav
064
Hash.wav
Hash
065
RecdCncl.wav
2587
No.
Filename
Prompts/Responses
066
Chkinwel.wav
It is our pleasure receiving you. We will do our best to make your stay
comfortable
067
Extstamp.wav
Extension Number
068
Extra4.wav
Blank
069
Extra5.wav
Blank
070
Entextn.wav
071
NoDISA.wav
072
NoDISAStn.wav
073
EntPwd.wav
074
ConvBetn.wav
075
And.wav
and
076
Callnum.wav
077
DeptOrStn.wav
078
MmmmCustmized.wav
079
Extra6.wav
Blank
080
Extra7.wav
Blank
081
Extra8.wav
Blank
082
Invalcmd.wav
083
Okcmd.wav
084
Extra9.wav
Blank
085
Extra10.wav
Blank
086
Num1000.wav
Thousand
087
LeavemsgI.wav
088
MB80Full.wav
Your Mailbox is 80% Full. Please Delete old messages of your mailbox.
089
MBFull.wav
090
Whichbox.wav
091
DialName.wav
092
Extra13.wav
Blank
093
Xfrornot.wav
094
MatchFor.wav
095
PlsHold.wav
Please hold
096
Xfrcalto.wav
097
Opr.wav
Operator
098
XfrCall.wav
2588
No.
Filename
Prompts/Responses
099
Noreply.wav
100
Extbusy.wav
101
Recname.wav
Record your name after the beep and press any digit to end
102
Callfrom.wav
103
Takecall.wav
Dial '1' to take the call, '2' and disconnect if you do not want to take the call
104
Noattnd.wav
105
LeavemsgG.wav
Dial '1' to leave a message, '2' to dial an extension, Dial '3' to dial by name,
Dial '4' to go to operator, Dial '0' to go to home position, Dial '#' to disconnect.
106
Mblockd.wav
107
Edestopn.wav
108
Thankyou.wav
109
LeavemsgE.wav
Enter the extension number for which you wish to leave message.
110
LeavemsgH.wav
Enter the Room number or the Extension number for which you wish to leave
message
111
NoMailbox.wav
112
Checkout.wav
113
XferMailbox.wav
114
Match.wav
115
Msgfail.wav
116
Msgread.wav
117
PerMB.wav
118
DeptMB.wav
119
GraphTimeout.wav
120
Enterpwd.wav
121
Youhave.wav
You have
122
Newmsg.wav
New Message
123
Newmsgs.wav
New Messages
124
Nonewmsg.wav
125
Mmmm.wav
To listen to new message, press '1', to listen to old message press '2', to send
a message press '3', to change your mailbox settings press '4', to go to home
position press '0'
126
Msgrecon.wav
127
Mmsmo.wav
To replay the message press '1', for message details press '2', to delete the
message press '3', to play the next message press '4', to forward the message
press '5', to save the message as new press '6', to go to previous menu press
'0'
128
Fwdhow.wav
To forward the message without comment press '1', to forward the message
with comment press '2', to go to previous menu press '0'
2589
No.
Filename
Prompts/Responses
129
MsgdestI.wav
130
Askcfrm.wav
To request read receipt press '1', to ignore read receipt press '2'.
131
Pending.wav
132
Msgsent.wav
133
Nooldmsg.wav
134
PreEntry.wav
More than one match found for the extension. Names will be played one by
one. Press '1' after the name to make selection, to skip to next name press '2',
to go to home position press '0'
135
Chgmbset.wav
For Mailbox Name, Press '1', For message redirection, Press '2', To delete
all old messages of your mailbox, press '3', For Mailbox Greetings, press
4, For Assistance Number Programming, press 5 For Personal Number
Programming, press 6, To go to previous menu press '0.
136
GrtChange.wav
Your Mailbox Greeting has been changed. To listen to your Mailbox Greeting,
please select desired time zone.
137
MbGrtZone.wav
For Working Hours Greetings, press 1, For Break Hours Greetings, press 2,
For Non-working Hours Greetings, press 3, To go to previous menu, press 0.
138
MbGrtPlay.wav
139
Extra17.wav
Blank
140
Extra18.wav
Blank
141
Extra19.wav
Blank
142
Extra20.wav
Blank
143
Extra21.wav
Blank
144
Extra22.wav
Blank
145
Extra23.wav
Blank
146
Extra24.wav
Blank
147
MsgNtfy.wav
Message notification
148
Extra25.wav
Blank
149
Extra26.wav
Blank
150
MbMsgRd.wav
To set message redirection press '1', to cancel message redirection press '2',
to go to previous menu press '0'
151
MsgRdHdl.wav
152
Mbfulstg.wav
"To refuse new messages press '1', to deliver new message to general
mailbox press '2', to overwrite old messages press '3', to go to previous
menu press '0'".
153
At.wav
At
154
MsgreadBy.wav
155
MsgfailTo.wav
156
MsgDelCn.wav
2590
No.
Filename
Prompts/Responses
157
MsgSavCn.wav
158
MailboxName.wav
To record Name, press '1'. To play Name, press '2'. To go to previous menu,
press '0'
159
HoldMusic.wav
<Hold Music>
160
Extra32.wav
Blank
161
Beep.wav
<Beep>
162
Extra28.wav
Blank
163
RecMsgStopCode.wav
164
RecMsgEnd.wav
to end.
165
Pound.wav
Pound
166
RecNameStopCode.wav
167
DelAllCnf.wav
You are about to delete all old messages of your mailbox. To proceed, press 1,
to cancel, press 2.
168
DelAllDone.wav
169
LeavemsgU.wav
170
MsgdestU.wav
171
EntertimeI.wav
Enter the time, HH:MM in Twenty four hour format. To cancel all alarms, press
Pound (#).
172
EntertimeU.wav
Enter the time, HH:MM in Twelve hour format. To cancel all Alarm, press
Pound (#). For AM, press 1. For PM, press 2.
173
WakeupCancel.wav
174
SetOnceDaily.wav
175
WakeupVeri.wav
176
DailyWakeupVeri.wav
177
WakeupSet.wav
178
DailyWakeupSet.wav
179
Am.wav
A.M.
180
Pm.wav
P.M.
181
AlarmConf.wav
182
Thankservice.wav
183
NoInput.wav
184
InvalidInput.wav
185
AlarmDateI.wav
Enter the Date in DD MM YYYY format. To Cancel all reminders, press Pound
(#). For example, to enter date 17th March 2008, Dial One Seven Zero Three
Two Zero Zero Eight
2591
No.
Filename
Prompts/Responses
186
AlarmDateU.wav
Enter the Date in DD MM YYYY format. To Cancel all reminders, press Pound
(#). For example, to enter date March 17th 2008, Dial Zero Three One Seven
Two Zero Zero Eight
187
ReminderCancel.wav
188
ReminderVeri.wav
189
ReminderSet.wav
190
Alarmnoset.wav
Sorry! Your Wake Up Alarm cannot be set. Please call Operator for further
assistance
191
Remindernoset.wav
Sorry! Your Reminder cannot be set. Please call Operator for further
assistance
192
RemoteExt.wav
Enter the Extension number for which you have to set or cancel Wake Up
Alarm
193
RemoteExtH.wav
Enter the Room number for which you have to set or cancel Wake Up Alarm
194
Extra29.wav
Blank
195
Extra30.wav
Blank
196
PerWakeupVeri.wav
197
AutoWakeupVeri.wav
198
PerWakeupSet.wav
199
AutoWakeupSet.wav
200
DailyPerWakeupSet.wav
201
DailyAutoWakeupVeri.wav
202
DailyPerWakeupVeri.wav
203
DailyAutoWakeupSet.wav
204
PerReminderVeri.wav
205
AutoReminderVeri.wav
206
PerReminderSet.wav
207
AutoReminderSet.wav
208
Alarmmode.wav
209
Alarmnocancel.wav
210
Remindernocancel.wav
211
WakeUpgreeting.wav
212
DailyWakeUpgreeting.wav
213
Remindergreeting.wav
214
SWakeUpgreeting.wav
This is your Wake up Call. For Acknowledge, Please Press '0'. Music of 5
seconds
215
SDailyWakeUpgreeting.w
av
This is your Daily Wake up Call. For Acknowledge, Please Press '0'. Music of 5
seconds.
2592
No.
Filename
Prompts/Responses
216
SRemindergreeting.wav
This is your Reminder call. For Acknowledge, Please Press '0'. Music of 5
seconds
217
Acknowledge.wav
218
Entertime2I.wav
219
Entertime2U.wav
Enter the Time, HH MM in Twelve hour format. For AM, Press 1. For PM,
Press 2.
220
RemoteExtRem.wav
Enter the Extension number for which you have to set or cancel Reminder.
221
RemoteExtHRem.wav
Enter the Room number for which you have to set or cancel Reminder
222
RemAcknowledge.wav
223
Extra31.wav
Blank
224
MbGrt.wav
For Personal greetings, press '1', For Conditional greetings, press '2',
To go to previous menu, press '0'.
225
MbGrtCond.wav
For Busy, press '1', For No Reply, press '2', For Unconditional, press '3',
To go to previous menu, press '0'.
226
MsgPhoneI.wav
227
MsgPhoneU.wav
228
Extra36.wav
Blank
229
Extra37.wav
Blank
230
Extra38.wav
Blank
2593
Features at a Glance
Abbreviated Dialing
Personal/Global Abbreviated Dialing
8-Location Code
1071-Location Code-Number-#*-TAC
Account Code
Account Code by Number
1058-Account Code
1059-Account Name
Alarms
Once Only Alarm
161-Hours-Minutes-1
Daily Alarms
161-Hours-Minutes-2
161-#
102
Auto Redial
Auto Redial
17
1070
1174
Background Music
Enable/Disable Background Music
1099
Barge-In
Barge-In
2594
1075
Call Chaining
Call Chaining
1050
Call Forward
Call Forward-All Calls to Another Station
131-Station/Department Group/VMS
132-Station/Department Group/VMS
133-Station/Department Group/VMS
134-Station/Department Group/VMS
1361
1360
130
1175-1-1-Destination Number
1175-1-2-Destination Number
1175-1-3-Destination Number
1175-1-4-Destination Number
1175-1-5-1
1175-1-5-0
1175-1-0
1175-2-1-Destination Number
1175-2-2-Destination Number
1175-2-3-Destination Number
1175-2-4-Destination Number
1175-2-5-1
2595
1175-2-5-0
1175-2-0
1175-3-1-Destination Number
1175-3-2-Destination Number
1175-3-3-Destination Number
1175-3-4-Destination Number
1175-3-5-1
1175-3-5-0
1175-3-0
1175-0
Call Hold
Put the caller on Hold
Flash
Flash
Call Park
To Park a Call
115-Orbit Number
116-Orbit Number
Call Pick Up
Call Pick Up-General
12-Station
2596
Call Toggle
Call Toggle (Toggle)
Call Transfer
Call Transfer to Station
Flash-1078-Station
103
1051
Conference 3-Party
Conference 3-Party
Flash-*3
Conference Dial-In
Schedule a Conference
Initiate a Conference
Cancel a Conference
*19-0
Conference Multiparty
To Temporarily Leave / Rejoin a Conference
191
Terminate Conference
190
Conversation Recording
Conversation Recording
Flash-1095
2597
Department Call
Department Call
Do Not Disturb
Set Do Not Disturb with Text Message
18-0
DND Override
Door Phone
Set Door Phone as Scheduled Mode
1173
Dynamic Lock
Set Dynamic Timer
142-User Password-Minutes
141-User Password-Level
Emergency Conference
Make Emergency Conference
Emergency Dialing
Dial Emergency Number
Flashing on Trunks
Flashing on Trunks
2598
Flash-*
Floor Service
To access Floor Service
38
Follow Me
Set Call Follow Me
135-Station-User Password
130
Forced Answer
Forced Answer
#*
Hot Desking
Set Hot Desk
Hotline
Set Hotline
151-Station
154-Seconds
Cancel Hotline/HOD
150
Interrupt Request
Interrupt Request
1092
1094-1
To Deactivate LCS
1094-0
2599
1098-Destination Station
Maid-In
Maid Status from a Room
1054-Code
Meet Me Paging
Meet Me Paging-Caller
Message Wait
Message Wait Set/Cancel
1076-Station-Code
1077
Mini Bar
To check the utilities of the room
1056-Item Number-Quantity
Mute
Mute
1052
0/5/6-OGTBG Index
Operator
Call to Operator
Paging
Paging
Presence
Publishing Presence by extension user
104-Password-Index No.
1097-Extension Number
2600
*37
*38
Raid
Raid
RCOC
RCOC in DISA Mode
**
on dial tone
Reminder
Set Reminder
162-DD-MM-YYYY-HH-MM
Cancel Reminder
162-#
Room Monitor
Room Monitor
1073-Station
1057
1#92-SA Password
Exit SA Mode
1#92
SA Command
1072
1#91-SE Password
Exit SE Mode
00
Trunk Reservation
Reserve a Trunk
102
2601
User Absent/Present
User Absent
104-User Password-0
User Present
104-User Password-1
User Password
Change User Password
Voice Help
Voice Help
1090
To Walk-Out
111-0
2602
2603
2604
SA Commands
Call Phases
Feature
Number
Access
Code
Dial
Check-In
001
1072-901
Check-Out
002
1072-902
Guest Name
003
1072-903
Guest Group
004
1072-904
Guest-In/Out
005
1072-905
Guest Title
006
1072-906
007
1072-907
008
1072-908
009
1072-909
Room Shift
010
1072-910
011
1072-911
012
1072-912
013
1072-913
014
1072-914
Set DND-Remote
015
1072-001
016
1072-002
017
1072-003
018
1072-004
Assign/De-assign Mailbox to a
Station/Department Group Remote
019
1072-005
020
1072-006
021
1072-007
022
1072-008
023
1072-009
024
1072-010
025
1072-011
Feature Name
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
2605
Call Phases
Feature
Number
Access
Code
Dial
026
1072-012
027
1072-013
028
1072-014
Change SA password
029
1072-015
030
1072-016
031
1072-017
032
1072-018
033
1072-019
034
1072-020
035
1072-021
036
1072-022
037
1072-023
038
1072-024
039
1072-315
040
1072-026
041
1072-027
042
1072-028
043
1072-029
044
1072-101
045
1072-102
046
1072-103
047
1072-104
048
1072-105
049
1072-106
050
1072-107
051
1072-108
052
1072-109
Feature Name
2606
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
Call Phases
Feature
Number
Access
Code
Dial
053
1072-110
054
1072-111
055
1072-112
056
1072-113
057
1072-114
058
1072-115
059
1072-120
060
1072-121
061
1072-122
062
1072-123
063
1072-124
064
1072-125
065
1072-131
066
1072-132
067
1072-133
068
1072-136
069
1072-137
070
1072-138
071
1072-141
072
1072-142
073
1072-143
Feature Name
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
2607
Call Phases
Feature
Number
Access
Code
Dial
074
1072-144
075
1072-145
076
1072-150
077
1072-151
078
1072-152
079
1072-153
080
1072-154
081
1072-155
082
1072-156
083
1072-157
084
1072-158
085
1072-159
086
1072-160
087
1072-161
088
1072-162
089
1072-163
090
1072-164
091
1072-165
092
1072-166
093
1072-167
094
1072-168
Feature Name
2608
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
Call Phases
Feature
Number
Access
Code
Dial
095
1072-169
096
1072-170
097
1072-171
098
1072-172
099
1072-173
100
1072-174
101
1072-175
102
1072-180
103
1072-030
104
1072-031
105
1072-032
106
1072-181
107
1072-176
108
1072-177
109
1072-178
110
1072-915
111
1072-916
112
1072-917
Remote Reminder
113
1072-033
114
1072-034
115
1072-035
116
1072-314
117
1072-316
118
1072-036
Feature Name
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
2609
Call Phases
Feature
Number
Access
Code
Dial
119
1072-037
Enable/Disable Scheduled
Reminder Report
120
1072-038
121
1072-039
Request Database
Synchronisation to PMS
122
1072-040
123
1072-041
124
1072-042
Enable/Disable Scheduled
Change of Room Clean Status
125
1072-043
126
1072-044
127
1072-317
128
1072-045
129
1072-191
130
1072-920
131
1072-183
132
1072-184
133
1072-185
134
1072-186
135
1072-187
136
1072-188
137
1072-189
138
1072-192
139
1072-193
140
1072-194
141
1072-195
Feature Name
2610
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
Call Phases
Feature
Number
Access
Code
Dial
142
1072-196
143
1072-197
144
1072-198
145
1072-199
146
1072-200
147
1072-201
148
1072-202
149
1072-203
150
1072-204
151
1072-205
152
1072-206
153
1072-207
154
1072-208
155
1072-209
156
1072-210
157
1072-211
158
1072-212
159
1072-213
160
1072-214
161
1072-215
162
1072-216
163
1072-217
164
1072-218
165
1072-219
166
1072-220
167
1072-221
Feature Name
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
2611
Call Phases
Feature
Number
Access
Code
Dial
168
1072-046
169
1072-222
170
1072-223
171
1072-116
172
1072-117
173
1072-118
174
1072-047
To Broadcast Message
175
1072-301
176
1072-302
177
1072-303
178
1072-304
179
1072-305
180
1072-306
181
1072-307
182
1072-308
183
1072-309
184
1072-310
185
1072-311
186
1072-312
187
1072-313
Feature Name
2612
Routing
Blocked
Placed
Matured
2-Way
Matured
3-way
System Commands
Abbreviated Dialing
Program telephone number in a personal directory
1905-1-SLT-Personal Directory
1905-1-SLT-00
1906-1-DKP-Personal Directory
1906-1-DKP-00
1907-1-ISDN-Personal Directory
1907-1-ISDN-00
To clear a PM Group
1901-1-PM Group
1801-Location Code-Number
1801-Location Code-#*
1802-Location Code-Name
1802-Location Code-#*
1803-1-Location Code-OGTBG
1800-1-Location Code
Access Codes
Program the access code for features
3161-1-Feature Number
Account Codes
Program account name for the account code
2613
AC Impedance Test
To run the AC Impedance Test using the Accurate
Mode
3362
Alarms
Program alarm/reminder ring timer
2201-Seconds
2204-Code
2208-Flag
2209-Flag
To program Macros
To clear a Macro
1810-Macro Index-#*
3115-1-Macro Index
Auto Answer
To set auto answer on DKP
5503-1-SLT-Template Number
2614
5504-1-DKP-Template Number
3801-Seconds
Auto Redial
Program time duration between two trials for low
priority
1704-Seconds
1705-Count
1706-Seconds
1707-Count
1702-Seconds
1703-Seconds
1712-Gadget Number-DOP
1717-Index-Scheduled ON Time
5001-1-DOP-Contact Type
Barge-In
To program Barge-In Timer
3803-Seconds
2615
3310-1-CO-Code
3313-1-CO-Call Back on
3317-1-CO-OGTB Group
8010-1-Mobile-Code
8013-1-Mobile-Call Back on
8037-1-Mobile-OGTB Group
7712-1-SIP-Code
7746-1-SIP-Call Back on
7750-1-SIP-OGTB Group
2616
6249-1-BRI-OGTB Group
6149-1-T1E1-OGTB Group
Call Budget
To program default Call Budget Amount
3301-1-CO-Budget Type
3302-1-CO-Budget Amount
3303-1-CO-Minutes
3305-1-CO-Date
3309-1-CO-Number of Calls
7733-1-SIP-Budget Type
7734-1-SIP-Budget Amount
7735-1-SIP-Minutes
7737-1-SIP-Date
7730-1-SIP-Number of Calls
8019-1-Mobile-Budget Type
8020-1-Mobile-Budget Amount
8021-1-Mobile-Minutes
8033-1-Mobile-Number of Calls
2617
8023-1-Mobile-Date
6214-1-BRI-Budget Type
6215-1-BRI-Budget Amount
6216-1-BRI-Minutes
6205-1-BRI-Number of Calls
6217-1-BRI-Reset Mode
6218-1-BRI-Date
6122-1-T1E1-Budget Type
6123-1-T1E1-Budget Amount
6124-1-T1E1-Minutes
6125-1-T1E1-Number of Calls
6139-1-T1E1-Date
3306-1-CO
7738-1-SIP
8024-1-Mobile
6219-1-BRI
6140-1-T1E1
2603-Service Charge
2604-Percentage
2606
2618
2611
2622-Reverse SE Password
2630-Day-Code
4202-1-CDC Table-Code
4203-1-CDC Table-Code
4204-1-CDC Table-Code
4201-1-CDC Table
Call Hold
To select Default Call Hold Type
5318-Hold Type
2619
3805-Seconds
3812-Minutes
Call Logs
To enable/disable Log Internal Calls in Missed Calls
5361-Code
5362-Code
5363-Code
Call Park
Set call park timer
3809-Minutes
3810-Minutes
Call Pick Up
To assign call pickup group for SLT
3901-1-SLT-00
3902-1-DKP-00
3903-1-ISDN Terminal-00
3501-Region Code
3502-Seconds
3503-Seconds
3504-Seconds
3505-Seconds
3506-Seconds
3509-Seconds
3508-Seconds
5307-Flag
3541-Code
3542-Seconds
2620
Call Taping
To enable/disable beeps during conversation
recording
5332-Code
Call Transfer
To program transfer while ringing timer
3806-Seconds
3807-Seconds
3808-Minutes
5334-Code
5335-Replacement String-#*
5335-#*
1301-1-CoS Group
2621
4101-Index-Telephone Number-#*
4101-Index-#*
4102-Index-Name-#*
4102-Index-#*
4104-1-Index
Clock Synchronization
To program the clock sources
4502-1-Route Index-#*
4503-1-Route Index-OGTBG
4505-1-Route Index-Code
4501-1-Route Index
Communication Ports
To set data transfer rate of a COM port
3201-Port-Speed
3202-Port-Data Bits
3203-Port-Parity
3204-Port-Stop Bits
3205-Port-Flow Control
3206-Port-DSR Sensing
3210-Port
2118-Time
2119
2622
2120-User Number
2102-Code
2101-Code
2116-Connection Type
2110-IP Address
2111-Subnet Mask
2112-Gateway IP Address
2121-Port
2125-Code
2126-DDNS User ID
2130
2132-IP Address
2134- Port
2135 - Interval
2150
2151
2152
2159
2160
2161
2162
2623
Configuring Extensions
To change the default value of a SLT Hardware
Template
5701-1-Template Number
5703-1-SLT-Template Number
5501-1-Template Number
5503-1-SLT-Template Number
5504-1-DKP-Template Number
5507-1-ISDN-Template Number
5505-1-E&M-Template Number
5506-1-T1E1PRI-Template Number
5509-1-BRI-Template Number
5601-1-Template Number
5603-1-SLT-Template Number
5604-1-DKP-Template Number
5607-1-ISDN-Template Number
5605-1-E&M-Template Number
5606-1-T1E1PRI-Template Number
2624
5609-1-BRI-Template Number
1101-SLT-00-00
3101-1-SLT-Access Code-#*
3101-1-SLT-#*
3151-1-SLT
5402-1-SLT-Name
5402-1-SLT-#*
5703-1-SLT-Template Number
5503-1-SLT-Template Number
5603-1-SLT-Template Number
3901-1-SLT-00
1905-1-SLT-Personal Directory
1905-1-SLT-00
3911-1-SLT-Priority
1102-DKP-00-00
3102-1-DKP-Access Code-#*
3102-1-DKP-#*
3152-1-DKP
5403-1-DKP-Name
2625
5403-1-DKP-#*
5504-1-DKP-Template Number
5604-1-DKP-Template Number
1201-1-DKP-Call Capacity
3902-1-DKP-00
1906-1-DKP-Personal Directory
1906-1-DKP-00
3912-1-DKP-Priority
1224-1-DKP-Language
1204-1-DKP-Ringer Mode
1220-1-DKP-Ring Destination
1202-1-DKP-Ring Tune
1203-1-DKP-Ringer Volume
1241-1-DKP-DTMF Generation
2626
1103-DKP-DSS-00-00
7301-1-ISDN Terminal-BRI
7301-1-ISDN Terminal-00
3103-1-ISDN Terminal-#*
3153-1-ISDN Terminal
5409-1-ISDN Terminal-Name-#*
5409-1-ISDN Terminal-#*
1907-1-ISDN Terminal-00
3913-1-ISDN Terminal-Priority
3903-1-ISDN Terminal-00
Configuring Region
To select Region
5301-Region Code
Configuring Operator
To assign a Time Table to an Operator
1602-1-Operator-Time Table
1611-1-Operator-Routing Group
1612-1-Operator-Routing Group
2627
1613-1-Operator-Routing Group
To default an Operator
1601-1-Operator
Configuring Trunks
To change the default value of a CO Hardware
Parameter in a Template
6001-1-Template Number
6003-1-E&M-Template Number
6004-1-T1E1-Template Number
Configuring CO Trunks
To assign Hardware Slot and Port to the CO Port
1104-CO-00-00
5404-1-CO-Name
5404-1-CO-#*
3307-1-CO-Flag
2628
3308-1-CO-Cost Factor
3310-1-CO-Code
3313-1-CO-Call Back on
3317-1-CO-OGTB Group
1108-Mobile-00-00
8000-1-Mobile-Flag
5408-1-Mobile-Name-#*
5408-1-Mobile-#*
8005-1-Mobile-Mode
8030-1-Mobile-Code
8001-1-Mobile-Cost Factor
8010-1-Mobile-Code
8013-1-Mobile-Call Back on
2629
8037-1-Mobile-OGTB Group
8013-1-Mobile-Number List
8029-1-Mobile-Code
8014-1-Mobile-Pause Timer
8015-1-Mobile-DTMF ON Time
8051-1-Mobile-Mode
8052-1-Mobile-Duration
8018-1-Mobile-Category
8017-1-Mobile-DTMF String
8041-1-Mobile-SIP Rx Gain
8042-1-Mobile-SIP Tx Gain
8032-1-Mobile-1
1109-VoIP Port-Slot
1109-VoIP Port-00
7763-1-VoIP Port-Name-#*
2630
7823-1-VoIP Port-Flag
7824-1-VoIP Port-Flag
7829-1-VoIP Port-Flag
7781-1-VoIP Port-Flag
7780-1-VoIP-IP Address
7795-1-VoIP Port-Flag
7796-1-VoIP Port-Timer
7783-1-VoIP Port-Code
7775-1-VoIP Port-Flag
7776-1-VoIP Port-VLAN ID
7771-1-VoIP Port
2631
7772-1-VoIP-SIP Trunk
7701-1-SIP-VoIP Port
7701-1-SIP-00
7702-1-SIP-Code
5410-1-SIP-Name-#*
5410-1-SIP-#*
7704-1-SIP-SIP User ID
7707-1-SIP-Re-Registration Timer
7709-1-SIP-Authentication User ID
7740-1-SIP-Flag
5808-1-SIP-Template Number
7703-1-SIP-Cost Factor
2632
7743-1-SIP-Code
7891-1-SIP-Rx Gain
7892-1-SIP-Tx Gain
7713-1-SIP-DTMF Type
7714-1-SIP-Fax Type
7883-1-SIP-Max. Rate
7884-1-SIP-Packet Period
7746-1-SIP-Flag
7747-1-SIP-Tail Length
7748-1-SIP-Tail Length
7881-1-SIP-Optimization Factor
7882-1-SIP-Minimum Delay
7739-1-SIP-Flag
7744-1-SIP-Flag
7745-1-SIP-Flag
7718-1-SIP-Flag
7742-1-SIP-Code
7711-1-SIP-Code
2633
7729-1-SIP-Digest Authentication
4118-Index-User ID
7731-1-SIP-Default Transport
7712-1-SIP-Code
7746-1-SIP-Call Back on
7750-1-SIP-OGTB Group
7722-1-SIP-IC Reference ID
7723-1-SIP-IC Reference ID
7724-1-SIP-IC Reference ID
7721-1-SIP-OG Reference ID
7720-1-SIP-Pause Timer
7725-1-SIP-DTMF ON Time
7727-1-SIP-Gateway Application
To configure DTMF String for Gateway ApplicationAnswer Signaling on the SIP trunk
5324-1-SIP- Flag
3918-1-SIP Extension-Priority
2634
7872-1-SIP Extension-Code
3108-1-SIP Extension-SIP ID
3157-1-SIP Extension
7874-1-SIP Extension-Authentication ID
3918-1-SIP Extension-Priority
2635
1105-E&M-00-00
3321-1-E&M-Code
5406-1-E&M-Name
5406-1-E&M-#*
6003-1-E&M-Template Number
5505-1-E&M-Template Number
5605-1-E&M-Template Number
3915-1-E&M-Priority
3322-1-E&M-Cost Factor
6801-1-Magneto-Code
3107-1-Magneto-Access Code-#*
3107-1-Magneto-#*
3156-1-Magneto
5411-1-Magneto-Name-#*
5411-1-Magneto-#*
3919-1-Magneto-Priority
5357-Flag
2636
1111-LD-Slot-Port offset
1111-LD-00-00
5416-1-LD-Name-#*
3941-1-LD-Flag
3920-1-LD-Priority
5512-1-LD-SBFT
3942-1-LD-Cost Factor
3943-1-LD-Flag
3944-1-LD-Flag
3945-1-LD-Flag
3946-1-LD-Option
To program the Call Cost Calculation Time ScheduleT1-Start Time for LD trunk port
3947-1-LD-Start Time
To program the Call Cost Calculation Time ScheduleT1-End Time for LD trunk port
3948-1-LD-End Time
To program the Call Cost Calculation Time ScheduleT2-Start Time for LD trunk port
3949-1-LD-Start Time
To program the Call Cost Calculation Time ScheduleT2-End Time for LD trunk port
3950-1-LD-End Time
To program the Call Cost Calculation Time ScheduleT3-Start Time for LD trunk port
To program the Call Cost Calculation Time ScheduleT3-End Time for LD trunk port
3952-1-LD-End Time
To program the Call Cost Calculation Time ScheduleT4-Start Time for LD trunk port
3953-1-LD-Start Time
To program the Call Cost Calculation Time ScheduleT4-End Time for LD trunk port
3954-1-LD-End Time
3109-*
2637
5414-1-Virtual Extension-#*
2152-1-Virtual Extension-#*
3919-1-Virtual Extension-Priority
Configuring LCR
To program Time Zone at a Time Zone index
3401
3411-Number Index-#*
3412-Number Index-CF1-CF2-CF3-CF4
3410
3422-Number Index-#*
3420
3441-Number Index-#*
3442-Number Index-CF1-CF2-CF3-CF4
3440
1404-1-OGTBG-LCR Type
2638
3116-Index-Emergency Number-#*
Conflict Dialing
To program conflict dialing timer
5351-Seconds
Conversation Recording
To enable/disable conversation recording beeps
5332-Code
Customer Name
To program the customer name
5401-Customer Name-#*
5401-#*
4801-Code
1010-DST Mode
1013-DST Type
2639
Default Settings
To load the default parameters
5302-Reverse SE Password
Department Call
To program the destination in the routing group
Digest Authentication
To program SIP-Id in the Digest Authentication Table
4118-Index-User ID
4119-Index-User Password
4901-1-DIP-Code
4902-1-DIP-Code
4901-1-DIP
1221-1-DKP-0
2640
1254-1-DKP-DSS-Key Number-00-000
3111-1-76-Access Code
3161-1-76
2411-Seconds
2412-Seconds
2413-Seconds
2414-Seconds
2415-Seconds
2416-Seconds
2417-Seconds
5338-Code
5336-Code
5337-Code
2420-Seconds
2421-Minutes
Distinctive Rings
To demonstrate the ring
4003-Ring Pattern
3542-Seconds
4001
4002-Event-Ring Pattern
2641
1501
Door Phone
To program access code of Door Phone
3155-1-Door Phone
5413-1-Door Phone-Name-#*
3231-1-Door Phone-DOP
5110-Code
Emergency Dialing
To program emergency numbers
3116-Index-Emergency Number-#*
Flexible Numbers
To program the access code for a SLT
3101-1-SLT-Access Code-#*
3101-1-SLT-#*
2642
3151-1-SLT
3102-1-DKP-Access Code-#*
3102-1-DKP-#*
3152-1-DKP
Floor Service
To program a routing group with member extensions
IC Reference Table
To default the IC Reference Table
Interrupt Request
To set interrupt request timer
3802-Seconds
ISDN BRI
To assign a name to the BRI port
5405-1-BRI-Name
6201-1-BRI-Port Status
6202-1-BRI-SP
6204-1-BRI-Orientation Type
6205-1-BRI-Companding
6207-1-BRI-Idle Code
6208-1-BRI-Timer
6210-1-BRI-DTMF ON Time
2643
6291-1-BRI-Level-Code
6221-1-BRI-Caller TON
6222-1-BRI-Caller NPI
6249-1-BRI-OGTB Group
6231-1-BRI-OG Reference ID
6241-1-BRI-OG Reference ID
6242-1-BRI-OG Reference ID
6232-1-BRI-IC Reference ID
6233-1-BRI-IC Reference ID
6234-1-BRI-IC Reference ID
6235-1-BRI-Channel Count
6236-1-BRI-Channel Count
6237-1-BRI-Channel Count
6239-1-BRI-TEI Value
3153-1-ISDN Terminal
7301-1-ISDN Terminal-BRI
7301-1-ISDN Terminal-00
2644
6225-1-BRI-Layer 1 Mode
To clear a macro
1810-Macro Index-#*
3115-1-Macro Index
3165-1-Macro Index
Logical Partition
To define call permission across and between
Categories
5317-Category-Category-Flag
3811-Seconds
Message Wait
To program Message Wait Ring Count
Music on Hold
To demonstrate music on hold
3551-Code
3552-Code
3553-Code
Number List
To program a number in a Number List
4301-1-List Number
OFF-Hook Alert
To enable/disable OFF-Hook Alert to Operator
5333-Code
2645
OG Reference Table
To program an OG reference table
OG Trunk Bundle
To program the feature in OG trunk bundle
1401-1-OGTBG Number
1403-1-OGTBG Number-Flag
3112-1-OGTBG Index-#*
3162-1-OGTBG Index
Paging
To enable/disable Analog Output Port (AOP) in a Page
Zone
2301-1-Page Zone-Flag
2302-1-Page Zone-Member-DKP
2303-1-Page Zone-Member-Flag
Presence
To enable/disable 'Display Presence Status during Call
on DKP'
5320-Flag
Peer-to-Peer Calling
To program the number string in peer-to-peer table
7801-1-Index-Number String-#*
7801-1-Index-#*
7802-1-Index-Destination Address-#*
7802-1-Index-#*
7803-1-Index-Name-#*
2646
7803-1-Index-#*
Priority
To assign priority to SLT
3911-1-SLT-Priority
3912-1-DKP-Priority
3918-1-SIP Extension-Priority
3914-1-T1E1-Priority
3915-1-E&M-Priority
3916-1-BRI-Priority
3919-1-Magneto-Priority
RCOC
To enable RCOC on a SIP Trunk
7743-1-SIP-Code
6145-1-T1E1-Code
6220-1-BRI-Code
8030-1-Mobile-Code
3521- Minutes
3503-Seconds
1000-Date Format
1001-Date-Month-Year
1002-Time Zone
To set time
1003-Hours-Minutes-Seconds
Reminder
Program alarm/reminder ring timer
2201-Seconds
3701-Flag
2204-Code
Routing Group
To program the destination in the routing group
2647
6501-1-Routing Group
6510-1-Routing Group
5202-Flag
5203-Time Table
5204-TimeZone-Trigger on
5205-TimeZone-DOP
5206-TimeZone-Routing Group
5207-TimeZone-Index-External Number-#*
5211-Number of Attempts
5212-Flag
5213-Routing Group
5201
1101-SLT-00-00
1102-DKP-00-00
2648
1103-DKP-DSS-00-00
1104-CO-00-00
1105-E&M-00-00
1106-BRI-00-00
1107-T1E1-00-00
1108-Mobile-00-00
1110-Magneto-00-00
7811-1-Index-Destination Address
7812-1-Index-Subnet Mask
7813-1-Index-Gateway Address
7814-1-Index
2930-Code
2932-IP Address
2933-IP Port
2830-Code
2832-IP Address
2649
2833-IP Port
2730-Code
2732-IP Address
2733-IP Port
8200-Column Position
8201-Field Length
8202-Alignment
8203-Fill Character
8204-Reset
8205-Column Position
8206-Reset
8207-Column Position
8208-Field Length
8210-Column Position
8211-Field Length
8212-Alignment
8213-Fill Character
8214-Column Position
8215-Format Type
8216-Column Position
8217-Field Length
8218-Alignment
8219-Fill Character
8220-Date Format
8222-Column Position
8223-Field Length
8224-Alignment
8225-Fill Character
8226-Time Format
8227-Column Position
8228-Field Length
8229-Alignment
2650
8230-Fill Character
8231-Duration Unit
8232-Column Position
8233-Field Length
8234-Alignment
8235-Fill Character
8237-Column Position
8238-Field Length
8239-Alignment
8240-Fill Character
8242-Column Position
8243-Field Length
8244-Alignment
8245-Number Format
8246-Column Position
8247-Field Length
8248-Alignment
8249-Number Format
8250-Column Position
8251-Field Length
8252-Alignment
8253-Column Position
8254-Field Length
8255-Alignment
8256
2701-Code
8330-Code
2651
8312-ENQUIRE Signal
8313-ENQUIRE
8316-Start of Packet
8317-End of Packet
8318-BCC Flag
8300
8100-Column Position
8101-Field Length
8102-Alignment
8103-Fill Character
8104-Reset
8105-Column Position
8106-Reset
8174-Starting Character
8107-Column Position
8108-Field Length
8110-Column Position
8111-Field Length
8112-Alignment
8113-Fill Character
2652
8114-Column Position
8115-Format Type
8116-Column Position
8117-Field Length
8118-Alignment
8119-Fill Character
8120-Date Format
8122-Column Position
8123-Field Length
8124-Alignment
8125-Fill Character
8126-Time Format
8127-Column Position
8128-Field Length
8129-Alignment
8130-Fill Character
8131-Duration Unit
8132-Column Position
8133-Field Length
8134-Alignment
8135-Fill Character
8136-Column Position
8137-Field Length
8138-Alignment
8139-Fill Character
8140-Amount Format
8141-Column Position
8142-Field Length
8143-Alignment
8144-Fill Character
8145-Character1.......Character8
2653
8146-Column Position
8147-Field Length
8148-Alignment
8150-Column Position
8151-Field Length
8152-Alignment
8154-Column Position
8155-Field Length
8156-Alignment
8157-Number Format
8158-Column Position
8159-Field Length
8160-Alignment
8161-Fill Character
8165-Code
8166-Column Position
8167-Field Length
8168-Alignment
8169
8321-Index-Country Code-#*
8322-Index-Country Code-#*
8333-Code
8331-Address-Address-Address-Address
8332-Destination IP Port
8334-Listening Port
2931-Code
2934-IP Address
2935-IP Port
2831-Code
2834-IP Address
2654
2835-IP Port
2731-Code
2734-IP Address
2735-IP Port
2901-Storage Flag
2902-Flag
2903-Flag
2904-Flag
2905-Flag
2906-Flag
2907-Seconds
2908-Seconds
2909-Seconds
2915
2801-Storage Flag
2802-Seconds
2815
2701-Storage Flag
2702-Number List
2703-Seconds
2704-Unit
2715
2716-Toggle Flag
2717-Originating Flag
6401-Storage Flag
6402-Port
6404-IP Address
6405-IP Port
6403-Port
6406-IP Address
6407-IP Port
2655
6410
System Debug
To program a port for debug
2103-Port
2104-Value
2181-1-Code
2184-1-Code
2184-2-Code
2194
6451-Flag
6452-Port
6454-IP Address
6455-IP Port
6453-Port
6456-IP Address
6457-IP Port
System Parameters
To assign Station Type to SLT
3921-1-SLT-Station Type
3922-1-DKP-Station Type
2187-Slot No
5305-Reverse SE Password
5303
5304
2656
5321-Code
5319-Language
5322-Type
To monitor a port
7902-Slot-LED Number-Port
5309-Flag
2121-Port
5315-Code
2122
System Security
To change SE password
5306-New SE Password
5310-New SA Password
T1 Maintenance
To enable/disable T1 FDL on a T1E1 port
6164-1-T1E1-T1 FDL
T1 RBS Parameters
To program the T1 line signaling variants for the T1E1
port
6182-1-T1E1-Wink Timer
6185-1-T1E1-Delay Duration
6166-1-T1E1-Code
2657
6167-1-T1E1-Character
6168-1-T1E1-Code
6169-1-T1E1-Character
6163-1-T1E1-Code
T1E1 Trunks
To enable/disable the port
6101-1-T1E1-Port Status
5407-1-T1E1-Name
6108-1-T1E1-Carrier
6103-1-T1E1-Line Coding
6195-1-T1E1-Line Coding
6104-1-T1E1-Framing
6196-1-T1E1-Framing
6105-1-T1E1-Line Type
6151-1-T1E1-Line Type
6108-1-T1E1-Interface Companding
6110-1-T1E1-Mode
6112-1-T1E1-Glare Option
6121-1-T1E1-Category
6113-1-T1E1-Idle Code
6114-1-T1E1-Timer
6109-1-T1E1-Pause Timer
6117-1-T1E1-DTMF ON Time
6106-1-T1E1-Orientation Type
6126-1-T1E1-Source TON
6127-1-T1E1-Source NPI
6128-1-T1E1-Destination TON
6129-1-T1E1-Destination NPI
6130-1-T1E1-Flag
6115-1-T1E1-Flag
6116-1-T1E1-Flag
2658
6131-1-T1E1-OG Reference ID
6132-1-T1E1-IC Reference ID
6133-1-T1E1-IC Reference ID
6134-1-T1E1-IC Reference ID
6102-1-T1E1-Cost Factor
6141-1-T1E1-Loopback
6142-1-T1E1-Code
6143-Port
6144-Port
6154-1-T1E1-Flag
6162-1-T1E1-Code
6171-1-T1E1-Flag
6155-1-T1E1-Flag
2659
7108-1-T1E1-End of DNIS
7118-1-T1E1-ANI Length
7119-1-T1E1-Ask ANI
7124-1-T1E1-Ordinary Subscriber
7125-1-T1E1-Priority Subscriber
7126-1-T1E1-Maintenance Equipment
7127-1-T1E1-Operator
7128-1-T1E1-Pay Phone
7129-1-T1E1-Data Transmission
7130-1-T1E1-Interception Operator
2660
To program congestion
7148-1-T1E1-congestion
7149-1-T1E1-Unallocated Number
7151-1-T1E1-Reject Call
7157-1-T1E1-Congestion
2661
7161-1-T1E1-CD Bits
7162-1-T1E1-Invert Bit A
7163-1-T1E1-Invert Bit B
7164-1-T1E1-Invert Bit C
7165-1-T1E1-Invert Bit D
7169-1-T1E1-Release Timer
6105-1-T1E1-Line Type
6004-1-T1E1-Template Number
7191-1-T1E1-Code
7162-1-T1E1-Invert Bit A
7163-1-T1E1-Invert Bit B
7164-1-T1E1-Invert Bit C
7165-1-T1E1-Invert Bit D
6162-1-T1E1-Code
6119-1-T1E1-Gateway Application-Answer
Signaling Flag
6120-1-T1E1-Gateway Application-Answer
Signaling DTMF String
6149-1-T1E1-OGTB Group
6191-1-T1E1-Level-Code
2662
6191-1-T1E1-1-Code
6191-1-T1E1-2-Code
6192-1-T1E1-1-Code
6192-1-T1E1-2-Code
Time Tables
To program a timetable
1051-1-Time Table
Toll Control
To program Local Numbers Allowed List
4303-Index-Number String-#*
4304-Index-Number String-#*
4311-Reverse SE Password
4305-Index-Number String-#*
4306-Index-Number String-#*
4312-Reverse SE Password
4307-Index-Number String-#*
4308-Index-Number String-#*
4313-Reverse SE Password
4301-1-List Number
5502-1-Template-08-Number List
5502-1-Template-09-Number List
5502-1-Template-11-Number List
5502-1-Template-12-Number List
5502-1-Template-14-Number List
5502-1-Template-15-Number List
2663
Trunk Reservation
To program trunk reservation timer
3804-Minutes
5301-Region Code
5302-Reverse SE Password
5061-Address-Address-Address-Address-#*
5062-Address-Address-Address-Address-#*
5063-Address-Address-Address-Address-#*
5064-Connection Type-#*
5066-PPPoE Password-#*
5069- Address-Address-Address-Address-#*
5081-Code-#*
5082-Code-#*
5096-Code-#*
5086-Access Code-#*
5087-Code-#*
5088-Function-Code-#*
5090-Code-#*
5091-Code-#*
5092-Code-#*
2664
5093-Code-#*
5094-Code-#*
5036-Graph-Node-Node Type-#*
5037-Graph-Node-Digit-Destination Node-#*
5038-Graph-Node-Extension Number-#*
5031-Profile Number-Graph #*
5041-Code-#*
5042-Code-#*
5043-Display Name-#*
5044-Email ID-#*
To program User ID
5045-User ID-#*
To program Password
5046-Password-#*
5050-Reconnection Interval-#*
5711-1-SLT-Flag-#*
5712-1-SLT-Mailbox Size-#*
2665
5714-1-SLT-Delivery Option-#*
5716-1-SLT-Password Option-#*
5718-1-SLT-Profile Number-#*
5719-1-SLT-Abbreviated Name-#*
5722-1-SLT-Notification code-#*
5723-1-SLT-Code-#*
5724-1-SLT-E-mail ID-#*
1271-1-DKP-Flag-#*
1272-1-DKP-Mailbox Size-#*
1274-1-DKP-Delivery Option-#*
1276-1-DKP-Password Option-#*
1278-1-DKP-Profile Number-#*
1279-1-DKP-Abbreviated Name-#*
1282-1-DKP-Notification code-#*
1283-1-DKP-Code-#*
2666
1284-1-DKP-E-mail ID-#*
6261-1-ISDN Terminal-Flag-#*
6273-1-ISDN Terminal-Code-#*
2011-1-Department Group-Flag-#*
2667
2501-Code
To verify a message
2503-Voice Module
2504-Voice Module-Duration
2668
Troubleshooting
All servicing to be undertaken ONLY by qualified service personnel. There are no user serviceable parts
inside the unit.
Always switch off "MAINS" and "BATTERY" marked switches of the system before opening the system and
remove power cable from Mains plug, to avoid risk of electric shock.
Check the Mains Fuse (6Amp Slow Blow, glass fuse provided in AC Mains socket of ETERNITY ME Card
10SAC/16SAC).
Check for loose connection of PT3 connector (connecting AC Mains socket to ETERNITY ME Card
10SAC/16SAC).
If FCBC is used, check battery voltage interfaced with FCBC, it has to be 43-56V.
Dial 130 from 2001 (your call might have been forwarded).
2669
Check wiring.
Check the Time programmed in the PBX. This is a time sensitive feature.
Please check up with your Telephone Company (Service Provider) for CLI facility.
Please check whether the Station where you are checking CLI function is programmed as CLI Phone.
Please check for the polarity. Refer to the figure shown in topic Digital Output Port (DOP).
2670
Please check the connections of the Panic Switch/Sensor to the Digital Input Port (DIP).
Please check antenna connection with antenna port of ETERNITY ME Card GSM or ETERNITY ME Card
GSM.
Please check SIP trunks are programmed correctly in the Routing Group.
2671
Acronyms
ACB
AIP
AIS
ANSI
ANT
AO/DI
AOP
APM
ATA
AVD
Alternate Voice data signaling (Also called clear channel, out-of-band signaling)
BCC
BGM
Background Music
BH
Break Hour
BI
Barge-In
BOS
BPS
BRI
BSCs
BSS
BTSs
CAS
CCC
CCS
CCS
CCWT
CD
Carrier Detect
CDC
CDC
CDR
CESID
CI
Call Incoming
CLIP
CLIR
2672
CO
Call Outgoing
CO
Central Office
COLP
COLR
COS
Class of Service
CPC
CPD
CPTG
CPU
Call Pick-Up
CPW
CRC
CTS
Clear to Send
CPE
CUG
CVT
DCD
DCE
DDI
Direct Dialing-In
DHCP
DIP
DISA
DKP
DLC
DND
Do Not Disturb
DNS
DOP
T1E1
DSR
DSS64
DST
DTE
DTMF
DTR
E1
E-Carrier1 (30B+D)
EID
Exchange ID
2673
ENQ
Enquiry
ETX
End of Text
E&M
FAS
FCBC
FIFO
FM
Frequency Modulation
FSK
FTP
GDT
GND
Ground
GSM
GPRS
HDSL
HLR
HOG
IC
Incoming call
ICWT
IMEI
IP
Internet Protocol
IR
Interrupt
ISDN
ISP
ITU
IVDT
LA
Left Align
LAN
LAN NIC
LCD
LCM
LCR
LCS
LE
Local Exchange
LED
LIFO
2674
LNR
LOF
Loss of Frame
LOS
Loss of signal
LSR
MAC
MCC
MCI
MDF
MFA
Multi-Frame Alignment
MNC
MOH
Music on Hold
MOV
MS
Mobile Station
MSC
MSN
MTBF
NH
NPI
NSS
NT1/2
OG
Outgoing
OMC
OOF
Out of Frame
OSI
OSS
PAS
PBX
PC
Personal Computer
PCM
PCOL
PFT
PIN
PISN
PLCC
PMS
2675
POTS
PPDC
PPM
PPS
PRI
PS
Power Supply
PSK
PSTN
PUK
QAM
QSIG
Q-Signaling
RA
Right Align
RBT
RF
Radio Frequency
RI
Ring Indicator
RLSD
RTC
RTS
Request to Send
RXD
Receive data
SA
System Administrator
SAL
SE
System Engineer
SEFS
SES
SFL
SID
Exchange Identity
SIP
SIM
SLIC
SLT
SMDR
SMPS
SP
Service Provider
STX
Start of Text
SPID
2676
T1
T-Carrier (23B+D)
TA
Terminal Adaptor
TAC
TAG
TCM
TCP/IP
TE1/2
TEI
TLG
TON
TON
Type of Number
TWT
UART
UPS
VIC
VLR
VMS
VMA
WAN
WH
Working Hour
2677
Warranty Statement
Matrix warrants that its products will be free from defects in material and workmanship, under normal use and
service for a period of twelve (12) months from the date of installation.
Matrix warranties the replacement or repair of any product or component(s) found to be defective during the
applicable period and return the same, or grant a reimbursement credit with respect to the product or component.
Parts repaired or replaced will be under warranty throughout the remainder of the original warranty period only. In
case of software program design defect(s) that prevents the program from performing the specified functionality,
affecting service and beneficial use of the product, Matrix reserves the right to incorporate solutions in its new
release of the software and make it available to the customer within a reasonable period of time. The above said
with regard to the software design defect, constitutes the sole obligation of Matrix and its authorized installer with
respect to the product.
Matrix does not, however, affirm or stand for that the functions or features contained in the system will satisfy its
end-user's particular purpose and /or requirements or that the operation of the program will be uninterrupted or
error free.
This warranty is voidable by Matrix:
If the product is used other than under normal use and is not properly serviced and maintained by qualified
technicians.
If the product is not maintained under proper environmental conditions.
If the product is subjected to abuse, damage, misuse, neglect, fire, power flow, acts of God, accident.
If the product is installed or used in combination or in assembly with the products that are not supplied or
authorized by Matrix or are of inferior quality or design than Matrix supplied products, which may cause
reduction or degradation in functionality.
If the product is operated outside the product's specifications or used without designated protections.
If the completely filled warranty cards have not been received by Matrix within 15 days of the installation.
In no event will Matrix be liable for any damages, including lost profits, lost business, lost savings, downtime or
delay, labor, repair or material cost, injury to person, property or other incidental or consequential damages arising
out of use of or inability to use such product, even if Matrix has been advised of the possibility of such damages or
losses or for any claim by any other party.
Except for the obligations specifically set forth in this Warranty Policy Statement, in no event shall Matrix be liable
for any direct, indirect, special, incidental or consequential damages, whether based on contract or any other legal
theory, and where advised of the possibility of such damages.
Neither Matrix nor any of its channel partners makes any other warranty of any kind, whether expressed or implied,
with respect to Matrix products. Matrix and its distributors, dealers or sub-dealers specifically disclaim the implied
warranties of merchantability and fitness for a particular purpose.
This warranty is not transferable and applies only to the original user of the Product. All legal course of action
subjected to Vadodara (Gujarat, India) jurisdiction only.
2678
Regulatory Information
Customer Information-ACTA
Federal Communications Commission Statement
Part 15:
Note: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to
Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference
when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate
radio frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful
interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful
interference in which case the user will be required to correct the interference at his/her own expense.
Part 68:
This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA (Administrative
Council for Terminal Attachments). On the bottom side of this equipment is a label that contains, among other
information, a product identifier in the format US: MTXMF01BETERNITY. If requested, this number must be
provided to the telephone company.
REN Number
The REN is used to determine the quantity of devices that may be connected to the telephone line. Excessive
RENs on the telephone line may result in devices not ringing in response to an incoming call. In most, but not all
areas, the sum of RENs should not exceed 5.0. To be certain of the number of devices that may be connected to a
line, as determined by the total RENs, contact the local telephone company. On the bottom of this equipment is a
label that contains, among other information, a product identifier in the format US: MTXMF01BETERNITY. The
digits represented by 01 are the ringer equivalence number (REN) without a decimal point (for example, 03 is a
REN of 0.3). If requested, this number must be provided to the telephone company.
If this equipment 'ETERNITY' causes harm to the telephone network, the telephone company will notify you in
advance that temporary discontinuance of service may be required. But if advance notice is not practical, the
telephone company will notify the customer as soon as possible. Also, you will be advised of your right to file a
complaint with the FCC if you believe it is necessary.
The telephone company may make changes in its facilities, equipment, operations or procedures that could affect
the operation of the equipment. If this happens, the telephone company will provide advance notice in order for you
to make necessary modifications to maintain uninterrupted service.
If trouble is experienced with this equipment 'ETERNITY', for repair or warranty information, please contact your
dealer. If the equipment is causing harm to the telephone network, the telephone company may request that you
disconnect the equipment until the problem is resolved.
It is recommended that repairs be performed by the representatives of your Sales Representative.
The equipment cannot be used on public coin phone service provided by the telephone company. Connection to
party line service is subject to state tariffs. Contact the state public utility commission, public service commission or
corporation commission for information.
2679
calls are:
b. This equipment returns answer supervision on all DID calls forwarded to the PSTN. Permissible
exceptions are:
A call is unanswered
A busy tone is received
A reorder tone is received
Equal Access:
This equipment is capable of providing the end user equal access to the carrier of the user's choice. This
equipment is capable of providing users access to interstate providers of operator services through the use of
2680
access codes. Modification of this equipment by call aggregators to block access dialing codes is a violation of the
Telephone Operator Consumers Act of 1990.
Electrical Safety:
Telephone companies report that electrical surges, typically lightning transients, are very destructive to customer
terminal equipment connected to AC power sources. This has been identified as a major nationwide problem.
However Matrix provides all protection against lightning transients in the equipment; the user must provide a
suitable surge arrestor while integrating the equipment with other networking equipments.
2681
TEC Certificate:
2682
2683
2684
2685
2686
2687
CE Certificate:
2688
CE Certificate:
2689
2690
RoHS Certificate:
2691
The firmware of this product also includes some of the Open-Source software released under GNU
General Public License (GPL) Version 2. Terms of this license is printed in full below.
The source of the open source software used in this product is available on CD, upon written request from:
R&D Team
Matrix Comsec Pvt Ltd
394, Makarpura GIDC,
Vadodara - 390 010
Gujarat
India.
Customer shall bear the shipping and handling charges.
2692
We protect your rights with two steps: (1) copyright the software, and
(2) offer you this license which gives you legal permission to copy,
distribute and/or modify the software.
Also, for each author's protection and ours, we want to make certain
that everyone understands that there is no warranty for this free
software. If the software is modified by someone else and passed on, we
want its recipients to know that what they have is not the original, so
that any problems introduced by others will not reflect on the original
authors' reputations.
Finally, any free program is threatened constantly by software
patents. We wish to avoid the danger that redistributors of a free
program will individually obtain patent licenses, in effect making the
program proprietary. To prevent this, we have made it clear that any
patent must be licensed for everyone's free use or not licensed at all.
The precise terms and conditions for copying, distribution and
modification follow.
GNU GENERAL PUBLIC LICENSE
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
0. This License applies to any program or other work which contains
a notice placed by the copyright holder saying it may be distributed
under the terms of this General Public License. The "Program", below,
refers to any such program or work, and a "work based on the Program"
means either the Program or any derivative work under copyright law:
that is to say, a work containing the Program or a portion of it,
either verbatim or with modifications and/or translated into another
language. (Hereinafter, translation is included without limitation in
the term "modification".) Each licensee is addressed as "you".
Activities other than copying, distribution and modification are not
covered by this License; they are outside its scope. The act of
running the Program is not restricted, and the output from the Program
is covered only if its contents constitute a work based on the
Program (independent of having been made by running the Program).
Whether that is true depends on what the Program does.
1. You may copy and distribute verbatim copies of the Program's
source code as you receive it, in any medium, provided that you
conspicuously and appropriately publish on each copy an appropriate
copyright notice and disclaimer of warranty; keep intact all the
notices that refer to this License and to the absence of any warranty;
and give any other recipients of the Program a copy of this License
along with the Program.
You may charge a fee for the physical act of transferring a copy, and
you may at your option offer warranty protection in exchange for a fee.
2. You may modify your copy or copies of the Program or any portion
2693
of it, thus forming a work based on the Program, and copy and
distribute such modifications or work under the terms of Section 1
above, provided that you also meet all of these conditions:
a) You must cause the modified files to carry prominent notices
stating that you changed the files and the date of any change.
b) You must cause any work that
whole or in part contains or is
part thereof, to be licensed as
parties under the terms of this
2694
2695
2696
10. If you wish to incorporate parts of the Program into other free
programs whose distribution conditions are different, write to the author
to ask for permission. For software which is copyrighted by the Free
Software Foundation, write to the Free Software Foundation; we sometimes
make exceptions for this. Our decision will be guided by the two goals
of preserving the free status of all derivatives of our free software and
of promoting the sharing and reuse of software generally.
NO WARRANTY
11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN
OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS
TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE
PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
REPAIR OR CORRECTION.
12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED
TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
POSSIBILITY OF SUCH DAMAGES.
END OF TERMS AND CONDITIONS
How to Apply These Terms to Your New Programs
If you develop a new program, and you want it to be of the greatest
possible use to the public, the best way to achieve this is to make it
free software which everyone can redistribute and change under these terms.
To do so, attach the following notices to the
to attach them to the start of each source file
convey the exclusion of warranty; and each file
the "copyright" line and a pointer to where the
program. It is safest
to most effectively
should have at least
full notice is found.
<one line to give the program's name and a brief idea of what it does.>
Copyright (C) <year> <name of author>
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
2697
See the
You should have received a copy of the GNU General Public License along
with this program; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
Also add information on how to contact you by electronic and paper mail.
If the program is interactive, make it output a short notice like this
when it starts in an interactive mode:
Gnomovision version 69, Copyright (C) year name of author
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
The hypothetical commands `show w' and `show c' should show the appropriate parts
of the General Public License. Of course, the commands you use may be called
something other than `show w' and `show c'; they could even be mouse-clicks or
menu items--whatever suits your program.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
`Gnomovision' (which makes passes at compilers) written by James Hacker.
<signature of Ty Coon>, 1 April 1989
Ty Coon, President of Vice
This General Public License does not permit incorporating your program into
proprietary programs. If your program is a subroutine library, you may
consider it more useful to permit linking proprietary applications with the
library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License.
2698
Index
Navigation keys
Down key 146
Enter key 146
Number Lists 2060
Paging 2085
PC Port 143, 153, 294, 403, 413, 485, 597, 607
PCAP Trace 2500
Peer-to-Peer Calling 2091
Peer-to-Peer table 2096
PLCC-An Introduction 2099
Power Adapter 485, 607
Power Adaptor 143, 153, 284, 294, 403, 413, 495,
597
Power jack 143, 153, 284, 294, 413, 495, 607
power jack 403, 485, 597
Power over Ethernet (PoE) 143, 153, 284, 294,
403, 413, 485, 495, 597, 607
Presence 2107
Prioritization of traffic 150, 291
prioritization of traffic 410, 492, 604
Priority 2119
Priority Calls in E&M MFCR2 Signaling 2115
Privacy 2122
Q
QSIG 2124
Quick Dial 2141
R
Raid 2143
RCOC (Return Call to Original Caller) 2145
Real Time Clock (RTC) 2150
Reminder 2156
Reminder - Personalized 2156
Reminder - Snooze 2158
Reminder - Voice-guided 2156
Reminder Status 2162
Remote Programming 2165
Room Monitor 2167
Routing Group 2169
S
T1 Maintenance 2364
T1 RBS Parameters 1077
T1E1 Trunks 1034
Time Tables 2372
Time Zone Display 2376
Toll Control 2377
traffic type 150, 291, 410, 492, 604
Trunk Access Group (TAG) 2392
Trunk Auto Answer 2393
Trunk Call Waiting 2396
Trunk Landing Group (TLG) 2397
Trunk Reservation 2400
U
MATRIX COMSEC
Manufacturing Unit:
19-GIDC, Waghodia - 391760, Dist. Vadodara, India.
Tel.: +91 2668 263172/73
Customer Care:
Tel.: +91 265 2630555
E-mail: Customer.Care@MatrixComSec.com, Support@MatrixComSec.com
www.MatrixComSec.com
Head Office:
394-GIDC, Makarpura, Vadodara - 390010, India.
Tel.:+91 265 2630555, Fax: +91 265 2636598
E-mail: Info@MatrixComSec.com