9600 SERIES VoIP

USER GUIDE

9600IPMWD5, 9600IPMWD, 1-Line IP Handset Kit

9602IPMWD5, 9602IPMWD, 2-Line IP Handset Kit
Table of Contents
9600 Series VoIP Phone Maps . . . . . . . . . . . . . . . . . . . 3 3.8 System Manage. . . . . . . . . . . . . . . . . . . . . . . . . 20
3.8.1 Account Manage . . . . . . . . . . . . . . . . . . . 20
Functions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 3.8.2 Syslog Configuration. . . . . . . . . . . . . . . . 20
3.8.3 Phone Book. . . . . . . . . . . . . . . . . . . . . . . 20
Standards and Protocols. . . . . . . . . . . . . . . . . . . . . . . . 5
3.8.4 Time Set—via Simple Network
1. Introduction. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Time Protocol (SNTP) . . . . . . . . . . . . . . . . . . . 20
1.1 Overview of Hardware . . . . . . . . . . . . . . . . . . . . 6 3.8.5 MMI Configuration . . . . . . . . . . . . . . . . . 21
1.1.1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 3.8.6 Logout and Reboot System . . . . . . . . . . 21
1.1.2. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. Operating Method for Dialing. . . . . . . . . . . . . . . . . 21
1.2 Overview of Software . . . . . . . . . . . . . . . . . . . . . 6
4.1 How to Dial an IP Phone. . . . . . . . . . . . . . . . . . 21
2. Handset Keys for 9600 Series. . . . . . . . . . . . . . . . . . 7
4.2 Set Up the Phone to Register to a SIP Server.21
2.1 Function Table of Keystrokes. . . . . . . . . . . . . . . 7 4.2.1 Set the WAN Interface . . . . . . . . . . . . . . 21
2.1.1 MWI—Message Waiting 4.2.2 SIP Configuration . . . . . . . . . . . . . . . . . . 22
Indication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Reference—Quick Keys. . . . . . . . . . . . . . . . . . . . . . . . 23
2.1.2 Call Volume Control . . . . . . . . . . . . . . . . . 7
Reference—SIP Quick Start to Register
2.1.3 Hold Function . . . . . . . . . . . . . . . . . . . . . . 7
a Single Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
2.1.4 Redial Function. . . . . . . . . . . . . . . . . . . . . 7
2.1.5 Speed Dial Keys. . . . . . . . . . . . . . . . . . . . . 7 Program Your Connection Under the
VoIP -> SIP Configuration Screen. . . . . . . . . . . . . . 24
3. Web Browser User Interface to
Configure the Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . 8 Speed Dial and OneTouch Message Keys
Are Programmed Under Advance ->
3.1 Login . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Memory Key. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
3.2 Current Status. . . . . . . . . . . . . . . . . . . . . . . . . . 8
Reset to Factory Default. . . . . . . . . . . . . . . . . . . . . 24
3.3 Network. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Reference—DECT Handset Base
3.3.1 WAN Configuration. . . . . . . . . . . . . . . . . . 8
Registration / Deregistration . . . . . . . . . . . . . . . . . . . 25
3.3.2 LAN Configuration . . . . . . . . . . . . . . . . . . 9
Register One Handset . . . . . . . . . . . . . . . . . . . . . . 25
3.4 VoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.4.1 SIP Configuration . . . . . . . . . . . . . . . . . . . 9 Register Up To 4 Additional Handsets. . . . . . . . . . 25
3.5 Advance. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12 Delete Handset Registrations
3.5.1 DHCP Server. . . . . . . . . . . . . . . . . . . . . . 12 from DECT-SIP Base . . . . . . . . . . . . . . . . . . . . . . . 25
3.5.2 NAT Configuration. . . . . . . . . . . . . . . . . . 12 Changing Handset Ring Tone:
3.5.3 STUN Configuration . . . . . . . . . . . . . . . . 13 6 Ring Tones to Choose From . . . . . . . . . . . . . . . . 25
3.5.4 Net Service . . . . . . . . . . . . . . . . . . . . . . . 13
3.5.5 Firewall Configuration . . . . . . . . . . . . . . 13 Important Safety Instructions. . . . . . . . . . . . . . . . . . . 26
3.5.6 VLAN Configuration QoS Configuration.14 FCC Interference Information. . . . . . . . . . . . . . . . . . . 26
3.5.7 Digital Map Configuration . . . . . . . . . . . 15
FCC RF Radiation Exposure Statement. . . . . . . . . . . 27
3.5.8 Call Service. . . . . . . . . . . . . . . . . . . . . . . 16
Industry of Canada Requirements. . . . . . . . . . . . . . . . . . 27
3.5.9 Memory Key Programming . . . . . . . . . . 17
3.5.10 MMI Filter . . . . . . . . . . . . . . . . . . . . . . . 18 Requirements of Part 15—FCC Rules . . . . . . . . . . . . 28
3.5.11 DSP Configuration/Audio Requirements of Part 68—FCC Rules . . . . . . . . . . . . 28
Settings. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18 Technical Specifications . . . . . . . . . . . . . . . . . . . . . . . 29
3.6 Configuration Management. . . . . . . . . . . . . . . 18 Dimensions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
3.7 Update Firmware . . . . . . . . . . . . . . . . . . . . . . . 19 Weight . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
3.7.1 Update Configuration. . . . . . . . . . . . . . . 19 User Guides . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
3.7.2 Autoprovision Firmware Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
Update. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Statement of Limited Warranty . . . . . . . . . . . . . . . . . 29

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9600 Series VoIP Phone Map

13. Handset Locate key

14. Base Microphone
15. Speaker key with LED indicator
16. Mute key with LED indicator
17. OneTouch Message Waiting key
18a. Line 1 with LED indicator
18b. Line 2 with LED indicator
19. Electronic HOLD key with On/Off
indicator and remote release
20. Dial Pad
21. Interchangeable handset cradle
BASE FEATURES AND CONTROLS 22. Handset Charging location
1. Industrial grade construction, 23. In Use—Ringer indicator
smooth plastic finish 24. Charging Indicator
2. Speaker (half duplex) 25. Desi strip display area
3. Message Waiting selector switch
(submerged) HANDSET KIT CONTROLS
4. Programmable Store key 26. Cordless Handset
(submerged) 27. Message Waiting Indicator
5. Programmable Flash key 28. ADA Compliant Handset Speaker
(submerged) 29. Handset Faceplate and Plastic
6. Line Cord receptacle Overlay
7a. HSIA (High Speed Internet Access) 30. Handset Ringer Volume switch
7b. HSIA (High Speed Internet Access) 31. Line Selector key
8. Faceplate and Plastic Overlay 32. Hold/Conf. Handset key
9. Base Ringer Volume switch 33. Line Selector key
10. Guest Services Memory keys 34. Handset Dial Pad
11. Programmable Pause/Redial key 35. Last Number Redial key
(submerged) 36a. Handset Volume Up
12a. ADA Compliant Speaker Volume 36b. Handset Volume Down
key (up) 37. Flash key
12b. ADA Compliant Speaker Volume 38. Charging Location Contacts
key (down) 39. Handset Microphone

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9600 Series VoIP Phone Map

UNDERSIDE CONTROLS CORDLESS HANDSET

39. Base underside sticker CHARGING STATION
information: 42. Message Waiting indicator
Certification—FCC, CE, ETL, RoHS 43. Cordless Handset
Version label 44. Handset Pop-Off Cover for
Model name battery compartment
REN number 45. Interchangeable Accent Piece
Disposal label 46. AC Power Plug-In (back of
QC label stand)
Series number 47. Battery Charging and
Warranty Void If Seal Broken Synchronization programming
label indicator light.
40. Voltage Settings—24V, 48V 48. Cordless Handset Charging
and OFF for Line 1 and Line 2 Stand
(factory set at 24V)
Line Voltage 24-52VDC (+/-) POWER ADAPTOR
Line Current 15mA to 80mA 49. Phone Line from wall goes into
Ringer Equivalance—1.3B Line receptacles
41. Non-Slip rubber feet (4) 50. Line to Phone from Adaptor

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Functions Standards and Protocols
• Support failover connection to a redundant • IEEE 802.3 / 802.3 u 10 Base T / 100 Base TX
SIP server • PPPoE
• Support NAT (Network Address Translation), • DHCP Client and Server
Firewall • Support G.711a/u,G729, G7231 5.3/6.3 audio
• Support DHCP for phone to accept IP via WAN Codec
port or assign IP address via LAN port • SIP RFC3261, RFC 2543
• Support PPPoE (used when connecting the • Support IAX2
ADSL or cable modem) • TCP/IP: Internet transfer and control protocol
• Update the firmware and/or configuration file (Transport Control Protocol/Internet Protocol)
program through HTTP, FTP, and TFTP • RTP: Real-time Transport Protocol
• Hold function • RTCP: Real-time Control Protocol
• Hotline function (dial a specific number as • VAD/CNG save bandwidth (Voice Activity
soon as going off-hook or set a delay before Detection)
dialing) • Telnet: Remote login protocol
• Speed-dial • DNS: Domain Name Server
• Call-forward, Three-way conference call (in • TFTP: Trivial File Transfer Protocol
select 2-line models)
• Caller ID display
• DND (Do Not Disturb), Black List, Limit List
• Auto-answer
• Program settings through a standard web
browser
• Remote management functionality via Telnet
• Classification management for common
user’s password and superuser’s password

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1. Introduction Voice Quality
•  VAD: Voice Activity Detection
•  CNG: Comfortable Noise Generator
This is the User Guide for the 9600 Series VoIP phone. •  LEC: Line Echo Canceller
This phone is a standard SIP (Session Initiation •  Packet Loss Compensation
Protocol) phone, and some configuration is •  Adaptive Jitter Buffer
necessary before it is ready for use. This manual
Call Function
will illustrate how to set up the phone through
keyboard and Web User Interface (UI). •  Call Hold •  Call Forward
• Call Waiting • Three-way Conference
1.1 Overview of Hardware Tone
1.1.1 •  Ring Tone •  Dial Tone
The two RJ-45 network interfaces support •  Ring Back Tone •  Busy Tone
10/100M Ethernet. The default WAN interface is a Phone Function
DHCP Client. The user connects the WAN interface •  Volume Adjustment •  Phone Book
to a Power Over Ethernet (POE) enabled switch •  Speed Dial Key
to obtain power and an IP address. (Alternatively,
the phone can be assigned a static IP address.) IP Assignment
The LAN port is commonly used for bridging a • IP (Static IP)  • DHCP  • PPPoE
guest computer to the network. You can use the Security
default administrator’s user name “admin” and
password “admin” to login and setup the device’s • HTTP 1.1 Basic/Digest Authentication Web
configuration. Setup
• MD5 for SIP Authentication (RFC2069/
1.1.2 RFC2617)
Only the WAN interface supports POE (802.3af). QoS
•  QoS Field
1.2 Overview of Software NAT Traversal
Network Protocol • STUN
•  SIP v1 (RFC2543) Configuration
•  SIP v2 (RFC3261)
• Web Browser  • Console/Telnet  • Keypad
• IP/TCP/UDP/RTP/RTCP
• IP/ICMP/ARP/RARP/SNTP DTMF
•  TFTP Client/DHCP Client/PPPOE Client •  DTMF RELAY •  DTMF SIP Info
•  Telnet/HTTP Server •  DTMF RFC 2833
•  DNS Clients
SIP Server
Codec
• Support two SIP servers working at the
•  G.711: 64K bit/s (PCM) same time
•  G.723.1: 63k/5.3k bit/s •  Provide a backup SIP server
•  G.726: 16k/24k/32k/40k bit/s (ADPCM)
Firmware Upgrade
•  G.729A: 8k bit/s (CS-ACELP)
•  G.729B: Adds VAD & CNG to G.729 • TFTP  • 
HTTP  • FTP

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2. Handset Keys for 2.1.1 MWI—MESSAGE WAITING
INDICATION
9600 Series The Messages key serves as an indicator that
messages are present as well as a speed dial
key to dial in and retrieve those messages. If
2.1 Function Table of Keystrokes messages are present, press the flashing red MWI
key labeled Messages to go off-hook and dial the
Name Status Function/Display programmed voicemail number. First, program
Hold Call Hold on the speed dial to the correct voice mail port of
your PBX in the Web UI (found under Advanced ->
MWI On-Hook Press flashing
Memory Key -> MWI Number).
light to retrieve
messages 2.1.2 CALL VOLUME CONTROL
Mute Press Mute voice path Press “Volume +” to increase the volume and
Redial Call/ Call the number “Volume –“ to decrease the volume.
Dialing dialed last time
2.1.3 HOLD FUNCTION
Spkr Speaker
Hold the current line’s call.
Volume + Call Increase the volume
Volume – Call Decrease the 2.1.4 REDIAL FUNCTION
volume Redial the last number dialed.
M1–M7 On-Hook 7 speed dial
numbers
2.1.5 SPEED DIAL KEYS
1 Dialing “1” The 9600 Series phone has 7 speed dial keys
programmable through the Web UI under
2 Dialing “2” Advanced -> Memory Key.
3 Dialing “3”
4 Dialing “4”
5 Dialing “5”
6 Dialing “6”
7 Dialing “7”
8 Dialing “8”
9 Dialing “9”
0 Dialing “0”
* Dialing “*”
# Dialing It can be regarded
as the first number
being dialed out or
the end mark for
ending number

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3. Web Browser User the telephone numbers in SIP Line 1 and SIP Line
2. (SIP Line 2 is commonly used as a failover to a

Interface to Configure second IP/PBX in case of emergency.)

the Phone
In most cases, the phone will obtain an IP
address from the local DHCP server. You may
find this IP address from the phone by using a
simple diagnostic code. On the phone’s dialpad,
press * * 4 7 # (* * I P #). The phone will read
out the IP address (and display it on the LCD if 3.3 Network
the phone has a screen). To access the phone, 3.3.1 WAN CONFIGURATION
it is necessary that your computer’s IP is on the
same network with the phone IP, so make sure WAN PORT NETWORK SETTING PAGE
it gets a compatible IP address within the same Supports Static IP, DHCP (Dynamic Host Control
network. Then open a web browser, input the Protocol), and PPPoE.
phone’s IP address in the address field, and you
will enter the web-based User Interface pages of
the 9600 Series phone.

3.1 Login
The default Username and Password are admin/
admin and guest/guest.

CONFIGURE STATIC IP

3.2 Current Status

The Current Status page shows the working state – Enable Static
of the VoIP phone. The network part shows the – Set 9600 Series’ IP address in the IP Address
connection state of the WAN interface and the field
LAN interface as well as the network setting – Set the Subnet mask in the Netmask field
(DHCP or Static): the MAC address of the phone – Set router IP address in the Gateway field
and the Firmware version. Here you can also see – DNS Domain
whether the phone is registered to the IP/PBX – Set local DNS server IP addresses in the
server or not. The Phone Number section shows Primary DNS and the secondary IP address in
the Alternate DNS field

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Configure to dynamically obtain the IP. will automatically modify the DHCP Lease Table
and save the configuration according to the IP
Enable DHCP.
and netmask information. The DHCP server
If there is a DHCP server in your local network, configuration won’t take effect until you reboot
the 9600 Series phone will automatically obtain the device.
the WAN port network information from your
NAT: Enable Network Address Translation (NAT)
DHCP server.
for navigating through firewalls.
CONFIGURE PPPOE Bridge Mode: Enable this option to switch to
Bridge Mode. An IP phone won’t assign an
internally generated IP for its LAN port in Bridge
Mode. Most hotel properties will bridge the LAN
and WAN port, but separate the Guest network
from the Hotel’s IP/PBX by implementing VLANs
– Enable PPPoE (Virtual LAN’s). (This setting won’t take effect
– PPPoE Server: Enter “ANY” if not specified unless you save the configuration and reboot the
from your ITSP device.)
– Enter PPPoE username and pin in the
Username and Password entries 3.4 VoIP
The 9600 Series phone will automatically obtain 3.4.1 SIP CONFIGURATION
the WAN port network information from your SETTING PAGE OF PUBLIC SIP SERVER
ITSP if the PPPoE setting and the setup are
correct. Note: If a user accesses the IP phone
through a WAN port he or she should use the new
IP address to access the IP phone when the WAN
port address was changed.

3.3.2 LAN CONFIGURATION

Server Address: The IP address of the SIP server

that the SIP phone will attempt to register to.
Server Port: The Internet Socket Port number
of the SIP registration server. The default port is
LAN IP: This is where you can set the IP address
5060.
of the phone (at its LAN port). This is seldom used
because in most environments Bridge Mode is Account Name: Username of your SIP account.
enabled, causing the LAN port to act as a bridged (Usually the same as the phone number.)
extension of the WAN port. Password: Password of your SIP account.
Netmask: Set the IP Subnet mask for the LAN. Phone Number: Phone number of your SIP
DHCP Service: Enable DHCP service on the LAN account.
port. After the user changes LAN IP, the phone

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Enable Register: Enable/Disable SIP register. time to the server recommended setting if it is
The 9600 Series phone won’t send registration different from the SIP server.
information to the SIP server if the register is Forward Type: Call forwarding has three types:
disabled. Always, Busy, and No Answer. You may select
Display Name: This field will display the User the type of forwarding here, and then you must
Agent (UA) in the header. complete the configuration by Disabling Call
Waiting (Found in Advanced -> Call Service).
Proxy Server Address: IP address of proxy SIP
server. (A SIP provider usually uses the same IP No answer: If no answer after a set period of
for the register server and the proxy server. In time (No Answer Time is found in Advanced ->
this case you don’t need to configure the proxy Call Service) it will forward to the appointed
server information.) phone.
Proxy Server Port: Signal port of the SIP proxy. Always: The caller is always forwarded to
the appointed phone.
Proxy Username: Proxy server username.
Proxy Password: Proxy server password. NAT Keep Alive Interval: The Network Address
Translation (NAT) is the process of modifying IP
Domain Realm: SIP domain—enter the SIP address information in IP packet headers while
domain if any. Otherwise, the 9600 Series phone in transit across a traffic routing device. Keep
will use the proxy server address as the SIP Alive is a method of maintaining that the service
domain. remains active.
Enable Message Waiting: The configuration Forward Phone Number: The designated phone
allows or forbids Message Waiting. number calls will forward to.
ADVANCED SIP SETTING User Agent: The User Agent (UA) is an identifier
for the phone and is part of the SIP Header used
Advanced SIP Settings rarely need to be changed.
with every packet.
Most of the settings should be left at Default, and
many are not applicable to hospitality phones. Server Type: The particular IP/PBX system
Two of the settings you may (in rare instances) supplier carries out the sign and speeches to
need to change are DTMF Mode and Forward encrypt. The default is COMMON and should
Type (however, forwarding in hospitality is usually always be used.
handled by the PBX). Signal Key: Not applicable in Hospitality
environments.
DTMF Mode: DTMF (Dual Tone Multi-Frequency)
signal sending mode: supports RFC2833, DTMF_
RELAY (inband audio), and SIP information where
DTMF is sent as a SIP message. (DTMF is the
acronym for the tones that are send from the
number dialpad.)
DTMF SIP INFO Mode: DTMF Tones can
be sent as SIP messages as well as audio
Register Expire Time: Registration expiration streamed tones. SIP INFO mode has two
time. The default is 60 seconds. The 9600 Series differentiations in this drop-down menu.
phone will automatically configure this expire The default of “Send 10/11” is recognized by

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 most IP/PBXs, but some vendors (Avaya CM) NAT Keep Alive: Usually not applicable in
recognized “Send */#”. Hospitality environments. The default is
unchecked. “Keep Alive” is a term used to describe
Media Key: Not applicable in Hospitality
the act of keeping a persistent connection. Ports
environments.
are opened and closed in IP communications.
RFC Protocol Edition: SIP versions are identified A keep-alive prompt/command ensures the
by the IETF (Internet Engineering Task Force). connection is NOT dropped, but is instead kept
SIP version 1 is RFC 2543. The default is the open. When the client sends another request,
newer SIP 2 RFC 3261. An example of when it uses the same connection. This will continue
you might revert back to RFC2543 is if the gate until either the client or the server decides that
needs to communicate to older devices (such as the conversation is over, and one of them drops
CISCO5300) using the SIP 1.0 (RFC2543 protocol). the connection.
Default is RFC 3261.
Rtp Encode: Not applicable in Hospitality
Local Port: The Standard Internet Port for SIP environments. Video H.273 encode related.
signaling is 5060. However, sometimes these
Enable Via Rport: Default is enabled/checked.
port numbers are changed to enhance security.
This checkbox enables or disables rport
They would be changed in Basic Settings as well
parameter for the Via header field. The default
as in Advanced Settings.
setting allows a client to request that the server
Transport Protocol: Default is UDP (User send the response back to the source IP address
Datagram Protocol) and should be used instead and port from which the request originated.
of the TCP (Transport Control Protocol) option. However, in some environments it might be
Ring Type: The alerting ring that the phone desired to switch this parameter off.
presents to the called party is set here. Default is Enable Session Timer: Default is unchecked. Not
the USA conventional ring Type 1. applicable in Hospitality environments.
Subscribe Expire Time: Message Waiting Enable PRACK: Default is unchecked/disabled.
Indication is an event package that must be SIP PRACK (Provisional Acknowledgement) is a
subscribed to. The default is 300 seconds to way to enable reliability for SIP 1xx messages like
check and re-subscribe with the server. 180 ringing and 183 session in progress. PRACK
Enable Subscribe: Enable the subscription to the messages flow from calling party to called party.
voice mail service. PRACK reliability has been found to be important
in several cases, including interoperability
Enable URI Convert: Default is enabled/checked.
scenarios with the PSTN and Proxy servers with
A Uniform Resource Identifier (URI) is a string of
SIP v1 RFC2543. The default is DISABLED.
characters used to identify a name or a resource.
Such identification enables interaction with Answer with Single Codec: Not applicable
representations of the resource over a network in Hospitality environments. The default is
(typically the World Wide Web) using specific unchecked/disabled.
protocols. Leave this parameter selected. Long Contact: Not applicable in Hospitality
Enable Keep Authentication: Not applicable environments. The default is unchecked/disabled.
in Hospitality environments. The default is Auto TCP: Not Applicable in Hospitality
unchecked. environments. The default is unchecked/disabled.
Signal Encode: Not applicable in Hospitality Auto TCP refers to TCP Auto-tuning. TCP tuning
environments. The default is unchecked. techniques adjust the network congestion

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avoidance parameters of TCP connections over DHCP Lease Table: Shows the IP-MAC
high-bandwidth, high-latency networks. corresponding table assigned by the DHCP
server.
Click to Talk: Not applicable in Hospitality
environments. The default is unchecked/disabled. Note: This setting won’t take effect unless you
save the configuration and reboot the device.
3.5 Advance
3.5.2 NAT CONFIGURATION
3.5.1 DHCP SERVER
ADVANCE NETWORK ADDRESS TRANSLATION
DHCP SERVER MANAGE PAGE (NAT) SETTING
User may trace and modify DHCP server Maximum of 10 items for TCP and UDP port
information on this page. mapping.

IPSec ALG: Enable/Disable IPSec ALG.

DHCP Lease Table: Displays the IP-MAC
corresponding table that the server distributed. FTP ALG: Enable/Disable FTP ALG.
Lease Table Name: Lease table name. PPTP ALG: Enable/Disable PPTP ALG.
Start IP: The starting IP of the lease table or Transfer Type: Transfer type using port mapping.
DHCP Scope. Outside Port: WAN port for port mapping.
End IP: The ending IP of the lease table or DHCP Inside IP: LAN device IP for port mapping.
Scope. Any network device connecting to the
9600  Series phone’s LAN port can dynamically Inside Port: LAN device port for port mapping.
obtain the IP range between the start IP and the Click Add to add new port mapping item and
end IP. Delete to delete current port mapping item.
Lease Time: DHCP server lease time defining DMZ CONFIGURATION
how long the IP address handed out to the
network device is valid.
Netmask: Subnet mask of the lease table.
Gateway: Default gateway of the lease table.
DNS: Default DNS (Domain Name Server) of the
lease table.
The purpose of a DMZ is to add an additional
DNS Relay: Enables the Domain Name Server
layer of security to an organization’s Local Area
relay function.

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Network (LAN). An external attacker only has Use STUN: The STUN setting that allows (enables)
access to equipment in the DMZ, rather than any or forbids use.
other part of the network. The name is derived
from the term “demilitarized zone,” an area 3.5.4 NET SERVICE
between nation states in which military action is
not permitted.

3.5.3 STUN CONFIGURATION

This page is used to set the private SIP server,
STUN server, and back up SIP server information.

STUN SERVER SETTINGS HTTP Port: Configures the Hyper Text Transfer
STUN means Simple Traversal of UDP through Port; the default HTTP port is 80. User may change
NATs (Network Address Translation). It is a this port to enhance the system’s security (the
protocol used for assisting devices behind a NAT number must be between 1024 and 65000). For
firewall or router with their packet routing. Used example, if you were to change to port 34687, then
to penetrate through NAT, start by configuring when this port is changed, you would use a generic
IP and port of STUN server (default is 3478) and IP address of format http://xxx.xxx.xxx.xxx:34687/
Enabling SIP Stun. This should enable routing to reconnect.
of SIP UDP packets from the phone through a Telnet Port: Configures the Telnet transfer port;
NAT firewall. STUN only supports three NAT the default is 23.
methodologies: FULL CONE, restricted, and port
restricted. RTP Initial Port: Real-time Transport Protocol
initial port. RTP is the standard packet format for
delivering audio and video over IP networks.
RTP Port Quantity: Maximum number of
unpriveliged UDP ports to be used for delivering
the RTP packets; the default is 200.
Note: Settings on this page won’t take effect
unless you save and reboot the device.
If you need to change the Telnet port or HTTP
port, use a port greater than 1024, because ports
under 1024 are reserved ports.
STUN Server Addr: Configures the STUN server
address. 3.5.5 FIREWALL CONFIGURATION
STUN Server Port: Configures the STUN server Firewall Settings Page: User may set up a firewall to
port. The default is 3478. prevent unauthorized Internet users from accessing
private networks connected to the Internet (input
STUN Effect Time: STUN detects the NAT type rule), or prevent unauthorized private network
circle; unit: minute. devices from accessing the Internet.
Local SIP Port: The SIP port of this phone. Access List Supports Two Type Limits: The
Load: Load the choices of the SIP line. input_access limit or output_access limit. Each
type supports 10 items maximum.

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9600 Series firewall filter is the base WAN port. So 3.5.6 VLAN CONFIGURATION/QOS
the source address or input destination address CONFIGURATION
should be the WAN port IP address.
The 9600 Series phones implement Virtual LANs
FIREWALL CONFIGURATION and Quality of Service (QoS) within this section
of the menu. The QoS is based on the 802.1p
protocol. The QoS is used to mark the network
communication priority in the data link/MAC sub-
layer. An 9600 Series phone will sort the packets
using QoS and send it to the destination.

VLAN Enable: If enabled the traffic over the WAN

In_access Enable: Enables the in_access rule. port (and the bridged LAN port) will separate
Out_access Enable: Enables the out_access rule. voice, signaling, and data transmission. This
is an essential security feature in hospitality
Input/Output: Select rule type input rule or
environments to separate voice on one Virtual LAN
output rule.
and guest data traffic on a second Virtual LAN.
Deny/Permit: Select rule type deny rule or permit
VLAN ID Check Enable: VLAN ID is the
rule.
identification of the Virtual Local Area Networks.
Protocol Type: Protocol used in this rule: TCP/IP/
DiffServ Enable: If enabled the VLAN service
ICMP/UDP.
indicates to use the DSCP mode (DiffServ Code
Port Range: Port range selections are more than, Point) to differentiate three layers of QoS (Quality
less than, equal, or not equal. of Service).
Src Addr: Source address. It can be a single IP 802.1P Priority: QoS (Quality of Service) is based
address or a network address range. on the 802.1p protocol. QoS is used to tag the
Des Addr: Destination address. It can be an IP packets with a Priority number (0–7 with 7 being
address or a network address range. highest priority and 0 being lowest). Usually
voice packets will be tagged in order to ensure
Src Mask: Source address mask. Indicates the that voice quality will be preserved. This tagging
source is dedicated IP if it is set to 255.255.255.255. happens in the Layer 2 data link/MAC sub-layer.
Otherwise, this is the network ID. A 9600 Series phone will sort the packets using
Des Mask: Destination address mask. Indicates QoS and send them to the destination.
the source is dedicated IP if it is set to Voice VLAN ID: Is used for establishing a secure
255.255.255.255. Otherwise, this is the network separate Virtual LAN for Voice signaling and
ID. traffic to the voice server. The value ranges are
0–4095. Default value is 256.

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IMPORTANT: Voice/Data VLAN differentiated IS A digital map is based on rules judging when the
ALWAYS SET TO “Undifferentiated” in the drop down user stops dialing and when to send the string
menu. of numbers to the server. 9600 Series phones
support the following digital map:
DiffServ Value: The value range: 0x28,0x30,
0x38,0x48,0x50,0x58,0x68,0x70,0x78,0x88,0x90,0x – End With “#”: Use # as the end of dialing.
98,0xb8. (Default is 0xb8.) 0xb8 stands for best – Fixed Length: When the length of the dialing
fast transmission. 28–38 is for the transmission matches, the call will be sent.
priority of the first rank. 48–58 is for the – Timeout: Specify the timeout of the last digit
transmission priority of the second rank. 68–78 dialed. The call will be sent after the timeout.
is for the transmission priority of the third rank. [ ] represents the range of digits used. It can be
88–98 is for the transmission priority of the fourth a range such as [1–4] or it can use commas such
rank. as [1,3,5] or it can use a list such as [234].
Data 802.1P Priority: QoS (Quality of Service is x represents any one digit between 0–9.
a 3-bit field called the Priority Code Point (PCP)
within an Ethernet frame header when using Tn represents the last digit for a timeout. It is
VLAN tagged frames, as defined by IEEE 802.1Q. necessary for n to represent the time from 0–9
It specifies a priority value of between 0 and 7 seconds. Tn must be the last alpha and digit in
inclusive that can be used by QoS disciplines to the entry. If Tn is not included in the entry, we use
differentiate traffic. (0 is lowest priority traffic T0 as default: which means the system will send
and 7 is highest priority traffic.) the number immediately if the number matches
the entry.
Data VLAN ID: Is used for establishing a secure
separate Virtual LAN for Guest Data traffic to the EXAMPLES:
Internet. The value ranges are 0–4095. Default
8[2-8]xxxxx All numbers from 8200000 to
Value is 254.
8899999 will be sent immediately.
3.5.7 DIGITAL MAP CONFIGURATION 955xx5 Six–digit numbers that begin with 9 will
be sent immediately.
10060 The number 10060 will be sent immediately.
22xxxxx T1 Seven-digit numbers that begin with
22 will be sent after one second.
39[3,9]xxxx Seven-digit numbers beginning with
393 or 399 will be sent immediately.

BEST PRACTICES: DIAL PLAN PATTERNS

Below are some examples of best practices.
These examples will help those in a hospitality
The digital map is a set of rules to determine environment to set up a useable dial plan.
when the user has finished dialing.
The Dial Plan (Digital Map) is a set of patterns.
9600 Series phones support the following digital As the user dials, as soon as the digits entered
map: match one of the patterns, dialing is considered
complete and the call is initiated.

9600 SERIES VoIP USER GUIDE   www.telematrix.net 15

ELEMENTS IN A DIAL PLAN PATTERN: Hot Line: Configures the hotline number. If this
option is set, 9600 Series immediately dials this
X Matches any dialed digit, #, or *
number when the handset is picked up, or the
0–9, *, # Matches the digit
speaker key is pressed, taking the phone off-
[<digits>] Matches any one of the digits
hook.
between brackets
, Play a secondary dial tone if the No Answer Time: The no answer/call forward
patter up to this point has been time setting. Default is 20 seconds before a call
matched, e.g. “9,1xxxxxxxxx” forward.
means play a dial tone after a 9 is P2P IP Prefix (Peer-to-Peer IP Prefix): Not
entered. applicable for Hospitality environments. Sets the
Tn Additional digits may be dialed, but Prefix for Peer-to-Peer IP calls. For example, if
the pattern will be matched when you want to dial the number 192.168.1.119, you
n seconds elapse with no further define the P2P IP Prefix as “192.168.1.”. So you
dialing dial only 119 to reach the number 192.168.1.119.
IN THE U.S., A RECOMMENDED MINIMUM DIAL Default is “.”. If there is no “.” set, the result
PLAN INCLUDES: disables the dialing IP.

Item1 Rule :0 Auto Answer: Enable/disable the auto-answer

Item2 Rule :911 #Emergency function.
Number Do Not Disturb: DND (Do Not Disturb) enables
Item3 Rule :9911 #Emergency this option to refuse all calls.
Number
Ban Outgoing: Enabling this bans outgoing calls.
Item4 Rule :[2–7]xxx #as needed to cover
internal numbers Enable Call Transfer: This is used in business
Item5 Rule :91xxxxxxxxxx #as needed to cover sets to FWD (Forward) or transfer a call. It is not
external numbers enabled in hospitality sets.
Enable Call Waiting: Enable/disable the call
3.5.8 CALL SERVICE
waiting option. (It is also necessary to disable Call
Waiting when enabling Call Forwarding under
VOIP -> SIP Configuration -> Advanced Set.)
Enable Three Way Call: Default is enabled,
allowing for conference calling.
Accept Any Call: Default is enabled. If this option
is disabled, the 9600 Series phone refuses the
incoming call when the called number is different
from 9600 Series phone number.
Black List: Any incoming call from these phone
numbers will be refused.
The user can also configure Call Service options
Limit List: Any outgoing calls using these phone
such as a hotline, call forward, call transfer, call
numbers will be refused.
waiting, 3-way conference call, auto-answer, etc.,
on this page.

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3.5.9 MEMORY KEY PROGRAMMING 3.5.9.2 CALL-PARK/RETRIEVE
This page layout shows the phone numbers used Call-Park/Retrieve allows the phone’s user to
by the speed dial keys. There are several special send a call to a ‘Park’ location (setup within the
functions that can be programmed into the PBX), where another user may then ‘Retrieve’ the
memory keys, such as Transfer, Transfer-Silent, call from a different phone.
and Call-Park/Retrieve. It is programmed as follows in two steps:
1. Under SIP Config -> Advanced Set you will
find Park Mode. Default is Off, and Park 1
is enabled or On.

3.5.9.1 MEMORY KEY SPECIAL

PROGRAMMING—TRANSFER, TRANSFER
SILENT, AND 1-SECOND PAUSE
Transfer occurs when there is an active/live call
in progress. If no call is active, the ‘T’ is ignored,
and the memory key will just call the programmed
number (1107). In the settings below, Memory
1 (T1107) will transfer to extension 1107 when
pressed. The user will hear the DTMF tones as
they are transferred to 1107. 2. Under Advanced -> Memory Key, program
the Memory Key HdActive to Transfer
Silent the call to the Park Location (70 in
the example below)—for example, the code
used would then be TS70.
In the settings above, Memory 2 (TS1107) will
transfer “silently” to extension 1107. ‘Silent’
means the user will not hear the DTMF tones.

In the settings below in Memory 3, the transfer
to 1107 will occur, after the connection is Also program Memory Key HdIdle to
established, a pause will happen then digits are Retrieve from the Parked Location, in
sent. P equals a 1-second pause, so PPP equals the above case TS71 is used. The Park
a 3-second pause, then the digits 12345 are sent. and Retrieve numbers (70 and 71) are
arbitrary, and subject to the programmer
of the PBX. They are used here as an
example only.

9600 SERIES VoIP USER GUIDE   www.telematrix.net 17

3.5.10 MMI FILTER Ring Volume: The volume of the alerting ring as
heard by the called party.
The MMI filter is used limit access to the 9600
Series phone. When an MMI filter is enabled, only G729 Payload Length: G729 payload length.
the IP addresses within the start IP and end IP G.729 is an audio data compression algorithm for
can access that 9600 Series phone. voice that compresses digital voice in packets of
10 milliseconds duration. It is officially described
as Coding of speech at 8 kbit/s. The payload is
the user data being transmitted, and the length
is measured in milliseconds.
Signal Standard: Signal standard for different
areas—this refers to the alerting ring tone that
the calling party hears (ringing in their ear) as
they call another party.
3.5.11 DSP CONFIGURATION/AUDIO VAD: Enable/disable the VAD (Voice Activity
SETTINGS Detection) option—VAD can facilitate speech
processing, and can also be used to deactivate
some processes during non-speech section
of an audio session. It can avoid unnecessary
coding/transmission of silence packets in Voice
over Internet Protocol applications, saving on
computation and on network bandwidth.

3.6 Configuration Management

CODEC (First, Second, Third, and Fourth): Select
the preferred CODEC. Supports uLAW, aLAW,
G729, and G723 The CODEC (Coder/Decoder)
compression algorithms are negotiated with
the IP/PBX server, and negotiation attempts
will happen in the order seen here. The First
CODEC will have priority over the others listed in
descending order when attempting to negotiate
recognized compression algorithms with the
server.
Default Ring Type: This is the alerting “Ring” tone
presented to the user when the phone is ringing. Save Configuration: Saves the current settings.
Handdown Time: Handset down detect time. The Backup Configuration: Backup the configuration
amount of time before the handset goes on-hook. file by using the right key of mouse to click on
save target as. A save window appears. Type the
Input Volume: Handset in volume (receiver).
configuration file name in the File name area.
Output Volume: Handset out volume (speaker). (The file type is a text file.)
Handfree Volume: The speaker phone volume. Clear Configuration: Restores to the default
settings.

18 www.telematrix.net   9600 SERIES VoIP USER GUIDE

Update Configuration: Update the current obtain the configuration file from your updated
configuration through configuration files. server if it is configured. To obtain the original
configuration file, you can use the FTP/TFTP back
Note: Clearing the configuration in admin mode
up as described above. Put the configuration file
restores all settings to factory default. Clearing
in the root directory of the updated server when
the configuration in guest modem restores
finished editing.
all settings to factory default except SIP and
advanced SIP. 3.7.2 AUTOPROVISION FIRMWARE
UPDATE
3.7 Update Firmware
3.7.1 UPDATE CONFIGURATION

Current Config Version: The system displays the

current version number.
Web Update: Updates the application or Server Address: The FTP/TFTP server address.
configuration files of the phone. The telephone’s
firmware is in a .z format and the configuration Username: The FTP server username.
files are in a .txt format. Password: The FTP server password.
Click the “Browse” button (left of Update) to Config File Name: The name of the configuration
open the firmware (.z) file or the configuration file.
file xxx.3300IP.txt), then click on the “Update”
Config Encrypt Key: The encryption key of the
button. After the upgrade, the 9600 Series phone
confirmation file.
will automatically restart.
Protocol Type: The protocol type used for
FTP Update: Upload or download the configuration
upgrading.
file with an FTP or TFTP server. Or download
firmware from an FTP or TFTP server. Back up Update Interval Time: The interval time that the
the configuration file to your FTP/TFTP server. terminals search for a new configuration file.
The configuration file uses a .txt extension. Update Mode: The auto provision mode. Disable:
The “Type” includes two parts: a configuration no automatic update. Update after reboot:
file export and a configuration file import. automatically updates after a reboot. Update
at time interval: automatically updates after a
Configuration File Export: Exports the
certain time.
configuration file.
Configuration file version is found in the top
Configuration File Import: Imports the
section of the .txt file in the <<VOIP CONFIG
configuration file.
FILE>> Version 1.0007 and <GLOBAL CONFIG
The 9600 Series phone supports FTP and TFTP MODULE> ConfFile Version.
auto updating. The gateway will automatically

9600 SERIES VoIP USER GUIDE   www.telematrix.net 19

For instance: and we recommend using the free software
Wireshark for capturing network data packets
Gateway original version is:
instead of Syslog. Warning: if the power is cycled
<<VOIP CONFIG FILE>>Version: 1.0000 on the phone, Syslog is set back to its default
<GLOBAL CONFIG MODULE> ConfFile Version: 6 “off” state.
User may edit the configure file version to: Click “Apply” after setting this up.
<<VOIP CONFIG FILE>>Version: 1.0007
<GLOBAL CONFIG MODULE> ConfFile Version: 7 3.8.3 PHONE BOOK
Cetis VoIP phones have a standard “base”
3.8 System Manage firmware that is used across several models. Not
all of the models have the same functionality, but
3.8.1 ACCOUNT MANAGE
the firmware may still show and be present in
Set the Web access account or keypad password the User Interface (UI). Some functions may be
for the 9600 Series phone. inappropriate for hospitality environments, or not
show up in a phone without a screen, but useful
in commercial applications or display sets. Some
of those examples are below.

3.8.2 SYSLOG CONFIGURATION

Sets the system log. Phone Book is not used in Hospitality
environments, but remains in the universal
firmware set for commercial IP phones.

3.8.4 TIME SET—VIA SIMPLE NETWORK

TIME PROTOCOL (SNTP)
This page configures setting the time on the
phone.

Server IP: Sets the Syslog server address.

Server Port: Sets the Syslog server port.
MGR Log Level: Sets the MGR log level.
SIP Log Level: Sets the SIP log level.
IAX2 Log Level: Sets the IAX2 log level.
Enable Syslog: Turns the system log information
capture on. This information is highly technical

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Server: Enter the IP address of the SNTP time
server. 4. Operating Method
Time Zone: Select the correct time zone from the
list in the box.
for Dialing
Time Out: The longest response time for SNTP
server. 4.1 How to Dial an IP Phone
12 Hours Format: Select the 12 hours format. You can make a call after your phone is properly
set up. Please confirm whether all the network
SNTP: Select the SNTP server. wires are connected correctly, and the phone
Daylight: Whether Daylight Saving Time is being credentials have been added (under VOIP section)
used. to register successfully to the IP/PBX server.
Manual Configuration: Set up the date and time To make a call, dial the number and then press
here instead of using a time server. “#”.
You can find the phone’s IP address by dialing
3.8.5 MMI CONFIGURATION
* * 4(I) 7(P) #.
Set the greeting information on the LCD display
Input the IP address of the 9600 Series phone in the
through the Man-Machine-Interface (MMI).
browser, press the Enter key, and then you can visit
the settings layout of the 9600 Series phone. The
administrator’s account is admin/admin; the
common user account is guest/guest.

4.2 Set Up the Phone to

3.8.6 LOGOUT AND REBOOT SYSTEM
Register to a SIP Server
4.2.1 SET THE WAN INTERFACE
Here are the connection ways of entering the
Network -> WAN Configuration layout in the
phone’s net port:
The 9600 Series phone can be connected to the
Internet by using the static IP, DHCP IP, or PPPoE
dialing.
Logout: Logout the Web entry.
Reboot Phone: Logout and reboot the phone.
Whenever a user modifies a configuration parameter
of the phone that will take effect after the phone is
rebooted, the user enters into this screen and clicks
“Reboot”. The phone reboots automatically.
Note: About rebooting an IP phone: some settings
require a reboot in order for the new settings to
work. Always save the configuration before the
reboot, otherwise the settings will return to the
previous settings.

9600 SERIES VoIP USER GUIDE   www.telematrix.net 21

CONFIGURE STATIC IP

– Choose Static
– Fill in the IP address of the 9600 Series phone
in the Static IP Address field Server Address: This is the IP address or FQDN
– Fill in the subnet mask in Netmask (Fully Qualified Domain Name) of the IP/PBX SIP
– Fill in the router address or up Gateway Registration server.
address in the Gateway Server Port: This is the Internet Socket Port for
– Fill in the local DNS server address in the SIP information. The common UDP port for a SIP
Primary DNS and Alter DNS respectively server is 5060. Default port is 5060.
USE THE CONFIGURATION TO DYNAMICALLY Account Name: Username of your SIP account.
OBTAIN IP TO GET THE IP ADDRESS (Often the same as the phone number.)
– Choose the DHCP option Password: Password of your SIP account.
Now, if the network has a DHCP server, then the Phone Number: Phone number of your SIP
9600 Series phone will receive the IP Address, account.
Netmask, Gateway, Primary DNS, and Alter DNS
Enable Register: Choose Enable Register to
from this DHCP server automatically.
allow the phone to begin the SIP registration
USE PPPOE DIALING TO CONNECT THE Call/Response process.
INTERNET Display Name: This field will display the User
Agent (UA) in the header.
Proxy Server Address: (usually left blank)—
IP address of proxy SIP server. (A SIP provider
usually uses the same IP for the register server
and the proxy server. In this case you don’t need
– Choose the PPPoE option to configure the proxy server information.)
– Fill in the account and the password that
Proxy Server Port: (usually left blank) Signal
PPPoE dialed in the PPPoE Username and
port of the SIP proxy.
Password areas
Proxy Username: (usually left blank) Proxy
So the 9600 Series phone could connect to the
server username.
Internet through PPPoE dialing and automatically
receive the IP Address, Netmask, Gateway, Primary Proxy Password: (usually left blank) Proxy server
DNS, and Alter DNS information, and so on. password.
Domain Realm: (often left blank) SIP domain—
4.2.2 SIP CONFIGURATION
enter the SIP domain if any. Otherwise, the
Enter into the VoIP -> SIP Configuration window 9600  Series phone will use the proxy server
to set up the SIP account information: address as the SIP domain.
22 www.telematrix.net   9600 SERIES VoIP USER GUIDE
Enable Message Waiting: (usually enabled)—The phone should set VLAN Enabled and the Voice
configuration allows or forbids Message Waiting. VLAN ID. If necessary the LAN should be set to
Bridge Mode also.
Note: Choose Save Configuration in the
Configuration Management window after setting Show Firmware Version—**39# (**FW#)
up the new information or the existing setting Description: When **39# is entered, the phone
information will be reset when the phone is will display or audibly respond with the firmware
rebooted. version number such as “1.7.240.235”.
Show TFTP Server Address—**83# (**TF#)
Description: When entered, the phone responds
Reference—Quick with the IP address of the TFTP server. This
should be the TFTP server address actually being
Keys used. It might come from the configuration file,
option 66, option 125, or option 43.
Re-Program New ConfigID—
The TeleMatrix family of SIP phones has **77*password*ConfigID# (**RP*xxx*yyyy#)
implemented an innovative way to determine Description: After entering **77*, the user must
various states of the SIP phone and the network it enter the Keypad Password followed by a *
is attached to via the phone’s dial-pad. Press the then the ConfigID of the new configuration file,
* key twice in succession followed by a series of followed by #. For example, if the password in the
numbers shown below (the numbers are generally existing configuration on the phone is 123, and
matched to the command’s corresponding letters). the new configuration file is 1234.3300ip.txt, then
Many of these features are newly added in the user would enter **77*123*1234#. The user
Firmware 1.8.0 and higher. should NOT be prompted for the password. The
phone will then download the new configuration
Show IP Address—**47# (**IP#)
file 1234.3300ip.txt.
Description: Keying **47# causes the phone to
read out the IP address currently assigned. If the Reset Phone to Factory Default—
phone has an LCD panel, it should be displayed **33*<password># = (**DE*123#)
there. If the phone has no display panel, or the Description: After entering **33*123#, the phone
address cannot be displayed, it should be read out will reset to the factory default, which implies that
audibly on the speaker (or through the handset if there is no current programming information, so
it is off-hook). the phone will prompt for a ConfigID at reboot.
Show VLAN—**85# (**VL#) Re-boot Phone (without reset)—**72# (**RB#)
Description: When this is entered the phone Description: After entering 72#, the phone will
responds with the Voice VLAN ID. If the VLAN is reboot.
disabled “Zero” to indicate “off” is spoken. SIP TeleWorker Configuration File Download –
Set VLAN—**87# (**87*password*VLANid#) **89*<tftp server IP address>*<ConfigID>#
Description: This allows the user to set the Description: To download a configuration file
VLAN ID. The Keypad Password is also required. for a phone remotely use this **TW code to
As an example: to set the VLAN ID when the designate a TFTP server IP address and ConfigID
password is 9382 and the VLAN ID is 30, enter to download.
“**87*123*301#”. When a VLAN ID is set, the

9600 SERIES VoIP USER GUIDE   www.telematrix.net 23

Reference—SIP Quick The Server Address of the IP/PBX, Extension
Number/Account Name, Phone Number, Display

Start to Register a Name, Enable Register, and Enable Message

Waiting indicators are all programmed here.

Single Phone Speed Dial and OneTouch Message Keys Are

Programmed Under Advance -> Memory Key

The Cetis, Inc., SIP phones are SIP-only endpoints

powered only by POE/802.3af and will acquire an
IP address when connected to a DHCP server.
Begin by plugging your POE Ethernet cable into
the WAN port. To retrieve an IP address, press
* * 4 7 # (* * I P #). The phone’s IP address will
be displayed on the LCD or spoken by audio
through the speaker. Point your Web browser to
that IP address. You will be presented with a login
screen.
Login = admin Password = admin
MWI Number is the field for programming the red
This is the Default Current Status Page
Message Waiting Indicator OneTouch.
In this example, it is programmed to dial number
7001 on the IP/PBX for retrieving voice mail
messages.

Reset to Factory Default

To reset the phone to the factory default settings,
The scope of this document is to provide the basic press “#” during the startup procedure. (At power
screenshots necessary to quickly configure the up you can see a black progress bar.) The phone
guest phone for basic registration and show the will enter into “post” mode, then input * # 1 6 8.
user where to program the voice mail retrieval Then you will see “clearing conf” on the screen,
keys and the speed dials. In our example, the SIP next you see “conf reset”. Now you have a reset to
phone is programmed as extension 1001. The the default settings. Lastly, cycle power (remove
firmware version of the phone is circled in red. and replace WAN power cable) to restart the
Program Your Connection Under the VoIP -> SIP phone with factory default settings.
Configuration Screen

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Reference—DECT more handsets (LEAVE BATTERIES IN ALL
HANDSETS).

Handset Base Only 5 total handsets can be registered to a base.

When a sixth one tries to register to the base it
Registration / fails. The next procedure deletes the handset
registrations so that you can start again.
Deregistration Delete Handset Registrations
from DECT-SIP Base
The VoIP-DECT cordless phone models have a base
Press and hold down the submerged (right side
station that is powered via Power Over Ethernet (POE)
button under faceplate) FLASH key on the base
802.3af. The max power consumption is 3.5W, thus
unit for three seconds. The speaker will give a
falling into the POE class spectrum of Class 2. The
single beep, then
auxiliary handset stations for E Series are POE, while
the 9600 series and I Series NDC sets are powered Press “0” on the base unit to delete all the
via standard electrical plugs. All newer models (2011 handsets.
and beyond) will auto-sync by placing the handset OR
in the charging cradle of the SIP base station. Press “1” on the base unit to delete the
Other older models require the manual procedure first handset.
described below. ALL models can register handsets Press “2” on the base unit to delete the
via the manual method described here. second handset.
Press “3” on the base unit to delete the
Register One Handset third handset.
1. Plug the DECT SIP base into a Power Over Press “4” on the base unit to delete the
Ethernet (POE) layer 2 switch. fourth handset.
2. Wait for ConfigID tone-prompt. Press “5” on the base unit to delete the
3. Enter ConfigID and # to retrieve file from fifth handset.
TFTP server OR press # to bypass ConfigID.
Note: After pressing the number keys, the charge
4. Press Speaker key on base to confirm dial tone.
indicator light will flash, which means the handset
5. Press Speaker key again to hang up.
is being deleted by the phone. When the charge
6. Install battery in remote handset.
indicator light stops flashing, it means that the
7. On the DECT-SIP base—press and hold
deletion is finished.
Locate or Page key until charging light blinks.
8. On handset—press and hold the * key.
Changing Handset Ring Tone:
Listen for short beep and continue holding
the * key until you hear the second longer 6 Ring Tones to Choose From
sing-song series of beeps. Hold FNC/FLASH key on handset for 2 seconds
9. On handset—press On/Off key and listen to then release. Press any key 1~6 (on handset) to
confirm you have dial tone. preview the handset ringtone—1 is default tone.
Press FLASH to exit selection state and save any
Register Up To 4 Additional of those 1~6 tones. Lower the ring volume on the
Handsets side of the handset.
It is possible to register up to 4 remote handset
kits to the base. Repeat this process for three
9600 SERIES VoIP USER GUIDE   www.telematrix.net 25
Important Safety Instructions voltages or other risks. Incorrect
reassembly can cause electric shock when
When using your telephone equipment, basic the appliance is subsequently used.
safety precautions should always be followed to 9. Unplug this product from the wall outlet
reduce the risk of fire, electric shock, and injury and refer servicing to qualified service
to persons, including the following: personnel under the following conditions:
1. Read and understand all instructions. – When the power supply cord or plug is
2. Follow all warnings and instructions marked damaged or frayed.
on the product. – If liquid has been spilled into the
3. Unplug the product from the wall outlet product.
before cleaning. Do not use liquid cleaner – If the product has been exposed to rain
or aerosol cleaners. Use a damp cloth for or water.
cleaning. – If the product does not operate normally
4. Do not use this product near water—for by following the operating instructions.
example, near a bathtub, wash bowl, Adjust only those controls that are
kitchen sink or laundry tub, in a wet covered by the operating instructions, as
basement, or near a swimming pool. improper adjustment of other controls
5. Do not place this product on an unstable may result in damage and may require
cart, stand, or table. The product may fall, extensive work by a qualified technician
causing serious damage to the product. to restore the product to normal
6. Slots and openings in the cabinet and operation.
the back or bottom are provided for – If the product has been dropped or the
ventilation, to protect it from overheating. cabinet has been damaged.
These openings must not be blocked or – If the product exhibits a distinct change
covered. The openings should never be in performance.
blocked by placing the product on the bed, 10. Avoid using a telephone (other than a
sofa, rug, or any other similar surface. cordless type) during an electrical storm.
This product should never be place near There may be a remote risk of electric
or over a radiator or heat register. This shock from lightning.
product should not be placed in a built-in 11. Do not use the telephone to report a gas
installation unless proper ventilation is leak in the vicinity of the leak.
provided. PLEASE SAVE THESE INSTRUCTIONS.
7. Never push objects of any kind into this
product through cabinet slots as they may
touch dangerous voltage points or short FCC Interference Information
out parts that could result in a risk of fire This device complies with Part 15 of the FCC
or electric shock. Never spill liquid of any Rules. Operation is subject to the following two
kind on the product. conditions:
8. To reduce the risk of electric shock do
1. This device may not cause harmful
not disassemble this product. Take it to a
interference.
qualified service facility if service or repair
2. This device must accept any interference
work is required. Opening or removing
received, including interference that may
covers may expose you to dangerous
cause undesired operation.

26 www.telematrix.net   9600 SERIES VoIP USER GUIDE

This equipment has been tested and found to Industry of Canada Requirements
comply with the limits for a Class B digital device,
pursuant to Part 15 of the FCC Rules. These limits Note: This equipment meets the applicable
are designed to provide reasonable protection Industry Canada Terminal Equipment Technical
against harmful interference in a residential Specifications. This is confirmed by the
installation. This equipment generates, uses, registration number. The abbreviation, IC, before
and can radiate radio frequency energy and, if the registration number signifies that registration
not installed and used in accordance with the was performed based on a Declaration of
instructions, may cause harmful interference Conformity indicating that Industry Canada
to radio communications. However, there is no technical specifications were met. It does not imply
guarantee that interference will not occur in a that Industry Canada approved the equipment.
particular installation. Before installing this equipment, users should
If this equipment does cause harmful interference ensure that it is permissible to be connected to
to radio or television reception, which can be the facilities of the local telecommunications
determined by turning the equipment off and on, the company. The equipment must also be installed
user is encouraged to try to correct the interference using an acceptable method of connection. The
by one or more of the following measures: customer should be aware that compliance with
the above conditions may not prevent degradation
– Reorient or relocate the receiving antenna of service in some situations.
for the radio or television that is receiving the
interference). Repairs to certified equipment should be
– Reorient or relocate and increase the coordinated by a representative designated by the
separation between the telecommunications supplier. Any repairs or alterations made by a user
equipment and receiving antenna. to this equipment, or equipment malfunctions,
– Connect the telecommunications equipment may give the telephone communications company
into an outlet on a circuit different from that to cause to request the user to disconnect the
which the receiving antenna is connected. equipment.
Users should ensure for their own protection, that
FCC RF Radiation Exposure the electrical ground connections of the power
Statement utility, telephone lines, and internal metallic
water pipe system, if present, are connected
The installation of the base unit should allow
together. This precaution may be particularly
at least 20 centimeters between the base
important in rural areas.
and persons to be in compliance with FCC RF
exposure guidelines. For body-worn operation, Caution: Users should not attempt to make such
the portable part (handset) has been tested and connections themselves, but should contact
meets FCC RF exposure guidelines. the appropriate electric inspection authority, or
electrician, as appropriate.
This device must not be co-located or operating
in conjunction with any other antenna or Notice: The Ringer Equivalence Number (REN)
transmitter. The changes or modifications not assigned to each terminal device provides an
expressly approved by the party responsible for indication of the maximum number of terminals
compliance could void the user’s authority to allowed to be connected to a telephone interface.
operate the equipment. The termination on an interface may consist

9600 SERIES VoIP USER GUIDE   www.telematrix.net 27

of any combination of devices subject only to Requirements of Part 68—
the requirement that the sum of the Ringer
Equivalence Numbers of all the devices does not
FCC Rules
exceed 5. This equipment complies with Part 68 of the FCC
Rules and the requirements adopted by ACTA.
REN: Z
On the bottom of this telephone is a label that
For warranty and service in Canada, please contains, among other information, a product
contact: identifier in the format US:AAAEQ##TXXXX. If
Williams Telecommunications requested, this number must be provided to
5610 Kennedy Road the telephone company. The USOC Jack for this
Mississauga, Ontario, L4Z2A9 equipment is RJ11C.
Canada A plug and jack used to connect this equipment
Phone: 905-712-4242 to the premises wiring and telephone network
Fax: 905-712-1754 must comply with the applicable FCC Part 68
rules and requirements adopted by ACTA. A
Requirements of Part 15— compliant telephone cord and modular plug are
FCC Rules provided with this telephone. It is designed to be
connected to a compatible modular jack that is
Note: This equipment has been tested and found to
also compliant. See installation instructions for
comply with the limits for a Class B digital device,
details.
pursuant to Part 15 of the FCC Rules. These limits
are designed to provide reasonable protection The Ringer Equivalence Number (REN) is used
against harmful interference in a residential to determine the number of devices that may be
installation. This equipment generates, uses, connected to a telephone line. Excessive RENs
and can radiate radio frequency energy and, if on a telephone line may result in the devices
not installed and used in accordance with the not ringing in response to an incoming call. In
instructions, may cause harmful interference most but not all areas, the sum of RENs should
to radio communications. However, there is no not exceed five (5.0). To be certain of the number
guarantee that interference will not occur in a of devices that may be connected to a line, as
particular installation. If this equipment does determined by the total RENs, contact the local
cause harmful interference to radio or television telephone company. For products approved after
reception, which can be determined by turning July 23, 2001, the REN for this product is a part
the equipment off and on, the user is encouraged of the product identifier that has the format
to try to correct the interference by one or more US:AAAEQ##TXXXX. The digits represented by
of the following measures: ## are the REN without a decimal point (e.g., 03
is a REN of 0.3). For earlier products, the REN is
1. Move the telephone away from the
separately shown on the label.
receiver.
2. Consult the dealer or an experienced If this telephone causes harm to the telephone
radio/TV technician for help. network, the telephone company will notify you
in advance that temporary discontinuance of
Any changes made by the user not approved by
service may be required. But if advance notice is
the manufacturer can void the user’s authority to
not practical, the telephone company will notify
operate the telephone.

28 www.telematrix.net   9600 SERIES VoIP USER GUIDE

the customer as soon as possible. Also, you will These telephone devices are intended for
be advised of your right to file a complaint with commercial use only, primarily in hotel guestrooms.
the FCC if you believe it is necessary. They must be used with a PBX (private branch
exchange), and are not intended to be connected
The telephone company may make changes in its
directly to a PSTN line (public switched telephone
facilities, equipment, operations, or procedures
network). There are no user-serviceable parts
that could affect the operation of the equipment.
inside the equipment.
If this happens, the telephone company will
provide advance notice in order for you to
make the necessary modifications to maintain Technical Specifications
uninterrupted service. DIMENSIONS
If trouble is experienced with this equipment, for 9.5” (w) x 6.75” (l) x 3.5” (h)
repair or warranty information, please contact
TeleMatrix at (800) 462-9446. If the equipment WEIGHT
is causing harm to the telephone network, the With handset: 36.6 lbs.
telephone company may request that you disconnect
the equipment until the problem is resolved. User Guides
There are no user-serviceable parts contained in If additional User Guides are needed, please go
this equipment. to www.telematrix.net to download the PDF.
Connection to party line service is subject to
state tariffs. Contact the state public utility Service
commission, public service commission, or When problems arise that cannot be resolved
corporation commission for information. using this or related documents, please go
If your home has specially wired alarm equipment to www.telematrix.net for information about
connected to the telephone line, ensure the customer support, technical support, warranty,
installation of this product does not disable your and product returns.
alarm equipment. If you have questions about
what will disable alarm equipment, consult your Statement of Limited Warranty
telephone company or a qualified installer. TeleMatrix product warranty information is
This telephone is hearing aid compatible. available at www.telematrix.net.

9600 SERIES VoIP USER GUIDE   www.telematrix.net 29

Toll Free: +1.800.462.9446
Tel: +1.719.638.8821

Email: info@telematrix.net
www.telematrix.net

© 2015 Cetis, Inc. Product specifications and descriptions in this document subject to change without notice. CetisTM, Teledex®, TeleMatrix®, and Scitec®
are trademarks or registered trademarks of Cetis, Inc. TMX-9600VoIP-UG-122015