GSC35XX User Manual
GSC35XX User Manual
GSC35XX Series
User Manual
COPYRIGHT
CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this
product in any way other than as detailed by this guide, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with devices as it may cause damage to the products and void
the manufacturer warranty.
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FCC Caution
Any Changes or modifications not expressly approved by the party responsible for compliance could void
the user's authority to operate the equipment. This device complies with part 15 of the FCC Rules.
Operation is subject to the following two conditions: (1) This device may not cause harmful interference,
and (2) this device must accept any interference received, including interference that may cause undesired
operation.
Note: This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against
harmful interference in a residential installation. This equipment generates uses and can radiate radio
frequency energy and, if not installed and used in accordance with the instructions, may cause harmful
interference to radio communications. However, there is no guarantee that interference will not occur in a
particular installation. If this equipment does cause harmful interference to radio or television reception,
which can be determined by turning the equipment off and on, the user is encouraged to try to correct the
interference by one or more of the following measures:
- Reorient or relocate the receiving antenna.
- Increase the separation between the equipment and receiver.
- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
- Consult the dealer or an experienced radio/TV technician for help.
This device is complied with SAR for general population/uncontrolled exposure limits in ANSI/IEEE C95.1-
1992, and has been tested in accordance with the measurement methods and procedures specified in OET
Bulletin 65 Supplement C. This device has been tested and meets the FCC RF exposure guidelines when
tested with the device directly contacted to the body. RF exposure compliance with anybody-worn
accessory, which contains metal, was not tested and certified, and use such body-worn accessory should
be avoided.
Changes or modifications not expressly approved by the party responsible for compliance could void the
user’s authority to operate the equipment.
Hereby,Grandstream declares that this device is in compliance with the essential requirements and other
relevant provisions of Directive 1999/5/EC.
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CE Authentication
Hereby, Grandstream Networks, Inc. declares that the radio equipment GSC3510/GSC3505 is in
compliance with Directive 2014/53/EU.
The full text of the EU declaration of conformity is available at the following internet address:
http://www.grandstream.com/support/resources/
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GNU GPL INFORMATION
GSC3510/GSC3505 firmware contains third-party software licensed under the GNU General Public License
(GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General
Public License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
http://www.grandstream.com/sites/default/files/Resources/GSC35XX_gpl.zip
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Table of Contents
WELCOME ................................................................................................................... 14
Equipment Packaging.............................................................................................................................. 17
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Multicast Paging Application .................................................................................................................... 27
Call ....................................................................................................................................................... 48
Call History ........................................................................................................................................... 49
Call History All ............................................................................................................................................. 50
Call History Intercept Record ....................................................................................................................... 52
Web GUI Missed Call Notification support....................................................................................................... 53
Contacts ............................................................................................................................................... 54
Contacts List ..................................................................................................................................................... 54
Group ............................................................................................................................................................... 58
Black/White List Settings ..................................................................................................................... 58
Whitelist ........................................................................................................................................................... 59
Blacklist ............................................................................................................................................................ 60
Blocking Rules .................................................................................................................................................. 61
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Ethernet Settings ................................................................................................................................. 65
Bluetooth .............................................................................................................................................. 67
Wi-Fi Settings ....................................................................................................................................... 67
Connect to Wi-Fi Network ................................................................................................................................ 67
Wi-Fi Settings description ................................................................................................................................ 69
OpenVPN® Settings ............................................................................................................................ 70
Advanced Network Settings ................................................................................................................. 71
Upgrade ............................................................................................................................................... 77
System Diagnosis ................................................................................................................................ 81
Event Notification ................................................................................................................................. 83
LDAP Book........................................................................................................................................... 84
Recording ............................................................................................................................................. 85
Music .................................................................................................................................................... 86
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Table of Tables
Table of Figures
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Figure 30: Click-to-Dial Feature .................................................................................................................. 49
Figure 31: Outgoing call in progress and accepted .................................................................................... 49
Figure 32: Call History → All ........................................................................................................................ 50
Figure 33: Call details under Call History → All ........................................................................................... 51
Figure 34: Add number from call history to an existing contact .................................................................. 51
Figure 35: Call History → Intercept Record ................................................................................................. 52
Figure 36: Call details under Call History → Intercept Record .................................................................... 53
Figure 37: Web GUI Missed Call Notification .............................................................................................. 53
Figure 38: Contacts → Contacts List ........................................................................................................... 54
Figure 39: Add New Contact ....................................................................................................................... 54
Figure 40: Contacts → Group ...................................................................................................................... 58
Figure 41: Add New Group.......................................................................................................................... 58
Figure 42: Whitelist section ......................................................................................................................... 59
Figure 43: Add phonebook contacts to whitelist ......................................................................................... 59
Figure 44: Add blocked numbers to whitelist .............................................................................................. 60
Figure 45: Add Manually to Whitelist ........................................................................................................... 60
Figure 46: Blacklist Section ......................................................................................................................... 60
Figure 47: Add from Call History to Blacklist ............................................................................................... 61
Figure 48: Wi-Fi Basics Page...................................................................................................................... 68
Figure 49: Connect to Wi-Fi Network .......................................................................................................... 68
Figure 50: GSC3510/GSC3505 Connect to Wi-Fi-Show Advanced Options .............................................. 69
Figure 51: GSC3510/GSC3505 Backup Page ............................................................................................ 76
Figure 52: Backup content selection ........................................................................................................... 76
Figure 53: Generated Backup ..................................................................................................................... 77
Figure 54: Device Detection - Audio Loop Test ........................................................................................... 87
Figure 55: Device Detection - Built-in Speaker Test ................................................................................... 87
Figure 56: Device Detection - LED Test ...................................................................................................... 88
Figure 57: Device Detection - Reset Button Test ........................................................................................ 89
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DOCUMENT PURPOSE
This document describes how to configure the GSC3510 via web UI menu to fully manipulate device's
features. Please visit http://www.grandstream.com/support to download the latest “GSC3510 User Manual”.
Product Overview
Getting Started
Hardware Installation
o Wall Mount
o Ceiling Mount
o Anti-theft Installation
o Bluetooth Speaker
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CHANGE LOG
This section documents significant changes from previous versions of user manual for GSC35XX Series.
Only major new features or major document updates are listed here. Minor updates for corrections or editing
are not documented here.
No major changes.
No major changes.
Added ability to generate a zip file for multiple recording files. [Record]
Added the missed calls notification on Web UI. [Missed Call Notification]
Optimized web UI tooltips.
Optimized the sensor functions of the device.
Added the incoming rules options for each account [Incoming Call Rules]
Added to support failover mechanism based on DNS SRV. [Register Before DNS SRV Fail-over]
Added the ability to delay the auto-answer. [Automatic Answer Ringing Time (s]
Added the ability to adjust the Busy Tone Expiration. [Busy Tone Ring Time (s)]
Added support for visibility timeout to nearby devices. [Bluetooth]
Added the option to Enable/Disable web access. [Security Settings]
Added Ability to disable IP2location and Wifi Certification. [Allow AutoConfig service access]
Added the ability to play/stream the music based on set schedule. [Music]
Added the ability to support MP3 Format files on the Music Feature. [Music]
Added the ability to display date/time on the Web UI. [System Info]
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Added an ability to adjust the volume during playing the radio/music
Changed “DATE” for the time display under call history and recording files to “TIME AND DATE”
Optimized web UI prompts
Optimized the audio quality
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WELCOME
Thank you for purchasing Grandstream GSC3510/GSC3505 SIP Intercom speakers. The GSC3505 is a
one-way SIP Intercom Speaker and GSC3510 is a two-way full-duplex SIP intercom speaker/microphone,
both featuring one 100Mbps Ethernet port with PoE/PoE+, integrated dual-band 2.4G/5G Wi-Fi, integrated
Bluetooth, high fidelity 8W speaker and a multi-purpose input port supporting a wide range of peripherals,
3 directional microphones with Multichannel Microphone Array Design (MMAD) available for GSC3510 only.
Both GSC3505 and GSC3510 with their Hi-Fi speaker delivers full-band audio, while GSC3510 adds a
state-of-art microphone array with pickup distance up to 4.2 meters. The built-in whitelist and blacklist
features enable easy filtering of unwanted calls from the Internet. Its modern industrial design and rich
features make it ideal for classrooms, hospitals, apartments, dormitories and much more.
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PRODUCT OVERVIEW
Feature Highlights
Up to 16 SIP accounts.
GSC3510
GSC3505 Ethernet RJ45 10/100Mbps, PoE/PoE+, Integrated Bluetooth, Wi-Fi.
Both GSC3505 and GSC3510 HD with their Hi-Fi speaker delivers full-band
audio, Hands-free speakerphone with HD acoustic chamber, advanced
acoustic echo cancellation, while GSC3510 adds a state-of-art microphone
array with pickup distance up to 4.2 meters.
The following table resumes all the technical specifications including the protocols / standards supported,
voice codecs, telephony features, languages and upgrade/provisioning settings for GSC3510/GSC3505.
Voice Codec G.711µ/a, G.722 (wide-band), G.722.1, G.722.1C, G.726-32, iLBC, Opus,
G.729A/B in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO).
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Telephony Features Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup,
downloadable contacts, call record, call log, auto answer, click-to-dial, flexible dial
plan.
QoS Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS.
Security User and administrator level passwords, MD5 and MD5-sess based authentication,
256-bit AES encrypted configuration file, TLS, SRTP, HTTPS, 802.1x media access
control.
Upgrade/ Firmware upgrade via TFTP / HTTP / HTTPS or local HTTP upload, mass
Provisioning provisioning using TR-069 or AES encrypted XML configuration file.
Package Content GSC3510/GSC3505, Metal Bracket, Plastic Bracket, Wiring Seat, Hang rope
plate, 3x Screw (PM 3 x 50), 3x Screw (PA 3.5 x 20), 1 x Screw (M3 x 15),
Hexagonal Screwdriver, 3 x Plastic Expansion Bolt, 3 x M3 NUT, Quick Installation
Guide, GPL license.
Physical Unit Dimensions: 215.32mm (diameter) x 68.7mm (depth)Unit Weight: 0.8kg, Box
Weight: 1.39k
Compliance FCC: Part 15 (CFR 47) Class B; UL 60950 (power adapter); FCC Part 15C, FCC
Part 15E, MPE
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GETTING STARTED
This chapter provides basic installation instructions including the list of the packaging contents and also
information for obtaining the best performance with the GSC3510/GSC3505.
Equipment Packaging
Table 3: Equipment Packaging
GSC3510/GSC3505
1x GSC3510/GSC3505 Main Case.
1x Metal Bracket.
1x Plastic Bracket.
Wiring Seat.
Hang rope plate.
3x Screw (PM 3x50)
3x Screw (PA 3.5 x20).
1x Screw (M3x15)
Hexagonal Screwdriver.
3x Plastic Expansion Bolt.
3x M3 NUT.
1x Quick Installation Guide.
1x GPL license.
Note: Check the package before installation. If you find anything missing, contact your system
administrator.
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GSC3510/GSC3505 Ports
1 2 3
Figure 2: GSC3510/GSC3505 Ports
The GSC3510/GSC3505 contains 4 types of colored LEDs (Red, Green, White and Blue light) that are used
in some specific situations and operations. Please, refer to the following table describing each one of the
LED Indicators status:
Green Light Slow Flashing (On 1s, Off 2s) Call on hold.
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White Light Fast Flashing (every 1s) Upgrading the firmware.
Hardware Installation
GSC3510/GSC3505 can be mounted on the wall or ceiling. Please refer to the following steps for the
appropriate installation
Wall Mount
1. Locate the equipment holder on the desired position with arrow up. Drill three holes on the wall
3. Align the position line on device’s back cover with the positioning slot.
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Ceiling Mount
1. Put the ceiling mounting (metal bracket) in the ceiling’s center and mark the position of the three
screws holes.
2. Drill a round hole with a diameter of 18mm for Ethernet cable. The distance between its center and
3. Fix the plastic and metal brackets on the ceiling with flat-head screws and locknuts. Then place
4. Align the position line on device’s back cover with the positioning slot.–––
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Anti-theft Installation
After the device is assembled with the metal bracket support on the wall or ceiling, use the anti-detachable
screw (M3x15) in order to prevent theft.
The GSC3510/GSC3505 can be powered on using PoE/PoE+ switch or PoE injector using following steps:
Step 1: Plug a RJ45 Ethernet cable into the network port of the GSC3510/GSC3505.
Step 2: Plug the other end into the power over Ethernet (PoE) switch or PoE injector.
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Connecting Wiring Seat
GSC3505/GSC3510 support to connect a “Key & LED” or “Normal Key” to the 2-pin port via Wiring Seat
using following steps:
Step 2: Connect the “Key & LED” or “Normal Key” with the wiring seat (as shown in the figure
below)
Note: This port supports the parallel connection of an incandescent lamp (with less than 1W) or an LED
lamp (with less than 100mA).
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Figure 14: GSC3510 Web GUI – Login
Users can use a computer connected to the same network as the GSC3510/GSC3505 to discover and
access the GSC3510/GSC3505 Configuration Interface using its MAC Address.
Please, refer to the following steps in order to access the GSC3510/GSC3505 Web GUI:
1. Locate the MAC address on the MAC tag of the unit, which is on the underside of the device, or on
the package.
2. From a computer connected to same network as the GSC3510/GSC3505, type in the following
address using the GSC3510/GSC3505’s MAC address on your browser: https://gsc_<mac>.local
Example: if a GSC3510/GSC3505 has the MAC address C0:74:AD:xx:xx:xx, this unit can be accessed by
typing https://gsc_c074adxxxxxx.local on the browser.
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GSC3510/GSC3505 APPLICATION SCENARIOS
GSC3510/GSC3505 can be used as an Intercom System using built-in SIP accounts, once the SIP account
registered the device can receive paging/intercom calls and it will automatically answer calls coming from
whitelisted numbers.
To register a SIP account on the GSC3510/GSC3505 the user needs to go under Account Account X
General Settings, and enter the account information as below, then save and apply the configuration.
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Figure 16: SIP Account Configuration
Once the account registered correctly, the GSC3510/GSC3505 will show the account status as
“Registered” under Status Account Status.
By default, the GSC3510/GSC3505 Blocks non-whitelisted number under Calls Black/White List
Settings Blocking Rules, user needs to either allow Non-White list calls or to set up a Whitelist that
contains the number that will be allowed to call the GSC3510/GSC3505.
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Figure 18: Default Blocking Rules
As soon as a SIP call is received by the GSC3510/GSC3505, it first checks if the Caller ID number is
allowed on the Whitelist and then answers automatically.
Notes:
By default, GSC3510/GSC3505 plays a Warning tone when auto answering incoming calls, this
warning tone can be disabled under Account Account X Call Settings, “Play Warning Tone
for Auto Answer Intercom”.
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Multicast Paging Application
Multicast paging is an approach to let different SIP users to listen for paging calls from a common multicast
IP address. In multicast page call, an audio connection will be set up from sender to receiver, but the
receiver will be only able to receive audio, a one-way communication. The 2 entities, Sender/Receiver, must
be located on same LAN (same broadcast domain).
To receive multicast page, GSC3510/GSC3505 must be well configured to listen to the right address and
port. The configuration is located under Phone Settings Multicast Paging. Up to 10 listening addresses
are supported with priority levels from 1 to 10.
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Figure 21: Multicast Paging Listening Addresses
In above screenshot, Listening Address “237.11.10.11:6767” with label “Sales” has the highest priority.
Users can enable “Paging Priority Active” option (under Multicast Paging tab) to accept incoming paging
calls during active multicast paging. The paging call with higher priority than active one will be accepted.
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In the case of receiving a multicast paging call while on a unicast SIP call, the GSC3510/GSC3505 can
choose to either keep the SIP call or to hold this last and allow the multicast call depending on paging call
priority.
This is can be set using “Paging Barge” option. If the option is set to “Disabled” all incoming multicast paging
calls will be dropped while on a SIP call. If the multicast paging call has higher priority than the value set on
“Paging Barge”, the SIP call will be put on hold and GSC3510/GSC3505 will the incoming multicast paging.
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Bluetooth Speaker
The GSC3510/GSC3505 can be used as a Bluetooth speaker for another device and it needs to be
connected via Bluetooth to that device. Users need to turn on GSC3510/GSC3505’s Bluetooth function
first. The first time when using a new Bluetooth device with the GSC3510/GSC3505, "pair" the device with
GSC3510/GSC3505 so that both devices know how to connect securely to each other.
Please, refer to the following steps in order to pair and connect the GSC3510/GSC3505 to the device:
2. Enable “Bluetooth Settings” function, and enable the option “Visible to Nearby Bluetooth Device” in
order to make the GSC3510/GSC3505 visible for other devices for 2 minutes (When the 2 minutes
are achieved, and you still didn’t connect it, please enable again the option to keep it visible)
3. Go to your Device’s Bluetooth settings I order to search for visible devices. The
GSC3510/GSC3505 is going to be listed within the visible devices with the “Device Name”
configured on the Web GUI.
4. Click on the GSC3510/GSC3505 device’s name in order to pair and connect it to the device
Note: The GSC3510/GSC3505 will only play the role of a speaker when it is connected to another device
via Bluetooth. Users cannot use the GSC3510/GSC3505 to take control of calls made/received by the
device connected to it.
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2-pin Multi-Purpose Input Applications
GSC3510/GSC3505 supports 2-pin multi-purpose input that can connect a “Key with LED” or “Normal Key”.
By configuring the sensor settings users can enable the GSC3510/GSC3505 to play an audio file
(.wav/.mp3 format), trigger a SIP call to a pre-configured extension, or to start recording audio locally when
triggered.
Voice prompt
Play sound when trigger switch,
support upload custom ring/alarm sound
Call dial
Dial the configured number when switch is triggered
Call recording
Record with one key when the switch is triggered
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Under Basic Setting section, users can set “Sensor Type” and “Trigger Type”.
Two states are supported by the Input circuit for the “Sensor Type”:
1. Edge Triggered: When selected, the notification is triggered only when the level changes (high
level to low level, or low level to high level.
2. Level Triggered: When selected, only high level (1) will trigger the notification.
Under “Trigger time” section, users can click on “Add” in order to configure different schedules and a trigger
profile for each one as shown in the figure below:
Cycle Time: The alarm can be configured to be triggered all days of the week, in this case “All
days” option needs to be checked. Or to some specific days of the week with Start and End time,
in this case “Period of Time” option needs to be checked for users to be able to configure Time
and Frequency options.
Play Audio: If checked, GSC3510/GSC3505 will play a sound when the switch is triggered during
the schedule. Users can select a “Prompt Tone” from available tones or upload a customized tone.
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Figure 28: Sensor Setting - Linkage Function - Play Audio
Make Call: If checked, GSC3510/GSC3505 will dial out configured numbers on “Dial out extension”
fields (up to 2 numbers supported) when the switch is triggered during the schedule.
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GSC3510/GSC3505 WEB GUI SETTINGS
Account Status
Network Status
MAC Address Global unique ID of device, in HEX format. The MAC address will be used
for provisioning and can be found on the label coming with original box and
on the label located on the back of the device.
NAT Type Type of NAT connection used by the device.
Address Type Configured address type:
DHCP, Static IP or PPPoE.
IPv4 Address IP address of the device.
Subnet Mask Subnet mask of the device.
Default Gateway Default gateway of the device.
DNS Server 1 DNS Server 1 of the device.
DNS Server 2 DNS Server 2 of the device.
IPv6 Address Type Configured address type:
DHCP, Static IP or PPPoE.
IPv6 Address IPv6 address of the device.
IPv6 DNS Server 1 IPv6 DNS Server 1 of the device.
IPv6 DNS Server 2 IPv6 DNS Server 2 of the device.
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System Info
GSC3510/GSC3505 has 16 independent SIP accounts. Each SIP account has an individual configuration
page.
General Settings
Account Registration
Account Active Indicates whether the account is active. 1st account active by default.
Account Name Configures the name associated with each account.
SIP Server Specifies the URL or IP address, and port of the SIP server. This should
be provided by VoIP service provider (ITSP).
SIP User ID Configures user account information provided by your VoIP service
provider (ITSP). It's usually in the form of digits similar to phone number or
actually a phone number.
SIP Authentication ID Configures the SIP service subscriber's Authenticate ID used for
authentication. It can be identical to or different from the SIP User ID.
SIP Authentication Configures the account password required for the device to authenticate
Password with the ITSP (SIP) server before the account can be registered. After
saving, it will appear as hidden for security purpose.
Display Name Configures the subscriber's name (optional) that will be used for Caller ID
display.
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Tel URI Indicates E.164 number in “From” header by adding “User=Phone”
parameter or using “Tel:” in SIP packets, if the device has an assigned
PSTN Number.
Disabled: Will use “SIP User ID” information in the Request-Line and
“From” header.
User=Phone: “User=Phone” parameter will be attached to the
Request-Line and “From” header in the SIP request to indicate the
E.164 number. If set to "Enable".
Enabled: "Tel:" will be used instead of "sip:" in the SIP request.
Please consult your carrier before changing this parameter. Default is
“Disabled”.
Network Settings
Outbound Proxy Configures the IP address or the domain name of the primary outbound
proxy, media gateway or session border controller. It's used by the device
for firewall or NAT penetration in different network environments. If a
symmetric NAT is detected, STUN will not work and only an outbound
proxy can provide a solution
Secondary Outbound Sets IP address or domain name of the secondary outbound proxy, media
Proxy gateway or session border controller. The device will try to connect the
Secondary outbound proxy only if the primary outbound proxy fails.
DNS Mode Defines which DNS service will be used to lookup IP address for SIP
server’s hostname. There are three modes:
A Record
SRV
NATPTR/SRV
The device will always prefer to send SIP requests to the available server
having the lowest priority, and in case it’s down it contacts the next one,
but once the server having lowest priority is UP again, the device will switch
over to this one.
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Saved one until DNS TTL (Stay on responding IP until DNS
timeout):
On this mode, the device will resolve DNS SRV records and tries to send
the request to the server having lowest priority and if it doesn’t respond, it
will move on to the next IP until one of the servers responds, once this
happen the device will keep contacting this responding IP until DNS
timeout (30 minutes) before starting over.
On this mode, the device will send SIP requests to the last responding IP,
and it doesn’t failover/switchover to the next one until this responding
server is down.
Register Before DNS When this feature is enabled (default), the device will register using the
SRV Fail-over outbound proxy when the SIP server is failed to get the registration.
NAT Traversal Specifies which NAT traversal mechanism will be enabled on the device.
It can be selected from the dropdown list:
NAT NO
STUN
Keep-alive
UPnP
Auto
OpenVPN
If the outbound proxy is configured and used, it can be set to “NAT NO”.
If set to "STUN" and STUN server is configured, the device will periodically
send STUN message to the SUTN server to get the public IP address of
its NAT environment and keep the NAT port open. STUN will not work if
the NAT is symmetric type.
If set to “Keep-alive”, the device will send the STUN packets to maintain
the connection that is first established during registration of the device. The
“Keep-alive” packets will fool the NAT device into keeping the connection
open and this allows the host server to send SIP requests directly to the
registered phone.
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If it needs to use OpenVPN to connect host server, it needs to set it to
“VPN”. If the firewall and the SIP device behind the firewall are both able
to use UPNP, it can be set to “UPNP”. The both parties will negotiate to
use which port to allow SIP through. The default setting is "Keep-alive".
Proxy-Require Adds the Proxy-Required header in the SIP message. It is used to indicate
proxy-sensitive features that must be supported by the proxy. Do not
configure this parameter unless this feature is supported on the SIP server.
SIP Settings
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Add Auth Header On RE- Configure if the SIP account needs to add Auth header in RE-REGISTER.
REGISTER If the option is checked, device will always add authentication
header in REGISTER.
If the option is unchecked, device will only send authentication for
the first REGISTER.
Enable SIP OPTIONS Enables SIP OPTIONS to track account registration status so the device
Keep Alive will send periodic OPTIONS message to server to track the connection
status with the server. The default setting is "No".
SIP OPTIONS Keep Alive Configures the time interval when the device sends OPTIONS message to
Interval (s) SIP server. The default value is 30 seconds, in order to send an OPTIONS
message to the server every 30 seconds.
The default range is 1-64800.
SIP OPTIONS Keep Alive Configures the maximum times of sending OPTIONS message
Maximum Tries consistently from the device to server. Phone will keep sending OPTIONS
messages until it receives response from SIP server. The default setting is
"3", which means when the device sends OPTIONS message for 3 times,
and SIP server does not respond this message, the device will send RE-
REGISTER message to register again. The valid range is 3-10.
Use Privacy Header Controls whether the Privacy header will present in the SIP INVITE
message or not, whether the header contains the caller info. Do not use
the privacy header fields by default in Huawei IMS mode. If set to "Yes",
the Privacy Header will always show in INVITE. If set to "No", the Privacy
Header will not show in INVITE.
Use P-Preferred-Identity Controls whether the P-Preferred-Identity header will present in the SIP
Header INVITE message or not, whether the header contains the caller info. Do
not use the P-Preferred-Identity header fields by default in Huawei IMS
mode. If set to "Yes", the P-Preferred-Identity Header will always show in
INVITE. If set to "No", the P-Preferred-Identity Header will not show in
INVITE.
SIP Transport Determines which network protocol will be used to transport the SIP
message. It can be selected from TCP/UDP/TLS.
Default setting is "UDP".
Local SIP Port Determines the local SIP port used to listen and transmit. The default
setting is 5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066
for Account 4, 5068 for Account 5, and 5070 for Account 6. The valid range
is from 5 to 65535.
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SIP URI Scheme When Defines which SIP header, "sip" or "sips", will be used if TLS is selected
Using TLS for SIP Transport.
The default setting is "sip".
Use Actual Ephemeral Determines the port information in the Via header and Contact header of
Port in Contact with SIP message when the device use TCP or TLS. If set to No, these port
TCP/TLS numbers will use the permanent listening port on the device. Otherwise,
they will use the ephemeral port for the particular connection. The default
setting is "No".
Support SIP Instance ID Determines if the device will send SIP Instance ID. The SIP instance ID is
used to uniquely identify the device. If set to “Yes”, the SIP Register
message Contact header will include +sip.instance tag. Default is "Yes".
SIP T1 Timeout Defines an estimate of the round-trip time of transactions between a client
and server. If no response is received in T1, the figure will increase to 2*T1
and then 4*T1. The request re-transmit retries would continue until a
maximum amount of time define by T2. The default setting is 0.5 sec.
SIP T2 Interval Specifies the maximum retransmit time of any SIP request messages
(excluding the SIP INVITE message). The re-transmitting and doubling of
T1 continues until it reaches the T2 value. The default setting is 4 sec.
SIP Timer D Interval Defines the amount of time that the server transaction can remain when
unreliable response (3xx-6xx) received. The valid value is 0-64 seconds.
The default value is 0.
Remove OBP from Configures the device to remove the outbound proxy URI from the Route
Route header. This is used for the SIP Extension to notify the SIP server that the
device is behind a NAT/Firewall. If it is set to “Yes”, it will remove the Route
header from SIP requests. The default setting is "No".
Enable 100rel Actives PRACK (Provisional Acknowledgment) method. PRACK improves
the network reliability by adding an acknowledgement system to the
provisional Responses (1xx). It is set to “Yes”, the device will response to
the 1xx response from the remote party. Default is "No".
Session Timer
Enable Session Timer Allows the device to use the session timer, when set to “Yes”, it will be
added in the SIP INVITE message to notify the server.
Session Expiration (s) Configures the device’s SIP session timer. It enables SIP sessions to be
periodically "refreshed" via a SIP request (UPDATE, or re-INVITE). If there
is no refresh via an UPDATE or re-INVITE message, the session will be
terminated once the session interval expires.
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Session Expiration is the time (in seconds) where the session is considered
timed out, provided no successful session refresh transaction occurs
beforehand.
The default setting is 180. The valid range is from 90 to 64800.
Min-SE (s) Determines the minimum session expiration timer (in seconds) if the device
act as a timer refresher. Default is 90. The valid range is from 90 to 64800.
UAC Specify Refresher Sets which party will refresh the active session if the device makes
outbound calls. If it is set to “UAC” and the remote party does not support
Refresher feature, the device will refresh the active session.
If it is set to “UAS”, the remote party will refresh it. If it is set to “Omit”, the
header will be omitted so that it can be selected by the negotiation
mechanism. The default setting is "Omit".
UAS Specify Refresher Specifies which party will refresh the active session if the device receives
inbound calls. If it is set to “UAC”, the remote party will refresh the active
session. If it is set to “UAS” and the remote party does not support refresh
feature, the device will refresh it. The default setting is "UAC".
Caller Request Timer Sets the caller party to act as refresher by force. If set to "Yes" and both
party support session timers, the device will enable the session timer
feature when it makes outbound calls. The SIP INVITE will include the
content “refresher=uac”. The default setting is "No".
Callee Request Timer Sets the callee party to act as refresher by force. If set to "Yes" and the
both parties support session timers, the device will enable the session
timer feature when it receives inbound calls. The SIP 200 OK will include
the content “refresher=uas”. The default setting is "No".
Force Timer Configures the session timer feature on the device by force.
If it is set to "Yes", the device will use the session timer even if the
remote party does not support this feature.
If set to "No", the device will enable the session timer only when the
remote party supports this feature. To turn off the session timer, select
"No".
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Codec Settings
Preferred Vocoder
Preferred Vocoder Lists the available and enabled Audio codecs for this account. Users can
enable the specific audio codecs by moving them to the selected box and
set them with a priority order from top to bottom. This configuration will be
included with the same preference order in the SIP SDP message.
Codec Negotiation Configures the device to use which codec sequence to negotiate as the
Priority callee. When set to “Caller”, the device negotiates by SDP codec sequence
from received SIP Invite; When set to “Callee”, the device negotiates by
audio codec sequence on the device. The default setting is “Callee”.
Use First Matching Configures the device to use the first matching codec in the 200OK
Vocoder in 200OK SDP message. The default value is 0.
iLBC Frame Size Sets the iLBC (Internet Low Bitrate Codec) frame size if ILBC is used.
Users can select it from 20ms or 30ms. The default setting is 30ms.
G726-32 ITU Payload Configures G726-32 payload type for ITU packing mode. Payload 2 is static
Type and payload dynamic is dynamic. The default setting is “2”.
G726-32 Dynamic Specifies the G726-32 payload type, and the valid range is 96 to 127. The
Payload Type default setting is “126”.
Opus Payload Type Defines the desired value (96-127) for the payload type of the Opus codec.
The default value is 123.
DTMF Specifies the mechanism to transmit DTMF (Dual Tone Multi-Frequency)
signals.
There are 3 supported modes: in audio, RFC2833, or SIP INFO.
• In audio, which means DTMF is combined in the audio signal (not very
reliable with low-bit-rate codecs);
• RFC2833, which means to specify DTMF with RTP packet. Users could
know the packet is DTMF in the RTP header as well as the type of DTMF.
• SIP INFO, which uses SIP INFO to carry DTMF. The defect of this mode
is that it's easily to cause desynchronized of DTMF and media packet if the
SIP and RTP messages are required to transmitted respectively.
The default setting is "RFC2833".
DTMF Payload Type Configures the RTP payload type that indicates the transmitted packet
contains DTMF digits. Valid range is from 96 to 127. Default value is 101.
Jitter Buffer Type Selects either Fixed or Adaptive based on network conditions.
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Enable Audio RED with If set to "Yes", FEC will be enabled for audio call. The default setting is
FEC "No".
Audio FEC Payload Type Configures audio FEC payload type. The valid range is from 96 to 127. The
default value is 121.
Audio RED Payload Configures audio RED payload type. The valid range is from 96 to 127. The
Type default value is 124.
Silence Suppression Enables the silence suppression/VAD feature. If it is set to “Yes”, when
silence is detected, a small quantity of VAD packets (instead of audio
packets) will be sent during the period of no talking. If set to “No”, this
feature is disabled. The default setting is “No”.
Voice Frames Per TX Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the “ptime” value for the SDP will
change with different configurations here. This value is related to the codec
used and the actual frames transmitted during the in-payload call. For end
users, it is recommended to use the default setting, as incorrect settings
may influence the audio quality.
The default setting is 2.
RTP Settings
SRTP Mode Sets if the device will enable the SRTP (Secured RTP) mode. It can be
selected from dropdown list:
Disable
Enabled but not forced
Enabled and forced
SRTP uses encryption and authentication to minimize the risk of denial of
service. (DoS). If the server allows to use both RTP and SRTP, it should
be configured as “Enabled but not forced”.
The default setting is “Disable”.
SRTP Key Length Configures all the AES (Advanced Encryption Standard) key size within
SRTP. It can be selected from dropdown list:
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Enable SRTP Key Life Defines the SRTP key life time. When this option is set to be enabled,
Time during the SRTP call, the SRTP key will be valid within 231 SIP packets,
and phone will renew the SRTP key after this limitation.
The default setting is “Yes”.
RTCP Destination Configures a remote server URI where RTCP messages will be sent to
during an active call.
Symmetric RTP Configures if the device enables the symmetric RTP mechanism.
If it is set to “Yes”, the device will use the same socket/port for sending and
receiving the RTP messages.
The default setting is "No".
RTP IP Filter Receives the RTP packets from the specified IP address and Port by
communication protocol. If it is set to “IP Only”, the device only receives
the RTP packets from the specified IP address based on the
communication protocol; If it is set to “IP and Port”, the device will receive
the RTP packets from the specified IP address with the specified port
based on the communication protocol.
The default setting is “Disable”.
Call Settings
Call Features
Play Warning Tone for When this option is enabled, the device will play a warning tone When auto-
Auto Answer Intercom answering intercom. The default setting is “yes”.
Send Anonymous If set to "Yes", the "From" header in outgoing INVITE messages will be set
to anonymous, essentially blocking the Caller ID to be displayed.
Intercept Anonymous If set to "Yes", anonymous calls will be rejected.
Calls
Call Log Categorizes the call logs saved for this account. If it is set to “Log All”, all
the call logs of this account will be saved.
If set to “Log Incoming/Outgoing Calls (Missed Calls Not Record)”, the
whole call history will be saved other than missed call.
If it is set to “Disable Call All”, none of the call history will be saved. If it is
set to “Don’t Prompt Missed Call”, the device will log the missed call
histories, but there is no prompt to indicate the missed calls.
The default setting is “Log All”.
Ring Timeout (s) Defines the expiration timer (in seconds) for the rings with no answer. The
default setting is 60. The valid range is from 10 to 300.
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Refer-To Use Target Sets the device to use the target’s Contact header tag to the Refer-To
Contact header in the SIP REFER message during an attended transfer.
The default setting is “No”.
Incoming Call Rules Allow to set incoming call rules for each account registered. This
configuration will over rule the global incoming call rules. If set to "Block",
all greylist calls will be blocked. If set to "Set password", all greylist calls
will need to enter the correct password before they can be answered. If
set to "Auto answer", all greylist calls will be automatically answered. If set
to "Ringing", all greylist calls will continue to ringing. The default ring time
is 60s. You can customize the timeout under the Account Call setting
Ring timeout. The default value is "Disable".
Dial Plan
Dial Plan Prefix This parameter can be configured to define the prefix added to each dialed
number.
Disable DialPlan Enables/disables the Dial plan mechanism for different cases. If the specific
case is checked, the Dial plan mechanism will be disabled.
Dial Page: It controls the pattern of dialing numbers from the call
page.
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f) <2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
h) \+ - add + to the dialing number
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Advanced Settings
Security Settings
Check Domain Sets the device to check the domain certificates if TLS/TCP is used for SIP
Certificates Transport.
The default setting is "No".
Validate Certification Configures whether to validate certification chain, when TLS/TCP is
Chain configured for SIP Transport. If this is set to “Yes”, phone will validate server
against the new certificate list.
The default setting is “No”.
Validate Incoming SIP Specifies if the device will check the incoming SIP messages caller ID and
Messages CSeq headers. If the message does not include the headers, it will be
rejected.
The default setting is "No".
Allow Unsolicited It is used to configure whether to dial the number carried by Refer-to after
REFER receiving SIP REFER request actively.
If it is set to "Disabled", the device will send error warning and stop dialing. If
it is set to "Enabled/Force Auth", the device will dial the number after sending
authentication, if the authentication failed, then the dialing will be stopped. If
it is set to "Enabled", the device will dial up all numbers carried by SIP REFER.
The default is "Disabled".
Only Accept SIP Answers the SIP request from saved servers when set to “Yes”, only the SIP
Requests from requests from saved servers will be accepted; and the SIP requests from the
Known Servers unregistered server will be rejected. The default setting is “No”.
Check SIP User ID for Configures the device to check the SIP User ID in the Request URI of the SIP
Incoming INVITE INVITE message from the remote party. If it doesn't match the device's SIP
User ID, the call will be rejected. The default setting is “No”.
Allow SIP Reset It is used to configure whether to allow SIP Notification message to perform
factory reset on the device. The default setting is "No".
Authenticate Configures the device to authenticate the SIP INVITE message from the
Incoming INVITE remote party. If set to "Yes", the device will challenge the incoming INVITE for
authentication with SIP 401 Unauthorized response. Default is "No".
SIP Realm used for Configure this item to validate incoming INVITE, but you must enable
Challenge INVITE & authenticate incoming INVITE first to make it take effect. You can verify the
NOTIFY NOTIFY information for the provision, including check-sync, resync and
reboot, but only when SIP NOTIFY authentication enabled first to make it take
effect.
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MOH
Upload Local MOH Loads the MOH file to the device. Click on “Browse” button to upload the
Audio File music file from local PC. The MOH audio file has to be in .wav or .mp3 format.
(Music on Hold) Note: Please be patient while the audio file is being uploaded. It could take
more than 3 minutes to finish the uploading especially the file size is large.
The button will show as “Processing” during the uploading. Once done, it will
show as “Browse” again. Click on “Save” on the bottom of the web page and
“Apply” on the top of the web page to save the change.
Advanced Features
Virtual Account It is used to set to categorize accounts in server mode groups, the accounts
Group in the same group will be combined as one and the account widget will display
the Caller ID in the account with lowest ID. The device can answer any
incoming calls to each account in groups.
If user makes an outbound call, the device will use the lowest ID account by
default. If the account fails or SIP INVITE message is timeout, the device will
failover to the next account in the group with higher account ID.
If all the accounts are not available in the group, the device will traverse all
the accounts in the group and notify the users the session is failed.
Special Feature Configures phone’s settings to meet different vendors’ server requirements.
Users can choose from Standard, NEC, WorldStone, BroadSoft, China
Mobile, ZTE IMS, Mobotix, ZTE NGN, or Huawei IMS depending on the server
type.
Call
The GSC3510 allows users to manage their calls using the Click to Dial feature which permits to initiate
and receive calls using the Web GUI. To use the Click to Dial feature, please refer the following steps:
3. Type the number / IP Address to call and press Dial button as displayed on the following
screenshots:
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Figure 30: Click-to-Dial Feature
Once the number / IP address is dialed or a Call is received, a window pops up showing the call information
Call History
The GSC3510 Call History is divided into two sections: “All” and “Intercepted Record”:
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Call History All
This section shows all the calls that have been made or answered. Users can find two types of calls under
“Call History → All”:
Outgoing Calls.
Answered Calls.
By Tapping on the checkbox to select the call history entries, users can do the following operations:
Delete Call History: Users need to press the button after selecting the call history entries.
Add entries to Whitelist: Users may select the entries to be allowed to call the
GSC3510/GSC3505 by clicking on the button after selecting the right entries.
Add entries to Blacklist: Users can block the calls of some entries by selecting them and pressing
the button .
Make a call to one of the call history entries: Users can directly make a call to a number listed
in the call history by clicking directly on the button under “OPERATION”.
Show calls details: users can show the calls details of a number by clicking on the button and
a window will pop up to show all the calls sent/received with the selected number.
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Figure 33: Call details under Call History → All
From the Call details window, users can also add the number selected to local contacts by creating
a new contact , or by adding it to an existing contact .
Add number to an existing contact: Users can click on “Save to local contacts” in order to show
a window with all the contacts already registered in the GSC3510/GSC3505 local contacts and to
choose one of the contacts to link the selected number with:
Create a new contact: user can click on “Add to local contacts” in order to show a window where
all the information about the contact need to be entered.
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Contacts” for more information about creating a new contact or editing an existing one.
This section shows all the calls that have been blocked when received because of not having the permission
to make a call to the GSC3510/GSC3505. Users can find only one type of calls under “Call History →
Intercept Record”:
Blocked Calls
By checking the checkbox to select entries, users can do the following operations:
Delete Blocked Numbers Call History: Users need to press the button after selecting
the call history entries.
Add entries to Whitelist: Users may select the blocked entries to give them permission to call the
GSC3510/GSC3505 by clicking on the button after selecting the right entries.
Make a call to one of the entries: Users can directly make a call to a number listed in call history
→ Intercept Record, by clicking directly on the button under “OPERATION”.
Show calls details: users can show the calls details of a number by clicking on the button and
a window will pop up to show all the blocked calls received from the selected number.
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Figure 36: Call details under Call History → Intercept Record
The GSC GUI will display a popup window to notify about missed calls. The figure below contains an
example where the Extension 4444 couldn’t reach the GSC at 3333.
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Contacts
Contacts section is divided into two sections: “Contacts List” and “Group”.
Contacts List
: Users can select one or a bench of contacts and click on the “Delete” button order to
delete all the selected contacts.
: Users can select one or bench of contacts and click on the “Add to Whitelist”
button in order to directly add the selected contacts to the list of contacts allowed to call the
GSC3510/GSC3505.
: Users can select one or bench of contacts and click on the “Add to blacklist” button
in order to remove the permission to call the GSC3510/GSC3505 from the selected contacts.
: Users can create a new contact by clicking on the “New contact” button, then a
window pops up (Please, refer to the following figure) in order to enter the new contact’s details.
: Users need to click on the “More” button for more operations (Import contacts, Export
contacts, Download contacts).
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Import Contacts
Clear The Old List Determines if the device will delete the previous contacts when a new
contact file is imported. If set to "Yes", the previous contacts will be
removed. The default setting is "No".
Clear Old History Mode If set to "Clear all", the device will delete all previous records before
importing the new records. If set to "Keep Local Contacts", the new-added
local new contacts will not be deleted when importing new records.
Replace Duplicate Items Configures the device to keep the original contact entries when duplicated
contact entries are included in the contact file. If set to "Yes", the device
will replace the original entries to the new one. Otherwise, the device will
save both contact entries.
The default setting is "No".
Replace Duplicate If set to "Replace by name", replace the records of the same name
Entries Mode automatically when importing new records. If set to "Replace by number",
replace the records of the same number automatically when importing new
records.
File Encoding Specifies the encoding format for contacts file importing. It can be selected
from the dropdown list:
UTF-8
GBK
UTF-16
UTF-32
Big5
Big5-HKSCS
Shift-JIS
ISO8859-1
ISO8859-15
Windows-1251
EUC-KR
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Export Contacts
File Encoding Specifies the encoding format for contacts file exporting. It can be selected
from the dropdown list:
UTF-8 (Default)
GBK
UTF-16
UTF-32
Big5
Big5-HKSCS
Shift-JIS
ISO8859-1
ISO8859-15
Windows-1251
EUC-KR
File Type Sets the type format for contacts file exporting. It can be selected from the
dropdown list.
XML
vCard
The default setting is "XML".
Export Downloads the contacts file from the device to PC.
Download Contacts (XML Contacts)
Clear The Old List Sets the device to delete the previous contacts when a new contact file is
downloaded. If set to "Yes", the previous contacts will be removed. The
default setting is "No".
Clear Old History Mode If set to "Clear all", the device will delete all previous records before
downloading the new records. If set to "Keep Local Contacts", the new-
added local new contacts will not be deleted when downloading new
records.
Replace Duplicate Items Keeps the original contact entries when duplicated contact entries are
included in the contact file. If set to "Yes", the device will replace the original
entries to the new one. Otherwise, the device will save both contact entries.
The default setting is "No".
Replace Duplicate If set to "Replace by name", replace the records of the same name
Entries Mode automatically when downloading new records. If set to "Replace by
number", replace the records of the same number automatically when
downloading new records.
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Download Mode Enables the device to download contacts file and select the server and
protocol to download the contacts file. It can be selected from TFTP, HTTP,
and HTTPS. The default setting is “OFF”.
File Encoding Selects the encoding format for contacts file download. It can be selected
from the dropdown list:
UTF-8
GBK
UTF-16
UTF-32
Big5
Big5-HKSCS
Shift-JIS
ISO8859-1
ISO8859-15
Windows-1251
EUC-KR
None
2 Hour
4 Hour
6 Hour
8 Hour
12 Hour
Download Now Starts downloading the XML contacts to the device immediately.
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Group
Users could manage the groups of the existing contacts that can be found in “Contacts List”.
: Users can select one or a bench of groups and click on the “Delete” button in order to
delete all the selected groups.
: Users can create a new group by clicking on the “New group” button, then a window
pops up (Please, refer to the following figure) in order to enter the group’s name and specify the
contacts that will be included in it.
This section is for managing calling permissions to the GSC3510/GSC3505. Users can give or remove the
permission to call the GSC3510/GSC3505, this can be managed under the following three subsections:
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Whitelist
Users can specify the numbers allowed to call the GSC3510/GSC3505 and every time a number is added
it is listed in the below list:
: Users can remove one or a group of numbers from whitelist by clicking in “Remove
Whitelist” button.
Note: Users can also press on to remove one specific contacts from whitelist.
: Users can add the phonebook contacts to the whitelist by clicking on “Add from
contacts” button. A window pops up showing the existing contacts so that users can select the ones
wishing to give permission to.
: Users can add the numbers that the GSC3510/GSC3505 is blocking to the
Whitelist by clicking on “Add from blocked calls”. A window pops up showing all the blocked
numbers.
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Figure 44: Add blocked numbers to whitelist
: Users can add numbers manually to whitelist by clicking on “Add manually” button. A
window pops up allowing users to enter the number and its name.
Note: Users can modify the name of the number listed in the Whitelist by clicking on .
Blacklist
Users can specify the numbers to be blocked by the GSC3510/GSC3505 for incoming calls, and every time
a number is added to the blacklist, it is listed in the below list:
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: Users can remove one or a group of numbers from blacklist by clicking in “Remove
blacklist” button.
Note: Users can also press on to remove one specific contacts from blacklist.
: Users can add phonebook contacts to the blacklist by clicking on “Add from
contacts” button. A window pops up showing the existing contacts so that users may select the
ones wishing to give permission to (Please, refer to Figure 23).
: Users can add numbers from Call History to the blacklist by clicking on “Add
from call history” button. A window pops up showing all the calls listed in the GSC3510/GSC3505
call history.
: Users can add numbers manually to blacklist by clicking on “Add manually” button. A
window pops up allowing users to enter the number and its name (Please, refer to Figure 25).
Note: Users can modify the name of the number listed in the blacklist by clicking on .
Blocking Rules
This sub-section allows the user to define the blocking rules for Non-white list calls. The blocking rules
available for the users are:
Block: Configures the GSC3510/GSC3505 to block all the numbers that are not listed in the
Whitelist.
Answer: Configures the GSC3510/GSC3505 to allow all the calls received from any number but
the number listed in the blacklist.
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Phone Settings Page Definitions
General Settings
Note: This parameter must be set to “No” for Direct IP Calling to work.
Keep-alive Interval (s) Specifies how the device will send a Binding Request packet to the SIP
server in order to keep the “ping hole” on the NAT router to open. The
default setting is 20 seconds. The valid range is from 10 to 160.
STUN Server Configures the URI of STUN (Simple Traversal of UDP for NAT) server.
The device will send STUN Binding Request packet to the STUN server to
learn the public IP address of its network. Only non-symmetric NAT routers
work with STUN. The default setting is “stun.ipvideotalk.com”.
TURN Server Username Fill in the username to validate TURN server.
TURN Server Password Fill in the password to validate TURN server.
Use NAT IP Configures the IP address for the Contact header and Connection
Information in the SIP/SDP message. It should ONLY be used if it’s
required by your ITSP. By default, the box is blank.
Call Settings
Enable Call Waiting Enables call waiting feature. If it is disabled, the GSC3510 will reject the
second incoming call during an active session without user’s knowledge.
But this missed call record will be saved to remind users. The default
setting is checked (enabled).
Enable Call Waiting Tone Sets the GSC3510 to play the call waiting tone along with LED indicator if
there is another incoming call. If unchecked, only LED will indicate another
incoming call. The default setting is checked (enabled).
Automatic Answer Configures the ring time for the unanswered call. The call will be
Ringing Time (s) automatically answered after timeout. Default is 5.
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Busy Tone Ring Time (s) Configures the timeout for Busy Tone during the call. Default is 30.
Auto Mute on Entry Configures whether to mute the call on entry automatically.
Note: This function only take effect when the device is from the idle status
to call status. Users could click the Mute button on call interface to cancel
the current mute status.
Virtual Account Group If set to "Yes", when processing SIP Register 3XX Response, it will parse
Avaya Mode the address site in 3XX, modify the account server info "SIP Server: port"
& "SIP Transaction" in virtual account group and initiate registration again.
This feature is designed for the Avaya customers.
Filter Characters Sets the characters for filter when dial out numbers. Users could set up
multiple characters. For example, if set to “[()-]”, when dial (0571)-8800-
8888, the character “()-“ will be automatically filtered and dial
057188008888 directly.
Escape # as %23 in SIP Determines which characters will be included in the SIP INVITE URI if end
URI users input #. If it is set to “Yes” (Default), the device will replace the #
by %23. Otherwise, it will include # in the SIP INVITE message.
Record Mode Configures phone recording mode. If set to "Record locally", then will use
the local tape recorder for call recording, and the audio file will be saved in
accordance with the tape recorder setup. If set to "Record on PortaOne",
then will send the specified SIP messages to the corresponding server; If
set to "Record on UCM", then will send the recording feature code to the
UCM server to request for recording, and the recording function will be
executed by the server.
Environment Sets operating environment for the device.
When set to "Small and medium size room & used on desk", the
sound pickup range is increased and ENC is reduced;
When set to "Large room & used on empty area", the sound pickup
range is reduced and ENC is increased.
The default value is "Large room & used on empty area".
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Ring Tone
Auto Config CPT by Configures whether to choose Call Progress Tone automatically by region.
Region If set to "Yes", the device will configure CPT (Call Progress Tone)
according to different regions automatically. If set to "No", you can manual
configure CPT parameters. The default setting is "No".
Call Progress Tones: Configures tone frequencies according to user preference. By default, the
Dial Tone tones are set to North American frequencies. Frequencies should be
Second Dial Tone configured with known values to avoid uncomfortable high pitch sounds.
Ring Back Tone Syntax: f1=val,f2=val [,c=on1/off1[-on2/off2[-on3/off3]]];
Busy Tone (Frequencies are in Hz and cadence on and off are in 10ms)
Reorder Tone ON is the period of ringing ("On time" in "ms") while OFF is the period of
Confirmation Tone silence. In order to set a continuous ring, OFF should be zero. Otherwise
Call-Waiting Tone it will ring ON ms and a pause of OFF ms and then repeats the pattern.
Please refer to the document below to determine your local call progress
tones: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
Call-Waiting Tone Gain Adjusts the call waiting tone volume. Users can select "Low", "Medium" or
"High". The default setting is "Low".
Default Ring Cadence Defines the ring cadence for the device. The default setting is:
c=2000/4000.
Multicast Paging
Multicast Paging
Paging Barge Sets the threshold of paging calls. If the paging call’s priority is higher than
the threshold, the existing call will be hold and the paging call will be
answered. Otherwise, the existing call does not be affected. If it is set to
Disable (Default), any paging call will not be answered.
Paging Priority Active Determines if a new paging call whose priority is higher than the existing
paging call will be answered. The default is disabled. Check to enabled.
Multicast Paging Codec Selects the codec type for the multicast paging call. This list includes
PCMU, PCMA, G726-32, G722, and G729A/B, iLBC, Opus.
Multicast Listening
Priority Configures the IP address and port number for monitoring multicast paging
Listening Address call. Reboot the device to make changes take effect.
Label The valid IP address range is from 224.0.0.0 to 239.255.255.255. Users
may also fill the label for each listening address corresponding to priority.
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Network Settings Page Definitions
Ethernet Settings
Preferred Internet If IPv4 is selected, the device will be using IPv4 addressing, otherwise, it
Protocol will be using IPv6 addressing.
Default is Prefer IPv4.
Different Networks for Configures whether to set up different networks for the phone data and the
Data and VoIP Calls call. If set to "Yes", you need to configure the data network and VoIP
network respectively.
Note: Reboot is required to take effect.
IPv4
IPv4 Address Type Allows users to configure the appropriate network settings on the device.
Users could select "DHCP", "Static IP" or "PPPoE".
DHCP: Obtain the IP address via one DHCP server in the LAN. All
domain values about static IP/PPPoE are unavailable (although
some domain values have been saved in the flash.)
When set to “DHCP Option 132 and DHCP Option 133”, the
GSC3510/GSC3505 will get DHCP Option 132 as VLAN ID and
get DHCP Option 133 as VLAN priority, from the DHCP server
directly.
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When set to “Encapsulated in DHCP Option 43”, the
GSC3510/GSC3505 will get VLAN ID and VLAN priority value
from the DHCP Option 43 which has DHCP Option 132 and DHCP
Option 133 encapsulated. In this case, please make sure the
option “Allow DHCP Option 43 and Option 66 to Override Server”
is enabled under GSC3510/GSC3505 web UI Maintenance
Upgrade.
Layer 2 QoS 802.1p Assigns the priority value of the Layer 2 QoS packets for Ethernet.
Priority Value (Ethernet) The Default value is 0.
for Data Note: When “Different Networks for Data and VoIP Calls” is set to Yes,
for VoIP Calls user needs to set “Layer 2 QoS 802.1p Priority Value (Ethernet) for Data”
and “Layer 2 QoS 802.1p Priority Value (Ethernet) for VoIP Calls”.
IPv6
IPv6 Address Configures the appropriate network settings on the device. Users could
select from "Auto-configured" or “Statically configured”.
Preferred DNS Server Configures the Preferred DNS Server.
DNS Server 1 Configures the primary DNS IP address.
Static IPv6 Address Enter the static IPv6 address in "Statically configured" IPv6 address type.
IPv6 Prefix Length Enter the IPv6 prefix length in "Statically configured" IPv6 address type.
Default is 64.
802.1x Mode
802.1x mode Enables and selects the 802.1x mode for the device. The supported 802.1x
modes are: EAP-MD5, EAP-TLS and EAP-PEAP.
The default setting is "Disable".
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802.1x Identity Enters the identity information for the selected 802.1x mode. (This setting
will be displayed only if 802.1 X mode is enabled).
802.1x Secret Enters the secret for the 802.1x mode. This option will appear when 802.1x
mode is EAP-MD5 or EAP-PEAP.
CA Certificate Uploads the CA Certificate file to the device. (This setting will be displayed
only if the 802.1 X mode is enabled)
Client Certificate Loads the Client Certificate file to the device. (This setting will be displayed
only if the 802.1 X TLS mode is enabled)
Private Key Loads the private key file to the device. (This setting will be displayed only
if the 802.1 X TLS mode is enabled)
Bluetooth
Wi-Fi Settings
Users can connect wirelessly to a network using Wi-Fi under GSC3510/GSC3505 Web GUI Network
Settings Wi-Fi Settings. In order to connect to a network using Wi-Fi, please, refer to the following
steps:
3. Click on Scan to show the list of Wi-Fi networks available around the GSC3510/GSC3505
Note: The list of Wi-Fi Networks refreshes automatically every 15 seconds and user can force to refresh
by clicking again on “Scan”.
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Figure 48: Wi-Fi Basics Page
4. Identify the Wi-Fi network’s SSID and click on “Connect”, then enter the correct password
information to connect to the selected network:
5. Users can check the Wi-Fi parameters and change the setting by checking the “Show advanced
options” in the bottom.
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Figure 50: GSC3510/GSC3505 Connect to Wi-Fi-Show Advanced Options
Wi-Fi Basics
Wi-Fi Function Enables/disables the Wi-Fi feature. The default setting is "Disable".
Permits to scan and select the available Wi-Fi networks within the range if
ESSID the Wi-Fi feature is enabled. Click on "Connect" to select the Wi-Fi network
and to enter the needed password if it is required.
Add Network
ESSID Configures the hidden ESSID name.
Defines the security mode used for the wireless network when the SSID is
Security Mode for
hidden. Default is “None”. Users can choose: WEP, WPA/WPA2 PSK or
Hidden SSID
802.1x EAP.
Password Configures the hidden ESSID password.
Advanced Settings
Layer 2 QoS 802.1p Assigns the priority value of the Layer 2 QoS packets for Wi-Fi.
Priority Value (Wi-Fi) The Default value is 0.
Country Code Configures Wi-Fi country code.
The default value is "United States of America".
Note: Reboot is requested to take effect.
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OpenVPN® Settings
OpenVPN ® Settings
Enable OpenVPN® This enables/disables OpenVPN® functionality, and requires the user to
have access to an OpenVPN® server. The default setting is No.
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Advanced Network Settings
Enable CDP Configures whether to enable CDP to receive and/or transmit information
from/to CDP-enabled devices.
The default setting is "No".
Layer 3 QoS for SIP Defines the Layer 3 packet’s QoS parameter for SIP messages in decimal
pattern. This value is used for IP Precedence, Diff-Serv or MPLS.
Default setting is 48 which is equivalent to the DSCP name constant CS6.
Layer 3 QoS for Audio Defines the Layer 3 packet’s QoS parameter for RTP messages in decimal
pattern. This value is used for IP Precedence, Diff-Serv or MPLS.
Default setting is 48 which is equivalent to the DSCP name constant CS6.
HTTP/HTTPS User- Sets the user-agent for contacts.
Agent Note: Reboot the device to make changes take effect.
SIP User-Agent Sets the user-agent for SIP. Default is:
Grandstream GSC3510 $version
Grandstream GSC3505 $version
Proxy
HTTP/HTTPS Proxy Specifies the HTTP/HTTPS proxy hostname for the device to send packets
Hostname to. The proxy server will act as an intermediary to route the packets to the
destination.
HTTP/HTTPS Proxy Port Specifies the HTTP/HTTPS proxy port for the device to send packets to.
The proxy server will act as an intermediary to route the packets to the
destination.
Bypass Proxy For Defines the destination IP address where no proxy server is needed. The
device will not use a proxy server when sending packets to the specified
destination IP address.
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System Settings Page Definitions
Time Settings
Time Settings
Assign NTP Server Defines the URL or IP address of the NTP server. The phone may obtain
Address the current date and time information from the server. The default setting
is “pool.ntp.org”.
DHCP Option 42 Obtains NTP server address from a DHCP server using DHCP Option 42;
Override NTP Server it will override configured NTP Server. If set to “No”, the phone will use
configured NTP server to synchronize time and date even if a NTP server
is provided by DHCP server. The default setting is "Yes".
Note: Changes in this setting need reboot to take effect.
DHCP Option 2 to Obtains time zone setting (offset) from a DHCP server using DHCP Option
Override Time Zone 2; it will override selected time zone.
Settings If set to “No”, the phone will use selected time zone even if provided by
DHCP server. The default setting is Yes.
Note: Changes in this setting need reboot to take effect.
Time Zone Specifies the local time zone for the phone. It covers the global time zones
and user can selected the specific one from the drop-down list.
Time Display Format Specifies which format will be used to display the time. It can be selected
from 12-hour and 24-hour format.
Date Display Format Determines which format will be used to display the date.
It can be selected from the drop-down list.
Normal (YYYY/M/D)
YYYY/MM/DD
MM/DD/YYYY
DD/MM/YYYY
The default setting is MM/DD/YYYY
Security Settings
Web/SSH Access
Enable SSH Enables/disables SSH access to the device. The default setting is "Yes".
Allow web page access Allow web page access, If set to “NO”, the Web UI will not be accessible.
Please turn it on carefully. The setting requires reboot to take effect.
Enabled by Default.
SSH Port Customizes the SSH port. By default, SSH uses port 22.
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Access Method Determines which protocol will be used to access the device ‘s Web GUI.
It can be selected from HTTP and HTTPS. The default setting is HTTP.
Port Specifies which port to use to access the Web UI. By default, if HTTP, the
port number will be 80; if HTTPS is selected, the port number will be 443.
User Info Management
Current Admin Enter current logged-in user's password. This field is case sensitive. The
Password default password is "admin".
New Admin Password Allows the user to change the admin password. The password field is
purposely blank after clicking the “Save” button for security purpose. This
field is case sensitive with a maximum length of 32 characters.
Confirm Admin Enter the new Admin password again to confirm.
Password
New User Password Allows the administrator to set the password for user-level web GUI
access. This field is case sensitive with a maximum length of 32
characters. The default password is “123”.
Confirm New User Enter the new User password again to confirm.
Password
TLS
Minimum TLS Version Specifies the minimum TLS version allowed for the connection.
Maximum TLS Version Specifies the maximum TLS version allowed for the connection.
Default is 99.
SIP TLS
SIP TLS Certificate Defines the SSL certificate used for SIP over TLS.
SIP TLS Private Key Defines the SSL Private key used for SIP over TLS.
SIP TLS Private Key Defines the SSL Private key password used for SIP over TLS.
Password
Certificate Management
CA Certificate
Import Trusted CA Allows to upload the CA Certificate file to phone.
Certificates Note: Reboot is required to take effect.
Trusted CA Certificates Lists trusted CA certificates previously uploaded. Administrator can delete
a certificate from here.
User Certificate
Add User Certificate Allows to upload & Install User Certificate file to phone.
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User Certificates Lists Users Certificates previously uploaded. Administrator can delete a
certificate from here.
Custom Certificate
Import Custom Allows to upload & Install Custom Certificate file to device.
Certificate
Custom Certificate Lists Custom Certificates previously uploaded. Administrator can delete a
certificate from here.
Preferences
LED Management
Disable Missed Call If set to "Yes", the LED indicator will not light up when there is missed call
Indicator on the device.
Disable Contact Full If set to "Yes", the LED indicator will light up when the contact storage or
Indicator message storage is full.
Brightness Control
Red Light Control the intensity of the Red Color in the LED
Green Light Control the intensity of the Green Color in the LED
Blue Light Control the intensity of the Blue Color in the LED
Volume Settings
Call Volume Sets the volume of calls.
Ringtone volume Sets the volume of ringtones.
Media volume Sets the volume of media.
TR-069
Enable TR-069 Sets the device to enable the “CPE WAN Management Protocol” (TR-069).
The default setting is "No".
Note: Reboot the device to make changes take effect.
ACS URL Specifies URL of TR-069 ACS (e.g, http://acs.test.com), or IP address.
ACS Username Enters username to authenticate to ACS.
ACS Password Enters password to authenticate to ACS.
Periodic Inform Enable Sends periodic inform packets to ACS. Default is “No”.
Periodic Inform Interval Configures to sends periodic “Inform” packets to ACS based on specified
(s) interval. The default setting is 86400.
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Connection Request Enters username for the ACS to connect to the device.
Username
Connection Request Enters password for the ACS to connect to the device.
Password
Connection Request Enters the port for the ACS to connect to the device.
Port
CPE Cert File Uploads Cert File for the device to connect to the ACS via SSL.
CPE Cert Key Uploads Cert Key for the device to connect to the ACS via SSL.
Sensor Settings
Basic Settings
Basic Settings
Sensor Type Set the initial state of the sensor, when the selection is normally open, the
contact is disconnected when static; When the selection is normally closed,
the contact is connected when static. The normally open will be connected
when the electrical action is on the switch, and the normally closed will
disconnected. The default is normally open.
Trigger Type Set the type of the trigger mode, and when the selection is level triggered,
only high level (1) or low level (0) will trigger the notification. When the edge
trigger is selected, the notification is triggered only when the level changes
(high level to low level, or low level to high level). The default is level trigger.
Trigger time
The alarm can be configured to be triggered all days of the week, in this
case “All days” option needs to be checked. Or to some specific schedule,
Cycle Time
in this case “Period of Time” option needs to be checked for users to be
able to configure Time and Frequency options below.
Set the activation time, up to 3 times. When the activation time is not set,
Time
the default time is full day.
Set the activation frequency from Monday to Sunday, which can be
Frequency
selected from the whole week. The default value is not selected.
Play audio Play a sound when the switch is triggered during the scheduled time.
When the "voice prompt" is selected, you can upload the customized audio
Prompt tone
by clicking on “Upload” and choose the file.
Make call Dial the number when the sensor is activated.
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Enter the number you need to dial, and click the "add" button to set two
Dial out extension
numbers at the same time.
Recording Start recording when the sensor is activated.
Backup
GSC3510/GSC3505 Backup page is used to back up data or import backup files to restore data. Users can
start the Backup by clicking on “Start Backup”.
The following data on the GSC3510/GSC3505 can be backed up and restored to the device again using
the built-in Backup application: Contacts, Call history, Configuration information.
Once the wanted Backup contents are checked, users can generate the backup by clicking on “Backup”.
The backup will be shown with all the needed information (Back Content, Backup Time and Backup Size).
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Figure 53: Generated Backup
: Users can bring back to the GSC3510/GSC3505 all the backup contents selected during
Back generating operation by clicking on “Restore”.
: Users needs to click on “Backup” to generate a new backup and replace the existing one.
Upgrade
Firmware
Upgrade via Manually Upload
Complete Upgrade If enabled, all files will be replaced except user data.
Default setting is disabled.
Upload Firmware File to Allows users to load the local firmware to the device to update the firmware.
Update
Upgrade via Network
Firmware Upgrade Mode Allows users to choose the firmware upgrade method: TFTP, HTTP,
HTTPS or Manual Upload. The default setting is “HTTP”.
Firmware Server Path Sets IP address or domain name of firmware server. The URL of the server
that hosts the firmware release. Default is “fm.grandstream.com/gs”.
HTTP/HTTPS Username Enters the username for the firmware HTTP/HTTPS server.
HTTP/HTTPS Password Enters the password for the firmware HTTP/HTTPS server.
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Firmware File Prefix Checks if firmware file is with matching prefix before downloading it. This
field enables user to store different versions of firmware files in one
directory on the firmware server.
Firmware File Postfix Checks if firmware file is with matching postfix before downloading it. This
field enables user to store different versions of firmware files in one
directory on the firmware server.
Firmware Upgrade Click the "Detect" button to check whether the firmware in the firmware
server has an updated version, if so, update immediately.
Config File
Use Grandstream GAPS It is used to configure the download path and update mode for the
configuration file server.
If set to "Yes", the device will set the download path of the
configuration file to “fm.grandstream.com/gs” by default and use
HTTPS protocol to connect to the server.
If set to "No", then users can manually configure the path and update
mode for the configuration file server.
Config Upgrade Via Selects provisioning method: TFTP, HTTP or HTTPS. Default setting is
“HTTPS”.
Config Server Path Sets IP address or domain name of configuration server. The server hosts
a copy of the configuration file to be installed on the device. Default is
“fm.grandstream.com/gs”.
HTTP/HTTPS Username Configures the username for the config HTTP/HTTPS server.
HTTP/HTTPS Password Configures the password for the config HTTP/HTTPS server.
Always send HTTP Includes configured username and password in HTTP request before
Basic Authentication receiving authentication challenge from the server.
Information Default is “No”.
Config File Prefix Checks if configuration files are with matching prefix before downloading
them. This field enables user to store different configuration files in one
directory on the provisioning server.
Config File Postfix Checks if configuration files are with matching postfix before downloading
them. This field enables user to store different configuration files in one
directory on the provisioning server.
Authenticate Conf File Sets the device to authenticate configuration file before applying it. When
set to “Yes”, the configuration file must include value P1 with phone
system’s administration password. If it is missed or does not match the
password, the device will not apply it. Default setting is “No”.
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XML Config File Decrypts XML configuration file when encrypted. The password used for
Password encrypting the XML configuration file is using OpenSSL.
Download Device Downloads the device's configuration file in text format. The config file
Configuration includes all the P value parameters for phone's current settings except
password for security purpose. Users can use the Grandstream
configuration file generator to generate binary config file from this text file.
Upload Device Uploads configuration file to the device.
Configuration Note: The GSC35XX supports the following config file format:
“cfgMAC.xml”, where MAC is the MAC Address.
“cfgGSC35XX.xml”, where GSC35XX is the Product Model.
Provision
Automatic Upgrade
Automatic Upgrade Specifies when the firmware upgrade process will be initiated; there are 4
options:
No: The device will only do upgrade once at boot up.
Check every day: User needs to specify “Hour of the day (0-23)”.
Check every week: User needs to specify “Hour of the day (0-23)”
and “Day of the week (0-6)”.
Check at a period Time: User needs to specify “Hour of the day
(0-23)”
Note: Day of week is starting from Sunday. The default setting is “No”.
Starting – Ending Hour of Sets the time or time period of automatic upgrade, and automatically
the Day (0 – 23) update during the set time or time period.
Firmware Upgrade and Defines the device’s rules for automatic upgrade. It can be selected from:
Provisioning Always Check at bootup
Always Check at bootup, when F/W pre/suffix changes,
Skip the Firmware Check.
The default setting is “Always Check at bootup”.
DHCP Option
Allow DHCP option 43, If DHCP option 43, 160 and 66 is enabled on the LAN side, the device will
option 160 and option 66 reset the CPE, upgrade, network VLAN tag, and priority configuration
to Override Server according to option 43 sent by the server. At the same time, the update
mode and server path of the configuration upgrade mode will be reset
according to the option 160 and 66 sent by the server. The default setting
is "on". The default setting is "Yes".
Notes: Reboot the device to make changes take effect.
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DHCP Option 120 Configures the device to allow the DHCP offer message to override the
Override SIP Server Config Server Path via the Option 120 header.
The default setting is "Yes". Note: Reboot to take effect.
Allow DHCP Option 242 Enables DHCP Option 242. Once enabled, the device will use the
(Avaya IP Phones) configuration info issued by the local DHCP in Option 242 to configure
proxy, transport protocol and server path. The default setting is “Yes”.
Note: Reboot the device to make changes take effect.
Config Provision
Download and Process By default, the device will provision the first available config in the order of
All Available Config cfgMAC, cfgMAC.xml, cfgMODEL.xml, and cfg.xml (corresponding to
Files device specific, model specific, and global configs). If set to "Yes", the
device will download and apply (override) all available configs in the order
of cfgMAC, cfg.xml, cfgMODEL.xml, cfgMAC.xml.
Config Provision Device will download the configuration files and provision by the configured
order. Use arrow buttons to add and order configuration files.
PNP Feature
PnP(3CX) Auto Sets the device to broadcast the SIP SUBSCRIBE message during booting
Provision up to allow itself to be discovered and be configured by the SIP platform.
The default setting is "Yes". Note: Reboot to take effect.
Advanced Settings
Disable SIP NOTIFY Disables the SIP NOTIFY Authentication on the device. If set to “Yes”, the
Authentication GSC3510/GSC3505 will not challenge NOTIFY with 401. The default
setting is “No”.
Validate Certification Configures whether to validate the server certificate when download the
Chain firmware/config file. If it is set to "Yes", the device will download the
firmware/config file only from the legitimate server. Default setting is "No".
Allow AutoConfig Set to allow access to the AutoConfig service. If not checked, access to
service access service.ipvideotalk.com will be disabled. The default value is "Enabled".
mDNS Override Server Sets the device to broadcast the Multicast DNS (mDNS) message during
booting up to allow itself to be discovered and be configured by the SIP
platform. If it is set to “User Type A’, the device will broadcast the MDNS
message “A_grandstream-cfg.local”; if it is set to “Use Type SRV”, the
MDNS message will be “SRV_grandstream-cfg.local”.
The default setting is “Use Type A”.
Factory Reset Resets the device to the default factory setting mode.
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System Diagnosis
Syslog
Syslog Protocol Select the transport protocol over which log messages will be carried.
UDP: Syslog messages will be sent over UDP.
Note: Changing syslog level does not require a reboot to take effect.
Syslog Keyword Filter Only send the syslog with keyword, multiple keywords are separated by
comma. Example: set the filter keyword to “SIP” to filter SIP log.
Logcat
Log Tag Configures the filter to display the specified process log file.
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Debug
One-click Debugging
One-click Debugging Capture the checked info in the debugging list, click "Start" to debug if
including "Capture trace" item and click "Stop" to end, Click "Capture" in
another situation. All retrieved files will be generated to a package, and the
last package will be overwritten, while the trace file will stay remain.
Debug Info Menu Display a list of info items that can be debugged, currently supports system
logs, info log, capture package, tombstones and ANR log. The captured
data can be viewed in "Debug information list". The default is all selected.
Debug Info List You can select the existing debugging info package or grab package. Click
the "Delete" button on the right to delete the file.
View Debug Info You can select the existing debugging info package or grab package. Click
the "Delete" button on the right to delete the file.
Core Dump
Enable Core Dump Configures whether to generate and save the core dump file when the
Generation program crashes. The default setting is “No”.
Note: Reboot the device to make changes take effect.
Core Dump List Selects the existing core dump file in the drop-down box. Users could
delete the file by pressing on “Delete” button.
View Core Dump Press “List” button to view all existing core dump files. The files are listed
in chronological order, users could click the file name to download the file
to the local computer.
Record
Record Click to start capturing audio data, click the "Stop" button to end. To capture
the audio data of the device can help to locate audio issues. The default is
not enabled. You can record up to 1 minute audio data.
Recording List Choose the existing audio file. Click the "Delete" button on the right to
delete this file.
View Recording Click on the "List" button to view. The captured audio data will be sorted by
time.
Click to download the data to the computer for analysis.
Traceroute
Target Host The IP address or URL for the Target Host of the Traceroute.
Press Start to send traceroute request to configured target host.
Press Stop to end traceroute running process.
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Ping
Target Host The IP address or URL for the Target Host of the Ping.
Press Start to send traceroute request to configured target host.
Press Stop to end traceroute running process.
NSLookup
Hostname Enter a host name and find out the corresponding IP address. It will also
do reverse name lookup and find the host name for an IP address you
specify.
Event Notification
Set the URL for events on phone web GUI, and when the corresponding event occurs on the device, the
device will send the configured URL to SIP server. The dynamic variables in the URL will be replaced by
the actual values of the device before sending to SIP server, in order to achieve the purpose of events
notification. Here are the standards:
1. The IP address of the SIP server needs to be added at the beginning and separate the dynamic
variables with a "/".
2. The dynamic variables need to have a "$" at the beginning. For example: local=$local
3. If users need to add multiple dynamic variables in the same event, users could use "&" to connect with
different dynamic variables. For example: 192.168.40.207/mac=$mac&local=$local
4. When the corresponding event occurs on the device, the device will send the MAC address and phone
number to server address 192.168.40.207.
On Boot Completed Configures the event URL when phone boots up.
Incoming Call Configures the event URL when phone has an incoming call.
Outgoing Call Configures the event URL when phone has an outgoing call. (Available
for GSC3510 only)
Missed Call Configures the event URL when the device has new a missed call.
On Connected Configures the event URL when a call is established.
On Disconnected Configures the event URL when a call is disconnected.
Forward On Configures the event URL when the forward feature is enabled on the
device.
Forward Off Configures the event URL when the forward feature is disabled on the
device.
On Blind Transfer Configures the event URL when users transfer a call with blind transfer
on the device.
On Attended Transfer Configures the event URL when users transfer a call with attended
transfer on the device.
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Log On Configures the event URL when users log on the device successfully.
Log Off Configures the event URL when users log off the device.
On Register Configures the event URL when an account in the device is registered
successfully.
On Unregister Configures the event URL when an account in the device is unregistered.
LDAP Book
Connection Mode Selects which protocol will be used for LDAP searching, LDAP or LDAPS.
Server Address Configures the URI of the LDAP server.
Port Configures the LDAP server port. The default LDAP port number is 389.
Base DN Determines the LDAP search base. This is the location in the directory
where the search is requested to begin.
Example:
dc=grandstream, dc=com
ou=Boston, dc=grandstream, dc=com
User Name Configures the bind "Username" for querying LDAP servers. Some LDAP
servers allow anonymous binds in which case the setting can be left blank.
Password Specifies the bind "Password" for querying LDAP servers. The field can be
left blank if the LDAP server allows anonymous binds.
LDAP Name Attributes Configures the "name" attributes of each record which are returned in the
LDAP search result. This field allows the users to configure multiple space
separated name attributes.
Example:
cn sn description
LDAP Number Configures the "number" attributes of each record which are returned in the
Attributes LDAP search result. This field allows the users to configure multiple space
separated number attributes.
Example:
telephoneNumber
telephoneNumber Mobile
LDAP Mail Attributes Determines the "mail" attributes of each record which are returned in the
LDAP search result.
Example: mail
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LDAP Name Filter Configures the filter used for name lookups.
Examples:
(|(cn=%)(sn=%)) returns all records which has the "cn" or "sn" field starting
with the entered prefix;
(!(sn=%)) returns all the records which do not have the "sn" field starting
with the entered prefix;
(&(cn=%) (telephoneNumber=*)) returns all the records with the "cn" field
starting with the entered prefix and "telephoneNumber" field set.
LDAP Number Filter Defines the filter used for number lookups.
Examples:
(|(telephoneNumber=%)(Mobile=%) returns all records which has the
"telephoneNumber" or "Mobile" field starting with the entered prefix;
(&(telephoneNumber=%) (cn=*)) returns all the records with the
"telephoneNumber" field starting with the entered prefix and "cn" field set.
LDAP Mail Filter Determines the filter used for mail lookups.
Example: (mail=%)
Max Hits Specifies the maximum number of results to be returned by the LDAP
server. If set to 0, server will return all search results. Default setting is 50.
Search Timeout (s) Configures the interval (in seconds) for the server to process the request
and client waits for server to return. The default setting is 4 seconds.
LDAP Lookup For Dial Sets the device to do the LDAP number searching when making outgoing
calls. The default setting is "No".
LDAP Lookup For Sets the device to do LDAP number searching for incoming calls. The
Incoming Call default setting is "No".
LDAP Dialing Default Configures the default account that being used when dialing LDAP contact.
Account Users may choose the Account 1-16, the default setting is “Default”.
Recording
Notes:
Operation
A recording file cannot be deleted if it is locked.
Users can delete a bench of recording files by clicking on “Delete”.
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Music
Audio file address Set up the stream’s RTSP address. it supports playing audio files in .ts
(.mp3) format. For example: rtsp://ip_address/musicFile.ts.
Select the audio play mode:
"Manual" (Default): Click Start/Stop to play/halt music. Music will
automatically stop when there is an incoming call, and will not
Play mode continue to play after the end of the call.
"Auto": The audio file will be played automatically. When there is
an incoming call, the audio file will stop playing and continue to
play after the call.
Manual operation Click to Play/Stop the stream: Only available on Manual Play mode.
When Play mode parameter is set to Auto, users can schedule the time
and date to play the audio file. This is by configuring the Cycle Time
parameter (Appears only when Play mode is set to Auto). This parameter
Auto mode play time
can either be set to All days, this will then trigger the function all days. Or
to Period of Time, which will trigger the function on the period of time
according to the user’s schedule. The default value is All days.
Audio loop test is used to test the three MICs available on the GSC3510. Each one of the MICs is tested
separately.
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Figure 54: Device Detection - Audio Loop Test
Built-in Speaker Test is used to test the GSC3510/GSC3505 by playing a piece of music in order to verify
the sound quality.
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LED Test
Led Test is used to test the availability of the four colored LEDs and their intensity. The colors of LEDs
available on the GSC3510/GSC3505 are: Green, Red, Blue and White.
Certificate Verify
Reset Button Test is used to test the Reset button, during the test the reset button doesn’t trigger factory
reset, this feature allows the user to check if the button is responding.
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Figure 57: Device Detection - Reset Button Test
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EXPERIENCING THE GSC3510/GSC3505
Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware
releases, additional features, FAQs, documentation and news on new products.
We encourage you to browse our product related documentation, FAQs and User and Developer Forum
for answers to your general questions. If you have purchased our products through a Grandstream Certified
Partner or Reseller, please contact them directly for immediate support.
Our technical support staff is trained and ready to answer all of your questions. Contact a technical support
member or submit a trouble ticket online to receive in-depth support.
Thank you again for purchasing Grandstream SIP Speaker, it will be sure to bring convenience and color
to both your business and personal life.
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