Grandstream Networks, Inc.
GXP2120/GXP2110/GXP2100/GXP1450
SIP Enterprise Phones
GXP2120/GXP2110/GXP2100/GXP1450 USER MANUAL
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VOICE MESSAGES (MESSAGE WAITING INDICATOR)...........................................................29
SHARED CALL APPEARANCE (SCA) ........................................................................................29
CALL FEATURES ................................................................................................................................30
CUSTOMIZED LCD SCREEN & XML.................................................................................................31
CONFIGURATION VIA KEYPAD.........................................................................................................33
CONFIGURATION VIA WEB BROWSER ...........................................................................................37
DEFINITIONS ......................................................................................................................................37
STATUS PAGE DEFINITIONS .....................................................................................................38
ACCOUNT PAGE DEFINITIONS .................................................................................................38
SETTINGS/BASIC SETTINGS PAGE..........................................................................................46
SETTINGS/ADVANCED SETTINGS PAGE.................................................................................53
NAT SETTINGS ...................................................................................................................................60
WEATHER UPDATE............................................................................................................................61
PUBLIC MODE ....................................................................................................................................62
EDITING CONTACTS AND CLICK-TO-DIAL ......................................................................................62
UPGRADE VIA KEYPAD MENU .........................................................................................................65
UPGRAGE VIA WEB GUI....................................................................................................................65
NO LOCAL TFTP/HTTP SERVERS ....................................................................................................66
CONFIGURATION FILE DOWNLOAD................................................................................................66
Table of Tables
GXP2120/GXP2110/GXP2100/GXP1450 User Manual
Table 1: GXP2120/GXP2110/GXP2100/GXP1450 FEATURES IN A GLANCE............................................8
Table 2: GXP2120/GXP2110/GXP2100/GXP1450 COMPARISON GUIDE .................................................9
Table 3: GXP2120/GXP2110/GXP2100/GXP1450 TECHNICAL SPECIFICATIONS...................................9
Table 4: GXP2120/GXP2110/GXP2100/GXP1450 EQUIPMENT PACKAGING ........................................12
Table 5: GXP2120/GXP2110/GXP2100/GXP1450 CONNECTORS ..........................................................12
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Table 6: LCD DISPLAY DEFINITIONS........................................................................................................15
Table 7: LCD ICONS...................................................................................................................................17
Table 8: KEYPAD DEFINITIONS ................................................................................................................20
Table 9: CALL FEATURES..........................................................................................................................30
Table 10: CONFIGURATION MENU...........................................................................................................33
Table of Figures
GXP2120/GXP2110/GXP2100/GXP1450 User Manual
Figure 1: GXP2120/GXP2110 Extension Board Connection......................................................................13
Figure 2: Keypad MENU Flow.....................................................................................................................36
Figure 3: Web GUI - Contacts.....................................................................................................................63
Figure 4: Click-to-Dial..................................................................................................................................64
GUI Interface Examples
GXP2120/GXP2110/GXP2100/GXP1450 User Manual
1. Screenshot of Configuration Login Page
2. Screenshot of Status Page
3. Screenshot of Basic Setting Configuration Page
4. Screenshot of Advanced User Configuration Page
5. Screenshot of SIP Account Configuration Page
6. Screenshot of Saved Configuration Changes Page
7. Screenshot of Reboot Page
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GNU GPL INFORMATION
GXP2120/GXP2110/GXP2100/GXP1450 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:
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CHANGE LOG
This section documents significant changes from previous versions of user manuals for
GXP2120/GXP2110/GXP2100/GXP1450. Only major new features or major document updates are listed
here. Minor updates for corrections or editing are not documented here.
FIRMWARE VERSION 1.0.4.23
•
•
Updated XML Application documentation link. [CUSTOMIZED LCD SCREEN & XML]
Added "Use Privacy Header" and "Use P-Preferred-Identity Header" options in web GUI. [ACCOUNT
•
•
Added Date Display Format options in web GUI. [SETTINGS/BASIC SETTINGS PAGE]
Added "Use Phonebook Key for LDAP Search" option in web GUI. [SETTINGS/ADVANCED
•
•
Updated generic config file cfg.xml information. [CONFIGURATION FILE DOWNLOAD]
FIRMWARE VERSION 1.0.4.9
•
•
•
Added instructions for connecting the phone. [CONNECTING YOUR PHONE]
Updated Keypad MENU options and Keypad configuration flow. [CONFIGURATION VIA KEYPAD]
Added Line Key options Speed Dial, Dial DTMF, Call Return, Transfer, Voice Mail, Intercom, Call Park,
LDAP Search, and etc. [SETTINGS/BASIC SETTINGS PAGE]
•
Added Multi Purpose Key options Voice Mail, Call Park, LDAP Search, and etc. [SETTINGS/BASIC
•
•
•
•
•
•
•
•
•
Added IPv6 configuration options. [SETTINGS/BASIC SETTINGS PAGE]
Added LDAP options information. [SETTINGS/ADVANCED SETTINGS PAGE]
Added Public Mode information for hot desking feature. [PUBLIC MODE]
Added NAT setting information. [NAT SETTINGS]
Added Matching Incoming Caller ID function in Account Setting. [ACCOUNT PAGE DEFINITIONS]
Added Editing Contacts and Click-to-Dial information. [EDITING CONTACTS AND CLICK-TO-DIAL]
Added GNU GPL information. [GNU GPL INFORMATION]
Added Change Log for this user manual. [CHANGE LOG]
Merging GXP1450 User Manual content with GXP2120/GXP2110/GXP2100 User Manual.
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WELCOME
Thank you for purchasing Grandstream GXP2120/GXP2110/GXP2100/GXP1450 SIP Enterprise Phones.
Your Grandstream GXP2120/GXP2110/GXP2100/GXP1450 Enterprise IP phone is feature-enriched,
sophisticated, yet simple to use. It delivers superior HD audio quality, rich and leading edge telephony
features, personalized information and customizable application service, automated provisioning for easy
deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP
devices and leading SIP/NGN/IMS platforms.
The GXP2120/GXP2110/GXP2100/GXP1450 supports a broad range of codecs, security protection, PoE,
dual 10/100mbps Ethernet ports, along with customizable XML provisioning and application
features. Users can expect superior audio quality using the new high definition handset, hands-free
speakerphone, or headset. Also, it can support up to 5-way conferencing for
GXP2120/GXP2110/GXP2100 and 3-way conferencing for GXP1450, multi-languages and dual-color
LEDs. The GXP2120/GXP2110/GXP2110 also supports presence and Busy Lamp Field (BLF) in the Multi
Purpose Keys. It presents a large easy-to-read backlit graphical display along with multiple XML keys to
further enhance the user experience. Additionally, the GXP2120/2110 is expandable with one to two
expansion modules.
The GXP2120/GXP2110/GXP2100/GXP1450 is a perfect choice for enterprise users looking for a high
quality, feature rich multi-line IP phone with the best values.
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 User Manual, could void your manufacturer warranty.
Warning:
Please do not use a different power adaptor with the GXP2120/GXP2110/GXP2100/GXP1450 as it may
cause damage to the products and void the manufacturer warranty.
This document is subject to change without notice. The latest electronic version of this user manual is
available for download here:
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for
any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.
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PRODUCT OVERVIEW
FEATURE HIGHTLIGHTS
Table 1: GXP2120/GXP2110/GXP2100/GXP1450 FEATURES IN A GLANCE
GXP2120 is an executive SIP phone. It features:
GXP2120
6 lines
7 programmable Multi Purpose Keys
4 XML programmable soft keys
GXP2110 is an executive SIP phone. It features:
GXP2110
4 lines
18 programmable Multi Purpose Keys
3 XML programmable soft keys
GXP2100 is an executive SIP phone. It features:
GXP2100
4 lines
7 programmable Multi Purpose Keys
3 XML programmable soft keys
GXP1450 is an executive SIP phone. It features:
GXP1450
2 lines
3 XML programmable soft keys
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Table 2: GXP2120/GXP2110/GXP2100/GXP1450 COMPARISON GUIDE
Features
GXP2120
GXP2110
GXP2100
GXP1450
LCD Display
320x160 pixel
240x120 pixel
180x90 pixel
180x60 pixel
Number of Lines
6
7
4
4
7
2
Programmable
Hard Keys
18
N/A
Programmable
Soft Keys
4
3
3
3
Extension Module Yes,
Yes,
N/A
N/A
up to 2 Expansion up to 2 Expansion
Modules,
Modules,
56 nodes each
56 nodes each
GXP2120/GXP2110/GXP2100/GXP1450 TECHNICAL SPECIFICATIONS
Table 3: GXP2120/GXP2110/GXP2100/GXP1450 TECHNICAL SPECIFICATIONS
Protocols
Standards
and SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS
(A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE,
TR-069, 802.1x, LLDP, IPv6, TLS, SRTP
SIP PUBLISH method (RFC3903), SIP Presence Package (RFC3856, 3863) for
use of MPKs, SIP Dialog Package (RFC4235), SIP Message method (RFC3428)
Note:
MPKs are not applicable for GXP1450.
Network Interfaces
Graphic Display
Dual switched 10/100Mbps port, integrated PoE
LCD size comparison:
GXP2120
GXP2110
GXP2100
GXP1450
320x160 pixel
240x120 pixel
180x90 pixel
180x60 pixel
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Feature Keys
Feature key comparison:
GXP2120 GXP2110
Yes
SPEAKERPHONE Yes
GXP2100 GXP1450
HOLD
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Yes
No
SEND
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
TRANSFER
CONF
MUTE
DND
HEADSET
INTERCOM
PHONEBOOK
MSG
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
MENU
NAVIGATION (4)
Voice Codec
Support for G.723.1, G.729A/B, G.711u/a, G.726-32, G.722 (wide-band), iLBC,
in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Telephony Features Hold, transfer, forward, up to 5-way conference for GXP2120/GXP2110/GXP2100,
up to 3-way conference for GXP1450, call park, pickup, shared-call-appearance
(SCA)/bridged-line-appearance (BLA), downloadable phone book (XML, LDAP, up
to 2000 items for GXP2120/GXP2110/GXP2100, up to 500 items for GXP1450),
call waiting, call log (up to 2000 records for GXP2120/GXP2110/GXP2100, up to
500 records for GXP1450), XML customization of screen, off-hook auto dial, auto
answer, click-to-dial, flexible dial plan, hot desking, personalized music ringtones,
server redundancy and fail-over
HD Audio
Yes, HD handset with support for wideband audio
2.5mm and RJ9 headset jack
Headset Jack
Base Stand
Wall Mountable
Expansion Module
QoS
Yes, 2-angle positions available
Yes
GXP2120/GXP2110 only
Layer 2 (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, 802.1x media
access control
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Multi-language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
Simplified Chinese, traditional Chinese, Korean, Japanese, and etc
Upgrade
and Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or
Provisioning
AES encrypted XML configuration file
Power and Green Universal power adapter:
Energy Efficiency
Input: 100-240VAC 50-60Hz; Output: 5VDC, 800mA
Integrated Power-over-Ethernet (Built-in auto-sensing: Cisco and IEEE 802.3af
standard)
Max power consumption 2.5W (power adapter) or 3W (PoE)
Physical
GXP2120
GXP2110
GXP2100
GXP1450
Dimension
251mm(W)
252mm (W)
222mm (W)
186mm (W)
x 202mm(L) x 210mm (L) x 210mm (L) x 210mm (L)
x 77mm(H)
1.08KG
x 78mm (H)
1.08KG
x 93mm (H)
0.98KG
x 81mm (H)
0.8KG
Unit Weight
Package Weight
1.78KG
1.78KG
1.63KG
1.3KG
Operating
32-104 oF / 0-40 oC, 10-90% (non-condensing)
Temperature
Humidity
and
Package Content
Compliance
GXP2120/GXP2110/GXP2100/GXP1450 phone, handset with cord, base stand,
wall mount spacers, universal power supply, network cable, quick start guide
FCC Part 15 (CFR 47) Class B; EN55022 Class B, EN55024, EN61000-3-2,
EN61000-3-3, EN60950-1; AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS;
UL 60950 (power adapter)
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INSTALLATION
EQUIPMENT PACKAGING
Table 4: GXP2120/GXP2110/GXP2100/GXP1450 EQUIPMENT PACKAGING
Main Case
Yes (1)
Yes (1)
Yes (1)
Yes (1)
Yes (1)
Yes (2)
Yes (2)
Yes (1)
Handset
Phone Cord
Power Adaptor
Ethernet Cable
Phone Stand
Wall Mount Spacers
Quick Start Guide
CONNECTING YOUR PHONE
Table 5: GXP2120/GXP2110/GXP2100/GXP1450 CONNECTORS
RJ9 handset connector port
Handset Port
Headset Port
LAN Port
2.5mm and RJ9 headset connector port
10/100Mbps RJ-45 port connecting to Ethernet, integrated PoE
10/100Mbps RJ-45 port for PC connection
Extension module connection port for GXP2120/GXP2110
5V DC Power connector port
PC Port
EXT Port
Power Jack
To set up the GXP2120/GXP2110/GXP2100/GXP1450, follow the steps below:
1. Attach the phone stand or wall mount to the back of the phone where there are slots;
2. Connect the handset and main phone case with the phone cord;
3. Connect the LAN port of the phone to the RJ-45 socket of a hub/switch or a router (LAN side of the
router) using the Ethernet cable;
4. Connect the 5V DC output plug to the power jack on the phone; plug the power adapter into an
electrical outlet. If PoE switch is used in step 3, this step could be skipped;
5. The LCD will display provisioning or firmware upgrade information. Before continuing, please wait for
the date/time display to show up;
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6. Using the keypad configuration menu or phone's embedded web server (Web GUI) by entering the IP
address in web browser, you can further configure the phone.
GXP2120/GXP2110 EXTENSION MODULE
GXP2120/2110 supports two extension units, providing up to 112 additional programmable extensions.
Each GXP Extension unit has 56 multi-purpose keys, dual color LEDs (red/green) and support BLF (Busy
Lamp Field) and Presence.
GXP2120/2110 Extension package contains:
•
•
1 GXP Extension unit
2 connection cables
GXP2120 with GXP Extension
GXP Extension
Reverse side of connection with connection
plate
GXP2120/2110 has a special port on
the back
Figure 1: GXP2120/GXP2110 Extension Board Connection
Connect the first GXP EXT to the GXP2120/2110 using the connection cable found in the GXP Extension
package. The first GXP EXT draws power directly from the phone. Connect the second GXP Extension
unit using the connection plate and the connection cable. The GXP2120/2110 will automatically reboot and
power up the GXP Extensions. Grandstream recommends, though not required, to use a separate power
supply with the second GXP EXT.
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Note:
•
•
Should your system lose power, please unplug your devices and power up the GXP2120/2110 first.
Extension for GXP2120/2110 is the same for GXP2020/2010 models. However, GXP2120/2110 uses a
different-shaped connector for the special port (as shown above). Extension cables will be included
with the extension board.
•
•
Extension for GXP2120/2110 does not support hot-swap. Once connected, user should reboot the
phone to ensure the set up will work correctly.
GXP2120/2110 can drive 2 extension modules. Independent power adapters are not needed for
extension modules.
SAFETY COMPLIANCES
The GXP2120/GXP2110/GXP2100/GXP1450 phone complies with FCC/CE and various safety standards.
The GXP2120/GXP2110/GXP2100/GXP1450 power adapter is compliant with the UL standard. Use the
universal power adapter provided with the GXP2120/GXP2110/GXP2100/GXP1450 package only. The
manufacturer’s warranty does not cover damages to the phone caused by unsupported power adapters.
WARRANTY
If the GXP2120/GXP2110/GXP2100/GXP1450 phone was purchased from a reseller, please contact the
company where the phone was purchased for replacement, repair or refund. If the phone was purchased
directly from Grandstream, contact the Grandstream Sales and Service Representative for a RMA (Return
Materials Authorization) number before the product is returned. Grandstream reserves the right to remedy
warranty policy without prior notification.
Warning:
Use the power adapter provided with the phone. Do not use a different power adapter as this may damage
the phone. This type of damage is not covered under warranty.
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USING THE GXP2120/GXP2110/GXP2100/GXP1450
GETTING FAMILAR WITH THE LCD
GXP2120/GXP2110/GXP2100/GXP1450 has a dynamic and customizable screen. The screen displays
differently depending on whether the phone is idle or in use (active). The following table describes the
items displayed on the GXP2120/GXP2110/GXP2100/GXP1450 idle screen.
Table 6: LCD DISPLAY DEFINITIONS
Displays the current date and time. It can be synchronized with Internet time
DATE AND TIME
servers.
Displays company logo/name. This logo/name can be customized via xml screen
LOGO
customization. The maximum size for logo name string is 26 characters in
English (approximately).
Shows the status of network in the middle of the screen. It will indicate whether
the network is down or starting.
NETWORK STATUS
STATUS ICON
Shows the status of the phone for registration status, call features and etc, using
icons as shown in the next table.
LINE
STATUS
Displays the name of the account that is in use.
INDICATOR
The softkeys are context sensitive and will change depending on the status of
the phone. Typical functions assigned to softkeys are:
•
SwitchSCR
Toggles between different idle screens: default idle screen, weather
information, stock information and currency information.
SOFTKEYS in Idle
Screen
•
•
•
ForwardAll
Unconditionally forwards the phone line (account 1) to another phone.
MissedCalls
Shows up unanswered calls to this phone.
Redial
Redials the last dialed number in idle screen when there is existed call log.
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Note:
If XML application is used for GXP2120/GXP2110/GXP2100, the softkey for
XML application will show up in the default idle screen as configured.
The softkeys are context sensitive and will change depending on the call status
of the phone.
•
•
•
•
•
•
•
Redial
Redials the last dialed number after off hook when there is existed call log.
Dial
Dials the call out after off hook and entering the number.
Hold
Puts the current active call on hold.
AnswerCall
Answers the incoming call when the phone is ringing.
RejectCall
SOFTKEYS in Call
Screen
Rejects the incoming call when the phone is ringing.
EndCall
Ends the active call.
Transfer
Transfer softkey will show up after pressing TRAN button and entering
transfer target number. Press Transfer softkey to do blind transfer.
•
Split
In auto-attended transfer mode, after establishing the second call, press Split
to quit transfer and go back to normal talking status.
•
•
ConfCall
Conferences the active calls.
ReConf
Re-establish the conference among the calls on hold.
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Call Parking: Please refer to GXE5024/5028 Online User Manual for more
information.
•
CallPark
When the phone dials out, the Call Park softkey will display on screen. To
park the call, press the "Call Park" softkey and select MPK to park the call.
•
PickUp
When the phone goes off-hook, the Call Pickup softkey will display on
screen. To pick up the parked call, press the "Call Pickup" softkey.
SPECIAL
SOFTKEYS
(Only for GXP2120/
GXP2110/GXP2100
Call Queue: Please refer to GXE5024/5028 Online User Manual for more
information.
when
integrated
with GXE5024/5028)
•
SignIn
Press this button to sign in to the call queue. Agent will be prompted in the
LCD display to select the call queue to join. Press MENU button on keypad
to select OK. Once the agent completely signs in, the agent will be brought
back to the main screen.
•
SignOut
Press this button to sign out of the call queue. Press MENU button on
keypad to select OK. This will be displayed once the agent is signed in to the
call queue.
Table 7: LCD ICONS
DND Status.
OFF - Do Not Disturb disabled
ON - Do Not Disturb enabled
Call Forward All Status.
OFF - Call Forward All feature disabled
ON - Call Forward All feature enabled
Call Forward Busy Status.
OFF - Call Forward Busy feature disabled
ON - Call Forward Busy feature enabled
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Call Forward No Answer Status.
OFF - Call Forward No Answer feature disabled
ON - Call Forward No Answer feature enabled
Call Forward All and Call Forward No Answer Status.
OFF - Call Forward All and Call Forward No Answer feature disabled
ON - Call Forward All and Call Forward No Answer feature enabled
Keypad Status.
OFF - keypad is unlocked
ON - keypad is locked
Enter Keypad Unlock Password.
Voicemail Status.
OFF - No new voicemail
ON - New voicemail
Instant Message.
OFF - No new instant message
ON - New instant message
Voice Mail and Instant Message.
OFF - No new instant message or voicemail
ON - New instant message and voicemail
Network Status.
OFF - Network connection is up
ON - Network connection is down
Missed Call Status.
OFF - No missed call
ON - New missed call
Save Call Record Status.
OFF - No saving call record activity
ON - The phone is writing the call records into the flash at the moment
Waiting For Response.
Please wait for the phone to response before the keypad entry
Handset Status.
OFF - handset on hook
ON - handset off hook
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Speaker Status.
OFF - speaker off
ON - speaker on
Headset Status.
OFF - headset off
ON - headset on
Calling Out.
The phone is calling out
Calling In.
Phone is ringing with incoming call
Incoming Call.
The current call is an incoming call
Outgoing Call.
The current call is an outgoing call
Call Failed
Fail to establish call
SRTP Status.
OFF - SRTP is not used
ON - SRTP is used
MUTE Status.
OFF - No muted
ON - Muted
Call On Hold.
Call Active.
Conference Call.
Core Dump.
Core dump file can be downloaded from phone's Web GUI->Status page.
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GETTING FAMILAR WITH THE KEYPAD
The following table describes the buttons used on the GXP2120/GXP2110/GXP2100/GXP1450 keypad.
Table 8: KEYPAD DEFINITIONS
GXP21xx
GXP1450
Definition
LINE KEYS
HOLD
Open or switch line.
Place active call on hold, or resume the call on hold.
Send/Redial.
•
Send. Enter the digits and then press Send to dial out the
number;
SEND
•
Redial. Redial when there is a previously dialed call.
TRANSFER
CONF
TRANSFER
CONF
Transfer an active call to another number.
Establish conference call with other parties.
MUTE
Mute. Press to mute/unmute an active call.
Headset.
HEADSET
Do-Not-Disturb. Press to enable/disable DND feature.
For GXP2120/GXP2110, a dedicated DND button is available. For
GXP1450, MUTE button can be used as DND button when the phone
is in idle screen.
DND
INTERCOM
N/A
Paging/Intercom the configured Intercom User ID.
Navigation Keys/Menu.
•
•
Press the 4 navigation keys to move up/down/left/right;
Press the round button in the center to enter Keypad
Configuration MENU when phone is in idle;
NAVIGATION KEYS/MENU
The round button "MENU" can also be used as ENTER key when in
Keypad Configuration.
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PHONEBOOK/
CONTACT
Display phonebook list and options.
Volume (For GXP2100/GXP1450 only).
On GXP2100/GXP1450, press "-" or "+" to adjust the volume;
On GXP2120/GXP2110, when the phone is off hook, press UP or
DOWN button to adjust the volume.
0 - 9, *, #
Standard phone keypad.
Configure key mode and User ID to use the Multi Purpose key as
Speed Dial, BLF, Call Park and etc.
Multi Purpose Keys
MAKING PHONE CALLS
HANDSET, SPEAKER AND HEADSET MODE
The GXP2120/GXP2110/GXP2100/GXP1450 allows users to switch among handset, speaker or headset
when making calls. Press the Hook Switch to switch to handset; press the Headset button to switch to
headset; or press the Speaker button
to switch to speaker.
MULTIPLE SIP ACCOUNTS AND LINES
GXP2120 can support up to 6 independent SIP accounts; GXP2110/GXP2100 can support up to 4
independent SIP accounts; GXP1450 can support up to 2 independent accounts. Each account is capable
of independent SIP server, user and NAT settings. Each of the line buttons is "virtually" mapped to an
individual SIP account. In off hook state, select an idle line and the dial tone will be heard.
To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. The
user can switch lines before dialing any number by pressing the LINE buttons.
For example: If 2 independent accounts are registered on GXP2120/GXP2110/GXP2100/GXP1450, when
LINE 1 is pressed, the LINE 1 LED will light up in green. If LINE 2 is pressed, the LINE 2 LED will light up in
green and the subsequent call will be made through SIP account 2.
Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When the
"virtually" mapped line is in use, the phone will flash the other available LINE in red. A line is ACTIVE when
it is in use and the corresponding LED is red.
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COMPLETING CALLS
There are several ways to complete a call on GXP2120/GXP2110/GXP2100/GXP1450.
•
On hook dialing. Enter the number when the phone is on hook and then send out.
When the phone is in idle, enter the number to be dialed out;
Take handset off hook; or
Press Speaker button; or
Press Headset button with headset plugged in; or
Select an available LINE key;
The call will be dialed out.
•
Off hook and dial. Off hook the phone, enter the number and send out.
Take handset off hook; or
Press Speaker button; or
Press Headset button with headset plugged in; or
Press an available LINE key to activate speaker;
You shall hear dial tone after off hook;
Enter the number;
Press SEND key or # to dial out.
•
Redial. Redial the last dialed number.
Take handset off hook; or
Press Speaker button; or
Press Headset button with headset plugged in; or
Press an available LINE key to activate speaker; or
When the phone is in idle;
Press SEND key, or the REDIAL softkey.
•
Via Call History. Dial the number logged in phone's call history.
Press MENU button to bring up the main menu;
Enter Call History and select "Answered Calls", "Missed Calls", "Transferred Calls" or "Forwarded
Calls";
Select the entry you would like to call using the navigation arrow keys;
Press SEND button to dial out.
•
Via Phonebook. Dial the number from the phonebook.
Press MENU button to bring up the main menu;
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Select and enter Phonebook;
Select the phonebook entry you would like to call using the navigation arrow keys. Press MENU
button to enter the entry detail;
Select option "Dial" using the navigation arrow keys and then press MENU button again to dial out.
•
Speed Dial. Dial the number configured as Speed Dial on Line Key.
Go to phone's Web GUI->Basic Settings, configure the Line Key's Key Mode as Speed Dial. Select
the account to dial from, enter the Name and User ID (the number to be dialed out) for the Line
Key. Click on "Update" at the bottom of the Web GUI page;
Off hook the phone, or directly press the Speed Dial key to dial out.
•
•
Call Return. Dial the last answered call.
Go to phone's Web GUI->Basic Settings, configure the Line Key's Key Mode as Call Return.
Select the account to dial from, no Name or User ID has to be set on for Call Return;
Off hook the phone, or directly press the Call Return key to dial out.
Via Page/Intercom.
Take handset off hook; or
Press Speaker button; or
Press Headset button with headset plugged in; or
Press an available LINE key to activate speaker;
You shall hear dial tone after off hook;
Press MENU button to switch the call screen from "Dialing" to "Paging";
Enter the number;
Press SEND key or # to dial out.
Note:
•
After entering the number, the phone waits for the No Key Entry Timeout (Default timeout is 4 seconds,
configurable via Web GUI) before dialing out. Press SEND or # key to override the No Key Entry
Timeout;
•
•
•
If digits have been entered after handset is off hook, the SEND key will works as SEND instead of
REDIAL;
By default, # can be used as SEND to dial the number out. Users could disable it by setting "User # as
Dial Key" to "No" from Web GUI->Account page;
For Paging/Intercom, if the SIP Server/PBX supports the feature and has Paging/Intercom feature
code set up already, users might not necessarily need toggle to paging mode in the call screen of the
phone. Simply dial the feature code with extension as a normal call.
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MAKING CALLS USING IP ADDRESSES
Direct IP Call allows two phones to talk to each other in an ad-hoc fashion without a SIP proxy. VoIP calls
can be made between two phones if:
•
•
•
Both phones have public IP addresses; or
Both phones are on the same LAN/VPN using private or public IP addresses; or
Both phones can be connected through a router using public or private IP addresses (with necessary
port forwarding or DMZ).
To make a direct IP call, please follow the steps below:
•
•
•
•
•
Press MENU button to bring up main menu;
Select "Direct IP Call" using the navigation arrow keys;
Press MENU to enter the Direct IP Call mode;
Input the 12-digit target IP address (Please see example below);
Press the "More" softkey to make sure the softkey selection "IPv4" or "IPv6" is correctly selected
depending on your network environment;
•
Press "OK" softkey to dial.
For example:
If the target IP address is 192.168.1.60 and the port is 5062 (i.e., 192.168.1.60:5062), input the following:
192*168*1*60#5062. The * key represents the dot (.), the # key represents colon (:). Wait for about 4
seconds and the phone will initiate the call.
Quick IP Call Mode:
The GXP2120/GXP2110/GXP2100/GXP1450 also supports Quick IP Call mode. This enables the phone
to make direct IP calls using only the last few digits (last octet) of the target phone's IP address. This is
possible only if both phones are under the same LAN/VPN. This simulates a PBX function using the
CMSA/CD without a SIP server. Controlled static IP usage is recommended.
To enable Quick IP Call Mode, go to phone's Web GUI->Advanced Setting page, set "Use Quick IP Call
Mode" to "Yes". Click on "Update" on the bottom of the Web GUI page to take the change. To make Quick
IP Call, take the phone off hook first. Then dial #xxx where x is 0-9 and xxx<255. Press # or SEND and a
direct IP call to aaa.bbb.ccc.XXX will be completed. "aaa.bbb.ccc" is from the local IP address regardless
of subnet mask. The number #xx or #x are also valid. The leading 0 is not required (but it's OK).
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For example:
•
•
•
•
192.168.0.2 calling 192.168.0.3 -- dial #3 followed by # or “SEND”;
192.168.0.2 calling 192.168.0.23 -- dial #23 followed by # “SEND”;
192.168.0.2 calling 192.168.0.123 -- dial #123 followed by # “SEND”;
192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3.
Note:
•
•
The # will represent colon ":" in direct IP call rather than SEND key as in normal phone call;
If you have a SIP server configured, direct IP call still works. If you are using STUN, direct IP call will
also use STUN;
•
Configure the "User Random Port" to "No" when completing direct IP calls.
ANSWERING PHONE CALLS
RECEIVING CALLS
•
•
Single incoming call. Phone rings with selected ring tone. The corresponding LINE key will flash in
red. Answer call by taking handset off hook, or using Speaker/Headset, or pressing the flashing LINE
key;
Multiple incoming calls. When another call comes in while having an active call, the phone will
produce a Call Waiting tone (stutter tone). The other LINE key will flash in red. Answer the incoming
call by pressing the flashing LINE key. The current active call will be put on hold automatically.
DO NOT DISTURB
Do Not Disturb can be enabled/disabled by pressing the DND button on the phone when the phone is in
idle (For GXP2120/GXP2110 only. On GXP1450, press the MUTE button). Or users could set it from
phone's Menu following the steps below:
•
•
•
•
Press the Menu button and select "Preference" using navigation keys;
Press Menu button again to get into Preference options;
Select "Do Not Disturb" and press Menu button;
Use arrow keys to select and press Menu button to enable or disable "Do Not Disturb" feature.
When Do Not Disturb feature is turned on, the DND icon will appear on the right side of the LCD. The
incoming call will not be accepted or the call will directly go into voicemail.
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DURING A PHONE CALL
CALL WAITING/CALL HOLD
•
•
•
Hold. Place a call on hold by pressing the HOLD button. The active LINE key will blink in green;
Resume. Resume call by pressing the blinking LINE key;
Multiple calls. Automatically place active call on hold or switch between calls by pressing the LINE
key. Call waiting tone (stutter tone) will be audible on incoming call during the active call.
MUTE
During an active call, press the MUTE button to mute/unmute the microphone. The LCD will show "
Talking" or "MUTE" to indicate the mute status, with Mute icon displayed on the screen.
CALL TRANSFER
GXP2120/GXP2110/GXP2100/GXP1450 supports Blind Transfer, Attended Transfer and Auto-Attended
Transfer.
•
•
Blind Transfer.
During the first active call, press TRAN key and dial the number to transfer to;
Press SEND key or # to complete transfer of active call.
Attended Transfer.
During the first active call, press LINE key. The first call will be put on hold;
Enter the number for the second call in the new line and establish the call;
Press TRAN key;
Press the other LINE key which is on hold to transfer the call.
•
Auto-Attended Transfer.
Set "Auto-Attended Transfer" to "Yes" under Web GUI->Advanced Settings page. And then click
"Update" on the bottom of the page;
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Establish one call first;
During the call, press TRAN key. A new line will be brought up and the first call will be
automatically placed on hold;
Enter the number and press SEND key to establish the second call;
After the second call is established, press TRAN key again. The call will be transferred;
If users press the SPLIT softkey before the call is transferred in the step above, the second call will
be resumed.
Note:
•
•
To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains.
In auto-attended transfer, use SEND key to dial out the second call instead of using #, even when #
could be used as SEND in normal phone calls.
5-WAY CONFERENCING
GXP2120/GXP2110/GXP2100 can host up to 5-way conference call with other parties; GXP1450 can host
up to 3-way conference call with other parties.
•
Initiate a conference call.
Establish calls with 2 or more parties respectively;
While 1 call is active, the other call will be put on hold with its LINE key blinking in green;
Press CONF key;
Press the desired LINE key on hold, the conference will be established;
Repeat the previous 2 steps for all the other parties to join the conference.
•
•
Cancel Conference.
If after press the CONF key, the user decides not to conference, press Cancel softkey or the
current active LINE key;
This will resume the 2-way conversation with the current line.
Split and Re-conference.
During the conference, press HOLD key. The conference call will be split and the calls will be put
on hold separately with the LINE keys blinking in green;
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Select 1 LINE key and press to resume the 2-way conversation;
If users would like to re-establish conference call, before 1 separate LINE is selected, press the
ReConf softkey right after the conference call is held/split;
•
End Conference.
Press HOLD key to split the conference call. The conference call will be ended with both calls on
hold; Or
Users could press the EndCall softkey or simply hang up the call to terminate the conference call.
GXP2120/GXP2110/GXP2100/GXP1450 supports Easy Conference Mode, which can be used combined
with the traditional way to establish the conference.
•
•
•
Initiate a conference call.
Establish 1 call;
Press CONF key and a new line will be brought up using the same account;
Dial the number and press SEND key to establish the second call;
Press CONF key or press the ConfCall softkey to establish the conference.
Join more parties in established conference call.
Establish conference call;
Press CONF key and a new line will be brought up using the same account;
Dial the number and press SEND key to establish the second call;
Press CONF key or press the ConfCall softkey to join the new party in the established conference.
Split and Re-conference.
During the conference, press HOLD key. The conference call will be split and both calls will be put
on hold separately with 2 LINE keys blinking in green;
Select 1 LINE key and press to resume the 2-way conversation;
If users would like to re-establish conference call, before 1 separate LINE is selected, press the
ReConf softkey right after the conference call is split.
•
Cancel Conference.
If users decides not to conference after establishing the second call, press EndCall softkey instead
of the ConfCall softkey/CONF key;
This will end the second call and the screen will show the first call on hold.
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•
End Conference.
Press HOLD key to split the conference call. The conference call will be ended with both calls on
hold; Or
Users could press the EndCall softkey or simply hang up the call to terminate the conference call.
Note:
•
The party that starts the conference call has to remain in the conference for its entire duration, you can
put the party on mute but it must remain in the conversation. Also, this is not applicable when the
feature "Transfer on Conference Hangup" is turned on.
•
•
The option "Disable Conference" has to be set to "No" to establish conference.
When using Easy Conference Mode, use SEND key to dial out the second call instead of using #,
even when # could be used as SEND in normal phone calls.
VOICE MESSAGES (MESSAGE WAITING INDICATOR)
A blinking red MWI (Message Waiting Indicator) on the top right corner of the phone indicates a message
is waiting. Dial into the voicemail box to retrieve the message by entering the voice mail number of the
server or pressing the Voicemail button (Voice Mail User ID has to be properly configured as the voice mail
number under Web GUI->Account page). An IVR will prompt the user through the process of message
retrieval.
SHARED CALL APPEARANCE (SCA)
The GXP2120/GXP2110/GXP2100/GXP1450 phone supports shared call appearance by Broadsoft
standard. This feature allows members of the SCA group to shared SIP lines and provides status
monitoring (idle, active, progressing, hold) of the shared line. When there is an incoming call designated
for the SCA group, all of the members of the group will be notified of an incoming call and will be able to
answer the call from the phone with the SCA extension registered.
All the users that belong to the same SCA group will be notified by visual indicator when a user seizes the
line and places an outgoing call, and all the users of this group will not be able to seize the line until the line
goes back to an idle state or when the call is placed on hold (with the exception of when multiple call
appearances are enabled on the server side).
In the middle of the conversation, there are two types of hold: Public Hold and Private Hold. When a
member of the group places the call on public hold, the other users of the SCA group will be notified of this
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by the red-flashing button and they will be able to resume the call from their phone by pressing the line
button. However, if this call is placed on private-hold, no other member of the SCA group will be able to
resume that call.
To enable shared call appearance, the user would need to register the shared line account on the phone.
In addition, they would need to navigate to “Settings”->”Basic Settings” on the web UI and set the line to
"Shared Line". If the user requires more shared call appearances, the user can configure multiple line
buttons to be "shared line" buttons associated with the account.
CALL FEATURES
The GXP2120/GXP2110/GXP2100/GXP1450 supports traditional and advanced telephony features
including caller ID, caller ID with caller Name, call forward and etc.
Table 9: CALL FEATURES
Block Caller ID (for all subsequent calls)
*30
*31
*67
*82
•
•
Off hook the phone;
Dial *30.
Send Caller ID (for all subsequent calls)
•
•
Off hook the phone;
Dial *31.
Block Caller ID (per call)
•
•
Off hook the phone;
Dial *67 and then enter the number to dial out.
Send Caller ID (per call)
•
•
Off hook the phone;
Dial *82 and then enter the number to dial out.
Disable Call Waiting (per Call)
*70
*71
•
•
Off hook the phone;
Dial *70 and then enter the number to dial out.
Enable Call Waiting (per Call)
Off hook the phone;
•
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•
Dial *71 and then enter the number to dial out.
Unconditional Call Forward. To set up unconditional call forward:
•
•
•
Off hook the phone;
*72
*73
*90
*91
*92
*93
Dial *72 and then enter the number to forward the call;
Press OK softkey or SEND key.
Cancel Unconditional Call Forward. To cancel the unconditional call forward:
•
•
•
Off hook the phone;
Dial *73;
Hang up the call.
Busy Call Forward. To set up busy call forward:
•
•
•
Off hook the phone;
Dial *90 and then enter the number to forward the call;
Press OK softkey or SEND key.
Cancel Busy Call Forward. To cancel the busy call forward:
•
•
•
Off hook the phone;
Dial *91;
Hang up the call.
Delayed Call Forward. To set up delayed call forward:
•
•
•
Off hook the phone;
Dial *92 and then enter the number to forward the call;
Press OK softkey or SEND key.
Cancel Delayed Call Forward. To cancel the delayed call forward:
•
•
•
Off hook the phone;
Dial *93;
Hang up the call.
CUSTOMIZED LCD SCREEN & XML
The GXP2120/GXP2110/GXP2100/GXP1450 IP phone supports the following XML applications. Please
refer to the corresponding link for documentation and templates.
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•
•
•
XML custom idle screen (customize idle screen logo, softkey layout, and etc.)
XML downloadable phonebook
XML Application (For GXP21xx only)
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CONFIGURATION GUIDE
The GXP2120/GXP2110/GXP2100/GXP1450 can be configured via two ways:
•
•
LCD Configuration Menu using the phone's keypad;
Web GUI embedded on the phone using PC's web browser.
CONFIGURATION VIA KEYPAD
To configure via the LCD configuration menu using phone's keypad, follow the instructions below:
•
•
Enter MENU options. When the phone is in idle, press the round MENU button to enter the
configuration menu;
Navigate in the menu options. Press the arrow keys up/down/left/right to navigate in the menu
options;
•
•
•
Enter/Confirm selection. Press the round MENU button to enter the selected option;
Exit. Press LEFT arrow key to exit to the previous menu;
The phone automatically exits MENU mode with an incoming call, when the phone is off hook or the
MENU mode if left idle for more than 20 seconds.
The MENU options are listed in the following table.
Table 10: CONFIGURATION MENU
Call History
Status
Displays call logs for answered calls, dialed calls, missed calls,
transferred calls and forwarded calls.
Displays network status, account registration status, software version
number, MAC address, hardware version number, P/N number.
•
Network status.
Press to enter the sub menu for IP setting information
(DHCP/Static IP/PPPoE), IPv4 address, IPv6 address, Subnet
Mask, Gateway and DNS server.
Phone Book
Displays phonebook. Users could add, edit, search and delete contacts
here, or download phonebook XML to the phone.
LDAP Directory
Configures LDAP directory options, displays LDAP directory by
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searching.
Instant Messages
Direct IP Call
Preference
Displays received instant messages.
Makes direct IP call.
Preference sub menu includes the following options:
•
Do Not Disturb
Enables/disables Do Not Disturb on the phone.
Forward Call
•
Configures call forward feature on selected account, forward
type and number.
•
•
•
•
•
Ring Tone
Configures different ring tones for incoming call.
Ring Volume
Adjusts ring volume by pressing left/right arrow key.
LCD Contrast
Adjusts LCD contrast by pressing left/right arrow key.
LCD Brightness
Adjusts LCD brightness by pressing left/right arrow key.
Download SCR XML
Triggers the phone to download the XML idle screen file
immediately. The XML idle screen server path and downloading
method need to be set up correctly in Web GUI->Advanced
Settings.
•
•
Erase Custom SCR
Erases custom XML idle screen previously loaded on the phone.
After erasing it, the phone will show default idle screen.
Display Language
Selects the language to be displayed on the phone. Users could
select Automatic for local language based on IP location if
available.
•
Time Settings
Configures date and time on the phone.
Config
Config sub menu includes the following options:
SIP
•
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Configures SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth
ID, SIP Password, SIP Transport and Audio information to
register SIP account on the phone.
Upgrade
•
Configures firmware server and config server for upgrading and
provisioning the phone.
•
•
Factory Reset
Resets the phone to factory default settings.
Layer 2 QoS
Configures 802.1Q/VLAN Tag and priority value.
Factory Functions
Factory Functions sub menu includes the following options:
•
•
•
Audio Loopback
Speak to the phone using speaker/handset/headset. If you can
hear your voice, your audio is working fine. Press Menu button
to exit audio loopback mode.
Diagnostic Mode
All LEDs will light up. Press any key (except MENU key) on the
keypad to display the button name in the LCD. Lift and put back
the handset or press Menu button to exit diagnostic mode.
Keyboard Diagnostic
Press all the available keys on the phone. The LCD will display
the name for the keys to be pressed to finish the keyboard
diagnostic mode.
Network
Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE account
ID and password; Configures IP address, Netmask, Gateway, DNS
Server 1 and DNS Server 2; Configures 802.1x mode.
Call Features
Voice Mails
Configures call forward features for Forward All, Forward Busy, Forward
No Answer and No Answer Timeout.
Displays voicemail message information in the format below:
new messages/all messages (urgent messages/all urgent messages)
Reboot
Exit
Reboot the phone.
Exit from this menu.
The following picture shows the keypad MENU configuration flow.
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Answered Calls
Dialed Calls
Missed Calls
Transferred Calls
Forwarded Calls
Clear All
First Name
Last Name
Number
Acct
Groups
MENU
Confirm Add
Cancel & Return
Back
Server Address
Port
Base
User Name
Password
LDAP Number Filter
LDAP Name Filter
LDAP Version
...
Groups
New Entry
Search
Download Phonebook XML
Delete All Entries
Back
Call History
Status
Phone Book
Search
LDAP Configuration
Back
LDAP
Enable DND
Disable DND
Back
Directory
Do Not Disturb
Forward Call
Ring Tone
Ring Volume
Instant
Default Ring
Ring1
Ring2
Ring 3
Back
Messages
LCD Contrast
LCD Brightness
Download SCR XML
Erase Custom SCR
Display Language
Time Settings
Back
Direct IP Call
Preference
Config
Account
SIP Proxy
Outbound Proxy
SIP User ID
SIP Auth ID
SIP Password
SIP Transport
Audio
SIP
Upgrade
Factory Reset
Layer 2 QoS
Back
Factory
Functions
Save
Cancel
Audio Loopback
Diagnostic Mode
Keyboard Diagnostic
Back
Network
Call Features
Voice Mails
Reboot
Firmware Server
Config Server
Upgrade Via
Back
IP Setting
PPPoE Settings
IP
Netmask
Gateway
DNS Server 1
DNS Server 2
802.1X
802.1Q/VLAN Tag
Priority value
Reset Vlan Config
Back
Exit
Back
Forward All
Forward Busy
Forward No Answer
No Answer Timeout
Account X
Figure 2: Keypad MENU Flow
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CONFIGURATION VIA WEB BROWSER
The GXP2120/GXP2110/GXP2100/GXP1450 embedded Web server responds to HTTP/HTTPS
GET/POST requests. Embedded HTML pages allow a user to configure the IP phone through a Web
browser such as Microsoft’s IE, Mozilla Firefox and Google Chrome.
To access the Web GUI:
1. Connect the computer to the same network as the phone;
2. Make sure the phone is turned on and shows its IP address. You may check the IP address by
pressing SwitchSCR softkey or go to MENU->Status;
3. Open a Web browser on your computer;
4. Enter the phone’s IP address in the address bar of the browser;
5. Enter the administrator’s login and password to access the Web Configuration Menu.
Note:
•
The computer has to be connected to the same sub-network as the phone. This can be easily done by
connecting the computer to the same hub or switch as the phone connected to. In absence of a
hub/switch (or free ports on the hub/switch), please connect the computer directly to the PC port on the
back of the phone;
•
If the phone is properly connected to a working Internet connection, the IP address of the phone will
display in MENU->Status. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a
number from 0-255. Users will need this number to access the Web GUI. For example, if the phone
browser;
•
The default login name for the administrator is "admin". The default administrator password is set to
"admin". The default login name for the end user is "user" while the default user password is set to
"123".
•
When changing any settings, always SUBMIT them by pressing the UPDATE button on the bottom of
the page. After submitting the changes in all the Web GUI pages, reboot the phone to have the
changes take effect if necessary. All the options under Basic Setting and Account Setting, and most of
the options under Advanced Settings do not require reboot after submitting the changes. Under
Advanced Setting, the parameters on network configuration require reboot after update.
DEFINITIONS
This section describes the options in the phone's Web GUI. As mentioned, you can log in as an
administrator or an end user.
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•
•
•
•
Status: Displays the Account status, Network status, and System Info of the phone;
Account: To configure the SIP account;
Basic Settings: To configure basic network settings, time settings, Line keys, and etc;
Advanced Settings: To configure advanced network settings, upgrading and provisioning, language
settings, call features, and etc.
•
Extension Boards: To configure the Multi Purpose Keys on the connected extension boards for
GXP2120/GXP2110.
STATUS PAGE DEFINITIONS
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.
MAC Address
IPv4 Address
IPv6 Address
Product Model
Part Number
The IPv4 address obtained on the phone.
The IPv6 address obtained on the phone.
Product model of the phone.
Product part number.
•
•
•
•
boot: boot version number;
core: core version number;
base: base version number;
Software Version
prog: program version number. This is the main firmware release
number, which is always used for identifying the software system of
the phone;
•
dsp: DSP version number.
System Up Time
System Time
Registered
System up time since the last reboot.
Current system time on the phone system.
SIP account registration status.
PPPoE Link Up
Service Status
Core Dump
PPPoE connection status.
GUI and Phone service status: running or stopped.
Core dump file that could be downloaded for troubleshooting purpose.
ACCOUNT PAGE DEFINITIONS
This field indicates whether the account is active. The default setting is
"Yes".
Account Active
Account Name
The name associated with the SIP account.
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The URL or IP address, and port of the SIP server. This is provided by
your VoIP service provider (ITSP).
SIP Server
The URL or IP address, and port of the SIP server. This will be used
when the primary SIP server fails.
Secondary SIP Server
IP address or Domain name of the Primary Outbound Proxy, Media
Gateway, or Session Border Controller. It's used by the phone 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.
Outbound Proxy
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 User ID
SIP service subscriber's Authenticate ID used for authentication. It can
be identical to or different from the SIP User ID.
Authenticate ID
Authenticate Password
Name
The account password required for the phone to authenticate with the
ITSP (SIP) server before the account can be registered. After it is saved,
this will appear as hidden for security purpose.
The SIP server subscriber's name (optional) that will be used for Caller
ID display.
This parameter controls how the Search Appliance looks up IP
addresses for hostnames. There are four modes: A Record, SRV,
NATPTR/SRV, Use Configured IP. The default setting is "A Record". If
the user wishes to locate the server by DNS SRV, the user may select
"SRV" or "NATPTR/SRV". If "Use Configured IP" is selected, please fill
in the three fields below:
DNS Mode
•
Primary IP: The primary IP address where the phone sends DNS
query to;
•
•
Backup IP 1;
Backup IP 2.
If the phone has an assigned PSTN telephone number, this field should
be set to "User=Phone". Then a "User=Phone" parameter will be
attached to the Request-Line and "TO" header in the SIP request to
indicate the E.164 number. If set to "Enable", "Tel:" will be used instead
of "SIP:" in the SIP request. The default setting is "Disable".
TEL URI
Selects whether or not the phone will send SIP Register messages to
the proxy/server. The default setting is "Yes".
SIP Registration
If set to "Yes", the SIP user's registration information will be cleared
when the phone reboots. The SIP Contact header will contain "*" to
Unregister On Reboot
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notify the server to unbind the connection. The default setting is "No".
Specifies the frequency (in minutes) in which the phone refreshes its
registration with the specified registrar. The default value is 60 minutes.
The maximum value is 64800 minutes (about 45 days).
Register Expiration
Specifies the time frequency (in seconds) that the phone sends
re-registration request before the Register Expiration. The default value
is 0.
Reregister Before Expiration
Local SIP Port
Defines the local SIP port used to listen and transmit. The default value
is 5060 for Account 1 and 5062 for Account 2.
SIP Registration Failure Retry
Wait Time
Specifies the interval to retry registration if the process is failed. The
default value is 20 seconds.
SIP T1 Timeout
SIP T2 interval
SIP T1 Timeout. The default setting is 0.5 seconds.
SIP T2 Interval. The default setting is 4 seconds.
Determines the network protocol used for the SIP transport. Users can
choose from TCP, UDP and TLS.
SIP Transport
SIP URI Scheme when using
TLS
Specifies if "sip:" or "sips:" will be used when TLS/TCP is selected for
SIP Transport. The default setting is "sips:".
Defines whether the actual ephemeral port in contact with TCP/TLS will
be used or not. This is used when TLS/TCP is selected for SIP Transfer.
The default setting is "No".
Use Actual Ephemeral Port in
Contact with TCP/TLS
Defines whether the domain certificates will be checked or not when
TLS/TCP is used for SIP Transport. The default setting is "No".
Check Domain Certificates
Remove OBP from route
Configures to remove outbound proxy from route. This is used for the
SIP Extension to notify the SIP server that the device is behind a
NAT/Firewall.
Defines whether the incoming messages will be validated or not. The
default setting is "No".
Validate Incoming Messages
Support SIP Instance ID
Defines whether SIP Instance ID is supported or not. The default setting
is "Yes".
This parameter configures whether the NAT traversal mechanism is
activated. Users could select the mechanism from No, STUN,
Keep-Alive, UPnP, Auto or VPN. If set to "STUN" and STUN server is
configured, the phone will route according to the STUN server. If NAT
type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone
will try to use public IP addresses and port number in all the SIP&SDP
messages. The phone will send empty SDP packet to the SIP server
periodically to keep the NAT port open if it is configured to be
NAT Traversal
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"Keep-Alive". Configure this to be "No" if an outbound proxy is used.
"STUN" cannot be used if the detected NAT is symmetric NAT.
When set to "Yes", a SUBSCRIBE for Message Waiting Indication will
be sent periodically. The phone supports synchronized and
non-synchronized MWI. The default setting is "No".
SUBSCRIBE for MWI
When set to "Yes", a SUBSCRIBE for Registration will be sent out
periodically. The default setting is "No".
SUBSCRIBE for Registration
This feature is used for Broadsoft call feature synchronization. When it's
enabled, DND and Call Forward features can be synchronized with
Broadsoft server. The default setting is "Disabled".
Feature Key Synchronization
PUBLISH for Presence
Proxy-Require
Enables Presence feature on the phone. The default setting is "No".
A SIP Extension to notify the SIP server that the phone is behind a
NAT/Firewall. Do not configure this parameter unless this feature is
supported on the SIP server.
Allows you to access voice messages by pressing the MESSAGE button
on the phone. This ID is usually the VM portal access number. For
example, in Asterisk server, 8500 could be used.
Voice Mail UserID
Specifies the mechanism to transmit DTMF digits. There are 3
supported modes: in audio which means DTMF is combined in the audio
signal (not very reliable with low-bit-rate codecs), via RTP (RFC2833), or
via SIP INFO.
Send DTMF
Configures the payload type for DTMF using RFC2833. The default
value is 101.
DTMF Payload Type
Selects whether or not to enable early dial. If it's set to "Yes", the SIP
proxy must support 484 response. The default setting is "No".
Early Dial
Dial Plan Prefix
Sets the prefix added to each dialed number.
A dial plan establishes the expected number and pattern of digits for a
telephone number. This parameter configures the allowed dial plan for
the phone.
Dial Plan Rules:
Dial Plan
1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d;
2. Grammar: x - any digit from 0-9;
a) xx+ - at least 2 digit numbers
b) xx. - only 2 digit numbers
c) ^ - exclude
d) [3-5] - any digit of 3, 4, or 5
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e) [147] - any digit of 1, 4, or 7
f) <2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
•
Example 1: {[369]11 | 1617xxxxxxx}
Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617;
•
Example 2: {^1900x+ | <=1617>xxxxxxx}
Block any number of leading digits 1900 or add prefix 1617 for any
dialed 7 digit numbers;
•
Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}
Allows any number with leading digit 1 followed by a 3 digit number,
followed by any number between 2 and 9, followed by any 7 digit
number OR Allows any length of numbers with leading digit 2, replacing
the 2 with 011 when dialed.
Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. |
[3469]11 }
Explanation of example rule (reading from left to right):
•
•
^1900x. - prevents dialing any number started with 1900;
<=1617>[2-9]xxxxxx - allows dialing to local area code (617)
numbers by dialing 7 numbers and 1617 area code will be added
automatically;
•
1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with
11 digits length;
•
•
011[2-9]x - allows international calls starting with 011;
[3469]11 - allows dialing special and emergency numbers 311, 411,
611 and 911.
Note:
In some cases where the user wishes to dial strings such as *123 to
activate voice mail or other applications provided by their service
provider, the * should be predefined inside the dial plan feature. An
example dial plan will be: { *x+ } which allows the user to dial * followed
by any length of numbers.
BLF Call-pickup Prefix
Configures the prefix prepended to the BLF extension when the phone
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picks up a call with BLF key. The default setting is **. (Not applicable to
GXP1450)
Delayed Call Forward Wait
Time
Defines the timeout (in seconds) before the call is forwarded on no
answer. The default value is 20 seconds.
When enabled, Do No Disturb, Call Forward and other call features will
be supported locally provided ITSP support those features. The default
setting is "Yes". If set to "No", ForwardAll softkey will be hidden for
Account 1.
Enable Call Features
Call Log
Configures Call Log setting on the phone. You can log all calls, only log
incoming/outgoing calls or disable call log. The default setting is "Log All
Calls".
The SIP Session Timer extension that 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. Session Expiration
is the time (in seconds) where the session is considered timed out,
provided no successful session refresh transaction occurs beforehand.
The default value is 180 seconds.
Session Expiration
The minimum session expiration (in seconds). The default value is 90
seconds.
Min-SE
If set to "Yes" and the remote party supports session timers, the phone
will use a session timer when it makes outbound calls.
Caller Request Timer
Callee Request Timer
If set to "Yes" and the remote party supports session timers, the phone
will use a session timer when it receives inbound calls.
If Force Timer is set to "Yes", the phone will use the session timer even if
the remote party does not support this feature. If Force Timer is set to
"No", the phone will enable the session timer only when the remote party
supports this feature. To turn off the session timer, select "No".
Force Timer
As a Caller, select UAC to use the phone as the refresher; or select UAS
to use the Callee or proxy server as the refresher.
UAC Specify Refresher
UAS Specify Refresher
As a Callee, select UAC to use caller or proxy server as the refresher; or
select UAS to use the phone as the refresher.
The Session Timer can be refreshed using the INVITE method or the
UPDATE method. Select "Yes" to use the INVITE method to refresh the
session timer.
Force INVITE
Enable 100rel
The use of the PRACK (Provisional Acknowledgment) method enables
reliability to SIP provisional responses (1xx series). This is very
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important in order to support PSTN internetworking. To invoke a reliable
provisional response, the 100rel tag is appended to the value of the
required header of the initial signaling messages.
Allows users to configure the ringtone for the account. Users can choose
from different ringtones from the dropdown menu.
Account Ring Tone
Specifies matching rules with number, pattern or Alert Info text. When
the incoming caller ID or Alert Info matches the rule, the phone will ring
with selected distinctive ringtone. Matching rules:
•
•
Specific caller ID number. For example, 8321123;
A defined pattern with certain length using x and + to specify, where
x could be any digit from 0 to 9. Samples:
xx+ : at least 2-digit number;
xx : only 2-digit number;
Matching Incoming Caller ID
[345]xx: 3-digit number with the leading digit of 3, 4 or 5;
[6-9]xx: 3-digit number with the leading digit from 6 to 9.
Alert Info text
•
Users could configure the matching rule as certain text (e.g., priority)
and select the custom ring tone mapped to it. The custom ring tone
will be used if the phone receives SIP INVITE with Alert-Info header
in the following format:
Selects the distinctive ring tone for the matching rule. When the
incoming caller ID or Alert Info matches the rule, the phone will ring with
the selected ring.
Distinctive Ringtones
Defines the timeout (in seconds) for the rings on no answer. The default
setting is 60 seconds.
Ring Timeout
Defines the interval (in seconds) before the line can be seized when
Shared Line is used. The default value is 15 seconds.
Line Seize Timeout
Send Anonymous
Anonymous Call Rejection
Auto Answer
If set to "Yes", the "From" header in outgoing INVITE messages will be
set to anonymous, essentially blocking the Caller ID to be displayed.
If set to "Yes", anonymous calls will be rejected. The default setting is
"No".
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep.
If set to "Yes", the phone will automatically turn on the speaker phone to
Allow Auto Answer by Call-Info answer incoming calls after a short reminding beep, based on the SIP
info header sent from the server/proxy. The default setting is "No".
Refer-To Use Target Contact
If set to "Yes", the "Refer-To" header uses the transferred target's
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Contact header information for attended transfer. The default setting is
"No".
Transfer on Conference
Hangup
Defines whether or not the call is transferred to the other party if the
initiator of the conference hangs up. The default setting is "No".
If set to "Yes", SIP User ID will be checked in the Request URI of the
incoming INVITE. If it doesn't match the phone's SIP User ID, the call will
be rejected. The default setting is "No".
Check SIP User ID for
incoming INVITE
7 different vocoder types are supported on the phone, including G.711
U-law (PCMU), G.711 A-law (PCMA), G.723.1, G.729A/B, G.722 (wide
band), iLBC and G72-32. Users can configure vocoders in a preference
list that is included with the same preference order in SDP message.
Preferred Vocoder
Enables the SRTP mode based on your selection. The default setting is
"Disabled".
SRTP Mode
Defines whether symmetric RTP is supported or not. The default setting
is "No".
Symmetric RTP
Controls the silence suppression/VAD feature of the audio codec G.723
and G.729. If 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".
Silence Suppression
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.
Defines the timeout (in seconds) for no key entry. If no key is pressed
after the timeout, the digits will be sent out. The default value is 4
seconds.
No Key Entry Timeout (s)
Use # as Dial Key
Allows users to configure the "#" key as the "Send" key. If set to "Yes",
the "#" key will immediately dial out the input digits. In this case, this key
is essentially equivalent to the "Send" key. If set to "No", the "#" key is
included as part of the dialing string.
G723 Rate
Selects encoding rate for G723 codec. The default value is 5.3kbps.
Selects "ITU" or "IETF" for G726-32 packing mode.
G.726-32 Packing Mode
iLBC Frame Size
iLBC Payload Type
Selects iLBC packet frame size. The default value is 30ms.
Specifies iLBC Payload type. The default value is 97. The valid range is
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between 96 and 127.
Selects either Fixed or Adaptive based on network conditions. The
default setting is "Adaptive".
Jitter Buffer Type
Selects Low, Medium, or High based on network conditions. The default
setting is "Medium".
Jitter Buffer Length
Configures the eventlist BLF URI on the phone to monitor the extensions
in the list with Multi Purpose Key. If the server supports this feature,
users need to configure an eventlist BLF URI on the service side first
(i.e., [email protected]) with a list of extension included. On the
phone, in this "eventlist BLF URI" field, fill in the URI without the domain
(i.e., BLF1006). To monitor the entensions in the list, under Web
GUI->Basic Setting page, please select "eventlist BLF" in the Multi
Purpose Key key mode, choose account, enter Name and User ID of
each extension in the list. (Not applicable to GXP1450)
Eventlist BLF URI
Configures the conference URI when using Broadsoft N-way calling
feature.
Conference URI
Controls whether the Privacy Header will present in the SIP INVITE
message or not. The default setting is "default", which is when "Huawei
IMS" special feature is on, the Privacy Header will not show in INVITE. 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 Privacy Header
Controls whether the P-Preferred-Identity Header will present in the SIP
INVITE message or not. The default setting is "default", which is when
"Huawei IMS" special feature is on, the P-Preferred-Identity Header will
not show in INVITE. 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.
Use P-Preferred-Identity
Header
Different soft switch vendors have special requirements. Therefore users
may need select special features to meet these requirements. Users can
choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro
or Huawei IMS depending on the server type. The default setting is
"Standard".
Special Feature
SETTINGS/BASIC SETTINGS PAGE
Allows the administrator to set the password for user-level web GUI
access. This field is case sensitive with a maximum length of 30
End User Password
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characters.
Confirm Password
Internet Protocol
Confirms the end user password field to be the same as above.
Selects Prefer IPv4 or Prefer IPv6.
Allows users to configure the appropriate network settings on the phone
to obtain IPv4 address. Users could select "DHCP", "Static IP" or
"PPPoE". By default, it is set to "DHCP".
IPv4 Address Type
Specifies the name of the client. This field is optional but may be
required by some Internet Service Providers.
DHCP Host name (Option 12)
DHCP Vendor Class ID
(Option 60)
Used by clients and servers to exchange vendor class ID.
Allow DHCP Option 120 to
override SIP Server
Enables DHCP Option 120 from local server to override the SIP Server
on the phone. The default setting is "No".
PPPoE Account ID
PPPoE Password
PPPoE Service Name
IPv4 Address
Enter the PPPoE account ID.
Enter the PPPoE Password.
Enter the PPPoE Service Name.
Enter the IP address when static IP is used.
Enter the Subnet Mask when static IP is used for IPv4.
Enter the Default Gateway when static IP is used for IPv4.
Enter the DNS Server 1 when static IP is used for IPv4.
Enter the DNS Server 2 when static IP is used for IPv4.
Enter the Preferred DNS Server for IPv4.
Subnet Mask
Gateway
DNS Server 1
DNS Server 2
Preferred DNS Server
Allows users to configure the appropriate network settings on the phone
to obtain IPv6 address. Users could select "Auto-configured" or
"Statically configured" for the IPv6 address type.
IPv6 Address Type
Enter the static IPv6 address when Full Static is used in "Statically
configured" IPv6 address type.
Static IPv6 Address
IPv6 Prefix Length
IPv6 Prefix
Enter the IPv6 prefix length when Full Static is used in "Statically
configured" IPv6 address type.
Enter the IPv6 Prefix (64 bits) when Prefix Static is used in "Statically
configured" IPv6 address type.
DNS Server 1
Enter the DNS Server 1 for IPv6.
Enter the DNS Server 2 for IPv6.
Enter the Preferred DNS Server for IPv6.
DNS Server 2
Preferred DNS server
Allows the user to enable/disable 802.1x mode on the phone. The
default value is disabled. To enable 802.1x mode, this field should be set
802.1x mode
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to EAP-MD5.
Identity
Enter the Identity for the 802.1x mode.
Enter the MD5 Password for the 802.1x mode.
MD5 Password
Specifies the HTTP proxy URL for the phone to send packets to. The
proxy server will act as an intermediary to route the packets to the
destination.
HTTP Proxy
Specifies the HTTPS proxy URL for the phone to send packets to. The
proxy server will act as an intermediary to route the packets to the
destination.
HTTPS Proxy
Assigns a function to the corresponding line key. The key mode options
are:
•
Line
Regular line key to open up a line and switch line. The Name and
UserID can be left blank.
•
Share Line
Share line for Shared Line Appearance feature. Select the Account
registered as Shared line for the line key. The Name and UserID can
be left blank.
•
•
•
•
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
UserID field to be dialed.
Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension
number in UserID field to be monitored.
Line Key
X
Presence Watcher
This option has to be supported by a presence server and it is tied to
the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX
collects the information from the phones and sends it out in one
single notify message. PBX server has to support this feature.
Speed Dial via active account
•
•
Similar to Speed Dial but it will dial based on the current active
account. For example, if the phone is offhook and account 2 is
active, it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the UserID field to be dialed during
the call. "Enable MPK Sending DTMF" (under Advanced Setting)
has to be set to "Yes" first.
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•
•
Voice Mail
Select Account and enter the Voice Mail access number in the
UserID field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Name and UserID field should be left blank. Also, this option is not
binding to the account and the call will be returned based on the
account with the last answered call.
•
•
•
•
Transfer
Select Account, and enter the number in the UserID field to be
transferred (blind transfer) during the call.
Call Park
Select Account, and enter the call park extension in the UserID field
to park/pick up the call.
Intercom
Select Account, and enter the extension number in the UserID field
to do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP
search base in the Name field. It could be the same or different from
the Base in LDAP configuration under Advanced Settings. The Base
in LDAP configuration will be used if the Name field is left blank.
Enter the LDAP Name/Number filter in the UserID field.
Assigns a function to the corresponding Multi Purpose Key (Not
applicable to GXP1450). The key mode options are:
•
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
UserID field to be dialed.
•
BLF (Busy Lamp Field)
This option has to be supported on the PBX and it indicates the
status of the extension. The three possible states are idle (green),
busy (red), ringing (blinking red).
Multi Purpose Keys X
•
•
Presence Watcher
This option has to be supported by a presence server and it is tied to
the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX
collects the information from the phones and sends it out in one
single notify message. PBX server has to support this feature.
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•
•
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active
account. For example, if the phone is offhook and account 2 is
active, it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the UserID field to be dialed during
the call. "Enable MPK Sending DTMF" (under Advanced Setting)
has to be set to "Yes" first.
•
•
Voice Mail
Select Account and enter the Voice Mail access number in the
UserID field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Name and UserID field should be left blank. Also, this option is not
binding to the account and the call will be returned based on the
account with the last answered call.
•
•
•
•
Transfer
Select Account, and enter the number in the UserID field to be
transferred (blind transfer) during the call.
Call Park
Select Account, and enter the call park extension in the UserID field
to park/pick up the call.
Intercom
Select Account, and enter the extension number in the UserID field
to do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP
search base in the Name field. It could be the same or different from
the Base in LDAP configuration under Advanced Settings. The Base
in LDAP configuration will be used if the Name field is left blank.
Enter the LDAP Name/Number filter in the UserID field.
Configures the date/time used on the phone according to the specified
time zone.
Time Zone
This parameter allows the users to define their own time zone.
The syntax is: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0
Self-Defined Time Zone
MTZ+6MDT+5
This indicates a time zone with 6 hours offset with 1 hour ahead which is
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U.S central time. If it is positive (+) if the local time zone is west of the
Prime Meridian (A.K.A: International or Greenwich Meridian) and
negative (-) if it is east.
M4.1.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3rd Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon,
Tues, ... ,Sat)
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
Configures to enable or disable weather update on the phone. The
default setting is "Yes". If set to "No", the weather information screen will
not show.
Enable Weather Update
Configures weather city code for the phone to look up the weather
information. The default setting is "Automatic" and the weather
information will be obtained based on the IP location of the phone if
available. Otherwise, specify the self-defined city code. For example,
USCA0638 is the city code for Los Angeles, CA, United States.
City Code
Specifies the weather update interval (in minutes). The default value is
15 minutes.
Update Interval
Degree Unit
Specifies the degree unit for the weather information to display on the
phone.
Configures to enable or disable stock update on the phone (Not
applicable to GXP1450). The default setting is "Yes". If set to "No", the
stock information screen will not show.
Enable Stock Update
Stock Code
Configures stock code for the phone to look up and display the stock
information (Not applicable to GXP1450).
Configures to enable or disable currency update on the phone (Not
applicable to GXP1450). The default setting is "Yes". If set to "No", the
currency information screen will not show.
Enable Currency Update
Currency Code
Configures currency code for the phone to look up and display the
currency information (Not applicable to GXP1450).
Configures the LCD backlight brightness level (from 0 to 8) for phone's
idle status and active status. The default value is 6 for active status and
2 for idle status.
LCD Backlight Brightness
LCD Contrast
Configures the LCD contrast level (from 0 to 20). The default value is 10.
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Configures the date display format on the LCD. The following formats
are supported:
•
•
•
•
•
yyyy-mm-dd: 2012-07-02
Date Display Format
mm-dd-yyyy: 07-02-2012
dd-mm-yyyy: 02-07-2012
dddd, MMMM dd: Friday, October 12 (Not applicable to GXP1450)
MMMM dd, dddd: October 12, Friday (Not applicable to GXP1450)
Configures the time display in 12-hour or 24-hour format on the LCD.
The default setting is in 12-hour format.
Time Display Format
When it's set to "Yes", the DTMF digits entered during the call will not
display. The default setting is "No".
Disable in-call DTMF Display
Disable Missed Call Backlight
When it's set to "Yes", the LCD backlight will not be turned on when
there is a new missed call. The default setting is "No".
Configures to enable or disable the speaker to ring when headset is
used on "Toggle Headset/Speaker" mode. If set to "Yes", when the
phone is in Headset "Toggle Headset/Speaker" mode, both headset and
speaker will ring on incoming call. The default setting is "No".
Always Ring Speaker
When headset is connected to the phone, users could use the
HEADSET button in "Default Mode" or "Toggle Headset/Speaker".
•
Default Mode:
When the phone is in idle, press HEADSET button to off hook
the phone and making calls by using headset. Headset icon will
display on the left side of the screen in dialing/talking status.
When there is an incoming call, press HEADSET button to pick
up the call using headset.
When there is an active call using headset, press HEADSET
button to hang up the call.
Headset Key Mode
When Speaker/Handset is being used in dialing/talking status,
press HEADSET button to switch to headset. Press it again to
hang up the call. Or press speaker/Handset to switch back to
the previous mode.
•
Toggle Headst/Speaker:
When the phone is in idle, press HEADSET button to switch to
Headset mode. The headset icon will display on the left side of
the screen. In this mode, if pressing Speaker button or Line key
to off hook the phone, headset will be used.
When there is an active call, press HEADSET button to toggle
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between Headset and Speaker.
Call History Flash Writing:
Write Timeout
Defines the interval (in seconds) to save the call history to phone's flash.
The default value is 300 seconds.
Defines the number of unsaved logs before written to phone's flash. The
default value is 200 entries.
Max Unsaved Log
Configures the transmission gain of the headset. The default value is
0dB.
Headset TX gain
Headset RX gain
Handset TX gain
Configures the receiving gain of the headset. The default value is 0dB.
Configures the transmission gain of the handset. The default value is 0
dB.
SETTINGS/ADVANCED SETTINGS PAGE
Allows users to change the admin password. The password field is
Admin Password
purposely hidden after clicking the Update button for security purpose.
This field is case sensitive with a maximum length of 30 characters.
Confirm Password
Layer 3 QoS
Confirms the admin password field to be the same as above.
Defines the Layer 3 QoS parameter. This value is used for IP
Precedence, Diff-Serv or MPLS. The default value is 12.
Layer 2 QoS 802.1Q/VLAN
Tag
Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is
0.
Layer 2 QoS 802.1p Priority
Value
Assigns the priority value of the Layer2 QoS packets. The default value
is 0.
This parameter defines the local RTP port used to listen and transmit. It
is the base RTP port for channel 0. When configured, channel 0 will use
this port _value for RTP; channel 1 will use port_value+2 for RTP. Local
RTP port ranges from 1024 to 65400 and must be even. The default
value is 5004.
Local RTP Port
When set to "Yes", this parameter will force random generation of both
the local SIP and RTP ports. This is usually necessary when multiple
phones are behind the same full cone NAT. The default setting is "Yes"
(This parameter must be set to "No" for Direct IP Calling to work).
Use Random Port
Keep-alive Interval
Specifies how often the phone sends a blank UDP 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 NAT IP address used in SIP/SDP messages. This field is blank at
the default settings. It should ONLY be used if it's required by your ITSP.
Use NAT IP
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The IP address or Domain name of the STUN server. STUN resolution
results are displayed in the STATUS page of the Web GUI. Only
non-symmetric NAT routers work with STUN.
STUN Server
Specifies how firmware upgrading and provisioning request to be sent:
Always Check for New Firmware, Check New Firmware only when F/W
pre/suffix changes, Always Skip the Firmware Check.
Firmware Upgrade and
Provisioning
The password for encrypting the XML configuration file using OpenSSL.
This is required for the phone to decrypt the encrypted XML
configuration file.
XML Config File Password
HTTP/HTTPS User Name
HTTP/HTTPS Password
The user name for the HTTP/HTTPS server.
The password for the HTTP/HTTPS server.
Allows users to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
Upgrade Via
Defines the server path for the firmware server. It could be different from
the configuration server for provisioning.
Firmware Server Path
Config Server Path
Defines the server path for provisioning. It could be different from the
firmware server for upgrading.
Enables your ITSP to lock firmware updates. If configured, only the
firmware with the matching encrypted prefix will be downloaded and
flashed into the phone.
Firmware File Prefix
Firmware File Postfix
Config File Prefix
Enables your ITSP to lock firmware updates. If configured, only the
firmware with the matching encrypted postfix will be downloaded and
flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
configuration file with the matching encrypted prefix will be downloaded
and flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
configuration file with the matching encrypted postfix will be downloaded
and flashed into the phone.
Config File Postfix
Allow DHCP Option 43 and
Option 66 Override Server
If DHCP option 66 is enabled on the LAN side, the TFTP server can be
redirected. The default setting is "Yes".
Automatic Upgrade
Enables automatic upgrade and provisioning. The default setting is "No".
Authenticates configuration file before acceptance. The default setting is
"No".
Authenticate Conf File
Enable TR-069
ACS URL
Enables TR-069. The default setting is "No".
URL for TR-069 Auto Configuration Servers (ACS).
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TR-069 Username
TR-069 Password
ACS username for TR-069.
ACS password for TR-069.
Enables periodic inform. If set to "Yes", device will send inform packets
to the ACS. The default setting is "No".
Periodic Inform Enable
Periodic Inform Interval
Sets up the periodic inform interval to send the inform packets to the
ACS.
Connection Request
Username
The user name for the ACS to connect to the phone.
Connection Request Password The password for the ACS to connect to the phone.
Connection Request Port
CPE SSL Certificate
The port for the ACS to connect to the phone.
The Cert File for the phone to connect to the ACS via SSL.
The Cert Key for the phone to connect to the ACS via SSL.
CPE SSL Private Key
Configures to enable phonebook XML download. Users could select
HTTP/HTTPS/TFTP to download the phonebook file. The default setting
is "No".
Phonebook XML Download
Phonebook XML Server Path
Phonebook Download Interval
Configures the server path to download the phonebook XML. This field
could be IP address or URL, with up to 256 characters.
Configures the phonebook download interval (in minutes). If it's set to 0,
the automatic download will be disabled. The default value is 0. The
valid range is 5 to 720 minutes.
Remove Manually-edited
Entries on Download
If set to "Yes", when XML phonebook is downloaded, the entries added
manually will be automatically removed. The default setting is "Yes".
LDAP Directory: Server
Address
Configures the IP address or DNS name of the LDAP server.
Configures the LDAP server port.
LDAP Directory: Port
Configures the LDAP search base. This is the location in the directory
where the search is requested to begin.
Example:
LDAP Directory: Base
dc=grandstream, dc=com
ou=Boston, dc=grandstream, dc=com
Configures the bind "Username" for querying LDAP servers. Some
LDAP servers allow anonymous binds in which case the setting can be
left blank.
LDAP Directory: User Name
Configures the bind "Password" for querying LDAP servers. The field
can be left blank if the LDAP server allows anonymous binds.
LDAP Directory: Password
LDAP Number Filter
Configures the filter used for number lookups.
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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.
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;
LDAP Name Filter
(!(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.
Selects the protocol version for the phone to send the bind requests.
The default setting is "Version 3".
LDAP Version
Specify 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.
LDAP Name Attributes
Example:
gn
cn sn description
Specifies the "number" attributes of each record which are returned in
the LDAP search result. This field allows the users to configure multiple
space separated number attributes.
Example:
LDAP Number Attributes
telephoneNumber
telephoneNumber Mobile
Configures the entry information to be shown on phone's LCD. Up to 3
fields can be displayed.
Example:
LDAP Display Name
Max. Hits
%cn %sn %telephoneNumber
Specifies the maximum number of results to be returned by the LDAP
server. If set to 0, server will return all search results. The default setting
is 50.
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Specifies the interval (in seconds) for the server to process the request
and client waits for server to return. The default setting is 30 seconds.
Search Timeout
Sort Results
Specifies whether the searching result is sorted or not. The default
setting is "No".
Configures to enable LDAP number searching when dialing and
receiving calls.
LDAP Lookup
Configures the display name when LDAP looks up the name for
incoming call or outgoing call. This field must be a subset of the LDAP
Name Attributes.
Example:
Lookup Display Name
gn
cn sn description
User Phonebook Key for LDAP If set to "Yes", the Phonebook Key will be used to bring up LDAP search.
Search
The default setting is "No".
Configures to enable idle screen XML download. Users could select
HTTP/HTTPS/TFTP to download the XML idle screen file. The default
setting is "No".
Idle Screen XML Download
Download Screen XML At
Bootup
If set to "Yes", the idle screen XML file will be downloaded when the
phone boots up. The default setting is "No".
User Custom Filename
Specifies the custom file for the idle screen XML file to be downloaded.
Configures the server path to download the idle screen XML file. This
field could be IP address or URL, with up to 256 characters.
Idle Screen XML Server Path
Configures the server path to download the idle screen XML file (Not
applicable to GXP1450). This field could be IP address or URL, with up
to 256 characters.
XML Application Server Path
XML Application Softkey Label
Specifies the softkey name displayed on the idle screen for the users to
enter XML application (Not applicable to GXP1450). Please make sure
the label name does not exceed the softkey range on each model. The
default value is "XML Service".
Configures a User ID/extension to dial automatically when the phone is
off hook. The phone will use the first account to dial out. The default
setting is "No".
Offhook Auto Dial
Configures whether auto recover or not when the phone is running
abnormal. The default setting is "Yes".
Auto Recover From Abnormal
The URL or IP address of the syslog server for the phone to send syslog
to.
Syslog Server
Syslog Level
Selects the level of logging for syslog. The default setting is None. There
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are 4 levels: DEBUG, INFO, WARNING AND ERROR.
Syslog messages are sent based on the following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level);
NAT related info (INFO level);
sent or received SIP message (DEBUG level);
SIP message summary (INFO level);
inbound and outbound calls (INFO level);
registration status change (INFO level);
negotiated codec (INFO level);
ethernet link up (INFO level);
SLIC chip exception (WARNING and ERROR levels);
memory exception (ERROR level).
Configures whether the SIP log will be included in the syslog messages
or not. The default setting is "No".
Send SIP Log
NTP Server
Defines the URL or IP address of the NTP server. The phone may obtain
the date and time from the server.
Defines whether DHCP Option 42 should override NTP server or not.
When enabled, DHCP Option 42 will override the NTP server if it's set
up on the LAN. The default setting is "Yes".
Allow DHCP Option 42
Override NTP Server
Configures to turn on/off public mode for hot desking feature on the
phone. If set to "Yes", users would need fill in the SIP Server address for
account 1 as well. Then reboot the phone. When the phone boots up,
users will need enter SIP User ID and Password on the LCD to login and
use the phone.
Public Mode
Note:
When the phone is in public mode login screen, press HOLD button will
have the IP address of the phone displayed.
SSL Certificate
SSL Certificate used for SIP Transport in TLS/TCP.
SSL Private key used for SIP Transport in TLS/TCP.
SSL Private key password used for SIP Transport in TLS/TCP.
SSL Private Key
SSL Private Key Password
Configures system ring tone. The default value is North American
standard. Users could adjust system ring tone frequencies and
cadences based on local telecom standard.
System Ring Tone
Call Progresses Tones:
Dial Tone
Configures ring or tone frequencies based on parameters from local
telecom. The default value is North American standard. Frequencies
should be configured with known values to avoid uncomfortable high
pitch sounds.
Message Waiting
Ring Back Tone
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Call-Waiting Tone
Busy Tone
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
Reorder Tone
(Frequencies are in Hz and cadence on and off are in 10ms)
ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of
silence. In order to set a continuous ring, OFF should be zero. Otherwise
it will ring ON ms and a pause of OFF ms and then repeat the pattern.
Up to three cadences are supported.
Configures the intercom extension number for account 1 to dial out (Not
applicable to GXP1450). This User ID is mapped to the INTERCOM
button on the phone.
Intercom User ID
Disable Call-Waiting
Disables the call waiting feature. The default setting is "No".
Disables the call waiting tone when call waiting is on. The default setting
is "No".
Disable Call-Waiting Tone
Disable Direct IP Calls
Disables Direct IP Call. The default setting is "No".
When set to "Yes", users can dial an IP address under the same
LAN/VPN segment by entering the last octet in the IP address. To dial
quick IP call, off hook the phone and dial #XXX (X is 0-9 and XXX
<=255), phone will make direct IP call to aaa.bbb.ccc.XXX where
aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet
mask. #XX or #X are also valid so leading 0 is not required (but OK). No
SIP server is required to make quick IP call. The default setting is "No".
Use Quick IP-Call mode
Disable Conference
Disables the Conference function. The default setting is "No".
Enables Multi Purpose Key to send DTMF during the call (Not applicable
to GXP1450). The default setting is "No".
Enable MPK sending DTMF
If set to "Yes", the DND button will not set the phone to DND. The default
setting is "No".
Disable DND Button
Disable Transfer
Disables the Transfer function. The default setting is "No".
If set to "Yes", the phone will use attended transfer by default. The
default setting is "No".
Auto-Attended Transfer
In-call dial number on pressing Configures the number for the phone to dial as DTMF during the call
transfer key
using TRAN button.
Configures the access control for the users to configure from keypad
Menu. There are three different options:
•
•
Unrestricted. All the options can be accessed in keypad Menu.
Basic settings only. The CONFIG option will not display for users
to access in keypad Menu.
Configuration via Keypad Menu
•
Constraint Mode. CONFIG, FACTORY FUNCTIONS and
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NETWORK options will not display for users to access in keypad
menu.
If set to "Yes", the keypad can be locked by pressing and holding the
STAR * key for about 4 seconds. A lock icon will show indicating the
keypad is locked. The default setting is "Yes".
Enable STAR key Keypad
locking
Note:
When the keypad is locked, users would need press and hold the STAR
* key for about 4 seconds again and then enter the password to unlock
it.
Configures the password to lock/unlock the keypad. The password field
allows number with up to 32 characters.
Password to lock/unlock
Offhook timeout
If configured, when the phone is on hook, it will go off hook after the
timeout (in seconds). The default value is 30 seconds.
Enables/Disables China Telecom Mode to use China Telecom special
features on the phone.
China Telecom Mode
Do Not Escape # as %23 in
SIP URI
Specifies whether to replace # by %23 or not for some special situations.
The default setting is "No".
Disable Telnet
Disables Telnet access. The default setting is "No".
Selects display language on the phone.
Display Language
Download Device
Configuration
Click to download the device configuration file in .txt format.
NAT SETTINGS
If the devices are kept within a private network behind a firewall, we recommend using STUN Server. The
following settings are useful in the STUN Server scenario:
•
STUN Server (under Advanced Settings page)
Enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the
internet and enter it on this field. If using Public IP, keep this field blank.
•
Use Random Ports (under Advanced Settings page)
This setting depends on your network settings. When set to "Yes", it will force random generation of
both the local SIP and RTP ports. This is usually necessary when multiple GXPs are behind the
same NAT. If using a Public IP address, set this parameter to "No".
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•
NAT Traversal (under Account Setting page)
Default setting is "No". Enable the device to use NAT traversal when it is behind firewall on a private
network. Select Keep-Alive, Auto, STUN (with STUN server path configured too) or other option
according to the network setting.
WEATHER UPDATE
To customize GXP2120/GXP2110/GXP2100/GXP1450 to display weather information for the preferred
city, users could go to web GUI->Settings->Basic Settings page and enter the city code in the following
options:
By default the City Code is set to "Automatic", which allows the phone to obtain weather information
based on the IP location detected. To use "Self-Defined City Code" option, please follow the steps below
to obtain the correct city code:
•
•
•
Enter the city name in the search field. For example, Boston, MA. And click on "SEARCH";
The searching result will show in a new window with URL in the browser's address bar. For example,
•
In the above link, USMA0046 is the city code to be filled in "Self-Defined City Code" option.
Users could then further configure the "Update Interval" and "Degree Unit" for weather information
display.
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PUBLIC MODE
The GXP2120/GXP2110/GXP2100/GXP1450 supports hot desking using public mode. Under public mode,
users could login the phone with the SIP account User ID and password. Please follow the steps below to
configure the phone for public mode:
•
•
•
Under Web GUI->Account 1 setting page, fill up the SIP server address for account 1. Click "Update"
on the bottom of the page;
Under Web GUI->Advanced setting page, set Public Mode option to "Yes". Click "Update" and reboot
the phone;
When phone boots up, SIP User ID and Password to register to the configured SIP server in account 1
will be required. Enter the correct account information to log in to the phone. When entering the
account information, press softkey "123"/"abc" to toggle input method;
In login page, pressing HOLD button on the phone will show phone's IP address;
After using the phone, go to LCD MENU->LogOut to log off the public mode.
•
•
EDITING CONTACTS AND CLICK-TO-DIAL
From GXP2120/GXP2110/GXP2100/GXP1450 Web GUI, users could view contacts, edit contacts, or dial
out with Click-to-Dial feature
on the top of the Web GUI. In the following figure, the Contact page
shows all the added contacts (manually or downloaded via XML phonebook). Here users could add new
contact, export phonebook XML, import phonebook XML, search contact, filter contacts by group, edit
selected contact, or dial the contact/number.
When clicking on the
icon on the top menu of the Web GUI, a new dialing window will show for you
to enter the number. Once Dial is clicked, the phone will go off hook and dial out the number from selected
account. Please see Figure 3 and Figure 4 in the following pages for more details.
Additionally, users could directly send the command for the phone to dial out by specifying the following
URL in PC's web browser, or in the field as required in other call modules.
http://ip_address/cgi-bin/api-make_call?phonenumber=1234&account=0&password=admin
In the above link, replace the fields with
•
ip_address:
Phone's IP Address.
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•
•
phonenumber=1234:
The number for the phone to dial out
account=0:
The account index for the phone to make call. The index is 0 for account 1, 1 for account 2, 2 for
account 3, and etc.
•
password=admin:
The admin login password of phone's Web GUI.
Click to select
group in the
Click
to
in
Click to input number
and dial from available
lines.
search
dropdown menu.
phonebook.
Click to edit this
contact.
Click to
add new
contacts.
Click to export
phonebook in
XML format.
Click to import Click to call
Click to download
the contact
phonebook
XML file.
this contact
from
the
information in .vcf
format
phone.
Figure 3: Web GUI - Contacts
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Figure 4: Click-to-Dial
SAVING THE CONFIGURATION CHANGES
After users makes changes to the configuration, press the Update button on the bottom of the Web GUI
page. We recommend rebooting or powering cycle the IP phone after saving changes.
REBOOTING FROM REMOTE LOCATIONS
Press the Reboot button on the bottom of the web GUI page to reboot the phone remotely. The web
browser will then display a reboot page with message "The device is rebooting now...". Wait for about 1
minute to log in again.
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UPGRADING AND PROVISIONING
The GXP2120/GXP2110/GXP2100/GXP1450 can be upgraded via TFTP/HTTP/HTTPS by configuring the
URL/IP Address for the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid
URL for TFTP or HTTP; the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com
fw.ipvideotalk.com/gs
There are two ways to setup a software upgrade server: The IVR Menu or the Web Configuration Interface.
UPGRADE VIA KEYPAD MENU
Follow the steps below to configure the upgrade server path via phone's keypad menu:
•
•
•
Press MENU button and navigate using Up/Down arrow to select Config;
In the Config options, select Upgrade;
Enter the firmware server path and select upgrade method. The server path could be in IP address
format or FQDN format;
•
•
Press the "OK" softkey. A reboot message window will be prompt;
Reboot the phone to have the change take effect.
When upgrading starts, the screen will show upgrading progress. When done you will see the phone
restarts again. Please do not interrupt or power cycle the phone when the upgrading process is on.
UPGRAGE VIA WEB GUI
Open a web browser on PC and enter the IP address of the phone. Then, login with the administrator
username and password. Go to Settings->Advanced Settings page, enter the IP address or the FQDN for
the upgrade server in "Firmware Server Path" field and choose to upgrade via TFTP or HTTP/HTTPS.
Update the change by clicking the "Update" button. Then "Reboot" or power cycle the phone to update the
new firmware.
When upgrading starts, the screen will show upgrading progress. When done you will see the phone
restarts again. Please do not interrupt or power cycle the phone when the upgrading process is on.
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Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. We
recommend completing firmware upgrades in a controlled LAN environment whenever possible.
NO LOCAL TFTP/HTTP SERVERS
For users that would like to use remote upgrading without a local TFTP/HTTP server, Grandstream offers a
NAT-friendly HTTP server. This enables users to download the latest software upgrades for their phone via
this server. Please refer to the webpage:
Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A
free windows version TFTP server is available for download from :
Instructions for local firmware upgrade via TFTP:
1. Unzip the firmware files and put all of them in the root directory of the TFTP server;
2. Connect the PC running the TFTP server and the phone to the same LAN segment;
3. Launch the TFTP server and go to the File menu->Configure->Security to change the TFTP server's
default setting from "Receive Only" to "Transmit Only" for the firmware upgrade;
4. Start the TFTP server and configure the TFTP server in the phone’s web configuration interface;
5. Configure the Firmware Server Path to the IP address of the PC;
6. Update the changes and reboot the phone.
Microsoft IIS web server.
Note:
When the phone boots up, it will send a TFTP or HTTP request to download the configuration file
"cfgxxxxxxxxxxxx" where "xxxxxxxxxxxx" is the MAC address of the phone. If it is a normal TFTP or HTTP
upgrade, the following messages “TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File does
not exist. Configuration File Download” can be ignored in the TFTP/HTTP server log.
CONFIGURATION FILE DOWNLOAD
Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File
(binary or XML) through TFTP or HTTP/HTTPS. The "Config Server Path" is the TFTP or HTTP/HTTPS
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server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format.
The "Config Server Path" can be the same or different from the "Firmware Server Path".
A configuration parameter is associated with each particular field in the web configuration page. A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with the “Admin Password” in the Web GUI->Settings->Advanced Settings.
For a detailed parameter list, please refer to the corresponding firmware release configuration template.
When the GXP2xx/GXP1450 boots up or reboots, it will issue a request to download a configuration XML
file named "cfgxxxxxxxxxxxx.xml" followed by a file named "cfgxxxxxxxxxxxx", where "xxxxxxxxxxxx" is
the MAC address of the phone, i.e., "cfg000b820102ab.xml" and "cfg000b820102ab". If the download of
"cfgxxxxxxxxxxxx.xml" file is not successful, the provision program will download a generic cfg.xml file. The
configuration file name should be in lower case letters.
For more details on XML provisioning, please refer to:
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RESTORE FACTORY DEFAULT SETTINGS
Warning:
Restoring the Factory Default Settings will delete all configuration information on the phone. Please
backup or print all the settings before you restore to the factory default settings. Grandstream is not
responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
Please follow the instructions below to reset the phone:
•
•
•
•
•
Press MENU button to bring up the keypad configuration menu;
Select "Config" and enter;
Select "Factory Reset";
A warning window will pop out to make sure a reset is requested and confirmed;
Press the "OK" softkey to confirm and the phone will reboot. To cancel the Reset, press Cancel
softkey instead.
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EXPERIENCING THE GXP2120/GXP2110/GXP2100/GXP1450
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 IP phone, it will be sure to bring convenience and color to
both your business and personal life.
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