Grandstream Networks GXP2130 IP PHONE User Manual
Grandstream Networks, Inc. IP PHONE
User Manual
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| Document ID | 2217244 |
| Application ID | VELepn6N4Nb4bR/e7YOKbA== |
| Document Description | User Manual |
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| Document Type | User Manual |
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| Filesize | 164.24kB (2053012 bits) |
| Date Submitted | 2014-03-17 00:00:00 |
| Date Available | 2014-03-17 00:00:00 |
| Creation Date | 2014-02-26 11:28:03 |
| Producing Software | Microsoft® Word 2010 |
| Document Lastmod | 2014-02-26 11:28:03 |
| Document Title | User Manual |
| Document Creator | Microsoft® Word 2010 |
| Document Author: | Grandstream |
Grandstream Networks, Inc.
GXP2130/GXP2140/GXP2160
Enterprise IP Phone
GXP2130/GXP2140/GXP2160 USER MANUAL
GXP2130/GXP2140/GXP2160 User Manual
Index
GUI INTERFACE EXAMPLES .................................................................... 6
GNU GPL INFORMATION .......................................................................... 7
CHANGE LOG ........................................................................................... 8
FIRMWARE VERSION 1.0.1.14 ............................................................................................................ 8
WELCOME ................................................................................................. 9
PRODUCT OVERVIEW ............................................................................ 10
FEATURE HIGHTLIGHTS ................................................................................................................... 10
GXP2130/GXP2140/GXP2160 TECHNICAL SPECIFICATIONS ........................................................ 11
INSTALLATION ........................................................................................ 16
EQUIPMENT PACKAGING ................................................................................................................. 16
CONNECTING YOUR PHONE ........................................................................................................... 16
GXP2200EXT MODULE ...................................................................................................................... 17
SAFETY COMPLIANCES .................................................................................................................... 18
WARRANTY......................................................................................................................................... 18
GETTING TO KNOW GXP2130/GXP2140/GXP2160 ............................... 20
IDLE SCREEN ..................................................................................................................................... 20
USING THE KEYPAD .......................................................................................................................... 26
MAKING PHONE CALLS..................................................................................................................... 28
HANDSET, SPEAKER AND HEADSET MODE ........................................................................... 28
MULTIPLE SIP ACCOUNTS AND LINES ..................................................................................... 28
COMPLETING CALLS.................................................................................................................. 29
MAKING CALLS USING IP ADDRESSES ................................................................................... 31
ANSWERING PHONE CALLS ............................................................................................................ 32
RECEIVING CALLS...................................................................................................................... 32
DO NOT DISTURB ....................................................................................................................... 32
DURING A PHONE CALL .................................................................................................................... 33
CALL WAITING/CALL HOLD ....................................................................................................... 33
MUTE ............................................................................................................................................ 33
CALL TRANSFER ........................................................................................................................ 33
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Page 2 of 84
4-WAY/ 5-WAY CONFERENCING ............................................................................................... 34
VOICE MESSAGES (MESSAGE WAITING INDICATOR) ........................................................... 36
SHARED CALL APPEARANCE (SCA) ........................................................................................ 36
CALL FEATURES ................................................................................................................................ 37
CONFIGURATION GUIDE........................................................................ 40
CONFIGURATION VIA KEYPAD ......................................................................................................... 40
CONFIGURATION VIA WEB BROWSER ........................................................................................... 46
DEFINITIONS ...................................................................................................................................... 47
STATUS PAGE DEFINITIONS ..................................................................................................... 47
ACCOUNTS PAGE DEFINITIONS ............................................................................................... 48
SETTINGS PAGE DEFINITIONS ................................................................................................. 57
NETWORK PAGE DEFINITIONS ................................................................................................. 64
MAINTENANCE PAGE DEFINITIONS ......................................................................................... 66
PHONEBOOK PAGE DEFINITIONS ............................................................................................ 69
NAT SETTINGS ................................................................................................................................... 71
WEATHER UPDATE ............................................................................................................................ 72
PUBLIC MODE .................................................................................................................................... 73
EDITING CONTACTS AND CLICK-TO-DIAL ...................................................................................... 74
CONNECTING TO DEVICES ................................................................... 77
EHS HEADSET .................................................................................................................................... 77
BLUETOOTH ....................................................................................................................................... 77
TURNING BLUETOOTH ON OR OFF ......................................................................................... 77
PAIRING GXP2140/GXP2160 WITH A BLUETOOTH DEVICE ................................................... 78
CONNECTING TO A BLUETOOTH EARPHONE ........................................................................ 78
BT ON AND BT OFF ..................................................................................................................... 78
UNPAIRING AND DISCONNECTING A BLUETOOTH DEVICE ................................................. 78
UPGRADING AND PROVISIONING ........................................................ 79
UPGRADE VIA KEYPAD MENU ......................................................................................................... 79
UPGRAGE VIA WEB GUI .................................................................................................................... 79
NO LOCAL TFTP/HTTP SERVERS .................................................................................................... 80
CONFIGURATION FILE DOWNLOAD ................................................................................................ 80
RESTORE FACTORY DEFAULT SETTINGS ........................................... 82
EXPERIENCING THE GXP2130/GXP2140/GXP2160 .............................. 83
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GXP2130/GXP2140/GXP2160 USER MANUAL
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Table of Tables
GXP2130/GXP2140/GXP2160 User Manual
Table 1: GXP2130 /GXP2140/GXP2160 FEATURES IN A GLANCE ......................................................... 10
Table 2: GXP2130/ /GXP2140/GXP2160 COMPARISON GUIDE .............................................................. 11
Table 3 GXP2130 TECHNICAL SPECIFICATIONS .................................................................................... 11
Table 4: GXP2140 TECHNICAL SPECIFICATIONS ................................................................................... 12
Table 5: GXP2160 TECHNICAL SPECIFICATIONS ................................................................................... 14
Table 6: GXP2130/GXP2140/GXP2160 EQUIPMENT PACKAGING ......................................................... 16
Table 7: GXP2130/GXP2140/GXP2160 CONNECTORS ........................................................................... 16
Table 8: GXP2200EXT Packaging .............................................................................................................. 17
Table 9: LCD DISPLAY DEFINITIONS........................................................................................................ 21
Table 10: GXP2130/GXP2140/GXP2160 LCD ICONS ............................................................................... 23
Table 11: GXP2130/GXP2140/GXP2160 Keypad Buttons ......................................................................... 27
Table 12: CALL FEATURES ........................................................................................................................ 37
Table 13: CONFIGURATION MENU ........................................................................................................... 40
Table 14: Status Page Definitions ............................................................................................................... 47
Table 15: Account Page Definitions ............................................................................................................ 48
Table 16: Settings Page Definitions ............................................................................................................ 57
Table 17: Network Page Definitions ............................................................................................................ 64
Table 18: Maintenance Page Definitions ..................................................................................................... 66
Table 19: Phonebook Page Definitions ....................................................................................................... 69
Table of Figures
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GXP2130/GXP2140/GXP2160 User Manual
Figure 1: GXP2140 Back View with GXP2200EXT .................................................................................... 18
Figure 2: GXP2130 Idle Screen .................................................................................................................. 20
Figure 3: GXP2140 Idle Screen .................................................................................................................. 20
Figure 4: GXP2160 Idle Screen .................................................................................................................. 21
Figure 5: GXP2130 Keyboard ..................................................................................................................... 26
Figure 6: GXP2140 Keyboard ..................................................................................................................... 27
Figure 7: GXP2160 Keyboard ..................................................................................................................... 27
Figure 8: Keypad MENU Flow..................................................................................................................... 45
Figure 9: Web Service ................................................................................................................................. 73
Figure 10: Web GUI - Phonebook->Contacts ............................................................................................. 75
Figure 11: Click-to-Dial ................................................................................................................................ 75
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GUI INTERFACE EXAMPLES
http://www.grandstream.com/products/gxp_series/general/documents/gxp2140_2160_gui.zip
1. Screenshot of Login Page
2. Screenshots of Status Pages
3. Screenshots of Accounts Pages
4. Screenshots of Settings Pages
5. Screenshots of Network Pages
6. Screenshots of Maintenance Pages
7. Screenshots of Phonebook Pages
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GNU GPL INFORMATION
GXP2130/GXP2140/GXP2160 firmware contains third-party software licensed under the GNU General
Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the
GNU General Public License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
http://www.grandstream.com/support/faq/gnu_gpl.
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CHANGE LOG
This
section
documents
significant
changes
from
previous
versions
of
user
manuals
for
GXP2130/GXP2140/GXP2160. Only major new features or major document updates are listed here. Minor
updates for corrections or editing are not documented here.
FIRMWARE VERSION 1.0.1.14
This is the initial version.
FIRMWARE VERSION 1.0.1.14
GXP2130/GXP2140/GXP2160 USER MANUAL
Page 8 of 84
WELCOME
Thank
you
for
purchasing
Grandstream
GXP2130/GXP2140/GXP2160
Enterprise
IP
Phone.
GXP2130/GXP2140/GXP2160 is a next generation enterprise grade IP phone. GXP2140/GXP2160
features 4.3 inch TFT Color LCD, 5 programmable context-sensitive soft keys, dual Gigabit network ports,
integrated PoE and Bluetooth, 5-way conference, and Electronic Hook Switch (EHS). GXP2130 supports
2.8 inch TFT Color LCD, 4 programmable context-sensitive soft keys, 4-way conference and EHS with
Plantronics headsets. Also, the series can support up to 3 lines for GXP2130, 4 lines for GXP2140 and 6
lines for GXP2160. The GXP2130/GXP2140/GXP2160 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 GXP2130/GXP2160 supports
presence and Busy Lamp Field (BLF) in the Multi-Purpose Keys as well. The GXP2140 is expandable with
one to 4 expansion modules. The GXP2160/GXP2140/GXP2130 is the perfect choice for enterprise users
looking for a high quality, feature rich multi-line executive IP phone with advanced functionalities and
performance.
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 GXP2130/GXP2140/GXP2160 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:
http://www.grandstream.com/support
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.
FIRMWARE VERSION 1.0.1.14
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PRODUCT OVERVIEW
FEATURE HIGHTLIGHTS
Table 1: GXP2130 /GXP2140/GXP2160 FEATURES IN A GLANCE
GXP2130
GXP2140
GXP2160
FIRMWARE VERSION 1.0.1.14
3 lines
2.8 inch (320x240) TFT color LCD
4 programmable soft keys
8 programmable Multi Purpose Keys
5-way conference
4 lines
4.3 inch (480x272) TFT color LCD
5 programmable soft keys
Bluetooth V2.1
5-way conference
Expansion board
6 lines
4.3 inch (480x272) TFT color LCD
5 programmable soft keys
24 programmable Multi Purpose Keys
Bluetooth V2.1
5-way conference
GXP2130/GXP2140/GXP2160 USER MANUAL
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Table 2: GXP2130/ /GXP2140/GXP2160 COMPARISON GUIDE
Features
LCD Display
LCD
Backlight
Number of
Lines
Programmabl
e Hard Keys
Programmabl
e Soft Keys
Extension
GXP2130
GXP2140
GXP2160
320x240
480 x 272
480 x 272
Yes
Yes
Yes
N/A
24
N/A
Module
Yes,
N/A
up to 4 EXT Boards,
GXP2130/GXP2140/GXP2160 TECHNICAL SPECIFICATIONS
Table 3 GXP2130 TECHNICAL SPECIFICATIONS
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A
record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE,
LLDP, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6
Network Interfaces
Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with
integrated PoE
Graphic Display
2.8 inch (320x240) TFT color LCD
Feature Keys
3 line keys with up to 3 SIP accounts, 8 speed-dial/BLF extension keys with
dual-color LED, 4 programmable context sensitive softkeys, 5 navigation/menu
keys, 11 dedicated function keys for: MESSAGE (with LED indicator),
PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET,
MUTE,
SEND/REDIAL, SPEAKERPHONE, VOL+, VOLVoice Codec
Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), and iLBC, in-band
and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Auxiliary Ports
RJ9 headset jack (allowing EHS with Plantronics headsets), USB, extension module port
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Telephony Features
Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call-appearance
(SCA)/bridged-line-appearance (BLA), downloadable phonebook (XML, LDAP, up to 2000
items), call waiting, call log (up to 500 records), customization of screen, off-hook auto dial,
auto answer, click-to-dial, flexible dial plan, hot desking, personalized music ringtones and
music on hold, server redundancy and fail-over
Sample Applications
Weather, currency, GMI available for advanced custom application development
HD audio
Yes, both on handset and speakerphone
Base Stand
Yes, allow 2 angle positions
Wall Mountable
Yes
QoS
Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level passwords, MD5 and MD5-sess based authentication, AES
based secure configuration file, SRTP, TLS, 802.1x media access control
Multi-language
English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew, Hungarian,
Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia, Spanish, Turkish
Upgrade/Provisioning
Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or encrypted
XML configuration file
Power
Green
Energy Efficiency
Universal power adapter included: Input:100-240V ; Output: +12V, 1.5A ;
Integrated Power-over-Ethernet (802.3af)
Max power consumption 3W (power adapter) or 4.5W (PoE, Class 2)
Physical
Dimension: 193mm (W) x 211mm (L) x 84.5 mm (H);
Unit weight: 0.78kg ;
Temperature
and
Package weight: 1.3kg
32-104℉ / 0～40℃, 10-90% (non- condensing)
Humidity
Package Content
GXP2130 phone, handset with cord, base stand, universal power supply, network cable,
Quick Start Guide
Compliance
FCC Part15 Class B, EN55022 ClassB, EN61000-3-2, EN61000-3-3, EN55024,
EN60950-1, AS/NZS CISPR22 Class B
Table 4: GXP2140 TECHNICAL SPECIFICATIONS
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A
record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP,
LDAP, TR-069, 802.1x, TLS, SRTP, IPv6
Network Interfaces
Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with
integrated PoE
Graphic Display
4.3 inch (480x272) TFT color LCD
Bluetooth
Yes, Bluetooth V2.1
Feature Keys
4 line keys with up to 4 SIP accounts, 5 programmable context sensitive softkeys,
5 navigation/menu keys, 11 dedicated function keys for : MESSAGE (with LED
indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET,
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MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOLVoice Codec
Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), and in-band and
out-of-band DTMF (in audio, RFC2833, SIP INFO)
Auxiliary Ports
RJ9 headset jack (allowing EHS with Plantronics headsets), USB, extension
module port
Telephony Features
Hold,
transfer,
forward,
5-way
conference,
call
park,
call
pickup,
shared-call-appearance (SCA)/bridged-line-appearance (BLA), downloadable
phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 500
records), customization of screen, off-hook auto dial, auto answer, click-to-dial,
flexible dial plan, hot desking, personalized music ringtones and music on hold,
server redundancy and fail-over
Sample Applications
Weather, currency, GMI available for advanced custom application development
HD audio
Yes, both on handset and speakerphone
Extension Module
Yes, can power up up to 4 GXP2200EXT modules which features a 128x384
graphic LCD, 20 quick-dial/BLF keys which dual-color LED, 2 navigation keys,
and less than 1.2W power consumption per unit.
Base Stand
Yes, allow 2 angle positions
Wall Mountable
Yes
QoS
Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User
and
administrator
level
passwords,
MD5
and
MD5-sess
based
authentication, AES based secure configuration file, SRTP, TLS, 802.1x media
access control
Multi-language
English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew,
Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia,
Spanish, Turkish
Upgrade/Provisioning
Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or
encrypted XML configuration file
Power
Green
Energy Efficiency
Universal power adapter included:
Input:100-240V ; Output: +12V, 1A ;
Integrated Power-over-Ethernet (802.3af)
Max power consumption: 6W (without GXP2200EXT), 10W(with 4 cascaded
GXP2200EXTs )
Physical
Dimention: 222mm (W) x 210mm (L) x 93mm (H); Unit weight: 0.98kg; Package
weight: 1.55kg
Temperature
and
32-104℉ / 0～40℃, 10-90% (non- condensing)
Humidity
Package Content
GXP2140 phone, handset with cord, base stand, universal power supply, network
cable, Quick Start Guide
Compliance
FCC Part15 Class B, EN55022 ClassB, EN61000-3-2, EN61000-3-3, EN55024,
FIRMWARE VERSION 1.0.1.14
GXP2130/GXP2140/GXP2160 USER MANUAL
Page 13 of 84
EN60950-1, AS/NZS CISPR22 Class B
Table 5: GXP2160 TECHNICAL SPECIFICATIONS
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A
record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP,
LDAP, TR-069, 802.1x, TLS, SRTP, IPv6
Network Interfaces
Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with
integrated PoE
Graphic Display
4.3 inch (480x272) TFT color LCD
Bluetooth
Yes, Bluetooth V2.1
Feature Keys
6 line keys with up to 6 SIP accounts, 24 speed-dial/BLF extension keys with
dual-color LED, 5 programmable context sensitive softkeys, 5 navigation/menu
keys, 11 dedicated function keys for : MESSAGE (with LED indicator),
PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET,
MUTE,
SEND/REDIAL, SPEAKERPHONE, VOL+, VOLVoice Codec
Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), and in-band and
out-of-band DTMF (in audio, RFC2833, SIP INFO)
Auxiliary Ports
RJ9 headset jack (allowing EHS with Plantronics headsets), USB, extension
module port
Telephony Features
Hold,
transfer,
forward,
5-way
conference,
call
park,
call
pickup,
shared-call-appearance (SCA)/bridged-line-appearance (BLA), downloadable
phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 500
records), customization of screen, off-hook auto dial, auto answer, click-to-dial,
flexible dial plan, hot desking, personalized music ringtones and music on hold,
server redundancy and fail-over
Sample Applications
Weather, currency, GMI available for advanced custom application development
HD audio
Yes, both on handset and speakerphone
Base Stand
Yes, allow 2 angle positions
Wall Mountable
Yes
QoS
Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User
and
administrator
level
passwords,
MD5
and
MD5-sess
based
authentication, AES based secure configuration file, SRTP, TLS, 802.1x media
access control
Multi-language
English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew,
Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia,
Spanish, Turkish
Upgrade/Provisioning
Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or
encrypted XML configuration file
Power
Green
Universal power adapter included:
FIRMWARE VERSION 1.0.1.14
GXP2130/GXP2140/GXP2160 USER MANUAL
Page 14 of 84
Energy Efficiency
Input:100-240V ; Output: +12V, 1A ;
Integrated Power-over-Ethernet (802.3af)
Max power consumption:6W
Physical
Dimention: 222mm (W) x 210mm (L) x 93mm (H); Unit weight: 0.98kg; Package
weight: 1.62kg
Temperature
and
32-104℉ / 0～40℃, 10-90% (non- condensing)
Humidity
Package Content
GXP2160 phone, handset with cord, base stand, universal power supply, network
cable, Quick Start Guide
Compliance
FCC Part15 Class B, EN55022 ClassB, EN61000-3-2, EN61000-3-3, EN55024,
EN60950-1, AS/NZS CISPR22 Class B
FIRMWARE VERSION 1.0.1.14
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Page 15 of 84
INSTALLATION
EQUIPMENT PACKAGING
Table 6: GXP2130/GXP2140/GXP2160 EQUIPMENT PACKAGING
Main Case
Yes
Handset
Yes
Phone Cord
Yes
Power Adaptor
Yes
Ethernet Cable
Yes
Phone Stand
Yes
Wall Mount
Yes
Quick Start Guide
Yes
CONNECTING YOUR PHONE
Table 7: GXP2130/GXP2140/GXP2160 CONNECTORS
Handset Port
RJ9 handset connector port
Headset Port
RJ9 headset connector port (supporting EHS with Plantronics headset)
USB Port
USB port connecting a USB flash drive. (not on GXP2130 )
LAN Port
10/100/1000Mbps RJ-45 port connecting to Ethernet
PC Port
10/100/1000Mbps RJ-45 port connecting to PC
EXT Port
RJ11 connector port to connect the GXP2200EXT Board(GXP2140)
Power Jack
12V DC Power connector port
To set up the GXP2130/GXP2140/GXP2160, 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 RJ45 socket of a hub/switch or a router (LAN side of the
router) using the Ethernet cable;
4. Connect the 12V 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.
GXP2200EXT MODULE
The GXP2140 is expandable with GXP2200EXT modules. GXP2140 supports up to 4 extension modules,
adding 160 fully programmable phone extensions to the phone.
GXP2200EXT board package contains:
Table 8: GXP2200EXT Packaging
GXP2200EXT Main Case
Yes (1)
GXP2200EXT Stand
Yes (1)
RJ11-RJ11 Cable
Yes (1)
Connector Plate
Yes (1)
Screws
Yes (4)
Quick Installation Guide
Yes (1)
Note:
The GXP2200EXT board is an additional accessory for the GXP2140 and is not included in the GXP2140
box.
To set up the GXP2200EXT board with the GXP2140, please follow the steps below:
1. On the back of the GXP2140 and the GXP2200EXT, there are slots for connector plate. Attach the
connector plate between the slots for GXP2140 and the GXP2200EXT;
2. Apply the screws provided and securely tighten them in (See Figure 1: GXP2140 Back View with
GXP2200EXT ).
3. Connect the GXP2200EXT board to the GXP2140 via EXT port using the RJ11-RJ11 cable provided;
4. Install the phone stand on the GXP2140 and the GXP2200EXT board;
5. Power up the GXP2140. The GXP2200EXT board will show the booting up screen with version
information and connecting status;
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6. After successfully booting up, the GXP2200EXT board will stay in idle. Press and hold the Left button
for 3 to 5 seconds to check the version information and status.
Figure 1: GXP2140 Back View with GXP2200EXT
The GXP2200EXT board can be configured via the web GUI of the GXP2140 connected. After
successfully configured, press Left or Right button on the GXP2200EXT board and users could browse all
the MPK's status in different pages.
For more information on installing, connecting and configuring the GXP2200EXT board with GXP2140,
please refer to the GXP2200EXT board user manual:
http://www.grandstream.com/support
SAFETY COMPLIANCES
The GXP2130/GXP2140/GXP2160 phone complies with FCC/CE and various safety standards. The
GXP2130/GXP2140/GXP2160 power adapter is compliant with the UL standard. Use the universal power
adapter provided with the GXP2130/GXP2140/GXP2160 package only. The manufacturer’s warranty does
not cover damages to the phone caused by unsupported power adapters.
WARRANTY
If the GXP2130/GXP2140/GXP2160 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 Support for a RMA (Return Materials Authorization) number
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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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GETTING TO KNOW GXP2130/GXP2140/GXP2160
IDLE SCREEN
GXP2130/GXP2140/GXP2160 screen displays differently depending on whether the phone is idle or in
use (active). The following figures show the idle screen of GXP2130/GXP2140/GXP2160.
Figure 2: GXP2130 Idle Screen
Figure 3: GXP2140 Idle Screen
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Figure 4: GXP2160 Idle Screen
The following table describes the items displayed on the GXP2130/GXP2140/GXP2160 idle screen.
Table 9: LCD DISPLAY DEFINITIONS
DATE AND TIME
NETWORK ICON
STATUS ICON
LINE STATUS INDICATOR
Displays the current date and time. It can be synchronized with
Internet time servers.
Shows the status of network. It will indicate whether the network is
down or starting.
Shows the status of the phone for registration status, call features and
etc, using icons as shown in the next table.
Displays the name of the account that is in use.
The softkeys are context sensitive and will change depending on the
status of the phone. Typical functions assigned to softkeys are:
SwitchSCR/NEXTSCR
Toggles between different idle screens. For example, for
GXP2130/GXP2140/GXP2160, pressing most left soft key (switch
SOFTKEYS in Idle Screen
screen)
will
toggle
among
default
idle
screen,
weather
information, currency information and Grandstream Upcoming
Events;
ForwardAll
Unconditionally forwards the phone line (account 1) to another
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phone.
MissedCalls
Shows up unanswered calls to this phone.
Redial
Redials the last dialed number when there is existed dialed call
log.
Note:
If XML application is used for GXP2130/GXP2140/GXP2160, 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. Here are the main softkeys in call screen.
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.
AnswerCall
Answers the incoming call when the phone is ringing.
SOFTKEYS in Call Screen
RejectCall
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,
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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.
Call Parking: Please refer to UCM61xx Online User Manual for more
SPECIAL SOFTKEYS
information.
(Only for
GXP2130/GXP2140/GXP2160
when integrated with
UCM61xx)
CallPark
When the phone dials out, the Call Park softkey will display on
screen. To park the call, press the "CallPark" softkey and select a
green MPK to park the call on available parking lot.
Table 10: GXP2130/GXP2140/GXP2160 LCD ICONS
Network Status.
OFF - Network connection is up
ON - Network connection is down
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
Call Forward No Answer Status.
OFF - Call Forward No Answer feature disabled
ON - Call Forward No Answer feature enabled
Call Forward Busy and Call Forward No Answer Status.
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OFF - Call Forward All and Call Forward No Answer feature disabled
ON - Call Forward All and Call Forward No Answer feature enabled
Missed Call Status.
OFF - No missed call
ON - New missed call
Handset Status.
OFF - Handset is not in use
ON - Handset is in use
Headset Status.
OFF - Headset is not in use
ON - Headset is in use
Microphone MUTE Status.
OFF - No muted
ON - Muted
Bluetooth Status. (GX2140/GXP2160)
OFF - Bluetooth is off
ON - Bluetooth is on
USB Status. (GX2140/GXP2160)
OFF - No USB connection
ON - USB connected
Incoming Call.
The current call is an incoming call
Outgoing Call.
The current call is an outgoing call
Call On Hold.
OFF - No call on hold
ON – Call on hold
Call Connect.
OFF - No call
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ON - Call connected
Call Conference.
OFF - No Conference call
ON - Conference call is established
Click Hand
In call screen, this icon shows on the line available for transfer or
conference.
IP Call Status
The current call is an IP call.
SIP Call
Enter number for initiate call. The current call is a SIP call.
Call Time
During an active call, this icon shows the call time.
LCD Brightness Up
Brightness is increasing
LCD Brightness Down
Brightness is decreasing
Speaker Volume Up
Call screen icon. Speaker volume is increasing
Speaker Volume Down
Call screen icon. Speaker volume is decreasing
Handset Volume Up
Call screen icon. Handset volume is increasing
Handset Volume Down
Call screen icon. Handset volume is decreasing
Ringtone Volume Down
Call screen icon. The volume of ringtone is decreasing
Ringtone Volume Up
Call screen icon. The volume of ringtone is increasing
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Ringtone Volume Up
Call screen icon. The volume of ringtone is increasing
Ringtone Volume Down
Call screen icon. The volume of ringtone is decreasing
VOC status
Call screen icon.
OFF - VOC is off.
ON - VOC is on.
USING THE KEYPAD
Figure 5: GXP2130 Keyboard
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Figure 6: GXP2140 Keyboard
Figure 7: GXP2160 Keyboard
Table 11: GXP2130/GXP2140/GXP2160 Keypad Buttons
Contacts. Press to view and edit contact information.
Hold. Press to put call on hold or resume the call.
Mute. Press to mute or unmute the call.
Headset. Press to switch to headset mode.
Message. Press to view voicemail messages.
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Transfer. Press to transfer call.
Conference. Press to conference call.
Speaker.
Send/Redial.
Volume.
Up Navigation key. In idle screen, press to see the IP.
MAKING PHONE CALLS
HANDSET, SPEAKER AND HEADSET MODE
The GXP2130/GXP2140/GXP2160 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
GXP2160 can support up to 6 independent SIP accounts; GXP2140 can support up to 4 independent SIP
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 GXP2130/GXP2140/GXP2160, 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 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.
Predictive dialing. When dialing numbers, based on the entered digit or digits, the phone will predict
and list the candidates of target number. If the target number already saved in phonebook, when enter
the first several digits, phone will display a list of the matched numbers. If the target number appears
on the list, user could select the number by Up/Down key and dial out. GXP2130/GXP2140/GXP2160
have predefined call feature code (the first digit is *). When user enter * as the first digit of target
number, the phone will list all feature codes as candidates.
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 soft key.
Via Call History. Dial the number logged in phone's call history.
Press MENU button to bring up the main menu;
Enter Call History;
Select the entry you would like to call using the navigation arrow keys;
Press SEND button to dial out.
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Via Phonebook. Dial the number from the phonebook.
Press Contacts button;
Under Contacts, enter Local Phonebook/ Broadsoft Phonebook using the navigation arrow key;
Select the contact you would like to call using the navigation arrow key;
Press SEND button to dial the selected contact.
Speed Dial from Line Key. Dial the number configured as Speed Dial on Line Key.
Go to phone's Web GUI->Settings->Programmable Keys, 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->Settings->Programmable Keys, 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 Paging/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 X->Call Settings;
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For Paging/Intercom, if the SIP Server/PBX supports the feature and has Paging/Intercom feature
code set up already, users do not necessarily need toggle to paging mode in the call screen. Simply
dial the feature code with extension as a normal call.
When dial out via paging, user can see the “call-info” header contains “answer-after=0” and “the
“alert-info” header contains “info=alert-autoanswer;delay=0” in the outgoing INVITE.
During an active call, user can see call info on LCD:
shows the call lasting time;
shows
contact information if the number already be saved in phonebook. When dialing DTMF, user can press
round OK button to switch to call info page.
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:
When phone is at idle state, press any number key or * key to bring up “Onhook Dialing” page;
Press round menu key or "DirectIP" soft key;
Input the target IP address (Please see example below) ;
Press "OK" soft key 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 GXP2130/GXP2140/GXP2160 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 CSMA/CD
without a SIP server. Controlled static IP usage is recommended.
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To enable Quick IP Call Mode, go to phone's Web GUI->Settings->Call Features, set "Use Quick IP Call
Mode" to "Yes". Clicking on "Save and Apply" on the bottom of 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).
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 from phone's LCD by following the steps below:
Press the Menu button and select "Preference" using navigation keys;
Press Menu button again to get into Preference options;
When "Do Not Disturb" is highlighted, press Left/Right key to disable/enable DND;
Press “Save” soft key to save the change.
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When Do Not Disturb feature is turned on, the DND icon will appear on the top of the LCD. The incoming
call will not be accepted or the call will directly go into voicemail.
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 new incoming call during the active call.
MUTE
During an active call, press the MUTE button to mute/unmute the microphone. The LCD will show the Mute
icon on the top of screen, when the call is muted.
CALL TRANSFER
GXP2130/GXP2140/GXP2160 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.
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Auto-Attended Transfer.
Set "Auto-Attended Transfer" to "Yes" under Web GUI->Settings->Call Features. And then click
"Save and Apply" on the bottom of the page;
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;
Dial the number and press SEND or # to make a second call. (Once the number is entered, a
"Transfer" soft key will show. If "Transfer" soft key is pressed instead of SEND or #, a blind transfer
will be performed);
Press TRAN key again. The call will be transferred.
For Auto-Attended Transfer, after dialing out the number for the second call, a "Split" soft key will
show. If the second call is not established yet (ringing), pressing "Split" will hang up the second call.
If the second call is established (answered), pressing "Split" will resume the second call and keep
the first call on hold.
Note:
To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains.
4-WAY/ 5-WAY CONFERENCING
GXP2130 can host up to 4-way conference call with other parties (PCMU/PCMA). GXP2140/GXP2160
can host up to 5-way conference call.
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 on hold to join the conference.
Cancel Conference.
If after pressing the CONF key, the user decides not to conference, press Cancel soft key or the
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current active LINE key (LED in solid green);
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;
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 soft key 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 soft key or simply hang up the call to terminate the conference call.
GXP2130/GXP2140/GXP2160 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
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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.
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 x->General Settings). An IVR will prompt the user through the process
of message retrieval.
SHARED CALL APPEARANCE (SCA)
The GXP2130/GXP2140/GXP2160 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.
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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
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"->"Programmable Keys" on the web UI and set the
line key mode as "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 GXP2130/GXP2140/GXP2160 supports traditional and advanced telephony features including caller
ID, caller ID with caller Name, call forward and etc.
Table 12: CALL FEATURES
Block Caller ID (for all subsequent calls)
*30
Off hook the phone;
Dial *30.
Send Caller ID (for all subsequent calls)
*31
Off hook the phone;
Dial *31.
Disable Call Waiting
*50
Off hook the phone;
Dial *50.
Enable Call Waiting
*51
Off hook the phone;
Dial *51.
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Call with Caller ID Blocked (per call)
*67
Off hook the phone;
Dial *67 and then enter the number to dial out.
Call with Caller ID Enabled (per call)
*82
Off hook the phone;
Dial *82 and then enter the number to dial out.
Call with Call Waiting Disable (per Call)
*70
Off hook the phone;
Dial *70 and then enter the number to dial out.
Call with Call Waiting Enabled (per Call)
*71
Off hook the phone;
Dial *71 and then enter the number to dial out.
Unconditional Call Forward. To set up unconditional call forward:
*72
Off hook the phone;
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:
*73
Off hook the phone;
Dial *73;
Hang up the call.
Busy Call Forward. To set up busy call forward:
*90
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:
*91
Off hook the phone;
Dial *91;
Hang up the call.
Delayed Call Forward. To set up delayed call forward:
*92
Off hook the phone;
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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:
*93
Off hook the phone;
Dial *93;
Hang up the call.
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CONFIGURATION GUIDE
The GXP2130/GXP2140/GXP2160 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 the LCD 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 or “Select” soft key to enter the selected
option;
Exit. Press “Exit” soft key to exit to the previous menu;
Return to Home page.
In the Main menu, press Home soft key to return home screen;
In sub menu, press and hold “Exit” soft key until the Exit soft key changes to Home soft key, then
release the soft key.
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 60 seconds.
When the phone is in idle, pressing the UP navigation key can see phone’s IP address.
The MENU options are listed in the following table.
Table 13: CONFIGURATION MENU
Call History
Displays Local call logs and Broadsoft call log:
Local Call Log
Answered calls, dialed calls, missed calls, transferred calls and forwarded
calls.
Broadsoft Call Log
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Status
Displays account status, network status, software version number and Hardware
Account status.
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.
Software Version
Press to enter the sub menu for Boot, Core, Base and Prog version.
Hardware
Press to enter the sub menu for MAC address, Hardware version and P/N
number.
Contacts
Contacts sub menu includes the following options:
Local Phonebook
Local Group
Broadsoft Phonebook
LDAP Directory
Contacts sub menu is for Local Phonebook, Local Group, LDAP Directory and
Broadsoft Phonebooks. User could configure phonebooks/groups/LDAP
options here, download phonebook XML to the phone and search
phonebook/LDAP directory.
Messages
Message sub menu include the following options:
Instant Message
Displays received instant messages;
Voice Mails
Displays
voicemail message information in the format below:
messages/all messages (urgent messages/all urgent messages).
Preference
Preference sub menu includes the following options:
Do Not Disturb
Enables/disables Do Not Disturb on the phone.
Sounds
o Ring Tone
Configures different ring tones for incoming call.
o Ring Volume
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new
Adjusts ring volume by pressing left/right arrow key.
Appearance
o Active LCD Brightness
Adjusts active LCD brightness by pressing left/right arrow key
o Idle LCD Brightness
Adjusts idle LCD brightness by pressing left/right arrow key
o Display Language
Selects the language to be displayed on the phone's LCD. Users
could select Automatic for local language based on IP location if
available.
Time Settings
Configures date and time on the phone.
Phone
Phone sub menu includes the following options:
SIP
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.
Call Features
Configures call forward features for Forward All, Forward Busy, Forward No
Answer and No Answer Timeout.
System
System sub menu includes the following options:
Network
o IP Setting
Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE
account ID and password; Configures static IP address, Netmask,
Gateway, DNS Server 1 and DNS Server 2.
o 802.1X
Enables/Disables 802.1X mode; Configures 802.1x identity and
MD5 password.
o Layer 2 QoS
Configures 802.1Q/VLAN Tag and priority value.
Bluetooth Settings(Only on GXP2140/GXP2160)
o Power
Turns on/off the Bluetooth feature.
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o Start Scan
Starts to scan other Bluetooth devices around the phone. If found,
user could press “Pair” soft key, and enter Pin code to pair to other
Bluetooth devices.
o Discoverable
Defines if the phone is discoverable by other Bluetooth devices.
o MAC
Displays the MAC address of the phone.
Upgrade
o Firmware Server
Configures firmware server for upgrading the phone.
o Config Server
Configures config server for provisioning the phone.
o Upgrade Via
Specifies upgrade/provisioning via TFTP/HTTP/HTTPS.
o Start Provison
Starts Provision immediately.
Factory Functions
o Diagnostic Mode
All LEDs will light up. All keys’ name will display in red on LCD
screen before diagnosing. Press any key on the keypad to
diagnose the key’s function. When done, the key’s name will
display in green on LCD. Lift and put back the handset to exit
diagnostic mode.
o Audio Loopback
Speak to the phone using speaker/handset/headset. If you can
hear your voice, your audio is working fine. Press “Exit” soft key to
exit audio loopback mode.
o LCD on/off
Selects this option to turn off LCD. Press any button to turn on
LCD.
o LCD Diagnostic
Enters this option and press Left/Right Navigation key to do LCD
Diagnostic. Press “Exit” soft key to quite.
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Detect/connect UCM server to process auto-provision. Manually input the
IP and port of the UCM server phone wants to bind with; Or select from the
available UCM server in network.
Operations
o Factory Reset
Resets the phone to factory default settings.
Reboot
Reboots the phone.
The following picture shows the keypad MENU configuration flow.
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MENU
Call History
Status
Contacts
Account Status
Network Status
System information
Hardware
Local Phonebook
Local Group
Broadsoft Phonebook
LDAP Directory
Messages
Instant Messages
Voice Mails
Preference
Do Not Disturb
Star Key Lock
Sounds
Appearance
Time Settings
Phone
System
SIP
Call Features
Network
Bluetooth
Upgrade
Factory Functions
UCM Detect
Factory Reset
Reboot
Enable/Disable
Password
Ring Tone
Ring Volume
Active LCD Brightness
Idle LCD Brightness
Display Language
IP Settins
802.1X
Layer 2 QoS
Power
Start Scan
Discoverable
MAC
Firmware Server
Config Server
Upgrade via
Start Provision
Diagnostic Mode
Audio Loopback
LCD on/off
LCD Diagnostic
Figure 8: Keypad MENU Flow.
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CONFIGURATION VIA WEB BROWSER
The GXP2130/GXP2140/GXP2160 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
Google Chrome, Mozilla Firefox and Microsoft’s IE.
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 Up arrow button when phone is at idle state;
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->Network 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 has IP address 192.168.40.154, please enter "http://192.168.40.154" in the address bar of
the browser;
There are two default passwords for the login page:
User Level
User
Password
Web Pages Allowed
End User Level
user
123
Only Status and Basic Settings
Administrator Level
admin
admin
Browse all pages
The password is case sensitive with maximum length of 25 characters.
When changing any settings, always SUBMIT them by pressing the "Save" or "Save and Apply" button
on the bottom of the page. If the change is saved only but not applied, after making all the changes,
click on the "APPLY" button on top of the page to submit. 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
"Accounts" page and "Phonebook" page do not require reboot. Most of the options under "Settings"
page do not require reboot).
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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.
Status: Displays the Account status, Network status, and System Info of the phone;
Account: To configure the SIP account;
Settings: To configure call features, ring tone, audio control, LCD display, date and time, Web
services, XML applications, programmable keys and etc.;
Network: To configure network settings;
Maintenance: To configure web access, upgrading and provisioning, syslog, language settings,
TR-069, security and etc.;
Phonebook: To manage Phonebook and LDAP.
STATUS PAGE DEFINITIONS
Table 14: Status Page Definitions
Status -> Account Status
Account x
Account index.
SIP User ID
Displays the configured SIP User ID.
SIP Server
Displays the configured SIP Server address.
SIP Registration
Displays SIP registration status YES/NO.
Status -> Network Status
Global unique ID of device, in HEX format. The MAC address will be used for
MAC Address
provisioning and can be found on the label coming with original box and on the
label located on the back of the device.
IP Setting
DHCP, Static IP or PPPoE.
IPv4 Address
The IPv4 address obtained on the phone.
IPv6 Address
The IPv6 address obtained on the phone.
Subnet Mask
The subnet mask obtained on the phone.
Gateway
The gateway address obtained on the phone.
DNS Server 1
The DNS server address 1.
DNS Server 2
The DNS server address 2.
PPPoE Link Up
PPPoE connection status.
NAT Type
NAT traversal status for each account.
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Status -> System Info
Product Model
Product model of the phone.
Part Number
Product part number.
Software Version
Boot: boot version number;
Core: core version number;
Base: base version number;
Prog: program version number. This is the main firmware release number,
which is always used for identifying the software system of the phone;
Aux: Aux version number;
Dsp: DSP version number.
System Up Time
System up time since the last reboot.
System Time
Current system time on the phone system.
Service Status
GUI and Phone service status.
Core Dump
Core dump file that could be downloaded for troubleshooting purpose.
ACCOUNTS PAGE DEFINITIONS
Table 15: Account Page Definitions
Account x -> General Settings
Account Active
Activates/deactivates account. The default setting is "Yes".
Account Name
The name associated with the SIP account.
SIP Server
Secondary SIP Server
The URL or IP address, and port of the SIP server. This is provided by your
VoIP service provider (ITSP).
The URL or IP address, and port of the SIP server. This will be used when the
primary SIP server fails.
IP address or Domain name of the Primary Outbound Proxy, Media Gateway,
Outbound Proxy
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.
User account information, provided by your VoIP service provider (ITSP). It's
SIP User ID
usually in the form of digits similar to phone number or actually a phone
number.
Authenticate ID
Authenticate Password
SIP service subscriber's Authenticate ID used for authentication. It can be
identical to or different from the SIP User ID.
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
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appear as hidden for security purpose.
Name
The SIP server subscriber's name (optional) that will be used for Caller ID
display.
Allows you to access voice messages by pressing the MESSAGE button on
Voice Mail User ID
the phone. This ID is usually the VM portal access number. For example, in
Asterisk server, 8500 could be used.
Account x -> Network Settings
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:
Backup IP 1;
Backup IP 2.
If SIP server is configured as domain name, phone will not send DNS query,
but use “Primary IP” or “Backup IP x” to send SIP message if at least one of
them are not empty. Phone will try to use “Primary IP” first. After 3 tries without
any response, it will switch to “Backup IP x”, and then it will switch back to
“Primary IP” after 3 re-tries.
If SIP server is already an IP address, phone will use it directly even “User
Configured IP” is selected.
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
NAT Traversal
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
"Keep-Alive". Configure this to be "No" if an outbound proxy is used. "STUN"
cannot be used if the detected NAT is symmetric NAT.
A SIP Extension to notify the SIP server that the phone is behind a
Proxy-Require
NAT/Firewall. Do not configure this parameter unless this feature is supported
on the SIP server.
Account x -> SIP Settings -> Basic Settings
If the phone has an assigned PSTN telephone number, this field should be set
TEL URI
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
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number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP
request. The default setting is "Disable".
SIP Registration
Selects whether or not the phone will send SIP Register messages to the
proxy/server. The default setting is "Yes".
If set to "Yes", the SIP user's registration information will be cleared when the
Unregister On Reboot
phone reboots. The SIP Contact header will contain "*" to notify the server to
unbind the connection. The default setting is "No".
Specifies the frequency (in minutes) in which the phone refreshes its
Register Expiration
registration with the specified registrar. The default value is 60 minutes. The
maximum value is 64800 minutes (about 45 days).
Reregister Before
Specifies
the
time
frequency
(in
seconds)
that
the
phone
sends
Expiration
re-registration request before the Register Expiration. The default value is 0.
Defines the local SIP port used to listen and transmit. The default value is
Local SIP Port
5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066 for Account
4, 5068 for Account 5, 5070 for Account 6.
SIP Registration Failure
Specifies the interval to retry registration if the process is failed. The default
Retry Wait Time
value is 20 seconds.
SIP T1 Timeout
SIP T1 Timeout. The default setting is 0.5 seconds.
SIP T2 Timeout
SIP T2 Timeout. The default setting is 4 seconds.
SIP Transport
Determines the network protocol used for the SIP transport. Users can choose
from TCP, UDP and TLS.
SIP URI Scheme when
Specifies if "sip:" or "sips:" will be used when TLS/TCP is selected for SIP
using TLS
Transport. The default setting is "sips:".
Use Actual Ephemeral
Defines whether the actual ephemeral port in contact with TCP/TLS will be
Port in Contact with
used or not. This is used when TLS/TCP is selected for SIP Transfer. The
TCP/TLS
default setting is "No".
Remove OBP from route
Support SIP Instance ID
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 SIP Instance ID is supported or not. The default setting is
"Yes".
When set to "Yes", a SUBSCRIBE for Message Waiting Indication will be sent
SUBSCRIBE for MWI
periodically. The phone supports synchronized and non-synchronized MWI.
The default setting is "No".
SUBSCRIBE for
When set to "Yes", a SUBSCRIBE for Registration will be sent out periodically.
Registration
The default setting is "No".
Enable 100rel
The use of the PRACK (Provisional Acknowledgment) method enables
reliability to SIP provisional responses (1xx series). This is very important in
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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.
When set to "Auto", the phone will look for the caller ID in the order of
P-Asserted Identity Header, Remote-Party-ID Header and From Header in the
Caller ID Display
incoming SIP INVITE. When set to "Disabled", all incoming calls are displayed
with "Unavailable". When set to "From Header", the phone will display the
caller ID based on the From Header in the incoming SIP INVITE. The default
setting is "Auto".
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
Use Privacy Header
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.
Controls whether the P-Preferred-Identity Header will present in the SIP
INVITE message or not. The default setting is "default", which is when
Use P-Preferred-Identity
"Huawei IMS" special feature is on, the P-Preferred-Identity Header will not
Header
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.
Account x -> SIP Settings -> Advanced Features
Broadsoft Call Center
Hoteling Event
Call Center Status
Publish to Call Center
Feature Key
Synchronization
Line Seize Timeout
When set to "Yes", Feature Key Synchronization will be enabled regardless of
web settings. The default setting is "No".
Enables Broadsoft Hoteling event feature. The default setting is "No".
When set to "Yes", the phone will send SUBSCRIBE to the server to obtain call
center status. The default setting is "No".
When set to "Yes", users could select "Away", "Online" or "Busy" from LCD
menu and publish it to call center. The default setting is "No".
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".
Defines the interval (in seconds) before the line can be seized when Shared
Line is used. The default value is 15 seconds.
Configures the eventlist BLF URI on the phone to monitor the extensions in the
Eventlist BLF URI
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.,
BLF1006@myserver.com) with a list of extension included. On the phone, in
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this "eventlist BLF URI" field, fill in the URI without the domain (i.e., BLF1006).
To
monitor
the
extensions
in
the
list,
under
Web
GUI->Settings->Programmable Keys page, please select "eventlist BLF" in the
key mode, choose account, enter the value of each extension in the list.
Conference URI
Music On Hold URI
BLF Call-pickup Prefix
PUBLISH for Presence
Omit charset=UTF-8 in
MESSAGE
Configures the conference URI when using Broadsoft N-way calling feature.
Configures Music On Hold URI to call when a call is on hold. This feature has
to be supported on the server side.
Configures the prefix prepended to the BLF extension when the phone picks
up a call with BLF key. The default setting is **.
Enables presence feature on the phone. The default setting is "No".
Omit charset=UTF-8 in MESSAGE content-type
Enable User Presence
The phone will subscribe and notify about user presence to SIP server. The
Subscription
SIP server should support this feature.
Hide Login Soft Key On
Auto User Presence
Subscription
Auto User Presence ID
The soft key to login or logout will be hidden on idle screen if auto login ID is
configured below.
The phone will subscribe user presence automatically with this ID once the
account is registered.
Auto User Presence
The phone will subscribe user presence automatically with this password once
password
the account is registered.
Different soft switch vendors have special requirements. Therefore users may
Special Feature
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".
Account x -> SIP Settings -> Session Timer
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
Session Expiration
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.
Min-SE
Caller Request Timer
Callee Request Timer
The minimum session expiration (in seconds). The default value is 90
seconds.
If set to "Yes" and the remote party supports session timers, the phone will use
a session timer when it makes outbound calls.
If set to "Yes" and the remote party supports session timers, the phone will use
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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
Force Timer
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".
UAC Specify Refresher
UAS Specify Refresher
Force INVITE
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.
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.
Account x -> SIP Settings -> Security Settings
Check Domain
Defines whether the domain certificates will be checked or not when TLS/TCP
Certificates
is used for SIP Transport. The default setting is "No".
Validate Incoming
Defines whether the incoming messages will be validated or not. The default
Messages
setting is "No".
Check SIP User ID for
incoming INVITE
Accept Incoming SIP
from Proxy Only
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".
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
account, the call will be rejected. The default setting is "No".
Authenticate Incoming
If set to "Yes", the phone will challenge the incoming INVITE for authentication
INVITE
with SIP 401 Unauthorized response. The default setting is "No".
Account x -> Audio Settings
Specifies the mechanism to transmit DTMF digits. There are 3 supported
Send DTMF
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.
DTMF Payload Type
Configures the payload type for DTMF using RFC2833. The default value is
101.
7 different vocoder types are supported on the phone, including G.711 U-law
Preferred Vocoder
(PCMU), G.711 A-law (PCMA), G.729A/B, G.722 (wide band) and G72-32.
Users can configure vocoders in a preference list that is included with the
same preference order in SDP message.
Use First Matching
When set to "Yes", the device will use the first matching vocoder in the
Vocoder in 200OK SDP
received 200OK SDP as the codec. The default setting is "No".
SRTP Mode
Enables the SRTP mode based on your selection. The default setting is
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"Disabled".
Symmetric RTP
Defines whether symmetric RTP is supported or not. The default setting is
"No".
Controls the silence suppression/VAD feature of the audio codec G.729. If set
Silence Suppression
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".
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the "ptime" value for the SDP will
Voice Frames Per TX
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.
G.726-32 Packing Mode
Jitter Buffer Type
Jitter Buffer Length
Selects "ITU" or "IETF" for G726-32 packing mode.
Selects either Fixed or Adaptive based on network conditions. The default
setting is "Adaptive".
Selects Low, Medium, or High based on network conditions. The default
setting is "Medium".
Account x -> Call Settings
Early Dial
Dial Plan Prefix
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".
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:
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
Dial Plan
b) xx. - only 2 digit numbers
c) ^ - exclude
d) [3-5] - any digit of 3, 4, or 5
e) [147] - any digit of 1, 4, or 7
f)
<2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
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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.
Delayed Call Forward
Defines the timeout (in seconds) before the call is forwarded on no answer.
Wait Time
The default value is 20 seconds.
When enabled, Do No Disturb, Call Forward and other call features will be
Enable Call Features
supported locally provided ITSP support those features. The default setting is
"Yes". If set to "No", ForwardAll softkey will be hidden for Account 1.
Configures Call Log setting on the phone. You can log all calls, only log
Call Log
incoming/outgoing calls or disable call log. The default setting is "Log All
Calls".
Account Ring Tone
Allows users to configure the ringtone for the account. Users can choose from
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different ringtones from the dropdown menu.
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;
[345]xx: 3-digit number with the leading digit of 3, 4 or 5;
Match Incoming Caller ID
[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:
Alert-Info: ; info=priority
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.
Ring Timeout
Send Anonymous
Anonymous Call
Rejection
Auto Answer
Allow Auto Answer by
Call-Info
Defines the timeout (in seconds) for the rings on no answer. The default setting
is 60 seconds.
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
answer incoming calls after a short reminding beep, based on the SIP info
header sent from the server/proxy. The default setting is "No".
Custom Call-Info for Auto
Used in addition to match the contents of the info parameter in the Call-Info
Answer
header for auto answer.
Refer-To Use Target
If set to "Yes", the "Refer-To" header uses the transferred target's Contact
Contact
header information for attended transfer. The default setting is "No".
Transfer on Conference
Defines whether or not the call is transferred to the other party if the initiator of
Hangup
the conference hangs up. The default setting is "No".
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No Key Entry Timeout (s)
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.
Allows users to configure the "#" key as the "Send" key. If set to "Yes", the "#"
Use # as Dial Key
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.
DND Call Feature On
Configures DND feature code to turn on DND.
DND Call Feature Off
Configures DND feature code to turn off DND.
SETTINGS PAGE DEFINITIONS
Table 16: Settings Page Definitions
Settings -> General Settings
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
Local RTP Port
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.
When set to "Yes", this parameter will force random generation of both
Use Random Port
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).
Specifies how often the phone sends a blank UDP packet to the SIP
Keep-alive Interval
server in order to keep the "ping hole" on the NAT router to open. The
default setting is 20 seconds.
Use NAT IP
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.
The IP address or Domain name of the STUN server. STUN resolution
STUN Server
results are displayed in the STATUS page of the Web GUI. Only
non-symmetric NAT routers work with STUN.
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,
Public Mode
users will need enter SIP User ID and Password on the LCD to login and
use the phone.
Note:
When the phone is in public mode login screen, press CONF button will
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have the IP address of the phone displayed.
Settings -> Call Features
Configures a User ID/extension to dial automatically when the phone is
Off-hook Auto Dial
off hook. The phone will use the first account to dial out. The default
setting is "No".
Off-hook Timeout
Disable Call Waiting
Disable Call Waiting Tone
Disable Direct IP Call
If configured, when the phone is on hook, it will go off hook after the
timeout (in seconds). The default value is 30 seconds.
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".
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
Use Quick IP Call mode
<=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".
Disable Conference
Disable in-call DTMF Display
Disables the Conference function. The default setting is "No".
When it's set to "Yes", the DTMF digits entered during the call will not
display. The default setting is "No".
Enable sending DTMF via
Enables Multi Purpose Key to send DTMF during the call. The default
Speed Dial
setting is "No".
Disable Transfer
Disables the Transfer function. The default setting is "No".
In-call dial number on pressing
Configures the number for the phone to dial as DTMF during the call
transfer key
using TRAN button.
Auto-Attended Transfer
If set to "Yes", the phone will use attended transfer by default. The default
setting is "No".
Do Not Escape # as %23 in
Specifies whether to replace # by %23 or not for some special situations.
SIP URI
The default setting is "No".
Click-To-Dial Feature
Enables Click-To-Dial feature. The default setting is "Disabled".
Call History Flash Writing:
Defines the interval (in seconds) to save the call history to phone's flash.
Write Timeout
The default value is 300 seconds.
Call History Flash Writing:
Defines the number of unsaved logs before written to phone's flash. The
Max Unsaved Log
default value is 200 entries.
Settings -> Ring Tone
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Configures ring or tone frequencies based on parameters from local
Call Progresses Tones:
System Ring Tone
Dial Tone
Message Waiting
Ring Back Tone
Call-Waiting Tone
Busy Tone
Reorder Tone
telecom. The default value is North American standard. Frequencies
should be configured with known values to avoid uncomfortable high
pitch sounds.
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
(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.
Call Waiting Tone Gain
Configures the call waiting tone gain to adjust call waiting tone volume.
The default setting is "Low".
Settings -> Audio Control
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 make calls by using headset. Headset icon will
display on 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
between Headset and Speaker.
Headset Type
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Selects normal RJ9 headset or Plantronics EHS headset for the headset
type.
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Configures to enable or disable the speaker to ring when headset is used
Always Ring Speaker
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".
Headset TX gain
Headset RX gain
Handset TX gain
Configures the transmission gain of the headset. The default value is
0dB.
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 -> LCD Display
Backlight Brightness: Active
Backlight Brightness: Idle
Disable Missed Call Backlight
Configures the LCD brightness when the phone is active. Valid range is
10 to 100 where 100 is the brightest. Default value is 70.
Configures the LCD brightness when the phone is idle. Valid range is 10
to 100 where 0 is off and 100 is the brightest. Default value is 30.
Configures the backlight to be on or off when there is missed call.
Settings -> Date and Time
NTP Server
Allow DHCP Option 42
Override NTP Server
Time Zone
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".
Configures the date/time used on the phone according to the specified
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
MTZ+6MDT+5
This indicates a time zone with 6 hours offset with 1 hour ahead (when
Self-Defined Time Zone
daylight saving) which is 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,
rd
3 Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon,
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Tues, ... ,Sat)
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
Configures the date display format on the LCD. The following formats are
supported:
Date Display Format
Time Display Format
yyyy-mm-dd: 2012-07-02
mm-dd-yyyy: 07-02-2012
dd-mm-yyyy: 02-07-2012
dddd, MMMM dd: Friday, October 12
MMMM dd, dddd: October 12, Friday
Configures the time display in 12-hour or 24-hour format on the LCD. The
default setting is in 12-hour format.
Settings -> Web Service
Configures to enable or disable weather update on the phone. The
Enable Weather Update
default setting is "Yes". If set to "No", the weather information screen will
not show.
Configures weather city code for the phone to look up the weather
information. The default setting is "Automatic" and the weather
City Code
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.
Update Interval
Degree Unit
Specifies the weather update interval (in minutes). The default value is
15 minutes.
Specifies the degree unit for the weather information to display on the
phone.
Configures to enable or disable currency update on the phone. The
Enable Currency Update
default setting is "Yes". If set to "No", the currency information screen will
not show.
Currency Code
Configures currency code for the phone to look up and display the
currency information.
Settings -> XML Applications
Server Path
Softkey Label
Configures the server path to download the idle screen XML file. This
field could be IP address or URL, with up to 256 characters.
Specifies the soft key name displayed on the idle screen for the users to
enter XML application.
Settings -> Programmable Keys
Line Key X
FIRMWARE VERSION 1.0.1.14
Assigns a function to the corresponding line key. The key mode options
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are:
Line
Regular line key to open up a line and switch line. The Value field
can be left blank.
Shared Line
Share line for Shared Line Appearance feature. Select the Account
registered as Shared line for the line key. The Value field can be left
blank.
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
the Value field to be dialed.
Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension
number in the Value field to be monitored.
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 Value field to be dialed during
the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value
field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Value 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 Value field to be
transferred (blind transfer) during the call.
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Call Park
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Select Account, and enter the call park extension in the Value field to
park/pick up the call.
Intercom
Select Account, and enter the extension number in the Value 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 Value field. LDAP search
does not support entering Non-ASCII characters
Assigns a function to the corresponding Multi Purpose Key. The key
mode options are:
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
the Value 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).
Presence Watcher
This option has to be supported by a presence server and it is tied to
Multi-Purpose Key(Only for
GXP2160)
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
/Extension Boards(Only for
collects the information from the phones and sends it out in one
GXP2140)
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 Value field to be dialed during
the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value
field.
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Call Return
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The last answered calls can be dialed out by using Call Return. The
Value 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 Value field to be
transferred (blind transfer) during the call.
Call Park
Select Account, and enter the call park extension in the Value field to
park/pick up the call.
Intercom
Select Account, and enter the extension number in the Value 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 Value field.
NETWORK PAGE DEFINITIONS
Table 17: Network Page Definitions
Network -> Basic Settings
Internet Protocol
Selects Prefer IPv4 or Prefer IPv6.
Allows users to configure the appropriate network settings on the phone to
IPv4 Address Type
obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE". By
default, it is set to "DHCP".
DHCP Host name
Specifies the name of the client. This field is optional but may be required by
(Option 12)
some Internet Service Providers.
DHCP Vendor Class ID
(Option 60)
Used by clients and servers to exchange vendor class ID.
PPPoE Account ID
Enter the PPPoE account ID.
PPPoE Password
Enter the PPPoE Password.
PPPoE Service Name
Enter the PPPoE Service Name.
IPv4 Address
Enter the IP address when static IP is used.
Subnet Mask
Enter the Subnet Mask when static IP is used for IPv4.
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Gateway
Enter the Default Gateway when static IP is used for IPv4.
DNS Server 1
Enter the DNS Server 1 when static IP is used for IPv4.
DNS Server 2
Enter the DNS Server 2 when static IP is used for IPv4.
Preferred DNS Server
Enter the Preferred DNS Server for IPv4.
Allows users to configure the appropriate network settings on the phone to
IPv6 Address Type
obtain IPv6 address. Users could select "Auto-configured" or "Statically
configured" for the IPv6 address type.
Static IPv6 Address
IPv6 Prefix Length
IPv6 Prefix
Enter the static IPv6 address when Full Static is used in "Statically configured"
IPv6 address type.
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.
DNS Server 2
Enter the DNS Server 2 for IPv6.
Preferred DNS server
Enter the Preferred DNS Server for IPv6.
Network -> Advanced Settings
Allows the user to enable/disable 802.1X mode on the phone. The default
802.1X mode
value is disabled. To enable 802.1X mode, this field should be set to
EAP-MD5.
802.1X Identity
Enter the Identity for the 802.1X mode.
MD5 Password
Enter the MD5 Password for the 802.1X mode.
802.1X CA Certificate
802.1X Client Certificate
HTTP Proxy
HTTPS Proxy
Layer 3 QoS
Layer 2 QoS
802.1Q/VLAN Tag
Layer 2 QoS 802.1p
Priority Value
Upload 802.1X CA certificate to the phone; or delete existed 802.1X CA
certificate from the phone.
Upload 802.1X Client certificate to the phone; or delete existed 802.1X Client
certificate from the phone.
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.
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.
Defines the Layer 3 QoS parameter. This value is used for IP Precedence,
Diff-Serv or MPLS. The default value is 12.
Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0.
Assigns the priority value of the Layer2 QoS packets. The default value is 0.
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Configures the PC port mode. When set to "Mirrored", the traffic in the LAN
PC Port Mode
port will go through PC port as well and packets can be captured by
connecting a PC to the PC port. The default setting is "Enable".
Enable LLDP
Control the LLDP(Link Layer Discovery Protocol) service.
MAINTENANCE PAGE DEFINITIONS
Table 18: Maintenance Page Definitions
Maintenance -> Web Access
End User Password
Set new password for web GUI access as User. This field is case sensitive.
Confirm Password
Enter the new User password again to confirm.
Current Admin Password
The current admin password is required for setting a new admin password.
Admin Password
Set new password for web GUI access as Admin. This field is case sensitive.
Confirm Password
Enter the new Admin password again to confirm.
Maintenance -> Upgrade and Provisioning
Firmware Upgrade and
Provisioning
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.
XML Config File
The password for encrypting the XML configuration file using OpenSSL. This
Password
is required for the phone to decrypt the encrypted XML configuration file.
HTTP/HTTPS User
Name
The user name for the HTTP/HTTPS server.
HTTP/HTTPS Password
The password for the HTTP/HTTPS server.
Upgrade Via
Allows users to choose the firmware upgrade method: TFTP, HTTP or HTTPS.
Firmware Server Path
Config Server Path
Defines the server path for the firmware server. It could be different from the
configuration server for provisioning.
Defines the server path for provisioning. It could be different from the firmware
server for upgrading.
This field enables user to store different versions of firmware files in one single
Firmware File Prefix
directory on the firmware server. If configured, only the firmware file with the
matching prefix will be downloaded.
This field enables user to store different versions of firmware files in one single
Firmware File Postfix
directory on the firmware server. If configured, only the firmware file with the
matching postfix will be downloaded.
Config File Prefix
This field enables user to store different configuration files in one single
directory on the configuration server. If configured, only the configuration file
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with the matching prefix will be downloaded.
This field enables user to store different configuration files in one single
Config File Postfix
directory on the configuration server. If configured, only the configuration file
with the matching postfix will be downloaded.
Allow DHCP Option 43
and Option 66 Override
Server
If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be
redirected. The default setting is "Yes".
Allow DHCP Option 120
Enables DHCP Option 120 from local server to override the SIP Server on the
to override SIP Server
phone. The default setting is "No".
3CX Auto Provision
Phone will multicast SUBSCRIBE for provision if this feature is enabled.
Automatic Upgrade
Enables automatic upgrade and provisioning. The default setting is "No".
Hour of the Day (0-23)
Day of the Week (0-6)
Authenticate Conf File
Defines the hour of the day to check the HTTP/TFTP server for firmware
upgrades or configuration files changes. The default value is 1.
Defines the day of the week to check HTTP/TFTP server for firmware
upgrades or configuration files changes. The default value is 1.
Authenticates configuration file before acceptance. The default setting is "No".
Maintenance -> Syslog
Syslog Server
The URL or IP address of the syslog server for the phone to send syslog to.
Selects the level of logging for syslog. The default setting is "None". There are
4 levels: DEBUG, INFO, WARNING AND ERROR.
Syslog messages are sent based on the following events:
Syslog Level
Send SIP Log
Auto Recover From
Abnormal
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. The
default setting is "No".
If set to “Yes”, the phone will automatically recover when running abnormal.
Maintenance -> Language
Display Language
Selects display language on the phone.
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Maintenance -> TR-069
Enable TR-069
Enables TR-069. The default setting is "No".
ACS URL
URL for TR-069 Auto Configuration Servers (ACS).
TR-069 Username
ACS username for TR-069.
TR-069 Password
ACS password for TR-069.
Periodic Inform Enable
Periodic Inform Interval
Connection Request
Username
Connection Request
Password
Enables periodic inform. If set to "Yes", device will send inform packets to the
ACS. The default setting is "No".
Sets up the periodic inform interval to send the inform packets to the ACS.
The user name for the ACS to connect to the phone.
The password for the ACS to connect to the phone.
Connection Request Port
The port for the ACS to connect to the phone.
CPE SSL Certificate
The Cert File for the phone to connect to the ACS via SSL.
CPE SSL Private Key
The Cert Key for the phone to connect to the ACS via SSL.
Maintenance -> Security
Configures the access control for the users to configure from keypad Menu.
There are three different options:
Configuration via Keypad
Menu
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.
Constraint Mode. CONFIG, FACTORY FUNCTIONS and 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.
Enable STAR key
Keypad locking
The default setting is "Yes".
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.
Password to lock/unlock
Configures the password to lock/unlock the keypad.
SSL TLS Certificate
SSL Certificate used for SIP Transport in TLS/TCP.
SSL TLS Private Key
SSL Private key used for SIP Transport in TLS/TCP.
SSL TLS Private Key
SSL Private key password used for SIP Transport in TLS/TCP.
Password
Download Device
Click to download the device configuration file in .txt format.
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Configuration
Web Access Mode
Sets the protocol for web interface.
Disable SSH
Disables SSH access.
Specifies the time in minutes that the web or LCD login interface will be locked
Web/Keypad/Restrict
out to user after five login failures. This lockout time is used for web login,
mode Lockout Duration
STAR keypad unlock and LCD restrict mode admin login. Range is 0-60
minutes.
PHONEBOOK PAGE DEFINITIONS
Table 19: Phonebook Page Definitions
Phonebook -> Contacts
Add Contact
Edit Contact
Specifies Contact’s First Name, Last Name, Phone Number, Accounts and
Groups to add one new contact in phonebook.
Edits selected contact.
Phonebook -> Group Management
Add Group
Specifies Group’s name to add new group.
Edit Group
Edits selected group.
Phonebook -> Phonebook Management
Enable Phonebook XML
Download
Configures to enable phonebook XML download. Users could select
HTTP/HTTPS/TFTP to download the phonebook file. The default setting is
"Disabled".
Phonebook XML Server
Configures the server path to download the phonebook XML. This field could
Path
be IP address or URL, with up to 256 characters.
Phonebook Download
Interval
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
If set to "Yes", when XML phonebook is downloaded, the entries added
Entries on Download
manually will be automatically removed. The default setting is "Yes".
Download XML
Phonebook
Upload XML Phonebook
Phonebook Key Function
Click on "Download" to download the XML phonebook file to local PC.
Click on "Upload" to upload local XML phonebook file to the phone.
Control the behavior of phonebook key. There are four options: Default, LDAP
Search, Local Phonebook, Local Group.
Phonebook -> LDAP
Server Address
Configures the IP address or DNS name of the LDAP server.
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Port
Configures the LDAP server port.
Configures the LDAP search base. This is the location in the directory where
the search is requested to begin.
Base
Example:
dc=grandstream, dc=com
ou=Boston, dc=grandstream, dc=com
User Name
Password
Configures the bind "Username" for querying LDAP servers. Some LDAP
servers allow anonymous binds in which case the setting can be left blank.
Configures the bind "Password" for querying LDAP servers. The field can be
left blank if the LDAP server allows anonymous binds.
Configures the filter used for number lookups.
Examples:
(|(telephoneNumber=%)(Mobile=%) returns all records which has the
LDAP Number Filter
"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.
LDAP Version
Selects the protocol version for the phone to send the bind requests. The
default setting is "Version 3".
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
LDAP Name Attributes
name attributes.
Example:
gn
cn sn description
Specifies the "number" attributes of each record which are returned in the
LDAP Number Attributes
LDAP search result. This field allows the users to configure multiple space
separated number attributes.
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Example:
telephoneNumber
telephoneNumber Mobile
Configures the entry information to be shown on phone's LCD. Up to 3 fields
LDAP Display Name
can be displayed.
Example:
%cn %sn %telephoneNumber
Max. Hits
Search Timeout
Sort Results
LDAP Lookup
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.
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.
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.
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.
Lookup Display Name
Example:
gn
cn sn description
Phonebook -> Broadsoft
Name
Specifies phonebook name..
Server
Configures the Broadsoft phonebook server.
Port
Configures the Broadsoft phonebook port.
Username
Specifies the username for phonebook access.
Password
Specifies the password for phonebook access.
Selects the Broadsoft phonebook type from Group Directory, Personal
Directory, Enterprise Directory, Missed Call Log, Placed Call Log and
Type
Received Call Log. If the type is “Directory”, a new entry will be added under
“Phonebook” -> “Broadsoft Phonebook”. If the type is “Call Log”, the new entry
will be added under “Call History” -> “Broadsoft Call Log”.
NAT SETTINGS
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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 Settings->General Settings, 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
It is under Settings->General Settings. 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".
NAT Traversal
It is under Accounts X->Network Settings. 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 GXP2130/GXP2140/GXP2160 to display weather information for the preferred city, users
could go to web GUI->Settings->Web Service page and enter the city code in the following options:
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Figure 9: Web Service
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:
In a web browser, go to www.weather.com;
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,
http://www.weather.com/weather/today/Boston+MA+USMA0046:1:US
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.
PUBLIC MODE
The GXP2130/GXP2140/GXP2160 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:
Fill up the SIP server address for account 1 first;
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Under Web GUI->Settings->General Settings, set "Public Mode" option to "Yes". Click "Save and
Apply" and reboot the phone;
When the 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 CONF 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 GXP2130/GXP2140/GXP2160 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, edit
selected contact, or dial the contact/number.
Before using the Click-To-Dial feature, make sure the option "Click-To-Dial Feature" under web
GUI->Settings->Call Features is turned on. By default it's disabled and the dialing icon in web GUI is in
grey
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 11 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/123
In the above link, replace the fields with
ip_address:
Phone's IP Address.
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.
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password=admin/123:
The admin login password or user login password of phone's Web GUI.
Click to dial from
Add contacts.
available lines.
Click to call this
Edit contact.
contact from the phone.
Figure 10: Web GUI - Phonebook->Contacts
Figure 11: Click-to-Dial
SAVING THE CONFIGURATION CHANGES
After users makes changes to the configuration, press the "Save" button will save but not apply the
changes until the "Apply" button on the top of web GUI page is clicked. Or, users could directly press
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"Save and Apply" button. We recommend rebooting or powering cycle the phone after applying all the
changes.
REBOOTING FROM REMOTE LOCATIONS
Press the "Reboot" button on the top right corner of the web GUI page to reboot the phone remotely. The
web browser will then display a reboot message. Wait for about 1 minute to log in again.
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CONNECTING TO DEVICES
The GXP2130/GXP2140/GXP2160 supports EHS headset (Plantronics). GXP2140/2160 is also capable
to connect to USB and Bluetooth devices.
EHS HEADSET
The GXP2130/GXP2140/GXP2160 supports normal RJ11 headset and Plantronics EHS headset. To use
Plantronics EHS headset, go to the GXP2130/GXP2140/GXP2160 web GUI->Settings->Audio Control:
Headset Type. Select "Plantronics EHS" and reboot the phone to take effect.
1. Connect EHS Headset (Plantronics) to GXP2130/GXP2140/GXP2160. Insert headset connector into
the RJ11 headset port on the back of GXP2130/GXP2140/GXP2160;
2. To use headset mode, press headset button
icon
on the GXP2130/GXP2140/GXP2160. A headset
will show on the GXP2130/GXP2140/GXP2160 status bar.
BLUETOOTH
Bluetooth is a proprietary, open wireless technology standard for exchanging data over short distances
from fixed and mobile devices, creating personal area networks with high levels of security. On
GXP2140/GXP2160, users could use Bluetooth to transfer files, share contact information with other
Bluetooth portable devices, or use Bluetooth headset for making calls as well as listening to media (music
or other audio output) from the GXP2140/GXP2160.
To connect to a Bluetooth device, turn on GXP2140/GXP2160’s Bluetooth radio first. The first time when
using a new Bluetooth device with the GXP2140/GXP2160, "pair" the device with the phone so that both
devices know how to connect securely to each other. After that, users could simply connect to a paired
device. Turn off Bluetooth if it's not used.
Bluetooth related settings are under GXP2140/GXP2160's LCD Menu->System->Bluetooth.
TURNING BLUETOOTH ON OR OFF
1. Go to GXP2140/GXP2160's LCD menu->System->Bluetooth;
2. Highlight Power option by pressing Up/Down arrow key;
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3. When the Power option is highlighted, pressing menu round button to turn on/off Bluetooth.
PAIRING GXP2140/GXP2160 WITH A BLUETOOTH DEVICE
Before connecting GXP2140/GXP2160 with other Bluetooth device, users must pair them first. They will
stay paired afterwards unless they are unpaired.
1. Go to GXP2140/GXP2160's LCD menu->System->Bluetooth;
2. Enter Start Scan option;
3. The GXP2140/GXP2160 scans and displays the IDs of all available Bluetooth devices in range. If the
device you would like to pair with does not show up in the list, ensure that the device is turned on and
set to be discoverable. If need, user could press “Rescan” soft key to restart the scan;
4. The discovered device will show in the scan result list with device ID. If it's unpaired, it will show
"Unpaired" behind the device ID. Select the device ID, and press “Pair” soft key to pair them. Then a
“Pincode Input” window will prompt up. User may enter the Pin code, and press “OK” soft key to start
pairing.
CONNECTING TO A BLUETOOTH EARPHONE
After successfully paired, the GXP2140/GXP2160 will show "Paired" behind the Bluetooth device ID in the
scan list. Highlight the device ID and press “Connect” soft key to connect.
Once connected, the GXP2140/GXP2160 will show “Connected” behind the Bluetooth device ID in the
scan list.
BT ON AND BT OFF
After successfully connected, the GXP2140/GXP2160 will show “BT OFF” soft key on call screen.
Pressing the “BT OFF” soft key, it will change to “BT ON” and the Bluetooth will be turn off. When “BT
OFF” soft key appears, the Bluetooth is in use and user could answer the incoming call via Bluetooth.
UNPAIRING AND DISCONNECTING A BLUETOOTH DEVICE
In the Bluetooth settings, highlight the Bluetooth device ID and press “Disconnect”/”Unpair” soft key to
disconnect/unpair the Bluetooth device.
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UPGRADING AND PROVISIONING
The GXP2130/GXP2140/GXP2160 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/HTTPS; 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 LCD Keypad 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 System;
In the System options, select Upgrade;
Enter the firmware server path and select upgrade method. The server path could be in IP address
format or FQDN format;
Select Start Provision option, and press the "Select" soft key.
A warning window will be prompt for provision confirmation. Press “YES” soft key to start
upgrading/provisioning immediately.
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 Maintenance->Upgrade and Provisioning 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 "Save and Apply" button. Then "Reboot" or power cycle
the phone to update the new firmware.
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When upgrading starts, the screen will show upgrading progress. When done you will see the phone
restart again. Please do not interrupt or power cycle the phone when the upgrading process is on.
Firmware upgrading takes 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:
http://www.grandstream.com/support/firmware
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 :
http://www.solarwinds.com/products/freetools/free_tftp_server.aspx
http://tftpd32.jounin.net/.
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.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use
Microsoft IIS web server.
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
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".
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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 "New Password" in the Web GUI->Maintenance->Web Access
page->Admin Password. For a detailed parameter list, please refer to the corresponding firmware release
configuration template.
When the GXP2130/GXP2140/GXP2160 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:
http://www.grandstream.com/general/gs_provisioning_guide_public.pdf
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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 "System" and enter;
Select "Operations - Factory Reset";
A warning window will pop out to make sure a reset is requested and confirmed;
Press the "Yes" soft key to confirm and the phone will reboot. To cancel the Reset, press “No” soft key
instead.
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EXPERIENCING THE GXP2130/GXP2140/GXP2160
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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FCC Caution:
Any Changes or modifications not expressly approved by the party responsible for compliance could void
the user's authority to operate the equipment.
This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:
(1) This device may not cause harmful interference, and (2) this device must accept any interference
received, including interference that may cause undesired operation.
Note: This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against
harmful interference in a residential installation. This equipment generates, uses and can radiate radio
frequency energy and, if not installed and used in accordance with the instructions, may cause harmful
interference to radio communications. However, there is no guarantee that interference will not occur in a
particular installation. If this equipment does cause harmful interference to radio or television reception,
which can be determined by turning the equipment off and on, the user is encouraged to try to correct the
interference by one or more of the following measures:
—Reorient or relocate the receiving antenna.
—Increase the separation between the equipment and receiver.
—Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
—Consult the dealer or an experienced radio/TV technician for help.
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