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GXP21xx Series Administration Guide Documentation Center

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12 views122 pages

GXP21xx Series Administration Guide Documentation Center

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dexterargento

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Grandstream Networks, Inc.

GXP21xx Series - Administration Guide

The GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 is a state-of-the-art enterprise-grade IP phone. GXP2170 features 12
dual-color line keys (can be digitally programmed as up to 48 provisionable BLF/fast-dial keys). GXP2135 features 8 dual-color
line keys (can be programmed as up to 32 provisionable BLF/fast-dial keys). GXP2140/GXP2160/GXP2170 features a 4.3-inch
TFT Color LCD, 5 programmable context-sensitive Softkeys, dual Gigabit network ports, integrated PoE and Bluetooth, 5-way
conference, and Electronic Hook Switch (EHS). GXP2135 supports 2.8-inch TFT Color LCD, 4 programmable context-sensitive
Softkeys, 4-way voice conference, and EHS with Plantronics headsets. GXP2130 supports 2.8-inch TFT Color LCD, 4
programmable context-sensitive Softkeys, 4-way voice conference, and EHS with Plantronics headsets. Also, this series can
support up to 3 lines for GXP2130, 4 lines for GXP2140, 6 lines for GXP2160, 12 lines for GXP2170, and 8 lines for GXP2135.
The GXP2130 / GXP2140 / GXP2160 / GXP2170/GXP2135 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/GXP2170/GXP2135 supports presence and Busy Lamp Field (BLF) in the to cellphones (supporting
BluetoothMulti-Purpose Keys as well. The GXP2140/GXP2170 is expandable with one to 4 expansion modules. The GXP2130/
GXP2140/GXP2160/GXP2170/GXP2135 is the perfect choice for enterprise users looking for high quality, feature-rich multi-line
executive IP phone with advanced functionalities and performance.

The GXP21xx Series – Administration Guide is intended for administrators.

PRODUCT OVERVIEW

Feature Highlights

The following tables contain the major features of GXP21xx.

3 lines
2.8 inch (320×240) TFT color LCD
4 programmable soft keys
GXP2130
Bluetooth (GXP2130v2 only)
8 programmable Multi-Purpose Keys
4-way conference

4 lines
4.3 inch (480×272) TFT color LCD
5 programmable soft keys
GXP2140
Bluetooth
5-way conference
Expansion board

6 lines
4.3 inch (480×272) TFT color LCD
5 programmable soft keys
GXP2160
Bluetooth
5-way conference24 programmable Multi-Purpose Keys

12 dual-color line keys that can be digitally programmed as up to 48 provisionable

BLF/fast-dial keys
4.3 inch (480×272) TFT color LCD
GXP2170 5 programmable soft keys
Bluetooth
5-way conference
Expansion board
8 dual-color line keys that can be digitally programmed as up to 32 provisionable
BLF/fast-dial keys
2.8 inch (320×240) TFT color LCD
GXP2135
4 programmable soft keys
Bluetooth
4-way conference

Table 1: GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 Features in a Glance

Features GXP2130 GXP2140 GXP2160 GXP2170 GXP2135

LCD Display 320×240 480 x 272 480 x 272 480 x 272 320×240

LCD Backlight Yes Yes Yes Yes Yes

Number of Lines 3 4 6 12 8

Programmable Hard Keys 8 N/A 24 48 32

Programmable Softkeys 4 5 5 5 4

Extension Module N/A Yes, up to 4 EXT Boards N/A Yes, up to 4 EXT Boards N/A

Table 2: GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 Comparison Guide

GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 Technical Specifications

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs,
telephony features, languages, and upgrade/provisioning settings for the GXP21xx series.

Protocols
SIP RFC3261, TCP/IP/UDP, RTP/RTCP/RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP,
/Standar
PPPoE, SSH, TFTP, FTP/FTPS, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6
ds

Network
Interface Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with integrated PoE
s

Graphic
2.8 inch (320×240) TFT color LCD
Display

Bluetoot
Yes, Bluetooth (GXP2130v2 only, GXP2130v1 does not support Bluetooth)
h

3 line keys with up to 3 SIP accounts, 8 speed-dial/BLF extension keys with dual-color LED, 4 programmable
Feature contexts sensitive Softkeys, 5 navigation/menu keys, 11 dedicated function keys for MESSAGE (with LED
Keys indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE,
VOLUME+, VOLUME-

Voice Support for G.729A/B, G723.1, G.711µ/a-law, G.726, G.722 (wide-band), ILBC, OPUS and in-band and out-of-
Codec band DTMF (in audio, RFC2833, SIP INFO)

Auxiliary
RJ9 headset jack (allowing EHS with Plantronics headsets)
Ports
Hold, transfer, forward, 4-way conference, call park, call pickup, shared-call-appearance (SCA), bridged-line-
Telephon
appearance (BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 500
y
records), customization of the screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot
Features
desking, personalized music ringtones and music on hold, server redundancy and fail-over

HD audio Yes, both on the handset and full-duplex hands-free speakerphone

Base
Yes, allow 2 angle positions
Stand

Wall
Mountab Yes
le

QoS Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

User and administrator level passwords, MD5 and MD5-sess based authentication, AES-based secure
Security
configuration file, SRTP, TLS, 802.1x media access control

LCD Language: ‫( العربية‬Arabic) Català (Catalan) Czech Deutsch (German) English Español (Spanish) Français
(French) ‫( עברית‬Hebrew) Hrvatski (Croatian) Magyar (Hungarian) Italiano (Italian) 日本語 (Japanese) 한국어
(Korean) Latviešu valoda (Latvian) Nederlands (Dutch) Polski (Polish) Português (Portuguese) Русский (Russian)
Multi- Slovenščina (Slovenian) Svenska (Swedish) Türkçe (Turkish) Ukrainian 正體中文 (Traditional Chinese) 简体中文
language (Simplified Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) 繁體中文 (Traditional Chinese) ‫( العربية‬Arabic) Czech
Deutsch (German) Español (Spanish) Français (French) ‫( עברית‬Hebrew) Hrvatski (Croatian) Magyar (Hungarian)
Italiano (Italian) 日本語 (Japanese) 한국어 (Korean) Nederlands (Dutch) Polski (Polish) Português (Portuguese)
Русский (Russian) Slovenščina (Slovenian) Türkçe (Turkish)

Upgrade
Firmware upgrade via TFTP/FTP/FTPSHTTP/HTTPS, mass provisioning using TR-069 or encrypted XML
/Provisio
configuration file
ning

Power &
Green Universal power adapter included: Input:100-240 VAC; Output: +12VDC, 0.5A;
Energy
Efficienc Integrated Power-over-Ethernet (802.3af)
y

Dimension : 193mm (W) x 211mm (L) x 84.5 mm (H)

Physical Unit weight: 0.78kg

Package weight: 1.3kg

Temperat
ure and 32-104℉ / 0～40℃, 10-90% (non- condensing)
Humidity

Package
GXP2130 phone, handset with cord, base stand, universal power supply, network cable, Quick Start Guide
Content

Complia FCC Part15 Class B, EN55022 ClassB, EN61000-3-2, EN61000-3-3, EN55024, EN60950-1, AS/NZS CISPR22 Class
nce B

Table 3: GXP2130 Technical Specifications

Protocol
SIP RFC3261, TCP/IP/UDP, RTP/RTCP/RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP,
s/Standa
PPPoE, SSH, TFTP, FTP/FTPS NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6
rds

Network
Interface Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with integrated PoE
s

Graphic
4.3 inch (480×272) TFT color LCD
Display

Bluetoot
h Bluetooth supported

4 line keys with up to 4 SIP accounts, 5 programmable contexts sensitive Softkeys, 5 navigation/menu keys, and
Feature
11 dedicated function keys for MESSAGE (with LED indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD,
Keys
HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME+, VOLUME-

Voice Support for G.723.1, G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), OPUS, iLBC and in-band and out-of-
Codec band DTMF (in audio, RFC2833, SIP INFO)

Auxiliary
RJ9 headset jack (allowing EHS with Plantronics headsets), USB, extension module port
Ports

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call-appearance (SCA)/bridged-line-
Telepho
appearance (BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 500
ny
records), customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot desking,
Features
personalized music ringtones and music on hold, server redundancy and fail-over

HD
Yes, both on handset and full-duplex handsfree speakerphone
audio

Extensio
Yes, can power up to 4 GXP2200EXT modules which features a 128×384 graphic LCD, 20 quick-dial/BLF keys
n
which dual-color LED, 2 navigation keys, and less than 1.2W power consumption per unit.
Module

Base
Yes, allow 2 angle positions
Stand

Wall
Mounta Yes
ble

QoS Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

User and administrator level passwords, MD5 and MD5-sess based authentication, AES based secure
Security
configuration file, SRTP, TLS, 802.1x media access control
LCD Language: ‫( العربية‬Arabic) Català (Catalan) Czech Deutsch (German) English Español (Spanish) Français
(French) ‫( עברית‬Hebrew) Hrvatski (Croatian) Magyar (Hungarian) Italiano (Italian) 日本語 (Japanese) 한국어
(Korean) Latviešu valoda (Latvian) Nederlands (Dutch) Polski (Polish) Português (Portuguese) Русский (Russian)
Multi-
Slovenščina (Slovenian) Svenska (Swedish) Türkçe (Turkish) Ukrainian 正體中文 (Traditional Chinese) 简体中文
languag
(Simplified Chinese)
e
WebUI Language: English 简体中文 (Simplified Chinese) 繁體中文 (Traditional Chinese) ‫( العربية‬Arabic) Czech
Deutsch (German) Español (Spanish) Français (French) ‫( עברית‬Hebrew) Hrvatski (Croatian) Magyar (Hungarian)
Italiano (Italian) 日本語 (Japanese) 한국어 (Korean) Nederlands (Dutch) Polski (Polish) Português (Portuguese)
Русский (Russian) Slovenščina (Slovenian) Türkçe (Turkish)

Upgrade
Firmware upgrade via TFTP/FTP/FTPS/HTTP/HTTPS, mass provisioning using TR-069 or encrypted XML
/Provisio
configuration file
ning

Power & Universal power adapter included: Input:100-240 VAC; Output: +12VDC, 1.0A;
Green
Energy Integrated Power-over-Ethernet (802.3af)
Efficienc
y Max power consumption 6W (without GXP2200EXT), 10W (with 4 cascaded GXP2200EXTs)

Physical Dimension: 222mm (W) x 210mm (L) x 93mm (H); Unit weight: 0.98kg; Package weight: 1.55kg

Tempera
ture and
32-104℉ / 0～40℃, 10-90% (non- condensing)
Humidit
y

Package
GXP2140 phone, handset with cord, base stand, universal power supply, network cable, Quick Start Guide
Content

Complia FCC Part15 Class B, EN55022 ClassB, EN61000-3-2, EN61000-3-3, EN55024, EN60950-1, AS/NZS CISPR22 Class
nce B

Table 4: GXP2140 Technical Specifications

Network
Interface Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with integrated PoE
s

Graphic
4.3 inch (480×272) TFT color LCD
Display

Bluetoot
Bluetooth supported
h

6 line keys with up to 6 SIP accounts, 24 speed-dial/BLF extension keys with dual-color LED, 5 programmable
Feature contexts sensitive Softkeys, 5 navigation/menu keys, 11 dedicated function keys for: MESSAGE (with LED
Keys indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE,
VOLUME+, VOLUME-

Voice Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), iLBC(pending) and in-band and out-of-band
Codec DTMF (in audio, RFC2833, SIP INFO)
Auxiliary
RJ9 headset jack (allowing EHS with Plantronics headsets), USB
Ports

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call-appearance (SCA)/bridged-line-
Telephon
appearance (BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 500
y
records), customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, Hot Desking,
Features
personalized music ringtones and music on hold, server redundancy and fail-over

HD audio Yes, both on handset and full-duplex handsfree speakerphone

Base
Yes, allow 2 angle positions
Stand

Wall
Mountab Yes
le

QoS Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

User and administrator level passwords, MD5 & MD5-sess based authentication, AES based secure
Security
configuration file, SRTP, TLS, 802.1x media access control

Upgrade
Firmware upgrade via TFTP/FTP/FTPS/HTTP/HTTPS, mass provisioning using TR-069 or encrypted XML
/Provisio
configuration file
ning

Power & Universal power adapter included: Input:100-240V; Output: +12V, 1.0A;
Green
Energy Integrated Power-over-Ethernet (802.3af)
Efficienc
y Max power consumption: 6W

Dimension : 222mm (W) x 210mm (L) x 93mm (H).

Physical
Unit weight: 0.98kg; Package weight: 1.62kg

Temperat
ure and 32-104℉ / 0～40℃, 10-90% (non- condensing)
Humidity

Package
GXP2160 phone, handset with cord, base stand, universal power supply, network cable, Quick Start Guide
Content

Complia FCC Part15 Class B, EN55022 ClassB, EN61000-3-2, EN61000-3-3, EN55024, EN60950-1, AS/NZS CISPR22 Class
nce B

Table 5: GXP2160 Technical Specifications

Network
Interface Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with integrated PoE
s

Graphic
4.3 inch (480×272) TFT color LCD
Display

Bluetoot
Bluetooth supported
h

12 line keys with up to 6 SIP accounts or 48 provisionable BLF/fast-dial keys, 5 programmable contexts
Feature sensitive Softkeys, 5 navigation/menu keys, 11 dedicated function keys for: MESSAGE (with LED indicator),
Keys PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME+,
VOLUME-

Voice Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), in-band and out-of-band DTMF (in audio,
Codec RFC2833, SIP INFO)

Auxiliary
RJ9 headset jack (allowing EHS with Plantronics headsets), USB, extension module port
Ports

HD audio Yes, both on handset and full-duplex handsfree speakerphone

Base
Stand /
Wall Yes, allow 2 angle positions
Mountab
le

QoS Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

User and administrator level passwords, MD5 and MD5-sess based authentication, AES based secure
Security
configuration file, SRTP, TLS, 802.1x media access control

Universal power adapter included: Input:100-240V; Output: +12V, 1.0A;

Power & Integrated Power-over-Ethernet (802.3af)

Green
Energy Max power consumption:
Efficienc
y 5.4W (without GXP2200EXT)

or 9.2W (with 4 cascaded GXP2200EXTs)

Dimension : 228mm (W) x 206mm (L) x 46.5mm (H).

Physical
Unit weight: 0.98kg; Package weight: 1.55kg

Temperat
ure and 0 ~ 40ºC (32 ~ 104ºF), 10 ~ 90% (non-condensing)
Humidity

Package
GXP2170 phone, handset with cord, base stand, universal power supply, network cable, Quick Start Guide
Content

Complia FCC Part 15 (CFR 47) Class B ; EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1, EN62479,
nce AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS ; UL 60950 (power adapter)

Table 6: GXP2170 Technical Specifications

Network
Interface Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with integrated PoE
s

Graphic
2.8 inch (320×240) TFT color LCD
Display

Bluetoot
Bluetooth supported
h

8 line keys with up to 4 SIP accounts or 32 provisionable BLF/fast-dial keys, 4 programmable contexts sensitive
Feature
Softkeys, 5 navigation/menu keys, 11 dedicated function keys for: MESSAGE (with LED indicator), PHONEBOOK,
Keys
TRANSFER, CONFERENCE, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME+, VOLUME-

Voice Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), in-band and out-of-band DTMF (in audio,
Codec RFC2833, SIP INFO)

Auxiliary
RJ9 headset jack (allowing EHS with Plantronics headsets)
Ports
Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call-appearance (SCA)/bridged-line-
Telephon
appearance (BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 500
y
records), customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, Hot Desking,
Features
personalized music ringtones and music on hold, server redundancy and fail-over

HD audio Yes, both on handset and full-duplex handsfree speakerphone

Base
Stand /
Wall Yes, allow 2 angle positions
Mountab
le

QoS Layer 2 (808.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

User and administrator level passwords, MD5 and MD5-sess based authentication, AES based secure
Security
configuration file, SRTP, TLS, 802.1x media access control

Upgrade
Firmware upgrade via TFTP/FTP/FTPS/HTTP/HTTPS, mass provisioning using TR-069 or encrypted XML
/Provisio
configuration file
ning

Power &
Green Universal power adapter included: Input:100-240VAC; Output: +12VDC, 0.5A;
Energy
Efficienc Integrated Power-over-Ethernet (802.3af) Max power consumption 3W
y

Dimension : 228mm (W) x 206mm (L) x 46.5mm (H)

Physical Unit weight: 0.98kg

Package weight: 1.55kg

Temperat
ure and 0 ~ 40ºC (32 ~ 104ºF), 10 ~ 90% (non-condensing)
Humidity

Package
GXP2135 phone, handset with cord, base stand, universal power supply, network cable, Quick Start Guide
Content

Complia FCC Part 15 (CFR 47) Class B ; EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1, EN62479,
nce AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS ; UL 60950 (power adapter)

Table 7: GXP2135 Technical Specifications

CONFIGURATION GUIDE
The GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 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” Softkey to enter the selected option.

Exit. Press “Exit” Softkey to exit to the previous menu.

Return to Home page.

In the Main menu, press Home Softkey to return home screen.

In sub menu, press and hold “Exit” Softkey until Exit Softkey changes to Home Softkey, then release the Softkey.

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, IP setting, MAC address and
software address.

The MENU options are listed in the following table.

Call Displays Local call logs:

Histo
ry All Calls/Answered Calls/Dialed Calls/Missed Calls/Transferred Calls

Displays account status, network status, software version number and Hardware

Account status.

Network status.

Statu Press to enter the sub menu for MAC address, IP setting information (DHCP/Static IP/PPPoE), Ipv4 address, Ipv6
s address, Subnet Mask, Gateway and DNS server.

System Information

Press to enter the sub menu for Hardware version, P/N number. Boot, Core, Base, Prog version and IP Geographic
Information.

Contacts sub menu includes the following options:

Local Phonebook

Local Group
Conta
cts 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.
Message sub menu include the following options :

Instant Message

Mess Displays received instant messages;

ages
Voice Mails

Displays voicemail message information in the format below: new messages/all messages (urgent messages/all
urgent messages).
Prefe Preference sub menu includes the following options:
rence
Do Not Disturb

Enables/disables Do Not Disturb on the phone.

Star Key Lock

Turns on/off keypad lock feature and configures keypad lock password. The default keypad lock password is null. If
user enabled Star Key lock without configuring password, user can unlock keypad by holding * key 4 seconds and
pressing “OK” button.

Sounds
Ring Tone

Configures different ring tones for incoming call.

Ring Volume

Adjusts ring volume by pressing left/right arrow key.

Appearance
Active LCD Brightness

Adjusts active LCD brightness by pressing left/right arrow key

Idle LCD Brightness

Adjusts idle LCD brightness by pressing left/right arrow key

Active LCD Timeout

Adjusts the minute of active backlight timeout.

Screensaver

Enables/Disables Screensaver

Screensaver Timeout

Configures the minutes of idle before the screensaver activates.

Valid range is 3 to 6.

Language and Input

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. By default, it is Auto.

Default Input Selection

Selects the Input mode from Multi-Tap and Shiftable. By default, it is Multi-Tap.

Multi-Tap: User may tap the same key multiple times to switch to the desired character.

Shiftable: After pressing the number button, user will see the IDs of the characters that matching to the button.
User can select the desired character by entering the corresponding ID on keypad.

Date Time
Allow DHCP Option 42 to override NTP server

Allow DHCP Option 2 to override Time Zone setting

Time Settings
It is used to configure date and time on the phone.

Search Mode

Specifies the phonebook search mode to QuickMatch or ExactMatch. By default, it is QuickMatch.

Phone sub menu includes the following options:

SIP

Phon Configures SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth ID, SIP Password, SIP Transport and Audio information
e 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.
Syste System sub menu includes the following options:
m
Network
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.

802.1X

Enables/Disables 802.1X mode; Configures 802.1x identity and MD5 password.

Layer 2 QoS

Configures 802.1Q/VLAN Tag and priority value. Select “Reset VLAN Config” to reset VLAN configuration.

Bluetooth Settings (GXP2130v2/GXP2140/GXP2160/GXP2170/GXP2135)

Bluetooth Status

Displays the status of Bluetooth

Bluetooth MAC

Displays the GXP phone’s MAC address

Power

Turns on/off the Bluetooth feature.

Handsfree Mode

Enables/Disables Handsfree mode

Bluetooth Name

Specifies GXP phone name when discovered by other Bluetooth devices.

Start Scan

Starts to scan other Bluetooth devices around the phone. If found, user could press “Pair” Softkey, and enter Pin
code to pair to other Bluetooth devices.

Upgrade
Firmware Server

Configures firmware server for upgrading the phone.

Config Server

Configures config server for provisioning the phone.

Upgrade Via

Specifies upgrade/provisioning via TFTP/FTP/FTPS/HTTP/HTTPS.

Start Provision

Starts Provision immediately.

Language Download
Auto Language Download

Language Download

Factory Functions
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 blue on LCD. Lift and
put back the handset to exit diagnostic mode.

Audio Loopback

Speak to the phone using speaker/handset/headset. If you can hear your voice, your audio is working fine. Press
“Exit” Softkey to exit audio loopback mode.

LCD on/off

Selects this option to turn off LCD. Press any button to turn on LCD.

LCD Diagnostic

Enters this option and press Left/Right Navigation key to do LCD Diagnostic. Press “Exit” Softkey to quit.

Certificate Verification

This is used to validate certificate chain for the server’s certificate.

UCM Detect

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.

Authentication
Admin Password

This is used to change the admin password for Web UI access.

End User Password

This is used to change end user password for Web UI access.

Settings

Turns on/off Test Password Strength feature. This will allow only passwords with some constraints to ensure better
security.

Operations
Factory Reset

It is used to restore the phone to factory default settings.

Rebo
Reboots the phone.
ot

Table 8: Configuration Menu

The following picture shows the keypad MENU configuration flow:

Figure 1: Keypad MENU Configuration

Configuration via Web Browser

The GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 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.

Notes:

The computer must 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 access levels for the login page:

User Level User Password Web Pages Allowed

End User Level user Set by admin Only Status and Basic Settings

Administrator Level admin admin Browse all pages

When accessing the GXP2130/2140/2160/2170/2135 for the first time or after factory reset, users will be asked to change
the default administrator password before accessing GXP21xx Web interface.

The new password field is case sensitive with a maximum length of 25 characters. Using strong password including letters,
digits and special characters is recommended for better security.

Since end user level access is disabled by default, administrators will have to enable user web access under Maintenance
🡪 Web Access, and set a new user password on the same page to enable users to access the Web UI.

Figure 2: Change Password on First Boot

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).

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 etc.

Network: To configure network settings.

Maintenance: To configure web access, upgrading and provisioning, syslog, language settings, TR-069, security etc.

Phonebook: To manage Phonebook and LDAP.

Status Page Definitions

Status 🡪 Account Status

Account Account index.

For GXP2130: up to 3 SIP accounts

For GXP2140: up to 4 SIP accounts

For GXP2160: up to 6 SIP accounts

For GXP2170: up to 6 SIP accounts

For GXP2135: up to 4 SIP accounts

SIP User ID Displays the configured SIP User ID for the account.

Displays the configured SIP Server address, URL or IP address, and port of the SIP
SIP Server
server.

Displays SIP registration status for the SIP account, it will display Yes/No with
SIP Registration
Green/Red background.

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 The configured address type: DHCP, Static IP or PPPoE.

Ipv4 Address The Ipv4 address obtained on the phone.

Ipv6 Address The Ipv6 address obtained on the phone.

OpenVPN IP The OpenVPN IP 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 obtained on the phone.

DNS Server 2 The DNS server address 2 obtained on the phone.

PPPoE Link Up PPPoE connection status.

NAT Type The type of NAT connection used by the phone.

NAT Traversal Display the status of NAT connection for each account on the phone.

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;
Locale: locale version number;
Recovery: recovery version number.

City: displaying phone location;

IP Geographic Information Language: displaying language;
Time Zone: displaying time zone;

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.

System Information Download system information

User Space Shows the percentage of the user space used and the status of the Database

Shows the status of the core dump and the core dump files generated if any. It also
Core Dump
gives the ability to generate GUI/Phone core dump files manually.

Status 🡪 Programmable Keys

Status 🡪Virtual Multi-Purpose Keys

● Mode
VPKs Status ● Account
● Description
● Value

Status 🡪 Programmable Keys

Status 🡪 Multi-Purpose Keys

● Mode
MPKs Status ● Account
● Description
● Value

Status 🡪 Programmable Keys

Status 🡪 Softkeys

● Mode
Softkeys ● Account
● Description
● Value

Status 🡪 Open Sources licenses Downloads the gpl open source licenses.

Accounts Page Definitions

Account x 🡪
General
Settings
This field indicates whether the account is active.
Account
Active
The default setting is “Yes”.

Account
The name associated with each account to be displayed on the LCD.
Name

The URL or IP address, and port of the SIP server.

SIP Server
This is provided by your VoIP service provider (ITSP).

The URL or IP address, and port of the SIP server. When configured, phone will register to both Primary and
Secondary
Secondary SIP Server. If Primary SIP Server is not reachable then the phone will use Secondary SIP Server
SIP Server
for phone services (including making/receiving calls).

IP address or Domain name of the Primary Outbound Proxy, Media Gateway, or Session Border
Controller. It’s used by the phone for Firewall or NAT penetration in different network environments.
Outbound Proxy
If a symmetric NAT is detected, STUN will not work and ONLY an Outbound Proxy can provide a
solution.

Backup Outbound IP address or Domain name of the Secondary Outbound Proxy which will be used when the primary
Proxy proxy cannot be connected.

BLF Server Optional server used for SUBSCRIBE requests to indicate other extensions status on the SIP server.

User account information, provided by your VoIP service provider (ITSP). It’s usually in the form of
digits like phone number or actually a phone number.

Field Improvement:
SIP User ID
– Users are able to register an account with a SIP user ID that carries “@”. (For example:
“[email protected]”, so the phone will register the account as “[email protected]” instead of 111)

Note: The server domain will not be included in the SIP from header.

SIP service subscriber’s Authenticate ID used for authentication.

Authenticate ID
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
Authenticate account can be registered.
Password
After it is saved, this will appear as hidden for security purpose.

Name The SIP server subscriber’s name (optional) that will be used for Caller ID display.

Voice Mail User This parameter allows you to access voice messages by pressing the MESSAGE button on the phone.
ID This ID is usually the VM portal access number. For example, in UCM6xxx IPPBX, *97 could be used.

Monitored Access Allows users to access the voice messages of monitored extension. This value is used together with
Number the voicemail programmable keys.

Picture Specifies account’s picture that will be sent to the caller/callee when making calls.
This option allows you to configure how your SIP account label will be displayed on the phone’s
screen.
Account Display
If set to “User Name”, LCD account label will display the Account Name configured for this SIP
account. If set to “User ID”, it will then display the SIP User ID configured for this SIP account.

Account x 🡪 Dial
Plan

Name Enter the name for the configured rules.

Rule Enter the rule settings (number pattern, prefix to add …etc).

Type Choose the type of the rule (pattern, block, dial now, prefix & second tone).

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:

1. Primary IP:
DNS Mode 2. Backup IP 1;
3. 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.

The option will decide which IP is going to be used in sending SIP packets after IPs for the SIP server host
are resolved with DNS SRV.

1. Default: If the option is set with “default”, it will again try to send registered messages to one IP at a
time, and the process repeats.
DNS SRV Fail- 2. Saved one until DNS TTL: If the option is set with “Saved one until DNS TTL”, it will send register
over Mode messages to the previously registered IP first. If no response, it will try to send one at a time for each IP.
This behavior lasts if DNS TTL (time-to-live) is up.
3. Saved one until no responses: If the option is set with “Saved one until no responses”, it will send
register messages to the previously registered IP first, but this behavior will persist until the registered
server does not respond.

Register This option allows to choose the behavior for registering before DNS SRV Fail-over.
Before DNS If set to “No”, a REGISTER request will not be initiated when a server failover occurred under DNS SRV mode.
SRV Failover If set to “Yes”, a REGISTER request will be initiated when a server failover occurred under DNS SRV mode.

Configures the primary IP address where the phone sends DNS query to when “Use Configured IP” is
Primary IP
selected for DNS mode.

Configures the backup IP1 address where the phone sends DNS query to when “Use Configured IP” is
Backup IP1
selected for DNS mode.

Configures the backup IP2 address where the phone sends DNS query to when “Use Configured IP” is
Backup IP2
selected for DNS mode.
This parameter configures whether the NAT traversal mechanism is activated. Users could select the
mechanism from No (Default), STUN, Keep-alive, UPnP, Auto or VPN.
If set to “STUN” and STUN server is configured, the phone will route according to the STUN server. If NAT
type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and
NAT Traversal
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. Set this to “VPN” if OpenVPN is used.

This option enables the utilization of symmetric response routing. When enabled, the "rport" field gets
appended to the Via header in the SIP Request. The necessary data is then extracted from the SIP 200OK
Support Rport
Response during a SIP Register operation, allowing the rewriting of SIP Contact information and its
(RFC 3581)
application in subsequent SIP Requests.
Enabled by Default.

A SIP Extension to notify the SIP server that the phone is behind a NAT/Firewall. Do not configure this
Proxy-Require
parameter unless this feature is supported on the SIP server.

Indicate whether or not a SBC server is used, if users want to work under SBC associated with 3CX, they
Use SBC
should enable this feature to have better communication with the server.

Account x 🡪 SIP Settings 🡪 Basic Settings

If the phone has an assigned PSTN telephone number, this field should be set to“User=Phone”.
Then a “User=Phone” parameter will be attached to the Request-Line and “TO” header in the SIP request to
TEL URI
indicate the E.164 number.
If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is “Disable”.

SIP
Selects whether the phone will send SIP Register messages to the proxy/server. The default setting is “Yes”.
Registration

Allows the SIP user’s registration information to be cleared when the phone reboots. The SIP REGISTER
message will contain “Expires: 0” to unbind the connection. Three options are available: The default setting
is “No”.
Unregister On
If set to “All”, the SIP user’s registration information will be cleared when the phone reboots. The SIP Contact
Reboot
header will contain “*” to notify the server to unbind the connection.
If set to “Instance”, the SIP user will be unregistered on current phone only.
If set to “No”, the phone will not unregister the SIP account when rebooting.

Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified registrar.
Register
The default value is 60 minutes.
Expiration
The maximum value is 64800 minutes (about 45 days).

Subscribe Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified
Expiration registrar. The maximum value is 64800 (about 45 days). The default value is 60 minutes.

Reregister
Specifies the time frequency (in seconds) that the phone sends re-registration request before the Register
Before
Expiration. The default value is 0.
Expiration

Enable
OPTIONS Keep Enable OPTIONS Keep Alive to check SIP Server.
Alive

OPTIONS Keep
Time interval for OPTIONS Keep Alive feature in Second.
Alive Interval

OPTIONS Keep
Number of max lost packets for OPTIONS Keep Alive feature before the phone re-registration.
Alive Max Lost
Defines the local SIP port used to listen and transmit. The default value is 5060 for Account 1, 5062 for
Local SIP Port Account 2, 5064 for Account 3, 5066 for Account 4, 5068 for Account 5, 5070 for Account 6. The valid range
is from 1 to 65535.

SIP
Registration Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600.
Failure Retry The default value is 20 seconds.
Wait Time

SIP T1 Timeout is an estimate of the round trip time of transactions between a client and server. If no
SIP T1
response is received the timeout is increased, and request re-transmit retries would continue until a
Timeout
maximum amount of time define by T2. The default setting is 0.5 seconds.

SIP T2 Timeout is the maximum retransmit time of any SIP request messages (excluding the INVITE
SIP T2
message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. Default is 4
Timeout
seconds.

Determines the network protocol used for the SIP transport. Users can choose from TCP, UDP and TLS. The
SIP Transport
default setting is “UDP”.

Based on option “SIP Transport” and this option “SIP Listening Mode”, GXP will decide which transport
protocol it should listening to from the incoming request. The default setting is “Transport Only”.

SIP Listening 1. Transport Only

Mode 2. Dual
3. Dual (Secured)
4. Dual (BLF Enforced)

SIP URI
Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is
Scheme when
“sips”.
using TLS

Use Actual This option is used to control the port information in the Via header and Contact header. If set to No, these
Ephemeral port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port
Port in Contact for the connection.
with TCP/TLS The default setting is “No”.

Outbound The Outbound proxy mode is placed in the route header when sending SIP messages, or they can be always
Proxy Mode sent to outbound proxy.

Support SIP
Defines whether SIP Instance ID is supported or not. Default setting is “Yes”.
Instance ID

SUBSCRIBE for When set to “Yes”, a SUBSCRIBE for Message Waiting Indication will be sent periodically. The phone
MWI 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. The default setting is “No”.
Registration

The use of the PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional
responses (1xx series). This is very important to support PSTN internetworking. To invoke a reliable
Enable 100rel
provisional response, the 100rel tag is appended to the value of the required header of the initial signaling
messages. The default setting is “No”.

When set to “Auto”, the phone will update the callee ID in the order of P-Asserted Identity Header, Remote-
Callee ID
Party-ID Header and To Header in the 180 Ringing. If “Disabled”, callee ID will be displayed as “Unavailable”.
Display
When set to “To Header”, caller ID will not be updated and displayed as To Header.
When set to “Auto”, the phone will look for the caller ID in the order of P-Asserted Identity Header, Remote-
Caller ID Party-ID Header and From Header in the incoming SIP INVITE. When set to “Disabled”, all incoming calls are
Display 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”.

Add Auth
Header on To define whether authorization Header will be added on initial REGISTER from the first REGISTER. The
Initial default setting is “No”.
REGISTER

Allow SIP This is used to perform a factory reset through SIP NOTIFY. When the phone receives the NOTIFY with event:
Reset RESET, the phone should perform a factory reset after the authentication. The default setting is “No”.

Ignore Alert- This option is used to configure default ringtone. If set to “Yes”, configured default ringtone will be played.
Info header The default setting is “No”.

Account x 🡪 SIP Settings 🡪 Custom SIP Headers

Controls whether the Privacy header will present in the SIP INVITE message or not, whether the header
Use Privacy
contains the caller info. If set to “Yes”, the Privacy Header will always show in INVITE. If set to “No”, the
Header
Privacy Header will not show in INVITE. Default setting is “Default”.

Use P-
Preferred- Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message. If set to “Yes”, the
Identity P-Preferred-Identity Header will always show in INVITE.
Header

Use X-
Enables / disables the use of X-Grandstream-PBX header in SIP request. When disabled, the SIP message
Grandstream-
sent from the phone will not include the selected header. Default setting is “Yes”.
PBX Header

Use P-Access-
Enables / disables the use of P-Access-Network-Info header in SIP request. When disabled, the SIP message
Network-Info
sent from the phone will not include the selected header. Default setting is “Yes”.
Header

Use P-
Enables / disables the use of P-Emergency-Info header in SIP request. When disabled, the SIP message sent
Emergency-
from the phone will not include the selected header. Default setting is “Yes”.
Info Header

Use X-switch- Configure whether X-switch-info Header is included in SIP REGISTER request.
info Header Set to "No" By Default

If Yes except REGISTER, the sip message for register or unregister will contain MAC address in the header,
and all the outgoing SIP messages except the REGISTER message will attach the MAC address to the User-
Agent header;
Use MAC If Yes to ALL SIP, the sip message for register or unregister will contain MAC address in the header, and all
Header the outgoing SIP messages including REGISTER will attach the MAC address to the User-Agent header;
If No, neither will the MAC header be included in the register or unregister message nor the MAC address be
attached to the User-Agent header for any outgoing SIP message.
The default setting is “No”.

Account x 🡪 SIP Settings 🡪 Advanced Features

For Shared Call Appearance, phone must send a SUBSCRIBE-request for the line-seize event package
Line Seize
whenever a user attempt to take the shared line off hook. “Line Seize Timeout” is the line-seize event
Timeout
expiration timer. The default value is 15 seconds. The valid range is from 15 to 60.
Configures the Eventlist BLF URI on the phone to monitor the extensions in the list with Multi-Purpose Key. If the
server supports this feature, users need to configure an Eventlist BLF URI on the service side first (i.e.,
Eventlis
[email protected]) with a list of extensions included. On the phone, in this “Eventlist BLF URI” field, fill in
t BLF
the URI without the domain (i.e., BLF1006). To monitor the extensions in the list, under Web
URI
GUI🡪Settings🡪Programmable Keys page, please select “Eventlist BLF” in the key mode, choose account, enter
the value of each extension in the list.

Auto
Provisi
When option is enabled, empty multi-purpose keys will be automatically provisioned to the monitored
on
extensions in the Eventlist BLF. The default setting is “Disabled”.
Eventlis
t BLFs

Confere
Configures Conference URI for N-way conference (Broadsoft Standard).
nce URI

Music
On
Configures Music On Hold URI to call when a call is on hold. This feature must be supported on the server side.
Hold
URI

Configures BLF Call-pickup method:

If select Auto:

The phone will do either Prefix or barge in code for BLF pickup depend on which on is set.
BLF
Call- If select Force BLF Call-pickup by prefix:
pickup
The phone will only use Prefix as BLF pickup method.

If select Disabled:

The phone will ignore both BLF pickup method, now the monitored VPK will only dial the extension if pressed

BLF
Call- Configures the prefix prepended to the BLF extension when the phone picks up a call with BLF key. The default
pickup setting is **.
Prefix

Call
Pickup
Set feature access code of Call Pickup with Barge-In feature.
Barge-
In Code

PUBLIS
H for
Enables presence feature on the phone. The default setting is “No”.
Presenc
e

Omit
charset
=UTF-8
Omit charset=UTF-8 in MESSAGE content-type
in
MESSA
GE
Allow
Unsolici
Allow Unsolicited REFER to accomplish an outgoing call.
ted
REFER

Different soft switch vendors have special requirements. Therefore, users may need select special features to
Special
meet these requirements. Users can choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro,
Feature
PhonePower and UCM Call center depending on the server type. The default setting is “Standard”.

When set to “Yes”, a Softkey “BSCCenter” is displayed on LCD. User can access different Broadsoft Call Center
agent features via this Softkey.

Broadsoft Please note that “Feature Key Synchronization” will be enabled regardless of this setting. Default setting is
Call Center “No”.

Note: To activate this feature, users need to change the special feature to Broadsoft and setup the Broadsoft
Call Center to take effect.

Hoteling Broadsoft Hoteling event feature. Default setting is “No”. With “Hoteling Event” enabled, user can access the
Event Hoteling feature option by pressing the “BSCCenter” softkey.

Call Center When set to “Yes”, the phone will send SUBSCRIBE to the server to obtain call center status. The default
Status setting is “No”.

Broadsoft
Executive When enabled, Feature Key Synchronization will be enabled regardless of web settings.
Assistant

Feature
This feature is used for Broadsoft call feature synchronization. When it’s enabled, DND, Call Forward features
Key
and Call Center Agent status can be synchronized between Broadsoft server and phone. Default is
Synchroniz
“Disabled”.
ation

Broadsoft When enabled, it will send SUBSCRIBE to Broadsoft server to obtain Call Park notifications. The default
Call Park setting is “Disabled”.

Account x
🡪 SIP
Settings 🡪
Session
Timer

Enable
This option is used to enable or disable session timer on the phone side when server side can provide both
Session
session timer UPDATE or session audit UPDATE. The default setting is “Yes”.
Timer

The SIP Session Timer extension (in seconds) that enables SIP sessions to be periodically “refreshed” via a SIP
request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be
Session
terminated once the session interval expires. Session Expiration is the time (in seconds) where the session is
Expiration
considered timed out, provided no successful session refresh transaction occurs beforehand. The default
setting is 180. The valid range is from 90 to 64800.

The minimum session expiration (in seconds). The default value is 90 seconds. The valid range is from 90 to
Min-SE
64800.
Caller
If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it
Request
makes outbound calls. The default setting is “No”.
Timer

Callee
If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it
Request
receives inbound calls. Default setting is “No”.
Timer

If Force Timer is set to “Yes”, the phone will use the session timer even if the remote party does not support
Force
this feature. If Force Timer is set to “No”, the phone will enable the session timer only when the remote party
Timer
supports this feature. To turn off the session timer, select “No”. The default setting is “No”.

UAC
As a Caller, select UAC to use the phone as the refresher; or select UAS to use the Callee or proxy server as
Specify
the refresher. The default setting is “Omit”.
Refresher

UAS
As a Callee, select UAC to use caller or proxy server as the refresher; or select UAS to use the phone as the
Specify
refresher. The default setting is “UAC”.
Refresher

The Session Timer can be refreshed using the INVITE method or the UPDATE method. Select “Yes” to use the
Force INVITE method to refresh the session timer.
INVITE
The default setting is “No”.

Account x
🡪 SIP
Settings 🡪
Security
Settings

Check
Choose whether the domain certificates will be checked or not when TLS/TCP is used for SIP Transport. The
Domain
default setting is “No”.
Certificates

Validate
Certificate Validate certification chain when TCP/TLS is configured. Default setting is “No”.
Chain

Validate
Incoming Choose whether the incoming messages will be validated or not. The default setting is “No”.
Messages

Check SIP
User ID for If set to “Yes”, SIP User ID will be checked in the Request URI of the incoming INVITE. If it doesn’t match the
Incoming phone’s SIP User ID, the call will be rejected. The default setting is “No”.
INVITE

Accept
Incoming When set to “Yes”, the SIP address of the Request URL in the incoming SIP message will be checked. If it
SIP from doesn’t match the SIP server address of the account, the call will be rejected. The default setting is “No”.
Proxy Only

Authentica
te If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized
Incoming response. Default setting is “No”.
INVITE
Account x
🡪 Audio
Settings

Preferred Multiple vocoder types are supported on the phone, the vocoders in the list is a higher preference. Users can
Vocoder configure vocoders in a preference list that is included with the same preference order in SDP message.

Use First
Matching When it is set to “Yes”, the device will use the first matching vocoder in the received 200OK SDP as the
Vocoder in codec. The default setting is “No”.
200OK SDP

Codec Configures the phone to use which codec sequence to negotiate as the callee. When set to “Caller”, the
Negotiatio phone negotiates by SDP codec sequence from received SIP Invite. When set to “Callee”, the phone
n Priority negotiates by audio codec sequence on the phone. The default setting is “Callee”.

Hide When option Hide Vocoder is set as Yes, the coded will be hidden from call screen as bellow. The default
Vocoder setting is “No”.

Disable
When it is set to “No”, the device will reply with multiple m lines; Otherwise, it will reply 1 m line. The default
Multiple m
setting is “No”.
line in SDP

SRTP
Enable SRTP mode based on your selection from the drop-down menu. The default setting is “Disabled”.
Mode

Allows users to specify the length of the SRTP calls. The available options are: AES 128&256 bit, AES 128 bit
SRTP Key and AES 256 bit.
Length
Default setting is: AES 128&256 bit

Crypto Life Enable or disable the crypto life time when using SRTP. If users set to disable this option, phone does not add
Time the crypto life time to SRTP header. The default setting is “Yes”.

Symmetric
Defines whether symmetric RTP is supported or not. Default setting is “No”.
RTP

Silence Controls the silence suppression/VAD feature of the audio codecs except forG.723 (pending) and G.729. If set
Suppressio to “Yes”, a small quantity of RTP packets containing comfort noise will be sent during the periods of silence. If
n set to “No”, this feature is disabled. Default setting is “No”

Jitter
Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The default setting is
Buffer
“Adaptive”.
Type

Jitter
Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default setting is
Buffer
“300ms”.
Length

Configures the number of voice frames transmitted per packet. When configuring this, it should be noted
that the “ptime” value for the SDP will change with different configurations here. This value is related to the
Voice
codec used and the actual frames transmitted during the in-payload call. For end users, it is recommended to
Frames Per
use the default setting, as incorrect settings may influence the audio quality.
TX

The default setting is 2.

This option determines the encoding rate for G723 codec. Users can choose from 6.3kbps encoding rate and
5.3kbps encoding rate.
G723 Rate

The default setting is “5.3kbps encoding rate”.

G.726-32
Packing Selects “ITU” or “IETF” for G726-32 packing mode. The default setting is “ITU”.
Mode

iLBC This option determines the iLBC packet frame size. Users can choose from 20ms and 30ms. The default
Frame Size setting is “30ms”.

iLBC
Payload This option is used to specify iLBC payload type. Valid range is 96 to 127. The default setting is “97”.
Type

OPUS
Specifies OPUS payload type. Valid range is 96 to 127. Cannot be the same as iLBC or DTMF Payload Type.
Payload
Default value is 123.
Type

DTMF
Payload Configures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or OPUS payload type.
Type

This parameter specifies the mechanism to transmit DTMF digits. There are 3 supported modes:

• In audio: DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs);

Send • RFC2833 sends DTMF with RTP packet. Users can check the RTP packet to see the DTMFs sent as well as the
DTMF number pressed.

• SIP INFO uses SIP INFO to carry DTMF.

Default setting is “RFC2833”.

DTMF
Configures the delay between sending DTMF during MPK/VPK use (in milliseconds).
Delay

Account x
🡪 Call
Settings

Selects whether to enable early dial. If it’s set to “Yes”, the SIP proxy must support 484 responses. Early Dial
means that the phone sends for each pressed digit a SIP INVITE message to SIP server. SIP server considers
its extensions and, if no match happened yet, it sends back a “484 Address Incomplete” message. Otherwise,
Early Dial
it executes the action.

The default setting is “No”.

Dial Plan
Configures the prefix to be added to each dialed number.
Prefix
Dial Plan 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. Default setting is “{ x+ | \+x+ | *x+ | *xx*x+ }”. 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

X – digits from 0-9, and letters from a-z, A-Z.

1. xx+ – at least 2-digit numbers

2. xx — only 2-digit numbers

3. ^ — exclude

4. [3–5] — any digit of 3, 4, or 5

5. [147] — any digit of 1, 4, or 7

6. <2=011> – replace digit 2 with 011 when dialing

7. | — the OR operand

8. {X123} — match Z123, e123, 5123, …

9. Flag T when adding a “T” at the end of the dial plan, the phone will wait for 3 seconds before dialing out.
This gives users more flexibility on their dial plan setup. E.g. with dial plan 1XXT, phone will wait for 3
seconds to let user dial more than just 3 digits if needed. Originally the phone will dial out immediately
after dialing the third digit.

10. Back slash “\” — can be used to escape specific letters. E.g. if { \p\a\r\k\+60 } dial plan is configured,
park+60 should be able to pass dial plan check. This also can be used to escape Mark and User-
unreserved characters.

Mark = “-“ / “_” / “.” / “!” / “~” / “*” / “’” / “(“ / “)”
User-unreserved = “&” / “=” / “+” / “$” / “,” / “;” / “?” / “/”

Example 1: {[369]11 | 1617xxxxxxx}

Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617;

Example 2: {^1900x+ | <=1617>xxxxxxx}

Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers;

Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}

Allows any number with leading digit 1 followed by a 3-digit number, followed by any number between 2
and 9, followed by any 7-digit number OR Allows any length of numbers with leading digit 2, replacing the 2
with 011 when dialed.

Example 4: If we set the dial plan with {\*123}, it should allow input *123 to pass dial plan check.

Example 5: If we set the dial plan with {\$123}, it should allow input $123 to pass dial plan check.

Example 6: If we set the dial plan with {12\_3}, it should allow input 12_3 to pass dial plan check.

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.

Configures Call Log setting on the phone. You can log all calls, only log incoming/outgoing calls (missed calls
Call Log
will not be logged), or disable call log. The default setting is “Log All Calls”.

Allows users to configure the ringtone for the account. Users can choose from different ringtones from the
Account
dropdown menu.
Ring Tone
Note: User can also choose silent ring tone.

Specifies matching rules with number, pattern or Alert Info text or string (up to 10 matching rules). 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; [6-9]xx: 3-digit number with the leading digit from 6 to 9.

Match
Incoming Note: Custom ringtone can be defined for matching rule of Caller ID beginning with +
Caller ID 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: <http://127.0.0.1>; info=priority

Alert Info string

Users could configure the matching rule as certain string (e.g., string) 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: <string> Selects the distinctive ring tone for the matching rule. When the
incoming caller ID or Alert Info test or string matches one of the 10 rules, the phone will ring with the
associated ringtone.

Ring Defines the timeout (in seconds) for the rings on no answer. The default setting is 60. The valid range is from
Timeout 10 to 300.

Send
If set to “Yes”, the “From” header in outgoing INVITE messages will be set to anonymous, blocking the Caller
Anonymou
ID to be displayed. Default is “No”.
s

Anonymou
s Call If set to “Yes”, anonymous calls will be rejected. The default setting is “No”.
Rejection

Auto If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short
Answer reminding beep. Default setting is “No”.
This function allows users to have the phone configured with a pre-defined list of numbers that will perform
auto answer.

There are different situations for auto answer setup:

1) Auto Answer disable → no auto answer;

Auto
2) Auto Answer enable, no auto answer number specified → all numbers are auto answered;
Answer
Numbers
3) Auto Answer enable, auto answer number specified → only numbers specified will be auto answered.

For “Auto Answer Numbers”, it accepts:

Digits :1,2,3,4,5,6,7,8,9; x – any digit from 0-9; xx – any two digits from 0-9; [1-5] – any digit from 1 to 5; +: it
matches the previous character as many time as needed like regular expression. Please note Auto Answer
Numbers can be split with “;”, for example: 1x;2xxx;3x+

Refer-To
If set to “Yes”, the “Refer-To” header uses the transferred target’s Contact header information for attended
Use Target
transfer. The default setting is “No”.
Contact

Transfer
on If set to “Yes”, when the phone hangs up as the conference initiator, the conference call will be transferred to
Conferenc the other parties so that other parties will remain in the conference call. The default setting is “No”.
e Hang-up

Disables recovery to the call to the transferee on failing blind transfer to the target. The default setting is
“No”.

Notes:

Disable
1) This feature only applies to blind transfer;
Recovery
on Blind
2) This feature depends on how server handles transfer. If there is any NOTIFY from server, this feature won’t
Transfer
take effect. If server responds 4xx, phone should try to recover regardless of this option.

3) During blind transfer, after transferor received 200/202 for REFER, but there is no NOTIFY from server after
7 seconds, transferor will decide to recover the call with transferee or not depending on the options. This is
the only case that this option will be applied.

Blind
Transfer
Defines the timeout (in seconds) for waiting SIP frag response in blind transfer. Valid range is 30 to 300.
Wait
Timeout

No Key
Defines the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the digits will be sent
Entry
out. The default value is 4 seconds. The valid range is from 1 to 15.
Timeout

Use # as Allows users to configure either the “*” or “#” keys as the “Send” key. Please make sure the dial plan is
Dial Key properly configured to allow dialing * and # out. The default setting is “Pound (#)”.

On Hold
If set to “Enabled”, phone will play a reminder tone when it has a call on hold. The default setting is
Reminder
“Disabled”.
Tone

RFC2543 Allows users to toggle between RFC2543 hold and RFC3261 hold. RFC2543 hold (0.0.0.0) allows user to
Hold disable the hold music sent to the other side. RFC3261 (a line) will play the hold music to the other side.
Hide
Dialing Allows users to hide the password when the dialing number matches the configured prefix.
Password

Disable
Enables / disables the call waiting feature for the current account. When set to “Default”, global call feature
Call
setting will be used. Default setting is Default.
Waiting

Account x
🡪
Intercom
Settings

Allow
Auto
Allows the phone to automatically turn on the speaker phone to answer incoming calls after a short
Answer by
reminding beep when enabled, based on the SIP Call-Info/Alert-Info header sent from the server/proxy.
Call-
Default setting is “Yes”.
Info/Alert-
Info

Allow
When enabled, the phone will automatically put the current call on hold and answer the incoming call based
Barging by
on the SIP Call-Info/Alert-Info header sent from the server/proxy. However, if the current call was answered
Call-
based on the SIP Call-Info/Alert-Info header, then all other incoming calls with SIP Call-Info/Alert-Info
Info/Alert-
headers will be rejected automatically. Default setting is “No”.
Info

Mute on
answer
When enabled, the phone will mute the incoming intercom call.
Intercom
call

Play
warning
tone for
When enabled, the phone will play warning tone when auto answer Intercom.
Auto
Answer
Intercom

Custom
Alert-Info Allows to customize Alert-Info header for auto answer. The phone will auto answer only if matching content
for Auto of the custom Alert-info header.
Answer

Account x
🡪 Feature
Codes

When enabled, Do Not Disturb, Call Forwarding and other call features can be used via the local feature
codes on the phone. Otherwise, the provisioned feature codes from the server will be used. User configured
feature codes will be used only if server provisioned feature codes are not provided. And once feature codes
are configured, either via server provisioning or local setting, a Softkey named “Features” will show on the
Enable
LCD screen.
Local Call
Features

Note: If the device is registered with Broadsoft account, it doesn’t matter if local call features are enabled or
disabled, once the Broadsoft account is set, special feature to Broadsoft and Feature Key Synchronization is
enabled, the call feature will be handled by Broadsoft server, not by the phone.
Do Not
Disturb
Configures DND feature code to turn on DND.
(DND)—
On

Do Not
Disturb
Configures DND feature code to turn off DND.
(DND)—
Off

Call
Forward
Unconditio Configures Call Forward All feature code to activate unconditional call forwarding.
nally (All)
—On

Call
Forward
Unconditio Configures Call Forward All feature code to deactivate unconditional call forwarding
nally (All)
—Off

Target Configures the extension that the call will be forwarded to.

Call
Forward Configures Call Forward Busy feature code to activate busy call forwarding.
Busy—On

Call
Forward Configures Call Forward Busy feature code to deactivate busy call forwarding.
Busy—Off

Target Configures the extension that the call will be forwarded to.

Call
Forward
Delayed
Configures Call Forward Delayed feature code to activate no answer call forwarding.
(No
Answer)—
On

Call
Forward
Delayed
Configures Call Forward Delayed feature code to activate no answer call forwarding.
(No
Answer)—
Off

Target Configures the extension that the call will be forwarded to.

Delayed
Call Defines the timeout (in seconds) before the call is forwarded on no answer. The default value is 20 seconds.
Forward The valid range is 1 to 120.
Wait Time

Accounts
🡪 Account
Swap
Allows users to swap the two accounts that they have configured. This will Increase the flexibility of account
Swap
management.
Account
Settings
Note: Make sure to press “Start” to complete the process.

Table 10: Account Page Definitions

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 this port _value for RTP;
Local RTP Port
channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024 to 65400
and must be even. Default value is 5004.

Gives users the ability to define the parameter of the local RTP port used to listen and
transmit. This parameter defines the local RTP port from 48 to 10000. This range will
Local RTP Port Range
be adjusted if local RTP port + local RTP port range is greater than 65486. Default
setting is 200.

When set to “Yes”, this parameter will force random generation of both the local SIP
and RTP ports. This is usually necessary when multiple phones are behind the same
Use Random Port
full-cone NAT. The default setting is “Yes”
Note: This parameter must be set to “No” for Direct IP Calling.

Specifies how often the phone sends a blank UDP packet to the SIP server to keep the
Keep-alive Interval “ping hole” on the NAT router to open. The default setting is 20 seconds. The valid
range is from 10 to 160.

The NAT IP address used in SIP/SDP messages. This field is blank at the default
Use NAT IP
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 results are
STUN Server
displayed in the STATUS page of the Web GUI.

Delay Registration Configures specific time that the account will be registered after booting up.

Only allow password with these constraints to ensure better security:

Password needs at least 9 characters and 3 of the following options:
1) Numeric (0-9)
Test Password Strength 2) Capital letters (A-Z)
3) Lower case (a-z)
4) Special characters (!, @, #, $, %, ^, &, *, (, ), etc)
Default setting is “No”.

Configures to turn on/off the public mode for hot desking feature. The default setting
is “No”.
Enable Public Mode
Note: When the public mode is enabled, the local phonebook can be shown on the
device

Enable Fix For RTP Timestamp Makes RTP timestamps be continuous to fix the audio loss issue when there is a jump
Jump in RTP timestamp. Default is No.

Public Mode Username Prefix Used as prefix of public mode login, when public mode is enabled

Public Mode Username Suffix Used as suffix of user name in public mode login, when public mode is enabled.
Settings 🡪 Broadsoft 🡪 Broadsoft XSI

Configures XSI Directory.

Server
Configure the BroadWorks Xsi server URI. If the server uses HTTPS, please add the
header “HTTPS” ahead of the Server URI.
For instance, “https://SERVER_URI”.
Port
Configure the BroadWorks Xsi server port. The default port is 80. If the server uses
HTTPS, please configure 443.
XSI Actions Path
This feature allows users to configure the deployment path for Broadsoft XSI Actions.
If it is empty, the path “com.broadsoft.xsi-actions” will be used.
Broadsoft Contact Download Interval
Configures the Broadsoft phonebook download interval (in minutes).
If set to 0, automatic download will be disabled. Valid range is 5 to 720.
Broadsoft Contacts Download Limitation
Configures Broadsoft contacts download limit. Default is 100. Range is 0-2000
Broadsoft Contacts Search Limitation
Configures Broadsoft contacts Search limit. Default is 100. The range is 0-2000
XSI
XSI Authentication Type:
Login Credentials
SIP Credentials
Account ½/3/4/5/6
Select XSI Authentication Type. SIP User ID need to be configured if SIP account is
selected.
Login Credentials:
Login Username.
Configure the Username for the BroadWorks Xsi server.
Login Password.
Configure the password for the BroadWorks Xsi server.
SIP Credentials
SIP User Name.
Configure SIP Username for the BroadWorks Xsi server.
SIP User ID.
Configure SIP User ID for the BroadWorks Xsi server.
SIP Password
Configure SIP Password for the BroadWorks Xsi server.

Network Directories Enable/Disable Broadsoft Network directories and defines the directory name. The
directory types are:
Group Directory
Enable/Disable and rename the BroadWorks Xsi Group Directory features on the
phone. If keep the Name box blank, the phone will use the default name “Group” for it.
Enterprise Directory
Enable/Disable and rename the BroadWorks Xsi Enterprise Directory features on the
phone. If keep the Name box blank, the phone will use the default name “Enterprise”
for it.
Group Common
Enable/Disable and rename the BroadWorks Xsi Group Common Directory features on
the phone. If keep the Name box blank, the phone will use the default name “Group
Common” for it.
Enterprise Common
Enable/Disable and rename the BroadWorks Xsi Enterprise Common Directory
features on the phone. If keep the Name box blank, the phone will use default name
“Enterprise Common” for it.
Personal Directory
Enable/Disable and rename the BroadWorks Xsi Personal Directory features on the
phone. If keep the Name box blank, the phone will use the default name “Personal” for
it.
Missed Call Log
Enable/Disable and rename the BroadWorks Xsi Missed Call Log features on the
phone. If keep the Name box blank, the phone will use the default name “Missed” for it.
Placed Call Log
Enable/Disable and rename the BroadWorks Xsi Placed Call Log features on the
phone. If keep the Name box blank, the phone will use the default name “Outgoing” for
it.
Received Call Log
Enable/Disable and rename the BroadWorks Xsi Placed Call Log features on the
phone. If keep the Name box blank, the phone will use the default name “Incoming” for
it.

Settings 🡪 Broadsoft 🡪 Broadsoft IM&P

Server
Broadsoft IM&P server address. Usually not necessary to configure and can already be
found in the Broadsoft IM&P username.
Port
Login Credentials Port for the Broadsoft IM&P server. Default port is 5222.
Username
Broadsoft IM&P username, not the Broadsoft account username.
Password
Broadsoft IM&P password, not the Broadsoft account password.

Enables Broadsoft Instant Message and Presence feature. The default setting is
Broadsoft IM&P
“Disabled”.

Associated Broadsoft Account Specifies the associated account. User could choose each account on the phone.

Choose to whether login to the Broadsoft IM&P account at boot-up. The default setting
Auto Login
is “No”.

Choose whether to display non-xmpp contacts associated with the Broadsoft IM&P
Display Non XMPP Contacts user. Non-xmpp contacts will not display a presence or status message. The default
setting is “No”.

Settings 🡪 External Service

Order Displays the order of the service.

Specifies the service’s type. Two options are available: None or GDS.
Default setting is None. Note: The GXP21xx supports up 10 GDS items.
Service Type
For more details, refer to https://documentation.grandstream.com/knowledge-
base/connecting-gds37xx-with-gxp-phones/.

Account Specifies the account on which the service will be applied.

Specifies the name to identify the service.

System Identification The field “System identification” will show up on LCD next to the GDS number as
indicated:

Specifies the system number, in case the service type option is set to GDS, the system
System Number number is the SIP user ID configured on GDS3710, or the IP address of the GDS3710
itself if it’s using IP call.

Determines the access password in case the service type option is set to GDS, the
Access Password access password is the one configured on “Remote PIN to Open the Door” field on
GDS3710 settings.

Settings 🡪 Call Features

Allows user to select a default account when other accounts have not been selected.
Preferred default Account The chosen account will be used for live DialPad and auto Redial. However, if this
account is not active, then the first account that is active will be used.
Allow users to show/hide the predictive dialing feature, when disabled, users will not
Predictive Dialing Feature
see any predictive numbers while dialing a number.

The predictive dialing feature will sequentially search the number based on the
Predictive Dialing Source selected sources from these: Call History, Local Phonebook, Remote Phonebook,
Feature Code, and LDAP.

When disabled, on-hook dialing won’t be interrupted by an incoming call. Default

Onhook Dial Barging
setting is Disabled.

Disable Mute Key in Call This feature allows users to disable mute key during a call.

Configures a User ID/extension to dial automatically when the phone is off hook. The
Off-hook Auto Dial
phone will use the first account to dial out. Default setting is “No”.

Configures the number of seconds during which the phone will wait before dialing out
Off-hook Auto Delay
when off-hood auto dial number is configured. The default is 0.

If configured, when the phone is off hook, it will go on hook after the timeout (in
Off-hook Timeout
seconds). The default value is 30 seconds. Valid range is from 10 to 60.

If enabled, When the phone is Offhook it will automatically dial out the number
Enable Live DialPad
punched in after the number of seconds that the user had set.

Set the Live DialPad expire time before initiating the call using Live DialPad feature.
Live DialPad Expire Time
Interval is between 2s and 15s. Default value is 5s.

If enabled, the phone will redial the number a configured number of times with a
Enable Automatic Redial
configured interval (in seconds) in between each redial.

Automatic Redial Times The number of times to attempt to call using Automatic Redial feature.

Automatic Redial Interval The interval between each call attempt using Automatic Redial feature.

Bypass Dial Plan Through Call Enable/Disable the dial plan check while dialing through the call history and any
History and Directories phonebook directories. The default setting is “No”.

Disable Call Waiting Disables the call waiting feature. The default setting is “No”.

Disable Call Waiting Tone Disables the call waiting tone when call waiting is on. Default setting is “No”.

Disables / enables the call waiting tone when the call waiting feature is enable. Default
Ring For Call Waiting
is disabled.

Disable Busy Tone on Remote Disables the busy tone heard in the handset when call is disconnected remotely. The
Disconnect default setting is “No”.

Disable Direct IP Call Disables Direct IP Call. The default setting is “No”.

When set to “Yes”, users can dial an IP address under the same LAN/VPN segment by
entering the last octet in the IP address.
To dial quick IP call, off hook the phone and dial #XXX (X is 0-9 and XXX <=255), phone
Use Quick IP Call mode 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”.
Disables the Conference function.
Disable Conference
The default setting is “No”.

If enabled, the phone only allows a call associated with the same account to build a
Only Same Account in Conference
conference.

When it’s set to “Yes”, the DTMF digits entered during the call will not be displayed on
Disable in-call DTMF Display
phone LCD. The default setting is “No”.

Enable Sending DTMF via specific Allows certain MPKs to send DTMF in-call.
MPKs This option doesn’t affect Dial DTMF.

When enabled, active MPK page on the extension board will be disabled. Default
Disable Active MPK Page
setting is disabled.

When option is enabled, Active VPK Page will be displayed on LCD when there are
Enable Active VPK Page active VPKs.
Default setting is Disabled.

If set to “No”, the user can not turn on Do Not Disturb feature via MUTE key, MPK, or
menu on LCD. Default is Yes.
Enable DND Feature Note: Now DND function can be support by 3CX server. It will display on CTI and web
client page. When DUT uses DND mode, it will send SIP INFO. That will sync with the
server and show on CTI and Web Client.

Preserve DND Status Configures whether DND status should be saved after a reboot.

Specifies the function of mute key in idle. Default setting is “DND”.

When select “Idle Mute” and press Mute key while idle, the future incoming call will be
Mute Key Functions While Idle
answered with mute. When select “Disabled”, Mute key will not take effect while idle.
The default setting is “DND”.

Allows the phone to accept certain incoming calls while set to DND mode.
Off: all incoming calls will not be accepted.
Allow all: all incoming calls will be allowed.
DND Override Allow Only Contacts: only incoming calls from numbers in the local phonebook will be
accepted.
Allow Only Favorites: only incoming calls from favorite numbers in the local
phonebook will be accepted.

Enables/disables transfer feature. If disabled, call transfer will not be possible. Default
Disable Transfer
setting is “No”.

In-call Dial Number on Pressing

Configures the number to be dialed as DTMF using TRANSFER button.
Transfer Key

If set to “Dynamic”, attended transfers will be performed by default. The default setting
Attended Transfer Mode is “Static”. For more details about “Static” and “Dynamic” transfer, refer to the user
guide.

When this option is set to “Yes”, users can see how long their call has been hold.
Show On Hold Duration
Default is “No”.

This feature allows users to filter out specific characters in dial-out calls. such as "(",
Filter Characters
"+"...

Specifies whether to replace # by %23 or not for some special situations. The default
Do Not Escape # as %23 in SIP URI
setting is “No”.
Enables Click-To-Dial feature. If this feature is enabled, user could click the green dial
Click-To-Dial Feature button on left top corner of phone’s Web GUI, then choose the account and dial to the
target number. The default setting is “Disabled”.

Sets the default call log list after select MENU🡪CALL HISTORY. Broadsoft Call Log or
Default call log type Local Call Log option will only show its own list. Default option will keep both call log
lists.

Return Code When Refusing When refusing the incoming call. The phone will send the selected type of SIP
Incoming Call message of the call. Default setting is “Busy 486”.

When DND is enabled, the phone will send the selected type of SIP message. Default
Return Code When Enable DND
setting is “Busy 486”.

By enabling BLF Pickup Screen, when monitored BLF is ringing, GXP should pop up a
Enable BLF Pickup Screen
BLF information window. The default setting is “No”.

Gives the user the ability to set sound notification to the monitoring BLF line when it’s
Enable BLF Pickup Sound ringing, GX21xx should play a sound to inform user.
The default setting is “No”.

Configures the list to be playing BLF sound notification for all except extensions in this
BLF Pickup Sound Except List
list. Separate extensions by comma (,)

BLF Pickup Sound Only list Configures play BLF sound notification only for the list below.

Gives the ability to record calls locally while on the call screen.
Local Call Recording Feature
The default setting is “Disabled”.

Saved Local Call Recording Location Location where the recordings will be stored.

Download Local Call Recordings When there are recordings presented, you may download them here.

Enable IM Popup If set to “No”, phone will not show a pop up when receiving an IM.

Configures the number of seconds that the message will remain on screen. Default
Instant Message Popup Timeout
setting is “10”.

If enabled, phone will play a short tone when receiving an IM during idle state. Default
Play Tone On Receiving IM
setting is disabled.

This allows incoming calls after dialed but before ringing. This can be used under
Allow Incoming Call Before Ringing
custom user configuration based on need. Default setting is No.

Add a new option for input the user agent field with operator configurable value or
value that identifies the device. The option should be configurable to give the end point
User-Agent Prefix device specific identification.
Ex. The value could be Mobile, Fixed, Desktop, etc. The configured “User Agent” should
be prepend to vendor’s default User.

Users could select “Extension Boards” or “VPK” which will be used first when the
Auto Provision List Starting Point phone is being automatically provisioned with Eventlist BLF. The default setting is
“Extension Boards”.

Hide BLF Remote Status Allows users to hide the Caller ID from showing at the BLF VPK and EXT
Disabled: The VPK will flash between the Caller ID and the BLF account.
Enabled: The VPK will stay under the monitored account and only notify that there is an
incoming call.

Show SIP Error Response Allows users to disable the SIP error message that will be shown on the call screen.

Allows users to show/hide the notification popup for missed calls. Default is “No”
Enable Missed Call Notification which will hide call notification popup.
Note: Currently the manually rejected calls are counted as missed calls.

When the automatic redial and call completion service are enabled, and the user
Enable Call Completion Service makes a call to callee, when the callee is busy at the moment, phone will monitor
callee’s status. Once the callee is available, phone will ask if user wants to redial again.

If set to “Yes”, phone will pop up an incoming call window to notify the call.
If set to “No”, there will be no notification pop up on LCD when there is an incoming
Enable Incoming Call Popup
call. This way users will not get disrupted by unexpected popup call but still get
notified by the flashing line LED.

Enable Enhanced Acoustic Echo Allows users to choose whether to enable or disable the echo canceller on their phone
Canceller in speaker mode.

If set to “Yes”, when forwarding a call, the recipient will display a “diverted from”
Enable Diversion Information message. If set to “No”, when phone receives a forwarded call, phone will not display
Display the “diverted from” message.
Default is “Yes”.

When set to “Yes”, disable hook switch completely; When set to “For Answer Call”,
Disable Hook Switch hook switch cannot be used for answering call.
Default is “No”.

Settings 🡪 Call History

Delete Users can select an entry, then click “Delete” to remove it from the list.

Click on Delete All in order to remove all Call History stored in the phone.
Note: Users could use the drop-down list to show only selected call history type (All,
Delete all
Answered, Dialed, Missed, Transferred) and also use navigation keys to browse pages
when many entries exist.

Settings 🡪 Multicast Paging

Allowed In DND Mode Allow Multicast Paging when DND mode is enabled.

During active call, if incoming multicast page is higher priority (1 being the highest)
Paging Barge than this value, the call will be held and multicast page will be played. The default
setting is “Disabled”.

If enabled, during a multicast page if another multicast is received with higher priority
Paging Priority Active
(1 being the highest) that one will be played instead. The default setting is “Disabled”.

The codec for sending multicast pages, there are 5 codecs could be used: PCMU,
Multicast Paging Codec
PCMA, G.726-32, G.729A/B, G.722 (wide band). Default setting is “PCMU”.

Multicast Channel Number (0-50). 0 for normal RTP packets, 1-50 for Polycom
Multicast Channel Number
multicast format packets.

Outgoing caller ID that displays to your page group recipients (for multicast channel 1
Multicast Sender ID
– 50).
Defines multicast listening addresses and labels. For example:
“Listening Address” should match the sender’s Value such as “237.11.10.11:6767”
Multicast Listening
“Label” could be the description you want to use.
For details, please check the “Multicast Paging User Guide” on our Website.

Settings 🡪 Outbound Notification

For detailed instruction for this part, please refer to: [Outbound Notification Support]
Section in this Administration Guide.
Setup Completed
Registered
Unregistered
Register Failed
Off Hook
On Hook
Incoming Call
Outgoing Call
Missed Call
Answered Call
Rejected Call
Forwarded Call
Established Call
Terminated Call
Idle to Busy
Busy to Idle
Action URL Open DND
Close DND
Open Forward
Close Forward
Open Unconditional Forward
Close Unconditional Forward
Open Busy Forward
Close Busy Forward
Open No Answer Forward
Close No Answer Forward
Blind Transfer
Attended Transfer
Transfer Finished
Transfer Failed
Hold Call
UnHold Call
Mute Call
IP Change
Auto-Provision Finish

Up to 10 destinations can be configured here. For detailed instruction for this part,
Destination
please refer to: [Outbound Notification Support] Section in this Administration Guide.

Specifies the message body of the notification for each event that can be customized
with embedded dynamic attributes.
Notification
For more details, refer to: [Outbound Notification Support] section in this
Administration Guide.

Settings 🡪 Preferences 🡪 Audio Control

Headset Key Mode 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.
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 Headset/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.

Selects whether the connected headset is normal RJ11 headset, Plantronics EHS
Headset Type
headset. Default setting is “Normal”.

Allows user to enable the ringtone from Plantronics EHS headset and play the ringtone
in the headset.
EHS Headset Ring Tone
Note: It also requires to set “Headset Key Mode” to “Toggle Headset/Speaker” and
manually press the HEADSET button on the keypad to switch to Headset mode.

Configures to enable or disable the speaker to ring when headset is used on “Toggle
Headset/Speaker” mode.
If set to “Yes, both”, when the phone is in Headset “Toggle Headset/Speaker” mode,
Always Ring Speaker
both headset and speaker will ring on incoming call.
If set to “Yes, speaker only”, when the phone is in Headset “Toggle Headset/Speaker”
mode, only speaker will ring on incoming call.

If enabled, the phone will display a soft key while on call to enable the speaker
Group Listen with Speaker
listening when the handset or headset is used.

Configures the transmission gain of the headset.

Headset TX gain
The default value is 0 dB.

Configures the receiving gain of the headset.

Headset RX gain
The default value is 0 dB.

Configures the transmission gain of the handset.

Handset TX gain
The default value is 0 dB.

Settings 🡪 Preferences 🡪 Date and Time

Defines the URL or IP address of the NTP server. The phone may obtain the date and
NTP Server time from the server.
The default setting is “pool.ntp.org”.

Defines the URL or IP address of the NTP server. The phone may obtain the date and
Secondary NTP Server time from the server. Allow user to configure 2 NTP server domain names. GXP will
loop through all of the IP addresses resolved from them.

Time interval for updating time from the NTP server. Valid time value is in between 5 to
NTP Update Interval 1440 minutes.
The default setting is “1440” minutes.

Defines whether DHCP Option 42 should override NTP server or not. When enabled,
Allow DHCP Option 42 Override NTP
DHCP Option 42 will override the NTP server if it’s set up on the LAN. The default
Server
setting is “Yes”.

Configures the date/time used on the phone according to the specified time zone.
Time Zone
Note: Daylight Saving Time (DST) is no longer added on Mexico city timezone.

Self-Defined 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 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, 3rd Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, … ,Sat)
Therefore, this example is the DST which starts from the First Sunday of April to the
1st Sunday of November.

Configures the date display format on the LCD. The following formats are supported.
The default setting is yyyy-mm-dd:

1. yyyy-mm-dd: 2012-07-02
Date Display Format 2. mm-dd-yyyy: 07-02-2012
3. mm-dd-yyyy: 07-02-2012
4. dddd, MMMM dd: Friday, October 12
5. MMMM dd, dddd: October 12, Friday

Configures the time display in 12-hour or 24-hour format on the LCD. The default
Time Display Format
setting is in 12-hour format.

Allows users to display time and date on the top panel of the LCD screen, you can
select to show time only or to show both Date and Time.
Show Date and Time on Status Bar The default setting is Disabled.
Note: For GXP2135 and GXP2170, the time and date will be displayed on top of the
LCD when the top VPK on the right side of the LCD screen is not configured.

Shows the Clock Widget on the main screen.

Show Clock Widget
Enabled by Default.

Settings 🡪 Preferences 🡪 LCD Display

Configures the LCD brightness when the phone is active. Valid range is 10 to 100
Backlight Brightness: Active
where 100 is the brightest. Default value is 100.

Configures the LCD brightness when the phone is idle. Valid range is 0 to 100 where 0
Backlight Brightness: Idle
is off and 100 is the brightest. Default value is 60.

Allows user to set up the backlight time (in minutes) for the extension board. Valid
Active Backlight Timeout range from 0 to 90. Default value is 1.
Note: When Active Backlight Timeout is set to 0, the backlight will be constantly on.

Disables/enables LCD backlight when there is a missed call notification.

Please note the user must select “Color Background” in “Wallpaper Source” option in
order to use the configurable color background code.
Disable Missed Call Backlight If set to “Yes, but flash MWI LED”, the phone will turn off LCD backlight but MWI will
not be deemed when there is a missed call.
If set to “No”, the phone will not turn off LCD backlight when there is a missed call.
The default setting is “No”.

Specifies the wallpaper source mode: Default, Download, USB, Uploaded and Color
Background. User could upload a wallpaper source into your phone, or download it
Wallpaper Source
from file server with the server path, or plug your USB drive with wallpaper source into
GXP2140/GXP2160/GXP2170 to upload the wallpaper.

(Note: USB is only for GXP2140, Note: If you choose “Color Background”, you need to enter a HEX color code based on
GXP2160 and GXP2170) your preference. The color codes could be found here:
Specifies the wallpaper server path. This option will take effect when wallpaper source
Wallpaper Server Path
is “Download”.

Click on the “Upload” button to browse and upload the desired wallpaper file. This
Upload Wallpaper
option will take effect when wallpaper source is “Uploaded”.

Enter a color you wish to use in HEX format. E.g. #000000

Reference: http://htmlcolorcodes.com
Color Background
Please note the user must select “Color Background” in “Wallpaper Source” option in
order to use the configurable color background code.

Configures Screensaver Feature, or “to enable Screensaver feature if no VPK is active”.

Please note this option is also available under LCD🡪 Menu🡪 Preference🡪
Screensaver Appearance. The phone will consider the page active if VPK is in Early (ringing), Trying
(dialing) and Confirmed (talking) status when VPK is configured with mode “BLF”,
“Eventlist BLF” or “Presence”. By default, screensaver is enabled.

Sets the location where screensaver is loaded from. If from USB, please have a folder
Screensaver Source
named “screensavers” containing your pictures.

Show Date and Time Allows to see time and date on phone’s screensaver mode

Configures the minutes of idle before the screensaver activates. Valid range is 3 to 6.
Screensaver Timeout
The default time is 3 minutes.

Screensaver Server Path Configures the server path which contains download screensaver definition XML.

Configures the screensaver XML download interval (in minutes). If set to 0, automatic
Screensaver XML Download Interval
download will be disabled. Valid range is 5 to 720.

Settings 🡪 Preferences 🡪 LED Control

This is used to configure the color and pattern of the LED based on status updates.
BLF LED Pattern The default setting is “Default”.
The BLF LED Patterns are listed in [Table 15: BLF LED Patterns].

The VM/MSG light cannot flash even though there’s an unread voice mail or message
Disable VM/MSG power light flash
when set to “Yes”. Default settings is “No”.

Users could view the color and pattern of the LED status based on the BLF status
BLF LED Pattern Explanation Form
update.

Settings 🡪 Preferences 🡪 Ring Tone

Call Progresses Tones

Configures ring or tone frequencies based on parameters from local telecom. The
System Ring Tone
default value is North American standard. Frequencies should be configured with
Dial Tone
known values to avoid uncomfortable high pitch sounds.
Second Dial Tone
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
Message Waiting
(Frequencies are in Hz and cadence on and off are in 10ms)
Ring Back Tone
ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence.
Call-Waiting Tone
In order to set a continuous ring, OFF should be zero. Otherwise, it will ring ON ms and
Busy Tone
a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.
Reorder Tone

Call-Waiting Tone Gain Configures the call waiting tone gain to adjust call waiting tone volume. Default is Low.

Configures speaker ring volume. The valid range is 0 to 7.

Speaker Ring Volume
The default setting is 5.
Notification Tone Volume Configures notification tone volume. The valid range is 0 to 7 and default setting is 5.

Locks volume adjustment when the option is enabled.

● If set to “No” (Default), the volume of the speaker will not be locked.
Lock Speaker Volume ● If set to "Ring", the volume of the speaker will be locked in a ringing state and idle.
● If set to “Talk”, the volume of the speaker will be locked in the calling state.
● If set to "Both", the volume of the speaker will be locked.

Allows to set Default Ringtone as their Global ringtone.

Note: The ringtone set in individual accounts has higher priority than this setting. If the
user wants the default ring tone to be used globally, he needs to set the ring tone of
Default Ringtone
each account to Default Ring Tone; Otherwise, it will be whichever ring tone you set.
Important: The Priority goes as Contact Ring Tone 🡪 Account Ring Tone 🡪 Default
Ring Tone.

Total Number of Custom Ringtone Configures the total number of custom ringtone update that can be downloaded during
Update provisioning process. Default is 3.

Settings 🡪 Programmable Keys

Show Label Background

If enabled, the VPK label’s background will match the status of the VPK and will no
longer be transparent
Use Long Label
If enabled, the VPK label will extend as far as possible.
Key Mode
If set to “Line Mode”, the amount of VPKs will be the amount of lines you can have. If
set to “Account Mode”, the lines will be grouped by account, so the VPKs could hold
more lines in one account.
For example, with line mode, when the line is in use, by pressing the VPK, nothing is
going to happen. In Account Mode, when the line is in use, by pressing the VPK, a new
line will be initiated.
Virtual Multi-Purpose Keys Settings
Transfer Mode via VPK
Allows users to configure “Transfer” VPK to do either Blind or Attended Transfer. They
can also set their Transfer key to make a new call with the configured number.
Enable Transfer via non-Transfer MPK
MPK with type BLF, Speed dial, etc. will perform as transfer MPK under active call.
Show VPK Icon
Show call screen VPK icon. When hidden, call screen can reserve more room for label
Show Keys Label
If set to “Show” side labels will be shown during calls.
If set to “Hide”, side labels will be hidden during calls for more space to display the
user information. If set to “Toggle”, a softkey will appear so that users can click to
Show/Hide the side labels.

Virtual Multi-Purpose Keys Assigns a function to the corresponding line key. The key mode options are:
Line
Regular line key to open up a line and switch line.The 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.
Note: Users can either show or hide VPK shared line display description, This only can
be done with provisioning using the Pvalue P8484 (Value = 0; No . Value = 1; Yes)
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to
be dialed, or enter the IP address to set the Direct IP call as Speed Dial.
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 Voice Mail access number in the Value field, you can define a
description for the Voicemail as well if needed.
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.
Call Park
Select Account, and enter the call park extension in the Value field to park/pick up the
call.
Monitored Call Park
Select account from Account field, and enter the call park extension in the Value field
to park/pick up the call, and also monitor the parked call via Line Key’s light.
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
Description 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
Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For
example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since the Base for LDAP server configuration is: “dc=grandstream,dc=com”,
“ou=video,ou=SZ” is added to narrow the LDAP search scope. “sn=Li” is the example to
filter the last name.
Conference
Allow user to set their Multi-Purpose Key to “Conference” mode to trigger a
conference.
By setting the extension number in the value box, the users will be able to activate a 3-
way conference by simply press the assigned MPK button.
Multicast Paging
This option is for multicast sending. Enter Line key description in Description field and
multicast sending address in Value field.
Record
This option is for Recording calls. Enter Line key description in Description filed and
the recorded extension number in Value field. Please make sure whether your VOIP
provider supports this feature before using it.
Call Log
Select Account and enter account number in the Value field to allow configuration of
call log for other extension.
Menu
Select this feature in order to display the Menu from the MPK buttons, no field dis
required for configuration.
XML Application
Select this feature in order to start the XML Application from the MPK buttons, no field
dis required for configuration.
Information
Select this feature in order to display the Information popup to show the firmware
version, MAC address, IP address and IP Settings from the MPK buttons, no field dis
required for configuration.
Message
Select this feature in order to display the Message menu from the MPK buttons, no
field dis required for configuration
Forward
Set the MPK Button to perform call forwarding to the destination number configured
on the “Value Field”. During ringing press the button to perform the call forward.
DND
Press the configured key to enabled/Disable DND.
Redial
On this mode, the configured key can be used to redial numbers.
Instant Messages
On this mode, the configured key can be used to enter IM menu and send new
messages.
Multicast Listen Address
The MPK button can be used to access directly to the Multicast listening IP list.
Keypad Lock
Configure the VPK button to be used to lock/unlock the keypad.

Physical Multi-Purpose Keys Assigns a function to the corresponding physical MPK. This feature is available on
GXP2130/GXP2160 only. 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, or enter the IP address to set the Direct IP call as Speed Dial.
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.
Note: 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.
Call Park
Select Account, and enter the call park extension in the Value field to park /pick up the
call.
Monitored Call Park
Select account from Account field, and enter the call park extension in the Value field
to park/pick up the call, and also monitor the parked call via Line Key’s light.
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
Description 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 Description field is left blank.
Enter the LDAP Name/Number filter in the Value field. For example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since the Base for LDAP server configuration is: “dc=randstream,dc=com”,
“ou=video,ou=SZ” is added to narrow the LDAP search scope. “sn=Li” is the example to
filter the last name.
Conference
Allow user to set their Multi-Purpose Key to “Conference” mode to trigger a
conference. By setting the extension number in the value box, the users will be able to
activate a 3-way conference by simply press the assigned MPK button.
Multicast Paging
This option is for multicast sending.
Enter Line key description in Description field and the multicast sending address in
Value field.
Record
This option is for Recording calls. Enter Line key description in Description filed and
the recorded extension number in Value field. Please make sure whether your VOIP
provider supports this feature before using it.
Call Log
Select Account and enter account number in the Value field to allow configuration of
call log for other extension.
Menu
Select this feature in order to display the Menu from the MPK buttons, no field dis
required for configuration.
XML Application
Select this feature in order to start the XML Application from the MPK buttons, no field
dis required for configuration.
Information
Select this feature in order to display the Information popup to show the firmware
version, MAC address, IP address, and IP Settings from the MPK buttons, no field is
required for configuration.
Message
Select this feature in order to display the Message menu from the MPK buttons, no
field is required for configuration
Forward
Set the MPK Button to perform call forwarding to the destination number configured
on the “Value Field”.
During ringing press the button to perform the call forward.
DND
Press the configured key to enable/Disable DND.
Redial
On this mode, the configured key can be used to redial numbers.
Instant Messages
On this mode, the configured key can be used to enter IM menu and send new
messages
Multicast Listen Address
The MPK button can be used to access directly to the Multicast listening IP list.
Keypad Lock
Configure the MPK button to be used to lock/unlock the keypad.

Softkeys Settings More Softkey Display Mode

Allows users to choose from the original Toggle mode or the enhanced Menu mode.
With the enhanced Menu mode, the MORE softkey now will not need the user to tap
multiple times on MORE to get to the next pages, instead, pressing MORE will have a
popup window and allow users to choose from the list. With Toggle mode, users need
to press the MORE softkey to switch between options.
Show Target Softkey
Allows users to remove target softkey by toggling the Yes/No option during the off-
hook dial screen and transfer screen.
Custom Softkey Layout
Enables/Disables custom softkey layout.
Enforce Softkey Layout Position
Whether to enforce the custom softkey layout position.
When enabled, GUI will still preserve the space if the configured softkey is unable to
show.
Hide System Softkey on Main Page
Configures to hide the system-generated softkey (Next, History, ForwardAll, Redial) on
the main page. The default value is none.
Dialing State
Custom softkey layout when the device is under Dialing State.
Available Softkeys: Phonebook(BT), BT On/Off, EndCall, ReConf, ConfRoom, Redial,
Dial, Backspace, PickUp.
On-hook Dialing State
Custom softkey layout when the device is under On-hook Dialing State.
Available Softkeys: Phonebook(BT), DirectIP, Cancel, Dial, Backspace
Ringing State
Custom softkey layout when the device is under Ringing State.
Available Softkeys: Answer, Reject, Forward, ReConf.Calling State
Calling State
Custom softkey layout when the device is under calling State.
Available Softkeys: BT On/Off, EndCall, ReConf, ConfRoom, ConfCall.
Call Connected State
Custom softkey layout when the device is under call connected State.
Available Softkeys: Phonebook(BT), BT On/Off, EndCall, ReConf, ConfRoom, ConfCall,
Cancel, New Call, Swap, Transfer, Trnf>VM, DialDTMF, BS-Ccenter, Record
On/Off(UCM), Record On/Off, CallPark(UCM), PrivateHold, CallPark.
Conference Connected State
Custom softkey layout when the device is under Conference Connected State.
Available Softkeys: BT On/Off, EndCall, Kick.
On Hold State
Custom softkey layout when the device is under On Hold State.
Available Softkeys: ReConf, Resume, Transfer, ConfCall, Add.
Call Failed State
Custom softkey layout when the device is under Call Failed State.
Available Softkeys: EndCall, ReConf, ConfRoom.
Transfer State
Custom softkey layout when the device is under Transfer State.
Available Softkeys: BT On/Off, Cancel, BlindTrnf, AttTrnf, Backspace.
Conference State
Custom softkey layout when the device is under Conference State.
Available Softkeys: BT On/Off, Cancel, Dial, Backspace.

Idle Screen Softkeys Assigns a function to the corresponding Softkeys. GXP2140, GXP2160 and GXP2170
supports 3 configurable Softkeys; GXP2130/GXP2135 supports 2 configurable
Softkeys.
Note: The first and last Softkeys are reserved for Exit/More functionality.
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.
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 off-hook and account 2 is active, it will call the configured Speed Dial
number using account 2.
Voice Mail
Select Account & 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.
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
Description 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 Description field is left blank. Enter
the LDAP Name/Number filter in the Value field.
For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since the Base for LDAP server configuration is “dc=randstream,dc=com”,
“ou=video,ou=SZ” is added to narrow the LDAP search scope. “sn=Li” is the example to
filter the last name.
Call Log
Select Account and enter the account number in the Value field to access the Call Log
of that selected account.
Menu
Select this feature in order to display the Menu from the MPK buttons, no field is
required for configuration.
Information
Select this feature in order to display the Information popup to show the firmware
version, MAC address, IP address, and IP Settings from the MPK buttons, no field is
required for configuration.
Message
Select this feature in order to display the Message menu from the MPK buttons, no
field is required for configuration

Assigns a function to the corresponding Call Screen Softkeys.

Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to
be dialed.
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 & 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.
Intercom
Select Account, and enter the extension number in the Value field to do the intercom.
LDAP Search
Call Screen Softkeys This option is to narrow the LDAP search scope. Enter the LDAP search base in the
Description 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 Description field is left blank. Enter
the LDAP Name/Number filter in the Value field.
For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since the Base for LDAP server configuration is “dc=randstream,dc=com”,
“ou=video,ou=SZ” is added to narrow the LDAP search scope. “sn=Li” is the example to
filter the last name.
Call Log
Select Account and enter account number in the Value field to access to the Call Log
of that selected account.
Information
Select this feature in order to display the Information popup to show the firmware
version, MAC address, IP address and IP Settings from the MPK buttons, no field dis
required for configuration.
Message
Select this feature in order to display the Message menu from the MPK buttons, no
field dis required for configuration

Extension Boards

One Page Display Mode

Each extension board only shows 20 Extensions, that is, EXT 1 ~ 80 could be displayed
EXT setting (Available only for
on 4 connected boards if the mode is enabled.
GXP2140/2170)
Sync Backlight with LCD
If set to yes, the Extension Board backlight will turn off when LCD is idle.
EXT (1-4) (Available only for Assigns a function to the corresponding Extension Board key.
GXP2140/2170) The key mode options are:
None
Select this option in order to disable the key.
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to
be dialed, or enter the IP address to set the Direct IP call as Speed Dial.
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.
Call Park
Select Account, and enter the call park extension in the Value field to park/pick up the
call.
Monitored Call Park
Select account from Account field, and enter the call park extension in the Value field
to park/pick up the call, and also monitor the parked call via Line Key’s light.
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
Description 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 Description field is left blank.
Enter the LDAP Name/Number filter in the Value field.
For example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since the Base for LDAP server configuration is “dc=randstream,dc=com”,
“ou=video,ou=SZ” is added to narrow the LDAP search scope. “sn=Li” is the example to
filter the last name.
Conference
Allow user to set their Multi-Purpose Key to “Conference” mode to trigger a
conference.
By setting the extension number in the value box, the users will be able to activate a 3-
way conference by simply press the assigned MPK button.
Multicast Paging
This option is for multicast sending.
Enter Line key description in Description field and multicast sending address in Value
field.
Record
This option is for Recording calls. Enter Line key description in Description filed and
the recorded extension number in Value field. Please make sure whether your VOIP
provider supports this feature before using it.
Call Log
Select Account and enter account number in the Value field to allow configuration of
call log for other extension.
Menu
Select this feature in order to display the Menu from the MPK buttons, no field dis
required for configuration.
XML Application
Select this feature in order to start the XML Application from the MPK buttons, no field
dis required for configuration.
Information
Select this feature in order to display the Information popup to show the firmware
version, MAC address, IP address and IP Settings
from the MPK buttons, no field dis required for configuration.
Message
Select this feature in order to display the Message menu from the MPK buttons, no
field dis required for configuration.

Settings 🡪 Web Service

Configures to enable or disable auto location services on the phone. (Reboot

Use Auto Location Service
Required). The default setting is “Yes”.

Settings 🡪 XML Application

Configures the server path to download the idle screen XML file. This field could be IP
Server Path
address or URL, with up to 256 characters.

Specifies the Softkey name displayed on the idle screen for the users to enter XML
Softkey Label application.
The default Softkey Label is “XMLApp”.

nters a color to use in HEX format. Default will be transparent.

Default Background Color
E.g. #000000. Reference: http://htmlcolorcodes.com

Permits to block auto-switching to call screen when XML application is running.

Block Call Screen
Default is disabled.

Enable XML Application Auto With this option is enabled, the phone will launch XML application automatically when
Launch there is an incoming call. Default is No.

Network Page Definitions

Network 🡪 Basic Settings

Internet
Selects Prefer Ipv4 or Prefer Ipv6. The default setting is “Prefer Ipv4”.
Protocol

Ipv4 Allows users to configure the appropriate network settings on the phone to obtain Ipv4 address. Users could
Address select “DHCP”, “Static IP” or “PPPoE”. By default, it is set to “DHCP”.

Host name
Specifies the name of the client. This field is optional but may be required by some Internet Service Providers.
(Option 12)
DHCP
Used by clients and servers to exchange vendor class IDs.
Vendor
The default setting is “Grandstream GXP2130” for GXP2130, ”Grandstream GXP2140” for GXP2140,
Class
“Grandstream GXP2160” for GXP2160, “Grandstream GXP2170” for GXP2170 and “Grandstream GXP2135” for
ID (Option
GXP2135.
60)

PPPoE
Enter the PPPoE account ID.
Account ID

PPPoE
Enter the PPPoE Password.
Password

PPPoE
Service Enter the PPPoE Service Name.
Name

Ipv4
Enter the IP address when static IP is used.
Address

Subnet
Enter the Subnet Mask when static IP is used for IPv4.
Mask

Gateway Enter the Default Gateway when static IP is used for IPv4.

DNS Server
Enter the DNS Server 1 when static IP is used for IPv4.
1

DNS Server
Enter the DNS Server 2 when static IP is used for IPv4.
2

Preferred
Enters the Preferred DNS Server for Ipv4.
DNS Server

Ipv6
Allows users to configure the appropriate network settings on the phone to obtain IPv6 address. Users could
Address
select “Auto-configured” or “Statically configured” for the IPv6 address type.
Type

Static Ipv6
Enter the static IPv6 address when Full Static is used in “Statically configured” Ipv6 address type.
Address

Ipv6 Prefix
Enter the IPv6 prefix length when Full Static is used in “Statically configured” Ipv6 address type.
Length

Ipv6 Prefix Enter the IPv6 Prefix (64 bits) when Prefix Static is used in “Statically configured” IPv6 address type.

DNS Server
Enter the DNS Server 1 for IPv6.
1

DNS Server
Enter the DNS Server 2 for IPv6.
2

Preferred
Enter the Preferred DNS Server for IPv6.
DNS server

Network 🡪 Advanced Settings

802.1X Allows the user to enable/disable 802.1X mode on the phone. The default value is disabled. To enable 802.1X
mode mode, this field should be set to EAP-MD5, users may also choose EAP-TLS, or EAP-PEAP.

802.1X Enter the Identity information for the 802.1x mode.

Identity Note: Letters, digits, and special characters including @ and – are accepted.

MD5 Enter the MD5 Password for the 802.1X mode.

Password Note: Letters, digits and special characters including @ and – are accepted.

802.1X CA Uploads / deletes the 802.1X CA certificate to the phone; or delete existed 802.1X CA certificate from the
Certificate phone.

802.1X
Uploads / deletes 802.1X Client certificate to the phone; or delete existed 802.1X Client certificate from the
Client
phone.
Certificate

Specifies the HTTP proxy URL for the phone to send packets to. The proxy server will act as an intermediary to
HTTP Proxy
route the packets to the destination.

HTTPS Specifies the HTTPS proxy URL for the phone to send packets to. The proxy server will act as an intermediary
Proxy to route the packets to the destination.

Bypass
Enter host names that do not require a proxy to reach. Those names should be separated by commas.
Proxy For

Layer 3 QoS Defines the Layer 3 QoS parameter for SIP. This value is used for IP Precedence, Diff-Serv or MPLS. The default
for SIP value is 26.

Layer 3 QoS Defines the Layer 3 QoS parameter for RTP. This value is used for IP Precedence, Diff-Serv or MPLS. The default
for RTP value is 46.

Release
Allows users to determine whether to release DHCP upon reboot. Enabled by default. This option change
DHCP On
requires a reboot before taking effect.
Reboot

Enable Enables auto-configure for VLAN settings through DHCP.

DHCP VLAN Disabled by default.

Enable
Manual
Enables/disables manual VLAN configuration. When this option is set to Disabled, the phone will bypass VLAN
VLAN
configuration and only use the DHCP VLAN to configure VLAN tag and priority. Default is “Enabled”.
Configuratio
n

Layer 2 QoS
802.1Q/VLA Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0.
N Tag

Layer 2 QoS
802.1p
Assigns the priority value of the Layer2 QoS packets. The default value is 0.
Priority
Value

PC Port Configure the PC port mode. When set to “Mirrored”, the traffic in the LAN port will go through PC port as well
Mode and packets can be captured by connecting a PC to the PC port. The default setting is “Enabled”.

PC Port
Assigns the VLAN Tag of the PC port. The default value is “0”.
VLAN Tag
PC Port
Priority Assigns the priority value of the PC port. The default value is “0”.
Value

Enable CDP Enables/Disables CDP “Cisco Discovery Protocol”.

Enable LLDP Controls the LLDP (Link Layer Discovery Protocol) service. The default setting is “Enabled”.

LLDP TX
Defines LLDP TX Interval (in seconds). Valid range is 1 to 3600.
Interval

Network 🡪 Remote Control

Action URI
Enable/disabled action URI feature on the phone.
Support

Remote
control Pop
Indicates whether the phone is enabled to pop up allow remote control.
up window
support

List of allowed IP address from which the phone receives action URI. The Allowed IP addresses followed by
Action URI
their subnet mask are separated by a comma such as “192.168.1.1/24,192.168.1.2/24”. Set this field to “any”
allowed IP
to allow any IP address to send Action URL to the phone. The default value is empty string which means no IP
list
address is allowed for remotely control the phone.

CSTA Indicates whether CSTA Control feature is enabled. Change of this configuration will need the system to reboot
Control to take effect.

Network 🡪 Affinity Settings

Allows communication with GS Affinity CTI application to manage telephone calls from a computer. If enabled,
Affinity
a reboot is required to establish the communication. The default is “Disabled”.
Support
GS Affinity CTI Application is available HERE and its User Guide is HERE.

Preferred
Selects the account on which CTI support is enabled.
Account

Network
🡪 Bluetooth
Settings

Bluetooth
Configures Bluetooth to power on, off or off with hiding menu from LCD. Default setting is “On”.
Power

Handsfree Enable / disable Bluetooth handsfree feature.

Mode Default setting is “Off”.

Persistent This option is used to retain bluetooth devices throughout public mode.
Bluetooth Disabled by Default.

Bluetooth
Specifies the Bluetooth device name.
Name

Network 🡪 OpenVPN® Settings

OpenVPN® Enable/Disable OpenVPN® feature. Default is No.

Enable

Configures the OpenVPN configuration mode, options are simple mode and Expert mode.
OpenVPN®
The default value is the simple mode.
mode
If the user chooses expert mode, he will have to upload an OpenVPN® config zip file.

Upload
OpenVPN®
Uploads the OpenVPN config .zip file
config zip
file

OpenVPN®
Server Specify the IP address or FQDN for the OpenVPN® Server.
Address

OpenVPN®
Specify the listening port of the OpenVPN® server. Default is 1194.
Port

OpenVPN®
Specify the Transport Type of OpenVPN® whether UDP or TCP. Default is UDP.
Transport

OpenVPN® Click on “Upload” to upload the Certification Authority of OpenVPN®. For a new upload, users could click on
CA “Delete” to erase the last certificate, and then upload a new one.

OpenVPN® Click on “Upload” to upload OpenVPN® certificate. For a new upload, users could click on “Delete” to erase the
Certificate last certificate, and then upload a new one.

OpenVPN® Click on “Upload” to upload OpenVPN® Key.

Client Key For a new upload, users could click on “Delete” to erase the last certificate, and then upload a new one.

OpenVPN®
Using the Upload button, users can upload their OpenVPN® TLS Auth key. Clicking "Delete" will erase the last
TLS Auth
certificate, and the new one will be uploaded.
key

OpenVPN®
Specifies the Cipher method used by the OpenVPN® server. The available options are: Blowfish, AES-128, AES-
Cipher
256 and Triple-DES. Default setting is: Blowfish.
Method

OpenVPN®
Configures the optional username for authentication if the OpenVPN server supports it.
Username

OpenVPN®
Configures the optional password for authentication if the OpenVPN server supports it.
Password

Configures whether to enable OpenVPN® Comp-lzo compression feature. When Comp-lzo is enabled on the
OpenVPN®
OpenVPN server, it must also be enabled on the phone. Otherwise, the network will fail to connect.
Comp-lzo
Default Value is "Yes"

Additional options to be appended to the OpenVPN® config file, separated by semicolons. For example, comp-
Additional lzo no;auth SHA256
Options Note: Please use this option with caution. Make sure that the options are recognizable by OpenVPN® and do
not unnecessarily override the other configurations above.

Network 🡪 SNMP Settings

Enable
Enables/Disables the SNMP feature. Default settings is No.
SNMP
Version SNMP version.

Port SNMP port (Default 161).

SNMP Trap
IP address of the SNMP trap receiver.
IP

SNMP Trap
Port of the SNMP trap receiver (Default 162)
Port

SNMP Trap
The interval between each trap sent to the trap receiver
Interval

Trap version Trap version of the SNMP trap receiver

Community SNMP Community

SNMP Trap
Community string associated to the trap. It must match the community string of the trap receiver.
Community

SNMP
Username for SNMPv3
Username

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.
Security
authUser: Users with security level authNoPriv and context name as auth.
Level
privUser: Users with security level authPriv and context name as priv.

Authenticati
Select the Authentication Protocol: “None” or “MD5” or “SHA”.
on Protocol

Privacy
Select the Privacy Protocol: “None” or “DES” or “AES”.
Protocol

Authenticati
Enter the Authentication Key.
on Key

Privacy Key Enter the Privacy Key.

SNMP Trap
User name for SNMPv3 Trap.
Username

Trap ● noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.
Security ● authUser: Users with security level authNoPriv and context name as auth.
Level ● privUser: Users with security level authPriv and context name as priv.

Trap
Authenticati Select the Authentication Protocol: “None” or “MD5” or “SHA”.
on Protocol

Trap Privacy
Select the Privacy Protocol: “None” or “DES” or “AES”.
Protocol

Trap
Authenticati Enter the Trap Authentication Key
on Key
Trap Privacy
Enter the Trap Privacy Key.
Key

Table 12: Network Page Definitions

Maintenance Page Definitions

Maintenance 🡪 Web Access

Administrator can disable or enable user web access. This option is disabled by
Enable User Web Access
default.

New 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 Password The current admin password is required for setting a new admin password.

Set new password for web GUI access as Admin. The admin password is case
New Password
sensitive with a maximum length of 25 characters.

Confirm Password Enter the new Admin password again to confirm.

Maintenance 🡪 Upgrade and Provisioning

Allows users to upload the firmware file locally by pressing Start, after selecting the
Upgrade Firmware correct firmware file from the local storage, the phone will start the firmware upgrade
automatically.

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
Firmware Upgrade and Provisioning
Skip the Firmware Check.
The default setting is “Always Check for New Firmware”.

Always Authenticate Before Only applies to HTTP/HTTPS. If enabled, the phone will send credentials before being
Challenge challenged by the server. The default setting is “No”.

Default setting is “Yes”. DHCP option 66 originally was only designed for TFTP server.
Later on it was extended to support an HTTP URL. GXP phones support both TFTP
Allow DHCP Option 43 and Option 66 and HTTP server via option 66. Users can also use DHCP option 43 vendor specific
Override Server option to do this.
DHCP option 43 approach has priorities. The phone is allowed to fall back to the
original server path configured in case the server from option 66 fails.

When enabled, users could select Option 150 or Option 160 to override the firmware
server instead of using the configured firmware server path or the server from option
Additional Override DHCP Option 43 and option 66 in the local network. Please note this option will be effective only
when option “Allow DHCP Option 43 and Option 66 to Override Server” is enabled. The
default setting is “None”.

Allow DHCP Option 120 to override Enables DHCP Option 120 from local server to override the SIP Server on the phone.
SIP Server The default setting is “No”.

Enables automatic provision feature on the phone when 3CX is used as the SIP
3CX Auto Provision
server. The default setting is “Yes”.
Automatic Upgrade Enables automatic upgrade and provisioning. The default setting is “No”.

Randomized Automatic Upgrade within the range of hours of the day or postpone the
Randomized Automatic Upgrade
upgrade every X minute(s) by random 1 to X minute(s).

Defines the hour of the day to check the HTTP/TFTP/FTP server for firmware
Hour of the Day (0-23)
upgrades or configuration files changes. The default value is 1.

Defines the day of the week to check HTTP/TFTP/FTP server for firmware upgrades
Day of the Week (0-6)
or configuration files changes. The default value is 1.

Device will not challenge NOTIFY with 401 when set to “Yes”.
Disable SIP NOTIFY Authentication
Default setting is “No”.

If set to “Yes”, the phone will ask the user to upgrade. If there is no response, the
phone will proceed with the upgrade.
Firmware Upgrade Confirmation
If set to “No”, the phone will automatically upgrade without user input.
Default is Yes.

Config

Allows users to choose the config upgrade method: TFTP, FTP, FTPS, HTTP or HTTPS.
Config Upgrade Via
The default setting is “HTTPS”.

Config Server Path Defines the server path for provisioning.

Config HTTP/HTTPS User Name The user name for the HTTP/HTTPS server.

Config HTTP/HTTPS Password The password for the HTTP/HTTPS server.

Enables your ITSP to lock configuration updates. If configured, only the configuration
Config File Prefix
file with the matching encrypted prefix will be downloaded and flashed into the phone.

Enables your ITSP to lock configuration updates. If configured, only the configuration
Config File Postfix file with the matching encrypted postfix will be downloaded and flashed into the
phone.

The password for encrypting XML configuration file using OpenSSL. This is required
XML Config File Password
for the phone to decrypt the encrypted XML configuration file.

Sets the phone system to authenticate configuration file before applying it. When set
to “Yes”, the configuration file must include value P1 with phone system’s
Authenticate Conf File administration password. If it is missed or does not match the password, the phone
system will not apply it.
Default setting is “No”.

Click to download phone’s configuration file in .txt format.

Download Device Configuration
Note: Configuration backup file doesn’t include passwords or CA/Custom certificate

When user protection is on, pvalues that user sets will not be changed by provision or
provider.
If “User protection” is OFF, everyone (Provider, user or admin) has access to most of
User protection
the Pvalues.
If “User protection” is ON, only those (normally user or admin) who have privilege can
modify the configuration.
By default, device will provision the first available config in the order of cfgMAC,
cfgMAC.xml, cfgMODEL.xml and cfg.xml (corresponding to device specific, model
Download and Process All Available specific and global configs).
Config Files If this option is enabled, the phone will inverse the downloading process to cfg.xml >
cfggxp21xx.xml > cfgMAC.bin > cfgMAC.xml. The following files will override the files
that has already been load and processed.

This allows users to download part of the configuration that does not include any
personal settings like Username and Passwords. Also, it will include all the changes
Download User configuration manually made by user from web UI, or config file uploaded from “Upload Device
Configuration”, but not include the changes from the server provision via
TFTP/FTP/FTPS/HTTP/HTTPS.

Upload Device Configuration Uploads configuration file to phone.

Export backup Package Export backup package which contains device configuration along with personal data.

Restore from Backup package Click to upload backup package and restore.

Firmware

Allows users to choose the firmware upgrade method:

Firmware Upgrade Via
TFTP, FTP, FTPS, HTTP or HTTPS. The default setting is “HTTPS”.

Firmware Server Path Defines the server path for the firmware server.

Firmware HTTP/HTTPS User Name The user name for the HTTP/HTTPS server.

Firmware HTTP/HTTPS Password The password for the HTTP/HTTPS server.

Enables your ITSP to lock firmware updates. If configured, only the firmware with the
Firmware File Prefix
matching encrypted prefix will be downloaded and flashed into the phone.

Enables your ITSP to lock firmware updates. If configured, only the firmware with the
Firmware File Postfix
matching encrypted postfix will be downloaded and flashed into the phone.

Maintenance 🡪 Syslog

If set to SSL/TLS, the syslog messages will be sent through secured TLS protocol to
syslog server.
Syslog Protocol
Default setting is UDP.
Note: The CA certificate is required to connect with the TLS server.

The URL or IP address of the syslog server for the phone to send syslog to.
Syslog Server Note: By adding port number to the Syslog server field (i.e 172.18.1.1:1000), the
phone will send syslog to the corresponding port of that IP.

Syslog Level 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:

● 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).

Syslog will be filtered based on keywords provided. If you enter multiple keywords, it
Syslog Keyword Filtering
should be separated by ‘,’. Please note that no spaces are allowed.

Configures whether the SIP log will be included in the syslog messages. The default
setting is “No”.
Send SIP Log
Note: By setting Send SIP Log to Yes, the phone will still send SIP log from syslog
even when Syslog Level set to NONE.

Show Internet Down Message If enabled, the internet down warning message will display when internet is down.

If set to “Yes”, the phone will automatically recover when running abnormal. The
Auto Recover From Abnormal
default setting is “Yes”.

USB Console Log If enabled, console log will be saved into USB drive.

Maintenance 🡪 Language

Selects display language on the phone. There are 21 languages can be set as display
Display Language language, user could also choose “Auto” or “Downloaded Language” as display
language. The default setting is “Auto”.

Configures the default input selection.

The default setting is “Multi-Tap”.
Default Input Selection
Multi-Tap: multi-tap to switch character;
Shiftable: select input from available characters.

This is used to configure the device to download language files automatically from
Auto language download
server. The default setting is “No”.

Maintenance 🡪 TR-069

Enable TR-069 Enables/Disables TR-069 service.

URL for TR-069 Auto Configuration Servers (ACS).

ACS URL
Default setting is: https://acs.gdms.cloud

TR-069 Username ACS username for TR-069.

TR-069 Password ACS password for TR-069.

Enables periodic inform. If set to “Yes”, device will send inform packets to the ACS.
Periodic Inform Enable
The default setting is “No”.

Sets up the periodic inform interval to send the inform packets to the ACS.
Periodic Inform Interval
Default is 86400.

Connection Request Username The user name for the ACS to connect to the phone.

Connection Request Password The password for the ACS to connect to the phone.

Connection Request Port 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.

When enabled, this option allows users to randomize the sending of TR069 INFORM
Randomized TR069 Startup
packets.

Maintenance 🡪 Security Settings🡪 Security

Configures the access control for the users to configure from keypad Menu. There are
three different options:

● Unrestricted: All the options can be accessed in keypad Menu.

● Basic settings only: The SIP option under Phone submenu, and Network, Upgrade,
UCM Detect and Factory Reset options under System submenu will not be
Configuration via Keypad Menu available in LCD Menu.
● Constraint Mode: The phone will require administration password to change the
Network, Upgrade and Factory Reset options under System submenu, and SIP
option under Phone submenu as well.
● Locked Mode: The phone menu and changing MPK/VPK/Line are disabled. The
default setting is “Unrestricted”.

This feature allows users to decide whether or not to disable password request when
performing factory reset with hard keys.
There are three options:

● Default: Password is needed when “configuration via keypad menu” is not

Factory Reset Security Level “Unrestricted”.
● Always Require password: Password is needed no matter what “configuration via
keypad menu” mode is.
● No password: No password is needed no matter what “configuration via keypad
menu mode” is.

After enabling this feature, phone will validate the server’s certificate. If the server that
Validate Server Certificates our phone tries to register on is not on our list, it will not allow server to access the
phone.

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 need to press and hold the STAR * key for
about 4 seconds again and then enter the password to unlock it.

SIP TLS Certificate SSL Certificate used for SIP Transport in TLS/TCP.

SIP TLS Private Key SSL Private key used for SIP Transport in TLS/TCP.

SIP TLS Private Key Password SSL Private key password used for SIP Transport in TLS/TCP.

The uploaded custom certificate will be used for SSL/TLS communication instead of
Custom Certificate
the GXP phone default certificate.

Web Access Mode Sets the protocol for web interface. The default setting is “HTTP”.

HTTP Web Port Configures the HTTP port under the HTTP web access mode.

Configures the HTTPS port under the HTTPS web access mode. Default setting is
HTTPS Web Port
“443”.

Disable SSH Disables SSH access. The default setting is “No”.

This feature allows users to customize the SSH access port to connect to the phone.
SSH Port
Default settings is “22”.

This option allows you to use authentication keys for SSH access. The public key
should be loaded to phone’s web UI while the private key should be used in the SSH
SSH Public Key
tool side.
Note: This will allow upcoming SSH access without password.

Specifies the time in minutes that the web or LCD login interface will be locked out to
Web/Keypad/Restrict mode Lockout
user after five login failures. This lockout time is used for web login, STAR keypad
Duration
unlock, and LCD restrict mode admin login. Range is 0-60 minutes.

Configures timer to logout web session during idle. Default is 10 min. Range is 2-60
Web Session Timeout
min.

Web Access Attempt Limit Configures attempt limit before lockout. Default is 5. Range is 1-10.

The function allows users to choose minimum TLS version for HTTPS provisioning
and SIP transport. This setting requires reboot to take effect on HTTPS web access.
Minimum TLS Version
Provisioning and sip transport don’t need reboot.
Default value is “TLS 1.1”

The function allows users to choose maximum TLS version for HTTPS provisioning
and SIP transport. This setting requires reboot to take effect on HTTPS web access.
Maximum TLS Version
Provisioning and sip transport don’t need reboot.
Default value is “Unlimited”

This feature could force the TLS version/Cipher suites for HTTPS provisioning and the
TLS version for sip transport (TLS/TCP) and HTTPS web access.

● Enable Weak TLS Ciphers Suites

Enable/Disable Weak Ciphers ● Disable Symmetric Encryption RC4/DES/3DES
● Disable Symmetric Encryption SEED
● Disable All Weak Symmetric Encryption
● Disable Symmetric Authentication MD5
● Disable All Weak TLS Cipher Suites

Maintenance 🡪 Security Settings🡪 Trusted CA Certificates

Allows to upload and delete the CA Certificate file to phone.

Trusted CA Certificates Note: Users can either upload the file directly from web or they can choose to
provision it from their cfg.xml file.

Users are able to specify which certificate they are going to use:

Load CA Certificates ● All Certificates: (Default) Both built-in and uploaded Certificates.
● Default Certificates: Built-in Certificates;
● Custom Certificates: Uploaded Certificates;

Maintenance🡪Security Settings🡪 Keypad Lock

If set to “Yes” the keypad can be locked either manually by pressing for 4 seconds *
Enable Keypad Locking key or pressing a VPK/MPK which set to “keypad lock” mode, also the keypad will be
locked automatically after the configured timer.

If set to “Functional Keys” then only functional keys will be locked and users still are
Keypad Lock Type
allowed to dial configured emergency numbers.
Password to Lock/Unlock Set the password to Unlock the keypad.

Keypad Lock Timer Configure idle screen timer after which the keypad will be locked.

Enter list of allowed emergency numbers when keypad is locked (separate the
Emergency
numbers with “,”.

Maintenance 🡪 Packet Capture

Displays packet capture status. When user starts to capture trace file, it will show
Status
“RUNNING” status, otherwise, it will show “STOPPED”.

Capture Location Location where the capture will be stored, either “Internal Storage” or “USB”

Defines whether the packet capture file contains RTP or not. The default is No.
Note: GXP21xx supports sending RTP events for hook flash, enhancing call control
With RTP Packets and telephony features, this update allows the system to better interpret and respond
to "hook flash" actions, improving the overall user experience and expanding the
possibilities for call management and features within the system.

When set to "Yes", the downloaded packet will include a secret key, to decrypt the
captured TLS packet
With Secret Key Information
Note: When "With Secret Key Information" is enabled, Packet capture will
automatically stop when the size threshold limit is reached.

USB Filename Filename of the capture. Only required for USB.

Table 13: Maintenance Page Definitions

Phonebook Page Definitions

Phonebook 🡪
Contacts

Search Bar Allows users searching for phonebook entries.

Specifies Contact’s First Name, Last Name, Phone Number, Accounts and Groups Blacklist, Whitelist,
Work, Friends and Family) to add one new contact in phonebook.

Add Contact
Note: If the contact number belongs to Blacklist group, the call from this number will be blocked. If the
contact number belongs to Whitelist group, when the phone is on DND mode, the call from whitelist
number will be allowed.

Edit Contact Edits selected contact.

Deletes all contacts from phonebook.

Delete All
Contacts NOTE: a message prompt will be displayed so that users will confirm to delete or cancel the operation, in
order to prevent users from losing contacts when deleting them accidentally.

Phonebook 🡪
Group
Management

Add Group Specifies Group’s name to add new group. More than 30 Groups can be added.
Edit Group Edits selected group.

Phonebook 🡪
Phonebook
Management

Enable Configures to enable phonebook XML download. Users could select HTTP/HTTPS/TFTP to download the
Phonebook phonebook file.
XML
Download The default setting is “Disabled”.

HTTP/HTTPS
The user name for the HTTP/HTTPS server.
User Name

HTTP/HTTPS
The password for the HTTP/HTTPS server.
Password

Phonebook Configures the server path to download the phonebook XML.

XML Server
Path This field could be IP address or URL, with up to 256 characters.

Configures the phonebook download interval (in minutes).

Phonebook
Download
If set to 0, automatic download will be disabled. The default value is 0. Valid range is 5 to 720 minutes.
Interval
Here’s how to trigger an [Immediate Download]

Remove
Manually- If set to “Yes”, when XML phonebook is downloaded, the entries added manually will be automatically
edited Entries removed. The default setting is “Yes”.
on Download

Import Group When set to “Replace”, existing groups will be completely replaced by imported one; When set to
Method “Append”, the imported groups will be attended with the current one.

Sort phonebook based on the selection of first name or last name.

Sort
Phonebook by
The default setting is “Last Name”.

Download
XML Click on “Download” to download the XML phonebook file to local PC
Phonebook

Upload XML
Click on “Upload” to upload local XML phonebook file to the phone.
Phonebook

Control the behavior of phonebook key. There are five options: Default, LDAP Search, Local Phonebook,
Phonebook
Local Group, and Broadsoft Phonebook. The default setting is “Default”, when user presses it, phone LCD
Key Function
will show the five options.

Default search
Configures default phonebook search mode. Default setting is “Quick match”.
mode

Phonebook 🡪
LDAP

Configures the LDAP protocol to LDAP or LDAPS. The default setting is “LDAP”. LDAPS is a feature to
LDAP Protocol
support LDAP over TLS.
Server Address Configures the IP address or DNS name of the LDAP server.

Port Configures the LDAP server port. The default port number is “389”.

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

Configures the bind “Username” for querying LDAP servers. Some LDAP servers allow anonymous binds
User Name
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
Password
allows anonymous binds.

Configures the filter used for number lookups. Examples:

(|(telephoneNumber=%)(Mobile=%) returns all records which has the “telephoneNumber” or “Mobile”

LDAP Number
field starting with the entered prefix;
Filter

(&(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”.

Specifies the “name” attributes of each record which are returned in the LDAP search result. This field
allows the users to configure multiple space separated name attributes. Example:
LDAP Name
Attributes gn

cn sn description

Specifies the “number” attributes of each record which are returned in the LDAP search result.

This field allows the users to configure multiple space separated number attributes. Example:
LDAP Number
Attributes
telephoneNumber

telephoneNumber Mobile

Configures the entry information to be shown on phone’s LCD. Up to 3 fields can be displayed. Example:
LDAP Display
Name
%cn %sn %telephoneNumber
Specifies the maximum number of results to be returned by the LDAP server. If set to 0, server will return
Max. Hits
all search results. The default setting is 50.

Search Specifies the interval (in seconds) for the server to process the request and client waits for server to
Timeout return. The default setting is 30 seconds.

Sort Results Specifies whether the searching result is sorted or not. Default setting is “No”.

LDAP Lookup Configures to enable LDAP number searching when dialing / 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 Example:
Display Name
gn

cn sn description

With LDAP Lookup Incoming call, Outgoing call selected, DUT will performs LDAP search during
Exact Match incoming and outgoing call. If exact match search enabled, during the LDAP search, DUT will only get the
Search result that matches the search input exactly. i.e. if 100 is the incoming/outgoing number only 100 will get
searched, *100* will not. Default is “disabled”.

Table 14: Phonebook Page Definitions

BLF LED Patterns

Pattern: Default Pattern: Analog

Call’s state Light Indication Call’s state Light Indication

Offline Off Offline Off

Idle Solid Green Idle Solid Green

Trying Solid Red Trying Solid Red

Talking Solid Red Talking Solid Red

Proceeding Flashing Red Proceeding Solid Red

Incoming call Flashing Red Incoming call Flashing Red

Pattern: Directional Mode: Inverse

Call’s state Light Indication Call’s state Light Indication

Offline Off Offline Off

Idle Solid Green Idle Solid Red

Trying Flashing Green Trying Solid Green

Talking Solid Red Talking Solid Green

Proceeding (Initiator) Flashing Green Proceeding Flashing Green

Proceeding (Receiver) Flashing Red Incoming call Flashing Green

Incoming call Flashing Red

Mode: Reserved Mode: Reserved

(Red) (Green)

Call’s state Light Indication Call’s state Light Indication

Off (Extension Board Icon: Off (Extension Board Icon:

Offline Offline
Off) Off)

Off (Extension Board Icon: Off (Extension Board Icon:

Idle Idle
Idle) Idle)

Trying Solid Red Trying Solid Green

Talking Solid Red Talking Solid Green

Proceeding Solid Red Proceeding Solid Green

Incoming
Incoming call Flashing Red Flashing Green
call

Table 15: BLF LED Patterns

Eventlist BLF Listening Transport Protocol

Web Configuration

User can find the new option at Web GUI🡪Accounts(x) 🡪SIP Settings🡪 Basic Settings.
Figure 3: SIP Listening Mode

Functionality

Based on option “SIP Transport” and new option “SIP Listening Mode”, GXP will decide which transport protocol it should
listening to from the incoming request.

SIP Listening
Mode / SIP UDP TCP TCP/TLS
Transport

Accept incoming
request using
Accept incoming request using UDP. Accept incoming request using TCP.
TLS/TCP.
Transport Only All outgoing request will go out using All outgoing request will go out using
All outgoing
UDP. TCP.
request will go out
using TLS/TCP.

Accept incoming request using both Accept incoming request using both
TCP and UDP. TCP and UDP.
Dual –
All outgoing request will go out using All outgoing request will go out using
UDP. TCP.
SIP Listening
Mode / SIP UDP TCP TCP/TLS
Transport

Accept incoming
Accept incoming request using both request using both
TLS/TCP and UDP. TLS/TCP and UDP.
Dual (Secured) –
All outgoing request will go out using All outgoing
UDP. request will go out
using TLS/TCP.

Accept incoming request using both Accept incoming request using both
TCP and UDP. TCP and UDP.
Dual (BLF All outgoing request will go out using All outgoing request will go out using
–
Enforced) UDP except for the BLF/Eventlist TCP except for the BLF/Eventlist
subscription the phone will add subscription the phone will add
Transport=TCP into the contact header. Transport=TCP into the contact header.

NAT Settings

If the devices are kept within a private network behind a firewall, we recommend using STUN Server. The following settings
are useful in the STUN Server scenario:

STUN Server

Under 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.

Dial Plan Configuration

Dial plan sets the rules to manage outgoing calls, in order to allow or block some type of calls or change the number format
before dialing out. Users can configure dial plan rules either under web GUI menu “Account X 🡪 Call Settings 🡪 Dial Plan”
or through a simpler and well-designed interface under menu “Account X 🡪 Dial Plan”.

For explanation purposes, we will be using the dial plan user interface.

Figure 4: Dial Plan Configuration

The current interface features are as follow:

1. Name: Users can name their dial plans for identification.

2. Rule: The rules can be typed out separately or in combination with “Type”

3. Type: We now support the following types.

Pattern: The general rule and it will not change the dial plan you configured.

Block: The rules you set in combination with this type will be blocked.

Dial Now: The rules you set in combination with this type will be dialed out once the DTMF matches the Dial Plan.

Prefix: The rules you set in combination with this type will include configured prefix automatically. If Replaced was set,
your used prefix will replace the “Replaced” value.

For example: If Dialed 3456, the DTMF will send 123456. See configuration below.

Second tone: The rules you set in combination with this type will play second tone if matching the Trigger.

4. Automatically update the configured data to the Dial Plan in Call Settings.

5. Dial Plan Verification.

Note:

This feature is not supported by config files (both .xml and .txt).

Users can increase or decrease the priority of each Pattern by pressing to move it up and to move it down.

When you input dial plan from Call Settings, it will not automatically choose a type for you. The default type is Pattern.

Entering Dial Plan from Call Settings🡪Dial Plan will cause bypassing the verification.

For more information about how to set a Dial Plan, please refer to Dial Plan Rules.

Blind Transfer and Attended Transfer Softkey

This feature works when option “Attended Transfer” under web UI🡪Call Features is set to “Dynamic”. When the user tries to
transfer an ongoing call, after pressing “Transfer” Softkey and entering the number
to be transferred to, the user will be able to select Softkey “BlindTrnf” for blind transfer or Softkey “AttTrnf” attended transfer.
Figure 5: Transfer Softkey During Call

Figure 6: Blind/Attended Softkeys During Call

Display SIP Message Text on LCD

During an active call, if the phone receives SIP message request that has message body with line-based text data defined, the
content will be displayed on the phone’s LCD. In the following example, the phone LCD will display “Total $5” as defined in the
SIP message text.

Figure 7: Display SIP Message Text on LCD

The option Enable IM POPUP should be enabled web UI-> Settings -> Call Features to show the instant messages on screen.

Link Command

The Link allows user to have an overview about the port status, speed, Duplex mode, and Auto negotiation.

Figure 8: Link Command

TLS Negotiation

TLS (transport layer security) is a common protocol, which provides privacy to your communication. It will also manage the
communication between IP phones to prevent the communications from tampering each other.

The GXP21XX series support TLS 1.0 (RFC2246), 1.1 (RFC4346), and 1.2 (RFC5246)

Editing Contacts and Click-To-Dial

Click-To-Dial
From GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 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. If no account registered, the icon will be in grey ; If click to dial is disabled, but account is registered, the icon
will be in green, and clicking on the icon will do nothing.

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&login=admin&password=admin

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.

password=admin/123:

The admin login password or user login password of phone’s Web GUI.

Figure 10: Click To Dial feature

Edit contacts

Users can navigate under the web GUI menu « Phonebook 🡪Contacts » and edit all the related settings to each contact. The
following fields are available for configuration:

First Name.

Last Name.

Favorite.

Company

Department.

Job.

Job Title.
Work.

Home.

Mobile.

Account.

Groups

Ring Tone (Set specific ring tone for the contact).

Picture.

Note: for the ring tone, currently only .wav file is supported. Users can upload their customized .wav files as custom ringtones.
(File size and format are restricted to 500KB or less.)

Figure 11: Edit contacts

Immediate Download

Once the Phonebook download is enabled, three ways would make the phone trigger the download:

The download Soft key:

Go to the phone’s Local phonebook and press the Download Softkey.

Phonebook Download Interval:

After each time the interval set for “Phonebook Download Interval” passes, the phones will download the phonebook.

Sending a SIP NOTIFY with “Event: sync-contacts” header:

– If a SIP account is registered on the phone, use the Authentication credentials.
– If no SIP account is registered on the phone: use Admin access credentials.

Send Instant Message

Instant messages are used to send text between IP Phones via SIP messages.

The GXP2130/2140/2160/2170/2135 allow users to send instant message with the instant message feature on the top of
the Web GUI as shown in the following figure.
Figure 12: Instant Message

Clicking on , will show the following pop up.

Select the account from where to send the message.

Select the number where to send the number.

Enter the content of the instant message.

Press button to send the message.

Figure 13: Send Instant Message

Wallpaper

GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 support 4 approaches for wallpaper configurations: “Default”, “Download”,

and “Upload” and “Color Background”.
GXP2140/GXP2160/GXP2170 also provide loading wallpaper file from USB drive.

Default Mode

Under Default mode, the phone will display the wallpaper supplied by firmware.

Download Mode

Under Download mode, the phone will download the wallpaper from the specified server path under “Wallpaper Server Path”
option. The Wallpaper Server Path option will take effect only when Download mode specified. See Figure 5 Download
wallpaper from server. The server path must begin with tftp:// or http:// or https://, otherwise, phone will assume HTTP
mode.

Figure 14: Download Wallpaper from Server

USB Mode (For GXP2140/GXP2160/GXP2170 only)

Under USB mode, when USB drive is connected on phone, it will look for a wallpaper.jpg file under the USB root directory. If
no such file found, phone will display default wallpaper.

Uploaded Mode

Under uploaded mode, user can browse and upload a .jpg or .jpeg format wallpaper file. The image must be smaller 500 KB.
See [Figure 9: Web Service].

Figure 15: Upload Selected Wallpaper to Phone

Color Background Mode

Users could find option “Color Background” under web UI🡪Settings🡪LCD Display: Wallpaper category. Enter any HEX color
code based on your preference. The color codes could be found here:

http://htmlcolorcodes.com/

If an invalid code is configured, the phone will use default value #000000 instead.

Figure 16: Wallpaper – Color Background Mode

Please note that the user must select “Color Background” in “Wallpaper Source” option in order to use the configurable color
background code.
Figure 17: Wallpaper Source

Example:

If the user users default color code #000000, the idle screen will load “black” as background. This color will also affect MENU
configuration page.

Contact Picture Support

The GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 supports adding pictures to each account, this can be done by
navigating on the webGUI under “Accounts > Account X > General Settings”.

Click “Select” under “Picture” as shown below.

Figure 18: Select Picture

The following window will pop up to select from where to upload the picture, from local disk or set a URL to the picture.

Figure 19: Upload Picture

Click “Save and Apply” after choosing the picture.

During the call, the callee will see the picture/icon that the caller sets. Users can find the Call-Info header that contains the jpg
file from sip messages as shown below. (Currently only support openser)

Figure 20: Picture Call-Info Header

Screensaver Pictures Downloading

GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 support downloading screensavers from HTTP/TFTP servers.

Please refer to following configuration steps:

1. Log into Web GUI > Settings > LCD Display > Screensaver.

2. Set Screensaver Source to “Download”.

3. Enter the following Path on Screensaver Server Path:

http://Server_IP/screensaver.xml or tftp://Server_IP/screensaver.xml

Figure 21: Screensaver Settings

4. On screensaver.xml file enter following tags:

5. Put picture files on HTTP server directory. Please refer to following example using HFS HTTP server:

Figure 22: HFS HTTP Server

6. Press Save and Apply button to save the new configuration

Saving 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 “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.

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. GXP2130v2/GXP2135/2140/GXP2160/GXP2170 supports
Bluetooth. On GXP2130v2/GXP2135/2140/GXP2160/GXP2170, users could connect to cellphones (supporting Bluetooth) via
hands free mode or use Bluetooth headset for making calls.

To connect to a Bluetooth device, turn on GXP2130v2/GXP2135/2140/GXP2160/GXP2170’s Bluetooth radio first. The first time
when using a new Bluetooth device with the GXP2130v2/GXP2135/GXP2140/GXP2160/GXP2170, “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 GXP2130v2/2140/GXP2160/GXP2170’s LCD Menu🡪System🡪Bluetooth. GXP2130v1

does not support Bluetooth function, only GXP2130v2 supports Bluetooth, you could differentiate by P/N as well as by FCC ID.

For more details on Bluetooth features, please refer to:

https://documentation.grandstream.com/knowledge-base/how-to-use-bluetooth-functionality/

Packet Capture

GXP2130/GXP2135/GXP2140/GXP2160/GXP2170 is embedded with packet capture function on firmware 1.0.4.10 or greater.

The related options are under Maintenance🡪Packet Capture.

Figure 23: Packet Capture in Idle

User can also define whether RTP packets will be captured or not from With RTP Packets option.

When the capture configuration is set, press Start button to start packet capture. The Status will become RUNNING while
capturing, as showed in Figure 24: Packet Capture when running. Press Stop button to end capture.

Press Download button to download capture file to local PC. The capture file is in .pcap format.
Click on clear, to clear the traces previously captured so you don’t have the repetitive files downloaded.
Figure 24: Packet Capture when running

Screenshots

Users can take screenshots of the GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 phones, by holding key HOLD and then
pressing MENU key, the output will be shown on the phone webGUI under “Status 🡪 System Info” as shown in the figure
below.

Figure 25: Screenshots

Users need to click on “Download” in order to view the screenshot.

Multicast Paging

GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 supports multicast paging, including sending and listening. On the phone,
users could send multicast page by setting the multicast address and port. Also, users can listen to at most 10 different
multicast IP address.

Multicast sender related settings are under Web UI, Settings🡪Programmable keys. Select Multicast paging as the key mode
for dial page call. Multicast paging listening related settings are under Web UI Settings🡪Multicast Paging.

For more details on Multicast paging features, please visit http://www.grandstream.com/support to download the latest
“GXP2130/GXP2140/GXP2160 Multicast Paging User Guide”.

Configuring Eventlist BLF

Grandstream GXP2130/2140/2160/2170/2135 Enterprise IP Phones support both Grandstream UCM Busy Lamp Filed and
Event List BLF features and allows end users, such as attendant, to monitor the call status of users in the list.
GXP2130/2140/2160/2170/2135 supports this feature by sending out the subscription request to the UCM and changing the
indicator status of the Line keys, MPKs, or virtual MPKs that associated with the monitored users. Additionally, the phone is
also able to pick up the calls to the monitored extensions by using a pre-defined feature code called BLF- Call-pickup Prefix.

For more details on Eventlist BLF configuration guide, please refer to:
https://documentation.grandstream.com/knowledge-base/how-to-configure-eventlist-blf/

Outbound Notification Support

Outbound notification options can be found under device web UI🡪Settings🡪Outbound Notifications. In the web UI, there
are three sections under Outbound Notifications: “Action URL”, “Destination” and “Notification”.

Action URL

To use Outbound Notification🡪Action URL, users need to know the supported events and the dynamic variables for the
supported events. The dynamic variables for the supported events will be replaced by the actual values on the phone in order
to notify the event to SIP server.

Figure 26: Action URL Settings Page

Supported Events

Setup Completed

Registered

Unregistered

Off Hook

On Hook
Supported Events

Incoming Call

Outgoing Call

Missed Call

Established Call

Terminated Call

Open DND

Close DND

Open Forward

Close Forward

Blind Transfer

Attended Transfer

Hold Call

UnHold Call

Table 16: Action URL – Supported Events

Supported Dynamic Variables Description

$phone_ip The IP address of the phone

$mac The MAC address of the phone

$product The product name of the phone

$program_version The software version of the phone

$hardware_version The hardware version of the phone

$language The display language of the phone

$local The called number on the phone

$display_local The display name of the called number on the phone

$remote The call number on the remote phone

$display_remote The display name of the call number on the remote phone

$active_user The account number during a call on the phone

Table 17: Action URL – Supported Dynamic Variables

After the user finishes setting Action URL on phone’s web UI, when the specific phone event occurs on the phone, phone will
send the Action URL to the specified SIP server. The dynamic variables in the Action URL will be replaced by the actual values.

Here is an example:

Configure the following Action URL on the phone’s web UI🡪Settings🡪Outbound Notification🡪Action URL:

Incoming Call: 172.18.24.103/mac=$mac&local=$local

Outgoing Call: 172.18.24.103/remote=$remote&phone_ip=$phone_ip

On hold: 172.18.24.103/program_version=$program_version

During incoming call, outgoing call and call hold, capture the trace on the phone and exam the packets. We can see the
phone send Action URL with actual values to SIP server to notify phone events. In the following screenshot, from top to
bottom, the phone events for each HTTP message are: Outgoing Call, Incoming Call and On Hold in the format of the defined
action URL with the parameters replaced with actual values.

Figure 27: Action URL Packet

The P values listed in below table are for the options under phone web UI🡪Settings🡪Outbound Notification🡪Action URL.

P-Value Web UI Option Value Format

P8304 Setup Completed String

P8305 Registered

P8306 Unregistered

P8308 Off Hook

P8309 On Hook

P8310 Incoming Call

P8311 Outgoing Call

P8312 Missed Call

P8313 Established Call

P-Value Web UI Option Value Format

P8314 Terminated Call

P8316 Open DND

P8317 Close DND

P8318 Open Forward

P8319 Close Forward

P8320 Blind Transfer

P8321 Attended Transfer

P8324 Hold Call

P8325 UnHold Call

Table 18: Action URL Parameters P-values

Destination

The options under phone’s web UI🡪Settings🡪Outbound Notification🡪Destination configures the server information
destination of the outbound notification. Click on “Add Destination” and users will see following window to configure
destination server information.

Figure 28: Action URL – Add Destination

The following table describes each option in the above interface.

Destination Server
Description
Option

Destination Name Identify the destination name. It must be unique.

Configure the protocol associated with the destination server. Currently XMPP and SMTP are
Protocol
supported.

Enable SSL Configure whether to use SSL to encrypt for SMTP protocol. This option is not editable for XMPP.

Destination
Configure destination server address, e.g., talk.google.com.
Address

Port Configure destination server port, e.g., 5222.

Domain Configure the destination server domain for XMPP protocol. This option is not editable for SMTP.

User Name Configure the authorization user name of the destination server.

Password Configure the authorization user password for the destination server.

From Configure the sender name for SMTP protocol. This option is not editable for XMPP.

To Configure the receiver’s address.

Extra Attribute Configure extra attribute’s name reserved for protocol specific attributes such as “jid” for XMPP
Name protocol. If “jid” is specified, user name and domain will be overridden.

Extra Attribute Configure extra attribute’s value reserved for protocol specific attributes such as “[email protected]”
Value for “jid” of XMPP protocol. If it’s specified, user name and domain will be overridden.

Table 19: Action URL – Add Destination Settings

Up to 10 destinations can be configured here. The P-values are listed in below table.

P Desti
Val natio Value Format
ue n

String. Each P value consists of all the options configured for this destination.

Example 1 – Destination 1 with protocol XMPP and 2 extra Attributes configured:

P9910=serverName=destination1&protocol=XMPP&serverAddress=talk.google.com&port=5222&user=user
name1&password=password1&from=&to=to1&domain=gmail.com&extraAttrName1=extraAttrValue1&extr
P9 Destin aAttrName2=extraAttrValue2
91 ation
0 1 Example 2 – Destination 2 with protocol SMTP and 3 extra Attributes configured:

P9911=serverName=destination2&protocol=SMTP&serverAddress=smtps://smtp.gmail.com&port=465&use
r=username2&password=password2&from=username2&to=to2&domain=&extraAttrName1=extraAttrValue
1&extraAttrName2=extraAttrValue2&extraAttrName3=extraAttrValue3

The highlighted strings in above examples are the actual values configured in each field for the destination.

P9 Destin
91 ation
1 2
P9 Destin
91 ation
2 3

P9 Destin
91 ation
3 4

P9 Destin
91 ation
4 5

P9 Destin
91 ation
5 6

P9 Destin
91 ation
6 7

P9 Destin
91 ation
7 8

P9 Destin
91 ation
8 9

P9 Destin
91 ation
9 10

Table 20: Action URL – Destination P-values

Notification

After configuring destination server, users can configure notification information under phone’s web
UI🡪Settings🡪Outbound Notification🡪Notification. Click on “Add Notification” and users will see following window to
configure notification.
Figure 29: Action URL – Add Notification

Notification Option Description

Event Configures the event, which will trigger an outbound notification.

Destination Configures the name of the destination where the outbound notification will be sent to.

Configures the subject of Email notification. This option is only applicable to SMTP protocol and it is
Subject
not editable for other protocols.

Message Configures the message body or the outbound notification.

Extra Attribute
Configure extra attribute’s name reserved for specific attributes for a given notification in the future.
Name

Extra Attribute
Configures extra attribute’s value reserved for specific attributes for a given notification in the future.
Value

Table 21: Action URL – Notification Options

The message body of the notification for each event can be customized with dynamic attributes embedded. The following
table shows the mapping between event and dynamic attribute.

Event Dynamic Attribute Name Dynamic Attribute Description

Line number associated with the

Call_Missed line
call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

Event Dynamic Attribute Name Dynamic Attribute Description

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

The time stamp when the missed call event

time
occurs

This is for DND status. The value

DND status
can be “enabled” or “disabled”

This is for the type of the call. The

Call_Forward callType value can be “incoming” or
“outgoing”

line Line number associated with the call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

time The timestamp when the call is forwarded

fwNumber Call Forward destination number

fwReason Call Forward reason

OAM_Login OAMUser OAM user name such as “admin”

OAM login source. The value can be “SSH”

OAMLoginSource
or “WebGUI”

OAM login From IP address. The value is the

OAMLoginFromIP IP address of the PC who will log in phone’s
web UI or SSH

OAM login result code. The value can be

OAMLoginCode
“succeeded” or “failed”

time OAM login time stamp

OAM_Lockout OAMUser OAM user name such as “admin”

Event Dynamic Attribute Name Dynamic Attribute Description

OAM login source. The value can be “SSH”

OAMLoginSource
or “WebGUI”

OAM login From IP address. The value is the

OAMLoginFromIP IP address of the PC who will log in phone’s
web UI or SSH

OAM lockout result code. The value can be

OAMLockoutCode
“locked” or “unlocked”

OAMLockoutTime OAM lockout time stamp

Incoming_Call callingNumber Calling party number

Type of the call. The value can be

callType
“incoming” or “outgoing”

line Line number associated with the call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

The time stamp when the incoming call

time
event occurs

Type of the call. The value can be

Outgoing_Call callType
“incoming” or “outgoing”

line Line number associated with the call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

Event Dynamic Attribute Name Dynamic Attribute Description

The time stamp when the outgoing call

time
event occurs

Type of the call. The value can be

Call_Established callType
“incoming” or “outgoing”

line Line number associated with the call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

The time stamp when outgoing call event

startTime
occurs

Type of the call. The value can be

Call_Terminated callType
“incoming” or “outgoing”

line Line number associated with the call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

startTime The time stamp when the call is established

Call_Forward_Status duration The call duration

The account number associated with the call

account
forward status change

forwardNumberAll The forward number for Call Forward All

Event Dynamic Attribute Name Dynamic Attribute Description

forwardNumberBusy The forward number for Call Forward Busy

The forward number for Call Forward No

forwardNumberNoAns
Answer

Type of the call. The value can be

Call Hold callType
“incoming” or “outgoing”

line Line number associated with the call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

startTime The time stamp when the call is on hold

Type of the call. The value can be

Call_Resume callType
“incoming” or “outgoing”

line Line number associated with the call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

startTime The time stamp when the call is resumed

Line number associated with the

Blind_Transfer line
call

account Account number associated with the call

remoteNumber The remote party number

Event Dynamic Attribute Name Dynamic Attribute Description

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

time The time stamp when the call is transferred

transferName Transferred party name

transferNumber Transferred party number

Line number associated with the

Attended_Transfer Line
call

account Account number associated with the call

remoteNumber The remote party number

remoteName The remote party name

localNumber The local party number

localName The local party name

sipServer The SIP server address of the account

call-id The SIP dialog call ID

Time The time stamp when the call is transferred

transferName Transferred party name

transferNumber Transferred party number

Account register status. The value

Register_Status registerStatus can be “registered” or
“unregistered”

Bootup_Complete N/A N/A

The dynamic attributes in this row

are common attributes that can be mac MAC address of the phone
applied to all events

phone_ip IP address of the phone

program_version Software version of the phone

Event Dynamic Attribute Name Dynamic Attribute Description

hardware_version Hardware version of the phone

product Product name of the phone

language Display language on the phone

Table 22: Action URL Notification – Events and Dynamic Attributes

All above dynamic attributes’ value is generated by phone system and can be used as dynamic attributes with a pair of curved
braces around them. For example, if the message body is specified as following:

Your call from {remoteName}:{remoteNumber} to {localName}:{localNumber} was forwarded to {fwdNumber} by reason

{fwdReason}.

Then the message received in the outbound notification will look like this:

Your call from Daniel:2070 to Jasmine:2071 was forwarded to 777777 by reason unconditional.

Only attributes in curved braces will be replaced by the run time value. Other content will remain the same as static text.

For each event, at most 3 notifications can be configured. In total, up to 75 notifications can be configured. The P value for
each notification is listed in below table.

P-Value Notification Value Format

String. Each P value consists of all the options configured for this notification.

Example 1 – Notification 1 for event “Call_Missed” to destination 1, with 2 extra Attributes

configured:

P9920=eventName=Call_Missed&destName=destination1&subject=&msg=You have a missed

call from {remoteName}:{remoteNumber} on Line {line}, account {account} at
{time}.&extraAttrName1=extraAttrValue1&extraAttrName2=extraValue2
Notification
P9920
1 Example 2 – Notification 2 for event “Incoming_Call” to destination 2, with 2 extra Attributes
configured:

P9921= eventName=Incoming_Call&destName=destination2&subject=Incoming Call

Alert&msg=You have an {callType} call from {remoteName}:{remoteNumber} on Line {line},
account {account} at {time}.&extraAttrName1=extraAttrValue1&extraAttrName2=extraAttrValue2

The highlighted strings in above examples are the actual values configured in each field for the
notification.

Notification
P9921
2

Notification
P9922
3

Notification
P9923
4

Notification
P9924
5

Notification
P9925
6
P-Value Notification Value Format

Notification
P9926
7

Notification
P9927
8

Notification
9

P9928
Notification
P9929
10

…
…

P9993
Notification
73
P9994

Notification
74

Notification
P9995
75

Table 23: Action URL Notification P-values

Virtual Multi-Purpose Keys

Web UI Configuration

Users can find new Virtual Multi-Purpose Keys (VPK) configuration under phone’s web UI🡪Settings🡪Programmable
Keys🡪Virtual Multi-Purpose Keys tab. It is recommended to select “Reset” on this page before configuring VPK here. By
default, all fixed VPKs are listed.

Figure 30: VPK Page

Click on “Edit VPK” for the line (fixed VPK) you would like to configure. A new window will pop up for VPK configuration. Users
can configure Mode, Account, Description and Value for the VPK.
If the VPK Description is set, it will show the description on the LCD screen. If the Description is left empty, Default value will
be Account name.

Up to 20 mode options can be selected for the VPK. Once done, press “Save” on this window and press “Save VPK” on the
bottom of the Virtual Multi-Purpose Keys page again to apply the change.

Figure 31: Edit VPK – Fixed VPK

If users would like to configure more VPKs than the ones displayed on the page, the users can click on “Add VPK” to configure
dynamic VPK. The dynamic VPK supports up to 17 mode options.
Figure 32: Edit VPK – Dynamic VPK

Please note:

1. Dynamic VPK does not support LINE and Shared LINE mode. These two mode options are only available for fixed VPKs.

2. Dynamic VPK does not support NONE mode. If users do not need this VPK, click on “Edit VPK” for it and select “Delete” to
remove this VPK.

3. All settings require user to click on “Save” on the prompted window and “Save VPK” button on the bottom of Virtual
Multi-Purpose Keys page to take effect.

P-Value for VPK Mode in String Format

Mode Name Mode String Mode P-Value

None None -1

Default Line 0

Shared Line Shared line 1

Speed Dial Speed dial 10

Busy Lamp Field BLF 11

Presence Watcher presencewatcher 12

Eventlist BLF eventlistblf 13

Speed Dial via Active Account speeddialaa 14

Dial DTMF dialdtmf 15

Voice Mail voicemail 16

Call Return callreturn 17

Transfer transfer 18

Call Park callpark 19

Intercom Intercom 20

LDAP search Ldap Search 21

Conference Conference 22

Multicast Paging Multicast paging 23

Record Record 24

Call Log Call log 25

Monitored Call Park Monitoredcp 26

Menu Menu 27
Mode Name Mode String Mode P-Value

XML Application Xmlapp 28

Information Information 29

Message message 30

Table 24: VPK Modes in String Format

The string could be capital or lower-case letters, but there must be no “space” in between. For example, in the cfg.xml,
“Transfer” or “transfer” is the same as “18”, it will configure Virtual Multi-Purpose Key 3 as transfer mode.

Figure 33: Line Key as Transfer Mode

LCD Indication and Configuration

The configured fixed VPKs are displayed next to the corresponding line. If dynamic VPKs are configured, the users can see a
page number shown on the upper left corner on the LCD. The following figures show page 1 and page 2 of the VPKs on LCD.
Pressing “RIGHT” arrow key or “Next” Softkey will switch to the next page; pressing “LEFT” arrow key will switch back to the
previous page.

Figure 34: VPK – LCD Indication

The users could also edit and add VPK from LCD.

1. To edit (fixed) VPK, press and hold the line key for about 4 seconds, a configuration window will pop up for the user to
configure.

2. To add (dynamic) VPK, press and hold the RIGHT arrow key for about 4 seconds, a configuration window will pop up for
the user to configure.
Up to 20 modes can be supported on fixed VPK and up to 17 modes can be supported on dynamic VPK. Each mode is
indicated by a different icon on the LCD and the icon will be different when in different status.

Please find the icon indications below for different mode of VPK.

LED
VPK Mode State Icon
Status

Unregistered
LINE (No IM, Voice mail, No Call OFF
Forward)

Registered + Idle
OFF
(No IM, Voice mail, No Call Forward)

Unregistered + IM + Voice mail OFF

Registered + IM + Voice mail OFF

Unregistered + IM (No Voice mail) OFF

Registered + IM (No Voice mail) OFF

Unregistered + Voice Mail (No IM) OFF

Registered + Voice Mail (No IM) OFF

Unregistered + Call Forward All

OFF
(No IM, No Voice Mail)

Registered + Call Forward All

OFF
(No IM, No Voice Mail)

Unregistered + Call Forward Delay + Call Forward Busy

OFF
(No IM, No Voice Mail)

Registered + Call Forward Delay + Call Forward Busy

OFF
(No IM, No Voice Mail)

Unregistered + Call Forward Delay

OFF
(No IM, No Voice Mail, No Call Forward Busy)

Registered + Call Forward Delay

OFF
(No IM, No Voice Mail, No Call Forward Busy)

Unregistered + Call Forward Busy

OFF
(No IM, No Voice Mail, No Call Forward Delay)

Registered + Call Forward Busy

OFF
(No IM, No Voice Mail, No Call Forward Delay)

Registered + Ringing Flashing RED

LED
VPK Mode State Icon
Status

Registered + On Hold Flashing GREEN

Registered + Connected + Incoming Call GREEN

Registered + Connected + Outgoing Call GREEN

Shared Line Unregistered OFF

Registered + Not support SCA Call-info header OFF

Registered + Not support SCA or SCA Failed OFF

Registered + Idle OFF

Registered + Seized RED(Alternate DUT)

Flashing
Registered + Processing
GREEN(Alternate DUT)

Registered + Alert Flashing RED

Registered + Hold by user Flashing GREEN

Registered + Hold by the other party Flashing RED

Registered + Connected GREEN

BLF/
Offline, Unknown OFF
Eventlist BLF

Terminated GREEN

Proceeding RED

Ringing (Early) Flashing RED

Trying Flashing GREEN

Confirmed RED

Presence Watcher Offline, Unknown OFF

LED
VPK Mode State Icon
Status

Available GREEN

Handsfree Unpair OFF

Paired, but not connected OFF

Connected OFF

Speed Dial Account Unregistered OFF

Account Registered OFF

Speed Dial Via Active Account OFF

Dial DTMF OFF

Call Return OFF

Transfer Account Unregistered OFF

Account Registered OFF

Call Park Account Unregistered OFF

Account Registered OFF

Intercom Account Unregistered OFF

Account Registered OFF

LDAP Search OFF

Multicast Paging OFF

Record Idle OFF

Recording Flashing

Call Log OFF

LED
VPK Mode State Icon
Status

Menu OFF

Voice Mail Account not registered OFF

Account Registered
OFF
(No new voice mail)

Account Registered
OFF
(Have new voice mail)

Table 25: VPK Modes Icon Indications

Please note that no matter how each line is configured on the idle screen, all the lines in call screen will keep line or shared
line displayed for the corresponding accounts. For example, even if the user has configured all lines as VPK (with non-LINE
mode), he/she can still use the configured account to dial out by offhook or pressing SPEAKER, HEADSET or any other
unconfigured LINE key to go to call screen.

Figure 35: Dial Screen

When the user is in call screen (during a call), he/she can press Softkey to switch back to VPK screen.

Figure 36: Making Call

When the user is in VPK screen during a call, he/she can press Softkey or corresponding line key to switch back to call
screen.

Notes :

If a call is parked via VPK call park, the display on the VPK will change between CID of the active call and the parking
number.

When changing the VPK information that requires subscription, the phone will perform unsubscribe first, then perform a
new subscription. This way server will know that previous subscription has been void.
Programmable Keys Status On Web GUI

Users could access programmable key status under phone’s web UI🡪Status.

Web UI🡪Status🡪Programmable Keys Status Virtual Multi-Purpose Keys

Multi-purpose Keys

Web UI🡪Status🡪Extension Boards Status Extension 1 keys

Extension 2 keys

Extension 3 keys

Extension 4 keys

Select the tab you would like to check the status; the status of the specific keys will display. The screenshot below shows
virtual Multi-purpose keys status.

Figure 37: VPK Status

PAI Update for CallPark VPK/MPK

The phone will update the number in call history regarding the PAI that it receives from the server.

For instance, when your number is parked in the CallPark space, and the CallPark space has been set into a VPK/MPK, if the
VPK/MPK is used to retrieve the call, the number will be updated in the call history. However, if the VPK/MPK is not used and
a call is made directly into the CallPark space, the number will not be updated in the call history. In both cases, the number will
be updated in the talking states. When using VPK/MPK to park the call, you will see the dialing number (71) in call history.
Figure 38: VPK/MPK to Park the Call

Figure 39: Dial Park Space directly

Figure 40: Call History Updated if Call is Parked

using MPK/VPK

When parking a call using MPK/VPK it will have the same call leg, therefore the SIP server will send the PAI header that will
update the user number in the call history. While calling to the parking space is considered as a separate call therefore, no
update will be received from the server side, thus the phone will not update the call history.

The figure below shows an example of the PAI header received by the phone in order to update the call history.
Figure 41: Received PAI Header

If a user tries to park a call to an occupied parking lot, the parking process will fail and the conversation will resume.

UPGRADING AND PROVISIONING

The GXP2130 / GXP2140 / GXP2160 / GXP2170 / GXP2135 can be upgraded via TFTP / FTP / FTPS / HTTP / HTTPS by
configuring the URL/IP Address for the TFTP / HTTP / HTTPS / FTP / FTPS server and selecting a download method. Configure
a valid URL for TFTP, FTP/FTPS or HTTP/HTTPS, the server name can be FQDN or IP address.

Examples of valid URLs:

firmware.grandstream.com/BETA

fw.mycompany.com

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” Softkey.

A warning window will be prompt for provision confirmation. Press “YES” Softkey 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.

Shortcut of Upgrade and Provision via Keypad Menu

When GXP phone is in idle state, user could press HOLD key and RIGHT navigation key together to trigger provision functions.
Similarly, phone will pop up reboot banner while idle, if user presses HOLD key and LEFT navigation key together. After the
provision or reboot banner pops up on LCD screen, user could press YES/NO Softkey to confirm/cancel the action.

Upgrade 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 or FTP/FTPS. Update the change by clicking the “Save and
Apply” button. Then “Reboot” or power cycle the phone to update the new firmware.

When upgrading starts, the screen will show upgrading progress. When done you will see the phone 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/FTP/HTTP Servers

For users that would like to use remote upgrading without a local TFTP/FTP/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:

https://www.grandstream.com/support/firmware

Alternatively, users can download a free TFTP, FTP or HTTP server and conduct a local firmware upgrade. A free window
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, FTP/FTPS or HTTP/HTTPS. The “Config Server Path” is the TFTP, FTP/FTPS 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”.

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 5-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
configuration template.

When the GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 boots up or reboots, it will issue a request to download a

configuration file named “cfgxxxxxxxxxxxx” followed by an XML file named “cfgxxxxxxxxxxxx.xml”, where “xxxxxxxxxxxx” is the
MAC address of the phone, i.e., “cfg000b820102ab” and “cfg000b820102ab.xml”. If the download of “cfgxxxxxxxxxxxx.xml” file
is not successful, the phone will issue a request to download a specific model configuration file “cfg<model>.xml”, where
<model> is the phone model, i.e., “cfggxp2130.xml” for the GXP2130, “cfgxp2170” for the GXP2170. If this file is not available,
the phone will issue a request to download the generic “cfg.xml” file. The configuration file name should be in lower case
letters.

Figure 42: Config File Download

Note: (Attempt to download Config File again)

When doing provision on the phone, if your first config file contains p-values listed below, phone will try to download the
potential second cfg.xml file and apply the second file without rebooting. Maximum 3 extra attempts.

Those P-values are:

*212 — Config upgrade via

*234 — Config prefix

*235 — Config postfix

*237 — Config upgrade Server

*240 – Authenticate Config File

*1359 – XML Config File Password

*8463 – Validate Server Certificate

*8467 – Download and process ALL Available Config Files

*20713 – Always authenticate before challenge

*22011 – Bypass Proxy For

*22030 – Enable SSL host verification for provision

Note: (P-values that trigger Auto-Provision)

If the p-values listed below are changed while managing configuration on web UI or LCD, the provision process will be
triggered:

* 192 — Firmware upgrade server

* 232 — Firmware prefix

* 233 — Firmware postfix

* 6767 — Firmware Upgrade Via

* 6768 — Firmware HTTP/HTTPS Username

* 6769 — Firmware HTTP/HTTPS Password

* 237 — Config upgrade Server

* 212 — Config upgrade via

* 234 — Config prefix

* 235 — Config postfix

* 1360 — Config HTTP/HTTPS username

* 1361 — Config HTTP/HTTPS password.

Note: Certificates and Keys provisioning

Users can configure the phone to get all the needed certificates during boot up. Instead of putting the certificate/key content
in text directly from the Web interface or uploading them manually, they can choose to provision them from the configuration
file by putting the URL in the Pvalue field of each certificate and/or key. (e.g. http://ProvisionServer_address/SIP-TLS-
Certificate.pem) The phone will then process the URL, search for the appropriate certificate/Key file, download it and then
apply it into the phone.
Figure 43: Certificates Files Download

For more details on XML provisioning, please refer to:

https://documentation.grandstream.com/knowledge-base/sip-device-provisioning-guide/

No Touch Provisioning

After the phone sends, config file request to the Broadsoft provisioning server via HTTP/HTTPS, if the provisioning server
responds “401 Unauthorized” asking for authentication, the phone’s LCD will prompt a window for user to enter username
and password. Once correct username and password are entered, the phone will send config file request again with
authentication. Then the phone will receive the config file to download and get provisioned automatically.

Besides manually entering the username and password in LCD prompt, users can save the login credentials for provisioning
process as well. The username and password configuration is under phone’s web UI🡪Maintenance🡪Upgrade and
provisioning page: “HTTP/HTTPS Username” and “HTTP/HTTPS Password”. If the saved username and password saved are
correct, login window will be skipped. Otherwise, login window will be popped up to prompt users to enter correct username
and password again.

RESTORE FACTORY DEFAULT SETTINGS

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.

There three methods to perform factory reset on GXP21XX IP phone series which are described below.

Restore to factory using Web GUI

From the web GUI and as shown on the following screenshot, users can either click on the top right link to reset the phone
and wipe the data or click the button at the bottom of the page to lunch the reset.
Figure 44: Factory Reset from web GUI

Restore to factory using hard keys

In order to perform hard reset of the phone using keypad buttons please follow below steps:

1. Power cycle the phone.

2. Wait till you see “booting”.

3. When phone is “booting”, press KEY 1 + Key 9 immediately and hold it until LCD factory reset message or if a password is
required.

4. If it is required, enter the correct admin password to factory reset.

The admin password will be not required to perform factory reset when the option “Configuration via Keypad menu” under web
UI 🡪 Maintenance 🡪 Security is set to “Unrestricted”, otherwise if it’s set to “Basic Settings Only”, or “Constraint Mode”, or
“Locked Mode”, the admin password will be requested. If the password input is correct, phone will perform factory reset; if not,
the phone will reboot without factory reset.

Warning

If the admin password is lost while constraint mode is enabled, your device may become permanently unusable. Remember to
be careful when using constraint mode to avoid irreversible damage.

5. Factory reset is complete.

When users try to factory reset from keypad while booting, the phone will prompt confirmation information to make sure the
action (Press # to Factory Reset or * to cancel). This will avoid people from accidentally reset the phone.
Figure 45 : Confirmation for Factory
Reset

Restore to factory using LCD menu

Please follow the instructions below to reset the phone:

1. Press MENU button to bring up the keypad configuration menu.

2. Select “System” and enter.

3. Select “Operations – Factory Reset”.

4. A warning window will pop out to make sure a reset is requested and confirmed.

5. Press the “Yes” Softkey to confirm and the phone will reboot. To cancel the Reset, press “No” Softkey instead.

CHANGE LOG
This section documents significant changes from previous versions of admin manuals for
GXP2130/GXP2140/GXP2160/GXP2170/GXP2135. Only major new features or major document updates are listed here. Minor
updates for corrections or editing are not documented here.

Firmware version 1.0.11.84

Disabled User Web Access by default. [Enable User Web Access]

Firmware version 1.0.11.79

Added Support to use custom ringtone for matching rule of Caller ID beginning with +. [Match incoming Caller ID]

Added Support to show date and time on status bar. [SHOW DATE AND TIME ON STATUS BAR]

Added Support to show local phonebook in device when it’s in public mode. [Enable Public Mode]

Added Support to display or hide clock and weather widget on LCD. [SHOW CLOCK WIDGET]

Added Support Rport (RFC 3581). [SUPPORT RPORT (RFC3581)]

Added Support of the predictive dialing feature for LDAP. [Predictive Dialing Source]

Added Support to send RTP event for the hook flash. [With RTP Packets]

Firmware version 1.0.11.74

Added support to enable TR-069 [ENABLE TR-069]

Firmware version 1.0.11.71

Added Support to upload OpenVPN TLS Authentication key from web UI. [OpenVPN TLS Authentication key]

Added Support of Expert Mode for OpenVPN to allow uploading the certificate zip file. [OpenVPN MODE]

Added Support of Zoom X-switch-info SIP Header. [USE X-SWITCH-INFO HEADER]

Added Support of Filter Characters for the click-to-dial feature [FILTER CHARACTERS]

Added Support to keep Bluetooth connection when users log out in public mode. [PRESISTENT BLUETOOTH]

Added Support that when With Secret Key Information is enabled, package capture will stop when the size threshold limit
is reached. [With Secret Key Information]

Added Support to add a subnet (e.g.: 192.168.1.0/24) in “Action URI Allowed IP List”[Action URI allowed IP list]
Added Support for voicemail VPK to dial into target’s mailbox. [MONITORED ACCESS NUMBER]

Added Support to show “Description” for Voicemail VPK. [Voicemail]

Firmware version 1.0.11.64

Added Support to download sslkeylogfile on Web packet capture. [Maintenance Page Definitions]

Removed weather service.

Firmware version 1.0.11.57

No major changes.

Firmware version 1.0.11.56

Added Support to configure two passwords for the same GDS “System Number” and display both “Open Door “softkey
on device LCD. [Settings Page Definitions]

Firmware version 1.0.11.48

Added Support to disable mute key during a call [Disable Mute Key in Call]

Firmware version 1.0.11.39

Added Support for Alert-Info: <string> in the Match Incoming Caller ID field. [Match Incoming Caller ID]

Firmware version 1.0.11.35

Improved DNS SRV Failover Design. [Register Before DNS SRV Failover]

Added USB headset support for GXP2140/GXP2160/GXP2170.

Added Support to keep DND after reboot. [Preserve DND Status]

Added Support to display the string entered in “System Identification” field when communicating with GDS. [System
Identification]

Firmware version 1.0.11.16

Added Support for Central Africa Time zone “CAT”. [Time Zone]

Added Support for Beirut Time zone. [Time Zone]

Added Ability to choose TLS Version for HTTPS provisioning. [Minimum TLS Version] [Maximum TLS Version]

Added Ability to enable/disable weak ciphers [Enable/Disable Weak Ciphers]

Added Ability to customize SSH port. [SSH Port]

Added Support of Factory Reset Security Level. [Factory Reset Security Level]

Added Ability to auto answer pre-defined numbers. [Auto Answer Numbers]

Added support to not send the DHCP release upon reboot. [Release DHCP On Reboot]

Firmware version 1.0.11.10

No major changes.

Firmware Version 1.0.11.6

No major changes.

Firmware Version 1.0.11.3

No major changes.

Firmware Version 1.0.11.2

Added Support to use uploaded CA Certificates only. [Load CA Certificates]

Added Support for inputting “@” for SIP User ID field. [SIP User ID]

Added Support for Ukrainian language. [GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 Technical Specifications]

Added Support for Chile time zone. [Time Zone]

Added the ability to launch XML App automatically upon receiving calls. [Enable XML Application Auto Launch]

Added Support to bypass security settings when External Service is configured. [Accept Incoming SIP from Proxy Only]

Added Support to lock the audio volume of the speaker phone. [Lock Speaker Volume]

Added Support for GDMS (Grandstream Device Management System). [Maintenance 🡪 TR-069]

Firmware Version 1.0.9.135

Added support for the new OUI (C0:74:AD)

Firmware Version 1.0.9.132

Added Web UI option “Disable Hook Switch”. [Disable Hook Switch]

Added Web UI option “Total Number of Custom Ringtone Update” [Total Number of Custom Ringtone Update]

Firmware Version 1.0.9.127

Added Clear option for Packet Capture. [Packet Capture]

Added Ability to sync phonebook directly through SIP notify [Immediate Download]

Firmware Version 1.0.9.121

Added support to disable the BLF call pickup process for both methods “Replaces header” and “Prefix” [BLF Call-pickup]

Added support to disable/enable ringtone on headset [Always Ring Speaker]

Added support to show both call session timer and hold duration timer on LCD during call hold [Show On Hold Duration]

Added support to set only the same account in conference [Only Same Account in Conference]

Added support to automatically firmware upgrade [Firmware Upgrade Confirmation]

Added support to turn off LCD even if there is a missed call notification and flash MWI LED [Disable Missed Call Backlight]

Added support to park call real number [Virtual Multi-Purpose Keys]

Removed CA/Custom certificate from configuration backup [Download Device Configuration]

Added option to disable incoming call popup [Incoming Call Popup]

Added support of exact match lookup method for LDAP search [Exact Match Search]

Added support to display or hide Diversion info [Enable Diversion Information Display]

Added VPKs to TR069 Data Model

Removed subscription then changing subscribe information [Virtual Multi-Purpose Keys]

Added support for more of 30 groups in local phonebook [Group Management]

Added support of manual VLAN configuration [Enable Manual VLAN Configuration]

Firmware Version 1.0.9.108

No major changes.

Firmware Version 1.0.9.102

Added Support to download certificate files during provisioning. [Certificates and Keys provision]

Updated Korean LCD and web UI display translations. [Korean]

Added Option to enable or disable Acoustic Echo Cancellation. [Enable Enhanced Acoustic Echo Canceller]

Added Ability to randomize the sending of TR069 INFORM message. [Randomized TR069 Startup]

Added Support for 802.1x authentication with special letter like “@”,”-“. [802.1X Identity] [MD5 Password]

Added More fields for Distinctive Ring Tone. (Matching Rule) [Match Incoming Caller ID]
Firmware Version 1.0.9.96

Added support for SNMPv3 [SNMP Settings]

Added support for more Action URL notification [Action URL]

Added support for Group Listening softkey [Group Listen with Speaker]

Added support to configure device with custom certificate signed by custom CA certificate [Custom Certificate]

Added FTP/FTPS support for provisioning and firmware upgrade [UPGRADING AND PROVISIONING]

Added Catalan Language support. [GXP2130/GXP2140/GXP2160/GXP2170/GXP2135 Technical Specifications]

Added option to enable/disable instant messages on LCD screen [IM]

Added support to allow custom softkey values configurable in all phone states [Call Screen Softkeys]

Allow provision to fallback to origin server path if it fails from the server from DHCP option 66 [Allow DHCP Option 43
and Option 66 Override Server]

Added support to inverse the sequence of uploading cfg files when the phone is forced to load all config files [Download
and Process All Available Config Files]

Added max download/search result number support for Broadsoft XSI directory [Broadsoft Contacts Download Limitation]

Removed downloading all XSI directory on reboot for Broadsoft

Added kick option for network conference for Broadsoft

Added ability to generate core dump files manually according to Phone or GUI [Core Dump]

Allowed feature key sync for call forward with local call features disabled for Broadsoft [Broadsoft]

Allow BLF key to pick up a call parked on a monitored extension for Broadsoft

Added monitored call park on busy lot should be denied [park a call to an occupied parking lot]

Added Web UI Entry for Users to Append OpenVPN Config Options [Additional options to be appended to the
OpenVPN®]

Added ability to show/hide VPK shared line description value by provisioning [VPK shared line display description]

Removed Device’s unused P-Values from the configuration backup file downloaded from Web UI

Added Dial plan flag T support. [Flag T]

Updated Broadsoft XSI Contact Download Interval tooltip [Contact Download Interval]

Improved Broadsoft SCA feature for handling multiple call arrangement under Account Mode

Updated Broadsoft XSI download interval default value to 72 hours [Contact Download Interval]

Firmware Version 1.0.9.71

Removed Device’s unused P-Values from the configuration backup file downloaded from Web UI.

Firmware Version 1.0.9.69

Added Confirm window for killing program to generate core dump on web UI.

Enhanced syslog to run on other ports instead of default port. [Syslog Server]

Added Option to choose either to override or accumulate groups when uploading a new XML phonebook. [Import Group
Method]

Added Auto provision starts when certain p-values are changed. [P-values that trigger Auto-Provision]

Added Attempt to download config files. [Attempt to download Config File again]

Firmware Version 1.0.9.63

Added “Account Display” option to configure SIP account display label on LCD. [Account Display]

Added Login Prefix/Suffix for public mode. [Public Mode Username Prefix] [Public Mode Username Suffix]

Enabled strong password for admin/user password. [Test Password Strength]

Added SNMP support [SNMP Settings]

Added CSTA support [CSTA Control]

Added web UI option to upload SSH public key for SSH access. [SSH Public Key]

Added option to disable user web account [Enable User Web Access]

Added support to customize idle time to logout the web access [Web Session Timeout]

Added support to customize number of failed attempts to access web GUI [Web Access Attempt Limit]

Added an option to force the phone to download and process all available config files [Download and Process All
Available Config Files]

Added an option on web UI to warm deleting all contacts [Delete All Contacts]

Added support of display of Active VPK page. [Enable Active VPK Page]

Added Random firmware/config download time feature [Randomized Automatic Upgrade]

Added support for Password change upon initial login [Change Password on First Boot]

Add ability to increase/decrease the priority of each existing pattern under phone’s dial plan [Dial Plan Configuration]

Added Server Validation [Validate Server Certificates]

Added option to disable DND function [Enable DND Feature]

Allowed LCD backlight to be always on or always off [Active Backlight Timeout]

Added ability to choose the predictive dialing source [Predictive Dialing Source]

Added option to enable/disable incoming call popup [Incoming Call Popup]

Added option to include MAC address in the SIP User-Agent [Use MAC Header]

Added Use SBC configuration for 3CX [Use SBC]

Added support for VPK share line to display description value [Virtual Multi-Purpose Keys]

Added support for more of 30 groups on local phonebook [Group Management]

Added host name of the phone on the DHCP INFORM using DHCP Option 12.

Allowed BLF and speed dial to perform blind transfer during active call.

Added support to get OpenVPN files from HTTP/TFTP server.

Added DoNotDisturbEvent support for 3CX [Enable DND Feature]

Firmware Version 1.0.9.32

No major changes.

Firmware Version 1.0.9.26

No major changes.

Firmware Version 1.0.9.25

Added support for “Off-Hook Auto Dial Delay” [Off-hook Auto Delay]

Added keypad support for factory reset [Restore to factory using hard keys]

Added “Call Completion” feature [Enable Call Completion Service]

Added support to enable keypad lock with MPK/VPK/Softkey by one press [Keypad Lock]

Added account swapping feature [Swap Account Settings]

Added dial plan new web UI interface [Dial Plan Configuration]

Added MPK mode for multicast listening list [Multicast Listen Address]

Added new MPK mode: forward, DND, redial, SMS, paging [Physical Multi-Purpose Keys]

Added “Transfer Mode via VPK” feature [Transfer Mode via VPK]

Added action URI feature [Action URI Support]

Added support for Cisco Discovery Protocol (CDP) [Enable CDP]

Added support of wav playback feature and custom ringtone for individual Contact [Account Ring Tone]

Added support to export/import user configuration [Download User configuration]

Added “User Protection” feature [User protection]

Added UCM Call Center Feature [UCM Call center]

Added support to show status of user space [User Space]

Added support for date and time display on screensaver [Show Date and Time]

Added support for default ringtone [Default Ringtone]

Firmware Version 1.0.8.53

No major changes

Firmware Version 1.0.8.49

Added date in the top panel on the phone LCD Screen [Show Date on Status Bar]

Added support of Syslog over TLS [Syslog Protocol]

Added disable call waiting per account [Disable Call Waiting]

Added ability to lock the phone ringing volume.

Added ability to search with case insensitive in Web UI phonebook [Search Bar]

Added auto-answer ring alert.

Added support for AES-256 in SRTP Call [SRTP Key Length]

Added support of provisioning “Trusted CA Certificates” [Trusted CA Certificates]

Added ability to display Broadsoft call center status on idle screen [Broadsoft Call Center]

Added GDS support Integration [External Service]

Added OpenVPN® username/password authentication and OpenVPN® Cipher option on web [OpenVPN® Cipher
Method][OpenVPN® Username][OpenVPN® Password]

Firmware Version 1.0.8.47

Added Ability to customize the domain name on the XSI request [XSI Actions Path]

Firmware Version 1.0.8.46

Added Option to Show/Hide VPK label on call screen [Show Keys Label]

Added Option to disable and enable the notification popup window for the missed call [Enable Missed Call Notification]

Added Option to Sync Extension Board Backlight with LCD [Sync Backlight with LCD]

Allowed certificate upload for OpenVPN® [OpenVPN® Settings]

Added Support of PAI update for CallPark VPK/MPK [PAI Update for CallPark VPK/MPK]

Added support for Predictive Dialing Feature [Predictive Dialing Feature]

Attended Transfer Improvement [Attended Transfer Mode]

Removed G729 codec restriction

Updated NTP default value to “pool.ntp.org” [NTP Server]

Added customizable OPUS Payload Type [OPUS Payload Type]

Added DND Barge — Internal Calls and Paging [DND Override]

Added user option to enable Plantronics EHS headset ringtone [EHS Headset Ring Tone]

Hided Caller ID info on line key and BLF Presentation [Hide BLF Remote Status]

Enabled RTCP & RTCP-XR [Protocols/Standards]

Added “Link” command to display port status [Link Command]

Added support TLS negotiation over TLS v1, TLS v1.1 and TLS v1.2 for SIP [TLS Negotiation]

Added support to send SIP log without enabling debug level [Send SIP Log]

Added MAC address display in the header of SIP Register [Use MAC Header]

Added ability to remove “target” softkey [Show Target Softkey]

Added ability to remove SIP error on LCD [Show SIP Error Response]

Added ability to send Instant Messages from web GUI [Send Instant Message]

Added features that support configurable option for RFC2543 Hold (0.0.0.0) and RFC3261 (a line) [RFC2543 Hold]

Added the ability to manage Call History from Web GUI [Call History]

Enhanced intercom options [Intercom Settings]

Added customizable Softkey Layout [Custom Softkey Layout]

Added ability to take screenshots from the phone [Screenshots]

Added Affinity password support [Affinity]

Added Affinity support [Affinity Support]

Enhanced MORE softkey selection [More Softkey Display Mode]

Added contact picture/icon through SIP Call-Info header [Contact Picture Support]

Added LINE key mode support, coexisted with legacy mode [Key Mode]

Added Live DialPad Feature [Enable Live DialPad]

Added specific model configuration file [Configuration File Download]

Added Automatic Redial Function [Enable Automatic Redial]

Added Preferred Default Account [Preferred default Account]

Added GXP2200EXT LCD timeout [Active Backlight Timeout]

Split Firmware/Config Upgrade Method/Username/Password [Upgrade and Provisioning]

Firmware Version 1.0.7.97

Added options to enable / disable custom SIP header [Custom SIP Headers]

Added support for configuring RTP port range [Local RTP Port Range]

Added support for more keys as send [Use # as Dial Key]

Added support of configurable HTTP/HTTPS port for Web UI access [HTTP Web Port][HTTPS Web Port]

Added language input search support [Language]

Added single button call parking support [Call Pickup Barge-In Code]

Added option to set the DTMF delay [DTMF Delay]

Added option to enable/disable HTTP access [Web Access Mode]

Added silent ringtone option [Account Ring Tone]

Added option to lock or restrict to only call/receive functionality without menu access and ability to configure anything
from phone side [Configuration via Keypad Menu]

Added ability to change instant message display duration

Added ability to change screensaver pictures via HTTP server [Screensaver Server Path]

Added support for DHCP option 132 & 133 tunneled through DHCP option 43 [Enable DHCP VLAN]

Added ability to use MPK to trigger a conference [Programmable Keys]

Added ability to display mobile and home number when searching in local

Added ability to set call forwarding from the web GUI [Feature Codes]

Added ability to disable/enable a sound notification for each ringing monitored BLF [Enable BLF Pickup Sound]

Added option to allow the user to modify the configuration Bluetooth via Web UI [Bluetooth]

Added option to enable/disable voicemail indication [Disable VM/MSG power light flash]

Added random registration [Delay Registration]

Added return code when call is rejected or DND [Return Code When Refusing Incoming Call][Return Code When Enable
DND]

Added option to enable Plantronics EHS headset ringtone [EHS Headset Ring Tone]

Changed OPUS sampling rate to 48000 Hz.

Added ability to change screensaver pictures via HTTP [Screensaver Server Path]

Firmware Version 1.0.7.81

Added support for local firmware upgrade [Upgrade Firmware]

Added a web option to let user chose whether to display internet down warning window

Added OPUS codec support [Voice Codec]

Added support to accept P-value in string format for VPK mode configuration xml [P-Value for VPK Mode in String
Format]

Added pre-Dialing search to include Broadsoft directories [Broadsoft]

Firmware Version 1.0.7.25

Added support for Broadsoft XSI authentication type [Settings Page Definitions]

Added support to configure Broadsoft XSI SIP authentication method by selecting the account [Settings Page Definitions]

Added support to stop Screensaver when VPK is active [Settings Page Definitions]

Added option to disable Auto Location Service from IPVideoTalk server [Settings Page Definitions]

Added supports for secondary NTP server [Settings Page Definitions]

Added the ability to specify Eventlist BLF listening transport protocol [Eventlist BLF Listening Transport Protocol]

Added support to play sound notification when one or more monitored BLF is ringing [Settings Page Definitions]

Added support to populate configurable User Agent field [Settings Page Definitions]

Added support to remove audio codec information on call screen [Accounts Page Definitions]

Added support of BLF call pickup with Barge-In option [Accounts Page Definitions]

Added option to control Speakerphone RX gain [Settings Page Definitions]

Added support to display status detail on LCD Screen when Ethernet not connected, account not register or configured

Added DNS SRV Fail-over Mode option support [Accounts Page Definitions]

Added separate subscription expire options for each account [Accounts Page Definitions]

Added support for default Dial Plan { x+ | \+x+ | *x+ | *xx*x+ } [Accounts Page Definitions]

Firmware Version 1.0.7.15

Added support for No Touch Provisioning to prompt for username and password for XML config file download for
Broadsoft server. [No Touch Provisioning]

Changed the default provisioning protocol to HTTPS. This option “Upgrade via” is under phone’s web
UI🡪Maintenance🡪Upgrade and provisioning. [Maintenance Page ]

Added support for outbound notification. [Outbound Notification Support]

Added support for Virtual Multi-Purpose Keys. [Virtual Multi-Purpose Keys]

Added support to show programmable keys status on web UI. [Programmable Keys Status On Web GUI]

Added option “Auto Provision List Starting Point” on web UI. [Settings Page Definitions]

Added additional ability to customize DHCP option for provisioning server. [Maintenance Page ]

Added support for iLBC and G723. [Accounts Page Definitions]

Added options for G723 rate, iLBC frame size and payload type. [Accounts Page Definitions]

Added option to enable and disable session timer. [Accounts Page Definitions]

Added option to ring speaker for call waiting. [Settings Page Definitions]

Added configurable backlight timer. [Settings Page Definitions]

Added color background wallpaper selection. [Settings Page Definitions]

Added BLF LED Pattern Explanation Form on web UI. [Settings Page Definitions]

Disable screen saver when VPK is active. [Settings Page Definitions]

Added fully black support for the idle screen LCD brightness (i.e., allow idle brightness to be 0). [Settings Page Definitions]
Added Blind and Attended Transfer Softkey options. [Blind Transfer and Attended Transfer Softkey]

Added ability to display SIP MESSAGE text on LCD. [Display SIP Message Text on LCD]

Firmware Version 1.0.6.9

This is the initial version for GXP2135

Added support to configure whether to show label background on VPK [Settings Page Definitions]

Added support to show long label on VPK [Settings Page Definitions]

Added support to hide Softkeys on main page [Settings Page Definitions]

Firmware Version 1.0.6.6

Added VPK support for eventlist auto-provision. If there are more BLFs in the eventlist than idle VPK keys, extra BLFs will
be auto-provisioned to EXT board [Settings Page Definitions]

Added “None” mode for VPK [Settings Page Definitions]

Added 12 lines support (with 6 accounts)

Firmware Version 1.0.6.2

This is the initial version for GXP2170

Firmware Version 1.0.5.23

Updated logo for web GUI

Firmware Version 1.0.5.18

Added more features descriptions for the MPKs mode – Monitored Call Park and Call Park sections. [Settings Page
Definitions]

Added BLF LED Patterns Settings for LED Control section. [Settings Page Definitions]

Added “Features” Softkey explanation for feature codes section. [Accounts Page Definitions]

Firmware Version 1.0.5.17

Added option to factory reset the phone directly through SIP NOTIFY. [Accounts Page Definitions]

Added option to disable multiple line in SDP, to send only 1 m line or multiple m lines. [Accounts Page Definitions]

Added option to allow barging by Call-Info. [Accounts Page Definitions]

Added option to disable recovery on blind transfer. [Accounts Page Definitions]

Added option to play a reminder tone when you have a call on hold. [Accounts Page Definitions]

Added Feature Codes Configuration Part on WEB GUI to support call features using star codes locally. [Accounts Page
Definitions]

Added PC Port VLAN Tag and PC Port Priority Value options to assigns the VLAN tag and the priority value of the PC port.
[Network Page Definitions]

Added option to disable SIP NOTIFY Authentication. [Maintenance Page ]

Added option to configure the device to download language files automatically from server. [Maintenance Page ]

Added option to set the default call log type. [Settings Page Definitions]

Added option to enable Local Call Recording. [Settings Page Definitions]

Added option to download local call recordings. [Settings Page Definitions]

Added option to configure the color and pattern of the LED based on status updates. [Settings Page Definitions]

Added function to allow configuration of MPK or Line key to provide MWI for other extension. [Settings Page Definitions]

Added function to allow configuration of Call Log for other extension. [Settings Page Definitions]

Added MPK mode Monitored Call Park for other extension. [Settings Page Definitions]

Added function to allow user to upload certificate file to phone and to configure the CA certificate. [Maintenance Page ]
Firmware Version 1.0.4.23

Added support to display the status of NAT connection for each account on the phone.

Added option to auto provision Eventlist BLFs with monitored extensions. [Accounts Page Definitions]

Added crypto life time option for SRTP calls. [Accounts Page Definitions]

Added option to set the NTP update interval time. [Settings Page Definitions]

Changed the default value of Layer 3 QoS for SIP to 26. [Network Page Definitions]

Added option to set the Layer 3 QoS for RTP. [Network Page Definitions]

Added BroadSoft Phonebook option in Phonebook Key functions list. [Phonebook Page Definitions]

Added LDAP Protocol option to support LDAP over TLS. [Phonebook Page Definitions]

GXP2130v1 does not support Bluetooth function, GXP2130v2 supports Bluetooth. [Bluetooth]

Firmware Version 1.0.4.16

Added support to configure phone’s MPK from phone GUI. [Settings Page Definitions]

Firmware Version 1.0.4.15

Added option to configure to always use the prefix for BLF Call-pickup. [Accounts Page Definitions]

Added option to send SUBSCRIBE to BroadSoft server to obtain Call Park Notifications. [Accounts Page Definitions]

Added option to send credentials before being challenged by the server. [Maintenance Page ]

Added user name and password options for HTTP/HTTPS server authentication for phonebook XML downloading.
[Phonebook Page Definitions]

Added option to enable/disable the dial plan check while dialing through the call history and any phonebook directories.
[Settings Page Definitions]

Added option to enable/disable the busy tone heard in the handset when call is disconnected remotely. [Settings Page
Definitions]

Added XML Application support. [Settings Page Definitions]

Added Direct IP Call support on MPK and Phonebook. [Settings Page Definitions]

Added ability to dial the digits faster when using MPK as Dial DTMF. [Settings Page Definitions]

Added support to play short reminder beep when performing auto answer. [Settings Page Definitions]

Firmware Version 1.0.4.10

Added option to show account name only and not the User ID on the LCD screen for GXP2130/2140. [Accounts Page
Definitions]

Added option for adding Auth Header on initial REGISTER. [Accounts Page Definitions]

Added BroadSoft Network Directories features. [Settings Page Definitions]

Added Web UI option for downloading Language XML file. [Maintenance Page ]

Added Web UI option for auto language download. [Maintenance Page ]

Added Multicast paging support. [Settings Page Definitions]

Added packet capture support. [Maintenance Page ]

Added phonebook entry sorting option. [Phonebook Page Definitions]

Firmware Version 1.0.3.9

Added PhonePower special feature. [Accounts Page Definitions]

Added BroadSoft IM&P features. [Phonebook Page Definitions]

Added Screensaver options to LCD under Preference🡪Appearance. [Configuration via Keypad]

Added Web UI option to select default search mode for phonebook. [Configuration via Keypad]

Added Second dial tone support. [Settings Page Definitions]

Added Input character selection window. [Configuration via Keypad]

Added BLF server support. [Accounts Page Definitions]

Firmware Version 1.0.2.9

Add Bluetooth hands free mode. [Bluetooth]

Added Configuration file upload support via Web UI. [Maintenance Page ]

Add Screen saver support. [Settings Page Definitions]

Add Wallpaper support. [Wallpaper]

Add the support of STAR key keypad lock feature. [Maintenance Page ]

Add the support of Configuration via Keypad Menu. [Maintenance Page ]

Add Keypad shortcut to reboot and provisioning. [Shortcut of Upgrade and Provision via Keypad Menu]

Firmware Version 1.0.1.19

Added GXP2130

Firmware Version 1.0.1.6

Added Local group and BroadSoft phonebook in phonebook support. [Maintenance Page ]

Added Instant message. [Configuration via Keypad]

Added BroadSoft shared call appearance support. [Accounts Page Definitions]

Added BroadSoft call center support. [Accounts Page Definitions]

Added Eventlist BLF update support for BroadSoft. [Accounts Page Definitions]

Firmware Version 1.0.0.17

This is the initial version for GXP2140/GXP2160.

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