GXP21xx Series - Administration Guide

  • Updated on August 24, 2022

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 Multi-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.

   GXP21303 lines
2.8 inch (320×240) TFT color LCD
4 programmable soft keys
Bluetooth V2.1 (GXP2130v2 only)
8 programmable Multi-Purpose Keys
4-way conference  
  GXP21404 lines
4.3 inch (480×272) TFT color LCD
5 programmable soft keys
Bluetooth V2.1
5-way conference
Expansion board  
   GXP21606 lines
4.3 inch (480×272) TFT color LCD
5 programmable soft keys
Bluetooth V2.1
5-way conference24 programmable Multi-Purpose Keys  
 
 GXP217012 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
5 programmable soft keys
Bluetooth V2.1
5-way conference
Expansion board
 GXP21358 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
4 programmable soft keys
Bluetooth V2.1
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/Standards

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

Network Interfaces

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

Graphic Display

2.8 inch (320×240) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1 (GXP2130v2 only, GXP2130v1 does not support Bluetooth)

Feature Keys

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

Voice Codec

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-band DTMF (in audio, RFC2833, SIP INFO)

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets)

Telephony Features

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

HD audio

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

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes

QoS

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

Security

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

Multi-language

English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew, Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia, Spanish, Turkish, Catalan, Ukrainian

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included: Input:100-240 VAC; Output: +12VDC, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Physical

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

Unit weight: 0.78kg

Package weight: 1.3kg

Temperature and Humidity

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

Package Content

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

Compliance

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

Table 3: GXP2130 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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-band DTMF (in audio, RFC2833, SIP INFO)

Auxiliary Ports

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

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Extension Module

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

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes

QoS

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

Security

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

Multi-language

English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew, Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia, Spanish, Turkish, Catalan, Ukrainian

Upgrade/Provisioning

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

Power & Green Energy Efficiency

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

Integrated Power-over-Ethernet (802.3af)

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

Temperature and Humidity

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

Package Content

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

Compliance

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

Table 4: GXP2140 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets), USB

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes

QoS

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

Security

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

Multi-language

English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew, Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia, Spanish, Turkish, Catalan, Ukrainian

Upgrade/Provisioning

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

Power & Green Energy Efficiency

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

Integrated Power-over-Ethernet (802.3af)

Max power consumption: 6W

Physical

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

Unit weight: 0.98kg; Package weight: 1.62kg

Temperature and Humidity

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

Package Content

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

Compliance

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

Table 5: GXP2160 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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

Auxiliary Ports

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

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Extension Module

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

Base Stand / Wall Mountable

Yes, allow 2 angle positions

QoS

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

Security

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

Multi-language

English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew, Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia, Spanish, Turkish, Catalan, Ukrainian

Upgrade/Provisioning

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

Power & Green Energy Efficiency

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

Integrated Power-over-Ethernet (802.3af)

Max power consumption:

5.4W (without GXP2200EXT)

or 9.2W (with 4 cascaded GXP2200EXTs)

Physical

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

Unit weight: 0.98kg; Package weight: 1.55kg

Temperature and Humidity

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

Package Content

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

Compliance

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

Table 6: GXP2170 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

2.8 inch (320×240) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets)

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Base Stand / Wall Mountable

Yes, allow 2 angle positions

QoS

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

Security

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

Multi-language

English, Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew, Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenia, Spanish, Turkish, Catalan, Ukrainian

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included: Input:100-240VAC; Output: +12VDC, 0.5A;

Integrated Power-over-Ethernet (802.3af) Max power consumption 3W

Physical

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

Unit weight: 0.98kg

Package weight: 1.55kg

Temperature and Humidity

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

Package Content

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

Compliance

FCC Part 15 (CFR 47) Class B ; EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1, EN62479, 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 History

Displays Local call logs:

All Calls/Answered Calls/Dialed Calls/Missed Calls/Transferred Calls

Status

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

  • Account status.
  • Network status.

Press to enter the sub menu for MAC address, IP setting information (DHCP/Static IP/PPPoE), Ipv4 address, Ipv6 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

Contacts sub menu includes the following options:

  • Local Phonebook
  • Local Group
  • LDAP Directory

Contacts sub menu is for Local Phonebook, Local Group, LDAP Directory and Broadsoft Phonebooks. User could configure phonebooks/groups/LDAP options here, download phonebook XML to the phone and search phonebook/LDAP directory.

Messages

Message sub menu include the following options :

  • Instant Message

Displays received instant messages;

  • Voice Mails

Displays voicemail message information in the format below: new messages/all messages (urgent messages/all urgent messages).

Preference

Preference sub menu includes the following options:

  • Do Not Disturb

Enables/disables Do Not Disturb on the phone.

  • 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

Phone sub menu includes the following options:

  • SIP

Configures SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth ID, SIP Password, SIP Transport and Audio information to register SIP account on the phone.

  • Call Features

Configures call forward features for Forward All, Forward Busy, Forward No Answer and No Answer Timeout.

System

System sub menu includes the following options:

  • Network
    • 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.

Reboot

Reboots the phone.

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 default passwords for the login page:
User LevelUserPasswordWeb Pages Allowed

End User Level

user

123

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.
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.

SIP Server

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

SIP Registration

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

Status 🡪 Network Status

MAC Address

Global unique ID of device, in HEX format. The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device.

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.

IP Geographic Information

  • City: displaying phone location;
  • 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

Core Dump

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

Status 🡪 Programmable Keys Status 🡪Virtual Multi-Purpose Keys

VPKs Status

  • Mode
  • Account
  • Description
  • Value

Status 🡪 Programmable Keys Status 🡪 Multi-Purpose Keys

MPKs Status

  • Mode
  • Account
  • Description
  • Value

Status 🡪 Programmable Keys Status 🡪 Softkeys

Softkeys

  • Mode
  • Account
  • Description
  • Value

Status 🡪 Extension board Status

Extension 1/2/3/4 Keys

  • Mode
  • Account
  • Description
  • Value
Table 9: Status Page Definitions

Accounts Page Definitions

Account x 🡪 General Settings

Account Active

This field indicates whether the account is active.

The default setting is “Yes”.

Account Name

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

SIP Server

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

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

Secondary SIP Server

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

Outbound Proxy

IP address or Domain name of the Primary Outbound Proxy, Media Gateway, or Session Border Controller. It’s used by the phone for Firewall or NAT penetration in different network environments.

If a symmetric NAT is detected, STUN will not work and ONLY an Outbound Proxy can provide a solution.

Backup Outbound Proxy

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

BLF Server

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

SIP User ID

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:

– Users are able to register an account with a SIP user ID that carries “@”. (For example: “111@test.com”, so the phone will register the account as “111@test.com” instead of 111)

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

Authenticate ID

SIP service subscriber’s Authenticate ID used for authentication.

It can be identical to or different from the SIP User ID.

Authenticate Password

The account password required for the phone to authenticate with the ITSP (SIP) server before the account can be registered.

After it is saved, this will appear as hidden for security purpose.

Name

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

Voice Mail User ID

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

Picture

Specifies account’s picture that will be sent to the caller/callee when making calls.

Account Display

This option allows you to configure how your SIP account label will be displayed on the phone’s screen.

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

DNS Mode

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:

  • Primary IP:
  • Backup IP 1;
  • Backup IP 2.

If SIP server is configured as domain name, phone will not send DNS query, but use “Primary IP” or “Backup IP x” to send SIP message if at least one of them are not empty.

Phone will try to use “Primary IP” first. After 3 tries without any response, it will switch to “Backup IP x”, and then it will switch back to “Primary IP” after 3 re-tries.

If SIP server is already an IP address, phone will use it directly even “User Configured IP” is selected.

DNS SRV Fail-over Mode

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

  • Default

If the option is set with “default”, it will again try to send register messages to one IP at a time, and the process repeats.

  • Saved one until DNS TTL

If the option is set with “Saved one until DNS TTL”, it will send register 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.

  • 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 Before DNS SRV Failover

This option allows to choose the behavior for registering before DNS SRV Fail-over.

  • If set to “No”, a REGISTER request will not be initiated when a server failover occurred under DNS SRV mode.
  • If set to “Yes”, a REGISTER request will be initiated when a server failover occurred under DNS SRV mode.

Primary IP

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

Backup IP1

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

Backup IP2

Configures the backup IP2 address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

NAT Traversal

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 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.

Proxy-Require

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

Use SBC

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

Account x 🡪 SIP Settings 🡪 Basic Settings

TEL URI

If the phone has an assigned PSTN telephone number, this field should be set to “User=Phone”.

Then a “User=Phone” parameter will be attached to the Request-Line and “TO” header in the SIP request to indicate the E.164 number.

If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is “Disable”.

SIP Registration

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

Unregister On Reboot

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”.

  • If set to “All”, the SIP user’s registration information will be cleared when the phone reboots. The SIP Contact 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.

Register Expiration

Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified registrar. The default value is 60 minutes.

The maximum value is 64800 minutes (about 45 days).

Subscribe Expiration

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

Reregister Before Expiration

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

Enable OPTIONS Keep Alive

Enable OPTIONS Keep Alive to check SIP Server.

OPTIONS Keep Alive Interval

Time interval for OPTIONS Keep Alive feature in Second.

OPTIONS Keep Alive Max Lost

Number of max lost packets for OPTIONS Keep Alive feature before the phone re-registration.

Local SIP Port

Defines the local SIP port used to listen and transmit. The default value is 5060 for Account 1, 5062 for 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 Failure Retry Wait Time

Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600.

The default value is 20 seconds.

SIP T1 Timeout

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

SIP T2 Timeout

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

SIP Transport

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

SIP Listening Mode

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”.

  • Transport Only
  • Dual
  • Dual (Secured)
  • Dual (BLF Enforced)

SIP URI Scheme when using TLS

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

Use Actual Ephemeral Port in Contact with TCP/TLS

This option is used to control the port information in the Via header and Contact header. If set to No, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the connection.

The default setting is “No”.

Outbound Proxy Mode

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

Support SIP Instance ID

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

SUBSCRIBE for MWI

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

SUBSCRIBE for Registration

When set to “Yes”, a SUBSCRIBE for Registration will be sent out periodically. The default setting is “No”.

Enable 100rel

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

Callee ID Display

When set to “Auto”, the phone will update the callee ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and To Header in the 180 Ringing. If “Disabled”, callee ID will be displayed as “Unavailable”. When set to “To Header”, caller ID will not be updated and displayed as To Header.

Caller ID Display

When set to “Auto”, the phone will look for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP INVITE. When set to “Disabled”, all incoming calls are displayed with “Unavailable”. When set to “From Header”, the phone will display the caller ID based on the From Header in the incoming SIP INVITE. The default setting is “Auto”.

Add Auth Header on Initial REGISTER

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

Allow SIP Reset

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

Ignore Alert-Info header

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

Account x 🡪 SIP Settings 🡪 Custom SIP Headers

Use Privacy Header

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

Use P-Preferred- Identity Header

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

If set to “No”, the P-Preferred-Identity Header will not show in INVITE.

Use X-Grandstream-PBX Header

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

Use P-Access-Network-Info Header

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

Use P-Emergency-Info Header

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

Use MAC Header

If Yes except REGISTER, the sip message for register or unregister will contains MAC address in the header, and all the outgoing SIP messages except REGISTER message will attach the MAC address to the User-Agent header;

If Yes to ALL SIP, the sip message for register or unregister will contains MAC address in the header, and all the outgoing SIP message 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

Line Seize Timeout

For Shared Call Appearance, phone must send a SUBSCRIBE-request for the line-seize event package whenever a user attempt to take the shared line off hook. “Line Seize Timeout” is the line-seize event expiration timer. The default value is 15 seconds. The valid range is from 15 to 60.

Eventlist BLF URI

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., BLF1006@myserver.com) with a list of extensions included. On the phone, in this “Eventlist BLF URI” field, fill in the URI without the domain (i.e., BLF1006). To monitor the extensions in the list, under Web GUI🡪Settings🡪Programmable Keys page, please select “Eventlist BLF” in the key mode, choose account, enter the value of each extension in the list.

Auto Provision Eventlist BLFs

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

Conference URI

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

Music On Hold URI

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

BLF Call-pickup

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.

  • If select Force BLF Call-pickup by prefix:

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-pickup Prefix

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

Call Pickup Barge-In Code

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

PUBLISH for Presence

Enables presence feature on the phone. The default setting is “No”.

Omit charset=UTF-8 in MESSAGE

Omit charset=UTF-8 in MESSAGE content-type

Allow Unsolicited REFER

Allow Unsolicited REFER to accomplish an outgoing call.

Special Feature

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

Broadsoft Call Center

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

Please note that “Feature Key Synchronization” will be enabled regardless of this setting. Default setting is “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 Event

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

Call Center Status

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

Broadsoft Executive Assistant

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

Feature Key Synchronization

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

Broadsoft Call Park

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

Account x 🡪 SIP Settings 🡪 Session Timer

Enable Session Timer

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

Session Expiration

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 terminated once the session interval expires. Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default setting is 180. The valid range is from 90 to 64800.

Min-SE

The minimum session expiration (in seconds). The default value is 90 seconds. The valid range is from 90 to 64800.

Caller Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it makes outbound calls. The default setting is “No”.

Callee Request Timer

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

Force Timer

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

UAC Specify Refresher

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

UAS Specify Refresher

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

Force INVITE

The Session Timer can be refreshed using the INVITE method or the UPDATE method. Select “Yes” to use the INVITE method to refresh the session timer.

The default setting is “No”.

Account x 🡪 SIP Settings 🡪 Security Settings

Check Domain Certificates

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

Validate Certificate Chain

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

Validate Incoming Messages

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

Check SIP User ID for Incoming INVITE

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

Accept Incoming SIP from Proxy Only

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

Authenticate Incoming INVITE

If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response. Default setting is “No”.

Account x 🡪 Audio Settings

Preferred Vocoder

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

Use First Matching Vocoder in 200OK SDP

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

Codec Negotiation Priority

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

Hide Vocoder

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

Disable Multiple m line in SDP

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

SRTP Mode

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

SRTP Key Length

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

Default setting is: AES 128&256 bit

Crypto Life Time

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

Symmetric RTP

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

Silence Suppression

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

Jitter Buffer Type

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

Jitter Buffer Length

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

Voice Frames Per TX

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

The default setting is 2.

G723 Rate

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

The default setting is “5.3kbps encoding rate”.

G.726-32 Packing Mode

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

iLBC Frame Size

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

iLBC Payload Type

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

OPUS Payload Type

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

DTMF Payload Type

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

Send DTMF

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);

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

SIP INFO uses SIP INFO to carry DTMF.

Default setting is “RFC2833”.

DTMF Delay

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

Account x 🡪 Call Settings

Early Dial

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, it executes the action.

The default setting is “No”.

Dial Plan Prefix

Configures the prefix to be added to each dialed number.

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. [35] — 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.

Call Log

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

Account Ring Tone

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

Match Incoming Caller ID

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.

  • 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 Timeout

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

Send Anonymous

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

Anonymous Call Rejection

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

Auto Answer

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep. Default setting is “No”.

Auto Answer Numbers

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;

2) Auto Answer enable, no auto answer number specified → all numbers are auto answered;

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 Use Target Contact

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

Transfer on Conference Hang-up

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

Disable Recovery on Blind Transfer

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

Notes:

1) This feature only applies to blind transfer;

2) This feature depends on how server handles transfer. If there is any NOTIFY from server, this feature won’t 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 Wait Timeout

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

No Key Entry Timeout

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

Use # as Dial Key

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

On Hold Reminder Tone

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

RFC2543 Hold

Allows users to toggle between RFC2543 hold and RFC3261 hold. RFC2543 hold (0.0.0.0) allows user to disable the hold music sent to the other side. RFC3261 (a line) will play the hold music to the other side.

Hide Dialing Password

Allows users to hide the password when the dialing number matches the configured prefix.

Disable Call Waiting

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

Account x 🡪 Intercom Settings

Allow Auto Answer by Call-Info/Alert-Info

Allows the phone to automatically turn on the speaker phone to answer incoming calls after a short reminding beep when enabled, based on the SIP Call-Info/Alert-Info header sent from the server/proxy. Default setting is “No”.

Allow Barging by Call-Info/Alert-Info

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

Mute on answer Intercom call

When enabled, the phone will mute the incoming intercom call.

Play warning tone for Auto Answer Intercom

When enabled, the phone will play warning tone when auto answer Intercom.

Custom Alert-Info for Auto Answer

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

Account x 🡪 Feature Codes

Enable Local Call Features

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 LCD screen.


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 (DND)—On

Configures DND feature code to turn on DND.

Do Not Disturb (DND)—Off

Configures DND feature code to turn off DND.

Call Forward Unconditionally (All)—On

Configures Call Forward All feature code to activate unconditional call forwarding.

Call Forward Unconditionally (All)—Off

Configures Call Forward All feature code to deactivate unconditional call forwarding

Target

Configures the extension that the call will be forwarded to.

Call Forward Busy—On

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

Call Forward Busy—Off

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

Target

Configures the extension that the call will be forwarded to.

Call Forward Delayed (No Answer)—On

Configures Call Forward Delayed feature code to activate no answer call forwarding.

Call Forward Delayed (No Answer)—Off

Configures Call Forward Delayed feature code to activate no answer call forwarding.

Target

Configures the extension that the call will be forwarded to.

Delayed Call Forward Wait Time

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

Accounts 🡪 Account Swap

Swap Account Settings

Allows users to swap the two accounts that they have configured. This will Increase the flexibility of account management.

Note: Make sure to press “Start” to complete the process.

Table 10: Account Page Definitions

Settings Page Definitions

Settings 🡪 General Settings

Local RTP Port

This parameter defines the local RTP port used to listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP; channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024 to 65400 and must be even. Default value is 5004.

Local RTP Port Range

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 be adjusted if local RTP port + local RTP port range is greater than 65486. Default setting is 200.

Use Random Port

When set to “Yes”, this parameter will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is “Yes”

Note: This parameter must be set to “No” for Direct IP Calling to work.

Keep-alive Interval

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

Use NAT IP

The NAT IP address used in SIP/SDP messages. This field is blank at the default settings. It should ONLY be used if it’s required by your ITSP.

STUN Server

The IP address or Domain name of the STUN server. STUN resolution results are displayed in the STATUS page of the Web GUI.

Only non-symmetric NAT routers work with STUN.

Public Mode

Configures to turn on/off the public mode for hot desking feature. The default setting is “No”.

Enable Fix For RTP Timestamp Jump

Makes RTP timestamps be continuous to fix the audio loss issue when there is a 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.

Delay Registration

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

Test Password Strength

Only allow password with these constraints to ensure better security:

The password has to be more than 9 characters/digits and has to fulfill at least 3 options among 4 options below:

1) Numeric (0-9)

2) Capital letters (A-Z)

3) Lower case (a-z)

4) Special characters (!, @, #, $, %, ^, &, *, (, ), etc)

Default setting is “No”.

Settings 🡪 Broadsoft 🡪 Broadsoft XSI

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. Range is 0-2000

  • 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

Login Credentials

  • Server

Broadsoft IM&P server address. Usually not necessary to configure and can already be found in the Broadsoft IM&P username.

  • Port

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.

Broadsoft IM&P

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

Associated Broadsoft Account

Specifies the associated account. User could choose each account on the phone.

Auto Login

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

Display Non XMPP Contacts

Choose whether to display non-xmpp contacts associated with the Broadsoft IM&P 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.

Service Type

Specifies the service’s type. Two options are available: None or GDS.

Default setting is None. Note: The GXP21xx supports up 10 GDS items.

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.

System Identification

Specifies the name to identify the service.

The field “System identification” will show up on LCD next to the GDS number as indicated: <system identification>(space)<GDS number>

System Number

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

Access Password

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

Settings 🡪 Call Features

Preferred default Account

Allows user to select a default account when other accounts have not been selected. 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.

Predictive Dialing Feature

Allow users to show/hide predictive dialing feature, when disabled, users will not see any predictive numbers while dialing a number.

Predictive Dialing Source

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

Onhook Dial Barging

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

Disable Mute Key in Call

This feature allows users to disable mute key during a call.

Off-hook Auto Dial

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

Off-hook Auto Delay

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

Off-hook Timeout

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

Enable Live DialPad

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

Live DialPad Expire Time

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

Enable Automatic Redial

If enabled, the phone will redial the number a configured number of times with a 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 History and Directories

Enable/Disable the dial plan check while dialing through the call history and any 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”.

Ring For Call Waiting

Disables / enables the call waiting tone when the call waiting feature is enable. Default is disabled.

Disable Busy Tone on Remote Disconnect

Disables the busy tone heard in the handset when call is disconnected remotely. The default setting is “No”.

Disable Direct IP Call

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

Use Quick IP Call mode

When set to “Yes”, users can dial an IP address under the same LAN/VPN segment by entering the last octet in the IP address.

To dial quick IP call, off hook the phone and dial #XXX (X is 0-9 and XXX <=255), phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask. #XX or #X are also valid so leading 0 is not required (but OK). No SIP server is required to make quick IP call. The default setting is “No”.

Disable Conference

Disables the Conference function.

The default setting is “No”.

Only Same Account in Conference

If enabled, the phone only allows a call associated with the same account to build a conference.

Disable in-call DTMF Display

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

Enable Sending DTMF via specific MPKs

Allows certain MPKs to send DTMF in-call.

This option doesn’t affect Dial DTMF.

Disable Active MPK Page

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

Enable Active VPK Page

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

Default setting is Disabled.

Enable DND Feature

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.

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.

Mute Key Functions While Idle

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 answered with mute. When select “Disabled”, Mute key will not take effect while idle. The default setting is “DND”.

DND Override

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.
  • 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.

Disable Transfer

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

In-call Dial Number on Pressing Transfer Key

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

Attended Transfer Mode

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

Show On Hold Duration

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

Do Not Escape # as %23 in SIP URI

Specifies whether to replace # by %23 or not for some special situations. The default setting is “No”.

Click-To-Dial Feature

Enables Click-To-Dial feature. If this feature is enabled, user could click the green dial 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”.

Default call log type

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

Return Code When Refusing Incoming Call

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

Return Code When Enable DND

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

Enable BLF Pickup Screen

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

Enable BLF Pickup Sound

Gives the user the ability to set sound notification to the monitoring BLF line when it’s ringing, GX21xx should play a sound to inform user.

The default setting is “No”.

BLF Pickup Sound Except List

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

BLF Pickup Sound Only list

Configures play BLF sound notification only for the list below.

Local Call Recording Feature

Gives the ability to record calls locally while on the call screen.

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.

Instant Message Popup Timeout

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

Play Tone On Receiving IM

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

Allow Incoming Call Before Ringing

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

User-Agent Prefix

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 device specific identification.

Ex. The value could be Mobile, Fixed, Desktop, etc. The configured “User Agent” should be prepend to vendor’s default User.

Auto Provision List Starting Point

Users could select “Extension Boards” or “VPK” which will be used first when the 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.

Enable Missed Call Notification

Allows users to show/hide the notification popup for missed calls. Default is “No” which will hide call notification popup.

Note: Currently the manually rejected calls are counted as missed calls.

Enable Call Completion Service

When the automatic redial and call completion service are enabled, and the user 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.

Enable Incoming Call Popup

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 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 Canceller

Allows users to choose whether to enable or disable the echo canceller on their phone in speaker mode.

Enable Diversion Information Display

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

Default is “Yes”.

Disable Hook Switch

When set to “Yes”, disable hook switch completely; When set to “For Answer Call”, 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.

Delete all

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, 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.

Default Setting is “No”.

Paging Barge

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

Paging Priority Active

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

Multicast Paging Codec

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

Multicast Channel Number

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

Multicast Sender ID

Outgoing caller ID that displays to your page group recipients (for multicast channel 1 – 50).

Multicast Listening

Defines multicast listening addresses and labels. For example:

  • “Listening Address” should match the sender’s Value such as

“237.11.10.11:6767”

  • “Label” could be the description you want to use.

For details, please check the “Multicast Paging User Guide” on our Website.

Settings 🡪 Outbound Notification

Action URL

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
  • 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

Destination

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

Notification

Specifies the message body of the notification for each event that can be customized with embedded dynamic attributes.

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”.

  1. 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.
  1. 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.

Headset Type

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

EHS Headset Ring Tone

Allows user to enable the ringtone from Plantronics EHS headset and play the ringtone in the headset.

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.

Always Ring Speaker

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, 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.

The default setting is “No”.

Group Listen with Speaker

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

Headset TX gain

Configures the transmission gain of the headset.

The default value is 0 dB.

Headset RX gain

Configures the receiving gain of the headset.

The default value is 0 dB.

Handset TX gain

Configures the transmission gain of the handset.

The default value is 0 dB.

Settings 🡪 Preferences 🡪 Date and Time

NTP Server

Defines the URL or IP address of the NTP server. The phone may obtain the date and time from the server.

The default setting is “pool.ntp.org”.

Secondary NTP Server

Defines the URL or IP address of the NTP server. The phone may obtain the date and 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.

NTP Update Interval

Time interval for updating time from the NTP server. Valid time value is in between 5 to 1440 minutes.

The default setting is “1440” minutes.

Allow DHCP Option 42 Override NTP Server

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

Time Zone

Configures the date/time used on the phone according to the specified time zone.

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.

Date Display Format

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

  • yyyy-mm-dd: 2012-07-02
  • mm-dd-yyyy: 07-02-2012
  • dd-mm-yyyy: 02-07-2012
  • dddd, MMMM dd: Friday, October 12
  • MMMM dd, dddd: October 12, Friday

Time Display Format

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

Show Date on Status Bar

Allows users to display time and date on the top panel of the LCD screen. Default setting is Disabled. Note: For GXP2135 and GXP2170, the time and date will be displayed on top of LCD when the top VPK on the right side of LCD screen is not configured.

Settings 🡪 Preferences 🡪 LCD Display

Backlight Brightness: Active

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

Backlight Brightness: Idle

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

Active Backlight Timeout

Allows user to set up the backlight time (in minutes) for the extension board. Valid range from 0 to 90. Default value is 1.

Note: When Active Backlight Timeout is set to 0, the backlight will be constantly on.

Disable Missed Call Backlight

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

  • If set to “Yes”, the screen will turn off the LCD backlight even if there is a missed call on the phone.
  • 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”.

Wallpaper Source

(Note: USB is only for GXP2140, GXP2160 and GXP2170)

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

Note: If you choose “Color Background”, you need to enter a 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.

Wallpaper Server Path

Specifies the wallpaper server path. This option will take effect when wallpaper source is “Download”.

Upload Wallpaper

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

Color Background

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

Reference: http://htmlcolorcodes.com

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

Screensaver

Configures Screensaver Feature, or “to enable Screensaver feature if no VPK is active”. Please note this option is also available under LCD🡪 Menu🡪 Preference🡪 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.

Screensaver Source

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

Show Date and Time

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

Screensaver Timeout

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

Screensaver Server Path

Configures the server path which contains download screensaver definition XML.

Screensaver XML Download Interval

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

Settings 🡪 Preferences 🡪 LED Control

BLF LED Pattern

This is used to configure the color and pattern of the LED based on status updates. The default setting is “Default”.

The BLF LED Patterns are listed in [Table 15: BLF LED Patterns].

Disable VM/MSG power light flash

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

BLF LED Pattern Explanation Form

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

Settings 🡪 Preferences 🡪 Ring Tone

Call Progresses Tones

System Ring Tone

Dial Tone

Second Dial Tone

Message Waiting

Ring Back Tone

Call-Waiting Tone

Busy Tone

Reorder Tone

Configures ring or tone frequencies based on parameters from local telecom. The default value is North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.

Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
(Frequencies are in Hz and cadence on and off are in 10ms)

ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence.

In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.

Call-Waiting Tone Gain

Configures the call waiting tone gain to adjust call waiting tone volume. Default is Low.

Speaker Ring Volume

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

The default setting is 5.

Notification Tone Volume

Configures notification tone volume. The valid range is 0 to 7 and default setting is 5.

Lock Speaker Volume

Locks volume adjustment when the option is enabled.

– If set to “No” (Default), volume of the speaker will not be locked.

– If set to “Ring”, volume of the speaker will be locked in ringing state and idle.

– If set to “Talk”, volume of the speaker will be locked in calling state.

– If set to “Both”, volume of the speaker will be locked.

Default Ringtone

Allows to set Default Ringtone as their Global ringtone.

Note: The ring tone set in individual accounts have higher priority to this setting. If the user wants the default ring tone to be used globally, he needs to set the ring tone of each account to Default Ring Tone; Otherwise, it will be whichever the ring tone you set.

Important: The Priority goes as: Contact Ring Tone 🡪 Account Ring Tone 🡪 Default Ring Tone.

Total Number of Custom Ringtone Update

Configures the total number of custom ringtone update that can be downloaded during provisioning process. Default is 3.

Settings 🡪 Programmable Keys

Virtual Multi-Purpose Keys Settings

  • 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.

  • 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.

  • 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 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 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 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 MORE softkey to switch between options.

  • Show Target Softkey

Allows users to remove target softkey by toggle 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 main page. Default value is none.

  • Dialing State

Custom softkey layout when device is under Dialing State.

Available Softkeys: Phonebook(BT), BT On/Off, EndCall, ReConf, ConfRoom, Redial, Dial, Backspace, PickUp.

  • Onhook Dialing State

Custom softkey layout when device is under Onhook Dialing State.

Available Softkeys: Phonebook(BT), DirectIP, Cancel, Dial, Backspace

  • Ringing State

Custom softkey layout when device is under Ringing State.

Available Softkeys: Answer, Reject, Forward, ReConf.Calling State

  • Calling State

Custom softkey layout when device is under calling State.

Available Softkeys: BT On/Off, EndCall, ReConf, ConfRoom, ConfCall.

  • Call Connected State

Custom softkey layout when 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 device is under Conference Connected State.

Available Softkeys: BT On/Off, EndCall, Kick.

  • On Hold State

Custom softkey layout when device is under On Hold State.

Available Softkeys: ReConf, Resume, Transfer, ConfCall, Add.

  • Call Failed State

Custom softkey layout when device is under Call Failed State.

Available Softkeys: EndCall, ReConf, ConfRoom.

  • Transfer State

Custom softkey layout when device is under Transfer State.

Available Softkeys: BT On/Off, Cancel, BlindTrnf, AttTrnf, Backspace.

  • Conference State

Custom softkey layout when 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 offhook 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 account number in the Value field to access to the Call Log of that selected account.

  • Menu

Select this feature in order to display the Menu 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

Call Screen Softkeys

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

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

EXT setting (Available only for GXP2140/2170)

  • One Page Display Mode

Each extension board only shows 20 Extensions, that is, EXT 1 ~ 80 could be displayed on 4 connected boards if the mode is enabled.

  • 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 GXP2140/2170)

Assigns a function to the corresponding Extension Board key.
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

Use Auto Location Service

Configures to enable or disable auto location services on the phone. (Reboot Required). The default setting is “Yes”.

Settings 🡪 XML Application

Server Path

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

Softkey Label

Specifies the Softkey name displayed on the idle screen for the users to enter XML application.

The default Softkey Label is “XMLApp”.

Default Background Color

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

E.g. #000000. Reference: http://htmlcolorcodes.com

Block Call Screen

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

Enable XML Application Auto Launch

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

Table 11: Settings Page Definitions

Network Page Definitions

Network 🡪 Basic Settings

Internet Protocol

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

Ipv4 Address

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

Host name (Option 12)

Specifies the name of the client. This field is optional but may be required by some Internet Service Providers.

DHCP Vendor Class ID

(Option 60)

Used by clients and servers to exchange vendor class ID.

The default setting is “Grandstream GXP2130” for GXP2130, ”Grandstream GXP2140” for GXP2140, “Grandstream GXP2160” for GXP2160, “Grandstream GXP2170” for GXP2170 and “Grandstream GXP2135” for GXP2135.

PPPoE Account ID

Enter the PPPoE account ID.

PPPoE Password

Enter the PPPoE Password.

PPPoE Service Name

Enter the PPPoE Service Name.

Ipv4 Address

Enter the IP address when static IP is used.

Subnet Mask

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

Gateway

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

DNS Server 1

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

DNS Server 2

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

Preferred DNS Server

Enters the Preferred DNS Server for Ipv4.

Ipv6 Address Type

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

Static Ipv6 Address

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

Ipv6 Prefix Length

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

Ipv6 Prefix

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

DNS Server 1

Enter the DNS Server 1 for IPv6.

DNS Server 2

Enter the DNS Server 2 for IPv6.

Preferred DNS server

Enter the Preferred DNS Server for IPv6.

Network 🡪 Advanced Settings

802.1X mode

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

802.1X Identity

Enter the Identity information for the 802.1x mode.

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

MD5 Password

Enter the MD5 Password for the 802.1X mode.

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

802.1X CA Certificate

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

802.1X Client Certificate

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

HTTP Proxy

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

HTTPS Proxy

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

Bypass Proxy For

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

Layer 3 QoS for SIP

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

Layer 3 QoS for RTP

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

Release DHCP On Reboot

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

Enable DHCP VLAN

Enables auto configure for VLAN settings through DHCP.

Disabled by default.

Enable Manual VLAN Configuration

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

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0.

Layer 2 QoS 802.1p Priority Value

Assigns the priority value of the Layer2 QoS packets. The default value is 0.

PC Port Mode

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

PC Port VLAN Tag

Assigns the VLAN Tag of the PC port. The default value is “0”.

PC Port Priority Value

Assigns the priority value of the PC port. The default value is “0”.

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 Interval

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

Network 🡪 Remote Control

Action URI Support

Enable/disabled action URI feature on the phone.

Remote control Pop up window support

Indicates whether the phone is enabled to pop up allow remote control.

Action URI allowed IP list

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

CSTA Control

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

Network 🡪 Affinity Settings

Affinity Support

Allows communication with GS Affinity CTI application to manage telephone calls from computer. If enabled, a reboot is required to establish the communication. Default is “Disabled”.

GS Affinity CTI Application is available HERE and its User Guide HERE.

Preferred Account

Selects the account on which CTI support is enabled.

Network 🡪 Bluetooth Settings

Bluetooth Power

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

Handsfree Mode

Enable / disable Bluetooth handsfree feature.

Default setting is “Off”.

Bluetooth Name

Specifies the Bluetooth device name.

Network 🡪 OpenVPN® Settings

OpenVPN® Enable

Enable/Disable OpenVPN® feature. Default is No.

OpenVPN® Server Address

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

OpenVPN® Port

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

OpenVPN® Transport

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

OpenVPN® CA

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

OpenVPN® Certificate

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

OpenVPN® Client Key

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

OpenVPN® Cipher Method

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

OpenVPN® Username

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

OpenVPN® Password

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

Additional Options

Additional options to be appended to the OpenVPN® config file, separated by semicolons. For example, comp-lzo no;auth SHA256
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 SNMP

Enables/Disables the SNMP feature. Default settings is No.

Version

SNMP version.

Port

SNMP port (Default 161).

SNMP Trap IP

IP address of the SNMP trap receiver.

SNMP Trap Port

Port of the SNMP trap receiver (Default 162)

SNMP Trap Interval

The interval between each trap sent to the trap receiver

Trap version

Trap version of the SNMP trap receiver

Community

SNMP Community

SNMP Trap Community

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

SNMP Username

Username for SNMPv3

Security Level

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

Authentication Protocol

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

Privacy Protocol

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

Authentication Key

Enter the Authentication Key.

Privacy Key

Enter the Privacy Key.

SNMP Trap Username

User name for SNMPv3 Trap.

Trap Security Level

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

Trap Authentication Protocol

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

Trap Privacy Protocol

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

Trap Authentication Key

Enter the Trap Authentication Key

Trap Privacy Key

Enter the Trap Privacy Key.

Table 12: Network Page Definitions

Maintenance Page Definitions

Maintenance 🡪 Web Access

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.

New Password

Set new password for web GUI access as Admin. This field is case sensitive.

Confirm Password

Enter the new Admin password again to confirm.

Maintenance 🡪 Upgrade and Provisioning

Upgrade Firmware

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

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W pre/suffix changes, Always Skip the Firmware Check.

The default setting is “Always Check for New Firmware”.

Always Authenticate Before Challenge

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

Allow DHCP Option 43 and Option 66 Override Server

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 and HTTP server via option 66. Users can also use DHCP option 43 vendor specific 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.

Additional Override DHCP Option

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 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 SIP Server

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

3CX Auto Provision

Enables automatic provision feature on the phone when 3CX is used as the SIP server. The default setting is “Yes”.

Automatic Upgrade

Enables automatic upgrade and provisioning. The default setting is “No”.

Randomized Automatic Upgrade

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

Hour of the Day (0-23)

Defines the hour of the day to check the HTTP/TFTP/FTP server for firmware upgrades or configuration files changes. The default value is 1.

Day of the Week (0-6)

Defines the day of the week to check HTTP/TFTP/FTP server for firmware upgrades or configuration files changes. The default value is 1.

Disable SIP NOTIFY Authentication

Device will not challenge NOTIFY with 401 when set to “Yes”.

Default setting is “No”.

Firmware Upgrade Confirmation

If set to “Yes”, the phone will ask the user to upgrade. If there is no response, the phone will proceed with the upgrade.

If set to “No”, the phone will automatically upgrade without user input.

Default is Yes.

Config

Config Upgrade Via

Allows users to choose the config upgrade method: TFTP, FTP, FTPS, HTTP or HTTPS. 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.

Config File Prefix

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

Config File Postfix

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

XML Config File Password

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

Authenticate Conf 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 administration password. If it is missed or does not match the password, the phone system will not apply it.

Default setting is “No”.

Download Device Configuration

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

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

User protection

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 the Pvalues.

If “User protection” is ON, only those (normally user or admin) who have privilege can modify the configuration.

Download and Process All Available Config Files

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 specific and global configs).

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.

Download User configuration

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 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

Firmware Upgrade Via

Allows users to choose the firmware upgrade method:

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.

Firmware File Prefix

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

Firmware File Postfix

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

Maintenance 🡪 Syslog

Syslog Protocol

If set to SSL/TLS, the syslog messages will be sent through secured TLS protocol to syslog server.

Default setting is UDP.

Note: The CA certificate is required to connect with the TLS server.

Syslog Server

The URL or IP address of the syslog server for the phone to send syslog to.

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 Keyword Filtering

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

Send SIP Log

Configures whether the SIP log will be included in the syslog messages. The default setting is “No”.

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.

Auto Recover From Abnormal

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

USB Console Log

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

Maintenance 🡪 Language

Display Language

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

Default Input Selection

Configures the default input selection.

The default setting is “Multi-Tap”.

Multi-Tap: multi-tap to switch character;

Shiftable: select input from available characters.

Auto language download

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

Maintenance 🡪 TR-069

ACS URL

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

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

TR-069 Username

ACS username for TR-069.

TR-069 Password

ACS password for TR-069.

Periodic Inform Enable

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

Periodic Inform Interval

Sets up the periodic inform interval to send the inform packets to the ACS.

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.

Randomized TR069 Startup

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

Maintenance 🡪 Security Settings🡪 Security

Configuration via Keypad Menu

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 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”.

Factory Reset Security Level

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 “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.

Validate Server Certificates

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

Enable STAR key Keypad Locking

If set to “Yes”, the keypad can be locked by pressing and holding the STAR * key for about 4 seconds. A lock icon will show indicating the keypad is locked.

The default setting is “Yes”.

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.

Custom Certificate

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

Web Access Mode

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

Enable User Web Access

Administrator can disable or enable user web access. Default is Enabled.

HTTP Web Port

Configures the HTTP port under the HTTP web access mode.

HTTPS Web Port

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

Disable SSH

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

SSH Port

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

SSH Public Key

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 tool side.

Note: This will allow upcoming SSH access without password.

Web/Keypad/Restrict mode Lockout Duration

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

Web Session Timeout

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

Web Access Attempt Limit

Configures attempt limit before lockout. Default is 5. Range is 1-10.

Minimum TLS Version

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. Provisioning and sip transport don’t need reboot.

Default value is “TLS 1.1”

Maximum TLS Version

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. Provisioning and sip transport don’t need reboot.

Default value is “Unlimited”

Enable/Disable Weak Ciphers

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

  • 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

Trusted CA Certificates

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

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

Load CA Certificates

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

  • All Certificates: (Default) Both built-in and uploaded Certificates.

  • Default Certificates: Built-in Certificates;

  • Custom Certificates: Uploaded Certificates;

Maintenance🡪Security Settings🡪 Keypad Lock

Enable Keypad Locking

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

Keypad Lock Type

If set to “Functional Keys” then only functional keys will be locked and users still are 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.

Emergency

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

Maintenance 🡪 Packet Capture

Status

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

Capture Location

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

With RTP Packets

Defines whether the packet capture file contains RTP or not. Default is No

With Secret Key Information

When set to "Yes", the downloaded packet will include secret key, to decrypt the captured TLS packe

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.

Add Contact

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.

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.

Delete All Contacts

Deletes all contacts from phonebook.

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 Phonebook XML Download

Configures to enable phonebook XML download. Users could select HTTP/HTTPS/TFTP to download the phonebook file.

The default setting is “Disabled”.

HTTP/HTTPS User Name

The user name for the HTTP/HTTPS server.

HTTP/HTTPS Password

The password for the HTTP/HTTPS server.

Phonebook XML Server Path

Configures the server path to download the phonebook XML.

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

Phonebook Download Interval

Configures the phonebook download interval (in minutes).

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

Remove Manually-edited Entries on Download

If set to “Yes”, when XML phonebook is downloaded, the entries added manually will be automatically removed. The default setting is “Yes”.

Import Group Method

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

Sort Phonebook by

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

The default setting is “Last Name”.

Download XML
Phonebook

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

Upload XML Phonebook

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

Phonebook Key Function

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

Default search mode

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

Phonebook 🡪 LDAP

LDAP Protocol

Configures the LDAP protocol to LDAP or LDAPS. The default setting is “LDAP”. LDAPS is a feature to 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”.

Base

Configures the LDAP search base.

This is the location in the directory where the search is requested to begin.

Example:

dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com

User Name

Configures the bind “Username” for querying LDAP servers. Some LDAP servers allow anonymous binds in which case the setting can be left blank.

Password

Configures the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

LDAP Number Filter

Configures the filter used for number lookups. Examples:

(|(telephoneNumber=%)(Mobile=%) returns all records which has the “telephoneNumber” or “Mobile” field starting with the entered prefix;

(&(telephoneNumber=%) (cn=*)) returns all the records with the “telephoneNumber” field starting with the entered prefix and “cn” field set.

LDAP Name Filter

Configures the filter used for name lookups. Examples:

(|(cn=%)(sn=%)) returns all records which has the “cn” or “sn” field starting with the entered prefix;

(!(sn=%)) returns all the records which do not have the “sn” field starting with the entered prefix;

(&(cn=%) (telephoneNumber=*)) returns all the records with the “cn” field starting with the entered prefix and “telephoneNumber” field set.

LDAP Version

Selects the protocol version for the phone to send the bind requests. The default setting is “Version 3”.

LDAP Name Attributes

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:

gn

cn sn description

LDAP Number Attributes

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:

telephoneNumber

telephoneNumber Mobile

LDAP Display Name

Configures the entry information to be shown on phone’s LCD. Up to 3 fields can be displayed. Example:

%cn %sn %telephoneNumber

Max. Hits

Specifies the maximum number of results to be returned by the LDAP server. If set to 0, server will return all search results. The default setting is 50.

Search Timeout

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

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.

Lookup Display Name

Configures the display name when LDAP looks up the name for incoming call or outgoing call. This field must be a subset of the LDAP Name Attributes.

Example:

gn

cn sn description

Exact Match Search

With LDAP Lookup Incoming call, Outgoing call selected, DUT will performs LDAP search during incoming and outgoing call. If exact match search enabled, during the LDAP search, DUT will only get the 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 (Red)

Mode: Reserved (Green)

Call’s state

Light Indication

Call’s state

Light Indication

Offline

Off (Extension Board Icon: Off)

Offline

Off (Extension Board Icon: Off)

Idle

Off (Extension Board Icon: Idle)

Idle

Off (Extension Board Icon: Idle)

Trying

Solid Red

Trying

Solid Green

Talking

Solid Red

Talking

Solid Green

Proceeding

Solid Red

Proceeding

Solid Green

Incoming call

Flashing Red

Incoming call

Flashing Green

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 TransportUDPTCPTCP/TLS

Transport Only

Accept incoming request using UDP.
All outgoing request will go out using UDP.

Accept incoming request using TCP.
All outgoing request will go out using TCP.

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

Dual

Accept incoming request using both TCP and UDP.
All outgoing request will go out using UDP.

Accept incoming request using both TCP and UDP.
All outgoing request will go out using TCP.

Dual (Secured)

Accept incoming request using both TLS/TCP and UDP.
All outgoing request will go out using UDP.

Accept incoming request using both TLS/TCP and UDP.
All outgoing request will go out using TLS/TCP.

Dual (BLF Enforced)

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

Accept incoming request using both TCP and UDP.
All outgoing request will go out using TCP except for the BLF/Eventlist subscription the phone will add 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.
  1. Automatically update the configured data to the Dial Plan in Call Settings.
  2. 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.

http://192.168.5.136/export/screen/20170104-041940.png
Figure 5: Transfer Softkey During Call

http://192.168.5.136/export/screen/20170104-042042.png
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.

C:\Users\GS_SW\AppData\Local\Microsoft\Windows\INetCache\Content.Word\New Bitmap Image.bmp
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.

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.

58539
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)

https://lh3.googleusercontent.com/-1_U7cs7mOys/WGvKKJ9VpHI/AAAAAAAAAqc/Bb017vFJOiABF-BHIqnvBPd_FBBsYnQ3QCL0B/h257/2017-01-03.png
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

  1. On screensaver.xml file enter following tags:
<screensaver>
<image path="http://server_IP_address/picture1.jpg" />
<image path="http://server_IP_address/picture2.jpg" />
<image path="http://server_IP_address/picture3.jpg" />
<image path="http://server_IP_address/picture4.jpg" />
<image path="http://server_IP_address/picture5.jpg" />
</screensaver>
  1. Put picture files on HTTP server directory. Please refer to following example using HFS HTTP server:
Figure 22: HFS HTTP Server

  1. 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 Class 2 of version 2.1. 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

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 VariablesDescription

$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-ValueWeb UI OptionValue 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

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 OptionDescription

Destination Name

Identify the destination name. It must be unique.

Protocol

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

Enable SSL

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

Destination Address

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

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 Name

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

Extra Attribute Value

Configure extra attribute’s value reserved for protocol specific attributes such as “abc@gmail.com” 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 Value

Destination

Value Format

P9910

Destination 1

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=username1&password=password1&from=&to=to1&domain=gmail.com&extraAttrName1=extraAttrValue1&extraAttrName2=extraAttrValue2

Example 2 – Destination 2 with protocol SMTP and 3 extra Attributes configured:

P9911=serverName=destination2&protocol=SMTP&serverAddress=smtps://smtp.gmail.com&port=465&user=username2&password=password2&from=username2&to=to2&domain=&extraAttrName1=extraAttrValue1&extraAttrName2=extraAttrValue2&extraAttrName3=extraAttrValue3

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

P9911

Destination 2

P9912

Destination 3

P9913

Destination 4

P9914

Destination 5

P9915

Destination 6

P9916

Destination 7

P9917

Destination 8

P9918

Destination 9

P9919

Destination 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 OptionDescription

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.

Subject

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

Message

Configures the message body or the outbound notification.

Extra Attribute Name

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

Extra Attribute Value

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

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.

EventDynamic Attribute NameDynamic Attribute Description

Call_Missed

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 time stamp when the missed call event occurs

DND

status

This is for DND status. The value can be “enabled” or “disabled”

Call_Forward

callType

This is for the type of the call. The 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”

OAMLoginSource

OAM login source. The value can be “SSH” or “WebGUI”

OAMLoginFromIP

OAM login From IP address. The value is the IP address of the PC who will log in phone’s web UI or SSH

OAMLoginCode

OAM login result code. The value can be “succeeded” or “failed”

time

OAM login time stamp

OAM_Lockout

OAMUser

OAM user name such as “admin”

OAMLoginSource

OAM login source. The value can be “SSH” or “WebGUI”

OAMLoginFromIP

OAM login From IP address. The value is the IP address of the PC who will log in phone’s web UI or SSH

OAMLockoutCode

OAM lockout result code. The value can be “locked” or “unlocked”

OAMLockoutTime

OAM lockout time stamp

Incoming_Call

callingNumber

Calling party number

callType

Type of the call. The 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 time stamp when the incoming call event occurs

Outgoing_Call

callType

Type of the call. The 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

time

The time stamp when the outgoing call event occurs

Call_Established

callType

Type of the call. The 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

startTime

The time stamp when outgoing call event occurs

Call_Terminated

callType

Type of the call. The 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

startTime

The time stamp when the call is established

Call_Forward_Status

duration

The call duration

account

The account number associated with the call forward status change

forwardNumberAll

The forward number for Call Forward All

forwardNumberBusy

The forward number for Call Forward Busy

forwardNumberNoAns

The forward number for Call Forward No Answer

Call Hold

callType

Type of the call. The 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

startTime

The time stamp when the call is on hold

Call_Resume

callType

Type of the call. The 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

startTime

The time stamp when the call is resumed

Blind_Transfer

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 time stamp when the call is transferred

transferName

Transferred party name

transferNumber

Transferred party number

Attended_Transfer

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 time stamp when the call is transferred

transferName

Transferred party name

transferNumber

Transferred party number

Register_Status

registerStatus

Account register status. The value can be “registered” or “unregistered”

Bootup_Complete

N/A

N/A

The dynamic attributes in this row are common attributes that can be applied to all events

mac

MAC address of the phone

phone_ip

IP address of the phone

program_version

Software version of the phone

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-ValueNotificationValue Format

P9920

Notification 1

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

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.

P9921

Notification 2

P9922

Notification 3

P9923

Notification 4

P9924

Notification 5

P9925

Notification 6

P9926

Notification 7

P9927

Notification 8

P9928 P9929

P9993

P9994

Notification 9

Notification 10

Notification 73

Notification 74

P9995

Notification 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 NameMode StringMode 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

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.

VPK ModeStateIconLED Status

LINE

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

OFF

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

OFF

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
(No IM, No Voice Mail)

OFF

Registered + Call Forward All
(No IM, No Voice Mail)

OFF

Unregistered + Call Forward Delay + Call Forward Busy (No IM, No Voice Mail)

OFF

Registered + Call Forward Delay + Call Forward Busy (No IM, No Voice Mail)

OFF

Unregistered + Call Forward Delay
(No IM, No Voice Mail, No Call Forward Busy)

OFF

Registered + Call Forward Delay
(No IM, No Voice Mail, No Call Forward Busy)

OFF

Unregistered + Call Forward Busy
(No IM, No Voice Mail, No Call Forward Delay)

OFF

Registered + Call Forward Busy
(No IM, No Voice Mail, No Call Forward Delay)

OFF

Registered + Ringing

Flashing RED

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)

Registered + Processing

Flashing 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/
Eventlist BLF

Offline, Unknown

OFF

Terminated

GREEN

Proceeding

RED

Ringing (Early)

Flashing RED

Trying

Flashing GREEN

Confirmed

RED

Presence Watcher

Offline, Unknown

OFF

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

Menu

OFF

Voice Mail

Account not registered

OFF

Account Registered
(No new voice mail)

OFF

Account Registered
(Have new voice mail)

OFF

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.

http://192.168.5.136/export/screen/20170104-042351.png
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.

http://192.168.5.136/export/screen/20170104-044302.png
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.

vpk 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.

C:\Users\Soukaina\Desktop\20170323-071202.png
Figure 38: VPK/MPK to Park the Call

C:\Users\Soukaina\Desktop\20170323-071315.png
Figure 39: Dial Park Space directly
C:\Users\Soukaina\Desktop\20170323-071406.png
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.

C:\Users\Soukaina\Desktop\Capture110.PNG
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.

  1. 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 1.0.11.64

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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