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

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 (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 context-sensitive Softkeys, 5 navigation/menu keys, 11 dedicated function keys for MESSAGE (with LED indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, 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 / 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

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

WebUI Language: English 简体中文 (Simplified Chinese) 繁體中文 (Traditional Chinese) العربية (Arabic) Czech Deutsch (German) Español (Spanish) Français (French) עברית (Hebrew) Hrvatski (Croatian) Magyar (Hungarian) Italiano (Italian) 日本語 (Japanese) 한국어 (Korean) Nederlands (Dutch) Polski (Polish) Português (Portuguese) Русский (Russian) Slovenščina (Slovenian) Türkçe (Turkish)

Upgrade/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 Part 15 ClassB, EN55022 ClassB, EN61000-3-2, EN61000-3-3, EN55024, EN60950-1, EN62479
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950

Protocols/Standards

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

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 context-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 the 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

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

WebUI Language: English 简体中文 (Simplified Chinese) 繁體中文 (Traditional Chinese) العربية (Arabic) Czech Deutsch (German) Español (Spanish) Français (French) עברית (Hebrew) Hrvatski (Croatian) Magyar (Hungarian) Italiano (Italian) 日本語 (Japanese) 한국어 (Korean) Nederlands (Dutch) Polski (Polish) Português (Portuguese) Русский (Russian) Slovenščina (Slovenian) Türkçe (Turkish)

Upgrade/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 art 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)

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 / 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 & MD5-sess based authentication, AES based secure configuration file, SRTP, TLS, 802.1x media access control

Multi-language

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

WebUI Language: English 简体中文 (Simplified Chinese) 繁體中文 (Traditional Chinese) العربية (Arabic) Czech Deutsch (German) Español (Spanish) Français (French) עברית (Hebrew) Hrvatski (Croatian) Magyar (Hungarian) Italiano (Italian) 日本語 (Japanese) 한국어 (Korean) Nederlands (Dutch) Polski (Polish) Português (Portuguese) Русский (Russian) Slovenščina (Slovenian) Türkçe (Turkish)

Upgrade/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 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)

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, supported

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

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

WebUI Language: English 简体中文 (Simplified Chinese) 繁體中文 (Traditional Chinese) العربية (Arabic) Czech Deutsch (German) Español (Spanish) Français (French) עברית (Hebrew) Hrvatski (Croatian) Magyar (Hungarian) Italiano (Italian) 日本語 (Japanese) 한국어 (Korean) Nederlands (Dutch) Polski (Polish) Português (Portuguese) Русский (Russian) Slovenščina (Slovenian) Türkçe (Turkish)

Upgrade/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)

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

Bluetooth supported

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

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

WebUI Language: English 简体中文 (Simplified Chinese) 繁體中文 (Traditional Chinese) العربية (Arabic) Czech Deutsch (German) Español (Spanish) Français (French) עברית (Hebrew) Hrvatski (Croatian) Magyar (Hungarian) Italiano (Italian) 日本語 (Japanese) 한국어 (Korean) Nederlands (Dutch) Polski (Polish) Português (Portuguese) Русский (Russian) Slovenščina (Slovenian) Türkçe (Turkish)

Upgrade/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)

Call History

Displays Local call logs:

Status

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

Contacts

The Contacts sub menu includes the following options: 

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 includes the following options :

Preference

Preference sub menu includes the following options:

Enables/disables Do Not Disturb on the phone.

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.

Ring Tone: Configures different ring tones for incoming call.

Ring Volume: Adjusts ring volume by pressing left/right arrow key.

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.

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.The default setting is “Auto”.

Default Input Selection: Selects the Input mode from Multi-Tap and Shiftable. The default setting 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.

Configures the date and time on the phone.

Allow DHCP Option 42 to override NTP server
Allow DHCP Option 2 to override Time Zone setting
Time Settings

Specifies the phonebook search mode to QuickMatch or ExactMatch. The default mode is “QuickMatch”.

Phone

Phone sub menu includes the following options:

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.

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

Configures the ringtone for each SIP account.

System

IP Setting: Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE account ID and password; Configures static IP address, Netmask, Gateway, Preferred DNS Server, 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 Status: Displays the status of Bluetooth

Bluetooth MAC: Displays the 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.

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.

Configures the language download:

Auto Language Download
Language Download

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 Diagnostics: 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.

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.

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.

Restore the phone to factory default settings

Reboot

Reboots the phone.

User Level

User

Password

Web Pages Allowed

End User Level

user

Set by admin

Only Status and Basic Settings

Administrator Level

admin

admin

Browse all pages

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.

Affinity Broadcast

Affinity broadcasting status

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.

Individual Certificate Generation

Displays the Individual certificate generation.

Eg.: GEN 1 ( SHA 256 + RSA )

Security Version

Displays the security version.

Eg.: GS011.001.001

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;

Special Feature

OpenVPN® Support: displays if the device supports OpenVPN®.

System Time

System Up Time: system up time since the last reboot.

System Time: current system time on the phone system.

Service Status

GUI, Phone and CPE 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

Softkeys

Status → Extension Boards Status

Extension x Keys

Status → Open Sources licenses

Downloads the gpl open source licenses.

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.

Secondary 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.

SIP Authentication ID

SIP service subscriber’s Authenticate ID used for authentication.

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

SIP Authentication 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.

Voicemail Access Number

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.

Monitored Access Number

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

Picture

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

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

Account Display

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:

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 Failover Mode

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

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 IP 1

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

Backup IP 2

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.

Support Rport (RFC 3581)

This option enables the utilization of symmetric response routing. When enabled, the “rport” field gets appended to the Via header in the SIP Request. The necessary data is then extracted from the SIP 200OK Response during a SIP Register operation, allowing the rewriting of SIP Contact information and its application in subsequent SIP Requests.

Enabled by Default.

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.

Registration Retry Wait Time upon 403 Forbidden

Configures the retry interval for SIP registration attempts when registration fails due to a 403 Forbidden error.

The valid range is 1 to 3600 in second.

The default value is 1200.

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”.

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”.

Add Auth Header On RE-REGISTER

If enabled, an Authorization header will be added in the RE-REGISTER request.

The default setting is “Yes”.

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.

The 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.

The default setting is “Default”.

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.

The default setting is “Yes”.

Use P-Early-Media Header

P-Early-Media header (RFC 5009) is used to negotiate early media direction (e.g., sendrecv, inactive) during call setup. It informs the caller whether to expect audio before the call is answered. This helps the IP phone decide whether to play local ringback or wait for RTP.

Disabled by default

Use X-switch-info Header

Configure whether X-switch-info Header is included in SIP REGISTER request.

The default setting is “No”.

Use MAC Header

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

Account x → SIP Settings → Advanced Features

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 Eventlists

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:

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.

Call Park Feature Code

Configure the feature access code for parking current call to parking lot or another extension.

Call Park Retrieve Feature Code

Configure the feature access code for retrieving the parked call on accounts or parking lot.

Dialog Info Call Park Retrieve

It enables or disables the IP phone implements directed call park retrieve through SIP signals (such as Replaces header) for a specific account.

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.

Reboot on check-sync event without “reboot=”

If this option is enabled, the phone reboots when it receives a SIP NOTIFY check-sync event without “reboot=” header. If it’s disabled, the phone will provision instead.

The default setting is “Enabled’.

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.

The 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.

The 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.

The 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”.

Trusted Domain Name List

Configure the list of trusted domain names, which supports filling in the SAN list used only for domain name verification in TLS to obtain certificates. If it matches any item in the trusted domain name list, the certificate is trusted.

By default, the remote proxy domain name and SIP server domain name are trusted.

This field allows numbers/letters/-/./*. It supports wildcard domain names, such as “*.mycompany.com“, and trusts any domains ending with “.mycompany.com“.

Validate Certificate Chain

Validate certification chain when TCP/TLS is configured.

The 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.

The 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”.

Configures to enable or disable multiple m lines 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.

The 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.

The 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.

The 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:

The 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.

The default setting is “{ x+ | +x+ | *x+ | *xx*x+ }”.

Dial Plan Rules:

x – any digit from 0-9

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

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):

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.

Bypass Dial Plan

Allows selected items to bypass the dial plan.

Multi-Purpose Keys are enabled by default.

Ringback Tone at No Early Media

Plays a ringback tone when early media RTP is not received while P-Early-Media support is enabled.

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”.

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.

The 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.

The 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:

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 Hangup

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:

Blind Transfer Wait Timeout

Defines the timeout (in seconds) for waiting SIP frag response in blind transfer.

The valid range is 30 to 300.

Transfer to VM feature code

Configures the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the collected digits will be sent out.

Note: The default value for this setting is *.

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.

Key As Send

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. 

The default setting is “Default”.

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 or upload a new ringtone.

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:

Notes:

Custom ringtone can be defined for matching rule of Caller ID beginning with “+” :

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

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 30 to 3600.

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 “Yes”.

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 Intercom Auto Answer

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.

Off: Configures DND feature code to turn off DND.

Call Forward Always

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

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.

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 No Answer

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

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

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

Call Forward No Answer Timeout: 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.

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. The 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. The 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.

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.

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:

Password needs at least 9 characters and 3 of the following options:

1) Numeric (0-9)

2) Capital letters (A-Z)

3) Lower case (a-z)

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

Default setting is “No”.

Customized Station Label

Configures a customized station label that will be displayed on the status bar of the LCD screen. If this field is left blank (default), the banner will show the display name of the preferred account/first available account.

Enable Public Mode

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

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.

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 Limit

Configures Broadsoft contacts download limit. Default is 100. Range is 0-2000

BroadSoft Contacts Search Limit

Configures Broadsoft contacts Search limit. Default is 100. The 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

BroadSoft User ID: Configure the Username for the BroadWorks Xsi server.

Login Password: Configure the password for the BroadWorks Xsi server.

SIP Credentials

SIP Authentication ID: Configure SIP Username for the BroadWorks Xsi server.

BroadSoft User ID: Configure SIP User ID for the BroadWorks Xsi server.

SIP Authentication Password:Configure SIP Password for the BroadWorks Xsi server.

Sort Phonebook by

Configures to sort phonebook based on the selection of first name or last name.

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 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.

Mute Key Functions While Idle

If set to “DND”, clicking the MUTE key while idle will put the phone in DND mode. If set to “Idle Mute”, clicking the MUTE key while idle will cause the phone to be muted automatically when answering incoming calls.

Show SIP Error Response

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

Filter Characters

This feature allows users to filter out specific characters in dial-out calls. such as “(“, “+”…

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”.

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

Enable Enhanced Acoustic Echo Canceller

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

Disable Hook Switch

When set to “Yes”, the hook switch is disabled completely; When set to “For Answering Call”, the hook switch cannot be used for answering call.
The default is “No”.

Disable Speaker Key

Group Call Park Softkey Feature Code:

The default setting is “No”.

Contact Source Priority

Configure the priority of the ID source displayed on the phone when incoming/outgoing calls. Select the relevant ID sources and click the Up/Down arrow on the right to adjust the order.
The available sources :

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”.

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”.

Predictive Dialing Feature

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

Predictive Dialing Source

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

Onhook Dial Barging

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

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 Dial 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 Keypad

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 Keypad Expiration

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 Auto Redial

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

Auto Redial Times

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

Auto 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”.

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 Missed Call Notification

Allows users to show/hide the notification popup for missed calls.

The default is “No” which will hide call notification popup.
Note: Currently the manually rejected calls are counted as missed calls.

Return Code When Refusing Incoming Call

When refusing the incoming call. The phone will send the selected type of SIP message of the call.

The default setting is “Busy 486”.

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.

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.

The default setting is “No”.

Ring For Call Waiting

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

The default is disabled.

Enable Diversion Information Display

If set to “Yes”, the “Diverted From” message is displayed when receiving a forwarded call.
If set to “No”, the “Diverted From” message is not displayed for forwarded calls.
The default is “Yes”.

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.

Show On Hold Duration

When this option is set to “Yes”, users can see how long their call has been hold.

The default is “No”.

Disable Mute Key in Call

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

In-call Dial Number on Pressing Transfer Key

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

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 Transfer

Enables/disables transfer feature. If disabled, call transfer will not be possible.

The default setting is “No”.

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.

Enable Call Park Softkey

Allows call park service to use call park related softkeys.

The default setting is “Yes”.

Call Park Mode

Select different implementations of call park softkeys. Eg:
Call Park under XSI mode needs to configure XSI service;
Call Park under FAC mode needs to configure related FACs, which have lower priority than BLF call park related FACs;
Call Park under Transfer mode can also use similar FACs method to Call Park under FAC mode but its FAC is optional to park/blind transfer current call to parking lot or target number.

The default setting is “FAC”.

Call Park Softkey Feature Code

Configure the feature access code for parking current call to parking lot or another extension.

Call Park Retrieve Softkey Feature Code

Configure the feature access code for retrieving the parked call on accounts or parking lot.

Group Call Park Softkey Feature Code

Configure the feature access code for parking current call to the group current user is in.

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.

The default is “Yes”.
Notes:

Preserve DND Status

Configures whether DND status should be saved after a reboot.

Return Code Upon DND

When DND is enabled, the phone will send the selected type of SIP message.

The default setting is “Busy 486”.

DND Override

Configures which incoming calls are allowed to override the Do Not Disturb (DND) status when it’s enabled.

The default setting is “Off”.

Disable Conference

Disables the Conference function.
The default setting is “No”.

Only Same Account in Conference

If enabled, only calls associated with the same account are allowed to build a conference.

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 Only list

Configures play BLF sound notification only for the list below.

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 (,)

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.

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.

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.
Note: The mode and label of the VPK/MPK/Extension board keys can be retrieved by entering the following API URL in the web browser:
{IP}/cgi-bin/api-get_vpk_status
To retrieve the BLF/SCA state of monitored keys, enter the following API URL instead:
{IP}/cgi-bin/api-get_vpk_status?type=monitor

Default Call Log Type

Configures the default call log displayed after selecting Menu → Call History. By default, both BroadSoft and local call log lists will be displayed.

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.

Settings → Call History

Delete

Users can select an entry, then click “Delete” to remove it from the call history.
The call information displayed on the page is the name, number and time.

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.

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).

The 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
Enable DND
Disable DND
Enable Call Forward
Disable Call Forward
Open Forward Always
Close Forward Always
Open Call Forward Busy
Close Call Forward Busy
Open Call Forward No Answer
Close Call Forward No Answer
Blind Transfer
Attended Transfer
Transfer Completed
Transfer failed
Hold Call
Unhold Call
Mute Call
Unmute Call
IP Change
Auto Provision Completed

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”.

Default Mode:
When the phone is in idle, press HEADSET button to off hook the phone and make calls by using headset. Headset icon will display on the screen in dialing/talking status.
When there is an incoming call, press HEADSET button to pick up the call using headset.
When there is an active call using headset, press HEADSET button to hang up the call.
When Speaker/Handset is being used in dialing/talking status, press HEADSET button to switch to headset. Press it again to hang up the call. Or press speaker/Handset to switch back to the previous mode.
Toggle Headset/Speaker:
When the phone is in idle, press HEADSET button to switch to Headset mode. The headset icon will display on the left side of the screen. In this mode, if pressing Speaker button or Line key to off hook the phone, headset will be used.
When there is an active call, press HEADSET button to toggle between Headset and Speaker.

Headset Type

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

EHS Headset Ringtone

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.

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.
Note: Daylight Saving Time (DST) is no longer added on Mexico city timezone.

To allow DHCP option 2 to override the configured Time Zone Setting check the option.

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:

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 Clock Widget

Shows the Clock Widget on the main screen.
The default setting is “Yes”.

Show Date and Time on Status Bar

Allows users to display time and date on the top panel of the LCD screen, you can select to show time only or to show both Date and Time.

The default setting is “Show Time only”. 

Note: For GXP2135 and GXP2170, the time and date will be displayed on top of the LCD when the top VPK on the right side of the LCD screen is not configured.

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.
The 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.

The 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. 

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

Specifies the wallpaper source mode: Default, Download, USB, Uploaded and Color Background. User could upload a wallpaper source into your phone, download it from file server with the server path, choose a solid color wallpaper in HEX format, or plug your USB drive to upload the wallpaper

Note: USB upload is only for GXP2140, GXP2160 and GXP2170

Wallpaper Server Path

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

Color Background

Enter the code color based on your preference in HEX format. E.g. #000000
Reference: http://htmlcolorcodes.com
Note: This option will take effect  when wallpaper source is “Color Background”.

Screensaver

Configures the screensaver display on the LCD screen. If set to “On if no VPK is active”, the screensaver is displayed only when no VPK is active.
A VPK is considered active when:
In Early (ringing), Trying (dialing) and Confirmed (talking) state.
Configured with “BLF”, “Eventlist BLF” or “Presence” mode. 

Note: this option is also available on the LCD under Menu→Preference→Appearance.

The default setting is “Yes”.

Screensaver Source

Sets the location where screensaver is loaded from. If from USB, please have a folder named “screensavers” containing your pictures.
Note: USB upload is only for GXP2140, GXP2160 and GXP2170

Show Date and Time

Configures the clock display the LCD screensaver.
The default setting is “Yes”.

Screensaver Timeout

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

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.
The default setting is “0”.

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 under “BLF LED Pattern Explanation Form” section on the Web UI or in [BLF LED Patterns] table in this Adminstration guide.

Disable VM/MSG power light flash

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

Note: In addition to the LED indicator that notifies users of the voicemail status, users can also retrieve the voicemail status by using the following API command: {IP}/cgi-bin/api-get_vm_status. Example result:  

{
“response”: “success”,
“body”: [
{
“id”: 1,
“sip_id”: “1087”,
“vm(new/total)”: “0/1”
}
]
}
This result indicates that account 1 with SIP ID 1087 has 0 new voicemails and 1 total voicemail.

Call Forward Status Indicator

Enables/disables unconditional call forward status LED indicator.  if set to “Yes” the LED light will turn green when call forward is deactivated, and red when call forward is activated.
The default setting is “No”.

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.
The default setting 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.
The default setting is “5”.

Lock Speaker Volume

Locks volume adjustment when the option is enabled.

Default Ringtone

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

Alert-Info Remote Ringtone Download Timeout

Configures how long before the phone falls back to using the local ringtone if the Alert-Info remote ringtone fails to download.
Valid range is 1 to 5.

Total Number of Custom Ringtone Update

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

The default setting is “3”.

Settings → Programmable Keys

Virtual Multi-Purpose Keys Settings

Auto Provision List Starting Point

Configures if “Extension Boards” or “VPK” will be used first when the phone is being automatically provisioned with Eventlist BLF.
The default setting is “Extension Boards”.
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.

Enable Push-To-Talk
Configures whether to enable Push-To-Talk. When enabled, long pressing on an Intercom VPK would switch to the Push-To-Talk functionality that initiates a call and ends the call when the key is released.

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/MPK
Allows users to configure “Transfer” VPK or MPK 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:

Softkeys Settings

More Softkey Display Mode
Allows users to choose from the original Toggle mode or the enhanced Menu mode.

With the enhanced Menu mode, the MORE softkey now will not need the user to tap multiple times on MORE to get to the next pages, instead, pressing MORE will have a popup window and allow users to choose from the list. With Toggle mode, users need to press the MORE softkey to switch between options.

Show Target Softkey
Allows users to remove target softkey by toggling the Yes/No option during the off-hook dial screen and transfer screen.

Custom Softkey Layout
Enables/Disables custom softkey layout.

Enforce Softkey Layout Position
Whether to enforce the custom softkey layout position.

When enabled, GUI will still preserve the space if the configured softkey is unable to show.

Hide System Softkey on Main Page
Configures to hide the system-generated softkey (Next, History, ForwardAll, Redial) on the main page. The default value is none.

Dialing State
Custom softkey layout when the device is under Dialing State.

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

Onhook Dialing State
Custom softkey layout when the device is under On-hook Dialing State.

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

Ringing State
Custom softkey layout when the device is under Ringing State.

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

Calling State
Custom softkey layout when the device is under calling State.

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

Call Connected State
Custom softkey layout when the device is under call connected State.

Available Softkeys: Phonebook(BT), BT On/Off, EndCall, ReConf, ConfRoom, ConfCall, Cancel, New Call, Swap, Transfer, Trnf>VM, DialDTMF, BS-Ccenter, Record On/Off(UCM), Record On/Off, CallPark(UCM), PrivateHold, CallPark.

Conference Connected State
Custom softkey layout when the device is under Conference Connected State.

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

On Hold State
Custom softkey layout when the device is under On Hold State.

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

Call Failed State
Custom softkey layout when the device is under Call Failed State.

Available Softkeys: EndCall, ReConf, ConfRoom.

Transfer State
Custom softkey layout when the device is under Transfer State.

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

Conference State
Custom softkey layout when the device is under Conference State.

Available Softkeys: BT On/Off, Cancel, Dial, Backspace.

Idle Screen Softkeys

Assigns a function to the corresponding Softkeys. GXP2140, GXP2160 and GXP2170 supports 3 configurable Softkeys; GXP2130/GXP2135 supports 2 configurable Softkeys.

Note: The first and last Softkeys are reserved for Exit/More functionality.

The key mode options are:

Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed.

Speed Dial via Active Account
Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is off-hook and account 2 is active, it will call the configured Speed Dial number using account 2.

Voicemail
Select Account & enter the Voice Mail access number in the Value field.

Call Return
The last answered calls can be dialed out by using Call Return.

The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.

Intercom
Select Account, and enter the extension number in the Value field to do the intercom.

LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings.

The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field.

For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:

Description -> ou=video,ou=SZ,dc=grandstream,dc=com

Value -> sn=Li

Since the Base for LDAP server configuration is “dc=randstream,dc=com”, “ou=video,ou=SZ” is added to narrow the LDAP search scope. “sn=Li” is the example to filter the last name.

Call Log
Select Account and enter the account number in the Value field to access the Call Log of that selected account.

Menu
Select this feature in order to display the Menu from the MPK buttons, no field is required for configuration.

Information
Select this feature in order to display the Information popup to show the firmware version, MAC address, IP address, and IP Settings from the MPK buttons, no field is required for configuration.

Message
Select this feature in order to display the Message menu from the MPK buttons, no field is required for configuration.

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.

Voicemail
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

Settings → 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.

Voicemail
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 VPK button to be used to lock/unlock the keypad.
GDS OpenDoor Presss the VPK to open GDS door systems remotely.

Settings → Web Service

Use Auto Location Service

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

Username

Configures the Username of authentication to download the XML App file.

Password

Configures the password of authentication to download the XML App file.

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

Configures the background color in HEX format.

The default is 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.

Settings → E911 Service

Enable E911

Enable or disable E911.
The default setting is “No”.

HELD Protocol

Configures HELD transfer protocol.

The default setting is “HTTP”.

HELD Synchronization Interval

The valid synchronization interval is between 30 to 1440 minutes. The synchronization is off when the interval is 0.

Location Server

Configures the primary Location Information Server (LIS) adderss.

Location Server Username

Configures the username of the primary Location Information Server (LIS).

Location Server Password

Configures the password of the primary Location Information Server (LIS).

Secondary Location Server

Configures the secondary Location Information Server.

Secondary Location Server Username

Configure the username of the secondary Local Information Server.

Secondary Location Server Password

Configure the password of the secondary Location Information Server.

HELD Location Types

Configure “locationType” element in the location request.

HELD Use LLDP Information

If set to “Yes”, the information from the LLDP-support switch is used to generate ChassisID and portID; otherwise.
The default setting is “No”.

HELD NAI

If set to “Yes”, Network Access Identifier (NAI) is included as a device identity in the location request sent to the Location Information Service.
The default setting is “No”.

HELD Identity

The HELD Identity refers to Hierarchical Mobile IPv6 (HMIPv6) Enhanced Location-based Discovery (HELD) protocol.
This protocol is utilized to determine the location of a mobile device when making an emergency call, such as a 911 call, over an IP-based network.
Up to 10 HELD identities can be configured.

HELD Identity Value

Defines the HELD Identity value.

E911 Emergency Numbers

A user can configure multiple emergency numbers separated by the delimiter symbol “;”.
The default value is “911”.

Geolocation-Routing Header

If set to “Yes”, E.911 INVITE message includes the “Geolocation-Routing” header with the value “Yes”.

The default setting is “No”.

Geolocation Header Format

This setting specifies how the phone encodes and inserts location information (like building, floor, room) into SIP INVITE messages during emergency (911) calls.

The default setting is “Legacy”.

Priority Header

If set to “Yes”, E.911 INVITE messages includes the “Priority” header with the value “emergency”.
The default setting is “No”.

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 IDs.
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

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 (64)

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”.