GXP21xx Series - Administration Guide

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

Feature Keys

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

Voice Codec

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

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets)

Telephony Features

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

HD audio

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

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes

QoS

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

Security

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

Multi-language

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

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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

Auxiliary Ports

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

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Extension Module

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

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes

QoS

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

Security

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

Multi-language

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

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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

Auxiliary Ports

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

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes

QoS

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

Security

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

Multi-language

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

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD

Bluetooth

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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

Auxiliary Ports

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

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Extension Module

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

Base Stand / Wall Mountable

Yes, allow 2 angle positions

QoS

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

Security

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

Multi-language

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

Yes, Bluetooth V2.1

Feature Keys

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

Voice Codec

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

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets)

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Base Stand / Wall Mountable

Yes, allow 2 angle positions

QoS

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

Security

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

Multi-language

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:

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

Status

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

• Account status.
• Network status.

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

• System Information

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

Contacts

Contacts sub menu includes the following options:

• Local Phonebook
• Local Group
• LDAP Directory

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

Messages

Message sub menu include the following options :

• Instant Message

Displays received instant messages;

• Voice Mails

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

Preference

Preference sub menu includes the following options:

• Do Not Disturb

Enables/disables Do Not Disturb on the phone.

• Star Key Lock

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

• Sounds
  • Ring Tone

Configures different ring tones for incoming call.

• Ring Volume

Adjusts ring volume by pressing left/right arrow key.

• Appearance
  • Active LCD Brightness

Adjusts active LCD brightness by pressing left/right arrow key

• Idle LCD Brightness

Adjusts idle LCD brightness by pressing left/right arrow key

• Active LCD Timeout

Adjusts the minute of active backlight timeout.

• Screensaver

Enables/Disables Screensaver

• Screensaver Timeout

Configures the minutes of idle before the screensaver activates.

Valid range is 3 to 6.

• Language and Input
  • Display Language

Selects the language to be displayed on the phone’s LCD. Users could select Automatic for local language based on IP location if available. By default, it is Auto.

• Default Input Selection

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

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

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

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

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

• Search Mode

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

Phone

Phone sub menu includes the following options:

• SIP

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

• Call Features

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

System

System sub menu includes the following options:

• Network
  • IP Setting

Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE account ID and password; Configures static IP address, Netmask, Gateway, DNS Server 1 and DNS Server 2.

• 802.1X

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

• Layer 2 QoS

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

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

Displays the status of Bluetooth

• Bluetooth MAC

Displays the GXP phone’s MAC address

• Power

Turns on/off the Bluetooth feature.

• Handsfree Mode

Enables/Disables Handsfree mode

• Bluetooth Name

Specifies GXP phone name when discovered by other Bluetooth devices.

• Start Scan

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

• Upgrade
  • Firmware Server

Configures firmware server for upgrading the phone.

• Config Server

Configures config server for provisioning the phone.

• Upgrade Via

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

• Start Provision

Starts Provision immediately.

• Language Download
  • Auto Language Download
  • Language Download
• Factory Functions
  • Diagnostic Mode
    All LEDs will light up. All keys’ name will display in red on LCD screen before diagnosing. Press any key on the keypad to diagnose the key’s function. When done, the key’s name will display in blue on LCD. Lift and put back the handset to exit diagnostic mode.
  • Audio Loopback

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

• LCD on/off

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

• LCD Diagnostic

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

• Certificate Verification

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

• UCM Detect

Detect/connect UCM server to process auto-provision. Manually input the IP and port of the UCM server phone wants to bind with; Or select from the available UCM server in network.

• Authentication
  • Admin Password

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

• End User Password

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

• Settings

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

• Operations
  • Factory Reset

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

Reboot

Reboots the phone.

user

123

admin

admin

Status 🡪 Account Status

Account

Account index.

For GXP2130: up to 3 SIP accounts

For GXP2140: up to 4 SIP accounts

For GXP2160: up to 6 SIP accounts

For GXP2170: up to 6 SIP accounts

For GXP2135: up to 4 SIP accounts

SIP User ID

Displays the configured SIP User ID for the account.

SIP Server

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

SIP Registration

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

Status 🡪 Network Status

MAC Address

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

IP Setting

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

Ipv4 Address

The Ipv4 address obtained on the phone.

Ipv6 Address

The Ipv6 address obtained on the phone.

OpenVPN IP

The OpenVPN IP obtained on the phone.

Subnet Mask

The subnet mask obtained on the phone.

Gateway

The gateway address obtained on the phone.

DNS Server 1

The DNS server address 1 obtained on the phone.

DNS Server 2

The DNS server address 2 obtained on the phone.

PPPoE Link Up

PPPoE connection status.

NAT Type

The type of NAT connection used by the phone.

NAT Traversal

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

Status 🡪 System Info

Product Model

Product model of the phone.

Part Number

Product part number.

Software Version

• Boot: boot version number;
• Core: core version number;
• Base: base version number;
• Prog: program version number. This is the main firmware release number, which is always used for identifying the software system of the phone;
• Locale: locale version number;
• Recovery: recovery version number.

IP Geographic Information

• City: displaying phone location;
• Language: displaying language;
• Time Zone: displaying time zone;

System Up Time

System up time since the last reboot.

System Time

Current system time on the phone system.

Service Status

GUI and Phone service status.

System Information

Download system information

User Space

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

Core Dump

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

Status 🡪 Programmable Keys Status 🡪Virtual Multi-Purpose Keys

VPKs Status

• Mode
• Account
• Description
• Value

Status 🡪 Programmable Keys Status 🡪 Multi-Purpose Keys

MPKs Status

• Mode
• Account
• Description
• Value

Status 🡪 Programmable Keys Status 🡪 Softkeys

Softkeys

• Mode
• Account
• Description
• Value

Status 🡪 Extension board Status

Extension 1/2/3/4 Keys

• Mode
• Account
• Description
• Value

Account x 🡪 General Settings

Account Active

This field indicates whether the account is active.

The default setting is “Yes”.

Account Name

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

SIP Server

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

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

Secondary SIP Server

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

Outbound Proxy

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

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

Backup Outbound Proxy

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

BLF Server

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

SIP User ID

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

Field Improvement:

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

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

Authenticate ID

SIP service subscriber’s Authenticate ID used for authentication.

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

Authenticate Password

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

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

Name

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

Voice Mail User ID

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

Picture

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

Account Display

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

If set to “User Name”, LCD account label will display the Account Name configured for this SIP account. If set to “User ID”, it will then display the SIP User ID configured for this SIP account.

Account x 🡪 Dial Plan

Name

Enter the name for the configured rules.

Rule

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

Type

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

Eventlist BLF URI

Configures the Eventlist BLF URI on the phone to monitor the extensions in the list with Multi-Purpose Key. If the server supports this feature, users need to configure an Eventlist BLF URI on the service side first (i.e., BLF1006@myserver.com) with a list of extensions included. On the phone, in this “Eventlist BLF URI” field, fill in the URI without the domain (i.e., BLF1006). To monitor the extensions in the list, under Web GUI🡪Settings🡪Programmable Keys page, please select “Eventlist BLF” in the key mode, choose account, enter the value of each extension in the list.

Auto Provision Eventlist BLFs

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

Conference URI

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

Music On Hold URI

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

BLF Call-pickup

Configures BLF Call-pickup method:

• If select Auto:

The phone will do either Prefix or barge in code for BLF pickup depend on which on is set.

• If select Force BLF Call-pickup by prefix:

The phone will only use Prefix as BLF pickup method.

• If select Disabled:

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

BLF Call-pickup Prefix

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

Call Pickup Barge-In Code

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

PUBLISH for Presence

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

Omit charset=UTF-8 in MESSAGE

Omit charset=UTF-8 in MESSAGE content-type

Allow Unsolicited REFER

Allow Unsolicited REFER to accomplish an outgoing call.

Special Feature

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

Broadsoft Call Center

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

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

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

Hoteling Event

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

Call Center Status

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

Broadsoft Executive Assistant

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

Feature Key Synchronization

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

Broadsoft Call Park

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

Account x 🡪 SIP Settings 🡪 Session Timer

Enable Session Timer

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

Session Expiration

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

Min-SE

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

Caller Request Timer

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

Callee Request Timer

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

Force Timer

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

UAC Specify Refresher

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

UAS Specify Refresher

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

Force INVITE

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

The default setting is “No”.

Account x 🡪 SIP Settings 🡪 Security Settings

Check Domain Certificates

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

Validate Certificate Chain

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

Validate Incoming Messages

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

Check SIP User ID for Incoming INVITE

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

Accept Incoming SIP from Proxy Only

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

Authenticate Incoming INVITE

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

Account x 🡪 Audio Settings

Preferred Vocoder

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

Use First Matching Vocoder in 200OK SDP

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

Codec Negotiation Priority

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

Hide Vocoder

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

Disable Multiple m line in SDP

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

SRTP Mode

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

SRTP Key Length

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

Default setting is: AES 128&256 bit

Crypto Life Time

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

Symmetric RTP

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

Silence Suppression

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

Jitter Buffer Type

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

Jitter Buffer Length

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

Voice Frames Per TX

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

The default setting is 2.

G723 Rate

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

The default setting is “5.3kbps encoding rate”.

G.726-32 Packing Mode

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

iLBC Frame Size

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

iLBC Payload Type

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

OPUS Payload Type

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

DTMF Payload Type

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

Send DTMF

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

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

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

• SIP INFO uses SIP INFO to carry DTMF.

Default setting is “RFC2833”.

DTMF Delay

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

Account x 🡪 Call Settings

Early Dial

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

The default setting is “No”.

Dial Plan Prefix

Configures the prefix to be added to each dialed number.

Dial Plan

A dial plan establishes the expected number and pattern of digits for a telephone number. This parameter configures the allowed dial plan for the phone. Default setting is “{ x+ | \+x+ | *x+ | *xx*x+ }”. Dial Plan Rules:

1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d;
2. Grammar:

x – any digit from 0-9

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

1. xx+ – at least 2-digit numbers
2. xx — only 2-digit numbers
3. ^ — exclude
4. [3–5] — any digit of 3, 4, or 5
5. [147] — any digit of 1, 4, or 7
6. <2=011> – replace digit 2 with 011 when dialing
7. | — the OR operand
8. {X123} — match Z123, e123, 5123, …
9. Flag T when adding a “T” at the end of the dial plan, the phone will wait for 3 seconds before dialing out. This gives users more flexibility on their dial plan setup. E.g. with dial plan 1XXT, phone will wait for 3 seconds to let user dial more than just 3 digits if needed. Originally the phone will dial out immediately after dialing the third digit.
10. Back slash “\” — can be used to escape specific letters. E.g. if { \p\a\r\k\+60 } dial plan is configured, park+60 should be able to pass dial plan check. This also can be used to escape Mark and User-unreserved characters.

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

• Example 1: {[369]11 | 1617xxxxxxx}

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

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

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

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

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

• Example 4: If we set the dial plan with {\*123}, it should allow input *123 to pass dial plan check.
• Example 5: If we set the dial plan with {\$123}, it should allow input $123 to pass dial plan check.
• Example 6: If we set the dial plan with {12\_3}, it should allow input 12_3 to pass dial plan check.

Example of a simple dial plan used in a Home/Office in the US:

{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }

Explanation of example rule (reading from left to right):

• ^1900x. — prevents dialing any number started with 1900;
• <=1617>[2-9]xxxxxx — allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically;
• 1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits length.
• 011[2-9]x — allows international calls starting with 011;
• [3469]11 — allows dialing special and emergency numbers 311, 411, 611 and 911.

Note: In some cases, where the user wishes to dial strings such as *123 to activate voice mail or other applications provided by their service provider, the * should be predefined inside the dial plan feature.

An example dial plan will be: {*x+ } which allows the user to dial * followed by any length of numbers.

Call Log

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

Account Ring Tone

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

Match Incoming Caller ID

Specifies matching rules with number, pattern or Alert Info text or string (up to 10 matching rules). When the incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive ringtone. Matching rules:

• Specific caller ID number. For example, 8321123;
• A defined pattern with certain length using x and + to specify, where x could be any digit from 0 to 9. Samples:

xx+ : at least 2-digit number;

xx : only 2-digit number;

[345]xx: 3-digit number with the leading digit of 3, 4 or 5;

[6-9]xx: 3-digit number with the leading digit from 6 to 9.

• Alert Info text

Users could configure the matching rule as certain text (e.g., priority) and select the custom ring tone mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info: <http://127.0.0.1>; info=priority

• Alert Info string

Users could configure the matching rule as certain string (e.g., string) and select the custom ring tone mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info: <string>

Selects the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info test or string matches one of the 10 rules, the phone will ring with the associated ringtone.

Ring Timeout

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

Send Anonymous

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

Anonymous Call Rejection

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

Auto Answer

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

Auto Answer Numbers

This function allows users to have the phone configured with a pre-defined list of numbers that will perform auto answer.

There are different situations for auto answer setup:

1) Auto Answer disable → no auto answer;

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

3) Auto Answer enable, auto answer number specified → only numbers specified will be auto answered.

For “Auto Answer Numbers”, it accepts:

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

Refer-To Use Target Contact

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

Transfer on Conference Hang-up

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

Disable Recovery on Blind Transfer

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

Notes:

1) This feature only applies to blind transfer;

2) This feature depends on how server handles transfer. If there is any NOTIFY from server, this feature won’t take effect. If server responds 4xx, phone should try to recover regardless of this option.

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

Blind Transfer Wait Timeout

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

No Key Entry Timeout

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

Use # as Dial Key

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

On Hold Reminder Tone

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

RFC2543 Hold

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

Hide Dialing Password

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

Disable Call Waiting

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

Account x 🡪 Intercom Settings

Allow Auto Answer by Call-Info/Alert-Info

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

Allow Barging by Call-Info/Alert-Info

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

Mute on answer Intercom call

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

Play warning tone for Auto Answer Intercom

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

Custom Alert-Info for Auto Answer

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

Account x 🡪 Feature Codes

Enable Local Call Features

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

Note: If the device is registered with Broadsoft account, it doesn’t matter if local call features are enabled or disabled, once the Broadsoft account is set, special feature to Broadsoft and Feature Key Synchronization is enabled, the call feature will be handled by Broadsoft server, not by the phone.

Do Not Disturb (DND)—On

Configures DND feature code to turn on DND.

Do Not Disturb (DND)—Off

Configures DND feature code to turn off DND.

Call Forward Unconditionally (All)—On

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

Call Forward Unconditionally (All)—Off

Configures Call Forward All feature code to deactivate unconditional call forwarding

Target

Configures the extension that the call will be forwarded to.

Call Forward Busy—On

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

Call Forward Busy—Off

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

Target

Configures the extension that the call will be forwarded to.

Call Forward Delayed (No Answer)—On

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

Call Forward Delayed (No Answer)—Off

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

Target

Configures the extension that the call will be forwarded to.

Delayed Call Forward Wait Time

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

Accounts 🡪 Account Swap

Swap Account Settings

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

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

Phonebook 🡪 Contacts

Search Bar

Allows users searching for phonebook entries.

Add Contact

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

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

Edit Contact

Edits selected contact.

Delete All Contacts

Deletes all contacts from phonebook.

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

Phonebook 🡪 Group Management

Add Group

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

Edit Group

Edits selected group.

Phonebook 🡪 Phonebook Management

Enable Phonebook XML Download

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

The default setting is “Disabled”.

HTTP/HTTPS User Name

The user name for the HTTP/HTTPS server.

HTTP/HTTPS Password

The password for the HTTP/HTTPS server.

Phonebook XML Server Path

Configures the server path to download the phonebook XML.

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

Phonebook Download Interval

Configures the phonebook download interval (in minutes).

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

Remove Manually-edited Entries on Download

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

Import Group Method

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

Sort Phonebook by

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

The default setting is “Last Name”.

Download XML
Phonebook

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

Upload XML Phonebook

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

Phonebook Key Function

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

Default search mode

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

Phonebook 🡪 LDAP

LDAP Protocol

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

Server Address

Configures the IP address or DNS name of the LDAP server.

Port

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

Base

Configures the LDAP search base.

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

Example:

dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com

User Name

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

Password

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

LDAP Number Filter

Configures the filter used for number lookups. Examples:

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

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

LDAP Name Filter

Configures the filter used for name lookups. Examples:

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

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

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

LDAP Version

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

LDAP Name Attributes

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

gn

cn sn description

LDAP Number Attributes

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

This field allows the users to configure multiple space separated number attributes. Example:

telephoneNumber

telephoneNumber Mobile

LDAP Display Name

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