GRP261x/GRP262x/GRP263x/GRP2670/GRP2650 Series - Administration Guide

  • Updated on May 17, 2024

Thank you for purchasing Grandstream GRP26XX Carrier-Grade IP Phones.

The GRP26xx Professional Carrier-Grade IP Phones offer top-notch HD audio quality as well as a variety of telephony features. This series was designed for mass deployment and broad interoperability with most 3rd party SIP devices and platforms. With its sleek design, re-conceptualized user experience, the GRP26xx series comes with powerful out-of-the-box feature options including Bluetooth & Wi-Fi support, a dual LCD screen, support for up to 14 lines, 5-way conferencing for all models, and customizable face-plates for a touch of personalization. In addition to all the mentioned characteristics, the series is compatible with Grandstream’s cloud provisioning platform, GDMS (Grandstream Device Management System) for remote and cloud provisioning purposes. The GRP26xx is the perfect choice for enterprises looking for IP phones with advanced functionalities that are also easy to use and deploy.

PRODUCT OVERVIEW

Feature Highlights

The following table contains the major features of the GRP26XX phones:

GRP2612

GRP2612P

GRP2612W

GRP2612G

  • 4 dual-color line keys (can be digitally programmed as up to 16 provisionable BLF/fast-dial keys)

  • 2.4” (320×240) TFT color LCD.

  • 4 programmable context-sensitive soft keys.

  • 100M net­work ports. (1000M for GRP2612G)

  • Integrated PoE (GRP2612P, GRP2612G & GRP2612W only).

  • 5-way conference.

  • Electronic Hook Switch (EHS).

  • Wi-Fi support (GRP2612W only).

GRP2613

  • 6 dual-color line keys (can be digitally programmed as up to 24 pro­visionable BLF/fast-dial keys).

  • 2.8” (320×240) TFT color LCD.

  • 4 programmable context-sensitive soft keys.

  • 1000M network ports.

  • 5-way conference.

  • integrated PoE.

  • Electronic Hook Switch (EHS).

GRP2614

  • 4 dual-color line keys (can be digitally programmed as up to 16 pro­visionable BLF/fast-dial keys).

  • 2.8” (320×240) TFT color LCD.

  • 4 programmable context-sensitive soft keys

  • 2.4” (320×240) addi­tional screen dedicated to up to 24 multi-purpose keys.

  • 1000M network ports.

  • Integrated PoE.

  • Wi-Fi and Bluetooth support.

  • 5-way conference.

  • Electronic Hook Switch (EHS).

GRP2615

  • 10 dual-color line keys (can be digitally programmed as up to 40 provisionable BLF/fast-dial keys).

  • 4.3” (480×272) TFT color LCD.

  • 5 programmable context-sensitive soft keys.

  • 1000M network ports.

  • Integrated PoE.

  • Wi-Fi and Bluetooth support.

  • 5-way conference.

  • Electronic Hook Switch (EHS).

GRP2616

  • 6 dual-color line keys (can be digitally programmed as up to 24 provisionable BLF/fast-dial keys).

  • 4.3” (480×272) TFT color LCD.

  • 5 programmable context-sensitive soft keys.

  • 2.4” (320×240) addi­tional screen dedicated to up to 24 multi-purpose keys.

  • 1000M network ports.

  • Integrated PoE.

  • Wi-Fi and Bluetooth support.

  • 5-way conference.

  • Electronic Hook Switch (EHS).

GRP2624

  • 8 dual-color line keys (can be digitally programmed as up to 24 provisionable BLF/fast-dial keys).

  • 2.8 inch (320×240) TFT color LCD.

  • 4 programmable context-sensitive soft keys.

  • 2.4” (320×240) addi­tional screen dedicated to up to 24 multi-purpose keys.

  • 1000M network ports.

  • Integrated PoE.

  • Wi-Fi and Bluetooth support.

  • 5-way conference.

  • Electronic Hook Switch (EHS).

GRP2634

  • 8 dual-color line keys (can be digitally programmed as up to 24 provisional BLF/fast-dial keys).

  • 2.8 inch (320×240) TFT color LCD.

  • 4 programmable context-sensitive soft keys.

  • 2.4” (320×240) addi­tional screen dedicated to up to 24 multi-purpose keys.

  • 1000M network ports.

  • Integrated PoE.

  • Wi-Fi and Bluetooth support.

  • 5-way conference.

GRP2636

  • 12 lines with up to 6 SIP Accounts

  • 24 dual-color line keys (can be digitally programmed as up to 24 provisional BLF/fast-dial keys).

  • 4.3inch(480x272) TFT color LCD.

  • 5 programmable context-sensitive soft keys.

  • 1000M network ports.

  • Wi-Fi and Bluetooth support.

  • 5-way conference.

GRP2670

  • 12 lines with up to 6 SIP Accounts

  • 7” (1042x600) capacitive touch TFT color LCD.

  • 1000M network ports.

  • Integrated PoE.

  • Wi-Fi and Bluetooth support.

  • 5-way conference.

GRP2650

  • 14 line keys with up to 6 SIP accounts (With 56 Virtual Multi-Purpose Keys (VPK))

  • 5.0 inch (1280x720) TFT color LCD

  •  6 XML programmable context sensitive softkeys

  • Dual switched auto-sensing 10/100/1000 Mbps 

  • Integrated PoE

  • Wi-Fi and Bluetooth support

  • 5-way conference.

GRP261x/GRP2624/GRP2634/GRP2636/GRP2650/GRP2670 Features in a Glance

Technical Specifications

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for the GRP261x/GRP2624/GRP2634/GRP2670 series.

Protocols/Standards

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

Network Interfaces

Dual switched auto-sensing 10/100 Mbps Ethernet ports (GRP2612)
Dual switched auto-sensing 10/100 Mbps Ethernet ports with integrated PoE (GRP2612P
& GRP2612W)
Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with integrated PoE
(GRP2612G)

Graphic Display

2.4 inch (320×240) TFT color LCD

Feature Keys

4 line keys with up to 4 SIP accounts and up to 2 SIP accounts for legacy hardware (discontinued in 2020), 4 programmable contexts sensitive Softkeys, 5 navigation/menu keys, 9 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, 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), 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)

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 1000 items), call waiting, call log (up to 2000 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, (*wall mount sold separately)

QoS

Layer 2 (802.1Q, 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: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

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: +5VDC, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Physical

Dimension : 203mm x 193mm x 52.1mm

Unit weight : 554g

Package weight : 936g

Temperature and Humidity

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

Package Content

GRP2612/GRP2612P/GRP2612W/GRP2612G phone, handset with cord, base stand, universal power supply (except GRP2612P), network cable, Quick Installation Guide

Compliance

GRP2612/GRP2612P/GRP2612G:

FCC: Part 15 Class B; FCC Part 68 HAC.

CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN IEC 62368-1.

RCM: AS/NZS CISPR 32; AS/NZS 62368.1; AS/CA S004

IC: ICES-003; CS-03, Part V.

GRP2612W:

FCC: Part 15 Class B; Part 15 Subpart C, 15.247; Part 15 Subpart E, 15.407; FCC Part 68 HAC.

CE: EN 55032; EN 55035; EN IEC 61000-3-2; EN 61000-3-3; EN IEC 62368-1; EN 301 489-1; EN 301 489-17; EN 300 328; EN 301 893; EN 62311.

RCM: AS/NZS CISPR 32; AS/NZS 62368.1; AS/NZS 4268; AS/NZS 2772.2; AS/CA S004.

IC: ICES-003; CS-03, Part V; RSS-247; RSS-102.

GRP2612/GRP2612P/GRP2612W Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

2.8 inch (320x240) TFT color LCD – 2.4 inch MPK color LCD

Feature Keys

6 line keys with up to 4 SIP accounts and up to 3 SIP accounts for legacy hardware (discontinued in 2020), 5 navigation/menu keys, 9 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, 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), 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 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 2000 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, (*wall mount sold separately)

QoS

Layer 2 (802.1Q, 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: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

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, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Max power consumption: 6W

Physical

Dimension : 203mm x 193mm x 52.1mm

Unit weight : 554g

Package weight : 936g

Temperature and Humidity

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

Package Content

GRP2613 phone, handset with cord, base stand, universal power supply, network cable, Quick Installation Guide

Compliance

FCC: FCC Part 15B, Class B; FCC Part 15 Subpart C; FCC Part 15 Subpart E; FCC Part 68 HAC.

CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; EN 301 489-1/-17; EN 300 328; EN 301 893; EN 62311;

RCM: AS/NZS CISPR 32;AS/NZS 60950.1;AS/NZS 4268; AS/CA S004.

GRP2613 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

2.8 inch (320×240) TFT color LCD – 2.4 inch MPK color LCD

Bluetooth

Yes, Bluetooth integrated

Wi-Fi

Yes, dual-band

Feature Keys

4 line keys with up to 4 SIP accounts, 24 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, 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).

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 2000 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, (*wall mount sold separately)

QoS

Layer 2 (802.1Q, 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: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

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, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Max power consumption: 6W

Physical

Dimension : 234mm x 213mm x 82.2mm

Unit weight : 950g

Package weight : 1460g


Temperature and Humidity

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

Package Content

GRP2614 phone, handset with cord, base stand, universal power supply, network cable, Quick Installation Guide

Compliance

FCC: FCC Part 15B, Class B; FCC Part 15 Subpart C; FCC Part 15 Subpart E; FCC Part 68 HAC.

CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; EN 301 489-1/-17; EN 300 328; EN 301 893; EN 62311;

RCM: AS/NZS CISPR 32;AS/NZS 60950.1;AS/NZS 4268; AS/CA S004.

GRP2614 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD – 2.4 inch MPK color LCD

Bluetooth

Yes, Bluetooth integrated

Wi-Fi

Yes, dual-band

Feature Keys

10 line keys with up to 5 SIP accounts, 40 speed-dial/BLF extension keys with dual-color LED, 5 programmable contexts sensitive Softkeys, 5 navigation/menu keys, 9 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, 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), 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 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 2000 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, (*wall mount sold separately)

QoS

Layer 2 (802.1Q, 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: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

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, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Physical

Dimensions : 243mm x 210mm x 82.3mm

Unit weight:970g

Package weight:1480g

Temperature and Humidity

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

Package Content

GRP2615 phone, handset with cord, base stand, universal power supply, network cable, Quick Installation Guide

Compliance

FCC: FCC Part 15B, Class B; FCC Part 15 Subpart C; FCC Part 15 Subpart E; FCC Part 68 HAC.

CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; EN 301 489-1/-17; EN 300 328; EN 301 893; EN 62311;

RCM: AS/NZS CISPR 32;AS/NZS 60950.1;AS/NZS 4268; AS/CA S004.

GRP2615 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

4.3 inch (480×272) TFT color LCD – 2.4 inch MPK color LCD

Bluetooth

Yes, Bluetooth integrated

Wi-Fi

Yes, dual-band

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), 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 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 2000 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 (802.1Q, 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: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

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, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Temperature and Humidity

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

Package Content

GRP2616 phone, handset with cord, base stand, universal power supply, network cable, Quick Installation Guide

Compliance

FCC: FCC Part 15B, Class B; FCC Part 15 Subpart C; FCC Part 15 Subpart E; FCC Part 68 HAC.

CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; EN 301 489-1/-17; EN 300 328; EN 301 893; EN 62311;

RCM: AS/NZS CISPR 32;AS/NZS 60950.1;AS/NZS 4268; AS/CA S004.

GRP2616 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

2.8 inch (320×240) TFT color LCD

Bluetooth

Yes, Bluetooth integrated

Wi-Fi

Yes, dual-band

Feature Keys

8 line keys with up to 6 SIP accounts and up to 4 SIP accounts for legacy hardware (discontinued in 2020), 4 XML programmable context sensitive softkeys, 5 navigation/menu keys, 9 dedicated function keys for: MESSAGE(with LED indicator), TRANSFER, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL

Voice Codec

Support for 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 to support Grandstream’s GUV Series headsets and other USB headsets

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-callappearance(SCA)/bridged-line-appearance(BLA), downloadable phonebook(XML, LDAP, up to 2000 items), call waiting, call log(up to 2000 records), XML 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, HD handset and speakerphone with support for wideband audio, and dual microphone.

Base Stand

Yes, 2 angle positions available, Wall Mountable (Wall Mount *sold separately)

Wall Mountable

Yes

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, secure boot.

Multi-language

Upgrade/Provisioning

Firmware upgrade via FTP/TFTP / HTTP / HTTPS, mass provisioning using GDMS/TR069 or AES encrypted XML configuration file.

Power & Green Energy Efficiency

Universal power adapter included: Input: 100-240V ; Output: +12V, 1A ; Integrated Power-over-Ethernet (802.3af) Max power consumption 9.5W (power adapter) or 10.8W (PoE)

Temperature and Humidity

Operation: 0°C to 40°C Storage: -10°C to 60°C Humidity: 10% to 90% Non-condensing

Package Content

GRP2624 phone, handset with cord, phone stand, 12V power adapter, network cable, Quick Installation Guide, GPL license

Compliance

FCC: FCC Part 15B, Class B; FCC Part 15 Subpart C; FCC Part 15 Subpart E; FCC Part 68 HAC.

CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; EN 301 489-1/-17; EN 300 328; EN 301 893; EN 62311;

RCM: AS/NZS CISPR 32;AS/NZS 60950.1;AS/NZS 4268; AS/CA S004.

GRP2624 Technical Specifications

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069,
802.1x, TLS, SRTP, IPV6

Network Interfaces

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

Graphic Display

2.8 inch (320x240) TFT color LCD

Bluetooth

Yes, integrated

Wi-Fi

Yes, integrated dual-band Wi-Fi 802.11 a/b/g/n/ac (2.4Ghz & 5Ghz)

Feature Keys

8 line keys with up to 6 SIP accounts and up to 4 SIP accounts for legacy hardware (discontinued in 2020), 10 MPK extension keys with paper slot, 4 XML programmable context-sensitive softkeys, 5 navigation/menu keys, 9 dedicated
function keys for MESSAGE(with LED indicator), TRANSFER, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL

Voice Codec

Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band), G723, iLBC, OPUS,in-band and out-of-band DTMF(in audio, RFC2833, SIP INFO)

Auxiliary Ports

RJ9 headset jack allowing EHS with Plantronics headsets, USB to support Grandstream’s GUV Series headsets, and other USB 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 2000 records), XML 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, HD handset and speakerphone with support for wideband audio. Dual Microphone.

Extension Module

No

Base Stand

Yes, 2 angle positions available, Wall Mountable (Wall Mount *sold separately)

QoS

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

Security

User and administrator-level passwords, MD5 and MD5-sess-based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control,
secure boot.

Multi-language

LCD Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

Upgrade/Provisioning

Firmware upgrade via FTP/TFTP / HTTP / HTTPS, mass provisioning using GDMS/TR069 or AES encrypted XML configuration file.

Power & Green Energy Efficiency

Universal power adapter included:
Input:100-240V ; Output: +12V, 1A ;
Integrated Power-over-Ethernet (802.3af)
Max power consumption 9.5W (power adapter) or 10.8W (PoE)

Temperature and Humidity

Operation: 0°C to 40°C

Storage: -10°C to 60°C

Humidity: 10% to 90% non-condensing

Package Content

GRP2634 phone, handset with cord, phone stand, 12V power adapter, network cable, Quick Installation Guide, GPL license

Physical

Dimension: 220mmx 210mmx 82mm
Unit Weight: 880g ; Package Weight:1260g

Compliance

FCC: Part 15 (CFR 47) Class B
CE: EN55022 Class B; EN55024 Class B;EN61000-3-2; EN61000-3-3;EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950.1

GRP2634 Technical Specifications

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV,NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x,
TLS, SRTP, IPV6

Network Interfaces

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

Graphic Display

4.3inch(480x272) TFT color LCD

Bluetooth

Yes, integrated

Wi-Fi

Yes, integrated dual-band Wi-Fi 802.11 a/b/g/n/ac (2.4Ghz & 5Ghz)

Feature Keys

12 line keys with up to 6 SIP accounts, 24MPK extension keys with paper slot, 5 XML programmable context-sensitive softkeys, 5 navigation/menu keys, 8 dedicated
function keys for: MESSAGE(with LED indicator), TRANSFER, HEADSET, HOLD,MUTE,
SEND/REDIAL, SPEAKERPHONE, VOL+, VOL

Voice Codec

Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band), G723, iLBC, OPUS, inband and out-of-band DTMF(in audio, RFC2833, SIP INFO)

Auxiliary Ports

RJ9 headset jack allowing EHS with Plantronics headsets, USB to support Grandstream’s GUV Series headsets and other USB headsets

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call appearances(SCA)/bridged-line-appearance(BLA), downloadable phonebook(XML,
LDAP, up to 2000 items), call waiting, call log(up to 2000 records), XML 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, HD handset and speakerphone with support for wideband audio. Dual Microphone.

Extension Module

Yes

Base Stand

Yes, 2 angle positions available, Wall Mountable (Wall Mount *sold separately)

QoS

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

Security

User and administrator-level passwords, MD5 and MD5-sess-based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control,
secure boot.

Multi-language

LCD Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included:
Input:100-240V ; Output: +12V, 1A ;
Integrated Power-over-Ethernet(802.3af)
Max power consumption 3.2W(power adapter) or 4.3W(PoE)

Temperature and Humidity

Operation: 0°C to 40°C

Storage: -10°C to 60°C

Humidity: 10% to 90% non-condensing

Package Content

GRP2636 phone, handset with cord, phone stand, 12V power adapter, network cable,
Quick Installation Guide, GPL license

Physical

Dimension: 220mmx 210mmx 82mm
Unit Weight: 880g ; Package Weight:1260g

Compliance

FCC: FCC Part 15 Class B; FCC Part 15 Subpart C,15.247; FCC Part 15 Subpart E,15.407;
FCC Part 68 HAC.
CE: ETSI EN 301 893; ETSI EN 301 489-1/-17; ETSI EN 300 328; EN IEC 62311; EN
55032; EN 55035; EN 62368-1.
IC: RSS-247 Issue 2; RSS-Gen Issue 5; ICES-003 Issue 7; CS-03, Part V.
RCM: AS/NZS CISPR32; AS/NZS 4268; AS/NZS 2772.2; AS/NZS 62368.1; AS/CA S004.
UKCA: ETSI EN 301 893; ETSI EN 301 489-1/-17; ETSI EN 300 328; BS EN IEC 62311; BS
EN 55032; BS EN 55035; BS EN 62368-1.

GRP2636 Technical Specifications

Protocols/Standards

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

Network Interfaces

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

Graphic Display

7” (1042x600) capacitive touch TFT color LCD

Bluetooth

Yes, integrated

Wi-Fi

Yes, integrated dual-band Wi-Fi 802.11 a/b/g/n/ac (2.4Ghz & 5Ghz)

Feature Keys

5 navigation/menu keys, 9 dedicated function keys for MESSAGE (with LED indicator), TRANSFER, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-

Voice Codec

Support for 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 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 2000 records), XML 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, HD handset and speakerphone with support for wideband audio, and dual microphone.

Extension Module

Yes

Base Stand

Yes, 2 angle positions available, Wall Mountable (Wall Mount *sold separately)

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, secure boot.

Multi-language

LCD Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included:

Input: 100-240V.

Output: +12V, 1A.

Integrated Power-over-Ethernet (802.3af)

Max power consumption 6.5W (power adapter)

Temperature and Humidity

Operation: 0°C to 40°C

Storage: -10°C to 60°C

Humidity: 10% to 90% non-condensing

Package Content

GRP2670 phone, handset with cord, phone stand, 12V power adapter, network cable, Quick Installation Guide

Compliance

FCC: Part 15 Subpart B(Class B), Part 15 Subpart C 15.247, Part 15 Subpart C 15.407, Part 1 Subpart I, Part 68. 316/317.

IC: RSS-247, RSS-Gen, RSS-102, ICES-003, CS-03 Part V;
CE: EN 55032, EN 55035, EN 61000-3-2, EN 61000-3-3, EN 62368-1, EN 62311, EN 301 489-1, EN 301 489-17, EN 300 328, EN 301 893;
RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/NZS 4268, AS NZS 2772.2, AS/CA S004.

GRP2670 Technical Specifications

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x,
TLS, SRTP, IPV6

Network Interfaces

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

Graphic Display

5.0 inch (1280x720) TFT color LCD

Wi-Fi

Yes, integrated dual-band WiFi 802.11 a/b/g/n/ac (2.4Ghz & 5Ghz)

Bluetooth

Yes, integrated

Feature Keys

14 line keys with up to 6 SIP accounts, 6 XML programmable context-sensitive softkeys, 5 navigation/menu keys, 9 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE,
VOL+, VOL-

Voice Codec

Support for G.729A/B, G.711µ/a-law, G.726-32, G.722(wide-band), G723.1, iLBC, OPUS, 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-callappearance (SCA)/bridged-line-appearance (BLA), downloadable phonebook(XML,
LDAP, up to 2000 items), call waiting, call log(up to 2000 records), XML 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, HD handset and speakerphone with support for wideband audio, and dual microphone

Extension Module

Yes, GBX20

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control,
secure boot

Multi-language

LCD Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Català (Catalan) Čeština (Czech) Deutsch (German) Ελληνικά (Greek) 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) Svenska (Swedish) Slovenščina (Slovenian) Slovenčina (Slovak) Türkçe (Turkish) Українська (Ukrainian) 正體中文 (Traditional Chinese)
WebUI Language: English 简体中文 (Simplified Chinese) العربية (Arabic) Čeština (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) 繁體中文 (Traditional Chinese)

Upgrade/Provisioning

Firmware upgrade via FTP/TFTP / HTTP / HTTPS, mass provisioning using GDMS/TR069 or AES 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 6.2W(power adapter)

Physical

Unit weight:1050g ; Package weight:1620g
Dimension: 263mm x 210mm x 82mm

Temperature and Humidity

Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity: 10% to 90% Non-condensing

Package Content

GRP2650 phone, handset with cord, phone stand, 12V power adapter, network cable, Quick Installation Guide

Compliance

FCC: Part 15 (CFR 47) Class B
CE: EN55022 Class B; EN55024 Class B;EN61000-3-2; EN61000-3-3;EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950.1

GRP2650 Technical Specifications

GETTING STARTED

This chapter provides basic installation instructions including the list of the packaging contents and also information for obtaining the best performance with the GRP261x/GRP2624/GRP2634 phone.

Equipment Packaging

GRP261x/GRP2624/GRP263x/GRP2670/GRP2650

  • 1 x GRP261x/GRP2624/GRP263x/GRP2670/GRP2650 Main Case.

  • 1 x Handset.

  • 1 x Phone Stand.

  • 1 x Ethernet Cable.

  • 1 x Power Adapter.

  • 1 x Phone cord.

  • 1 x Quick Installation Guide.

Equipment Packaging

GRP261X/GRP2624/GRP2634 Package Content (GRP2670 as an example)
Note

Check the package before installation. If you find anything missing, contact your system administrator.

GRP261X/GRP2624/GRP263x/GRP2670/GRP2650 Phone Setup

The GRP261X/GRP2624/GRP263x/GRP2670/GRP2650 phones can be installed on the desktop using the phone stand or attached to the wall using the slots for wall mounting.

Phone Stand and Mounting Slots on the GRP261X/GRP2624/GRP2634/GRP2670

Using the Phone Stand

For installing the phone on the table with the phone stand, attach the phone stand to the bottom of the phone where there is a slot for the phone stand. (Upper half, bottom part).

Using the Slots for Wall Mounting

1. Attach the wall mount spacers to the slot for wall mount spacers on the back of the phone.

2. Attach the phone to the wall via the wall mount hole.

3. Pull out the tab from the handset cradle (See figure below).

4. Rotate the tab and plug it back into the slot with the extension up to hold the handset while the phone is mounted on the wall (see figure below).

Tab on the Handset Cradle

Connecting the GRP261X/GRP2624/GRP263x/GRP2670/GRP2650

To set up the GRP261X/GRP2624/GRP263x/GRP2670/GRP2650, follow the steps below:

1. Connect the handset and main phone case with the phone cord.

2. Connect the LAN port of the phone to the RJ-45 socket of a hub/switch or a router (LAN side of the router) using the Ethernet cable.

3. Connect the PSU output plug to the power jack on the phone; plug the power adapter into an electrical outlet. If a PoE switch is used in step 2, this step could be skipped.

4. The LCD will display provisioning or firmware upgrade information. Before con­tinuing, please wait for the date/time display to show up.

5. Using the phone-embedded web server or keypad configuration menu, you can further configure the phone using either a static IP or DHCP.

GRP261X/GRP2624/GRP263x/GRP2670/GRP2650 Back / Side View
Note

  • For easy deployment, GRP2612W/GRP2614/GRP2615/GRP2616/GRP2624/GRP263x/GRP2670/GRP2650 out of the box is preconfigured to connect to a default SSID named wp_master with a password (WPA/WPA2 PSK) equal to wp!987@dmin, for this to work the following criteria needs to be met:
    1. The IP phone should not be connected by a LAN cable.
    2. No SIP account should be registered on the IP phone.
    3. SSIDs from previous connections should not be saved on the IP phone, please factory reset the unit to confirm.

Configuration via Keypad

To configure the LCD menu using the phone’s keypad, follow the instructions below:

  • Enter MENU options. When the phone is idle, press the round MENU button to enter the configuration menu.
  • Navigate to the menu options. Press the arrow keys up/down/left/right to navigate to the menu options.
  • Enter/Confirm selection. Press the round MENU button or “Select” Softkey to enter the selected option.
  • Exit. Press “Exit” Softkey to exit the previous menu.
  • Return to Home page.

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

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

  • The phone automatically exits MENU mode with an incoming call, when the phone is off-hook or the MENU mode if left idle for more than 60 seconds.
  • When the phone is idle, pressing and holding the UP-navigation key for 3 seconds can see the phone’s IP address, IP setting, MAC address, and software address.

The MENU options are listed in the following table.

Call History

Displays Local call logs:

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

Status

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

  • Account status

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

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

  • Provider Status: 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.

– Keypad 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 minutesof idle before the screensaver activates. Valid range is 3 to 60.

– MPK LCD Settings (Available on GRP2614/GRP2616 only):

  • MPK LCD Display Order: Choose MPK LCD Display Order whether to be Sequential or Alternating

  • Display Contact on MPK LCD: Enable / Disable Display Contact on MPK Order

– 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 NTP server

  • 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: 

  • Internet Protocol :Selects Prefer IPv4 / Prefer IPv6 / IPv4 only or IPv6 only. The default setting is “Prefer IPv4”.

  • 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.

– Wi-Fi Settings (GRP2612W, GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2670 & GRP2650 only): 

  • Enables/disables Wi-Fi: Enables/disables Wi-Fi

  • Scan: Enables/disables Wi-Fi

– Bluetooth Settings (GRP2614/GRP2615, GRP2616, GRP2624, GRP2634, GRP2670 & GRP2650 only): 

  • Bluetooth Status: Displays the status of Bluetooth

  • Bluetooth MAC: Displays the GRP phone’s Bluetooth MAC address.

    (Bluetooth MAC address is GRP phone’s MAC address plus 1)

  • Power: Turns on/off the Bluetooth feature.

  • Handsfree Mode: Enables/Disables Handsfree mode

  • Bluetooth Name: Specifies GRP 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.

– Web Access: 

  • Web Access Mode

  • HTTP web port

  • HTTPs web port

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

  • 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.

– Operation: 

  • Factory Reset: It is used to restore the phone to factory default settings.

  • Ping and Traceroute: It is used to show the route taken by packets across to an URL.

  • Alternative firmware: It is used to show current and alternative firmware versions available on the phone. Users can “rollback” to alternative firmware from this menu.

Reboot

Reboots the phone.

Configuration Menu

The following picture shows the keypad MENU configuration flow:

Configuration via Web Browser

The GRP261X/GRP2624/GRP263x/GRP2670/GRP2650 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Google Chrome, Mozilla Firefox, and Microsoft’s IE. To access the Web GUI:

  1. Connect the computer to the same network as the phone.
  2. Make sure the phone is turned on and shows its IP address. You may check the IP address by pressing and holding the UP arrow button for 3 seconds when the phone is in an idle state.
  3. Open a Web browser on your computer.
  4. Enter the phone’s IP address in the address bar of the browser.
  5. Enter the administrator’s login and password to access the Web Configuration Menu.
Notes

  • The computer must be connected to the same sub-network as the phone. This can be easily done by connecting the computer to the same hub or switch as the phone is connected. In the absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the PC port on the back of the phone.
  • If the phone is properly connected to a working Internet connection, the IP address of the phone will display in MENU🡪Status🡪Network Status. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. Users will need this number to access the Web GUI. For example, if the phone has an IP address of 192.168.40.154, please enter “http://192.168.40.154” in the address bar of the browser.

There are two default passwords for the login page:

User Level

User

Password

Web Pages Allowed

End User Level

user

123

Only Status and Basic Settings

Administrator Level

admin

Random password available on the sticker at the back of the unit.

Browse all pages

When changing any settings, always SUBMIT them by pressing the “Save” or “Save and Apply” button at the bottom of the page. If the change is saved only but not applied, after making all the changes, click on the “APPLY” button on the top of the page to submit. After submitting the changes in all the Web GUI pages, reboot the phone to have the changes take effect if necessary (All the options under the “Accounts” page and “Phonebook” page do not require a reboot. Most of the options under “Settings” page do not require a reboot).

Saving Configuration Changes

After users make changes to the configuration, pressing the “Save” button will save but not apply the changes until the “Apply” button on the top of the web GUI page is clicked. Or, users could directly press the “Save and Apply” button. We recommend rebooting or powering cycle the phone after applying all the changes.

Rebooting from Remote Locations

Press the “Reboot” button on the top right corner of the web GUI page to reboot the phone remotely. The web browser will then display a reboot message. Wait for about 1 minute to log in again.

CONFIGURATION GUIDE

This section describes the options in the phone’s Web GUI. As mentioned, you can log in as an administrator or an end-user.

  • Status: Displays the Account status, Network status, and System Info of the phone.
  • Account: To configure the SIP account.
  • Settings: To configure call features, ring tone, audio control, LCD display, date and time, Web services, XML applications, programmable keys, etc.
  • Network: To configure network settings.
  • Maintenance: To configure web access, upgrading, provisioning, Syslog, language settings, TR-069, security, etc.
  • Directory: To manage Phonebook and LDAP.

Status Page Definitions

Status 🡪 Account Status

Account

Account index.

  • For GRP2612/GRP2612P/GRP2612W/GRP2612G: up 4 SIP accounts and up to 2 accounts for legacy hardware.

  • For GRP2613: up to 4 SIP accounts and and up to 3 accounts for legacy hardware.

  • For GRP2614: up to 4 SIP accounts.

  • For GRP2615: up to 5 SIP accounts.

  • For GRP2616: up to 6 SIP accounts.

  • For GRP2624: up to 6 SIP accounts and up to 4 SIP accounts for legacy hardware.

  • For GRP2634: up to 6 SIP accounts and up to 4 SIP accounts for legacy hardware.

  • For GRP2636: up to 6 SIP accounts.

  • For GRP2650: up to 6 SIP accounts.

  • For GRP2670: up to 6 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.

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

The status of Affinity Broadcast on the phone. (Available on GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2670 & GRP2650 only).

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.

Serial Number

Displays the Serial Number of the unit.

Certificate Type

Displays the ceritificate type used for encryption

Software Version

  • Boot: boot version number.

  • Core: core 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.

  • GUI: The GUI version used by the phone

  • Res: recovery version number.

IP Geographic Information

  • Country Code: displaying country code for Wi-Fi; (Available on GRP2612W, GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2636, GRP2670 & GRP2650 only).

  • City: displaying phone location.

  • Language: displaying language.

  • Recommended Time Zone: displaying time zone

System Up Time

System up time since the last reboot.

System Time

Current system time on the phone system.

System Time Zone

Displays the current system time zone

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.

Screenshot

Download captured screenshots.

Press “Start” button to clear screenshots.

Special Feature

OpenVPN® Support: displaying if the phone supports OpenVPN®.

Status 🡪 Call Status

Display the calls status. Refer to [Maintenance > Voice Monitoring > Display Report]

Status 🡪 Programmable Keys Status 🡪 Multi-Purpose Keys Status

MPKs Status

  • Mode

  • Account

  • Description

  • Value

Status 🡪 Programmable Keys Status 🡪Virtual Multi-Purpose Keys Status

VPKs Status

  • Mode

  • Account

  • Description

  • Value

Status 🡪 Programmable Keys Status 🡪 Softkeys Status 

Softkeys

  • Mode

  • Account

  • Description

  • Value

Status 🡪 Extension Boards Status (GRP2615 & GRP2650 & GRP2670 only)

Extension (1-4) Keys

EXT (1-160):

  • Mode

  • Account

  • Description

    Value

Status 🡪 Call Feature Status

Accounts

  • DND

  • Auto Answer

  • Forward All

  • Busy Forward

  • Delay Forward

Status 🡪 Energy Saving

Current Hours

Displays wether it is Office Hours or Non-Office Hours configuration.

Energy Saving Mode

Displays the Energy Saving mode.

Idle Time Tracking

Displays Energy Saving Feedback for achieved energy savings. By showing the duration that the phone has been powered up.

Status 🡪 Energy Saving 🡪 Applied Energy Saving Config

Backlight Brightness: Active

Displays the LCD brightness when the phone is active. Valid range is 10 to 100 where 100 is the brightest.

Backlight Brightness: Idle

Displays the LCD brightness when the phone is idle. Valid range is 0 to 100 where 0 is off and 100 is the brightest.

Active Backlight Timeout

Displays active backlight timeout (in minutes). The valid range is 0 to 90.

Blank Screen Timeout

Displays The actual applied power saving timeout value under Standard, Maximum, or Customized Mode

Enable Missed Call Backlight

If set to "Yes", LCD backlight will be turned on when there is a missed call on the phone.

Enable IEEE 802.3az EEE (Energy Efficient Ethernet)

Displays whether to enable IEEE 802.3az Energy Efficient Ethernet.

Enable Live Keypad

If enabled, phone will automatically dial out and turn on hands-free mode when keypad or softkey is pressed.

Status Page Definitions

Account Page Definitions

Account x 🡪 General Settings

Account Register

Account Active

Indicates whether the account is active.

The default setting is “No”.

Account Name

The name associated with each account to be displayed on the LCD. (e.g., MyCompany)

SIP Server

The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (e.g., sip.mycompany.com, or IP address)

Secondary SIP Server

The URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails

Tertiary SIP Server

The URL or IP address, and port of the SIP server. This will be used when the primary and secondary SIP server fail.

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.

SIP User ID

User account information, provided by your VoIP service provider.

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 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 (e.g., John Doe).

Tel URI

If the phone has an assigned PSTN telephone number, this field should be set to "user=phone". A "user=phone" parameter will be attached to the Request-URI 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.

Voice Mail Access Number

Allows users to access voice messages by pressing the MESSAGE button on the phone. This value is usually the VM portal access number.

Monitored Voicemail Access Number

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

BLF Server

Configures the BLF server (optional) used for SUBSCRIBE requests.

Picture

Configures picture for the account. It will be sent to the caller/callee for the call.

Account Display

When set to “Username”, the LCD will display the Username if it is not empty and when set to “User ID”, the LCD will display the User ID if it is not empty.

Network Settings

DNS Mode

This parameter controls how the Search Appliance looks up IP addresses for hostnames.
There are four modes: A Record, SRV, NATPTR/SRV, Use Configured IP.
The default setting is “A Record”.
If the user wishes to locate the server by DNS SRV, the user may select “SRV” or “NATPTR/SRV”.

If “Use Configured IP” is selected, please fill in the three fields below:

  • Primary IP

  • Backup IP 1

  • Backup IP 2

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

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

Max Number Of Sip Request Retries

Sets the maximum number of retries for the device to send requests to the server. In DNS SRV configuration, if the destination address does not respond, all request messages are resent to the same address according to the configured retry times. Valid range: 1-10.

DNS SRV Failover Mode

Configures the preferred IP mode for DNS SRV. If set to “default”, the first IP from the query result will be applied. If set to “Saved one until DNS TTL”, previous IP will be applied before DNS timeout is reached. If set to “Saved one until no response”, previous IP will be applied even after DNS timeout until it cannot respond.

  • Default

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

  • Saved one until DNS TTL

If the option is set with “Saved one until DNS TTL”, it will send register messages to the previously registered IP first. If no response, it will try to send one at a time for each IP. This behavior lasts if DNS TTL (time-to-live) is up.

  • Saved one until no responses

If the option is set with “Saved one until no responses”, it will send register messages to the previously registered IP first, but this behavior will persist until the registered server does not respond.

  • Failback follows failback expiration timer

 If "Failback follows failback expiration timer" is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until the failback timer expires.

Failback Expiration (m)

Specifies the duration (in minutes) since failover to the current SIP server or Outbound Proxy before making failback attempts to the primary SIP server or Outbound Proxy.

Register Before DNS SRV Failover
 

Indicates whether a REGISTER request will be initiated when a server failover occurred under DNS SRV mode.
Default setting is No.

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 IP 1 address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

Backup IP 2

Configures the backup IP 2 address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

NAT Traversal

Configures whether NAT traversal mechanism is activated. Please refer to user manual for more details.

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)

Configures to use symmetric response routing. If it is used, the "rport" field will be added to the Via header field in the SIP Request, and the information will be extracted from the SIP 200OK Response for SIP Register to rewrite the SIP Contact information and apply it 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.

Use SBC

Configures whether a SBC server is used. Note: If enabled, make sure an outbound proxy is set up.

Account x 🡪 SIP Settings

Basic Settings

SIP Registration

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

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 “No”, the phone will not unregister the SIP user’s registration information before new registration.

  • 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.

REGISTER Expiration

Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified registrar.

The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.

SUBSCRIBE Expiration

Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified registrar.

The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.

Re-Register before Expiration

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

Registration Retry Wait Time

Specifies the interval to retry registration if the process failed. Valid range is 1 to 3600. Default is 20.

SIP SUBSCRIBE Retry Wait Time

Configures the time interval to retry sending SIP SUBSCRIBE request if receive error response. Valid range is 1 to 3600. Default is 20.

Add Auth Header on Initial REGISTER

If enabled, the phone will add Authorization header in initial REGISTER request.

Enable OPTIONS Keep-Alive

Enable OPTIONS Keep Alive to check SIP Server.

Default is “No”.

OPTIONS Keep-Alive Interval

Time interval for OPTIONS Keep Alive feature in seconds.

Default is “30” seconds.

OPTIONS Keep Alive Max Tries

Configures the maximum number of times the phone will try to send OPTIONS message consistently to server without receiving a response. If set to "3", the phone will send OPTIONS message 3 times. If no response from the server, the phone will re-register.

Enable TCP Keep-Alive

Configures whether to enable Keep-Alive for TCP connection. Default 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”.

Use Privacy Header

Controls whether the Privacy header will present in the SIP INVITE message or not, whether the header contains the caller info.

  • Default: The Privacy Header will show in INVITE only when “Huawei IMS” special feature is on.

  • Yes: The Privacy Header will always show in INVITE.

  • 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.

  • Default: The P-Preferred-Identity Header will show in INVITE unless “Huawei IMS” special feature is on.

  • Yes: The P-Preferred-Identity Header will always show in INVITE.

  • No: The P-Preferred-Identity Header will not 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. The 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.  The 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-Asserted-Identity Header

Configure whether the "P-Asserted-Identity Header" is present in the SIP INVITE message. The default setting is "No".

Use X-switch-info Header

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

Use MAC Header

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

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

  • If No, neither will the MAC header be included in the register or unregister message nor the MAC address be attached to the User-Agent header for any outgoing SIP message.

The default setting is "No".

Add MAC in User-Agent

  • If Yes except REGISTER, all outgoing SIP messages will include the phone’s MAC address in the User-Agent header, except for REGISTER and UNREGISTER.

  • If Yes to All SIP, all outgoing SIP messages will include the phone’s MAC address in the User-Agent header.

  • If No, the phone’s MAC address will not be included in the User-Agent header in any outgoing SIP messages.

The default setting is "No".

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

Determines whether or not to listen to multiple SIP protocols.

  • Transport Only: will listen to configured transport protocol only.

  • Dual: will listen to TCP when UDP is selected.

  • Dual (Secured): will listen to TLS/TCP when UDP is selected. If TCP or TLS/TCP is selected, UDP will be listened to

  • Dual (BLF Enforced): will try to enforce BLF subscriptions to use TCP protocol by adding ‘transport=tcp’ to the Contact header.

The default setting is “Transport Only”.

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 1024 to 65400.

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”.

Support SIP Instance ID

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

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.

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.

  • In route

  • Not in route

  • Always send to

Default is “in route”.

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”

Use Route Set In NOTIFY (Follow RFC 6665)

Configures whether to use route set in NOTIFY (follow RFC 6665). If enabled, the Request URI of the refresh subscription will use the URI in the received NOTIFY Contact (RFC 6665); otherwise, the URI in the previously subscribed 200 OK Contact will be used. The default setting is "Yes".

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 “No”.

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 session 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 "UAC".

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 🡪 Codec Settings

Audio

Preferred Vocoder

(Choice 1 – 8)

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.

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”.

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”.

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

If enabled, the phone always responds one m line in SDP regardless multiple m lines are offered.

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”.

G.726-32 Packing Mode

Selects “ITU” or “IETF” for G726-32 packing mode.

The default setting is “ITU”.

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.

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).

Default is “250” ms.

DTMF Payload Type

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

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”.

RTP Settings

SRTP Mode

Enable SRTP mode based on your selection from the drop-down menu.

  • No

  • Enabled But Not forced

  • Enabled and Forced

  • Optional

The default setting is “No”.

SRTP Key Length

Allows users to specify the length of the SRTP calls. Available options are:

  • AES 128&256 bit

  • AES 128 bit

  • 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”.

Enable RTCP

Enables user to select to use RTCP, RTCP-XR, or disable the feature.
Set to RTCP-XR by Default.

RTCP Mode

Configure RTCP port negotiation rules. If set to "default", it will use the traditional RTCP port, which is "RTP port+1".
If set to "Negotiate RTCP Port", it will use attribute RTCP to negotiate.
Set to "Default".

Collector Address Selection

When set to 'Manual' mode, the VQ RTCP-XR Collector Name/Address/Port will serve as the destination for publishing VQ reports. In 'Auto' mode, the phone will automatically use the same address as the SIP REGISTER to publish VQ reports.
Set to "Manual" By Default.

VQ RTCP-XR Collector Name

Configures the host name of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Address

Configures the IP address of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Port

Configure the port of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages. Default is “5060”.

Symmetric RTP

Defines whether symmetric RTP is supported or not.

Default setting is “No”.

Account x 🡪 Call Settings

General

Key As Send

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 default setting is “Pound (#)”.

No Key Entry Timeout

Configures the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the collected digits will be sent out. The default setting is 4.

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”.

Enable Call Waiting

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

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.
The default setting is “No”.

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.

Call Log

Configures Call Log setting on the phone.

  • Log All Calls

  • Log incoming/Outgoing Only (missed calls NOT recorded)

  • Disable Call Log

The default setting is “Log All Calls”.

Auto Answer

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

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

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 times as needed like   

       regular expression.

Note: Auto Answer Numbers can be split with ";", for example:

1x;2xxx;3x+

Intercom

Play warning tone for Auto Answer Intercom

When enabled, the phone will play warning tone when auto answer Intercom.
The default value is “Yes”.

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.

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 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 answer Intercom call

When enabled, the phone will mute the incoming intercom call.
The default value is “No”.

Transfer

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”.

Enable Recovery on Blind Transfer

Disables recovery to the call to the transferee on failing blind transfer to the target. The default setting is “Yes”.

Notes:

  • This feature only applies to blind transfer.

  • 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.

  • 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.
Default setting is “30”.

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”.

Hide Dialing Password

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

  • Prefix for Dialing Password

  • Password Length

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”.

Call Forward

Enable Forward All

If set to "Yes", all calls will be forwarded to the number specified below.

All To

Specifies the number to be forwarded to when enabled Forward all.

Enable Busy Forward

If set to "Yes", call will be forwarded to the number specified below on busy.

Busy To

Specifies the number to be forwarded to for Call Forward On Busy.

Enable No Answer Forward

Specifies the number to be forwarded to for Call Forward On Busy.

No Answer To

Specifies the number to be forwarded to for Call Forward On Busy.

No Answer Timeout (s)

Specifies the number to be forwarded to for Call Forward On Busy.
The default value is 12 seconds.
The valid range is 1 to 120.

Dial Plan

Dial Plan Prefix

Configures a prefix added to all numbers when making outbound calls.

Bypass Dial Plan

Enable/Disable the dial plan bypass while dialing through:

  • Contact

  • Call History Incoming Call

  • Call History Outgoing Call

  • Dialing Page

  • MPK

  • API

The default setting is “MPK”.

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;

  • Grammar:  x – any digit from 0-9;

  • Grammar: X – any character from 0-9, a-z, A-Z.

  • xx+ – at least 2-digit numbers

  • xx – only 2-digit numbers

  • ^ – exclude

  • [3-5] – any digit of 3, 4, or 5

  • [147] – any digit of 1, 4, or 7

  • <2=011> – replace digit 2 with 011 when dialing

  • | – the OR operand

  • , – second dial tone. For example: {0,x+} will play second dial tone after dialing 0 and all digits will be sent including 0

  • {X123} — match Z123, e123, 5123, …

  • 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.

  • Back slash “” — can be used to escape specific letters. E.g. if { park+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 11 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 dialing7 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 Display

Caller ID Display

Determines from where to locate caller ID to display or not on the phone

  • 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.

  • Disabled: All incoming calls are displayed with “Unavailable”.

  • From Header: the phone will display the caller ID based on the From Header in the incoming SIP INVITE.

The default setting is “From Header”.

Callee ID Display

Determines from where to locate callee ID to display or not on the phone.

  • 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

  • Disabled: Callee ID will be displayed as “Unavailable”.

  • To Header: Callee ID will not be updated and displayed as To Header.

The default setting is “Auto”.

Ringtone

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.

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”.

Match Incoming Caller ID

Specifies matching rules with number, pattern, or Alert Info text (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

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

  •  Remote Ringtone via Alert Info

The remote ringtone feature enables the use of a ringtone stream via a remote URL. The functionality of this feature works as follows: the following audio file named test.wav is uploaded onto an HTTP server and the remote URL is "http://192.168.5.165:8080/test.wav;info=ring3", the IP phone then attempts to use the provided URL first to play the ringtone. If the URL is not functional for some reason, it will then use the info=ring3 parameter, as the default ringtone.

Account x 🡪 Advanced 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

This option allows you to populate a list of trusted domain names used for TLS certificate verification. When obtaining certificates, the system verifies if the domain name matches any entry in the trusted domain list. By default, the remote proxy domain name and SIP server domain name are trusted. You can enter alphanumeric characters, hyphens, periods, and asterisks in the list. Wildcard domain names like "*.grandstream.com" are supported, as well as any domain ending with ".grandstream.com" will be trusted.
You can submit up to 10 trusted domain names for TLS certificate verification.

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”.

Omit charset=UTF-8 in MESSAGE

Omit charset=UTF-8 in MESSAGE content-type.

The default setting is “Disabled”.

Allow Unsolicited REFER

Allow Unsolicited REFER to accomplish an outgoing call.

  • Disabled

  • Enabled

  • Enabled/Force Auth

The default setting is “Disabled”.

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”.

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”.

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”.

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”.

MOH

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 “Enabled”.

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.

Advanced Features

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, Huawei IMS, PhonePower and UCM Call center depending on the server type. The default setting is “Standard”.

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”.

Conference URI

Configures Conference URI for N-way conference (Broadsoft 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 “Disabled”.

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 “Disabled”. With “Hoteling Event” enabled, user can access the Hoteling feature option by pressing the “BSCCenter” softkey.

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

Call Center Status

When set to “Yes”, the phone will send SUBSCRIBE to the server to obtain call center status. The default setting is “Disabled”.

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

Broadsoft Executive Assistant

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

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

Broadsoft Call Park

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

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

BLF (Busy Lamp Field)

Presence Eventlist URI

Configures Presence Eventlist URI to monitor the extensions on Multi-Purpose Keys.

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

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” or “Presence Eventlist”.

  • Disabled

  • BLF Eventlist

  • Presence Eventlist

The default setting is “Disabled”.

BLF Call-pickup

Configures BLF Call-pickup method:

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

  • Force BLF Call-pickup by prefix: The phone will only use Prefix as BLF pickup method.

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

The default setting is “Auto”.

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

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

Call Park Retrieve Feature Code

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

PUBLISH for Presence

Enables presence feature on the phone. The default setting is “Disabled”.

SCA

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.

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 🡪 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.

DND

DND Call Feature On

Configures DND feature code to turn on DND.

DND Call Feature Off

Configures DND feature code to turn off DND.

Call Forward Always

On

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

Off

Configures Call Forward Always feature code to deactivate unconditional call forwarding.

Target

Configures the extension for the call to 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 for the call to be forwarded to.

Call Forward No Answer

On

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

Off

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

Target

Configures the extension for the call to be forwarded to.

Call Forward No Answer Timeout (s)

Configures the timeout (in seconds) before the call is forwarded when there is no answer. Valid range is 1 to 120. The default setting is 12 seconds.

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.

Account Page Definitions

Phone Settings Page Definitions

Phone Settings 🡪 General Settings

Basic Settings

Local RTP Port

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

Local RTP Port Range

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

Use Random Port

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

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

Enable Fix for RTP Timestamp Jump

Makes RTP timestamps be continuous, if there is audio loss caused by timestamp jump.
Default is “No”

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.

STUN Server

The IP address or Domain name of the STUN server. STUN resolution results are displayed in the STATUS page of the Web GUI.
Only non-symmetric NAT routers work with STUN.

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.

Delay Registration

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

Enable Outbound Notification

Indicates whether Outbound Notification feature is enabled. Default is “Enabled”. For more details refer to [OUTBOUND NOTIFICATION SUPPORT].

Public Mode

Enable Public Mode

Configures to turn on/off the public mode for hot desking feature.
The default setting is “Disabled”.

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.

Allow Multiple Accounts

If set to "No", then after the user logs in to the public mode account on LCD, only the public mode account can be used on the phone even though there are other configured SIP accounts. If set to "Yes", then after the user logs in to the public mode account on LCD, other configured SIP accounts on the phone can also be used.

Note: This option requires enabling public mode to take effect.

Enable Remote Synchronization

Enables phone to automatically download current account’s setting from remote server and upload to the server.
Default setting is “Disabled”.

Server Type

Allows users to choose the type of the server (TFTP, FTP or HTTP) that stores personal files of public account.
Default is “TFTP”

Server Path

Defines server path that stores personal files of public account.

FTP/HTTP User Name

Specifies User Name to access FTP/HTTP server.

FTP/HTTP Password

Specifies Password to access FTP/HTTP server.

Login Timeout

Configures Login timeout in Minute in public mode.
The default value is 10.

Settings 🡪 Settings

General

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.

The default setting is “Account Mode”.

Preferred Default Account

Selects the preferred default account when offhook/onhook dialing. When selected account is unavailable, system will fall back to use the first available account instead.

Select Account from LCD

Configures whether the user can use the Up/Down key to select an account in the idle screen.

Mute Key Functions While Idle

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

When select “Idle Mute” and press Mute key while idle, the future incoming call will be answered with mute. When select “Disabled”, Mute key will not take effect while idle.

The default setting is “DND”.

Last Call Forward Always

Configures to enable storing the last input number when entering number in the call screen after pressing the ForwardAll softkey.
Default is “No”.

Show SIP Error Response

Shows SIP error response information on LCD screen. The default setting is “Yes”.

Do Not Escape '#' as %23 in SIP URI

Replaces # by %23 for some special situations.

User-Agent Prefix

Configures the prefix in the User-Agent header.

Enable Enhanced Acoustic Echo Canceller

Enables/Disables Enhanced Acoustic Echo Canceller (EAC) providing acoustic echo reduction which is required for full-duplex handsfree speaker phone functions on the phone.
The default setting is “Yes”.

Enable Hook Switch

When set to "No", disable hook switch completely; When set to "Yes, except answering call", hook switch cannot be used for answering call. The default is "Yes".

Disable Speaker Key

When set to "Yes", the user can disable the speaker key completely. When set to "For Ongoing Call", the user can not hang up the call using the speaker key.

Contact Source priority

Configures the order of the contact sources for ID lookup in incoming/outgoing calls.

Outgoing

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”.

For more details refer to [CLICK-TO-DIAL].

Enable Paging Call Mode

Configures if a user is able to dial out a paging call.

Enable Direct IP Call

Enables Direct IP Call feature.

The default setting is “Yes”.

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, press # to switch to “Direct IP Call” mode 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

Allows users to show/hide predictive dialing feature, when disabled, users will not see any predictive numbers while dialing a number. Default setting is “Enabled”.

Predictive Dialing Source

Searches sequentially then number while dialing based on the selected sources from these: Call History, Local Phonebook, Remote Phonebook, Feature Code. Press “Modify” to edit available options.

Onhook Dial Barging

Allows incoming call to interrupt on-hook dialing when set to “Enabled”. Default setting is “Enabled”.

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 4.

Off-hook Timeout (s)

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

Enables to Dial out automatically the number punched in after the number of seconds that the user had set when the phone is off-hook.
Default value is “No”

Live Keypad Expire Time

Sets the Live Keypad expiration time before initiating the call using the Live Keypad feature. Interval is between 2s and 15s. The default value is 5s.

Enable Auto Redial

Enables the phone to redial automatically when called number is busy.

If enabled, the phone will prompt the user to start “automatic redial” or no. If yes, the phone will redial called number several times [Automatic Redial Times] with [Automatic Redial Interval] between each call. The user is guided via different prompts on phone’s LCD displaying number of remaining attempts, count-down to initiate next auto redial and allowing user to manually initiate the call without waiting for the specified interval [Automatic Redial Interval]. The phone will stop automatic redial after successful attempt (called party not busy) or after unsuccessful attempts [Automatic Redial Times].
Note: For auto redial feature to take effect, voicemail must be enabled for the called extension
The default setting is “No”.

Auto Redial Times

The number of times to attempt to call using Automatic Redial feature. The valid range is 1 – 200.
The default value is “10”.

Auto Redial Interval

The interval between each call attempt using Automatic Redial feature. The valid range is 1 – 360 seconds.
The default value is “10”.

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.
The default setting is “No”.

Incoming

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.
The default setting is “Yes”.

Enable Missed Call Notification

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

The default setting is “Yes”.

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. Available options are:

  • Busy (486).

  • Temporarily Unavailable (480).

  • Not found (404).

  • Decline (603).

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.
Default setting is “No”

Enable Call Waiting

Enable the call waiting feature.

The default setting is “Yes”.

Enable Call Waiting Tone

Enables Call Waiting alert tone when another incoming call is received while a call is in progress.

Default setting is “Yes”.

Ring For Call Waiting

Configures the phone to ring instead of playing call waiting tone when handset or headset is used.
The default setting is “No”.

Auto Answer Delay

Configure the delay for automatically answering the incoming call. Valid range is 0 to 10 (second). The default value is 0 (which means auto answer is disabled).

In Call

Enable in-call DTMF Display

Enables/disables the display of entered DTMF digits on the phone LCD during the call.

The default setting is “Yes”.

Enable Sending DTMF via specific MPKs

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

The default setting is “No”.

Show On Hold Duration

Show the duration of holding a call on the LCD.

The default setting is “Yes”.

Enable Auto Unmute

If the option is enabled, automatically unmute phone when an user unholds the call or establishes a new call. The default setting is "No".

In-call Dial Number on Pressing Transfer Key

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

Enable Busy Tone on Remote Disconnect

Enables the busy tone heard in the handset when call is disconnected remotely.
The default setting is “Yes”.

Transfer

Enable Transfer

Enables/disables transfer feature. If disabled, call transfer will not be possible.
Default setting is “Yes”.

Hold Call Before Completing Transfer

When set to "No", the phone will not hold the current call or the transfer target for an Attended Transfer. The default setting is "Yes".

Attended Transfer Mode

If set to “Static”, attended transfers can only be performed with pre-established calls. If set to “Dynamic”, attended transfers can be performed with pre-established calls OR be initiated during the transfer process. This option does not affect the user’s ability to perform blind transfers.

The default setting is “Dynamic”.

For more details about “Static” / “Dynamic” transfer, refer to the user guide.

DND

Enable DND Feature

If set to “No”, the user cannot turn on Do Not Disturb feature via MUTE key, MPK, or menu on LCD.

The default setting is “Yes”.

Return Code When Enable DND

When DND is enabled, the phone will send the selected type of SIP message. Available options are:

  • Busy (486).

  • Temporarily Unavailable (480).

  • Not found (404).

  • Decline (603).

Default setting is “Temporarily Unavailable (480)”.

DND Override

Allows the phone to accept certain incoming calls while set to DND mode.

  • Off: all incoming calls will not be accepted.

  • Allow all: all incoming calls will be allowed.

  • Allow Only Contacts: only incoming calls from numbers in the local phonebook will be accepted.

  • Allow Only Favorites: only incoming calls from favorite numbers in the local phonebook will be accepted.

The default setting is “Off”.

Conference

Enable Conference

Enables the Conference feature. The default setting is "Yes".

BLF

Enable BLF Pickup Screen

By enabling BLF Pickup Screen, when monitored BLF is ringing, GRP261x/GRP2624/GRP2634/GRP2670/GRP2650 will 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, GRP261x/GRP2624/GRP2634/GRP2670/GRP2650 will play a sound to inform user.
The default setting is “No”.

BLF Pickup Sound Except List

Configures the list to be playing BLF sound notification for “All Except” extensions in the list [BLF Pickup Sound Except List] or “Only Allow” extensions in the list [BLF Pickup Sound Only List].
The default setting is “Allow Except”.

Hide BLF Remote Status

Allows users to hide the Caller ID from showing at the BLF VPK and MPK.

  • No: The VPK will flash between the Caller ID and the BLF account.

  • Yes: The VPK will stay under the monitored account and only notify that there is an incoming call.

The default setting is “No”.

IM

Enable IM Popup

If set to “No”, phone will not show a pop up when receiving an IM.
The default setting is “Yes”.

Instant Message Popup Timeout

Configures the number of seconds that the message will remain on screen. The valid range is 10 – 900.

The default setting is “10”.

Play Tone On Receiving IM

If enabled, phone will play a short tone when receiving an IM during idle state.
The default setting is “Disabled”.

Call Features

Enable Active MPK Page

When the option is enabled, Active MPK Page on the extension boards will be disabled.

The default setting is “No”.

Enable Active VPK Page

Enables Active VPK Page to be displayed on LCD when there are active VPKs.

The default setting is “No”.

Enable Call Recording LCD Indicator

Configures whether to show the call recording indicator on LCD for local and remote call recording.

Enabled by Default.

Local Call Recording Feature

Gives the ability to record calls locally while on the call screen, the available options are:

  • No: the local record feature is disabled.

  • Yes: the local record feature is enabled.

  • Yes & Auto-Start: the local record feature is enabled and recording will automatically start.

The default setting is “No”.

Note: The IP phone displays a prompt when the storage for recording files is almost full. This alert helps users manage the space by deleting or transferring recordings to avoid storage issues.

Default call log type

Sets the default call log list after select MENU🡪CALL HISTORY.

Broadsoft Call Log or Local Call Log option will only show its own list. Default option will keep both call log lists.
The default setting is “Default”.

Saved Local Call Recording Location

Configures location where the recordings will be stored.

Replace the oldest call record

When enabled, the oldest call record will be replaced with the newest one when the storage is full. If the option is disabled, the call recording feature will stop recording automatically.
Default is “Disabled”.

Download Local Call Recordings

When there are recordings presented, you may download them here.
Note: The file name of the recording has been improved to include information such as the exact time and date of the call and the from/to accounts that are part of the call.

Settings 🡪 Ringtone

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 (Low, Medium or High). 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 and default setting is 5.

Call Tone Volume

Used to configure the call tones’ level in dB. Values range from -15 to 15.

Lock Speaker Volume

Lock volume adjustment when the option is enabled so it cannot be changed from phone LCD. The option can be set to: “No”, “Ring”, “Talk” or “Both”.

Default setting is “No”.

Default Ringtone

Allows to set Default Ringtone as their Global ringtone.

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

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

Alert-info Remote Ringtone Download Timeout

Configures how long (in seconds) before the phone falls back to using the local ringtone if the Alert-Info remote ringtone fails to download.

Provision

Total Number of Custom Ringtone Update

Configures the number of custom ringtones to update in the provisioning process. The default setting is 3. The valid range is 0 - 10.

Settings 🡪 Multicast Paging

Multicast Paging Function

Enable or disable multicast paging. The default setting is "No".

Allowed in DND Mode

Allow Multicast Paging when DND mode is enabled.
Default Setting is “No”.

Paging Barge

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

Paging Priority Active

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

Multicast Channel Number

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

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), G.723.1.
Default setting is “G.722(wide band)”.

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.

Phone Settings Page Definitions

Network Page Definitions

Network Settings 🡪 Ethernet Settings

Internet Protocol

Selects “IPv4 Only”, “IPv6 Only”, “Both, prefer IPv4” or “Both, prefer IPv6”. The default setting is “IPv4 only”.

IPv4 Address

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 Internet Service Providers.

Vendor Class ID (Option 60)

Used by clients and servers to exchange vendor class ID.

PPPoE Account ID

Enter the PPPoE account ID.

PPPoE Password

Enter the PPPoE Password.

PPPoE Service Name

Enter the PPPoE Service Name.

Ipv4 Address

Enter the IP address when static IP is used.

Subnet Mask

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

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

IPv6 Address Type

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

Static IPv6 Address

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

IPv6 Prefix Length

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

IPv6 Prefix(64 bits)

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.

802.1X

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-PEAPv0/MSCHAPv2.
Note:  EAP-PEAP encryption supports Microsoft servers.

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.

Network 🡪 Advanced Settings

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

Configures the destination IP address where no proxy server is needed. The phone will not use a proxy server when sending packets to the specified destination IP address.

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

Configures whether the phone will release the DHCP lease on reboot.

Disabled by Default.

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 valid range is 0 – 4094.
The default value is 0.

Layer 2 QoS 802.1p Priority Value

Assigns the priority value of the Layer2 QoS packets. The valid range is 0 – 7.
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 valid range is 0 – 4094.
The default value is 0.

PC Port Priority Value

Assigns the priority value of the PC port. The valid range is 0 – 7.
The default value is 0.

Enable CDP

Enables/Disables CDP “Cisco Discovery Protocol”.
The default setting is “Enabled”.

Enable LLDP

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

LLDP TX Interval

Defines LLDP TX Interval (in seconds). Valid range is 1 to 3600.
The default setting is “60”.

Maximum Transmission Unit (MTU)

Defines the MTU in bytes. The valid range is 576 – 1500.
The default value is 1500 bytes.

 Remote Control

Action URI Support

Enables/disables action URI feature on the phone.
The default setting is “Enabled”.

Remote control Pop up window support

Indicates whether the phone is enabled to pop up allow remote control.
The default setting is “Enabled”.

Action URI allowed IP list

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

CSTA Control

Indicates whether CSTA Control feature is enabled. Change of this configuration will need the system to reboot to take effect. The default setting is “Disabled”.

 CTI Settings (GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2636, GRP2650 & GRP2670 only)

Affinity Support

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

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

Preferred Account

Selects the account on which CTI support is enabled.

Static DNS Cache

NAPTR

NAPTR (Naming Authority Pointer) records are used to specify rules for rewriting one type of domain name to another, typically used for handling Uniform Resource Identifiers (URIs) within the domain, when you configure NAPTR in the static DNS cache, you are specifying custom rules for how specific URIs or domain names should be resolved, the options to configure are :

  • NAPTR DNS Cache Name: The domain name to which this resource record refers.

  • NAPTR DNS Cache Time Interval (s): The time interval that the resource record may be cached before the source of the information should again be consulted, Default value is 300 seconds.

  • NAPTR DNS Cache Order: A 16-bit unsigned integer specifying the order in which the NAPTR records must be processed to ensure the correct ordering of rules.

  • NAPTR DNS Cache Preference: A 16-bit unsigned integer that specifies the order in which NAPTR records with equal "order" values should be processed, with low numbers being processed before high numbers.

  • NAPTR DNS Cache Replacement: The next name to query for SRV records.

  • NAPTR DNS Cache Service: Specifies the service(s) available down this SRV record path.

SRV

SRV records are DNS records used to identify servers that provide specific services, such as email, SIP (Session Initiation Protocol) servers, or other services, Configuring SRV in the static DNS cache allows you to specify which servers should be used for particular services, helping ensure that your IP phone connects to the correct servers for specific functions, the available options to configure are:

  • SRV DNS Cache Name: The domain name string with SRV prefix.

  • SRV DNS Cache Time Interval (s):  Specifies the time interval that the resource record may be cached before the source of the information should again be consulted. The default value is 300 seconds.

  • SRV DNS Cache Priority: Set the priority of this target host.

  • SRV DNS Cache Weight: Set server selection mechanism.

  • SRV DNS Cache Target: The domain name of the target host.

  • SRV DNS Cache Port: Set the port on the target host of this service.

A

A records are used to map a domain name to an IPv4 address. They are the most common type of DNS record and are used to resolve domain names to IP addresses, Configuring A records in the static DNS cache allows you to manually specify the IP addresses associated with specific domain names, ensuring that your IP phone always connects to the intended destination, the options to configure are: 

  • A DNS Cache Name: Set Hostname.

  • A DNS Cache Time Interval: A DNS Cache Time Interval, Default is 300 seconds.

  • A DNS Cache IP Address: A DNS Cache IP Address.

Network 🡪 Bluetooth Settings (GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2636, GRP2650 & GRP2670 only)

Bluetooth Power

Configures Bluetooth to power “On”, “Off” or “Off & Hide Menu From LCD”.

If set “Off & Hide Menu From LCD”, Bluetooth will be disabled, and users will not find Bluetooth settings on phone LCD Menu, while if set to “No”, Bluetooth will be disabled, and Bluetooth Settings menu will be available, and user can enable it. The default setting is “On”.

Handsfree Mode

Enable / disable Bluetooth handsfree feature. Default setting is “Off”.

Bluetooth Name

Specifies the Bluetooth device name.

Network 🡪 OpenVPN® Settings

OpenVPN® Enable

Enables/Disables OpenVPN® feature. Default is “No”.

OpenVPN® Mode

Selects OpenVPN® mode to use:

  • Simple mode: Using simple mode, the administrator needs to configure the OpenVPN settings below.

  • Expert mode: After switching to “expert mode”, the administartor can manually upload a single OpenVPN client(.ovpn) file

Upload OpenVPN® config zip file

Upload OpenVPN® .zip file containing .ovpn file when OpenVPN® Mode is set to "Expert Mode"

Note: This field appears only when "OpenVPN® Mode" is set to "Expert Mode".

OpenVPN® Server Address

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

OpenVPN® Port

Specify the listening port of the OpenVPN® server. The valid range is 1 – 65535. The default value is “1194”.

OpenVPN® Transport

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


The default value is “UDP”

OpenVPN® CA

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

OpenVPN® Certificate

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

OpenVPN® Client Key

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

OpenVPN® Cipher Method

Specifies the Cipher method used by the OpenVPN® server. The available options are:

  • Blowfish

  • AES-128

  • AES-256

  • Triple-DES

The default setting is “Blowfish”.

OpenVPN® Username

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

OpenVPN® Password

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

OpenVPN® Comp-lzo

Configures enable/disable the LZO compression. When the LZO Compression is enabled on the OpenVPN server, you must turn on it at the same time. Otherwise, the network will be abnormal. Default value is YES.

Additional Options

Additional options to be appended to the OpenVPN® config file, separated by semicolons. For example, comp-lzo no;auth SHA25

Note: Please use this option with caution. Make sure that the options are recognizable by OpenVPN® and do not unnecessarily override the other configurations above.

Network 🡪 SNMP Settings

Enable SNMP

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

Version

SNMP version. Select Version 1, Version 2 or Version 3.


Default is “Version 3”.

Port

SNMP port. The valid range is 161, 1025-65535.
The default value is “161”.

Community

SNMP Community.

SNMP Trap Version

Choose the Trap version of the SNMP trap receiver.

  • Trap Version 1

  • Trap Version 2

  • Trap Version 3

The default is “Trap Version 2”.

SNMP Trap IP

IP address of trap destination.

SNMP Trap port

Port of the SNMP trap receiver. The valid range is 162, 1025-65535. The default value is “162”.

SNMP Trap Interval

The interval between each trap sent to the trap receiver. The valid range is 1 – 1440.The default value is “5”

SNMP Trap Community

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

SNMP Username

Username for SNMPv3

Security Level

  • noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.

  • authUser: Users with security level authNoPriv and context name as auth.

  • privUser: Users with security level authPriv and context name as priv.

Authentication Protocol

Select the Authentication Protocol:

  • None

  • MD5

  • SHA

The default setting is “None”.

Privacy ProtocolNone

Select the Privacy Protocol:

  • None

  • DES

  • AES

Authentication Key

Enter the Authentication Key.

Privacy Key

Enter the Privacy Key.

SNMP Trap Username

Username for SNMPv3 Trap.

Trap Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.

authUser: Users with security level authNoPriv and context name as auth.

privUser: Users with security level authPriv and context name as priv.

Trap Authentication Protocol

Select the Authentication Protocol: “None” or “MD5” or “SHA”.
The default setting is “None”.

Trap Privacy Protocol

Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
The default setting is “None”.

Trap Authentication Key

Enter the Trap Authentication Key.

Trap Privacy Key

Enter the Trap Privacy Key.

Network 🡪 WiFi Settings (Available on GRP2612W & GRP2614 & GRP2615 & GRP2616 & GRP2624 & GRP2634 & GRP2636 &  GRP2650 & GRP2670 only)

Enable/Disable WiFi

Enables / Disables the WiFi on the phone. Three options are available:

  • No: Disables WiFi. User has ability to enable WiFi from LCD Menu.

  • Off & Hide Menu from LCD: Disables WiFi and hides “WiFi Settings” menu from phone LCD.

  • Yes: Enables WiFi to connect to WiFi network.

Country

Specifies the Wi-Fi encryption type.

Access Point (1 – 10)

SSID: Enters WiFi SSID name to connect.

Password

Configures the authentication password to access WiFi Network

Security Type

Specifies the WiFi encryption type. The available options are the following: None, WEP, WPA, WPA Enterprise and Auto. Default settings is None.

VoWLAN Target Delay

Configures the amount of jitter buffer target delay over Wi-Fi. Low is 100ms, Medium is 200ms, and
High is 400ms.
The Default value is set to "Low"

Network Page Definitions

Programmable Keys Page Definitions

Programmable keys 🡪 Multi-Purpose Keys (GRP2614 & GRP2616 & GRP2634 & GRP2636 only)

Keys Settings

Mode

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.