Thank you for purchasing Grandstream GRP26XX Carrier-Grade IP Phones.
GRP2612/GRP2612P/GRP2612W/GRP2612G is featuring 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), integrated dual-band Wi-Fi (GRP2612W only), 3-way conference, and Electronic Hook Switch (EHS).
GRP2613 is featuring 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, integrated PoE, 3-way conference, and Electronic Hook Switch (EHS).
GRP2614 is featuring 4 dual-color line keys (which 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, 3-way conference and Electronic Hook Switch (EHS).
GRP2615 is featuring 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, 3-way conference and Electronic Hook Switch (EHS).
GRP2616 is a next-generation enterprise IP Phone featuring 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, USB port, 1000M network ports, integrated PoE, Wi-Fi, and Bluetooth support, 3-way conference and Electronic Hook Switch (EHS).
The GRP2624 is featuring 8-line dual-color line keys (which can be digitally programmed as up to 32 provisionable BLF/fast-dial keys), 2.8 inches (320×240) TFT color LCD, 4 programmable context-sensitive soft keys, 1000M network ports, integrated PoE, Wi-Fi and Bluetooth support, 5-way conference and Electronic Hook Switch (EHS).
The GRP2634 is featuring 8-line dual-color line keys (can be digitally programmed as up to 32 provisionable BLF/fast-dial keys), 2.8 inches (320×240) TFT color LCD, 4 programmable context-sensitive soft keys, 10 multi-purpose keys, 1000M network ports, integrated PoE, Wi-Fi, and Bluetooth support, 5-way conference and Electronic Hook Switch (EHS).
The GRP2636 is a professional 12-line model designed with zero-touch provisioning for mass deployment and easy management. It features a sleek design and a suite of next-generation features including 5-way voice conferencing to maximize productivity, integrated PoE & Wi-Fi, and full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity. The GRP series includes carrier-grade security features to provide enterprise-level security, including secure boot, dual firmware images, and encrypted data storage.
The GRP2670 is a next-generation enterprise IP Phone featuring 6 Account Lines, 7” (1042×600) capacitive touch TFT color LCD, 10/1001000M network ports, integrated PoE, integrated dual-band Wi-Fi (2.4GHz 5GHz), Integrated Bluetooth, 5-way conference, and Electronic Hook Switch (EHS).
The GRP2650 is a professional 14-line model designed with zero-touch provisioning for mass deployment and easy management. It features a sleek design and a suite of next-generation features
including 5-way voice conferencing to maximize productivity, integrated PoE & Wi-Fi, full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS support for Plantronics headsets, and integrated USB headset support.
The GRP26XX series delivers superior HD audio quality, rich and leading-edge telephony features, protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. GRP26XX series is the perfect choice for enterprise users looking for a high-quality, feature-rich multi-line executive IP phone with advanced functionalities and performance.
PRODUCT OVERVIEW
Feature Highlights
The following table contains the major features of the GRP26XX phones:
 GRP2612 GRP2612P GRP2612W GRP2612G |
|
 GRP2613 |
|
 GRP2614 |
|
 GRP2615 |
|
 GRP2616 |
|
 GRP2624 |
|
 GRP2634 |
|
 GRP2636 |
|
 GRP2670 |
|
 GRP2650 |
|
Table 1: GRP261x/GRP2624/GRP2634 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) |
Graphic Display | 2.4 inch (320×240) TFT color LCD |
Feature Keys | 4 line keys with up to 2 SIP accounts, 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) |
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. |
Table 2: 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 3 SIP accounts, 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) |
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. |
Table 3: 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 version 5 |
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) |
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. |
Table 4: 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 version 5 |
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) |
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. |
Table 5: 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 version 5 |
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) |
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. |
Table 6: 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 intergrated |
Wi-Fi | Yes, dual-band |
Feature Keys | 8 line keys with up to 4 SIP accounts, 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. |
Table 7: 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, |
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 4 SIP accounts, 10 MPK extension keys with paper slot, 4 XML programmable context-sensitive softkeys, 5 navigation/menu keys, 9 dedicated |
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, |
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, |
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) |
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: |
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 |
Compliance | FCC: Part 15 (CFR 47) Class B |
Table 8: 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, |
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 |
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, |
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, |
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) |
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: |
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, |
Physical | Dimension: 220mmx 210mmx 82mm |
Compliance | FCC: FCC Part 15 Class B; FCC Part 15 Subpart C,15.247; FCC Part 15 Subpart E,15.407; |
Table 9: 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) |
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; |
Table 10: 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, |
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, |
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, |
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, |
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) |
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: |
Physical | Unit weight:1050g ; Package weight:1620g |
Temperature and Humidity | Operation: 0°C to 40°C |
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 |
Table 11: 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 |
|
Table 12: Equipment Packaging
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.
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).
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.
- 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
|
Contacts | Contacts sub menu includes the following options:
Contacts sub menu is for Local Phonebook, Local Group, LDAP Directory and Broadsoft Phonebooks. User could configure phonebooks/groups/LDAP options here, download phonebook XML to the phone and search phonebook/LDAP directory. |
Messages | Message sub menu 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:
– Appearance:
– MPK LCD Settings (Available on GRP2614/GRP2616 only):
– Language and Input:
– Date Time:
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:
|
System | System sub menu includes the following options: – Network:
– Wi-Fi Settings (GRP2612W, GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2670 & GRP2650 only):
– Bluetooth Settings (GRP2614/GRP2615, GRP2616, GRP2624, GRP2634, GRP2670 & GRP2650 only):
– Web Access:
– Upgrade:
– Language Download:
– Factory Functions:
– 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:
– Operation:
|
Reboot | Reboots the phone. |
Table 13: 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:
- Connect the computer to the same network as the phone.
- 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.
- Open a Web browser on your computer.
- Enter the phone’s IP address in the address bar of the browser.
- Enter the administrator’s login and password to access the Web Configuration Menu.
- 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.
|
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. |
Software Version |
|
IP Geographic Information |
|
Special Feature |
|
System Up Time | System up time since the last reboot. |
System Time | Current system time on the phone system. |
Service Status | GUI, Phone and CPE service status. |
System Information | Download system information |
User Space | Shows the percentage of the user space used and the status of the Database |
Core Dump | Shows the status of the core dump and the core dump files generated if any. It also gives the ability to generate GUI/Phone core dump files manually. |
Screenshot | Download captured screenshots. Press “Start” button to clear screenshots. |
Special Feature | Displays wether OpenVPN® is supported. |
Status 🡪 Call Status | |
Display the calls status. Refer to [Maintenance > Voice Monitoring > Display Report] | |
Status 🡪 Programmable Keys Status 🡪Virtual Multi-Purpose Keys | |
VPKs Status |
|
Status 🡪 Programmable Keys Status 🡪 Softkeys Status | |
Softkeys |
|
Status 🡪 Extension Boards Status (GRP2615 & GRP2650 only) | |
Extension (1-4) Keys | EXT (1-160):
|
Status 🡪 Call Feature Status | |
Accounts |
|
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 | Configures the LCD brightness when the phone is active. Valid range is 10 to 100 where 100 is the brightest. |
Active Backlight Timeout | Configures 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) | Configures 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. |
Table 14: 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. If “Use Configured IP” is selected, please fill in the three fields below:
If SIP server is configured as domain name, phone will not send DNS query, but use “Primary IP” or “Backup IP x” to send SIP message if at least one of them are not empty. Phone will try to use “Primary IP” first. After 3 tries without any response, it will switch to “Backup IP x”, and then it will switch back to “Primary IP” after 3 re-tries. If SIP server is already an IP address, phone will use it directly even “User Configured IP” is selected. |
DNS SRV Failover Mode | 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.
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.
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.
If the option is set with “Saved one until no responses”, it will send register messages to the previously registered IP first, but this behavior will persist until the registered server does not respond. |
Register Before DNS SRV Failover | Indicates whether a REGISTER request will be initiated when a server failover occurred under DNS SRV mode. |
Primary IP | Configures the primary IP address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode. |
Backup IP 1 | Configures the backup 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. |
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”.
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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 Failure 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 1. |
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 “Yes”. |
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.
The default setting is “default”. |
Use P-Preferred-Identity Header | Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message.
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 paresent 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 |
The default setting is "No". |
Add MAC in User-Agent |
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.
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 1 to 65535. |
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.
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:
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.
The default setting is “No”. |
SRTP Key Length | Allows users to specify the length of the SRTP calls. Available options are:
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. |
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. |
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. |
Enable Call Waiting | Enables / disables the call waiting feature for the current account. When set to “Default”, global call feature setting will be used. |
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. |
Ring Timeout | Defines the timeout (in seconds) for the rings on no answer. The default setting is 60. |
Call Log | Configures Call Log setting on the phone.
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. |
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:
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. |
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. |
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. |
Mute on answer Intercom call | When enabled, the phone will mute the incoming intercom call. |
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. |
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:
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Blind Transfer Wait Timeout | Defines the timeout (in seconds) for waiting SIP frag response in blind transfer. Valid range is 30 to 300. |
Refer-To Use Target Contact | If set to “Yes”, the “Refer-To” header uses the transferred target’s Contact header information for attended transfer. |
Hide Dialing Password | Allows users to hide the password when the dialing number matches the configured prefix.
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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. |
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:
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;
Mark = “-“ / “_” / “.” / “!” / “~” / “*” / “’” / “(“ / “)”
Allow 311, 611, and 911 or any 11 digit numbers with leading digits 1617;
Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers;
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 of a simple dial plan used in a Home/Office in the US: { ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 } Explanation of example rule (reading from left to right):
Note: In some cases, where the user wishes to dial strings such as *123 to activate voice mail or other applications provided by their service provider, the * should be predefined inside the dial plan feature. |
Call Display | |
Caller ID Display | Determines from where to locate caller ID to display or not on the phone
The default setting is “From Header”. |
Callee ID Display | Determines from where to locate callee ID to display or not on the phone.
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:
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.
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.
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. |
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.
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. |
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”.
The default setting is “Disabled”. |
BLF Call-pickup | Configures BLF Call-pickup method:
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. |
Table 15: 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. |
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. |
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. |
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. |
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. |
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. |
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. |
Server Type | Allows users to choose the type of the server (TFTP, FTP or HTTP) that stores personal files of public account. |
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. |
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. |
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. |
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". |
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. |
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. |
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. |
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]. |
Auto Redial Times | The number of times to attempt to call using Automatic Redial feature. The valid range is 1 – 200. |
Auto Redial Interval | The interval between each call attempt using Automatic Redial feature. The valid range is 1 – 360. |
Bypass Dial Plan Through Call History and Directories | Enable/Disable the dial plan check while dialing through the call history and any phonebook directories. |
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. |
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. |
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:
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. |
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. |
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. |
Transfer | |
Enable Transfer | Enables/disables transfer feature. If disabled, call transfer will not be possible. |
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:
Default setting is “Temporarily Unavailable (480)”. |
DND Override | Allows the phone to accept certain incoming calls while set to DND mode.
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. |
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. |
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]. |
Hide BLF Remote Status | Allows users to hide the Caller ID from showing at the BLF VPK and MPK.
The default setting is “No”. |
IM | |
Enable IM Popup | If set to “No”, phone will not show a pop up when receiving an IM. |
Instant Message Popup Timeout | Configures the number of seconds that the message will remain on screen. 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. |
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 default setting is “Disabled”. 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. |
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. |
Download Local Call Recordings | When there are recordings presented, you may download them here. |
Settings 🡪 Ringtone | |
Call Progresses Tones:
| 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]]]; 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 default setting is 5. |
Notification Tone Volume | Configures notification tone volume. |
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. |
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. |
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. |
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. |
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. |
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:
For details, please check the “Multicast Paging User Guide” on our Website. |
Table 16: 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. |
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. |
Layer 3 QoS for RTP | Defines the Layer 3 QoS parameter for RTP. This value is used for IP Precedence, Diff-Serv or MPLS. |
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. |
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. |
Layer 2 QoS 802.1Q/VLAN Tag | Assigns the VLAN Tag of the Layer 2 QoS packets. The valid range is 0 – 4094. |
Layer 2 QoS 802.1p Priority Value | Assigns the priority value of the Layer2 QoS packets. The valid range is 0 – 7. |
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. |
PC Port VLAN Tag | Assigns the VLAN Tag of the PC port. The valid range is 0 – 4094. |
PC Port Priority Value | Assigns the priority value of the PC port. The valid range is 0 – 7. |
Enable CDP | Enables/Disables CDP “Cisco Discovery Protocol”. |
Enable LLDP | Controls the LLDP (Link Layer Discovery Protocol) service. |
LLDP TX Interval | Defines LLDP TX Interval (in seconds). Valid range is 1 to 3600. |
Maximum Transmission Unit (MTU) | Defines the MTU in bytes. The valid range is 576 – 1500. |
Network 🡪 Remote Control | |
Action URI Support | Enables/disables action URI feature on the phone. |
Remote control Pop up window support | Indicates whether the phone is enabled to pop up allow remote control. |
Action URI allowed IP list | List of allowed IP address from which the phone receives action URI. The Allowed IP addresses are separated by a comma such as “192.168.1.1,192.168.1.2”. Set this field to “any” to allow any IP address to send Action URL to the phone. The default value is empty string which means no IP address is allowed for remotely control the phone. |
CSTA Control | Indicates whether CSTA Control feature is enabled. Change of this configuration will need the system to reboot to take effect. The default setting is “Disabled”. |
Network 🡪 Affinity Settings (GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2670 & 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”. |
Preferred Account | Selects the account on which CTI support is enabled. |
Network 🡪 Bluetooth Settings (GRP2614, GRP2615, GRP2616, GRP2624, GRP2634, GRP2670 & 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:
|
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.
|
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. |
OpenVPN® Cipher Method | Specifies the Cipher method used by the OpenVPN® server. The available options are:
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. |
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.
|
Port | SNMP port. The valid range is 161, 1025-65535. |
Community | SNMP Community. |
SNMP Trap Version | Choose the Trap version of the SNMP trap receiver.
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 |
|
Authentication Protocol | Select the Authentication Protocol:
The default setting is “None”. |
Privacy ProtocolNone | Select the Privacy Protocol:
|
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”. |
Trap Privacy Protocol | Select the Privacy Protocol: “None” or “AES/AES128” or “DES”. |
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 & GRP2650 & GRP2670 only) | |
Enable/Disable WiFi | Enables / Disables the WiFi on the phone. Three options are available:
|
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 |
Table 17: Network Page Definitions
Programmable Keys Page Definitions
Programmable keys 🡪 Multi-Purpose Keys (GRP2614 & GRP2616 & GRP2634 & GRP2636 only) | |
Keys Settings | |
Mode | Speed Dial:
Busy Lamp Field (BLF):
Presence Watcher:
Eventlist BLF:
Speed Dial via active account:
Dial DTMF:
Voicemail:
Call Return:
Transfer:
Call Park:
Intercom
LDAP Search:
For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters: Description -> ou=video,ou=SZ,dc=grandstream,dc=com Value -> sn=Li Since the Base for LDAP server configuration is: “dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the LDAP search scope. “sn=Li” is the example to filter the last name. Conference:
Multicast Paging:
Record:
Call Log:
Monitored Call Park:
Menu:
Information:
Message:
Forward:
DND:
Redial:
Presence Eventlist:
Note: The PBX server has to support this feature. Provision:
Opendoor:
Multicast Listen Address:
Multicast Paging Address:
Note: An MPK configuration tutorial video link can be found on the MPK configuration page. |
Account | Select the account to be associated with the configured MPK. |
Value | Enter the value to be associated with the configured MPK. (Extension Number, Multicast address...) |
Label | Enter the name to be associated with the MPK. |
Preview | Shows a preview of the configured MPK label. After saving, you can print the card style in the preview. For more info about how to install the BLF paper label check the Quick Installation guide. |
Programmable Keys 🡪 Virtual Multi-Purpose Keys | |
Mode | Allows the user to configure VPKs with modes such as Shared line, BLF and Speed Dial. Modes:
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Account | Select the account to be associated with the configured MPK. |
Value | Enter the value to be associated with the configured MPK. (Extension Number, Multicast address...) |
Label | Enter the label of the configured MPK. |
Locked | Choose whether to lock a specific Virtual multi-purpose Key or not. |
Programmable Keys 🡪 Idle Screen Softkeys | |
Custom Idle Screen Softkey Layout | Enables/disables softkey layout.Default is disabled |
Custom Softkey | Press on the Add Custom Softkey radio button to add/configure up to 3 custom softkeys. Supported key modes are:
You can also specify the Account, label, and Value (User ID) |
Custom Softkey Layout | The softkeys listed under the "Enabled" tab are displayed on the phone's idle screen. Select the softkey from the "Available" list to enable it. Up to 6 softkeys can be selected. |