GRP260x Series - Administration Guide

GRP2601

GRP2601P

GRP2602

GRP2602P

GRP2602W

GRP2602G

GRP2603

GRP2603P

GRP2604

GRP2604P

Protocols/Standards

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

Network Interfaces

Dual switched auto-sensing 10/100 Mbps Ethernet ports, integrated PoE (GRP2601P only)

Graphic Display

132 x 48 (2.21’’) LCD display

Feature Keys

4 XML programmable context sensitive soft keys, 5 (navigation, menu) keys. 8 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL

Voice Codec

RJ9 headset jack (allowing EHS with Plantronics & Jabra & Sennheiser headsets)

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra & Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 3-way conference, call park, call pickup, downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 800 records), 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
 

Base Stand

Yes, 1 angle positions available

Wall Mountable 

Yes, (*wall mount sold separately)

QoS

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

Security

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

Multi-language

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

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA;

PoE: IEEE802.3af Class 1, 3.84W; IEEE802.3az (EEE) (GRP2601P Only)

Physical

Dimension: 208mm (L) x 180mm (W) x 63.4mm (H) (with handset)

Unit weight:650g; Package weight:810g (860g for GRP2601)

Temperature and Humidity

Operation: 0°C to 40°C

Storage: -10°C to 60°C

Humidity: 10% to 90% Non-condensing

Package Content

GRP2601/2601P phone, handset with cord, base stand, universal power supply (GRP2601 only), network cable, Quick Installation Guide

Compliance

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

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

RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004;

IC: ICES-003; CS-03;

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE,

LLDP, LDAP, TR-069, SNMP, 802.1x, TLS, SRTP, IPV6

Network Interfaces

Dual switched auto-sensing 10/100 Mbps (1000 Mbps for GRP2602G) Ethernet ports, integrated PoE (GRP2602P/GRP2602G only)

Graphic Display

132 x 48 (2.21’’) backlit graphical LCD display

Wi-Fi

Yes, Dual band support (GRP2602W only)

Feature Keys

2 SIP account keys with dual-color LED, 4 XML programmable context sensitive soft keys, 5 (navigation, menu) keys. 8 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-

Voice Codec

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

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 3-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 800 records), 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

Base Stand

Yes, 1 angle positions available

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

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

Multi-language

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA;

PoE: IEEE802.3af Class 1, 3.84W; IEEE802.3az (EEE) (GRP2602P Only)

Physical

Dimension: 208mm (L) x 180mm (W) x 63.4mm (H) (with handset)

Unit weight: 670g; Package weight:830g (880g for GRP2602)

Temperature and Humidity

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

Package Content

GRP2602 phone, handset with cord, base stand, universal power supply (GRP2602/GRP2602W only), network cable, Quick Installation Guide

Compliance

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

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

RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004;

IC: ICES-003; CS-03;

Protocols/Standards

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

Network Interfaces

Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports, integrated PoE (GRP2603P only)

Graphic Display

132 x 64 backlit graphical LCD display

Feature Keys

3 SIP account keys with dual-color LED, 4 XML programmable context sensitive soft 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 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), VAD, AEC, CNG, PLC, AGC

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 4-way conference, call park, call pickup, shared-call-appearance(SCA)/bridged-line-appearance(BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 800 records), 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

Base Stand

Yes, 2 angle positions available

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

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

Multi-language

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA;PoE: IEEE802.3af Class 1, 3.84W; IEEE802.3az (EEE) (GRP2603P Only)

Physical

Dimension: 214mm (L) x 206mm (W) x 68mm (H) (with handset)

Unit weight: 780g; Package weight: 1090g for GRP2603P &1140g for GRP2603

Temperature and Humidity

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

Package Content

GRP2603 phone, handset with cord, base stand, universal power supply (GRP2603 only), network cable, Quick Installation Guide

Compliance

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

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

RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004;

IC: ICES-003; CS-03;

Protocols/Standards

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

Network Interfaces

Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports, integrated PoE (GRP2603P only)

Graphic Display

132 x 64 backlit graphical LCD display

Feature Keys

3 SIP account keys with dual-color LED, 4 XML programmable context sensitive soft 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 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), VAD, AEC, CNG, PLC, AGC

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra & Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 4-way conference, call park, call pickup, shared-call-appearance(SCA)/bridged-line-appearance(BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 800 records), 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

Base Stand

Yes, 2 angle positions available

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

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

Multi-language

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA;PoE: IEEE802.3af Class 2, 3.84W-6.49W; IEEE802.3az (EEE) (GRP2604P Only)

Physical

Dimension: 208mm (L) x 180mm (W) x 63.4mm (H) (with handset)

Unit weight: 670g; Package weight:830g (880g for GRP2602)

Temperature and Humidity

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

Package Content

GRP2604 phone, handset with cord, base stand, universal power supply (GRP2604 only), network cable, Quick Installation Guide

Compliance

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

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

RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004;

IC: ICES-003; CS-03;

Status

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

Settings

Settings sub menu contains the following options:

- Account Settings: Enable/Disable SIP account, Configures Account Name, SIP server’s address, SIP User ID, SIP Auth ID, SIP Password, Outbound Proxy, and Voice Mail Access Number.

- Call Settings: Enable/Disable DND, Enable Disable Auto Answer for SIP account, Enable/Disable Call Forward (Forward All/Busy/No Answer).

- Basic Settings: 

- Advanced Settings:

- Ethernet Settings:

- Wi-Fi Settings (GRP2602W only):

Messages

Call History

Displays Local Call Logs: “All” Calls / “Missed” Calls/ “Dialed” Calls/ “Answered” Calls.

Contacts

Contacts sub menu includes the following options:

User could configure phonebooks/groups options here, download phonebook XML to the phone and search and dial from the local phonebook and search and dial from LDAP phonebook.

Factory Functions

Factory Functions sub menu includes the following options:

UCM Detect

Configures the IP address and port of the UCM on which the IP phone can be registered.

Factory reset

Performs a Factory reset on the phone.

Reboot

Reboots the phone.

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

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.

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

System Up Time

System up time since the last reboot.

System Time

Current system time on the phone system.

System Time-Zone

Displays the time zone that is configured by user

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.

Special Feature

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

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

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.

BLF Server

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

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.

UCM User Password

Input UCM user login password to connect UCM user settings.

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 Primary IP, Backup IP 1 and Backup IP 2.

Max Number Of Sip Request Retries

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

DNS SRV Failover Mode

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

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 registered messages to the previously registered IP first, but this behavior will persist until the registered server does not respond.

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

Failback Expiration (m)

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

Register Before DNS SRV Failover

Configures whether to send REGISTER requests to the failover SIP server or Outbound Proxy before sending INVITE requests in the event of a DNS SRV failover.

Primary IP

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

Backup IP 1

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

Backup IP 2

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

NAT Traversal

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

If set to “STUN” and STUN server is configured, the phone will route according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all the SIP&SDP messages.

The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be “Keep-alive”. Configure this to be “No” if an outbound proxy is used. “STUN” cannot be used if the detected NAT is symmetric NAT. Set this to “VPN” if OpenVPN is used.

Support rport (RFC3581)

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

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

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.

Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600. The default value is 20 seconds.

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

Default is “Disabled”.

Enable OPTIONS Keep Alive

Configures whether to enable SIP OPTIONS to track account registration status. If enabled, the phone will send periodic OPTIONS messages to server to track the connection status with the server.

Default is “Disabled”.

OPTIONS Keep Alive Interval

Configures the time interval the phone sends OPTIONS message to the server. If set to 30 seconds, it means the phone will send an OPTIONS message to the server every 30 seconds.

OPTIONS Keep Alive Max Lost

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.

SUBSCRIBE for MWI

When set to “Yes”, a SUBSCRIBE for Message Waiting Indication will be sent periodically.

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

Configures whether the “Privacy Header” is present in the SIP INVITE message.

The default setting is “default”.

Use P-Preferred- Identity Header

Configures whether the “P-Preferred-Identity Header” is present in the SIP INVITE message.

Use X-Grandstream-PBX Header

Configures to use X-Grandstream-PBX header in SIP request.

Default setting is “Yes”.

Use P-Access-Network-Info Header

Configures to use P-Access-Network-Info header in SIP request.

Default setting is “Yes”.

Use P-Emergency-Info Header

Configures to use P-Emergency-Info header in SIP request.

Default setting is “Yes”.

Use P-Asserted-Identity Header

Configure whether the "P-Asserted-Identity Header" is present in the SIP REGISTER message.

Use P-Early-Media Header

Configure if the "P-Early-Media Header" support is enabled.

Use Zoom E911 X-switch-info SIP Header

Configure whether the "Zoom E911 X-switch-info SIP Header" is present in the SIP REGISTER message.

Use MAC Header

The default setting is “No”.

Add MAC in User-Agent

The default setting is “No”.

SIP Transport

Selects the network protocol used for the SIP transport.

The default setting is “UDP”.

Enable TCP Keep-alive

Configures whether to enable TCP Keep-alive for the TCP connection between the terminal and the SIP server.

SIP Listening Mode

Configures whether or not to listen to multiple SIP protocols.

The default setting is “Transport Only”.

Local SIP Port

Configures the local SIP port used to listen and transmit.

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

Configures whether the actual ephemeral port in contact with TCP/TLS will be used when TLS/TCP is selected for SIP Transport.

The default setting is “No”.

Support SIP Instance ID

Configures whether SIP Instance ID is supported or not.

The 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

Configures whether to put the Outbound Proxy in the Route header, or if SIP messages should always be sent to Outbound Proxy.

Default is “in route”.

Enable 100rel

When enabled, the 100rel tag is appended to the value of the Supported 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).

Session Timer

Enable Session Timer

Configures whether to enable session timer function. It 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. If set to “Yes”, the phone will use the related parameters when sending session timer according to “Session Expiration”. If set to “No”, session timer will be disabled.

The default setting is “No”.

Session Expiration

Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand.

The default setting is 180. The valid range is from 90 to 64800.

Min-SE

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

Caller Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it makes outbound calls.

The default setting is “No”.

Callee Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it receives inbound calls.

The default setting is “No”.

Force Timer

If set to “Yes”, the phone will use the Session Timer even if the remote party does not support this feature. Otherwise, Session Timer is enabled only when the remote party supports it.

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. When set to “Omit”, the refresh object is not specified.

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

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

iLBC Frame Size

Selects iLBC packet frame size. Users can choose from 20ms and 30ms. The default setting is “30ms”.

iLBC Payload Type

Specifies iLBC payload type. Valid range is 96 to 127. Cannot be the same as Opus or DTMF 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”.

G.726-32 Dynamic Payload Type

Specifies G.726-32 payload type. Valid range is 96 to 127. Default is 127.

Opus Payload Type

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

Send DTMF

Specifies the mechanism to transmit DTMF digits. There are 3 supported modes:

Default setting is “RFC2833”.

DTMF Payload Type

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

Inbound DTMF Volume

Sets the volume size when using in-audio DTMF transmission mode. The higher the parameter value, the higher the volume value. the valid range is 0-32.
The default value is 18.

Enable Audio RED with FEC

If set to “Yes”, FEC will be enabled for audio call.

Audio FEC Payload Type

Configures audio FEC payload type. The valid range is from 96 to 126.

The default value is 121.

Audio RED Payload Type

Configures audio RED payload type. The valid range is from 96 to 126.

The default value is 124.

Silence Suppression

If set to “Yes”, when silence is detected, a small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. For codec G.723 and G.729 only.

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. It is recommended that the IS limit value of Ethernet packet is 1500 bytes or 120 kbps. 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 in the codec table or negotiate the payload type during the actual call. For example, if set to 2 and the first code is G.729, G.711 or G.726, the “ptime” value in the SDP datagram of the INVITE request is 20 ms. If the “Voice Frame/TX” setting exceeds the maximum allowed value, the phone will use and save the maximum allowed value for the selected first codec. It is recommended to use the default setting provided, and incorrect setting may affect voice quality.

The default setting is 2.

G.723 Rate

Selects encoding rate for G723 codec.

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

RTCP Mode

Configure RTCP port negotiation rules.

RTCP Keep-Alive Method

Configures the RTCP channel keep-alive packet type.

RTP Keep-Alive Method

Configures the RTP channel keep-alive packet type.

VQ RTCP-XR Collector Name

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

VQ RTCP-XR Collector Address

Configure 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.

Symmetric RTP

Configures whether Symmetric RTP is used or not. Symmetric RTP means that the UA uses the same socket/port for sending and receiving the RTP stream. The default setting is “No”.

RTP Timeout (s)

Configures the RTP timeout of the phone. If the phone does not receive the RTP packet within the specified RTP time, the call will be automatically disconnected. The default range is 0 and 6-600. If set to 0, the phone will not hang up the call automatically.

Account x 🡪 Call Settings

General

Key as Send

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

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 value is 4 seconds. The valid range is from 1 to 15.

Send Anonymous

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

Anonymous Call Rejection

If set to “Yes”, anonymous calls will be rejected.

The default setting is “No”.

Enable Call Waiting

Configures the call waiting function for this account. If set to “Default”, it will be configured according to global call waiting function. Default value is “Default”.

RFC2543 Hold

Allows users to toggle between RFC2543 hold and RFC3261 hold. RFC2543 hold (0.0.0.0) allows user to disable the hold music sent to the other side. RFC3261 (a line) will play the hold music to the other side. The default setting is “No”.

Ring Timeout

Configures the timeout (in seconds) for the phone to ring when an incoming call is not answered. Valid range is 30 to 3600. 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. Default setting is “No”.

Auto Answer Numbers

Allows the user to configure specific numbers to auto answer. If not set, all numbers will be auto answered If auto answer is enabled. Up to 10 numbers can be configured.

Intercom

Play warning tone for Auto Answer Intercom

If enabled, phone will play warning tone when auto answering Intercom.

Custom Alert-Info for Auto Answer

Used exclusively to match the contents of the Alert-Info header for auto answer. The default auto answer headers will not be matched if this is defined.

Allow Auto Answer by Call-Info/Alert-Info

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP Call-Info/Alert-Info header sent from the server/proxy.

Default is “Yes”.

Allow Barging by Call-Info/Alert-Info

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

Mute on Intercom Answer

If enabled, the phone will mute the mirophone after answer an intercom call via Call-Info/Alert-Info.

Record

Record Key Default Fuction

Configures whether to turn the recording function on or off when the "Record" key is pressed for the first time in a call using this account. For example, the SIP server can be configured with automatic call recording. In this case, Record key default function should be configured as "Record off".

Call Recording On

Configures the DTMF sequence sent when pressing the Record key during a call on this account. Toggles between this value and the off code if possible; otherwise always sends this code.

Call Recording Off

Configures the DTMF sequence sent when pressing the Recording key during a call on this account when turning recording off.

Transfer 

Transfer on Conference Hangup

Configures whether the call is transferred to the other party if the conference initiator hangs up.

The default setting is “No”.

Enable Recovery on Blind Transfer

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

Blind Transfer Wait Timeout

Configures the timeout (in seconds) when waiting for sipfrag response in blind transfer. Valid range is 30 to 300. Default setting is “30”.

Refer-To Use Target Contact

If set to “Yes”, the “Refer-To” header uses the transferred target’s Contact header information for attended transfer.

Call Forward

Enable Forward All

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

All To

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

Enable Busy Forward

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

Busy To

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

Enable No Answer Forward

If set to "Yes", call will be forwarded to the number specified below on no answer. Disabled by Default

No Answer To

Specifies the number to be forwarded to for Call Forward On No Answer.

No Answer Timeout (s)

Defines the timeout (in seconds) before the call is forwarded on no answer.

Enable Override Forward

If enabled, the local call forward is disabled when an incoming call comes in from the configured override forward number. Disabled by Default

Override Forward Numbers

Configures the number to override the local forward function. The Max number is 10.

Dial plan

Dial Plan Prefix

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

Bypass Dial Plan

Bypass the dial plan when dialing from one of the available items:

Dial Plan

Configures the dial plan rule. For syntax and examples, please refer to user manual for more details.

Dial Plan Rules:

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.

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

Note: In some cases, where the user wishes to dial strings such as *123 to activate voice mail or other applications provided by their service provider, the * should be predefined inside the dial plan feature. An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers.

Max length of dial plan is up to 1024 characters.
 

Call Display

Caller ID Display

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

Callee ID Display

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

Ringtone

Ringback Tone at No Early Media

Play ringback tone when there is no receiving early media RTP packets. Disabled by Default

Account RingTone

Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu.

Note: User can also choose silent ring tone or doorbell.

Ignore Alert-Info header

Configures to play default ringtone by ignoring Alert-Info header.

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  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

Configures whether the domain certificates will be checked when TLS/TCP is used for SIP Transport. The default setting is “No”.

Trusted Domain Name List

Please enter a list of trusted domain names used exclusively for domain verification in TLS to obtain certificates. If a domain in the list matches any item in the trusted domain name list, the certificate will be trusted. The default trusted domains include the remote proxy domain name and the SIP server domain name. You can enter alphanumeric characters, hyphens, periods, and asterisks. Wildcard domain names like "*.grandstream.com" are supported, and any domain ending with".grandstream.com" will also be trusted.
The Maximum number of domains you can add is 10.

Validate Certificate Chain

Validate certification chain when TCP/TLS is configured.

The default setting is “No”.

Validate Incoming SIP Messages

Specifies if the phone will check the incoming SIP messages Caller ID and CSeq headers. If the message does not include the headers, it will be rejected. The default setting is “No”.

Omit charset=UTF-8 in MESSAGE

Omit charset=UTF-8 in MESSAGE content-type

Allow Unsolicited REFER

Configures whether to dial the number carried by Refer-to header after receiving out-of-dialog SIP REFER request actively.

If set to “Disabled“, the phone will send error warning and stop dialing.

If set to “Enabled/Force Auth“, the phone will dial the number after sending authentication. If the authentication fails, it will stop dialing.

If set to “Enabled“, the phone will dial all numbers carried by SIP REFER.

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 does not 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 does not match the phone’s SIP User ID, the call will be rejected. The default setting is “No”.

Allow SIP Reset

Allow SIP Notification message to perform factory reset.

The default setting is “No”.

Authenticate Incoming INVITE

If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response.

The default setting is “No”.

MOH

On Hold Reminder Tone

Configures to play reminder tone when the call is on hold.

Music On Hold URI

Music On Hold URI to call when a call is on hold if server supports it.

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, Zoom, 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 is enabled, DND, Call Forward features and Call Center Agent status can be synchronized between Broadsoft server and phone. Default is “Disabled”.

Conference URI

Configures the conference URI when using Broadsoft N-way calling feature.

BLF (Busy Lamp Field) – GRP2604 only

Presence Eventlist URI

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

Eventlist BLF URI

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

Auto provision Eventlist

Select the type of Eventlist to get automatically provisioned onto available MPKs. Whether its BLF Eventlist or Presence Eventilist.

BLF Call-pickup

Configures BLF Call-pickup method:

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

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

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

BLF Call-pickup Prefix

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

Call Pickup Barge-In Code

Configures feature access code of Call Pickup with Barge-in feature.

PUBLISH for Presence

Enables presence feature on the phone.

The default setting is “No”.

SCA

Line-Seize Timeout

Configures the interval (in seconds) when the line-seize is considered timed out when Shared Call Appearance feature is used. Valid range is 15 to 60.

Call Park

Call Park Feature Code

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

Call Park Retrieve Feature Code

Configure the feature access code for retrieving the parked call on accounts or BLF keys.

Account x 🡪 Dial Plan

Name

Enter the name for the configured rules.

Rule

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

Type

Choose the type of the rule:

Account x 🡪 Hidden Number Plan

Hidden Number Feature

If set to “Yes”, incoming and outgoing number display will be handled according to hidden number plan.

Hidden Number Plan List

Currently, incoming and outgoing number display will be handled according to hidden number plan. Users are able to configure the hidden number rules, matching their syntax rules from top to bottom as follows:

Hidden Number Plan Test

Enter the number to test the hidden rules of the current page. After confirming, please save and apply it.​

Hidden Number Plan

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.

Note: If the device is registered with Broadsoft account, it does not 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

The extension the call will 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 No Answer feature code to deactivate no answer call forwarding.

Target

The extension the call will be forwarded to.

Call Forward No Answer Timeout (s)

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

Accounts 🡪 Account Swap

Swap Account Settings

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

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

Phone Settings 🡪 General settings

Basic Settings

Local RTP Port

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

Local RTP Port Range

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

Use Random Port

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

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

Keep-alive Interval

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

STUN Server

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

Only non-symmetric NAT routers work with STUN.

Use NAT IP

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

Delay Registration

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

Enable Outbound Notification

Configures whether to enable outbound notifications such as Action URL.

Clean User Data While Different Users Log In

When enabled, if the current login account is different with last one, Device under tesr will delete the contact and call history of the last account.

Public Mode

Enable Public Mode

Configures to turn on/off public mode for hot desking feature.

Public Mode Username Prefix

Configures the prefix of the username for public mode login.

Public Mode Username Suffix

Configures the suffix of the username for public mode login.

Allow Multiple Accounts

Configures remaining account with a SIP account and be able to make/receive calls on the configured account even if the user is not logged in to the public mode.

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

Enable Remote Synchronization

If enabled, the phone will automatically download the current account’s settings from the remote server and upload it to the server.

Server Type

Configures the Server type and protocol used to send requests to the Remote Synchronization Server.

This can be set to TFTP, FTP, HTTP

Server Path

Configures the server path for the Remote Synchronization feature.

FTP/HTTP Username

The username for FTP/HTTP server.

FTP/HTTP Password

The password for FTP/HTTP server.

Public Mode Timeout

Sets the public mode timeout (min), the time starts from the last idle state of the phone.
The Valid Range is 10-1440min 

The Default value is 10min

Pop-up Hidden Settings

Hide Pop-up Options

Sets options for hiding pop-ups in LCD screen, the available options are:

Disabled by Default.

Phone Settings 🡪 Call Settings

General

Key Mode

Account Mode

In Calling State, each key displays the call status of the corresponding account. Click to switch to the first line under this account or select the account to initiate a new call.

Line Mode

In Calling State, each key controls a line, and the call line can be switched by pressing the key.

Preferred Default Account 

Select the preferred default account for on-hook or off-hook dialing. When the selected account is unavailable, system will use the first available account to dial out.

Mute Key Functions While Idle

Select "Mute" key function while the phone is idle.

Last Call Froward Always

If enabled, the number put into the ForwardAll feature will be stored the next time you use the ForwardAll softkey.
Note: ForwardAll softkey currently only used for account 1. 

Show SIP Error Response

Configures to enable SIP error response information displayed on LCD screen.

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 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.

Enable Speaker Key

Sets whether to enable the speaker key.

In-call Contact Info Display

Configure the contact information displayed during a call, up to 5 lines to display. The following labels and their combinations can be entered:
● {number}
● {name}
● {title}
● {company}
● {department}
● {email}
● {firstName}
● {lastName}

Instance：
Line 1：{name}
Line 2：{title}-{company}
Line 3：{number} / {email}

After the configuration, the screen will scroll up and down according to the sequence.

You can configure up to three contact info displays.

Contact Source Priority

Configure the priority if the ID source displayed on the phone when incming/outgoing calls. Select on ID source and click Up/Down arrow on the right to adjust the order.
Note: If the "Caller ID Display" under the account is configured as "Disabled", the caller number cannot be obtained, the phone will only display "Unavailable".

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”.

Enable Direct IP Call

Enables Direct IP Call feature. 

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 of the IP address.

Enable Paging Call Mode

Configures enable/disable paging call mode of the phone. If set to "Yes", the feature of paging call will be enabled. Default is Yes.

Predictive Dialing Feature

Configures the predictive dialing feature on the call screen.

Predictive Dialing Source

Predictive dialing feature will sequentially search the number based on the selected sources.

Onhook Dial Barging

When the option is set to "Disabled", onhook dialing won't be interrupted by an incoming call.

Off-hook Auto Dial

Configures the digits to be dialed via the first account when the phone is off-hook.

Off-hook Auto Dial Delay

Configures the digits to be dialed via the first account when the phone is off-hook.

Off-hook/On-hook Timeout (s)

If configured, when the phone is in the off-hook or on-hook dialing state, it will go idle after the timeout (in seconds). Valid range is 10 to 60.

Enable Live Keypad

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

Live Keypad Expiration

Configures the expiration time for live keypad. Interval is between 2s and 15s. Default value is 5s.

Enable Auto Redial

Configures to redial automatically at a later time when the dialed number is currently busy.

Auto Redial Times

Configures the total times to redial if "Auto Redial" is enabled.

Auto Redial Interval

Configures the interval between each redial if "Auto Redial" is enabled.

Bypass Dial Plan Through Call History and Directories

Configures hether to check dial plan when dialing from call history and phonebook directories.

Enable Call Completion Service

If enabled, phone will automatically redial the previous failed call when the remote party becomes available.

Incoming

Enable Incoming Call Popup

If set to "Yes", phone will pop up an incoming call window to notify the call.

Enable Missed Call Notification

If set to "Yes", phone will show a prompt about the missed call information.

Return Code When Refusing Incoming Call

Configures the return code that phone will send to the call when it refuses an incoming call.

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

Disables the call waiting feature. The 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

Configures the delay for automatically answering the incoming call. Valid range is 0 to 10 (seconds)

In Call

Enable In-call DTMF Display

When set to "No", the DTMF digits entered during a call will not be displayed on LCD.
Enabled by Default.

Enable Sending DTMF via specific MPKs

Allows certain MPKs to send DTMF in-call. This option does not affect Dial DTMF.
Disabled by Default

Show on Hold Duration

Shows the duration of holding a call on the LCD
Enabled by Default.

Enable Auto Unmute

If the option is enabled, automatically unmute the phone when a user unholds the call or establishes a new call.
Enabled by Default

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

Configures the phone to Play busy tone when call is disconnected remotely.
Enabled by Default.

Enable Mute Key In Call

When set to "No", the mute key will not work while on call.
Enabled by Default.

Phonebook Matching

Enable Phonebook Matching Rules

Enable phonebook matching rules. Multiple options are supported, you can select the following below options:

This function is disabled by default.

Phonebook Matching Rules

Configure rules for processing numbers. Numbers 4,7, and 9 digits can be matched from top to bottom.
Expression
Matching rule/Replacement rule
Instance
● 9XXXXXXXX: matches a nine-digit number starting with 9 and is not modified
For example: 9XXXXXXXX: 958242234 - >; 958242234
● (6XXXXXX)/95

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.

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.

PPPoE Account ID

Enter the PPPoE account ID.

PPPoE Password

Enter the PPPoE Password.

PPPoE Service Name

Enter the PPPoE Service Name.

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.

802.1X Client Certificate

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

Network Settings 🡪 Wi-Fi Settings (GRP2602W Only)

Wi-Fi Function

Enables / Disables the Wi-Fi on the phone. Three options are available:

Wi-Fi Band

Set the type of Wi-Fi Band whether its 2G or 5G or 5G&2G.

Country Code

Configures Wi-Fi country code.

ESSID

This parameter sets the ESSID for the Wireless network. Press “Scan” to scan for the available wireless network. Click on “Connect” and enter the authentication credentials of the Wi-Fi network to connect to. Users can connect to hidden networks by pressing on “Add Network” and configure:

Network Settings 🡪 OpenVPN® Settings

OpenVPN® Enable

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

Import OpenVPN® Configuration

Imports the configuration file from the current computer. After importing, the local configuration will be overwritten and OpenVPN® function is automatically enabled.

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, TCP, UDP IPV4 Only, TCP IPV4 Only, UDP IPV6 Only, TCP IPV6 Only

The default value is “UDP”.

OpenVPN® CA

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

OpenVPN® Certificate

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

OpenVPN® Client Key

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

OpenVPN® Client Key Password

Allows user to set password for client.key file

OpenVPN® TLS Key

Uploads the OpenVPN® TLS .key file

OpenVPN® TLS Key Type

Selects the encryption type of the OpenVPN® TLS key. it can be set to : TLS-Auth, TLS-Crypt, TLS-Crypt V2

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 SHA256

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

Network Settings 🡪 Advanced Settings

Advanced Network Settings

DNS Refresh Timer (m)

Configures the refresh time (in minutes) for DNS query. If set to "0", the phone will use the DNS query TTL from DNS server response.
the Default value is "0"

DNS Failure Cache Duration (m)

Configures the duration (in minutes) of the previous DNS cache when the DNS query fails. If set to "0", the feature will be disabled. Note: Only valid for SIP registration.
The Default value is "0"

Enable LLDP

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

LLDP TX Interval

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

Enable CDP

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

Layer 3 QoS for SIP

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

Layer 3 QoS for RTP

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

Enable DHCP VLAN

Enables auto configure for VLAN settings through DHCP. Disabled by default.

Enable Manual VLAN Configuration

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

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets. The valid range is 0 – 4094.The default value is 0.

Layer 2 QoS 802.1p Priority Value

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

Maximum Transmission Unit (MTU)

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

The default value is 1500 bytes.

PC Port Mode

PC Port Mode

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

PC Port VLAN Tag

Assigns the VLAN Tag of the PC port. The valid range is 0 – 4094.

The default value is 0.

PC Port Priority Value

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

The default value is 0.

Proxy

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.

Remote Control

Action URI Support

Indicate whether the phone is enabled to receive and handle Action URI request.

Remote Control Pop up Window Support

Indicate whether the phone is enabled to pop up Allow Remote Control window.

Action URI Allowed IP List

List of allowed IP addresses from which the phone receives the Action URI

CSTA Control

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

CTI Settings

Affinity Support

Indicate whether Affinity feature is supported.

Preferred Account

Affinity target SIP account.

Network Settings 🡪 SNMP Settings

Enable SNMP

Enable/Disable SNMP service. Default is No.

Version

Choose between (Version 1, Version 2, or Version 3).

Port

Listening Port of SNMP daemon (Default 161).

Community

Name of SNMP community.

Security Level

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

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

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

SNMP Username

Username for SNMPv3.

Authentication Protocol

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

Privacy Protocol

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

Authentication Key

Enter the Authentication Key for SNMPv3.

Privacy Key

Enter the Privacy Key for SNMPv3.

SNMP Trap Version

Choose the Trap version of the SNMP trap receiver.

SNMP Trap IP

IP address of trap destination.

SNMP Trap port

Port of Trap destination (Default 162)

SNMP Trap Interval

Time interval between traps (Default is 5).

SNMP Trap Community

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

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” “AES” or “DES”.
Set to "None" By Default.

Trap Authentication Key

Enter the Trap Authentication Key.

Trap Privacy Key

Enter the Trap Privacy Key.

Programmable keys 🡪 Multi-Purpose Keys (GRP2604 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:

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:

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.

HTTP Command:

Call Flip:

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.

Basic Settings 

Transfer Mode via MPK

Perform blind transfer, attended transfer, or a new call with the specific in the Value field when a user presses “Transfer” multiple-purpose key

Enable Transfer via Non-Transfer MPK

MPK with type BLF, Speed dial, etc, will perform as transfer MPK under active call

Programmable Keys 🡪 Virtual Multi-Purpose Keys

Mode

Allows the user to configure VPKs with modes such as Shared line, BLF and Speed Dial. Modes:

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.

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 Extension

Note: For more information on configuring Virtual extensions, please refer to the following guide: GRP260x Virtual Extension User Guide

Access IP type

To access the Virtual Extension module interface, users have three options: They can use their browser and enter either the LAN IP or mDNS domain link. Alternatively, they can scan the QR code on the webpage using their phone while connected to the same network as the IP phone. After scanning, a new window will appear, prompting them to log in with their IP phone credentials.

Download App

To download the App, simply Hover over the icon of your operating system(IOS client, or Android client) scan the QR code and the app will be installed.

Mode

Speed Dial:
Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed or enter the IP address to set the Direct IP call as Speed Dial.
Busy Lamp Field (BLF):
Select the Account to monitor the BLF status. Enter the extension number in the Value field to be monitored.
Presence Watcher:
This option has to be supported by a presence server and it is tied to the “Do Not Disturb” status of the phone’s extension.
Eventlist BLF: This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it out in one single notify message. PBX server has to support this feature.
Speed Dial via active account: Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2
Dial DTMF: Enter a series of DTMF digits in the Value field to be dialed during the call. “Enable MPK Sending DTMF” has to be set to “Yes” first.
Voicemail: Select Account and enter Voicemail access number in the Value field.
Call Return: The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.
Transfer: Select Account and enter the number in the Value field to be transferred (blind transfer) during the call.
Call Park: Select Account and enter the call park extension in the Value field to park/pick up the call.
LDAP Search: This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings. The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field.
For example:

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

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

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

 
Conference: Allow user to set their Multi-Purpose Key to “Conference” mode to trigger a conference. By setting the extension number in the value box, the users will be able to activate a 3-way conference by simply press the assigned MPK button.
Multicast Paging: Allows the user to configure the address to send a multicast page to.
Call Log:
Select Account and enter account number in the Value field to allow configuration of call log for other extension.
Monitored Call Park:
Select account from Account field and enter the call park extension in the Value field to park/pick up the call, and also monitor the parked call via Line Key’s light.
Menu: Select this feature in order to display the Menu from the MPK buttons, no field dis required for configuration.
Information: Select this feature in order to display the Information popup to show the firmware version, MAC address, IP address and IP Settings from the MPK buttons, no field dis required for configuration.
Message: Select this feature in order to display the Message menu from the MPK buttons, no field dis required for configuration.
Forward: Set the MPK Button to perform call forwarding to the destination number configured on the “Value Field”. During ringing press the button to perform the call forward.
DND:

Press the configured key to enabled/Disable DND.
Redial:

On this mode, the configured key can be used to redial numbers.
Presence Eventlist:
This option is similar to the Presence Watcher option but in this case the PBX collects the information from the phones and sends it out in one single notify message.
Note: The PBX server has to support this feature.
Provision:
Select this feature in order to make the phone trigger an instant provisioning
Opendoor:
Select this feature in order to make the phone trigger an open-door action in conjunction with a GDS37xx
Multicast Listen Address:
This feature sets up a multicast listening address for the IP Phone.
Multicast Paging Address:
This Feature sets up a Multicast paging address for paging purposes.  
Note: An MPK configuration tutorial video link can be found on the MPK configuration page.
HTTP Command:
This Feature sets up a call through an HTTP command

Call Flip:
Call Flip with MPK is a feature that allows you to transfer an active call from one VoIP phone to another or to a different number using a programmable Multi-Purpose Key, ensuring a seamless transition.

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.

Preview

Shows a preview of the configured Virtual MPks label.

Programmable Keys 🡪 Idle Screen Softkeys

Custom Idle Screen Softkey Layout 

Enables/disables softkey layout.Default is disabled 

Custom Softkey

Press on Add Custom Softkey radio button to add/configure up to 3 custom softkeys. Supported key modes are speed dial, speed dial via active account, and voicemail.
Note: The softkey icons have been updated and now have the option to be customized on preference based on the key mode , either speed Dial , speed dial via active, or voicemail.

Custom Softkey Layout

The softkeys listed under "Enabled" tab is displayed on the phone's idle screen. Select the softkey from "Available" list to enable it. Up to 6 softkeys can be selected.

Programmable Keys 🡪 Call screen softkeys

Custom Call Screen Softkey Layout 

Enablesdisables custom softkey layout Default is disabled

Enforce Softkey Layout Position

Whether to enforce the custom softkey layout position When set to 'YES', GUI will still preserve the space if the configured softket is unable to show
Disabled by Default

Custom Softkey

Press on Add Custom Softkey radio button to add/configure up to custom softkeys Supported key modes are speed dial, speed dial via active account and voicemail.

Custom Softkey Layout

Dialing state:

Ringing State:

Calling State:

Call Connected State:

On Hold State: