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

Protocols/Standards

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

Network Interfaces

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

Graphic Display

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

Bluetooth

Yes, Bluetooth integrated

Wi-Fi

Yes, dual-band

Feature Keys

10 line keys with up to 5 SIP accounts, 40 speed-dial/BLF extension keys with dual-color LED, 5 programmable contexts sensitive Softkeys, 5 navigation/menu keys, 9 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME+, VOLUME-

Voice Codec

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

Auxiliary Ports

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

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

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

Multi-language

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

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included: Input:100-240V; Output: +12V, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Physical

Dimensions : 243mm x 210mm x 82.3mm

Unit weight:970g

Package weight:1480g

Temperature and Humidity

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

Package Content

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

Compliance

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

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

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

Protocols/Standards

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

Network Interfaces

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

Graphic Display

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

Bluetooth

Yes, Bluetooth integrated

Wi-Fi

Yes, dual-band

Feature Keys

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

Voice Codec

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

Auxiliary Ports

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

Telephony Features

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

HD audio

Yes, both on handset and full-duplex handsfree speakerphone

Base Stand

Yes, allow 2 angle positions

Wall Mountable

Yes

QoS

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

Security

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

Multi-language

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

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included: Input:100-240V; Output: +12V, 0.5A;

Integrated Power-over-Ethernet (802.3af)

Temperature and Humidity

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

Package Content

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

Compliance

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

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

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

GRP261x/GRP2624/GRP263x/GRP2670/GRP2650

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

PPPoE Link Up

PPPoE connection status.

NAT Type

The type of NAT connection used by the phone.

NAT Traversal

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

Status 🡪 System Info

Product Model

Product model of the phone.

Part Number

Product part number.

Serial Number

Displays the Serial Number of the unit.

Certificate Type

Displays the ceritificate type used for encryption

Software Version

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 current system time zone

Service Status

GUI and Phone service status.

System Information

Download system information

User Space

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

Core Dump

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

Screenshot

Download captured screenshots.

Press “Start” button to clear screenshots.

Special Feature

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

Status 🡪 Call Status

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

Status 🡪 Programmable Keys Status 🡪 Multi-Purpose Keys Status

MPKs Status

VPKs Status

Status 🡪 Programmable Keys Status 🡪 Softkeys Status 

Softkeys

Status 🡪 Extension Boards Status (GRP2615 & GRP2650 & GRP2670 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

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

Backlight Brightness: Idle

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

Active Backlight Timeout

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

Blank Screen Timeout

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

Enable Missed Call Backlight

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

Enable IEEE 802.3az EEE (Energy Efficient Ethernet)

Displays whether to enable IEEE 802.3az Energy Efficient Ethernet.

Enable Live Keypad

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

Account x 🡪 General Settings

Account Register

Account Active

Indicates whether the account is active.

The default setting is “No”.

Account Name

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

SIP Server

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

Secondary SIP Server

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

Tertiary SIP Server

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

Outbound Proxy

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

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

Secondary Outbound Proxy

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

SIP User ID

User account information, provided by your VoIP service provider.

SIP Authentication ID

SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from the SIP User ID.

SIP Authentication Password

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

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

Name

The SIP server subscriber’s name (optional) that will be used for Caller ID display (e.g., John Doe).

Tel URI

If the phone has an assigned PSTN telephone number, this field should be set to "user=phone". A "user=phone" parameter will be attached to the Request-URI and "To" header in the SIP request to indicate the E.164 number. If set to "Enable", "tel:" will be used instead of "sip:" in the SIP request.

Voice Mail Access Number

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

Monitored Voicemail Access Number

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

BLF Server

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

Picture

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

Account Display

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

Network Settings

DNS Mode

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

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

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

The Phone will try to use “Primary IP” first. After 3 tries without any response, it will switch to “Backup IP x”, and then it will switch back to “Primary IP” after 3 re-tries.

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

Max Number Of Sip Request Retries

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

DNS SRV Failover Mode

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

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.

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

Failback Expiration (m)

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

Register Before DNS SRV Failover
 

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

Primary IP

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

Backup IP 1

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

Backup IP 2

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

NAT Traversal

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

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

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

Support Rport (RFC 3581)

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

Proxy-Require

A SIP Extension to notify the SIP server that the phone is behind a NAT/Firewall.

Use SBC

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

Account x 🡪 SIP Settings

Basic Settings

SIP Registration

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

UNREGISTER on Reboot

Allows the SIP user’s registration information to be cleared when the phone reboots. The SIP REGISTER message will contain “Expires: 0” to unbind the connection. Three options are available: The default setting is “No”.

REGISTER Expiration

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

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

SUBSCRIBE Expiration

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

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

Re-Register before Expiration

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

Registration Retry Wait Time

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

SIP SUBSCRIBE Retry Wait Time

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

Add Auth Header on Initial REGISTER

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

Enable OPTIONS Keep-Alive

Enable OPTIONS Keep Alive to check SIP Server.

Default is “No”.

OPTIONS Keep-Alive Interval

Time interval for OPTIONS Keep Alive feature in seconds.

Default is “30” seconds.

OPTIONS Keep Alive Max Tries

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

Enable TCP Keep-Alive

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

Subscribe for MWI

When set to “Yes”, a SUBSCRIBE for Message Waiting Indication will be sent periodically. The phone supports synchronized and non-synchronized MWI. The default setting is “No”.

Subscribe for Registration

When set to “Yes”, a SUBSCRIBE for Registration will be sent out periodically. The default setting is “No”.

Use Privacy Header

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

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 present in the SIP INVITE message. The default setting is "No".

Use X-switch-info Header

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

Use MAC Header

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 1024 to 65400.

SIP URI Scheme When Using TLS

Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is “sips”.

Use Actual Ephemeral Port in Contact with TCP/TLS

This option is used to control the port information in the Via header and Contact header. If set to No, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the connection

The default setting is “No”.

Support SIP Instance ID

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

SIP T1 Timeout

SIP T1 Timeout is an estimate of the round-trip time of transactions between a client and server. If no response is received the timeout is increased, and request re-transmit retries would continue until a maximum amount of time define by T2. The default setting is 0.5 seconds.

SIP T2 Timeout

SIP T2 Timeout is the maximum retransmit time of any SIP request messages (excluding the INVITE message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. Default is 4 seconds.

Outbound Proxy Mode

The Outbound proxy mode is placed in the route header when sending SIP messages, or they can be always sent to outbound proxy.

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

The default setting is “97”.

G.726-32 Packing Mode

The default setting is “ITU”.

OPUS Payload Type

Default value is 123.

Send DTMF

DTMF Delay

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.

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

RTP Settings

SRTP Mode

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.
Set to RTCP-XR by Default.

RTCP Mode

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

Collector Address Selection

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

VQ RTCP-XR Collector Name

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

VQ RTCP-XR Collector Address

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

VQ RTCP-XR Collector Port

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

Symmetric RTP

Defines whether symmetric RTP is supported or not.

Default setting is “No”.

Account x 🡪 Call Settings

General

Key As Send

Defines the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the digits will be sent out. The default value is 4 seconds.
The default setting is “Pound (#)”.

No Key Entry Timeout

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

Send Anonymous

If set to “Yes”, the “From” header in outgoing INVITE messages will be set to anonymous, blocking the Caller ID to be displayed.

Default is “No”.

Anonymous Call Rejection

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

Enable Call Waiting

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

RFC2543 Hold

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

Ring Timeout

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

Call Log

Configures Call Log setting on the phone.

The default setting is “Log All Calls”.

Auto Answer

Auto Answer

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

Auto answer numbers

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

For “Auto Answer Numbers”, it accepts:

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

1x;2xxx;3x+

Intercom

Play warning tone for Auto Answer Intercom

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

Custom Alert-Info for Auto Answer

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

Allow Auto Answer by Call-Info/Alert-Info
 

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

Allow Barging by Call-Info/Alert-Info

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

Mute on answer Intercom call

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

Transfer

Transfer on Conference Hang-up

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

Enable Recovery on Blind Transfer

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

Notes:

Blind Transfer Wait Timeout

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

Refer-To Use Target Contact

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

Hide Dialing Password

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

Early Dial

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

The default setting is “No”.

Call Forward

Enable Forward All

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

All To

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

Enable Busy Forward

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

Busy To

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

Enable No Answer Forward

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

No Answer To

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

No Answer Timeout (s)

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

Dial Plan

Dial Plan Prefix

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

Bypass Dial Plan

Enable/Disable the dial plan bypass while dialing through:

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

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

Call Display

Caller ID Display

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

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.
You can submit up to 10 trusted domain names for TLS certificate verification.

Validate Certificate Chain

Validate certification chain when TCP/TLS is configured.

Default setting is “No”.

Validate Incoming Messages

Choose whether the incoming messages will be validated or not.

The default setting is “No”.

Omit charset=UTF-8 in MESSAGE

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

The default setting is “Disabled”.

Allow Unsolicited REFER

Allow Unsolicited REFER to accomplish an outgoing call.

The default setting is “Disabled”.

Accept Incoming SIP from Proxy Only

When set to “Yes”, the SIP address of the Request URL in the incoming SIP message will be checked. If it doesn’t match the SIP server address of the account, the call will be rejected.

The default setting is “No”.

Check SIP User ID for Incoming INVITE

If set to “Yes”, SIP User ID will be checked in the Request URI of the incoming INVITE. If it doesn’t match the phone’s SIP User ID, the call will be rejected.

The default setting is “No”.

Allow SIP Reset

This is used to perform a factory reset through SIP NOTIFY. When the phone receives the NOTIFY with Event: reset, the phone should perform a factory reset after the authentication. The default setting is “No”.

Authenticate Incoming INVITE

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

Default setting is “No”.

MOH

On Hold Reminder Tone

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

Music On Hold URI

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

Advanced Features

Special Feature

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

Feature Key Synchronization

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

Conference URI

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

Broadsoft Call Center

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

Please note that “Feature Key Synchronization” will be enabled regardless of this setting. Default setting is “Disabled”.

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

Hoteling Event

Broadsoft Hoteling event feature. Default setting is “Disabled”. With “Hoteling Event” enabled, user can access the Hoteling feature option by pressing the “BSCCenter” softkey.

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

Call Center Status

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

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

Broadsoft Executive Assistant

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.