Configure Grandstream GRP2612W

Summary of contents

☎ This Grandstream GRP2612W Setup Guide is an Administrator’s Manual that will make it easy for you to install and program advanced features easily. 📣 Read more below!

What are the technical specifications for Configuring Grandstream GRP2612W?

The following table summarizes all the technical specifications, including supported protocols/standards, voice codecs, telephony features, languages, and upgrade/provisioning settings for Configure GRP2612W. We show all the necessary information below:

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

10/100 Mbps Ethernet ports with automatic double-switching auto-sensing, with integrated PoE

Graphic display

LCD TFT color 2.4 inch (320×240)

Function keys

4 line keys with up to 2 SIP accounts, 4 context sensitive soft keys, 5 menu/navigation keys, 9 dedicated function keys for: MESSAGE (with LED indicator ), TRANSFER, STANDBY, HEADSET, MUTE, SEND/REDIAL, SPEAKER, VOLUME+, VOLUME-

Voice codec

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

Auxiliary ports

RJ9 headset (allowing EHS with Plantronics headsets)

Telephony functions

Hold, transfer, forward, 3-way conferencing, call parking, call capture, shared call appearance (SCA), bridged line appearance (BLA), downloadable phone directory (XML, LDAP, up to 1000 items), call waiting, call log (up to 2000 records), display customization, off-hook auto dialing, auto answer, click-to-dial, plan flexible dialing, shared desktop, custom music tones and music on hold, server redundancy and failover.

HD audio

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

Base mount

Yes , it allows 2 angle positions.

Wall

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 based on MD5 and MD5-sess authentication, secure configuration file based on AES, SRTP, TLS, 802.1x media access control.

Multiple languages

English,Arabic, Chinese, Croatian, Czech, Dutch, German, French, Hebrew, Hungarian, Italian, Japanese, Korean, Polish, Portuguese, Russian, Slovenian, eSpanish, Turkish.

Updating/provisioning

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

Energy and green Energy efficiency

Universal power adapter included: Input: 100-240 VAC; Output: +5 VDC, 0.5 A; Integrated Power-over-Ethernet (802.3af).

Physics

Dimension: 203 mm x 193 mm x 52.1 mm

Unit weight: 554 g

Package weight: 936 g

Temperature and humidity

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

Package Contents

GRP2612W handset, corded handset, base stand, universal power supply, network cable, Quick Installation Guide.

Fulfillment

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 61000-3-3-3; EN 62368-1; EN 301

489-1/-17; EN 300 328; EN 301 893; EN 62311;

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

How to install Grandstream GRP2612W?

Grandstream GRP2612W can be installed on the desktop using the phone cradle or attached to the wall using the wall mount slots.

How to install Gransdtream GRP2612W on bracket?

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

How to wall mount Grandstream GRP2612W?

Fasten the wall mount spacers to the wall mount spacer slot on the back of the phone.
Fasten the phone to the wall through the wall mount hole.
Remove the tab from the handset base (see figure below).
Turn the tab and plug it back into the slot with the extension facing up to hold the handset while the phone is mounted on the wall (see figure below).

How to connect Grandstream GRP2612W?

To connect Grandstream GRP2612W, follow the steps below:

Connect the handset and the main housing of the phone with the telephone cable.
Connect the LAN port of the handset to the RJ-45 jack of a hub/switch or router (LAN side of the router) using the Ethernet cable.
Connect the power supply output plug to the phone’s power connector; connect the power adapter to a power outlet. If the PoE switch is used in step 2, this step can be skipped.
The LCD will display firmware upgrade or provisioning information. Before continuing, wait for the date/time display to appear.
Using the phone’s built-in web server or the keypad configuration menu, you can configure GRP2612W using static IP or DHCP.

How to configure GRP2612W via web browser?

The built-in GRP2612W web server responds to HTTP/HTTPS GET/POST requests. The embedded HTML pages allow the user to configure GRP2612W via a web browser such as Google Chrome, Mozilla Firefox and Microsoft’s IE. To access the web GUI:

Connect the computer to the same network as the phone.
Make sure the phone is turned on and displaying its IP address. You can check the IP address by pressing and holding the UP arrow button for 3 seconds when the phone is idle.
Open a web browser on your computer.
Enter the IP address of the phone in the browser address bar.
Enter the administrator’s username and password to access the web configuration menu.

Notes:

- The PC must be connected to the same subnet as the phone. This can easily be done by connecting the PC to the same hub or switch that the phone is connected to. In the absence of a hub/switch (or free ports on the hub/switch), connect the computer directly to the PC port on the back of the phone;

- If the phone is properly connected to a working Internet connection, the phone’s IP address will be displayed under MENU -> Status -> Red. This address has the format: xxx.xxx.xxx.xxx.xxx, where xxx represents a number from 0 to 255. Users will need this number to access the Web GUI. For example, if the phone has the IP address 192.168.40.154, enter “http://192.168.40.154” in the browser address bar;

- There are two default passwords for the login page:

User Level	User	Password	Allowed web pages
End user level	user	123	Status and Basic Settings only
Administrator Level	admin	Random password available on the sticker on the back of the unit.	Scan all pages

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

How to save web configuration changes in Grandstream GRP2612W?

After users make configuration changes, click the “Save” button to save, but not apply the changes until the “Apply” button at the top of the web GUI page is clicked. Or, users can directly click the “Save and Apply” button. We recommend rebooting or turning the phone off and on after applying all changes.

How to reboot Grandstream GRP2612W?

Press the “Reset” button in the upper right corner of the web GUI page to reboot the phone remotely. The web browser will display a reboot message. Wait approximately 1 minute to log in again.

How to configure the GRP2612W?

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

Status: shows the account status, network status and system information of the phone.
Account: To configure GRP2612W SIP account.
Settings:To configure call features, ring tone, audio control, LCD display, date and time, web services, XML applications, soft keys, etc.
Network: to configure GRP2612W network settings.
Maintenance: for configuring web access, upgrade and provisioning, syslog, language settings, TR069, security, etc.
Directory:for managing the phonebook and LDAP.

What are meanings of the status page for Configure Grandstream GRP2612W?

Status -> EAccount status

Account

Account index.

- For GRP2612W: up to 2 SIP accounts

SIP user ID

Displays the SIP user ID configured for the account.

SIP Server

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

SIP registration

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

Status -> Network Status

MAC Address

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

IP Configuration

The type of address configured: 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 gateway address obtained on the phone.

DNS Server 1

The DNS server 1 address obtained on the phone.

DNS Server 2

The DNS server 2 address obtained on the phone.

Link

PPPoE PPPoE connection status.

Type of NAT

The type of NAT connection used by the phone.

Cross NAT

Displays the NAT connection status for each account on the phone.

Status -> System Information

Product Model

Phone Model.

Part Number

Product part number

Software Version

- Startup: startup version number;
- Kernel: kernel version number;
- Base: base version number;
- Prog: program version number. This is the main firmware version number, which is always used to identify the software system of the phone;
- Regional Settings: regional settings version number;
- Recovery: recovery version number.

Geographic IP information

City: display of phone location;
Language: display language;
Time zone: time zone display;
Country code: display country code for Wi-Fi;

Special Feature

OpenVPN® support:shows whether the phone supports OpenVPN®.

System uptime

System time since last reboot.

System Time

Current system time on the phone system.

Service Status

GUI, phone and CPE service status.

System Information

Download system information.

User Space

User space Displays the percentage of user space used and the database status.

Memory dump

Displays the status of the kernel dump and the generated kernel dump files, if any.

Also provides the ability to manually generate GUI/phone core dump files.

Capture

Download screenshots.

Press the “Home” button to delete screenshots.

Status -> Softkey Status -> Virtual Multipurpose Keys

Status VPKs

Mode
Account
Description
Value

Status -> Programmable Key Status -> Virtual Multipurpose Keys

MPKs Status

Mode
Account
Description
Value

Status -> Programmable Key Status -> Programmable Keys

Programmable keys

Mode
Account
Description
Value

What are the account page definitions for Configure GRP2612W?

Account X -> General Settings

Active account

This field 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 screen.

SIP Server

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

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

Secondary SIP server

The URL or IP address and port of the SIP server. When configured, the phone will register to the primary and secondary SIP server. If the primary SIP server cannot be accessed, the phone will use the secondary SIP server for telephone services (including making and receiving calls).

Outbound proxy

IP address of theThe phone uses it for Firewall or NAT penetration in different network environments.

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

Backup proxy

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

BLF Server

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

SIP user ID

User account information provided by your VoIP service provider (ITSP). Its usually in digit form as a phone number or actually a phone number.

Verify ID

Authentication ID of the subscriber of the SIP service used for authentication. It can be identical or different from the SIP user ID.

Verify Password

The account password required for the phone to authenticate with the ITSP (SIP) before the account can be registered. After saving it, it will appear as hidden for security reasons.

Name

The name of the SIP server subscriber (optional) to be used for caller ID show.

Voicemail Access Number

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

Image

Specifies the account image to be sent to the caller/caller when making calls.

Account Display

This option allows you to configure GRP2612W how your SIP account label will be displayed on the phone screen.

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

Account X -> Dial Plan

Name

Enter the name of the configured rules.

Rule

Enter the rule configuration (numeric pattern, prefix to add, etc.).

Type

Choose the type of rule (pattern, block, mark now, prefix and second tone).

Account X -> Network Settings

DNS Mode

This setting controls how Search Appliance looks up IP addresses for host names.

There are four modes: A Record, SRV, NATPTR/SRV, Use Configured IP.

The default setting is “A Record”.

If the user wants to locate the server by DNS SRV, the user can select “SRV” or “NATPTR/SRV”.

If “Use configured IP” is selected, complete the three fields below:.

main IP
backup IP 1
backup IP 2

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

The phone will try to use “main IP” first. After 3 attempts with no response, it will switch to “Backup IP x”, then switch back to “Primary IP” after 3 retries.

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

DNS SRV DNS failover mode

The option will decide which IP will be used to send SIP packets after the IPs for the SIP server host are resolved with DNS SRV.

Default

If the option is set to “default”, it will retry sending log messages to one IP at a time and the process repeats.

One saved up to DNS TTL

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

One saved until there are no responses

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

Register before SRV failover

Indicates whether a REGISTER request will be initiated when a server failover occurs in DNS SRV mode.

The default setting is No.

IP Main

Sets the primary IP address to which the phone sends the DNS query when “Use configured IP” is selected for DNS mode.

IP Backup 1

Configures the backup IP1 address to which the phone sends the DNS query when “Use configured IP” is selected for DNS mode.

Backup IP 2

Configures the backup IP2 address to which the phone sends DNS query when “Use configured IP” is selected for DNS mode.

NAT Traversal

This parameter configures whether the NAT traversal mechanism is enabled. Users can select the mechanism from No, STUN, Keep-alive, UPnP, Auto or VPN. The default setting is “Auto”.

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

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

is symmetric NAT. Set it to “VPN” if OpenVPN is used.

Proxy Requirements

A SIP extension to notify the SIP server that the phone is behind a NAT/Firewall. Do not configure this parameter unless this feature is supported by the SIP server.

Using SBC

Specify whether or not an SBC server is used. If users wish to work under SBC associated with 3CX, they must enable this feature to have better communication with the server.

Account X -> SIP Configuration -> Basic Configuration

TEL URI

If the phone has a PSTN phone number assigned to it, this field must be set to “User = Phone”.

A “User = Phone” parameter will then be attached to the request line 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. The default setting is disabled.”

SIP registration

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

Delete Registration on Reboot

Allows you to delete the SIP user registration information when the phone is rebooted. The SIP REGISTER message will contain “Expires: 0” to unlink the connection. Three options are available: the default setting is “No”.

- If set to “All“, the SIP user registration information will be cleared when the phone is rebooted. The SIP contact header will contain “*” to notify the server to unlink the connection.
- If set to “Instance“, the SIP user will not register to the current phone only.
If set to “No“, the phone will not unregister the SIP account when rebooted.

Registration Expiry

Specifies how often (in minutes) the phone updates its log with the specified logger. The default value is 60 minutes.

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

Subscription expiration

Specifies how often (in minutes) the phone updates its subscription with the specified logger. The maximum value is 64800 (about 45 days). The default value is 60 minutes.

Reregister before expiration

Specifies the frequency of time (in seconds) at which the phone sends a re-registration request before the registration expires. The default value is 0.

Enable OPTIONS Keep Alive

Enable OPTIONS Keep Alive to check the SIP server.

The default is “Yes”.

OPTIONS Keep Alive interval

Interval of time for OPTIONS Keep Alive function in seconds.

Default value is “30” seconds.

OPTIONS Keep Alive Max Lost

Maximum number of packets lost for the OPTIONS Keep Alive feature before re-registering the phone. The default value is “3”.

Local SIP port

Defines the local SIP port used for listening and transmitting. 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 1 to 65535.

SIP registration failure Retry timeout

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

T1 SIP timeout

SIP T1 Timeout is an estimate of the round-trip time for transactions between a client and a server. If no response is received, the timeout increases and retransmission retries of the request would continue up to a maximum amount of time defined by T2. The default setting is 0.5 seconds.

T2 SIP timeout

SIP T2 Timeout is the maximum retransmission time for any SIP request message (excluding INVITE message). T1 retransmission and duplication continues until it reaches the value of T2. The default value is 4 seconds.

SIP transport

Determines the network protocol used for SIP transport. Users can choose between TCP, UDP and TLS. The default setting is “UDP”.

SIP listening mode

Determines whether or not to listen to multiple SIP protocols.

Transport only: only will listen to the configured transport protocol.
Transport only: only will listen to the configured transport protocol.
Dual: will listen for TCP when UDP is selected.
Dual (secure): will listen for TLS/TCP when UDP is selected. If TCP or TLS/TCP is selected, it will listen to UDP
Dual (BLF Enforced): will attempt to enforce BLF subscriptions to use the TCP protocol by adding ‘transport = tcp’ to the contact header.

The default setting is “Transport only”.

SIP URI scheme when using TLS

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

Use real ephemeral port in contact with TCP/TLS

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

The default setting is “No”.

Outbound Proxy Mode

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

In route
Not en route
Always send to

The default is “en route”.

supported SIP instance ID

Defines whether or not the SIP instance ID is supported. The default setting 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”.

SIGN UP for registration

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

Enable 100rel

The use of the PRACK (Provisional Acknowledgment) method enables the reliability of SIP (1xx series) provisional responses. This is very important to support the interconnection of PSTN networks. To invoke a reliable interim response, the tag 100rel is added to the required header value of the initial signaling messages. The default setting is “No”.

Caller ID display

Determine from where to place the caller ID to display or not to display on the phone.

Automatic: the phone will update the caller ID in the order of P asserted identity header, remote party ID header and To header in ring 180.
Disabled: the caller ID will be displayed as “Unavailable”.
A header: the caller ID will not be updated and will be displayed as A header.

The default setting is “Auto”.

Caller ID display

Determine from where to place the caller ID to display or not to display it on the phone .

Automatic: the phone will look for the caller ID in the order of asserted identity header P, remote party ID header and source header in the incoming SIP INVITE.
Disabled: all incoming calls are displayed with “Unavailable”.
From Header: the phone will display the caller ID based on the From Header on the incoming SIP INVITE.

The default setting is “From Header.” .

Add authorization header in the initial REGISTRATION

To define whether to add an authorization header in the initial REGISTRATION from the first REGISTRATION. The default setting is “No”.

Enable SIP reset

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

Ignore alert information header

This option is used to set the default ringtone. If set to “Yes”, the configured default ringtone will be played. The default setting is “No”.

Account X -> SIP Settings -> Custom SIP headers

Use privacy header

Controls whether the privacy header will be presented in the SIP INVITE message or not, if the header contains the caller’s information.

Default: the privacy header will be displayed in INVITE only when the special “Huawei IMS” feature is enabled.
Yes: the privacy header will always be displayed in INVITE.
No: the privacy header will not be displayed in INVITE.

The default setting is “default.” .

Use P-Preferred identity header

Controls whether the P-Preferred-Identity header will be presented in the SIP INVITE message.

Default: the P-Preferred-Identity header will be displayed in INVITE unless the special “Huawei IMS” feature is enabled.
Yes: the P-Preferred-Identity header will always be displayed in INVITE.
No: the P-Preferred-Identity header will not be displayed in INVITE.

The default setting is “default.”

Use X-GrandstreamPBX header

Enables/disables the use of the X-Grandstream-PBX header in the 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 the P-Access-Network-Info header in the SIP request. When disabled, the SIP message sent from the phone will not include the selected header. The default setting is “Yes.”

Use emergency information header P

Enables/disables the use of the P-Emergency-Info header in the SIP request. When disabled, the SIP message sent from the phone will not include the selected header. The default setting is “Yes”.

Use MAC header

If Yes only for REGISTER, the SIP message to register or deregister will contain the MAC address in the header, and all outgoing SIP messages except the REGISTER message will append the MAC address to the User-Agent header;
If Yes to all SIP, the SIP message to register or deregister will contain the MAC address in the header, and all outgoing SIP messages, including REGISTER, will append the MAC address to the User-Agent header;
If No, the MAC header will not be included in the registration or deregistration message nor will the MAC address be attached to the user agent header for any outgoing SIP messages.

The default setting is “No”.

Add MAC in User-Agent

Users can choose to add the MAC address in the User-Agent header.

If set to “No” (default), the phone will not attach the MAC address to the User-Agent header for any outgoing SIP messages.
If set to “Yes, except REGISTER”, SIP messages to register or deregister will not contain the MAC address in the User Agent header and all other outgoing SIP messages will attach the MAC address to the User Agent header.
If set to “Yes to all SIP”, all outgoing SIP messages will attach the MAC address to the User Agent header.

Account X -> SIP Configuration -> Advanced Features

Timeout

For shared call appearance, the phone must send a SUBSCRIBE request for the line pickup event packet each time a user attempts to pick up the shared line. “Line Pickup Timeout” is the expiration timer for the line pickup event. The default value is 15 seconds. The valid range is from 15 to 60.

Presence Eventlist URI

Configures Presence Eventlist URI to monitor extensions in MultiPurpose Keys.

If the server supports this feature, users must first configure a service-side Presence Eventlist URI (i.e. [email protected]) with a list of extensions included. On the phone, in this “presence event list URI” field, fill in the URI without the domain (i.e. presence). To monitor the extensions in the list, in Web GUI🡪Settings🡪Programmable keys , select “Presence Watcher” in key mode, choose the account, enter the value of each extension in the list.

List of BLF URI events

Configures the BLF URI event list on the phone to monitor the extensions in the list with the multipurpose key. If the server supports this feature, users must first configure an event list BLF URI on the service side (i.e. [email protected]) with a list of extensions included. On the phone, in this “Event List BLF URI” field, fill in the URI without the domain (i.e. BLF1006). To monitor the extensions in the list, under Web .

GUI🡪Configuration🡪Programmable keys , select “BLF Event List” in key mode, choose the account, enter the value of each extension in the list.

Automatic provisioning of BLF

When the option is enabled, empty multipurpose keys will be automatically provisioned to monitored extensions in “Event List BLF” or “Presence Event List“.

disabled
BLF events
presence

The default setting is “Disabled”.

Conference URI

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

Music on hold URI

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

BLF call capture

Configure the BLF call capture method:.

Automatic:

the phone will use either a prefix or an intrusion code for BLF capture, depending on which one is configured.

Force BLF call capture by prefix:

the phone will only use the prefix as the BLF capture method.

Disabled:

the phone will ignore both BLF capture methods, now the monitored VPK will only dial the extension if pressed.

The default setting is “Auto”.

BLF call capture prefix

Sets the prefix prefix prefixed to the BLF extension when the phone captures a call with the BLF key. The default setting is **.

Call capture interrupt code

Set the access code for the Call Capture feature with the Interrupt feature.

PUBLISH for Presence

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

Omit character set=UTF-8 in MESSAGE

Omit character set=UTF-8 in MESSAGE content type.

Default setting is disabled”.

Allow REFERENCE

Allow unsolicited REFERENCE to make an outgoing call.

Disabled
Enabled
Enabled/Forcing authentication

The default setting is “Disabled”.

Special feature

Different soft switch vendors have special requirements. Therefore, users may need special features selected 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”.

Broadsoft

Broadsoft Call Center

When set to “Yes”, a “BSCCenter” softkey is displayed on the LCD screen. The user can access different Broadsoft Call Center agent functions through this softkey.

Please note that “Function Key Synchronization” will be enabled regardless of this setting. The default setting is “Disabled”.

Note: To enable this feature, users must switch the special feature to Broadsoft and configure the Broadsoft call center for it to take effect.

Hoteling Event

Broadsoft Hoteling event feature. The default setting is “Disabled.” With the “Hoteling Event.

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

Call Center Status

In the process of configuring GRP2612W, when set to “Yes”, the phone will send SUBSCRIBE to the server to get the call center status. The default setting is disabled.” .

Executive Assistant at Broadsoft

When enabled, function key synchronization will be enabled regardless of the web configuration.

Function key synchronization

This feature is used for Broadsoft call feature synchronization. When enabled, DND, call forwarding and call center agent status features can be synchronized between the Broadsoft server and the phone. The default is “Disabled”.

Broadsoft call parking

When enabled, it will send SUBSCRIBE to the Broadsoft server to get Call Parking notifications. The default setting is disabled.” When enabled.

VQ RTCP-XR

VQ RTCP-XR CollectorName

Configures the hostname 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 PUBLISH SIP messages.

VQ RTCP-XR Collector Port

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

Enable RTCP

Allows the user to select to use RTCP, RTCP-XR or disable the function.

X-account -> SIP Configuration -> Session Timer

Enable

This option is used to enable or disable the session timer on the phone side when the server side can provide both Session Timer UPDATE and Session Audit UPDATE. The default setting is “No”.

Session expiration

The SIP session timer extension (in seconds) that allows SIP sessions to be periodically “refreshed” via a SIP request (UPDATE or reINVITE). If there is no update via an UPDATE or reINVITE message, the session will terminate once the session interval expires. Session timeout is the time (in seconds) at which the session is considered to have timed out, provided that a successful session refresh transaction does not occur first. The default setting is 180. The valid range is 90 to 64800.

Min-SE

In the process of configuring GRP2612W, if set to “Yes” and the remote location supports session timers, the phone will use a session timer when making outgoing calls. The default setting is “No”.

Request Timer

If set to “Yes” and the remote location supports session timers, the phone will use a session timer when receiving incoming calls. The default setting is “No”.

Forced timer

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

UAC Specify Update

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

UAS Specify update

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

Force INVITE

The session timer can be updated using the INVITE method or the UPDATE method. Select “Yes” to use the INVITE method to update the session timer. The default setting is “No”.

Account X -> SIP Settings -> Security

Check domain certificates

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

Validate certificate chain

Validate the certificate chain when TCP/TLS is configured.

Default setting is “No”.

Validate messages

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

Check the SIP user ID for INVITE incoming

If set to “Yes”, the 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”.

Accept incoming SIP only from proxy

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”.

Validate INVITE

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

Account X -> Audio Settings

Preferred Vocoder (Option 1 – 8)

The phone supports several vocoder types, the vocoders in the list have a higher preference. Users can configure vocoders in a preference list that is included with the same preference order in the SDP message.

Use the first matching vocoder in 200OK SDP

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

Codec Negotiation Priority

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

Hide Vocoder

When the Hide Vocoder option is set to Yes, the code will be hidden from the call screen as shown below. The default setting is “No”.

Disable multiple m-lines in SDP

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

SRTP mode

Enable SRTP mode according to your selection in the drop-down menu.

No
Enabled but not forced
Enabled and enforced
Optional

The default setting is “No”.

SRTP key length

Allows users to specify the duration of SRTP calls. The available options are:

AES 128-bit and 256-bit
128-bit AES
256-bit AES

Default settings are: 128-bit and 256-bit AES.

Typographic lifetime

Enable or disable cryptographic time to live when using SRTP. If users configure to disable this option, the phone does not add the cryptographic time-to-live to the SRTP header. The default setting is “Yes.”

Symmetric SRTP

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

Silence suppression

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

Jitter buffer type

Select Fixed or Adaptive for the jitter buffer type, depending on network conditions. The default setting is “Adaptive”.

Jitter buffer length

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

Voice frames per TX

Configures the number of voice frames transmitted per packet. When configuring this, it should be noted that the value of “ptime” for the SDP will change with different settings here. This value is related to the codec used and the actual frames transmitted during the 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.

Rate G723

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

The default setting is “5.3 kbps encoding rate”.

G.726-32 packing mode

Select “ITU” or “IETF” for the G726-32 packaging mode.

The default setting is “ITU”.

iLBC raster size

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

iLBC payload type

This option is used to specify the iLBC payload type.Valid range is 96 to 127. .

The default setting is “97”.

Type of loada OPUS tool

Specifies the type of OPUS payload. The valid range is 96 to 127. It cannot be the same as the iLBC or DTMF payload type. The default value is 123.

DTMF payload type

Configures the payload type for DTMF using RFC2833. It cannot be the same payload type iLBC or OPUS.

Send DTMF

This parameter specifies the mechanism for transmitting DTMF digits. There are 3 supported modes:.

In audio: DTMF is combined into the audio signal (not very reliable with low bit-rate codecs);
RFC2833 sends DTMF with RTP packet. Users can check the RTP packet to see the DTMFs sent, as well as the number pressed.
SIP INFO uses SIP INFO to transport DTMFs.

The default setting is “RFC2833”.

DTMF Delay

Sets the delay between DTMF sending during MPK/VPK usage (in milliseconds). The default value is “250” ms.

Account X -> Call setup

Dialing anticipated

Selects whether advance dialing is enabled. If set to “Yes”, the SIP proxy must support 484 responses. Early Dial means that the phone sends for each digit pressed a SIP INVITE message to the SIP server. The SIP server considers its extensions and, if no match has occurred yet, sends a “484 Address Incomplete” message. Otherwise, it executes the action.

The default setting is “No”.

Dial plan prefix

Configures the prefix to be added to each dialed number.

Plan Dialing plan

A dialing plan sets the expected number and digit pattern for a phone number. This parameter configures the allowed dialing plan for the phone. The default setting is “{ x+ | | *x+ | *x+ | *xx*x+ }”. Dial plan rules: .

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

x : any digit from 0 to 9.

X : digits from 0 to 9 and letters from az, AZ.

- xx+ – at least 2-digit numbers

- xx – 2-digit numbers only

- ^ – exclude
- [3–5] – any digit of 3, 4 or 5
- [147] – any digit of 1, 4 or 7
- <2=011> – replace digit 2 with 011 when dialing

- | – the OR operand
- , – second dial tone invitation. For example: {0,x+} will play the second dial tone after dialing 0 and all digits including 0 will be sent
{X123} : match Z123, e123, 5123,….
Dial Tby adding a “T” at the end of the plan dial, the phone will wait 3 seconds before dialing. This gives users more flexibility in setting up their dial plan. For example, with the 1XXT dial plan, the phone will wait 3 seconds to allow the user to dial more than 3 digits if necessary. Originally, the phone will dial immediately after dialing the third digit.
The backslash ““: can be used to escape specific letters. For example, if the dial plan { { \b>60} is set, park+60 should be able to pass the dial plan check. This can also be used to escape unreserved Mark and User characters.

Mark = “-” / “_” / “.” / “.” / “!” / “~” / “*” / “‘” / “(” / “)”

Unreserved user = “&” / “=” / “+” / “$” / “,” / “;” / “¿?” / “/” .

Example 1: {[369]11 | 1617xxxxxxx}

Allows 311, 611 and 911 or any 10-digit number with leading digits 1617; .

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

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

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

Allows any number with the 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 the starting digit 2, replacing the 2 with 011 when dialed.

Example 4: If we set the dial plan with {\*123}, it should be allowed to enter *123 to pass the dial plan check.
Example 5: if we set the dial plan with {\$123}, it should allow entering $123 to pass the dial plan check.
Example 6: if we set the dial plan with {12\_3}, it should allow the 12_3 entry to pass the dial plan check.

Example of a simple dial plan used in a US home/office: .

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

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

^1900x. – Prevents dialing any number beginning with 1900;
<=1617>[2-9]xxxxxx: allows dialing local area code numbers (617) by dialing 7 numbers and the area code 1617 will be added automatically;
1[2-9]xx[2-9]xxxxxx |- allows dialing any US/Canada number with a length of 11 digits.
011[2-9]x: allows international calls beginning with 011;
[3469]11 – allows dialing special and emergency numbers 311, 411, 611 and 911.

Note: In some cases, when the user wishes to dial strings such as *123 to activate voicemail or other applications provided by their service provider, the * must be predefined within the dial plan feature.

An example dial plan will be: {*x+} which allows the user to dial * followed by any length of numbers.
Explanation of the example rule (reading from left to right): .

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

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

Omit Dialing Plan

Enable/disable skip dialing plan when dialing via:.

Contact
Call history Incoming call Call history Outgoing call
Call history Outgoing call
Call history
Dial page
MPK
API

The default setting is “MPK”.

Call log

Set the call log on the phone.

The default setting is “Log all calls”.

Send anonymous

If set to “Yes”, the “From” header in outgoing INVITE messages will be set to anonymous, which will block the caller ID from being displayed. The default value is “No”.

Reject anonymous calls

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

Auto-answer

If set to “Yes”, the phone will automatically turn on the speakerphone to answer incoming calls after a short reminder beep. The default setting is “No”.

Refer-To Use Target Contact (Reference Use Target Contact)

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

Transfer at end of conference

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 the other parties remain on the conference call. The default setting is “No”.

Disable recovery on blind transfer

Disables call recovery to the transfer receiver on blind transfer failure to the destination. The default setting is “No”. Notes:.

1. This function only applies to blind transfer; 2.

2. This feature depends on how the server handles the transfer. If there is any

NOTIFICATION from the server, this feature will have no effect. If the server responds 4xx, the phone should attempt to recover regardless of this option.

3. During blind transfer, after the transferor received 200/202 to REFER, but there is no NOTIFICATION from the server after 7 seconds, the transferor will decide to retrieve the call with the transferor or not, depending on the options. This is the only case where this option will apply.

Blind transfer timeout

Defines the timeout time (in seconds) to wait for SIP fragment response in blind transfer. The valid range is from 30 to 300. The default setting is “30”.

Wait timeout without key input

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

Key as send

Allows users to set the “*” or “#” keys as the “Send” key. Make sure the dial plan is configured correctly to allow dialing * and #.

The default setting is “Pad (#)”.

Reminder tone standby

If set to “Enabled”, the phone will play a reminder tone when you have a call waiting. The default setting is enabled.”

RFC2543

Reminder RFC2543

Allows users to toggle between RFC2543 hold and RFC3261 hold. RFC2543 hold (0.0.0.0.0) allows the user to disable hold music sent to the other side. RFC3261 (one line) will play the hold music to the other side. The default setting is “No”.

Hide dialing password

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

Disable call waiting

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

Call tone

Account Ringtone Allows users to set the account ringtone. Users can choose from different ringtones in the drop-down menu.

Note: user can also choose the silent ringtone.

Incoming caller ID

Specifies matching rules with numbers, patterns or alert information text (up to 10 matching rules). When the incoming caller ID or alert information matches the rule, the phone will ring with the selected distinctive ring tone. Matching rules: Specific caller ID number. For example, 8321123; .

A defined pattern with a certain length using x and + to specify, where x could be any digit from 0 to 9. Samples:.

xx+ : at least a 2-digit number; xx : 2-digit number only; .

[345]xx: 3-digit number with the leading digit of 3, 4 or 5; [6-9]xx: 3-digit number with the first digit from 6 to 9.

Alert information text .

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

Select the distinctive ring tone for the matching rule. When the incoming caller ID or alert information matches one of the 10 rules, the phone will ring with the associated ring tone.

Wait for ringing

Defines the wait time (in seconds) for ringing if there is no answer. The default setting is 60. Valid range is 10 to 300.

Count X -> Intercom configuration

Enable automatic response by call/info alert

Enables the phone to automatically turn on the speakerphone to answer incoming calls after a short reminder beep when enabled, according to the SIP Call-Info/Alert-Info header sent from the server/proxy. The default value is “Yes.”

Enable inbound by call info/alert info

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

Silence when answering Intercom call

When enabled, the phone will mute the incoming intercom call. The default is no.”

Play warning tone for the auto answer intercom

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

The default value is “Yes”.

Custom alert information for automatic response

Allows you to customize the alert information header for automatic response. The phone will automatically answer only if the content matches the customized alert information header.

Account X ->Codes

Enable functions

When enabled, Do Not Disturb, call forwarding and other call features can be used through the local feature codes on the phone. Otherwise, the feature codes provided by the server will be used. User configured feature codes will be used only if server provided feature codes are not provided. And once the feature codes are configured, either through server provisioning or local configuration, a softkey labeled “Features” will be displayed on the LCD.

Note : if the device is registered with a Broadsoft account, no matter local call features are enabled or disabled, once Broadsoft account is configured, special feature for Broadsoft and function key synchronization is enabled, the call function will be handled by Broadsoft server, not the phone.

Do Not Disturb (DND): enabled

Set the DND function code to enable DND.

Do Not Disturb (DND): disabled

Set the DND function code to disable DND.

Call forwarding unconditional call forwarding (all): enabled

Set the Total call forwarding feature code to enable unconditional call forwarding.

Call forwarding unconditionally (all): disabled

Set the Forward all calls feature code to disable unconditional call forwarding .

Destination

Set the extension to which the call will be forwarded.

Call forwarding enabled

Set the busy call forwarding feature code to enable busy call forwarding.

Busy call forwarding- Disabled

Set the Busy Call Forwarding feature code to disable busy call forwarding.

Destination

Set the extension to which the call will be forwarded.

Delayed calls (no answer): enabled

Set the Call Forward Delayed feature code to enable call forwarding without answer.

Calls delayed (no answer): disabled

Set the Call Forwarding Delayed feature code to enable no answer call forwarding.

Destination

Set the extension to which the call will be forwarded.

Delayed call time

Defines the time to wait (in seconds) before the call is forwarded if not answered. The default value is 20 seconds. Valid range is from 1 to 120.

Accounts -> Change

Change Settings

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

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

What are the definitions on the Grandstream GRP2612W configuration page?

Setup -> Configuration general

Local RTP port

This parameter defines the local RTP port used for listening and transmitting. It is the base RTP port for channel 0. When set, channel 0 will use this _port_value for RTP; channel 1 will use port_value+2 for RTP. The local RTP port varies from 1024 to 65400 and must be even. The default value is 5004.

Local RTP port range

Gives users the ability to define the local RTP port parameter used for listening and transmitting. This parameter defines the local RTP port from 48 to 10000. This range will be set if the local RTP port + local RTP port range is greater than 65486. The default setting is 200.

Use random port

When set to “Yes”, this parameter will force random generation of 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 calls to work.

Connection hold interval

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

Use IP NAT

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

STUN Server

The IP address or domain name of the STUN server. The STUN resolution results are displayed on the STATUS page of the web GUI.

Only non-symmetric NAT routers work with STUN.

Delayed registration

Set the specific time the account will register after startup.

Test password difficulty

Only allow passwords with these restrictions to ensure better security: Password must be longer than 9 characters/digits and must meet at least 3 options among the 4 options below:.

Numeric (0-9)
Uppercase letters (AZ)
Lower case (az)
Special characters (!, @, #, $, $, %, ^, &, *, (, ), etc.) Default setting is “No”.

Public mode

Enable public mode

Set to enable/disable public mode for the desktop sharing feature. The default setting is disabled.”

Enable correction for RTP Time stamp skipping

Makes RTP timestamps continuous, if there is audio loss caused by timestamp skipping. The default value is “No”.

Public Username Prefix Mode

Used as public mode login prefix, when public mode is enabled.

Mode Public username suffix

Used as a user name suffix in public mode login, when public mode is enabled.

Allows remote synchronization

Allows the phone to automatically download the current account settings from the remote server and upload them to the server. The default setting is “Disabled”.

Server type

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

Server path

Defines the path of the server storing public account personal files.

FTP/HTTP username

Specifies the username for accessing the FTP/HTTP server.

FTP/HTTP password

Specifies the password to access the FTP/HTTP server.

Outbound notification

Enable notification

Indicates whether the output notification feature is enabled. The default value is .

“Enabled”. For details, see [OUTPUT NOTIFICATION SUPPORT].

Configuration -> Broadsoft -> Broadsoft XSI

XSI

Configures the XSI directory.

Server

Configure the BroadWorks Xsi server URI. If the server uses

HTTPS, add the “HTTPS” header in front of the server URI. For example, “https://URI_SERVIDOR”.

Port

Configure the BroadWorks Xsi server port. The default port is 80. If the server uses HTTPS, configure 443.

XSI share path

This function allows users to configure the deployment path for Broadsoft XSI actions. If empty, the path “com.broadsoft.xsiactions” will be used.

Broadsoft contact download interval

Sets the Broadsoft phonebook download interval (in minutes). If set to 0, the automatic download will be disabled. The valid range is 5 to 4320. The default value is 360.

XSI authentication type:

Select the XSI authentication type. The SIP user ID must be configured if the SIP account is selected.

Session credentials

Login username.
Set the username for the BroadWorks XSI server. or Logon password.
Set the password for the BroadWorks XSI server.

SIP Credentials or SIP Username.

Configure the SIP username for the BroadWorks XSI server. orSIP user ID.

Configure the SIP user ID for the BroadWorks XSI server. or SIP password.

Set the SIP password for the BroadWorks XSI server.

Sort phonebook by

Select to sort phonebook entries by “Last name” or “First name”.

The default setting is “Last name”.

Network directories

Enable/disable Broadsoft network directories and define the directory name.

The directory types are:.

Group directory

Enable/disable and rename BroadWorks Xsi Group Directory features on the phone. If you leave the Name box blank, the phone will use the pr name.

determined “Group”.

Business directory

Enable/disable and rename the BroadWorks XSI enterprise directory features on the phone. If you leave the Name box blank, the phone will use the default name “Enterprise”.

Group Common

Enable/disable and rename BroadWorks XSI Group Common Directory features on the phone. If you leave the Name box blank, the phone will use the default name “Common Group”.

Enterprise Common

Enable/disable and rename BroadWorks XSI Enterprise Common Directory features on the phone. If you leave the Name box blank, the phone will use the default “Enterprise Common” name for it.

Personal Directory

Enable/disable and rename the BroadWorks XSI home directory features on the phone. If you leave the Name box blank, the phone will use the default name “Personal”.

Missed Call Log

Enable/disable and rename the BroadWorks XSI Missed Call Log features on the phone. If you leave the Name box blank, the phone will use the default name “Missed”.

Record calls made

Enable/disable and rename BroadWorks Call Logging features XSI on the phone. If you leave the Name box blank, the phone will use the default name “Outbound”.

Record of received calls

Enable/disable and rename the BroadWorks XSI incoming call log features on the phone. If you leave the Name box blank, the phone will use the default name “Incoming”.

Setup -> Broadsoft -> Broadsoft IM&P

Access credentials

Server

Broadsoft IM&P server address. This usually does not need to be configured and can already be found in the Broadsoft IM&P username.

Port

Port for the Broadsoft IM&P server. The default port is 5222.

Name

of Broadsoft IM&P, not the Broadsoft account username.

Password

of Broadsoft IM&P, not the Broadsoft account password.

Broadsoft IM&P

Enables Broadsoft’s instant messaging and presence feature. The default setting is disabled.”

Broadsoft associated account

Specifies the associated account. The user can choose each account on the phone.

Automatic login

Choose whether to log into Broadsoft IM&P account during startup. The default setting is “No”.

Show contacts non XMPP

Choose whether to show non-xmpp contacts associated with the Broadsoft IM&P user. Non-xmpp contacts will not display a presence or status message. The default setting is “No”.

Settings-> External Service

Order (1 – 10)

Displays the order of service.

Type of service

Specifies the type of service. Two options are available:.

None
GDS

The default setting is “None”.

Note: The GRP261x/GRP2624/GRP2634 supports up to 10 GDS items.

For more details, refer to the Connecting GDS3710 with GRP26XX

Account

Specifies the account to which the service will be applied.

System ID

Specifies the name to identify the service.

Number of the system

Specifies the system number, in case the service type option is set to GDS, the system number is the SIP user ID configured in GDS3710/GDS3705, or the IP address of the GDS3710/GDS3705 itself if you are using an IP call.

Password

Determine the access password, in case the service type option is set to GDS, the access password is the one set in the “Remote PIN to open door” field in the GDS3710/GDS3705 configuration.

Configuration -> Call functions

Default preferred account

Selects the preferred default account when dialing with the phone off-hook/hung up. When the selected account is not available, the system will revert to using the first available account instead.

Select account from LCD

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

Predictive dialing function

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

Predicative dialing font

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

Onhook Dial Barging

Allows incoming calls to interrupt dialing with the phone hung up when set to “Enabled”. The default setting is “Enabled”.

Auto off-hook dialing

Set a user/extension ID to automatically dial when the phone is off-hook. The phone will use the first account to dial. The default setting is “No”.

Auto delay off-hook

Set the number of seconds for which the phone will wait before dialing when the off-hook autodial number is set. The default value is 4.

Timeout for off-hook

If set, when the phone is off-hook, it will hang up after the timeout time (in seconds). The default value is 30 seconds. Valid range is 10 to 60. Enable Live DialPad Enables automatically dialing the entered number after the user set amount of seconds when the phone is off-hook. The default value is “No”.

Live DialPad expiration time

Set the Live DialPad expiration time before initiating the call using the Live DialPad feature. The interval is between 2s and 15s. The default value is 5s.

Forward all last calls

Set to enable storing the last number entered when entering the number in the call screen after pressing the Forward All softkey. The default value is “No”.

Enable Auto Redial

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

If enabled, the phone will prompt the user to initiate “auto redial” or not. If yes, the phone will redial the called number multiple times [Auto Redial Times] with [Auto Redial Interval] between each call. The user is guided through different prompts on the phone’s LCD display showing the number of attempts remaining, the countdown to initiate the next automatic redial and allowing the user to manually initiate the call without waiting for the specified [Auto Redial Interval] interval. The phone will stop automatic redial after a successful attempt (called party is not busy) or after failed attempts [Times]. The default setting is “No”.

Auto Redial Times

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

Auto redial interval

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

Bypass dial plan through call history and directories

Enable/disable dial plan checking while dialing through call history and any phone book directories. The default setting is “No”.

Disable call waiting

Disable the call waiting feature. The default setting is “No”.

Disable call waiting tone

Disables the call waiting tone when call waiting is enabled. The default setting is “No”.

Call waiting ringer

Disables/enables ringing instead of playing call waiting tone when audio is on the phone or headset. The default setting is “No”.

Disable busy tone on remote disconnect

Disable the busy tone heard in the handset when the call is disconnected remotely. The default setting is “No”.

Disable direct IP call

Disable direct IP calling.

The default setting is “No”.

Use fast IP calling mode

When set to “Yes”, users can dial an IP address on the same LAN/VPN segment by entering the last octet in the IP address.

To dial a fast IP call, pick up the phone, press # to switch to “Direct IP call” mode and dial XXX (X is 0-9 and XXX <=255), the phone will make a direct IP call to aaa.bbb.ccc .XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of the subnet mask. XX or X are also valid, so no leading 0 is required (but that’s OK). A SIP server is not required to make a fast IP call. The default setting is “No”.

Disable conferencing

Disable the conference feature. The default setting is “No”.

Disable screen

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

Default setting is “No”.

Enable DTMF sending via specific MPKs

Enable certain MPKs to send DTMF during a call. This option does not affect DTMF Dialing.

The default setting is “No”.

Disable active page MPKs

When the option is enabled, the active page MPK on the extension boards will be disabled.

The default setting is “No”.

Enable active page VPKs

Enables the active VPK page to be displayed on the LCD when there are active VPKs.

Default setting is “No”.

Enable DND function

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

The default setting is “Yes”.

Mute key functions while idle

Specifies the function of the mute key while idle. The default setting is “NOM.”

When you select “Silent Idle” and press the Silent key while idle, the future incoming call will be answered with silence. When you select “Disabled”, the Mute key will have no effect while inactive.

The default setting is “NOM”.

Override DND

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

Disabled: all incoming calls will not be accepted.
Allow all: all incoming calls will be allowed.
Allow contacts only: only incoming calls from numbers in the local phonebook will be allowed.
Allow favorites only: only incoming calls from favorite numbers in the local phonebook will be accepted.

The default setting is “Disabled”.

Disable transfer

Enables/disables the transfer function. If disabled, call transfer will not be possible. The default setting is “No”.

Dial number in call on pressing the transfer key

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

Assisted Transfer Mode

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

The default setting is “Dynamic”.

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

Transfer mode via VPK

Perform a “Blind Transfer”, an “Assisted Transfer” or a “New Call” with the number specified in the Value field when a user presses the “Transfer” multi-purpose softkey. The default is “Blind Transfer”.

Hold call on transfer

When set to “No”, the phone will neither hold the current call in the transfer window nor hold the call with the transfer destination before forwarding the call in the attended transfer. The default setting is “Yes”.

Show call waiting duration

Display the duration of a call on hold on the LCD screen.

The default setting is “Yes”.

Do not escape # as %23 in SIP URI

Specifies whether to replace # with %23 or not for some special situations. The default setting is “No”.

Click-to-dial function

Enables theIf this function is enabled, the user can click the green dial button in the upper left corner of the phone’s web GUI, then choose the account and dial the destination number. The default setting is disabled.”

For more details, see [CLICK-TO-DIAL].

Default call log type

Sets the default call log list after selecting MENU🡪HISTORY.

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

Return code when rejecting an incoming call

The phone will send the selected SIP message type of the call. The available options are:

Busy (486).
Temporarily unavailable (480).
Not found (404).
Declive (603).

The default setting is “Busy 486”.

Return code when enabling DND

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

Busy (486).
Temporarily unavailable (480).
Not found (404).
Declive (603).

The default setting is “Temporarily unavailable (480)”.

Enable BLF capture screen

By enabling the BLF capture screen, when the monitored BLF is ringing, GRP261x/GRP2624/GRP2634 will display a BLF information window. The default setting is “No”.

Enable BLF capture sound

Enable BLF Gives the user the ability to configure a sound notification for the BLF line monitoring when it is ringing, GRP261x/GRP2624/GRP2634 will play a sound to inform the user. The default setting is “No”.

Excluded list capture sound in BLF

Configures the list to play BLF sound notification for “All except” extensions in the [BLF Capture Sound Exception List] list or “Allow only” extensions in the [BLF Capture Sound Only List] list. The default setting is “Allow except”.

List capture sound “only” in BLF

Sets the BLF sound notification of playback only for the list below.

Local call recording function

Gives the ability to record calls locally while in the call screen. The default setting is disabled.”

Replace the oldest call log

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

Downloading local call recordings

When recordings are submitted, you can download them here.

Enable IM pop-up window

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

Instant pop-up timeout message

Set the number of seconds the message will remain on the screen.

The valid range is 10 – 900. The default setting is “10”.

Play tone when receiving IM

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

Allow incoming calls before ringing

This allows incoming calls after dialing but before ringing. This can be used under a custom user setting based on need. The default setting is “No”.

User agent prefix

Set the prefix in the user agent header.

Hide BLF remote control status

Allows users to hide the caller ID from being displayed in the BLF VPK and MPK.

No: the VPK will flash between the caller ID and the BLF account.
Yes: the VPK will remain under the monitored account and will only notify that there is an incoming call.

The default setting is “No”.

Show response of SIP error

Displays SIP error response information on the LCD. The default setting is “Yes.”

Enable missed call notification

Allows users to show/hide the missed call notification popup window.

The default setting is “Yes”.

Note: Currently, manually rejected calls are counted as missed calls.

Enable call completion service

When auto redial and call completion service are enabled, and the user places a call to the called party, when the called party is currently busy, the phone will monitor the status of the called party. Once the called party is available, the phone will ask the caller if he/she wants to redial. The default setting is “No”.

Enable incoming call popup

If set to “Yes”, the phone will display an incoming call pop-up window to notify the call. If set to “No”, no notification will appear on the LCD screen when there is an incoming call. In this way, users will not be interrupted by an unexpected pop-up call, but will still be notified by the flashing line LED.

The default setting is “Yes”.

Enable acoustic echo enhancement

Enables/disables the Enhanced Acoustic Echo Canceller (EAC) which provides the acoustic echo reduction required for full-duplex hands-free speakerphone functions on the phone. The default setting is “Yes.”

Auto answer delay

Set the delay to automatically answer the incoming call. The valid range is 0 to 10 (second). The default value is 0 (which means automatic answering is disabled).

Setting -> Multicast paging

Enabled in DND mode

Enables multicast paging when DND mode is enabled.

The default setting is “No”.

Paging Barge

During an active call, if the incoming multicast PA has higher priority (1 being the highest) than this value, the call will be put on hold and the multicast PA will be played. The default setting is disabled.”

Paging priority active

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

Multicast paging codec

The codec for sending multicast pages, 5 codecs can be used: PCMU, PCMA, G.726-32, G.729A/B, G.722 (wideband), G.723.1. The default setting is “G.722 (wideband)”.

Multicast channel number

multicast channel number (0-50). 0 for normal RTP packets, 1-50 for Polycom multicast format packets.

Multicast Sender ID

Outbound caller ID displayed to recipients in your paging group (for multicast) channel 1 – 50).

Listening multicast

Defines multicast listening tags and addresses. For example:.

“Listening address” must match the sender value, such as

“237.11.10.11:6767”.

“Label” could be the description you want to use.

For more information, please refer to the “Multicast Paging User Guide” on our website.

Configuration -> Outbound notification

URL action

For detailed instructions on this part, please refer to the [OUTGOING NOTIFICATION SUPPORT] section in this Administration Guide.

Configuration complete
Registered
Not registered
Failed registration
Hang-up
Hanging
Incoming call
Outgoing call
call missed
Callback contested
Callback rejected
Callback forwarded
Callback set
Call finalized
Inactive to busy
Busy to inactive
Open DND
Close DND
Open Resend
Close Resend
open Resend unconditional
Close Resend unconditional
Open Busy Resend
Close Busy Resend
Open No reply Resend
Close No Reply Resend
Blind Transfer
Transfer attended
Transfer completed
Transfer failed
Call waiting
Waitless call
Muted call
IP change
Autoprovisioning completed

Destination

Up to 10 destinations can be configured here. For detailed instructions on this part, please refer to the [OUTGOING NOTIFICATION SUPPORT] section in this Administration Guide.

Notification

Specifies the notification message body for each event that can be customized with embedded dynamic attributes.

For more details, refer to the [OUTGOING NOTIFICATION SUPPORT] section in this Administration Guide.

Settings -> Preferences ->Audio control

Headphones / Headphones

Headphone key mode

When the headset is connected to the phone, users can use the HEADSET button in “Default Mode” or “Toggle Headset/Speakerphone”.

1. Default mode:

When the phone is idle, press the HEADSET button to pick up the phone and make calls using the headset. The headset icon will be displayed on the screen in dial/talk status.
When there is an incoming call, press the HEADSET button to answer the call using the headset.
When there is an active call using headphones, press the HEADSET button to hang up the call.
When using the speakerphone/phone in the dial/talk state, press the HEADSET button to switch to headset. Press it again to hang up the call. Or press speaker/phone to return to the previous mode.

2. Toggle handset/speakerphone:

When the phone is idle, press the HEADSET button to switch to headset mode. The headset icon will be displayed on the left side of the screen. In this mode, if you press the Speaker button or Line key to pick up the phone, the headset will be used.
When there is an active call, press the HEADSET button to toggle between Headset and Speakerphone.

Headset type

Selects whether the connected headset is a normal RJ11 headset or a Plantronics EHS headset. The default setting is “Normal”.

EHS Headset Tone

Selects a normal ringtone or a Plantronics EHS ringtone for Plantronics EHS headsets. The default is “Normal.”

Note: You also need to set the “Headset Key Mode” to “Toggle .

Headset/Speakerphone” and manually press the HEADPHONES button on the keypad to switch to Headphone mode.

Always sound speakerphone

Set to enable or disable the speakerphone to sound when using headphones in “Toggle Headphone/Speakerphone” mode.

If set to “Yes, both”, when the phone is in “Alternate Earpiece/Speakerphone” headset mode, both the earpiece and speakerphone will ring on the incoming call.
If set to “Yes, speakerphone only “, when the phone is in headset mode “Alternate headset/speakerphone”, only the speakerphone will ring on the incoming call.

The default setting is “No”.

Enable EDRC function

Enable EDRC function, the remote party will not hear the ambient noise during a call and thus improve the communication quality.

Headset TX gain

Sets the TX gain of the headset.

Available values: -6dB, 0dB or +6dB. The default value is 0dB.

Headphone RX gain

Sets the receive gain of the headset.

Available values : -6dB, 0dB or +6dB Default value is 0dB .

Auricular

Headphone TX Gain

Sets the TX gain of the handset.

Available values: -6dB, 0dB or +6dB. The default value is 0dB.

Settings ->Preferences -> Date and time

NTP Server

Defines the URL or IP address of the NTP server. The phone can get the date and time from the server.

The default setting is “pool.ntp.org”.

Secondary NTP server

Defines the URL or IP address of the NTP server. The phone can get the date and time from the server. Allow the user to configure 2 domain names of the NTP server. GRP will traverse all resolved IP addresses from them.

NTP Update Interval

Interval of time to update the time from the NTP server. The valid time value is between 5 and 1440 minutes.

The default setting is “1440” minutes.

Allow DHCP option 42 to override the NTP server

Defines whether DHCP option 42 should override the NTP server or not. When enabled, DHCP option 42 will override the NTP server if it is configured on the LAN. The default setting is “No”.

Time zone

Sets the date/time used on the phone according to the specified time zone. The default setting is “Auto”.

Allow DHCP option 2 to override the time zone setting

Allow the device to be provisioned for the time zone from DHCP option 2 on the local server. The default setting is enabled.

Self-defined time zone

This parameter allows users to define their own time zone, when the “Time zone” parameter is set to “Self-defined time zone”.

The syntax is: std offset dst [offset], start [/time], end [/time] The default value is: MTZ+6MDT+5,M4.1.0,M11.1.0 .

MTZ+6MDT+5 .

This indicates a time zone with 6 hours difference with 1 hour ahead (when it is Daylight Saving Time), which is Central Time in the U.S. If is positive (+) if the local time zone is west of the Prime Meridian (AKA: International or Greenwich Meridian) and negative (-) if it is east.

M4.1.0,M11.1.0 .

The first number indicates the month: 1,2,3…, 12 (for January, February,…, December).

The second number indicates the nth iteration of the day of the week: (first Sunday,third .

TuesdayTuesday) .

The 3rd number indicates the day of the week: 0,1,2,. ,6(for Sunday, Monday, Tuesday, …, Saturday) Therefore, this example is Daylight Saving Time starting from the first Sunday in April to the 1ster Sunday in November.

Date display format

Set the date display format on the LCD. The following formats are supported.

yyyyyy-mm-dd: 2019-03-02
mm-dd-yyyy: 03-02-2019
dd-mm-dd-yyyy: 02-03-2019
dddd, MMMM dd: saturday, March 02
MMMM dd, dddd: March 02, Saturday The default setting is yyyy-mm-dd.

Display formatof the time

Configures the time display in either 12-hour or 24-hour format on the LCD. The default setting is in 12-hour format.

Display date in status bar

Allows users to display the time and date on the top panel of the LCD. The default setting is “No”.

Setting -> Preferences -> Office time

Allows users to configure office hours for each day.

Settings-> Preferences-> Language

Language Display language

Select the display language on the phone. There are 21 languages can be set as display language, user can also choose “Auto” or “Downloaded language” as display language. The default setting is “Auto”.

Default input selection

Set the default input selection.

Multiple Touch: Touch multiple to switch characters;
Shiftable: select input from available characters.

The default setting is “Multi-Toque”.

Automatic language download

Used to configure the device to download language files automatically from the server. The default setting is “No”.

Settings-> Preferences-> LCD Display

Backlight brightness: Active

Set the brightness of the LCD display when the phone is active. The valid range is from 10 to 100, where 100 is the brightest. The default value is “100”.

Backlight brightness: inactive

Set the brightness of the LCD display when the phone is idle. The valid range is 0 to 100, where 0 is off and 100 is the brightest. The default value is “60”.

Backlight brightness time

Allows the user to set the backlight time (in minutes). Valid range from 0 to 90. Default value is “1”.

Note: When the active backlight timeout is set to 0, the backlight will be on constantly.

Power saving timeout

Set how long to wait before the LCD automatically turns off after office hours. Valid range is 0 to 90 minutes, where 0 means disable the function.

Disable missed call backlight

Enable/disable the LCD backlight when there is a missed call notification.

If set to “Yes”, the display will turn off the LCD backlight even if there is a missed call on the phone.
If set to “Yes, but MWI LED is flashing”, the phone will turn off the LCD backlight, but MWI will not be considered when there is a missed call.
If set to “No”, the phone will not turn off the LCD backlight when there is a missed call.

The default setting is “No”.

Background

Origin of the wallpaper

Specifies the wallpaper source mode: Default, Download, Loaded and Background color. User can load a wallpaper source on their phone or download it from the file server with the server path.

Note: If you choose “Background Color”, you must enter a HEX color code according to your preference. The color codes can be found here:

http://htmlcolorcodes.com/ . If an invalid code is set, the phone will use the default #000000 instead.

Wallpaper server path

Specifies the path of the wallpaper server. This option will take effect when the wallpaper source is “Download”.

Load wallpaper

Click the “Load” button to browse for and load the desired wallpaper file. This option will take effect when the wallpaper source is “Loaded”.

Background color

Enter a color you want to use in HEX format. For example, #000000 .

Reference: http://htmlcolorcodes.com

Please note that the user must select “Color Background” in the “Wallpaper Source” option to use the configurable color background code.

Screen saver

Display

Set the screen saver function, or “to enable the screen saver function if there is no screen saver.

VPK active.” Note that this option is also available in LCD🡪 Menu -> Preference -> Aappearance_ The phone will consider the page active if VPK is in Early (ringing), Attempting (dialing) and Confirmed (talking) state when VPK is configured with “BLF”, “BLF Event List” or “Presence” mode. By default, the screen saver is set to “On if no VPK is active”.

Screen saver source

Set the location from where the screensaver is loaded. Show date and timeAllows you to view the time and date in the screensaver mode of the phone. The default setting is “Yes”. screen saver wait Screen saver wait Configures the minutes of inactivity before the screen saver is activated. The valid range is 3 to 60. The default time is 3 minutes.

Screen saver path

Set the server path containing the download screensaver definition XML file.

Download screensaver XML interval

Set the screen saver XML download interval (in minutes). If set to 0, the automatic download will be disabled. Valid range is 5 to 720. The default setting is “0”.

Setting -> Preferences ->LED control

BLF LED pattern

Configures the LED color and pattern based on status updates. The default setting is “Default”.

The BLF LED patterns are listed in [BLF LED PATTERNS].

Disable VM/MSG power-on light blinking

The VM/MSG light cannot blink even if there is an unread voicemail or message when set to “Yes”. The default setting is “No”.

Line LED color scheme

Configures the color scheme of the line key LEDs.

Default setting: off (inactive)/green (in use)
On: green (inactive)/red (in use)

BLF LED pattern explanation form

Users can see the color and pattern of the LED status according to the BLF status update.

Settings-> Tone of call

Call progress tones:

Call progress tone system
Dial tone
Second dial tone
Message waiting
Call waiting tone call waiting
callback tone
call waiting tone
Busy tone
Recording tone

Set ring or tone frequencies based on local telecommunications parameters. The default value is the North American standard. Frequencies should be set to known values to avoid annoying high-pitched sounds.

Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]];

(Frequencies are in Hz and on/off cadence is in 10ms).

ON is the call period (“On time” in ‘ms’) while OFF is the silent period.

To set a continuous ringing, OFF must be zero. Otherwise, it will ring ON ms and pause for OFF ms and then repeat the pattern. Up to three cadences are supported.

Call waiting ring tone gain

Set the call waiting tone gain to adjust the volume of the call waiting tone (low, medium or high). The default setting is “Low”.

Speaker volumespeakerphone

Set the ringer volume of the speaker. Valid range is from 0 to 7. The default setting is 5.

Notification tone volume

Configures the notification tone volume.

The valid range is from 0 to 7 and the default setting is 5.

Call tone volume

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.

Used to set the ringtone level in dB. The values range from -15 to 15.

Lock speaker volume

Block the volume setting when the option is enabled so that it cannot be changed from the phone’s LCD screen. The option can be set to: “No”, “Ring”, “Talk” or “Both”. The default setting is “No”.

Default ring tone

Allows you to set the default ringtone as your global ringtone.

Note: The ringtone set in individual accounts has higher priority than this setting. If the user wants the default ringtone to be used globally, he/she must set the ringtone of each account to Default Ringtone; Otherwise, it will be whatever ringtone he/she sets.

Important: priority is: Contact ringtone -> Account ringtone -> Default ringtone..

Settings-> Programmable keys -> Virtual multipurpose key settings

Idle screen configuration

Display Background tag

If enabled, the VPK label background will match the VPK status and will no longer be transparent.

The default setting is “No”.

Use long tag

If enabled, the VPK tag will be extended as far as possible.

The default setting is “Yes”.

Call screen settings

Key mode

If set to “Line Mode”, the amount of VPK will be the amount of lines you can have. If set to “Account Mode”, the lines will be grouped per account, so VPKs could have more lines in one account. .

For example, with Line Mode, when the line is in use, pressing the VPK, nothing will happen. In Account Mode, when the line is in use, pressing VPK will start a new line.

The default setting is “Account Mode”.

Transfer mode via VPK

Allows users to configure “Transfer” VPK for blind or assisted transfers. They can also configure their Transfer key to place a new call with the configured number. The default setting is “Blind transfer.”

Enable transfer through MPK without transfer

MPK with BLF type, speed dial, etc. will function as transfer MPK on an active call. The default setting is “No”.

Show key label

If set to “Show“, the side labels will be displayed during calls.
If set to “Hide“, the side labels will be hidden during calls to have more space to display user information.
If set to “Hide“, a softkey will appear for users to click to Show/Hide side labels.

The default setting is “Toggle.” .

Mode

Assigns a function to the corresponding line key. The key mode options are:.

None
Predetermined

Normal line key to open a line and switch lines. The Value field can be left blank.

Shared

Shared Shared Line for Shared Line Appearance functionSelect the Account registered as Shared Line for the line key. The Value field can be left blank.

Note: users can show or hide the VPK shared line display description. This can only be done with provisioning using Pvalue P8484 (Value = 0; No. Value = 1; Yes) .

Quick Dial

Select the account to dial from. And enter the speed dial number in the Value to dial field, or enter the IP address to set the direct IP call as speed dial.

Busy Lamp Field (BLF)

Select the account to monitor BLF status. Enter the extension number in the Value field to be monitored.

Presence Monitor

This option must be compatible with a presence server and is linked to the “Do Not Disturb” status of the phone extension.

List of BLF events

This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it in a single notification message. The PBX server has to support this feature.

Speed dial via an active account

Similar to Speed Dial but will dial based on the current active account. For example, if the phone is off-hook and account 2 is active, it will call the speed dial number configured using account 2.

Dial DTMF

Enter a series of DTMF digits in the Value field to dial during the call. “Enable DTMF MPK send” must be set to “Yes” first.

Voice Mail

Select Account and enter the voicemail access number in the Value field.

Callback

Last answered calls can be dialed using Callback. The Value field must be left blank. Also, this option is not account binding 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 to transfer (blind transfer) field during the call.

Call park

Select Account and enter the call park extension in the Value field to park/answer the call.

Monitored Calls

Select theaccount in the Account field and enter the call park extension in the Value field to park/answer the call, and also monitor the parked call through the line key light.

Intercom

Select Account and enter the extension number in the Value field to perform intercom.

LDAP Search

This option is to narrow the scope of the LDAP search. Enter the LDAP search base in the Description field. It can be the same or different from the .

base in the LDAP configuration under Advanced settings. The Base setting in LDAP will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For example:

if users configure MPK 1 as “LDAP Search” for “Account 1”, and configure filters: .

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

Value -> sn=Li .

from the base for LDAP server configuration is: .

“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the scope of the LDAP search. “sn=Li” is the example to filter the last name.

Conference

Enable the user to set their multipurpose key to “Conference” mode to initiate a conference.

By setting the extension number in the value box, users will be able to activate a 3-way conference by simply pressing the assigned MPK button.

Multicast paging

This option is for multicast sending. Enter the description of the line key in the Description field and the multicast sending address in the Value field.

Save

This option is for Record calls. Enter the description of the line key in the Description field and the recorded extension number in the Value field. Make sure your VOIP provider supports this feature before using it.

Call Logging

Select Account and enter the account number in the Value field to enable call log setup for another extension.

Menu

Select this function to display the Menu from the MPK buttons, no field is required for configuration.

XML application

Select this function to start XML application from MPK buttons, no field is required for configuration.

Information

Select this function to display the Information popup window to show the firmware version, MAC address, IP address and IP configuration from MPK buttons, no field is required for configuration.

Message

Select this function to display the Message menu from the MPK buttons, no field is required for configuration .

Forward

Set the MPK button to perform call forwarding to the destination number set in the “Value field”. During ringing, press the button to perform call forwarding.

Do not disturb

Press the key configured to enable/disable DND.

Remarking

In this mode, the configured key can be used to redial numbers.

Instant messages

In this mode, the configured key can be used to enter the IM menu and send new messages.

Multicast listen address

The MPK button can be used to directly access the .

multicast listening IP list.

Blocking

Configure the VPK button that will be used to lock/unlock the keyboard.

GDS DoorOpen

Configure the VPK button to be used as GDS DoorOpen. Set the PIN code in the value field.

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 in a single notification message.

Note: The PBX server must support this feature.

Provisioning

Select this feature to have the phone trigger an instant provisioning. .

Accounts

Select the account to be used with specific VPK.

Description

Enter a tag for VPK.

Value

Specifies the VPK value according to the selected mode.

Blocked

Blocks the VPK.

Settings -> Programmable keys -> Softkey settings.

Programmable key display mode

Allows users to choose between the original Toggle mode or the Enhanced Menu mode. With the enhanced Menu mode, the MORE softkey will now not require the user to tap MORE multiple times to move to the next pages; instead, pressing MORE will display a pop-up window and allow users to choose from the list. With Toggle mode, users must press the MORE softkey to switch between options. The default setting is “Menu”.

Show Destination Softkey

Allows users to remove the destination softkey by toggling the Yes/No option during the off-hook dialing screen and the transfer screen. The default setting is “Yes.”

Programmable key layout

Custom softkey layout

Enables/disables custom softkey layout.

The default setting is disabled.”

Enforce the layout setting

Either to enforce the custom softkey layout position.

When enabled, the GUI will still retain the space if the configured softkey cannot be displayed. The default setting is “No”.

Inactive display softkey layout

Hide system softkey on home page

Set to hide the system generated softkey (Next, History, Forward all, Redial all, Redial) on the main page. The default value is none.

Custom softkey layout of the call screen

Dialing status

Customize the softkey layout when the device is in Dialing State. .

Available softkeys: Phonebook (BT), BT On/Off, End Call, ReConf, Conference Room, Redial, Dial, Backspace, Pickup, Destination, Show/Hide Label, Custom Softkey 1, Custom Softkey 2, Custom Softkey 3
.
Default programmable keys: Phonebook (BT), BT on/off, End call, ReConf, Conference room, Redial, Dial, Backspace, Pickup, Destination.

Hanging dialing status

Customize the layout of the softkeys when the device is in the Hanging dialing state.

Programmable keys available: Phonebook (BT), DirectIP, Hangup, Cancel, Dial, Backspace, Show/Hide Destination Label, Custom Softkey 1, Custom Softkey 2, Custom Softkey 3.
Default programmable keys: Phonebook (BT), DirectIP, Hang up, Cancel, Dial,

Backspace, Destination

Call status

Customize the layout of the softkeys when the device is in call state. .

Programmable keys available: Answer, Reject, Resend, ReConf, Show/Hide Label, Mute, Custom Softkey 1, Custom Softkey 2, Custom Softkey 3.
Default programmable keys: Respond, Reject, Resend, Resend, ReConf.

Call status

Customize the layout of the softkeys when the device is in call state.

Programmable keys available: BT On/Off, Cancel, EndCall, ReConf, ConfRoom, ConfCall. Show/Hide Label, Custom Programmable Key 1, Custom Programmable Key 2, Custom Programmable Key 3.
Default programmable keys: On/Off BT, Cancel, End Call, ReConf, Conference Room, Conference Call.

Call connection status

Customize the layout of the softkeys when the device is in call connect state.

Programmable keys available: Phonebook (BT), BT on/off, End call, ReConf, Conference room, ConfCall, Cancel, New call, Swap, Transfer, Trnf>VM, DialDTMF, BS-CCenter, Record on/off (UCM), Record on/off, Call park (UCM), Private hold, Call park. Show/Hide Label, Standby,

Conference, Mute, Custom Programmable Key 1, Custom Programmable Key 2, Custom Programmable Key 3.

Default programmable keys: Phonebook (BT), Enable/Disable BT, End Call, ReConf, Conference Room, Conference Call, Cancel, New Call, Swap, Transfer, Trnf>VM,

DialDTMF, BS-CCenter, Enable/Disable Recording (UCM), Enable/Disable Recording, Call Parking (UCM), Private Hold, Call Parking.

Conference connected status Customize the layout of the softkeys when the device is in conference connected state.

Programmable keys available: Enable/Disable BT, End Call, Kick, New Call, Trnf>VM, DTMF Dial, BS-CCenter, Enable/Disable Recording (UCM), Enable/Disable Recording, Conference Room, Add, Show/Hide Label, Hold, Split, Mute, Custom Softkey 1, Custom Softkey 2, Custom Softkey 3.
Default softkeys: Enable/disable BT, End call, Kick, New call, Trnf>VM, DialDTMF, BS-CCenter, Enable/disable recording (UCM), Enable/disable recording , Conference room, Add.

Standby status

Customize the layout of the softkeys when the device is in standby status.

Programmable keys available: ReConf, Resume, Transfer, ConfCall, Add, Show/Hide Label, NewCall, EndCall, Custom Softkey 1, Custom Softkey 2, Custom Softkey 3.
Default programmable keys: ReConf, Resume, Transfer, ConfCall, Add.

Failed call status

Customize the layout of the softkeys when the device is in failed call status.

Available softkeys: End Call, ReConf, Conference Room, Show/Hide Label, New Call, Custom Softkey 1, Custom Softkey 2, Custom Softkey 3.
Default softkeys: End Call, ReConf, Conference Room.

Status in Transfer

Customize the layout of the softkeys when the device is in Transfer State.

Programmable keys available: BT On/Off, Cancel, BlindTrnf, AttTrnf, Backspace, Target, Show/Hide Label, Custom Softkey 1, Custom Softkey 2, Custom Softkey 3.
Default programmable keys: BT On/Off, Cancel, BlindTrnf, AttTrnf, Backspace , Target.

Conference Status

Customize the layout of the softkeys when the device is in Conference State.

Programmable keys available: BT on/off, cancel, dial, backspace, target, show/hide label, custom softkey 1, custom softkey 2, custom softkey 3.
Default softkeys: BT on/off, cancel, dial, backspace, destination.

Settings ->Programmable keys -> Idle screen programmable keys

Order (1 – 2)

Displays the idle screen softkey index. (1 – 2)

The GRP261x/GRP2624/GRP2634 supports 4 configurable programmable keys.

Note:The first and last softkey are reserved for the Exit/More function.

Mode

Assigns a function to the corresponding softkeys. The key mode options are:.

Speed dial

Select the account to dial from. And enter the speed dial number in the Value field to dial from.

Speed dial through an active account

Similar to Speed Dial but will dial based on the current active account. For example, if the phone is off-hook and account 2 is active, it will call the speed dial number configured using account 2.

Voicemail

Select Account and enter the voicemail access number in the Value field.

Callback

Intercom

Select Account and enter the extension number in the Value field to perform intercom.

LDDAP search

This option is to narrow the scope of the LDAP search. Enter the LDAP search base in the Description field. It can be the same or different from the base in the LDAP configuration under Advanced settings.

The Base configuration in LDAP will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field.

For example: if users configure MPK 1 as “LDAP Search” for “Account 1”, and configure filters: .

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

Value -> sn=Li .

from the base for LDAP server configuration is .

“dc=87randstream,dc=com”, “ou=video,ou=SZ” is added to narrow the scope of the LDAP search. “sn=Li” is the example to filter the last name.

Call log

Select Account and enter the account number in the Value field to access the Call Log for that selected account.

Menu

Select this function to display the Menu from the MPK buttons, no field is required for configuration.

Information

Select this function to display the Information popup window to show the firmware version, MAC address, IP address and IP configuration from the MPK buttons, no field is required for configuration.

Message

Select this function to display the Message menu from the MPK buttons, no field is required for configuration.

Account

Select the account to be used with a specific softkey.

Description

Enter a label for the softkey.

Value

Specifies the value of the softkey according to the selected mode.

Settings-> Programmable keys -> Call screen softkeys

Order (1 – 3)

Displays the softkey index of the idle screen. (1 – 3)

Note:Make sure that “Custom softkey 1/2/3” is selected in [Custom call screen softkey layout].

Call screen programmable keys

Assigns a function to the corresponding call screen softkeys.

Speed dial

Select the account to dial from. And enter the speed dial number in the Value field to dial from.

Speed dial through an active account

Similar to Speed Dial but will dial based on the current active account. For example, if the phone is off-hook and account 2 is active, it will call the speed dial number configured using account 2.

Dial DTMF

Enter a series of DTMF digits in the Value field to dial during the call. “Enable DTMF MPK send” must be set to “Yes” first.

Voice Mail

Select Account and enter the voicemail access number in the Value field.

Callback

Intercom

Select Account and enter the extension number in the Value field to perform intercom.

LDDAP search

The Base configuration in LDAP will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field.

For example: if users configure MPK 1 as “LDAP Search” for “Account 1”, and configure filters: .

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

Value -> sn=Li .

from the base for LDAP server configuration is .

“dc=89randstream,dc=com”, “ou=video,ou=SZ” is added to narrow the scope of the LDAP search. “sn=Li” is the example to filter the last name.

Call log

Select Account and enter the account number in the Value field to access the Call Log for that selected account.

Information

Select this function to display the Information popup window to show the firmware version, MAC address, IP address and IP configuration from the MPK buttons, no field is required for configuration.

Message

Select this function to display the Message menu from the MPK buttons, no field is required for configuration .

Account

Select the account to be used with a specific softkey.

Description

Enter a label for the softkey.

Value

Specifies the value of the softkey according to the selected mode.

Settings-> Web service

Use automatic location service

Configures to enable or disable automatic location services on the phone. (Restart required). The default setting is “Yes.”

Auto update services

Enables or disables automatic service update on the phone. This setting can enable automatic service update only when the Use automatic location service setting is enabled. The default value is “Yes”.

Settings -> XML applications

Route

Set the server path to download the XML file from the idle screen. This field can be either an IP address or a URL, with a maximum of 256 characters.

Programmable key label

Specifies the name of the softkey that is displayed on the idle screen for users to enter the XML application.

The default softkey label is “XMLApp”.

Default background color

Enter a color to use in HEX format. The default will be transparent.

For example, #000000. Reference: http://htmlcolorcodes.com

Block call screen

Allows to block automatic switching to call screen when XML application is running. The default value is disabled.

Settings-> Voice monitoring

Session report

VQ RTCP-XR session report

When enabled, the phone will send a session quality report to the central report collector at the end of each call. The default setting is disabled.”

Interval report

VQ RTCP-XR interval report

When enabled, the phone will send an interval quality report to the central report collector periodically during a call. The default setting is disabled.”

VQ RTCP-XR Periodic Interval Report

Set the interval (in seconds) of the phone sending an interval quality report to the central report collector periodically during a call. The default value is 20 seconds.

Alert Report

Warning threshold for MosIq

Set the threshold value of the MOS listening score (MOS-LQ) multiplied by 10. The MOS-LQ threshold value causes the phone to send a warning alert quality report to the central report collector. The valid range is 0 – 49.

The default value is “0”.

Critical threshold for MosIq

Set the threshold value of the MOS listening score (MOS-LQ) multiplied by 10. The MOS-LQ threshold value causes the phone to send a critical alert quality report to the central report collector. The valid range is 0 – 49.

The default value is “0”.

Warning threshold for delay

Set the one-way delay threshold value (in milliseconds) that causes the phone to send a warning alert quality report to the central report collector.

The valid range is 0 – 65535 The default value is “0”.

Critical threshold of delay

Set the threshold value of the one-way delay (in milliseconds) that causes the phone to send a warning alert quality report to the central report collector.

The valid range is 0 – 65535 The default value is “0”.

Show report

Show report on the user

When enabled, the phone will display a quality report in the web user interface.

The default setting is disabled”.

Show report on LCD display

When enabled, the phone will display a quality report on the LCD screen.

The default setting is “Disabled”.

Custom screen layout on the LCD

When enabled, the phone will display a quality report on the LCD screen.

Press the “Modify” button to select the information you want to display. The available/default options are:.

Start Time, End Time, Local User ID, Remote User ID, Local User IP.

Remote User IP, Local User Codec, Remote User Codec, Jitter, Jitter Buffer Max, Lost Packets, Network Packet Loss Rate, MOS-LQ, MOS-CQ, Round Trip Delay, End System Delay, Symmetric One-Way Delay.

What are the network page definitions in Grandstream GRP2612W?

Network -> Basic configuration

Internet Protocol

Select “IPv4 only”, “IPv6 only”, “Both, prefer IPv4” or “Both, prefer IPv6”.

The default setting is “IPv4 only”.

IPv4 address

Allows users to configure the appropriate network settings on the phone to obtain the IPv4 address. Users can 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 some ISPs.

DHCP provider class id (option 60)

Used by clients and servers to exchange provider class IDs.

The default setting is “Grandstream GRP2612W” for GRP2612W.

PPPoE account ID

Ienter the PPPoE account ID.

PPPoE password

Enter the PPPoE password.

PPPoE service name

Enter the PPPoE service name.

Ipv4 Address

Enter the IP address when using a static IP.

Subnet Mask

Enter the subnet mask when using static IP for IPv4.

Gateway

Enter the default gateway when using static IP for IPv4.

DNS Server 1

Enter DNS Server 1 when using static IP for IPv4.

DNS Server 2

Enter DNS Server 2 when using static IP for IPv4.

Preferred DNS Server

Enter the Preferred DNS Server for Ipv4.

IPv6 Address

IPv6 address type

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

Static IPv6 address

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

IPv6 prefix length

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

IPv6 Prefix

Enter the IPv6 prefix (64-bit) when using the static prefix in the “statically configured” IPv6 address type.

DNS Server 1

Enter DNS Server 1 for IPv6.

DNS Server 2

Enter DNS Server 2 for IPv6.

Preferred DNS Server

Enter the preferred DNS server for IPv6.

Network

-> Advanced configuration

802.1X mode

Allows the user to enable/disable 802.1X mode on the phone. The default is disabled. To enable 802.1X mode, this field should be set to EAPMD5, users can also choose EAP-TLS or EAP-PEAPv0/MSCHAPv2.

802.1X identity

Enter the Identity information for 802.1x mode.

Note: Letters, digits, and special characters, including @ and -, are accepted.

MD5 password

Enter the MD5 password for 802.1X mode.

Note: Letters, digits and special characters, including @ and -, are accepted.

AC 802.1X certified

Load/delete the CA 802.1X certificate on the phone; or delete the existing CA 802.1X certificate from the phone.

802.1X client certificate

Load/remove the 802.1X client certificate on the phone; or remove the existing 802.1X client certificate from the phone.

HTTP Proxy

Specifies the HTTP proxy URL for the phone to send packets. 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. The proxy server will act as an intermediary to route the packets to the destination. Omit proxy for Enter host names that do not require a proxy to access. These names must be separated by commas.

Layer 3 QoS for SIP

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

Layer 3 QoS for RTP

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

Enable DHCP VLAN

Enables automatic configuration for VLAN configuration via DHCP. Disabled by default.

Enable manual VLAN configuration

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

Layer 2 QoS 802.1Q/VLAN Tagging

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 Layer2 QoS packets. The valid range is 0 to 7. The default value is 0.

PC port mode

Set the PC port mode. When set to “Duplicate”, traffic on the LAN port will also pass through the PC port 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 PC port priority value. Valid range is from 0 to 7. The default value is 0.

Enable CDP

Enables/disables the CDP “Cisco Discovery Protocol. The default setting is enabled.”

Enable LLDP

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

Interval LLLDP transmission interval

Define theThe valid range is 1 to 3600. The default setting is “60”.

Maximum Transmission Unit (MTU)

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

Network -> CRemote Control

Action URI support

.
Enable/disable the action URI feature on the phone.

Default setting is enabled”.

Remote control pop-up support

Indicates whether the phone is enabled to allow remote control pop-up.

Default setting is enabled”.

Allowed IP list of action URIs

List of allowed IP addresses from which the phone receives action URIs. The

allowed IP addresses are separate by one comma , as

“192.168.1.1.1,192.168.1.2”. Set this field to “any” to allow any IP address to send action URLs to the phone. The default value is an empty string, which means that no IP address is allowed to control the phone remotely.

CSTA Control

Indicates whether the CSTA Control feature is enabled. Changing this setting will require a system reboot to take effect. The default setting is disabled.”

Network ->Configuration OpenVPN®

Enable OpenVPN®

Enable/Disable OpenVPN® . The default value is “No.”

OpenVPN® server address

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

OpenVPN® port

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

OpenVPN® Transport

Specify the OpenVPN®, transport type, either UDP or TCP. The default is “UDP.”

OpenVPN® CA

Click “Upload” to upload the OpenVPN® Certificate Authority. For a new upload, users can click “Delete” to delete the last certificate and then upload a new one.

OpenVPN® Certificate

Click “Upload” to upload the OpenVPN® server. For a new upload, users can click “Delete” to delete the last certificate and then upload a new one.

OpenVPN® Client Key

Click “Upload” to upload the OpenVPN®.

For a new upload, users can click “Delete” to delete the last certificate and then upload a new one.

OpenVPN® encryption method

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

Blowfish
AES-128
AES-256
Triple-DES

The default setting is “Blowfish”.

OpenVPN® username

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

OpenVPN® password

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

Additional options

Additional options to be added to the OpenVPN® configuration file, separated by semicolons. For example, comp-lzo no;auth SHA256 .

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

Network -> SNMP settings

Enable SNMP

Enable/disable the SNMP function. The default setting is “No”.

Version

Version of SNMP. Select either Version 1, Version 2 or Version 3.

default is “Version 3”.

Port

PortSNMP. Valid range is 161, 1025-65535. The default value is “161”.

Community

CommunitySNMP.

SNMP capture version

SNMP capture receiver capture version.

Trap version 1
Trap version 2
Trap version 3

The default is “Trap version 2”.

SNMP Trap IP

SNMP trap receiver IP address.

SNMP trap port

SNMP trap receiver port. Valid range is 162, 1025-65535. The default value is “162”.

SNMP Trap Range

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

SNMP capture community

Community string associated with the trap. Must match the community string of the trap receiver.

SNMP username

username for SNMPv3 .

Security level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.
authUser: Users with security level authNoPriv and context name as auth.
privUser: Users with security level authPriv and context name as priv.

Authentication protocol

Select the Authentication protocol:.

None
MD5
SHA

The default setting is “None”.

Privacy protocol

Select the Privacy Protocol:.

None
DES
AES

The default setting is “None”.

What are the maintenance page definitions in Grandstream GRP2612W?

Maintenance-> Web access

User Password

New Password

Set a new password to access the web GUI as a user. This field is case sensitive.

Confirm password

Enter the new user password again to confirm.

Administrator password

PasswordCurrent

The current administrator password is required to set a new administrator password.

New password

Set a new password to access the web GUI as administrator. This field is case sensitive.

Confirm password

Enter the new administrator password again to confirm.

Maintenance-> Updating and Provisioning

Updating and Provisioningfirmware update

Specifies how the firmware update and provisioning request should be sent: Always check for new firmware, Check for new firmware only when pre/suffix F/W changes, Always skip firmware check.

The default setting is “Always check for new firmware”.

Always Authenticate before the challenge

only applies to HTTP/HTTPS. If enabled, the phone will send the credentials before the server requests it. The default setting is “No”.

Validate host name in certificate

Allows to validate hostname in SSL certificate.

The default setting is “No”.

Allow option 43 and option 66 to override the server

The DHCP option 66 was originally only designed for the TFTP server. It was later extended to support an HTTP URL. GRP phones support both TFTP and HTTP server via option 66. Users can also use the vendor-specific option of DHCP option 43 to do this.

The DHCP option 43 approach has priorities. The phone can fall back to the original configured server route in case the option 66 server fails. The option 66 server fails.

The default setting is “Yes”.

Additional DHCP override option

When enabled, users can select option 150 or option 160 to override the firmware server instead of using the configured firmware server path or option 43 and option 66 server on the local network. Note that this option will be effective only when the “Allow DHCP option 43 and option 66 to override the server” option is enabled. The default setting is “None”.

Enable DHCP option 120 to override SIP Server

Enable local server DHCP option 120 to override the SIP server on the phone. The default setting is “No”.

Automatic 3CX Provisioning

Enables the automatic provisioning feature (PNP) on the phone when using 3CX as a SIP server. The default setting is “Yes.”

Actuautomatic provisioning

Enables automatic update and provisioning. The default setting is “No”.

Automatic random update

Updaterandomly within the range of hours of the day or postpone the update every X minute(s) by 1 to X minute(s) at random.

The default setting is “No”.

Time of day (0-23)

Defines the time of day to check the HTTP/TFTP/FTP server for firmware updates or configuration file changes. The default value is 1.

Day of the week (0-6)

Defines the day of the week to check the HTTP/TFTP/FTP server for firmware updates or configuration file changes. The default value is 1.

Disable authentication

device will not challenge NOTIFY with 401 when set to “Yes”. The default setting is “No.”

Confirmation firmware update

If set to “Yes”, the phone will prompt the user to update. If there is no response, the phone will continue with the update.

If set to “No”, the phone will automatically update without user intervention. The default value is Yes.

Config

Config Upgrade Via

Allows users to choose the configuration upgrade method: TFTP, FTP, FTPS, HTTP or HTTPS. The default setting is “HTTPS”.

Config Server Path

Defines the server path for provisioning.

Currently, the following variables are supported for the provisioning server URL: .

$PN: used to identify the name of the provisioning server directory where the boot files and corresponding configuration files are located.
$MAC: used to identify the MAC address of the IP phone.

$PN and $MAC variables can be embedded in the server URL configuration in the web UI and also in the DHCP option 66. Example (web user interface): /192.168.0.2/$PN_$MAC Example (DHCP option 66): tftp://192.168 .0.2/$PN_$MAC $PN will be replaced with the phone model, e.g., GRP2615 $MAC will be replaced with the MAC address of the phone, e.g., c074ad132f60 .

Config HTTP/HTTPS

User Name

The user name for the HTTP/HTTPS server.

Config HTTP/HTTPS

Password

The password for the HTTP/HTTPS server. Configuration file prefix Allows your ITSP to block configuration updates. If configured, only the configuration file with the matching encrypted prefix will be downloaded and displayed on the phone. Config File Postfix Allows your ITSP to lock configuration updates. If configured, only the config file with the matching encrypted suffix will be downloaded and displayed on the phone.

XML configuration file password

The password to encrypt the XML configuration file using OpenSSL. This is necessary for the phone to decrypt the encrypted XML configuration file.

Authenticate configuration file

Configures the phone system to authenticate the configuration file before applying it.

When set to “Yes”, the configuration file must include the P1 value with the phone system administration password. If it is lost or does not match the password, the phone system will not enforce it. The default setting is “No”.

Download configuration

Click to download the phone configuration file in .txt format.

Note: configuration file does not include passwords or custom CA/certificate.

Download device configuration (XML)

Click to download the phone configuration file in .xml format.

Note: the configuration file does not include passwords or custom CA/certificate.

User protection

When user protection is enabled, the provision or provider will not change the p-values that the user sets.

If “User protection” is disabled, everyone (provider, user or administrator) has access to most of the P-values.
If “User Protection” is enabled, only those (usually users or administrators) who have privileges can modify the settings.

The default setting is “Disabled”.

Download and process all available configuration files

By default, the device will provision the first available configuration in the order of cfgMAC, cfgMAC.xml, cfgMODEL.xml and cfg.xml (corresponding to device-specific, model-specific and global configurations).

If this option is enabled, the phone will reverse the download process to cfgg.xml > cfgGRP261x/GRP2624/GRP2634.xml > cfgMAC.bin > cfgMAC.xml. The following files will override the files that were already uploaded and processed.

The default setting is “No”.

Download user settings

This allows users to download part of the configuration that does not include any personal settings such as username and passwords. In addition, it will include all changes made manually by the user from the web user interface or the configuration file uploaded from .

“Load Device Configuration”, but will not include changes from server provisioning via TFTP/FTP/FTPS/HTTP/HTTP/HTTPS.

Load device configuration

Upload the configuration file to the phone.

Export backup package

Export backup package containing device configuration along with personal data.

Restore from backup package

Click to load the backup package and restore.

Firmware

Upgrade via Firmware

Allows users to choose the firmware update method:.

TFTP, FTP, FTPS, HTTP or HTTPS. The default setting is “HTTP”.

Firmware server path

Defines the server path for the firmware server.

Currently, the following variables are supported in the firmware server URL:.

$PN: used to identify the name of the firmware server directory where the boot files and corresponding configuration files are located.
$MAC: used to identify the MAC address of the IP phone.

$PN and $MAC variables can be embedded in the server URL configuration in the web UI and also in the DHCP option 66. Example (web user interface): /192.168.0.2/$PN_$MAC Example (DHCP option 66): tftp://192.168 .0.2/$PN_$MAC $PN will be replaced with the phone model, e.g. GRP2615 $MAC will be replaced with the phone MAC address, e.g. c074ad132f60 .

Firmware HTTP/HTTPS Username

The username for the HTTP/HTTPS server.

FirmwareHTTP/HTTPS Password

The password for the HTTP/HTTPS server.

Firmware file prefix

Allows your ITSP to block firmware updates. If configured, only firmware with the matching encrypted prefix will be downloaded and updated on the phone.

Firmware File Postfix

Allows your ITSP to block firmware updates. If configured, only firmware with the matching encrypted suffix will be downloaded and updated on the phone.

Maintenance-> Syslog

Syslog protocol

If set to SSL/TLS, system log messages will be sent via the secure TLS protocol to the system log server.

The default setting is “UDP”.

Note: CA certificate is required to connect to the TLS server.

Syslog server

The URL or IP address of the Syslog server to which the phone will send Syslog. Note: By adding the port number to the Syslog server field (i.e. 172.18.1.1:1000), the phone will send syslog to the corresponding port on that IP.

Syslog Level

Selects the logging level for syslog.

The default setting is “None”. There are 4 levels: DEBUG, INFO, WARNING and ERROR.

Syslog messages are sent based on the following events:.

Product model/version at startup (INFO level);
Information related to NAT (INFO level);
SIP message sent or received (DEBUG level);
SIP message summary (INFO level);
Incoming and outgoing calls (INFO level);
Registration status change (INFO level);
Negotiated codec (INFO level);
Ethernet link up (INFO level);
SLIC chip exception (WARNING and ERROR levels);
Memory exception (ERROR level).

Syslog keyword filtering

Syslog will be filtered based on the keywords provided. If you enter multiple keywords, it must be separated by ‘,’. Note that spaces are not allowed.

Send SIP Registration

Sets whether the SIP log will be included in the system log messages. The default setting is “No”.

Note: By setting Send SIP Log to Yes, the phone will still send SIP log from syslog even when the Syslog level is set to NONE.

Show Internet Down Message

If enabled, the Internet down warning message will be displayed when the Internet is down. The default setting is “No”.

Automatic recovery from anomalies

If set to “Yes”, the phone will automatically recover when operating abnormally. The default setting is “Yes”.

Maintenance-> TR-069

URL of ACS

Specifies the TR-069 ACS URL (e.g., http://acs.mycompany.com) or IP address. The default setting is “https://acs.gdms.cloud” .

Tr-069 user name

Name ACS user name for TR-069.

Tr-069 password

Password ACS password for TR-069.

Enable periodic reporting

Enable periodic reporting. If set to “Yes”, the device will send information packets to the ACS. The valid range is 1 – 4294967295. The default setting is “Yes”.

Periodic interval report

Configures the periodic reporting interval for sending information packets to the ACS. The default value is “86400”.

Connection request from user

The username for the ACS to connect to the phone.

Password connection request

The password for the ACS to connect to the phone.

Connection request port

The port for the ACS to connect to the phone. The default value is “7547”.

CPE SSL Certificate

The certificate file for the phone to connect to the ACS via SSL.

Private CPE SSL key

The certification key for the phone to connect to the ACS via SSL.

Random start TR069

When enabled, this option allows users to randomize the sending of TR069 INFORM packets. The default setting is disabled.”

Maintenance -> Security Configuration ->Security

Configuration via keyboard menu

Configure access control for users to configure from the keypad menu. There are three different options:

No restrictions:All options can be accessed from the keyboard menu.
Basic configuration only: the SIP option in the Phone submenu and the Network, Upgrade, UCM Detection and Factory Reset options in the System submenu will not be available in the LCD menu.
Restriction mode: the phone will require an admin password to change the Network, Upgrade and Factory Reset options in the System submenu, and also the SIP option in the Phone submenu.
Locked mode: Phone menu and MPK/VPK/Line switching are disabled.

The default setting is “No restrictions”.

Allow to configure MPK via LCD

Enable/disable MPK configuration via LCD by holding down the MPKs. This option is available only on GRP2614 and GRP2616. The default setting is “Yes”.

Validate certificates

After enabling this feature, the phone will validate the server’s certificate. If the server that our phone is trying to register to is not on our list, it will not allow the server to access the phone.

SIP TLS Certificate

SSL certificate used for SIP transport in TLS/TCP. SIP TLS private key SSL Private key used for SIP transport in TLS/TCP.

SIP TLS private key password

Private key password used for SIP transport in TLS/TCP.

Custom certificate

The uploaded custom certificate will be used for SSL/TLS communication instead of the default GRP phone certificate.

Web access mode

Sets the protocol for the web interface.

HTTPS
HTTP
Disabled
Both HTTP and HTTPS

The default setting is “HTTP”.

Enable web user access

The administrator can disable or enable user web access. The default setting is enabled.”

HTTP web port

Configures the HTTP port in HTTP web access mode. The valid range is 80 – 65535. The default value is “80”.

HTTPS web port

Set the HTTPS port in HTTPS web access mode. The valid range is 443 – 65535. The default setting is “443”.

Access control web

Web access control allows you to choose whether to follow a whitelist or blacklist scheme for choosing who to allow access to the web user interface.

Web Access Control List

The Web Access Control List will list the IP addresses it allows or disallows depending on what you have selected in Web Access Control.

Disable SSH

Disable SSH access. The default setting is “No”.

SSH public key

This option allows you to use authentication keys for SSH access. The public key must be uploaded to the phone’s web UI, while the private key must be used on the SSH tool side.

Note: This will enable the next SSH access without password.

Web/keyboard/restricted mode lockout duration

Specifies the time in minutes that the web or LCD login interface will be locked for the user after five login errors. This lockout time is used for web login, STAR keypad unlock, and LCD restricted mode administrator login. The range is 0 to 60 minutes. The default setting is “5”.

Web session expired

Set the timer to close the web session during inactivity. Valid range is 2-60 min.

Default value is 10 min .

Web access limit

Set the limit of attempts before lockout. The valid range is from 1 to 10. The default value is “5”.

Maintenance-> Security settings -> CA certificates of trust

Trusted CA Certificates (1 to 6)

Allows you to upload and delete the CA certificate file on the phone.

Note:users can upload the file directly from the web or they can choose to provision it from their cfg.xml file.

Load CA certificates

The phone will verify the server certificate based on the list of built-in, custom or trusted certificates. The default setting is “Default certificates”.

Maintenance -> Security Settings -> Keypad Lock

Enable Keyboard lock

If set to “Yes”, the keyboard can be locked manually by pressing for 4 seconds the * key or pressing a VPK/MPK which is set to “keyboard lock” mode, also the keyboard will be locked automatically after the set timer. The default setting is “No”.

Keyboard lock type

If set to “Functional Keys“, only “Functional Keys” will be locked, but you will still be able to make emergency calls.
If set to “All keys“, all keys will be blocked and emergency calls will not be allowed.

The default setting is “All keys”.

Password for lock/unlock

Set the password to unlock the keyboard.

Keyboard lock timer

Set the idle screen timer after which the keyboard will automatically lock. The valid range is 0 – 3600. The default value is “0”.

Emergency

Enter the list of allowed emergency numbers when the keypad is locked (separate the numbers with “,”). The default value is “112,911,110”.

Maintenance-> Capture of packets

Status

Displays the packet capture status. When the user starts capturing the trace file, it will show the status “RUNNING”, otherwise it will show “STOPPED”.

With RTP packets

Defines whether the packet capture file contains RTP or not. The default setting is “No”.

What are the page directory definitions

Directory-> Contacts

Search bar

Allows users to search for entries in the phone book.

Add contact

Specify contact first name, last name, phone number, account and group blacklist, whitelist, work, friends and family) to add a new contact in the phonebook.

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

Edit contact

Edit the selected contact.

Delete all contacts

Delete all contacts from the phonebook.

Note: a message will be displayed for users to confirm whether to delete or cancel the operation, in order to prevent users from losing contacts by accidentally deleting them.

Directory-> Group management

Add group

Specify the group name to add a new group. More than 30 groups supported.

Edit group

Edit the selected group.

Directory-> Phonebook management

Enable Download XML phonebook

Set to enable XML download of the phonebook. Users can select HTTP/HTTPS/TFTP to download the phonebook file.

Default setting is disabled”.

HTTP/HTTPS username

The username for the HTTP/HTTPS server.

HTTP/HTTPS password

The password for the HTTP/HTTPS server.

Phonebook XML path

Set the server path to download the phonebook XML.

This field can be an IP address or a URL, with a maximum of 256 characters.

Phonebook download interval

Set the phonebook download interval (in minutes).

If set to 0, the automatic download will be disabled. The default value is 0. The valid range is 5 to 720 minutes.

Delete manually edited entries when downloading

If set to “Yes”, when downloading the XML phonebook, manually added entries will be deleted automatically.

The default setting is “Yes”.

Import group method

When set to “Replace“, the existing groups will be completely replaced by an imported one;
When set to “Attach“, imported groups will be serviced with the current one.

The default setting is “Replace”.

Sort phonebook by

Sort phonebook by first or last name selection.

The default setting is “Last name”.

Download phone directory

Click “Download” to download the XML phone directory file to the local PC.

Upload phone directory

Click “Upload” to load the local XML phone directory file to the phone.

Phonebook key function

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

Default Search

Sets the default phonebook search mode.

Quick match: the quick search feature allows users to search for parts and strings of entries. For example, if users only remember the first name, last name or parts of the first name/phone number, they can use the string in the search bar.
Exact match: users can search for their contacts using alphabets in exact mode, allowing them to find their contacts even if they forget the numbers. To perform this type of search, make sure the search type is set to “Exact Match”, then you can enter the exact name of the contact to search.

The default setting is “Quick Match”.

Directory-> Call history

Delete

Users can select an entry and then click “Delete” to remove it from the list.

Delete All

Click Delete All to delete all call history stored on the phone.

Note: Users can use the drop-down list to display only the selected call history type (All, Answered, Dialed, Missed, Transferred) and also use the navigation keys to navigate through the pages when there are many entries.

Directory -> LDAP

LDAP protocol

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

Server address

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

Port

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

Base

Configures the LDAP search base.

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

Example:

dc=grandstream, dc=com ou=Boston, dc=grandstream, dc=com .

User name

Set the “Username” link to query LDAP servers. Some LDAP servers allow anonymous links, in which case the setting can be left blank. .

Password

Set the “Password” link to query LDAP servers. The field can be left blank if the LDAP server allows anonymous links. .

LDAP number filter

Configures the filter used for number lookups.

Examples:

(|(phone number=%)(Mobile=%) returns all records that have the “Phone Number” or “Mobile” field beginning with the entered prefix; (&(Phone Number=%) (cn=*)) returns all records with the .

“Phone number” field starting with the entered prefix and the set “cn” field.

LDAP name filter

Configures the filter used for name lookups.

Examples:

(|(cn=%)(sn=%)) returns all records that have the field “cn” or “sn” beginning with the entered prefix;

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

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

LDAP version

Select the protocol version (Version 2 or Version 3) for the phone to send binding requests. The default setting is “Version 3.” .

LDAP name attributes

Specifies the “name” attributes of each record that are returned in the LDAP search result. This field allows users to configure multiple name attributes separated by spaces.

Example:

gn cn sn description .

LDAP number

Specifies the “number” attributes of each record that are returned in the .

Attributes

LDAP.

This field allows users to configure multiple attributes of space-separated numbers.

Example:

phone number cell phone number .

Show LDAP name

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

Example: .

%cn %sn %phone number

max. hits

Specifies the maximum number of hits to be returned by the LDAP server. If set to 0, the server will return all search results. The default setting is 50.

Search timeout

Specifies the interval (in seconds) for the server to process the request and the client waits for the server to return. The default setting is 30 seconds.

Sort results

Specifies whether the search result is sorted or not. The default setting is “No.” .

LDAP search

Set to enable LDAP number lookup when dialing and receiving calls.

Search display name

Sets the display name when LDAP looks up the name for an incoming call or an outgoing call. This field must be a subset of the LDAP name attributes.

Example: .

cn sn description

Exact match search

With LDAP search Incoming call, Outgoing call selected, DUT will perform LDAP Search during incoming and outgoing calls. If exact match search enabled, during LDAP search, DUT will only get the result that exactly matches the search entry. i.e. if 100 is just the incoming/outgoing number 100 will be searched, *100* will not. The default value is “disabled”.

What are the BLF LED patterns?

Pattern: Default		Pattern: Analog
Call status	Light indication	Call status	Luminous indication
Disconnected	Disabled	Off	Disabled
Inactive	Continuous green	Inactive	Green
Fixed Testing	Fixed Red	Testing	Red
Speaking	Solid Red	Speaking	Red
Proceeding	Flashing red	Proceeding	Red
Incoming call	Flashing red	Incoming call	Flashing red

Pattern: Directional		Mode: Inverse
Call status	Light indication	Call status	Luminous indication
Disconnected	Disabled	Off	Disabled
Inactive	Continuous green	Inactive	Red
Intenting	blinking green	Testing	Green
Speaking	Solid Red	Speaking	Green
Proceeding (Initiator)	Flashing green	Proceeding	Flashing Green
Proceeding (Receiver)	Flashing Red	Incoming call	Flashing green
Incoming call	Flashing red

Out .Off (Extension Card Icon: Off) .
Inactive .

Extension plate icon.

Mode: Reserved (Red)		Mode: Reserved (Green)
call status	Light indication	Call status	Luminous indication
Disconnected	Disabled (Extension board icon: Disabled)
Inactive	Off (Extension card icon: Inactive)
Testing	fixed red	Testing	Green
Speaking	Solid Red	Speaking	Green
Proceeding	Solid Red	Proceeding	Green
Incoming call	Flashing red	Incoming call	Flashing green

How to configure GRP2612WNAT.

If the devices are kept within a private network behind a firewall, we recommend using the STUN server. The following settings are useful in the STUN server scenario:.

STUN server

In Settings -> general, enter a STUN server IP (or FQDN) that you may have, or search for a free public STUN server on the Internet and enter it in this field. If you are using a public IP, leave this field blank.

Use random ports

It is in Settings🡪General Settings. This setting depends on your network configuration. When set to “Yes”, it will force random generation of local SIP and RTP ports. This is usually necessary when multiple GRPs are behind the same NAT. If you are using a public IP address, set this parameter to “No”.

Cross NAT

It is in Accounts X🡪Network Settings. The default setting is “No.” Enable the device to use NAT traversal when behind a firewall on a private network. Select Keep-Alive, Auto, STUN (with the STUN server path configured as well) or other option depending on the network configuration..

How to configure click-to-dial on Grandstream GRP2612W?

From GRP2612W Web GUI, users can dial with Click-to-Dial feature at the top of the Web GUI.

Before using the Click-to-Dial feature, make sure that the “Click-to-Dial Feature” option in the GUI-> Settings -> Call Feature sare enabled. If there is no account registered, the icon will be grayed out ; If clicking to check is disabled, but the account is registered, the icon will be green and clicking on the icon will do nothing.

On clicking the icon in the top menu of the Web GUI, a new dial window will appear for you to enter the number. Once Dial is clicked, the phone will go off-hook and dial the selected account number.

In addition, users can directly send the command for the phone to dial by specifying the following URL in the PC web browser, or in the field as required in other calling modules.

http://dirección_ip/cgi-bin/api-make_call?phonenumber=1234&account=0&login=admin&password=admin .

In the link above, replace the fields with .

ip_address:

Phone IP address.

phone number =1234:

The number for the phone to dial.

account=0:

The account index of the phone to make the call. The index is 0 for account 1, 1 for account 2, 2 for account 3, etc.

password=admin/123:

The administrator’s login password or the phone’s web GUI user’s login password.

How to configure notification options on Grandstream GRP2612W?

Outgoing notification options can be found in the device’s web UI🡪Settings -> NOutgoing Notifications. In the web UI, there are three sections under Outgoing Notifications: “Action URL”, “Destination” and “Notification”.

Action URL

To use the outbound notification action URL, users must know the supported events and the dynamic variables for the supported events. The dynamic variables for supported events will be replaced with the actual values on the phone to notify the event to the SIP server.

Supported Events

- Configuration completed
- Completed call
- Registered
- Open DND
- Not registered
- Close DND
- Off-Hook
- Open Forward
- Hang up
- Close Forward
- Blind transfer of incoming calls
- Call forwarding
- Outgoing call transfer attended
- Missed call
- Call waiting
- Call established
- Call not held

Dynamic Variables Supported

Dynamic Variable	Description
$phone_ip	The IP address of the phone
$mac	The MAC address of the phone
$product	The product name of the phone
$program_version	The software version of the phone
$hardware_version	The hardware version of the phone
$language	The display language of the phone
$local	The number called on the phone
$display_local	$display_local	The display name of the called number on the phone
$remote	$remote	The number to call on the remote phone
$display_remote	$display_remote	The display name of the number to call on the remote phone
$active_user	The account number during a call on the phone

After the user finishes configuring the action URL in the phone’s web UI, when the phone-specific event occurs on the phone, the phone will send the action URL to the specified SIP server. The dynamic variables in the action URL will be replaced by the actual values. Here is an example: .

Configure the following action URL in the phone’s web UI🡪Configuration -> Notification -> Action URL: .

Incoming call: 172.18.24.103/mac=$mac&local=$local .

Outgoing call: 172.18.24. 103/remote=$remote&phone_ip=$phone_ip Standby: 172.18.24.103/program_version=$program_version

During incoming call, outgoing call and call waiting, capture the trace on the phone and examine the packets. We can see that the phone sends action URLs with real values to the SIP server to notify the phone events. In the following screenshot, from top to bottom, the phone events for each HTTP message are: Incoming Call, Outgoing Call and On Hold in the format of the defined action URL with the parameters replaced by actual values.

The P-values listed in the following table are for options in the phone’s web UI🡪Configuration -> Notification -> Action URL.

Action URL P-value parameters

P P-Value	Web UI option	Format
P8304	Configuration complete	Chain
P8305	Registered
P8306	P8306		Unregistered
P8308	P8308		Unregistered
P8309	Hanging Call
P8310	Incoming call
P8311	Outgoing call
P8312	Missed call
P8313	P8313		Call set
P8314	Call terminated
P8316	Open DND
P8317	Close DND
P8318	P8318		Send open
P8319	Send closed
P8320	Blind transfer
P8321	P8321		Transfer attended
P8324	Call waiting
P8325	Resume Call

Destination

The options in the phone web UI🡪Settings -> Notification Outgoing -> Destination configures the destination of the outgoing notification server information. Click “Add Destination” and users will see the following window to configure the destination server information.

The following table describes each option in the above interface.

Add Action URL Destination Settings

Destination Server Option	Description
Destination name	Identify the destination name. It must be unique.
Protocol	Configure the protocol associated with the target server. Currently XMPP and SMTP are supported.
Enable SSL	Configure whether to use SSL to encrypt for the SMTP protocol. This option is not editable for XMPP.
Address	destination Set the address of the destination server, e.g. talk.google.com.
Port	Set the port of the destination server, for example, 5222.
Domain	Configure the destination server domain for the XMPP protocol. This option is not editable for SMTP.
User name	Set the authorization user name of the target server.
Password	Set the password of the authorization user for the target server.
From	Set the sender name for the SMTP protocol. This option is not editable for XMPP.
For	Set receiver address.
Name	additional attribute name Set the additional attribute name reserved for protocol-specific attributes, such as “jid” for XMPP protocol. If “jid” is specified, the username and domain will be overridden.
Value	of additional attribute Set the additional attribute value reserved for protocol-specific attributes, such as “[email protected]” for “jid” of the XMPP protocol. If specified, the username and domain will be overridden.

Up to 10 destinations can be configured here. The P values are listed in the following table. .

Action URL values

P	destination	P value Format
P9910	from target 1	. Each P value consists of all the options configured for this destination. . Example 1: destination 1 with XMPP protocol and 2 additional attributes configured: . P9910=serverName=destination1&protocol=XMPP&serverAddress=talk. google. com&port=5222&user=user1&password=password1&from=. &to=to1&domain=gmail. com&extraAttrName1=extraAttrValue1&extraAttr. Name2=extraAttrValue2
P9911	Destination 2
P9912	Destination 3
P9913	Destination 4
P9914	Destination 5

P9915	Destination 6	Example 2: destination 2 with SMTP protocol and 3 additional attributes configured: P9911=serverName=destination2&protocol=SMTP&serverAddress=smtp s://smtp. gmail. com&port=465&user=username2&password=password2&from=username2&to=to2&domain=&extraAttrName1=extraAttrValue1&e xtraAttrName2=extraAttrValue2&extraAttrName3=extraAttrValue3 . The highlighted strings in the examples above are the actual values set in each field for the destination.
P9916	Destination 7
P9917	Destination 8
P9918	Destination 9
P9919	Destination 10

Notification .

After configuring the target server, users can configure the notification information in the phone’s web UI🡪Settings -> Notification Outgoing ->Notification. Click on “Add Notification” and users will see the following window to set up the notification.

Action URL Notification Options

Option	Description
Event	Set the event, which will trigger an outgoing notification.
Destination	Configures the name of the destination to which the outbound notification will be sent.
Subject	Set the subject of the email notification. This option is only applicable to the SMTP protocol and is not editable for other protocols.
Message	Configures the message body or outgoing notification.
Namere of additional attribute	Set the additional attribute name reserved for specific attributes for a given notification in the future
Vadditional attribute value	Sets the additional attribute value reserved for specific attributes for a given notification in the future.

The message body of each event’s notification can be customized with embedded dynamic attributes.

The following table shows the mapping between event and dynamic attribute.

Action URL notification: events and attributes

.is deviated.
detour.calls fwReason Reason .

Event	Name	Dynamic Description
Call_Missed	line	Line number associated with the call account
	Number	of account associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name of the local part
	sipServer	The SIP server address of the account
	call-id	The SIP call ID dialog
	time	The timestamp when the missed call event occurs
DND	status	This is for the DND status. The value can be “enabled” or “disabled”
. Call_Forward	callType	This is for the call type. The value can be “incoming” or “outgoing”
	inline	associated with the call account
	Number	account number associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name
	sipServer	The SIP server address of the account
	call-id	The SIP call ID dialog box
	time	The time stamp when the
	fwNumber	fwNumber	destination number
	of the
OAM_Login	OAMUser	OAMUser name as “admin”
OAM_Login	OAMLoginSource	OAMLoginSource	OAMLoginSource. The value can be “SSH” or “WebGUI”


	OAMLoginFromIP	OAMLoginFromIP address. The value is the IP address of the PC that will log in to the phone’s web UI or additional attribute
	OAMLoginCode	OAMLogin result codeValue can be . “success” or “failed”
	time	OAM
OAM_Lockout	OAMUser	OAMUser name as “admin”
	OAMLoginSource	OAMLoginSource	OAMLoginSource. The value can be “SSH” or . “WebGUI”
	OAMLoginFromIP Start of	OAMLoginFromIP Start of	OAM session From IP address. The value is the IP address of the PC that will log in to the phone’s web user interface or
	OAMLockoutCode	OAMLockout result codeValue can be . “locked” or “unlocked”
	OAMLockoutTime Mark	OAMLockoutTime
Incoming_CallcallingNumber	callType	calling party
	Number	Call type. The value can be “incoming” line or . “outgoing”
	inline	associated with the call account
	Number	account number associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name
	sipServer	The SIP server address of the account
	call-id	The SIP call ID dialog box
	time	The time stamp when the incoming call event occurs
Outgoing_Call	callType	callType. The value can be “incoming” line or . “outgoing”
	inline	associated with the call account
	Number	account number associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name


	hour	The timestamp when the outgoing call event occurs
Call_Established	callType	Type of the call. The value can be “incoming” line or . “outgoing”
	inline	associated with the call account
	Number	account number associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name
	sipServer	The SIP server address of the account
	call-id	call ID of the SIP dialog
	startTime	startTime	The timestamp when the outgoing call event occurs
Call_Terpressed	callType	callType. The value can be “incoming” line or . “outgoing”
	inline	associated with the call account
	Number	account number associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name
	sipServer	The SIP server address of the account
	call-id	call ID of the SIP dialog
	startTime	startTime	The timestamp when the call is established
Call_Forward_Status	duration	The
	account	The account number associated with the status change
	forwardNumberAll	forwardNumberAll	for Call ForwardAll
	forwardNumberBusy	forwardNumberBusy	forwardNumberBusy for Call Forward Busy
	forwardNumberNoAns	forwardNumberNoAns	ForwardNumberNoAns for Call Forwarding No Answer
Retrieve	calls	Call type Call type. The value can be “incoming” line or . “outgoing”
	inline	associated with the call account
	Number	account number associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber


	localName	The name
	sipServer	The SIP server address of the account
	call-id	The SIP call ID dialog
	startTime	The time stamp when the call is on hold
Call_Resume	callType	callType. The value can be “incoming” line or . “outgoing”
	inline	associated with the call account
	Number	account number associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name
	sipServer	The SIP server address of the account
	call-id	call ID of the SIP dialog
	startTime	startTime	The timestamp when the call resumes
Blind_Transfer	line	Line number associated with the caller’s account
	Number	of account associated with the call
	remoteNumber	remoteNumber
	remoteName	remoteName
	localNumber	localNumber
	localName	The name of the local part
	sipServer	The SIP server address of the account
	call-id	The SIP call ID dialog
	time	The timestamp when the call is transferred transferName
	Name	part
	transferred	transferNumber Number of the transferred part
Line	of	line associated with the caller’s account
	Number	account associated with the call
	remoteNumber	remoteNumber
	remoteName	Th e remote party name
	localNumber	localNumber
	localName	The name
	sipServer	The SIP server address of the account
	call-id	The SIP call ID dialog
	Time	The timestamp when the call is transferred transferName
	Name	part
	transferred	transferNumber Number of the transferred part
Register_Status	registerStatus	Account registration status. The value can be “registered” or “unregistered”
Bootup_Complete	N/A	N/A
The dynamic attributes in this row are common attributes that can be applied to all events	mac	phone MAC address
	phone_ip	IP address
	program_version	software version of the phone
	hardware_version	hardware_version of the
	product	phone language product name
	Language	Language	display on phone

All of the above dynamic attribute values are generated by the phone system and can be used as dynamic attributes with a pair of curved braces around them. For example, if the message body is specified as follows: Your call from {remoteName}:{remoteNumber} to {localName}:{localNumber} was forwarded to {fwdNumber} for reason {fwdReason}. .

Then, the message received in the outgoing notification will look like this:

Your call from Daniel:2070 to Jasmine:2071 was forwarded to 777777 for unconditional reason.

Only the attributes between curly braces will be replaced by the runtime value. The rest of the content will remain the same as the static text.

For each event a maximum of 3 notifications can be configured. In total, up to 75 notifications can be configured. The P-value for each notification is listed in the following table.

Action URL Notification P-values

P-value

Format

P9920

of notification 1

Each P-value consists of all the options configured for this notification.

Example 1: notification 1 for the event “Call_Missed” to destination 1, with 2 additional attributes configured:

P9920=eventName=Call_Missed&destName= destination1&subject=&msg=.

You have a missed call of {remoteName}:{remoteNumber} on line {line} ,

count {count} at .

{time}. &extraAttrName1=extraAttrValue1&extraAttrName2=extraValue2 .

Example 2: notification 2 for the event “Incoming_call” to destination 2, with 2 additional attributes configured: P9921=

eventName=Incoming_Call&destName= destination2&subject=Incoming .

Call Alert&msg=You have a {callType} call from

P9921

Notification 2

P9922

Notification 3 P9923

Notification 4 P9924

Notification 5 P9925

Notification 6 P9926

Notification 7 P9927

Notification 8

P9928 .

P9929

P9993

P9994

Notification 9.

Notification 10

Notification 73

Notification 74

{remoteName}: {remoteNumber} in line {line}, count {account} in .
an style=”font-weight: 400;”> {time}. &extraAttrName1=extraAttrValue1&extraAttrName2=extraAttrValue2 .

The highlighted stringsin the examples above are the actual values set in each field for the notification.

P9995

Notification 75

How to perform the Grandstream GRP2612W formware upgrade?

Grandstream GRP2612W can be upgraded via TFTP / FTP / FTPS / HTTP / HTTPS by setting the URL/IP address for the TFTP / HTTP / HTTPS / FTP / FTPS server and selecting a download method. Set a valid URL for TFTP, FTP/FTPS or HTTP/HTTPS, the server name can be FQDN or IP address.

Examples of valid URLs: firmware.grandstream.com/BETA fw.mycompany.com

There are two ways to configure a software update server: the LCD keypad menu or the web configuration interface.

How to upgrade Grandstream GRP2612W via the keypad menu?

Follow the steps below to configure the upgrade server path via the phone’s keypad menu:.

Press the MENU button and navigate using the up/down arrows to select System.
In the system options, select Upgrade.
Enter the path to the firmware server and select the upgrade method. The server path could be in IP address format or in FQDN format.
Enter the firmware server path and select the upgrade method.
Select Start Provisioning and press the “Select” softkey.
A warning window will appear to confirm the provisioning. Press the “YES” softkey to start updating/provisioning immediately.

When the update starts, the display will show the progress of the update. When it is finished, you will see the phone reboot again. Do not interrupt or turn the phone off and on when the update process is in progress.

How to upgrade Grandstream GRP2612W through the Web GUI?

Open a web browser on the PC and enter the IP address of the phone. Then log in with the administrator’s username and password. Go to the Maintenance🡪Upgrade and Provisioning page, enter the IP address or FQDN for the upgrade server in the “Firmware server path” field and choose to upgrade via TFTP or HTTP/HTTPS or FTP/FTPS. Update the change by clicking the “Save and Apply” button. Then “Restart” or power off and power on the phone to update the new firmware..

When the update starts, the screen will show the progress of the update. When it is finished, you will see the phone reboot again. Do not interrupt or turn the phone off and on when the update process is in progress. When the update is complete, you will see the phone restart again.

The firmware update takes about 60 seconds on a controlled LAN or 5 to 10 minutes over the Internet. We recommend completing firmware upgrades in a controlled LAN environment whenever possible.

How to upgrade Grandstream GRP2612W sin local TFTP/FTP/HTTP/HTTP servers?

For users who wish to use remote upgrade without a local TFTP/FTP/HTTP/HTTP server, Grandstream offers a NAT-compliant HTTP server. This allows users to download the latest software updates for their phone through this server. Please refer to the web page:

https://www.grandstream.com/support/firmware

Alternatively, users can download a free TFTP, FTP or HTTP server and perform a local firmware upgrade. A free window TFTP server version is available for download from: http://www.solarwinds.com/products/freetools/free_tftp_server.aspx

http://tftpd32.jounin.net/.

Instructions for local firmware upgrade via TFTP:.

Unzip the firmware files and place them in the root directory of the TFTP server.
Connect the PC running the TFTP server and the phone to the same LAN segment.
Start the TFTP server and go to the File menu🡪Configure🡪Security to change the default TFTP server settings from “Receive Only” to “Transmit Only” for firmware update.
Start the TFTP server and configure the TFTP server in the phone’s web configuration interface.
Configure the firmware server path to the IP address of the PC.
Update the changes and reboot the phone.

End users can also choose to download a free HTTP server from http://httpd.apache.org/or use the Microsoft IIS web server.

How to manage the Grandstream GRP2612W Configuration Files?

Where to download the Grandstream GRP2612W configuration file?

Grandstream SIP devices can be configured through the web interface as well as through a configuration file (binary or XML) via TFTP, FTP/FTPS or HTTP/HTTPS. The “Configuration Server Path” is the path of the TFTP, FTP/FTPS or HTTP/HTTPS server for the configuration file.

Must be set to a valid URL, either in FQDN format or IP address. The “Configuration Server Path” can be the same or different from the “Firmware Server Path.” .

A configuration parameter is associated with each particular field on the web configuration page. A parameter consists of a capital letter P and numeric numbers from 2 to 5 digits. i.e. P2 is associated with the “New password” in the web GUI🡪Maintenance-> Web login page -> Password. For a detailed list of parameters, please refer to the corresponding configuration template..

When Grandstream GRP2612W boots or reboots, it will issue a request to download an XML file named “cfgxxxxxxxxxxxxxxxxxxxxxxxx.xml”, where “xxxxxxxxxxxxxxxx” is the MAC address of the phone, i.e. “cfg000b820102ab” and “cfg000b820102ab .xml”. If the download of the “cfgxxxxxxxxxxxxxxxxxxxx.xml” file is not successful, followed by a configuration file named “cfgxxxxxxxxxxxxxxxxxxxx”, the phone will issue a request to download a model specific configuration file “cfg<model>.xml”, where <model> is the phone model, i.e. “cfggrp2613.xml” for the GRP2613, “cfgrp2614” for the GRP2614. If this file is not available, the phone will issue a request to download the generic “cfg.xml” file. The name of the configuration file must be in lower case. If not found, the phone will request a file named “dev[MacAddress].cfg” where “MacAddress” is the MAC address of the device. With this provision file, users can to provision the device with P-values and aliases;.

Values must be placed between lines beginning with a ‘#’ in order to provision. Lines beginning with a ‘#’ have their data ignored and can therefore be used as comments. For example:.

account.2.name=2225 P407=114

Note: (Try downloading the configuration file again)

When provisioning the phone, if your first configuration file contains the p-values listed below, the phone will attempt to download the possible second cfg.xml file and apply the second file without restarting. Maximum 3 extra attempts..

These P values are:.

*212 — Configuration update via.

*234 — Configuration prefix

*235 — Configuration postfix

*237 — Configuration update server *237 — Configuration update server

*240 — Authenticate configuration file *240 – Authenticate configuration file

*1359 – XML configuration file password *1359 — XML configuration file password

*8463 – Validate Certified server

*8467: download and process ALL available configuration files

*20713: always authenticate before the challenge

*22011: omit proxy for

*22030: enable SSL host verification for provisioning

Note: (P values that enable automatic provisioning)

If the p-values listed below are changed while managing the configuration in the web UI or LCD, the provisioning process will be triggered:.

192 — Firmware update server
232 — Firmware prefix
233 — Postfix
6767 — Firmware upgrade via
of 6768 — Firmware HTTP/HTTPS username
6769 — Firmware HTTP/HTTPS password
237 — Configuration update server
212 — Configuration update via
of 234 — Configuration prefix
235 — Configuration postfix
1360 — HTTP/HTTPS configuration username
1361 — HTTP/HTTPS configuration password.

Note: Certificate and key provisioning

Users can configure GRP2612W to obtain all necessary certificates during startup. Instead of putting the certificate/key content in text directly from the web interface or uploading it manually, they can choose to provision it from the configuration file by putting the URL in the Pvalue field of each certificate and/or key. (e.g. http://ProvisionServer_address/SIP-TLS-Certificate.pem) Then, the phone will process the URL, search for the appropriate certificate/key file, download it and then apply it to the phone.

For more details on XML provisioning, see: https://www.grandstream.com/hubfs/Product_Documentation/gs_provisioning_guide.pdf?hsLang=en

How to perform contactless provisioning on Grandstream GRP2612W?

After the phone sends, set up the file request to Broadsoft provisioning server via HTTP/HTTPS, if the provisioning server responds “401 unauthorized” requesting authentication, the phone LCD screen will display a window for the user to enter the username and password. Once the correct username and password are entered, the phone will send a configuration file request again with authentication. Then, the phone will receive the configuration file to download and automatically provision itself..

In addition to manually entering the username and password on the LCD display, users can also save the login credentials for the provisioning process. The username and password settings can be found in the phone’s web user interface🡪Maintenance -> Update and Provisioning page: “HTTP/HTTPS Username” and “HTTP/HTTPS Password”. If the saved username and password are correct, the login window will be skipped. Otherwise, a login window will appear to prompt users to re-enter the correct username and password.

What is the update and provisioning shortcut via the keyboard menu in Gransdtream GRP2612W?

When the GRP phone is in idle state, the user can press the HOLD key and the RIGHT navigation key together to activate the provisioning functions. Similarly, the phone will display a reboot banner while idle if the user presses the HOLD key and the LEFT navigation key at the same time. After the provisioning or reboot banner appears on the LCD, the user can press the YES/NO softkey to confirm/cancel the action.

What tools does the Grandstream GRP2612W contain?

From the web GUI under Maintenance🡪 Tools, 4 tools are provided:

Provisioning

Makes the phone trigger an instant provisioning.

Factory Reset

Resets the phone to factory default settings.

Warning:

resetting to factory default settings will delete all configuration information on the phone. Back up or print all settings before restoring factory default settings. Grandstream is not responsible for restoring lost settings and may not connect your device to your VoIP service provider.

Ping

Hcauses the phone to ping a URL to verify if it has access to it.

Traceroute

Checks the path taken by packets to the specified URL.

How to restore to Grandstream GRP2612W factory default settings?

Warning:

Restoring to factory default settings will delete all configuration information on the phone. Back up or print all settings before restoring factory default settings. Grandstream is not responsible for restoring lost settings and may not connect your device to your VoIP service provider.

How to restore Grandstream GRP2612W factory settings using the LCD menu?

Follow the instructions below to reset the phone:.

Press the MENU button to open the keypad settings menu.
Select “System” and enter.
Select “Operations -> Factory Reset.”
Select “Operations -> Factory Reset.
A warning window to make sure that a reset is requested and confirmed.

Press the “Yes” soft key to confirm and the phone will reset, or the “No” soft key to cancel the reset.

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So far everything you need to know to Configure GRP2612W, technical administrator level configuration manual. If you are interested in continuing to learn how to configure more Grandstream models or other brands of handsets, be sure to follow the manuals section of our blog!

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