GRANDSTREAM NETWORKS GXP1760 (01) PDF MANUAL

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What are the ways the GXP1760/GXP1760W/GXP1780/GXP1782 can be configured?

The GXP1760/GXP1760W/GXP1780/GXP1782 can be configured via two ways:

  • LCD Configuration Menu using the phone’s keypad.
  • Web GUI embedded on the phone using PC’s web browser.

How do I configure the LCD menu using the phone’s keypad?

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

  • Enter/Confirm selection: Press the round MENU button or “Select” softkey to enter the selected option.
  • Back: Press “Back” softkey to exit to the previous menu.
  • Return to Home page: In any menu, press “Back” softkey to return to idle screen.

What happens if the phone is in MENU mode?

The phone automatically exits MENU mode in these situations:

  • An incoming call when the phone is off hook
  • The MENU mode if left idle for more than 60 seconds.

What options are available in the Call History sub menu?

The Call History sub menu includes the following options:

  • Local Call Log: Displays answered calls/dialed calls/missed calls/transferred calls. Press “Clear All” to clear all local call history.
  • Broadsoft Call Log: Displays Broadsoft call logs.

What options are available in the Status sub menu?

The Status sub menu includes the following options:

  • Network status: Displays the MAC address, IP information (DHCP/Static IP/PPPoE), IPv4 address, IPv6 address, Subnet Mask, Gateway, and DNS server.
  • Account status: Shows the registration status of each account.
  • System Status: Displays the hardware version, part number, software version, IP geographic information and special features.

What options are available in the Phone Book sub menu?

The Phone Book sub menu includes the following options:

  • Local Phonebook
  • Local Group
  • Broadsoft Phonebook
  • LDAP Directory

User may configure phonebooks/groups/LDAP options here, transfer phonebook XML to the phone, and search phonebook/LDAP directory.

What options are available in the Messages sub menu?

The Message sub menu includes the following options:

  • Instant Messages: Displays or clears all received instant messages.
  • Voice Mails: Displays voicemail message information in the format below: new messages/all messages (urgent messages/all urgent messages).

What options are available in the Preference sub menu?

The Preference sub menu includes the following options:

  • Do Not Disturb: Enables/disables Do Not Disturb on the phone.
  • Ring Tone: Configures different ring tones for incoming call.
  • Ring Volume: Adjusts ring volume by pressing left/right arrow key.
  • LCD Contrast: Adjusts active LCD contrast by pressing left/right arrow key.
  • LCD Brightness: Adjusts LCD brightness of idle state and active state by pressing left/right arrow key.

What options are available in the Display sub menu?

The Display sub menu includes the following options:

  • Download SCR XML: Triggers the phone to download the XML idle screen file immediately. The XML idle screen server path and downloading method need to be set up correctly from Web GUI first.
  • Erase Custom SCR: Erases custom XML idle screen previously loaded on the phone. After erasing it, the phone will show default idle screen.
  • Display Language: Selects the language to be displayed on the phone’s LCD. Users may select “Automatic” for local language based on IP location if available. By default, it is Auto.

What options are available in the Date Time sub menu?

The Date Time sub menu includes the following options:

  • Configures time zone, date, and time display format and NTP server on the phone.

What options are available in the Security sub menu?

The Security sub menu includes the following options:

  • Configures the available security settings (Config via keypad menu, web access mode, disable SSH).
  • Headset Type: Selects headset types.
  • Star Key Lock: Turns on/off keypad lock feature and configures keypad lock password. The default keypad lock password is null. If user enabled Star Key Lock without configuring password, user can unlock keypad by holding * key 4 seconds and pressing “OK” button.

What options are available in the Direct IP Call sub menu?

The Direct IP Call sub menu includes the following options:

  • Makes direct IP call.

What options are available in the Phone sub menu?

The Phone sub menu includes the following options:

  • SIP: Configures SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth ID, SIP Password, SIP Transport and Audio information to register SIP account on the phone.
  • Call Features: Configures call forward features for Forward All, Forward Busy, Forward No Answer and No Answer Timeout.

What options are available in the Network sub menu?

The Network sub menu includes the following options:

  • Internet Protocol: Selects “Prefer IPv4” or “IPv6”.
  • IPv4 Settings: Selects IP mode (DHCP/Static IP/PPPoE).
  • DHCP Settings: Configures “Host Name (Option 12)” and “Vendor Class ID (Option 60)”.
  • PPPoE Settings: Configures PPPoE account ID and password.
  • Static IP Settings: Configures static IP address, Netmask, Gateway, DNS Server 1, and DNS Server 2.
  • 802.1X: Enables/Disables 802.1X mode, Configures 802.1x identity and MD5 password.
  • Layer 2 QoS: Configures LAN port 802.1Q/VLAN Tag and priority value. Select “Reset VLAN Config” to reset VLAN configuration on the LAN port.
  • PC Port Mode: Selects PC port mode (Enabled/Disabled/Mirrored). Configures PC port 802.1Q/VLAN Tag and priority value.
  • OpenVPN® Settings: Enable/Disable OpenVPN®, configure OpenVPN® server and port.
  • IPv6 Settings: Selects IPv6 mode, Auto-configured or Statically configured (Full Static / Prefix Static).
  • Wi-Fi Settings (Available on GXP1760W only): Enables Wi-Fi and configure Wi-Fi settings to connect to Wi-Fi networks.

What options are available in the Upgrade sub menu?

The Upgrade sub menu includes the following options:

  • Firmware Server: Configures firmware server for upgrading the phone.
    • Server Address: Specifies the address of the firmware server.
    • Upgrade Via: Specifies the upgrade method.
    • User Name: Specifies the User name for the HTTP/HTTPS Server.
    • File Prefix: Specifies the prefix of the firmware to be downloaded and flashed into the phone.
  • File Postfix: Specifies the postfix of the firmware to be downloaded and flashed into the phone.
  • Config Server: Configures config server for provisioning the phone.
    • Server Address: Specifies the address of the config server.
    • Upgrade Via: Specifies the provision method.
    • User Name: Specifies the User name for the HTTP/HTTPS Server.
    • Password: Specifies the Password for the HTTP/HTTPS Server.
    • File Prefix: Specifies the prefix of the configuration file to be downloaded and flashed into the phone.
    • File Postfix: Specifies the postfix of the configuration file to be downloaded and flashed into the phone.
  • DHCP Opt43/66 Override FW: Enables the DHCP option 43 or 66 on the LAN side, to redirect to the TFTP server.
  • 3CX Auto Provision: Enables the phone to multicast a SUBSCRIBE for provision.
  • Start Provision: Allows users to start provision at the moment.
  • UCM Detect: Detect/connect UCM server to process auto-provision. Manually input the IP and port of the UCM server phone wants to bind with or select from the available UCM server in network.

What options are available in the Authentication sub menu?

The Authentication sub menu includes the following options:

  • Admin Password: Indicates the password of the admin account.
  • End User Password: Turns on/off Test Password Strength feature. This will allow only passwords with some constraints to ensure better security.

What options are available in the Factory Functions sub menu?

The Factory Functions sub menu includes the following options:

  • Audio Loopback: Speak to the phone using speaker/handset/headset. If you can hear your voice, your audio is working fine. Press “Exit” softkey to exit audio loopback mode.
  • Diagnostic Mode: All LEDs will light up. Toggle the line keys LED color to green with “#” and to red with “*”. Press any key on the phone to diagnose the key’s function. The key’s name will display on the LCD. Press the menu button or onhook/offhook handset to exit the diagnostic mode.
  • Wi-Fi Signal Strength: Allows to test the strength of the Wi-Fi signal.
  • Keyboard Diagnostic: All keys’ names will display on LCD screen before diagnosing. Press each key on the phone to remove it from the list of remaining keys to be diagnosed. Lift and put back the handset to exit diagnostic mode after all keys have been diagnosed.
  • Certificate Verification: This is used to validate certificate chain for the server’s certificate.
  • Factory Reset: It is used to restore the phone to factory default settings.
  • Reboot: Reboots the phone.

How do I access the Web GUI?

To access the Web GUI:

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

What are some notes about connecting to the Web GUI?

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

What are the access levels for the login page?

There are two access levels for the login page:

  • Administrator
  • End User.

What is the password criteria?

  • The password is case sensitive with maximum length of 25 characters.
  • When accessing the phone for the first time or after factory reset, users will be asked to change the default Administrator Password before accessing GXP17xx Web interface.
  • The new password field is case sensitive with a maximum length of 25 characters. Using strong password including letters, digits and special characters is recommended for better security.

How do you enable user web access?

Administrators will have to enable user web access under Maintenance 🡪 Web Access, and set a new user password on the same page to enable users to access the Web UI.

What should I do when changing settings?

When changing any settings, always SUBMIT them by pressing the “Save” or “Save and Apply” button on the bottom of the page. If the change is saved only but not applied, after making all the changes, click on the “APPLY” button on top of the page to submit.

What needs to be done after submitting changes in the Web GUI?

After submitting the changes in all the Web GUI pages, reboot the phone to have the changes take effect if necessary (All the options under “Accounts” page and “Phonebook” page do not require reboot. Most of the options under “Settings” page do not require reboot).

What are the sections in the phone’s Web GUI?

The sections in the phone’s Web GUI are:

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

What web pages are allowed for each user level?

  • End User Level: Only Status page, Basic Settings in Advanced Settings page, and some settings in maintenance page.
  • Administrator Level: Browse all pages.

What does the Account Status page display?

The Account Status page displays:

  • Account index: For GXP1760 and GXP1760W up to 3 SIP accounts, for GXP1780 and GXP1782 up to 4 SIP accounts.
  • SIP User ID: Displays the configured SIP User ID for the account.
  • SIP Server: Displays the configured SIP Server address, URL or IP address, and port of the SIP server.
  • SIP Registration: Displays SIP registration status for the SIP account, it will display Yes/No with Green/Red background.

What does the Network Status page display?

The Network Status page displays:

  • MAC Address: Global unique ID of device, in HEX format. The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device.
  • IP Setting: Configured address type: DHCP, Static IP or PPPoE.
  • IPv4 Address: Displays the IPv4 address obtained on the phone.
  • IPv6 Address: Displays the IPv6 address obtained on the phone.
  • OpenVPN® IP: The OpenVPN® IP obtained on the phone.
  • Subnet Mask: Displays the subnet mask obtained on the phone.
  • Gateway: Displays the gateway address obtained on the phone.
  • DNS Server 1: Displays the DNS server address 1 obtained on the phone.
  • DNS Server 2: Displays the DNS server address 2 obtained on the phone.
  • PPPoE Link Up: PPPoE connection status.
  • NAT Type: Displays the type of NAT connection used by the phone.
  • NAT Traversal: Displays the status of NAT connection for each account on the phone.

What does the System Info page display?

The System Info page displays:

  • Product Model: Product model of the phone.
  • Software Version:
    • Boot: boot version number.
    • Core: core version number.
    • Base: base version number.
    • Prog: program version number. This is the main firmware release number, which is always used for identifying the software system of the phone.
    • Locale: locale version number.
    • Recovery: recovery version number.
  • IP Geographic Information:
    • City: displaying city.
    • Language: displaying language.
    • Time Zone: displaying time zone.
    • Country Code: displaying country code. For example: US.
  • System Up Time: System up time since the last reboot.
  • System Time: Current system time on the phone system.
  • Service Status: GUI and Phone service status.
  • Core Dump: Displays the core dump file when available and which could be downloaded for troubleshooting purpose.

What settings are available on the Account x 🡪 General Settings page?

The following settings are available on the Account x 🡪 General Settings page:

  • Account Active: Indicates whether the account is active. The default setting is “Yes”.
  • Account Name: Configures name associated with each account to be displayed on the LCD.
  • SIP Server: Specifies the URL or IP address, and port of the SIP server. This should be provided by VoIP service providers (ITSP).
  • Secondary SIP Server: Specifies the URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails.
  • Outbound Proxy: Configures the IP address or the domain name of the primary outbound proxy, media gateway, or session border controller. It is used by the phone for firewall or NAT penetration in different network environments. If a symmetric NAT is detected, STUN will not work and only an outbound proxy can provide a solution.
  • Backup Outbound Proxy: Configures secondary outbound proxy which will be used when the primary proxy cannot be connected.
  • BLF Server: Configures the optional server used for SUBSCRIBE requests to indicate other extensions status on the SIP server.
  • SIP User ID: Usually in the form of digits similar to phone number or actually a phone number.
  • Authenticate ID: Configures the SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from the SIP User ID.
  • Authenticate Password: Configures the account password required for the phone to authenticate with the ITSP (SIP) server before the account can be registered. After it is saved, this will appear as hidden for security purpose.
  • Name: Specifies SIP server subscriber’s name (optional) that will be used for Caller ID display.
  • Voice Mail Access Number: Sets if the phone system allows users to access the voice messages by pressing the MESSAGE key on the phone. For example, in UCM6XXX IPPBX, *97 could be used.
  • Account Display: Indicates whether to display the User Name or the User ID of the account. When set to “User Name”, LCD will display User Name if it is not empty; When set to “User ID”, LCD will display User ID if it is not empty.

What settings are available on the Account x 🡪 Network Settings page?

The following settings are available on the Account x 🡪 Network Settings page:

  • DNS Mode: Defines which parameter will control how the Search Appliance looks up IP addresses for hostnames. There are four modes: A Record, SRV, NATPTR/SRV, Use Configured IP. The default setting is “A Record”. If the user wishes to locate the server by DNS SRV, the user may select “SRV” or “NATPTR/SRV”. If “Use Configured IP” is selected, please fill in the three fields below: Primary IP Backup IP 1 Backup IP 2 If the SIP server is configured as a domain name, the phone will not send a DNS query but 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 “Primary IP” first. After 3 tries without any response, it will switch to “Backup IP x”, and then it will switch back to “Primary IP” after 3 re-tries. If the SIP server is already an IP address, the phone will use it directly even if “User Configured IP” is selected.
  • NAT Traversal: Specifies which NAT traversal mechanism will be enabled on the phone system. Users could select the mechanism from No, STUN, Keep-alive, UPnP, Auto or VPN. The default setting is “No”. If set to “STUN” and STUN server is configured, the phone will route according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all SIP & SDP messages. The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be “Keep-alive”. Configure this to be “No” if an outbound proxy is used. “STUN” cannot be used if the detected NAT is symmetric NAT. Set this to “VPN” if OpenVPN® is used.
  • Support Rport (RFC 3581): This option enables the utilization of symmetric response routing. When enabled, the “rport” field gets appended to the Via header in the SIP Request. The necessary data is then extracted from the SIP 200OK Response during a SIP Register operation, allowing the rewriting of SIP Contact information and its application in subsequent SIP Requests. Enabled by Default.
  • Proxy-Require: Determines 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 on the SIP server.

What settings are available on the Account x 🡪 SIP Settings 🡪 Basic Settings page?

The following settings are available on the Account x 🡪 SIP Settings 🡪 Basic Settings page:

  • TEL URI: “User=Phone”, so that the “User=Phone” parameter will 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 “Disable”.
  • SIP Registration: Selects whether or not the phone will send SIP Register messages to the proxy/server. The default setting is “Yes”.
  • Unregister On Reboot: Allows the SIP user’s registration information to be cleared when the phone reboots. The SIP REGISTER message will contain “Expires: 0” to unbind the connection. Three options are available: If set to “All“, the SIP user’s registration information will be cleared when the phone reboots. The SIP Contact header will contain “*” to notify the server to unbind the connection. If set to “Instance“, the SIP user will be unregistered on the current phone only. If set to “No”, the phone will not unregister the SIP account when rebooting. The default setting is “No”.
  • Register Expiration: Specifies the time interval (in minutes) in which the phone refreshes its registration with the specified registrar. The default value is 60 minutes. The maximum value is 64800 minutes (about 45 days).
  • Subscribe Expiration: Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified registrar. The maximum value is 64800 (about 45 days). The default value is 60 minutes.
  • Reregister Before Expiration: Specifies the time interval (in seconds) that the phone sends a re-registration request before the Register Expiration. The default value is 0.
  • Enable OPTIONS Keep Alive: Select whether or not the phone will keep sending a message to check the connection with the server.
  • OPTIONS Keep Alive Interval: Specifies the time interval (in second) in which the phone will send the Keep Alive message to the server.
  • OPTIONS Keep Alive Max Lost: Specifies the maximum number of allowed lost packet before the phone will refresh its registration.
  • Local SIP Port: Defines the local SIP port used to listen and transmit. The default value is 5060 for Account 1, 5062 for Account 2, 5064 for Account 3, and 5066 for Account 4. The valid range is from 1 to 65535.
  • SIP Registration Failure Retry Wait Time: Specifies the time interval to retry registration if the process is failed. The valid range is 1 to 3600. The default value is 20 seconds.
  • SIP T1 Timeout: SIP T1 Timeout is an estimate of the round-trip time of transactions between a client and server. If no response is received the timeout is increased, and request re-transmit retries would continue until a maximum amount of time define by T2. The default setting is 0.5 seconds.
  • SIP T2 Timeout: SIP T2 Timeout is the maximum retransmit time of any SIP request messages (excluding the INVITE message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. The default setting is 4 seconds.
  • SIP Transport: and TLS. The default setting is “UDP”.
  • SIP URI Scheme when using TLS: Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is “sips”.
  • Use Actual Ephemeral Port in Contact with TCP/TLS: This option is used to control the port information in the Via header and Contact header. If set to No, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the particular connection. The default setting is “No”.
  • Outbound Proxy Mode: Configures which mode will be placed in route header in sending SIP messages or to be always sent to the outbound proxy.
  • Support SIP Instance ID: Defines whether SIP Instance ID is supported or not. 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”.
  • SUBSCRIBE for Registration: When set to “Yes”, a SUBSCRIBE for Registration will be sent out periodically. The default setting is “No”.
  • Enable 100rel: The use of the PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is very important in order to support PSTN internetworking. To invoke a reliable provisional response, the 100rel tag is appended to the value of the required header of the initial signaling messages. The default setting is “No”.
  • Caller ID Display: When set to “Auto”, the phone will look for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP INVITE. When set to “Disabled”, all incoming calls are displayed with “Unavailable”. When set to “From Header”, the phone will display the caller ID based on the From Header in the incoming SIP INVITE. The default setting is “Auto”.
  • Callee ID Display: The phone supports Callee ID display as follows: When set to “Auto” (Default), the phone will update the callee ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and To Header in the 180 Ringing. When set to “Disabled”, callee id will be displayed as “Unavailable”. When set to “To Header”, caller id will not be updated and displayed as To Header
  • Add Auth Header On Initial Register: When this option is enabled, the device will send Authentication Header on the first Register instead of second Register.
  • Use Privacy Header: Controls whether the Privacy header will be present in the SIP INVITE message and if the header will contain the caller info. If set to “Yes”, the Privacy Header will always be shown on the INVITE. If set to “No”, the Privacy Header will not be shown on the INVITE. Default setting is “Default”.
  • Use P-Preferred-Identity Header: Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message. The default setting is “default”. If set to “Yes”, the P-Preferred-Identity Header will always show in INVITE. If set to “No”, the P-Preferred-Identity Header will not show in INVITE.
  • Ignore Alert-Info Header: Selects whether default ringtone will be played by ignoring alert-info header.

What settings are available on the Account x 🡪 SIP Settings 🡪 Custom page?

The following settings are available on the Account x 🡪 SIP Settings 🡪 Custom page:

  • Use Privacy Header: whether the header contains the caller info. If set to “Yes”, the Privacy Header will always show in INVITE. If set to “No”, the Privacy Header will not show in INVITE. The default setting is “Default”.
  • Use P-Preferred-Identity Header: Controls whether the P-Preferred-Identity Header will be present in the SIP INVITE message. The default setting is “default”: If set to “Yes”, the P-Preferred-Identity Header will always show in INVITE. If set to “No”, the P-Preferred-Identity Header will not show in INVITE.
  • Use X-Grandstream-PBX Header: Controls whether the P-Preferred-Identity Header will be present in the SIP INVITE message. The default setting is “default”: If set to “Yes”, the P-Preferred-Identity Header will always show in INVITE. If set to “No”, the P-Preferred-Identity Header will not show in INVITE.
  • Use P-Access-Network-Info Header: Enables / disables the use of P-Access-Network-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is “No”.
  • Use P-Emergency-Info Header: Enables / disables the use of P-Emergency-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is “No”.
  • Use MAC Header: Indicates whether to put or not the MAC address of the phone on the sip header. If Yes, the sip message for register or unregister will contains MAC address in the header.

What settings are available on the Account x 🡪 SIP Settings 🡪 Advanced Features page?

The following settings are available on the Account x 🡪 SIP Settings 🡪 Advanced Features page:

  • Line Seize Timeout: For Shared Call Appearance, phone must send a SUBSCRIBE-request for the line-seize event package whenever a user attempts to take the shared line off hook. “Line Seize Timeout” is the line-seize event expiration timer. The default value is 15 seconds. The valid range is from 15 to 60.
  • Eventlist BLF URI: Configures the Eventlist BLF URI on the phone to monitor the extensions in the list with Multi-Purpose Key. If the server supports this feature, users need to configure an Eventlist BLF URI on the service side first (i.e., BLF1006@myserver.com) with a list of extensions included. On the phone, in this “Eventlist BLF URI” field, fill in the URI without the domain (i.e., BLF1006). To monitor the extensions in the list, under WebGUI🡪Settings🡪Programmable Keys page, please select “Eventlist BLF” in the key mode, choose account, enter the value of each extension in the list.
  • Auto Provision Eventlist BLFs: When option is enabled, empty multi-purpose keys will be automatically provisioned to the monitored extensions in the Eventlist BLF. The default setting is “Disabled”.
  • Conference URI: Configures Conference URI for N-way conference (Broadsoft Standard).
  • Music On Hold URI: Configures Music On Hold URI to call when a call is on hold. This feature has to be supported on the server side.
  • Force BLF Call-pickup by prefix: Configures to always use the prefix for BLF Call-pickup. The default setting is “No”.
  • BLF Call-pickup Prefix: Configures the prefix prepended to the BLF extension when the phone picks up a call with BLF key. The default setting is **.
  • Call Pickup Barge-In Code: Set Feature Access Code of Call Pickup with Barge-In feature.
  • PUBLISH for Presence: Enables presence feature on the phone. The default setting is “No”.
  • Omit charset=UTF-8 in MESSAGE: Omit charset=UTF-8 in MESSAGE content-type.
  • Special Feature: special features to meet these requirements. Users can choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, and PhonePower depending on the server type. The default setting is “Standard”.
  • Broadsoft Call Center: Default setting is “No”. When set to “Yes”, a softkey “BSCCenter” is displayed on LCD. User can access different Broadsoft Call Center agent features via this softkey. Please note that “Feature Key Synchronization” will be enabled regardless of this setting.
  • Hoteling Event: Broadsoft Hoteling event feature. Default setting is “No”. With “Hoteling Event” enabled, user can access the Hoteling feature option by pressing the “BSCCenter” softkey.
  • Call Center Status: When set to “Yes”, the phone will send SUBSCRIBE to the server to obtain call center status. The default setting is “No”.
  • Feature Key Synchronization: This feature is used for Broadsoft call feature synchronization. When it is enabled, DND, Call Forward features and Call Center Agent status can be synchronized between Broadsoft server and phone. The default setting is “Disabled”.
  • Broadsoft Call Park: When enabled, it will send SUBSCRIBE to Broadsoft server to obtain Call Park notifications. The default setting is “Disabled”.

What settings are available on the Account x 🡪 SIP Settings 🡪 Session Timer page?

The following settings are available on the Account x 🡪 SIP Settings 🡪 Session Timer page:

  • Enable Session Timer: Enable/Disable session timer support.
  • Session Expiration Min-SE: The SIP session timer expiration (in seconds) that enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default setting is 180. The valid range is from 90 to 64800.
  • The minimum session expiration (in seconds). The default value is 90 seconds. The valid range is from 90 to 64800.
  • Caller Request Timer: If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it makes outbound calls. The default setting is “No”.
  • Callee Request Timer: If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it receives inbound calls. The default setting is “No”.
  • Force Timer: If Force Timer is set to “Yes”, the phone will use the session timer even if the remote party does not support this feature.
  • If Force Timer is set to “No”, the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select “No”. The default setting is “No”.
  • UAC Specify Refresher: As a Caller, select UAC to use the phone as the refresher, or select UAS to use the Callee or proxy server as the refresher. The default setting is “Omit”.
  • UAS Specify Refresher: As a Callee, select UAC to use caller or proxy server as the refresher, or select UAS to use the phone as the refresher. The default setting is “UAC”.
  • Force INVITE: The Session Timer can be refreshed using the INVITE method or the UPDATE method.

How do I configure domain certificates when using TLS/TCP for SIP Transport?

Choose whether the domain certificates will be checked or not in Account x 🡪 SIP Settings 🡪 Security Settings. The default setting is “No”.

How do I validate the certification chain when using TCP/TLS?

You can validate the certification chain in Account x 🡪 SIP Settings 🡪 Security Settings when TCP/TLS is configured. The default setting is “No”.

How do I validate incoming messages?

You can choose whether incoming messages will be validated or not in Account x 🡪 SIP Settings 🡪 Security Settings. The default setting is “No”.

How do I configure the phone to check the SIP User ID for incoming INVITE messages?

If you set the Check SIP User ID for incoming INVITE option to “Yes” in Account x 🡪 SIP Settings 🡪 Security Settings, 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”.

How do I set the phone to only accept incoming SIP messages from the proxy server?

Set Accept Incoming SIP from Proxy Only to “Yes” in Account x 🡪 SIP Settings 🡪 Security Settings. This will check the SIP address of the Request URL in the incoming SIP message. If it does not match the SIP server address of the account, the call will be rejected. The default setting is “No”.

How do I configure the phone to authenticate incoming INVITE messages?

If you set Authenticate Incoming INVITE to “Yes” in Account x 🡪 SIP Settings 🡪 Security Settings, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response. The default setting is “No”.

How do I configure the preferred vocoder?

Users can configure vocoders in a preference list in Account x 🡪 Audio Settings, and the vocoders in the list will have a higher preference. The vocoders are included with the same preference order in the SDP message.

How do I set the phone to use the first matching vocoder in the 200OK SDP?

When Use First Matching Vocoder in 200OK SDP is set to “Yes” in Account x 🡪 Audio Settings, the device will use the first matching vocoder in the received 200OK SDP as the codec. The default setting is “No”.

How do I configure the phone to reply with multiple m lines in SDP?

When Disable Multiple m line in SDP is set to “No” in Account x 🡪 Audio Settings, the device will reply with multiple m lines. Otherwise, it will reply with 1 m line. The default setting is “No”.

How do I enable SRTP mode?

Enable SRTP mode based on your selection from the drop-down menu in Account x 🡪 Audio Settings. The default setting is “Disabled”.

How do I enable or disable crypto life time when using SRTP?

Enable or disable the crypto life time option in Account x 🡪 Audio Settings when using SRTP. If you disable this option, the phone does not add the crypto life time to the SRTP header. The default setting is “Yes”.

How do I enable or disable Symmetric RTP?

You can define whether symmetric RTP is supported or not in Account x 🡪 Audio Settings. The default setting is “No”.

How do I configure silence suppression?

You can control the silence suppression/VAD feature of the audio codecs (except for G.723 and G.729) in Account x 🡪 Audio Settings. If set to “Yes”, a small quantity of RTP packets containing comfort noise will be sent during the periods of silence. If set to “No”, this feature is disabled. Default setting is “No”.

How do I select the Jitter Buffer Type?

Select either Fixed or Adaptive for jitter buffer type in Account x 🡪 Audio Settings, based on network conditions. The default setting is “Adaptive”.

How do I select the Jitter Buffer Length?

Select the jitter buffer length from 100ms to 800ms in Account x 🡪 Audio Settings, based on network conditions. The default setting is “300ms”.

How do I configure Voice Frames Per TX?

You can configure the number of voice frames transmitted per packet in Account x 🡪 Audio Settings. The default setting is 2.

How do I configure the encoding rate for the G723.1 codec?

Configure the encoding rate for G723.1 codec in Account x 🡪 Audio Settings.

How do I select the packing mode for G.726-32?

Select “ITU” or “IETF” for G726-32 packing mode in Account x 🡪 Audio Settings. The default setting is “ITU”.

How do I configure the iLBC packet frame size?

Configure the iLBC packet frame size in Account x 🡪 Audio Settings.

How do I specify the iLBC payload type?

Specify the iLBC payload type in Account x 🡪 Audio Settings. Valid type is 96-127.

How do I specify the OPUS payload type?

Specify the OPUS payload type in Account x 🡪 Audio Settings. Valid range is 96 to 127. It cannot be the same as iLBC or DTMF Payload Type.

How do I configure the payload type for DTMF using RFC2833?

Configure the payload type for DTMF using RFC2833 in Account x 🡪 Audio Settings. It cannot be the same as iLBC or OPUS payload type.

How do I configure the mechanism to transmit DTMF digits?

This parameter specifies the mechanism to transmit DTMF digits in Account x 🡪 Audio Settings. There are 3 supported modes:

  • In audio, which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs).
  • RFC2833, which means to specify DTMF with RTP packet. Users could know the packet is DTMF in the RTP header as well as the type of DTMF.
  • SIP INFO, which uses SIP info to carry DTMF. The defect of this mode is that it is easily to cause desynchronized of DTMF and media packet for the reason the SIP and RTP are transmitted, respectively. The default setting is “RFC2833”.

How do I enable early dial?

Select whether or not to enable early dial in Account x 🡪 Call Settings. If set to “Yes”, the SIP proxy must support 484 responses. Early Dial means that the phone sends a SIP INVITE message to the SIP server for each pressed digit. The SIP server looks into its extensions and, if no match happened yet, it sends back a “484 Address Incomplete” message. Otherwise, it executes the action. The default setting is “No”.

How do I configure a dial plan prefix?

You can configure the prefix to be added to each dialed number in Account x 🡪 Call Settings.

How do I configure a dial plan?

You can configure the allowed dial plan for the phone in Account x 🡪 Call Settings. A dial plan establishes the expected number and pattern of digits for a telephone number. The default setting is “{ x+ | +x+ | *x+ | xxx+ }”.

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

  • x – any digit from 0-9
  • X – digits from 0-9, and letters from a-z, A-Z
  • xx+ – at least 2-digit numbers
  • xx – only 2-digit numbers
  • ^ – exclude
  • – any digit of 3, 4, or 5
  • – any digit of 1, 4, or 7
  • <2=011> – replace digit 2 with 011 when dialing
  • | – the OR operand
  • {X123} – match Z123, e123, 5123, …
  • Flag T when adding a “T” at the end of the dial plan, the phone will wait for 3 seconds before dialing out.
  • Back slash “” Character can be used to escape specific letters.
  • Mark = “-” / “_” / “.” / “!” / “~” / “*” / “‘” / “(” / “)”
  • User-unreserved = “&” / “=” / “+” / “$” / “,” / “;” / “?” / “/”.

How can I set up a dial plan that allows 311, 611, and 911, or any 10-digit numbers with the leading digits 1617?

You can use the dial plan: {11 | 1617xxxxxxx}.

How can I set up a dial plan that allows any number with the leading digit 1 followed by a 3-digit number, then any digit between 2 and 9, followed by any 7-digit number; OR any length of numbers with the leading digit 2, replacing the 2 with 011 when dialed?

You can use the dial plan: {1xxxxxxxxx | <2=011>x+}.

*How do I set a dial plan to allow input 123 to pass the dial plan check?

You can set the dial plan with {*123}.

How do I set a dial plan to allow input $123 to pass the dial plan check?

You can set the dial plan with {$123}.

How do I set a dial plan to allow input 12_3 to pass the dial plan check?

You can set the dial plan with {12_3}.

How do I set a dial plan to allow input +123 to pass the dial plan check?

You can set the dial plan with {+123}.

How do I set up a dial plan for a home/office in the US?

You can use the dial plan: { ^1900x. | <=1617>xxxxxx | 1xxxxxxxx | 011x. | 11 }.

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

  • ^1900x. – prevents dialing any number started with 1900
  • <=1617>xxxxxx – allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically
  • 1xxxxxxxx |- allows dialing to any US/Canada Number with 11 digits length
  • 011x – allows international calls starting with 011
  • 11 – allows dialing special and emergency numbers 311, 411, 611 and 911.

How do I set up a dial plan that allows the user to dial * followed by any length of numbers?

You can set the dial plan with { *x+ }.

How do I configure call log settings?

You can configure the call log setting on the phone in Account x 🡪 Call Settings. You can log all calls, only log incoming/outgoing calls (missed calls will not be logged) or disable call log. The default setting is “Log All Calls”.

How do I set the phone to send anonymous calls?

If you set Send Anonymous to “Yes” in Account x 🡪 Call Settings, the “From” header in outgoing INVITE messages will be set to anonymous, essentially blocking the Caller ID to be displayed. The default setting is “No”.

How do I configure the phone to reject anonymous calls?

If you set Anonymous Call Rejection to “Yes” in Account x 🡪 Call Settings, anonymous calls will be rejected. The default setting is “No”.

How do I configure the phone to auto answer calls?

If you set Auto Answer to “Yes” in Account x 🡪 Call Settings, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep. The default setting is “No”.

How do I configure the phone to auto answer based on call-info?

If you set Allow Auto Answer by Call-Info to “Yes” in Account x 🡪 Call Settings, the phone will automatically turn on the speaker phone to answer incoming calls, based on the SIP info header sent from the server/proxy. The default setting is “No”.

How do I configure a custom call-info for auto answer?

You can configure the Custom Call-Info for Auto Answer field in Account x 🡪 Call Settings in addition to matching the contents of the info parameter in the Call-Info header for auto answer.

How do I configure the phone to use the transferred target’s Contact header information for attended transfer?

If you set Refer-To Use Target Contact to “Yes” in Account x 🡪 Call Settings, the “Refer-To” header uses the transferred target’s Contact header information for attended transfer. The default setting is “No”.

How do I set up the phone to transfer a conference call when the conference initiator hangs up?

If you set Transfer on Conference Hangup to “Yes” in Account x 🡪 Call Settings, when the phone hangs up as the conference initiator, the conference call will be transferred to the other parties so that other parties will remain in the conference call. The default setting is “No”.

How do I disable recovery on blind transfer?

You can disable recovery to the call to the transferee on failing blind transfer to the target by setting Disable Recovery on Blind Transfer to “Yes” in Account x 🡪 Call Settings. The default setting is “No”.

How can I configure the “#” key as a “Send” key?

You can configure the “#” key as the “Send” key in Settings 🡪 General Settings. If set to “Yes”, the “#” key will immediately dial out the input digits. If set to “No”, the “#” key is included as part of the dialing string, and make sure the dial plan is properly configured to allow dialing # out. The default setting is “Yes”.

How do I enable an on hold reminder tone?

You can enable an on hold reminder tone in Settings 🡪 General Settings that plays 3 beep tones as a reminder when a call is on hold. It plays the three beeps after 1 minute and then every 30 seconds. The default is “Disabled”.

How do I configure the ringtone for an account?

You can configure the ringtone for the account in Settings 🡪 General Settings. You can choose from different ringtones from the dropdown menu. “Default Ringtone” supports up to 10 custom ringtones, and each “Account -Ringtone” supports up to 10 custom ringtones.

How do I configure the phone to use a distinctive ringtone based on incoming caller ID?

You can specify matching rules with number, pattern, or Alert-Info text in Settings 🡪 General Settings. When the incoming caller ID or Alert Info matches the rule, the phone will ring with a selected distinctive ringtone.

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 2-digit number.
    • xx : only 2-digit number.
    • xx: 3-digit number with the leading digit of 3, 4 or 5.
    • xx: 3-digit number with the leading digit from 6 to 9.
  • Alert Info text. You could configure the matching rule as certain text (e.g., priority) and select the custom ring tone mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info: http://127.0.0.1; info=priority. Select the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info matches the rule, the phone will ring with the selected ring.

How do I configure the ring timeout for no answer?

You can define the timeout (in seconds) for the rings on no answer in Settings 🡪 General Settings. The default setting is 60. The valid range is from 30 to 3600.

How do I configure feature codes?

When Enable Local Call Features is enabled in Account x 🡪 Feature Codes, Do Not Disturb, Call Forwarding and other call features can be used via the local feature codes on the phone. Otherwise, the provisioned feature codes from the server will be used. User-configured feature codes will be used only if server-provisioned feature codes are not provided. And once feature codes are configured, either via server provisioning or local setting, a softkey named “Features” will show on the LCD screen.

How do I configure the DND feature code to turn on DND?

Configure the DND feature code to turn on DND in Account x 🡪 Feature Codes.

How do I configure the DND feature code to turn off DND?

Configure the DND feature code to turn off DND in Account x 🡪 Feature Codes.

How do I configure the delayed call forward wait time?

You can define the timeout (in seconds) before the call is forwarded on no answer in Account x 🡪 Feature Codes. The default value is 20 seconds. The valid range is 1 to 120.

How do I configure the local RTP port?

When configured in Settings 🡪 General Settings, channel 0 will use this port_value for RTP, and channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024 to 65400 and must be even. The default value is 5004.

How do I enable random port generation?

You can force random generation of both the local SIP and RTP ports in Settings 🡪 General Settings. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is “Yes”. This parameter must be set to “No” for Direct IP Calling to work.

How do I configure the keep-alive interval?

You can specify how often the phone sends a blank UDP packet to the SIP server in order to keep the “ping hole” on the NAT router to open in Settings 🡪 General Settings. The default setting is 20 seconds. The valid range is from 10 to 160.

How do I configure the NAT IP address?

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

How do I configure the STUN server?

You can configure the IP address or Domain name of the STUN server in Settings 🡪 General Settings. STUN resolution results are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT routers work with STUN.

How do I configure public mode for Hot Desking?

You can configure to turn on/off the public mode for Hot Desking feature in Settings 🡪 General Settings. The default setting is “No”.

How do I configure the password strength?

Only allow passwords with these constraints to ensure better security in Settings 🡪 General Settings:

  • The password has to be more than 9 characters/digits and has to fulfill at least 3 options among 4 options below:
    • Numeric (0-9)
    • Capital letters (A-Z)
    • Lower case (a-z)
    • Special characters (!, @, #, $, %, ^, &, *, (, ), etc) The default setting is “No”.

How do I configure onhook dial barging?

When the option is set to “Disabled” in Settings 🡪 Call Features, onhook dialing will not be interrupted by an incoming call. The default is “Enabled”.

How do I configure off-hook auto dial?

You can configure a User ID/extension to dial automatically when the phone is off hook in Settings 🡪 Call Features. The phone will use the first account to dial out. The default setting is “No”.

How do I configure off-hook timeout?

If configured in Settings 🡪 Call Features, when the phone is off-hook, it will go on-hook after the timeout (in seconds). The default value is 30 seconds. The valid range is from 10 to 60.

How do I enable automatic redial?

If enabled in Settings 🡪 Call Features, the phone will automatically redial when the called number is busy. The default is No.

How do I configure the number of automatic redial attempts?

The total times that need to be redial if Automatic Redial is enabled can be configured in Settings 🡪 Call Features. The range is between 1 and 200.

How do I configure the interval between each redial attempt?

The interval between each time redial if Automatic Redial is enabled can be configured in Settings 🡪 Call Features. The range is between 1 and 360.

How do I enable or disable the dial plan check while dialing through the call history and any phonebook directories?

You can enable/disable the dial plan check while dialing through the call history and any phonebook directories in Settings 🡪 Call Features. The default setting is “No”.

How do I disable call waiting?

You can disable the call waiting feature in Settings 🡪 Call Features. The default setting is “No”.

How do I disable the call waiting tone?

You can disable the call waiting tone when call waiting is on in Settings 🡪 Call Features. The default setting is “No”.

How do I configure the phone to ring instead of playing the call-waiting tone when audio is on the handset or headset?

You can configure the phone to ring instead of playing the call-waiting tone when audio is on handset or headset in Settings 🡪 Call Features.

How do I disable the busy tone on remote disconnect?

You can disable the busy tone heard in the handset when a call is disconnected remotely in Settings 🡪 Call Features. The default setting is “No”.

How do I disable Direct IP Call?

You can disable Direct IP Call in Settings 🡪 Call Features. The default setting is “No”.

How do I enable quick IP call mode?

When set to “Yes” in Settings 🡪 Call Features, users can dial an IP address under the same LAN/VPN segment by entering the last octet in the IP address. To dial quick IP call, off hook the phone and dial #XXX (X is 0-9 and XXX <=255), phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask. #XX or #X are also valid so leading 0 is not required (but OK). No SIP server is required to make quick IP call. The default setting is “No”.

How do I disable the Conference function?

You can disable the Conference function in Settings 🡪 Call Features. The default setting is “No”.

How do I disable the display of DTMF digits during a call?

When it is set to “Yes” in Settings 🡪 Call Features, the DTMF digits entered during the call will not be displayed on the phone LCD. The default setting is “No”.

How do I configure the mute key functions while idle?

You can specify the function of the mute key in idle in Settings 🡪 Call Features. The default setting is “DND”. When select “Idle Mute” and press the Mute key while idle, the future incoming call will be answered with mute. When select “Disabled”, the Mute key will not take effect while idle.

How do I enable auto unmute?

If the option is enabled in Settings 🡪 Call Features, the phone will automatically unmute when the user unholds the call or establishes a new call. The default is “Yes”.

How do I disable the Transfer function?

You can disable the Transfer function in Settings 🡪 Call Features. The default setting is “No”.

How do I configure the number to dial using the TRAN button?

You can configure the number for the phone to dial as DTMF during the call using the TRAN button in Settings 🡪 Call Features.

How do I configure the attended transfer mode?

When Attended Transfer Mode is set to “Static” in Settings 🡪 Call Features, only the Blind Transfer softkey is available on the LCD after the user presses the TRANSFER button and enters the number for transferring the call. When Attended Transfer Mode is set to “Dynamic”, both Blind Transfer and Attended Transfer softkeys are available.

How do I configure the phone to replace # with %23 in SIP URI?

You can specify whether to replace # by %23 or not for some special situations in Settings 🡪 Call Features. The default setting is “No”.

How do I enable the Click-To-Dial feature?

If this feature is enabled in Settings 🡪 Call Features, the user could click the green dial button on the left top corner of the phone’s Web GUI, then choose the account and dial to the target number. The default setting is “Disabled”.

How do I configure the interval to save the call history?

You can define the interval (in seconds) to save the call history to the phone’s flash in Settings 🡪 Call Features. 0 means this option is disabled. The default value is 300 seconds.

How do I configure the maximum number of unsaved logs?

You can define the number of unsaved logs before written to the phone’s flash in Settings 🡪 Call Features. 0 means this option is disabled. The default value is 200 entries.

How do I configure the default call log type?

This option is used for users to set the default call log list after selecting MENU🡪CALL HISTORY in Settings 🡪 Call Features. The Broadsoft Call Log or Local Call Log option will only show its own list. The Default option will keep both call log lists.

How do I configure the return code when refusing an incoming call?

You can send a selected type of SIP message to the call when refusing the incoming call in Settings 🡪 Call Features. Options include:

  • Busy (486)
  • Temporarily Unavailable (480)
  • Not Found (404)
  • Decline (603) The default is “Busy (486)”.

How do I configure the return code when enabling DND?

You can send a selected type of SIP message when enabling DND in Settings 🡪 Call Features. Options include:

  • Busy (486)
  • Temporarily Unavailable (480)
  • Not Found (404)
  • Decline (603) The default is “Busy (486)”.

How do I enable or disable the local call recording feature?

You can enable/disable the ability to record calls locally while on the call screen in Settings 🡪 Call Features. The default setting is “Disabled”.

How do I configure the saved local call recording location?

You can define the location where the recordings will be stored in Settings 🡪 Call Features, either on the internal storage or on the connected USB. The default setting is “Internal Storage”.

How do I download local call recordings?

You can download the available local call recordings in Settings 🡪 Call Features.

How do I configure the user-agent prefix?

You can configure the prefix in the “User-Agent” header in Settings 🡪 Call Features.

How do I enable predictive dialing?

While dialing, the phone will predict and list candidates of the target number based on the entered digits in Settings 🡪 Call Features. If the target number exists in the phonebook/call history, the phone will display a list of matched numbers, and the user could select the number using the Up/Down key and dial out. The default setting is “Enabled”.

How do I enable or disable the Pound (#) key for redial?

You can enable/disable the Pound(#) key for redial in Settings 🡪 Call Features. If set to “No”, the pound key will not act as redial when it is pressed in the idle screen.

How do I enable Diversion Information Display?

When Enable Diversion Information Display is set to “Yes” in Settings 🡪 Call Features, Diversion info will be displayed if the device receives an INVITE with a Diversion header.

How do I enable the Enhanced Acoustic Echo Canceller?

You can choose to enable or disable the echo canceller on the phone in speaker mode in Settings 🡪 Call Features.

How do I configure paging barge priority?

You can configure the paging barge priority in Settings 🡪 Multicast paging. During an active call, if an incoming multicast page is higher priority (1 being the highest) than this value, the call will be held and the multicast page will be played. The default setting is “Disabled”.

How do I configure paging priority?

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

How do I configure the multicast paging codec?

You can configure the codec for sending multicast pages in Settings 🡪 Multicast paging. There are 5 codecs that can be used: PCMU, PCMA, G.726-32, G.729A/B, G.722 (wide band), G.723 and iLBC. The default setting is “PCMU”.

How do I configure multicast listening addresses?

You can define multicast listening addresses and labels in Settings 🡪 Multicast paging. The “Listening Address” should match the sender’s value, and the “Label” could be the description you want to use. For example: “Listening Address”: “237.11.10.11:6767”, “Label”: the description you want to use.

How do I configure tones?

You can configure ring or tone frequencies based on parameters from the local telecom in Settings 🡪 Ring Tone. The default value is the North American standard. Frequencies should be configured with known values to avoid uncomfortable high-pitch sounds. Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]. (Frequencies are in Hz and cadence on and off are in 10ms)

ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. To set a continuous ring, OFF should be zero. Otherwise, it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.

How do I adjust the call waiting tone volume?

You can adjust the call waiting tone volume in Settings 🡪 Ring Tone by selecting “Low”, “Medium” or “High”. The default setting is “Low”.

How do I configure speaker ring volume?

You can configure speaker ring volume in Settings 🡪 Ring Tone. The valid range is 0 to 7.

How do I configure notification tone volume?

You can configure notification tone volume in Settings 🡪 Ring Tone. The valid range is 0 to 7. This option helps the user to adjust the volume for paging, intercom, auto answer call, and call hold reminder beep notification.

How do you enable 802.1X mode on the phone?

To enable 802.1X mode, set the field to EAP-MD5, or choose EAP-TLS, or EAP-PEAP.

What special characters are accepted for the 802.1X Identity and MD5 Password?

Letters, digits, and special characters including @ and – are accepted.

How do you upload an 802.1X CA certificate to the phone?

Upload the 802.1X CA certificate to the phone using the designated function.

How do you delete an existing 802.1X CA certificate from the phone?

Delete the existing 802.1X CA certificate from the phone using the designated function.

How do you upload an 802.1X Client certificate to the phone?

Upload the 802.1X Client certificate to the phone using the designated function.

How do you delete an existing 802.1X Client certificate from the phone?

Delete the existing 802.1X Client certificate from the phone using the designated function.

How do you configure the PC port mode?

The PC port mode can be configured to “Enabled”, “Disabled”, or “Mirrored”. When set to “Mirrored”, the traffic in the LAN port will go through the PC port as well.

How do you enable LLDP control?

LLDP (Link Layer Discovery Protocol) service is enabled by default.

How can you enable the OpenVPN feature?

The OpenVPN® feature can be enabled or disabled. The default setting is “No”.

How do you upload the Certification Authority of OpenVPN?

To upload a new Certification Authority of OpenVPN, you can click on “Delete” to erase the last certificate, and then upload a new one.

How do you upload an OpenVPN certificate?

To upload a new OpenVPN® certificate, you can click on “Delete” to erase the last certificate, and then upload a new one.

How do you upload an OpenVPN Client Key?

To upload a new OpenVPN® Key, you can click on “Delete” to erase the last certificate, and then upload a new one.

How do you enable Wi-Fi on the phone?

The Wi-Fi on the phone can be enabled or disabled. The default setting is “No”.

How do you configure a Wi-Fi SSID name?

Enter the Wi-Fi SSID name in the designated field.

How do you configure the authentication password to access a Wi-Fi Network?

Enter the password in the designated field.

What are the available options for Wi-Fi encryption type?

The available options are None, WEP, WPA, WPA Enterprise and Auto. The default setting is None.

How do you enable user web access?

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

How do you set a new password for web GUI access as a User?

Set the new password in the “New Password” field. The field is case sensitive.

How do you set a new password for web GUI access as an Admin?

The current admin password is required for setting a new admin password. Set the new password in the “New Password” field. The field is case sensitive.

How do you specify how firmware upgrading and provisioning requests are sent?

The options are: Always Check for New Firmware, Check New Firmware only when F/W pre/suffix changes, and Always Skip the Firmware Check. The default setting is “Always Check for New Firmware”.

How do you enable the phone to send credentials before being challenged by the server?

This can be enabled by setting the “Always Authenticate Before Challenge” option to “Yes”.

How do you enable DHCP Option 43 and Option 66 to override the server?

The default setting for “Allow DHCP Option 43 and Option 66 Override Server” is “Yes”.

How do you select Option 150 or Option 160 to override the firmware server?

Enable “Additional Override DHCP Option”, then select Option 150 or Option 160.

How do you enable DHCP Option 120 to override the SIP Server?

Set the “Allow DHCP Option 120 to override SIP Server” option to “Yes”.

How do you enable automatic provisioning on the phone when 3CX is used as the SIP server?

Set the “3CX Auto Provision” option to “Yes”.

How do you enable automatic upgrade and provisioning?

Set the “Automatic Upgrade” option to “Yes”.

How do you randomize automatic upgrades?

Enable “Randomized Automatic Upgrade” and set the range of hours of the day, or postpone the upgrade every X minute(s) by random 1 to X minute(s).

How do you define the period of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes?

Set the “Hour of the Day (0-23)” option to the desired period. The default value is starting at 1 and ending at 0.

How do you define the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes?

Set the “Day of the Week (0-6)” option to the desired day. The default value is 1.

How do you disable SIP NOTIFY authentication?

Set the “Disable SIP NOTIFY Authentication” option to “Yes”.

How do you determine the config upgrade method?

You can select the config upgrade method via TFTP, HTTP, or HTTPS. The default setting is “HTTP”.

How do you define the server path for provisioning?

The server path for provisioning can be defined in the “Config Server Path” field. The default is “fm.grandstream.com/gs”.

How do you define a username for the HTTP/HTTPS server for configuration?

Define the username in the “Config HTTP/HTTPS User Name” field.

How do you define a password for the HTTP/HTTPS server for configuration?

Define the password in the “HTTP/HTTPS Password” field.

How do you enable your ITSP to lock configuration updates using a prefix?

Configure the “Config File Prefix” to enable this.

How do you enable your ITSP to lock configuration updates using a postfix?

Configure the “Config File Postfix” to enable this.

How do you define the password for encrypting the XML configuration file?

Define the password in the “XML Config File Password” field.

How do you set the phone system to authenticate configuration files?

Set the “Authenticate Conf File” option to “Yes”. The configuration file must include value P1 with the phone system’s administration password.

How do you download the phone’s configuration file?

Click on “Download Device Configuration” to download the phone’s configuration file in .txt format.

How do you upload a configuration file to the phone?

Click on “Upload Device Configuration” to upload a configuration file to the phone.

How do you choose the firmware upgrade method?

You can choose the firmware upgrade method: TFTP, HTTP, or HTTPS. The default setting is “HTTP”.

How do you define the server path for the firmware server?

The server path for the firmware server can be defined in the “Firmware Server Path” field. The default is “fm.grandstream.com/gs”.

How do you define a username for the HTTP/HTTPS server for firmware upgrades?

Define the username in the “Firmware HTTP/HTTPS User Name” field.

How do you define a password for the HTTP/HTTPS server for firmware upgrades?

Define the password in the “Firmware HTTP/HTTPS Password” field.

How do you enable your ITSP to lock firmware updates using a prefix?

Configure the “Firmware File Prefix” to enable this.

How do you enable your ITSP to lock firmware updates using a postfix?

Configure the “Firmware File Postfix” to enable this.

How do you send syslog messages through a secured TLS protocol to a syslog server?

Set the Syslog Protocol to SSL/TLS.

What is required to connect with the TLS server for syslog messages?

The CA certificate is required to connect with the TLS server.

How do you specify the syslog server for the phone to send syslog messages to?

Enter the URL or IP address of the syslog server in the “Syslog Server” field.

How do you include the SIP log in the syslog messages?

Set the “Send SIP Log” option to “Yes”.

How do you enable the phone to automatically recover when running abnormally?

Set the “Auto Recover From Abnormal” option to “Yes”.

How do you select the display language on the phone?

Select the display language from the “Display Language” option. There are 21 languages available, you can also choose “Auto” or “Downloaded Language”.

How do you enable periodic inform?

If set to “Yes”, the device will send inform packets to the ACS.

How do you set up the periodic inform interval to send the inform packets to the ACS?

Set the “Periodic Inform Interval” option to the desired interval.

How do you configure the access control for users to configure from the keypad menu?

There are three options: Unrestricted, Basic settings only, and Constraint Mode.

What happens when “Unrestricted” is selected for keypad menu access control?

All the options can be accessed in the keypad Menu.

What happens when “Basic settings only” is selected for keypad menu access control?

The SIP option under the Phone submenu, and Network, Upgrade, UCM Detect, and Factory Reset options under the System submenu will not be available in the LCD Menu.

What happens when “Constraint Mode” is selected for keypad menu access control?

The phone will require an administration password to change the Network, Upgrade, and Factory Reset options under the System submenu, and SIP option under the Phone submenu as well.

What happens when “Locked Mode” is selected for keypad menu access control?

When “Locked Mode” is selected, after locking the phone, the Menu option of the device will be completely disabled.

How do you enable keypad locking using the STAR key?

Set the “Enable STAR key Keypad Locking” option to “Yes”. The keypad can be locked by pressing and holding the STAR * key for about 4 seconds.

How do you configure the password to lock/unlock the keypad?

Set the password in the “Password to Lock/ Unlock” field.

How do you validate server certificates?

Set the “Validate Server Certificates” option to “Yes” to validate server certificates with a trusted list.

How do you upload an SSL Certificate used for SIP Transport in TLS/TCP?

Upload the certificate in the “SIP TLS Certificate” field.

How do you upload an SSL Private key used for SIP Transport in TLS/TCP?

Upload the private key in the “SIP TLS Private Key” field.

How do you set the password for the SSL Private key used for SIP Transport in TLS/TCP?

Set the password in the “SIP TLS Private Key Password” field.

How do you use a custom certificate instead of the GXP phone default certificate?

Upload the custom certificate in the “Custom Certificate” field.

How do you set the protocol for Web interface?

You can set the protocol for the Web interface to “HTTPS”, “HTTP”, or both.

How do you disable SSH access?

Set the “Disable SSH” option to “Yes”.

How do you specify the time the web or LCD login interface will be locked out after five login failures?

Specify the time in minutes in the “Web/Keypad/Restrict mode Lockout Duration” field. The range is 0-60 minutes.

How do you choose the minimum TLS version for HTTPS provisioning and SIP transport?

Select the minimum TLS version in the “Minimum TLS Version” field.

How do you enable weak TLS ciphers suites?

Enable the “Enable Weak Ciphers” option. You can also disable weak ciphers DES/3DES and RC4, Symmetric Encryption SEED, Symmetric Authentication MD5, Protocol Version SSLv2/SSLv3, or disable all of the above weak TLS ciphers suites.

How do you specify which CA certificate the phone needs to trust?

You can choose “Default Certificates”, “Custom Certificates”, or “All Certificates”.

How do you choose the location where the packet capture will be stored?

Choose the location in the “Capture Location” field, either internal storage or a connected USB.

How do you include RTP packets in the packet capture file?

Set the “With RTP Packets” option to “Yes”.

How do you define the filename of the capture when using USB storage?

Define the filename in the “USB File Name” field.

How do you start, stop, and download the packet capture?

Click the “Start/Stop/Down load” button to start, stop, and download the packet capture.

How do you enable the download of the phonebook file?

Set the “Enable Phonebook XML Download” option to “Enabled”.

How do you configure the server path to download the phonebook XML?

Configure the server path in the “Phonebook XML Server Path” field. This field could be an IP address or URL.

How do you configure the phonebook download interval?

Configure the interval (in minutes) in the “Phonebook Download Interval” field. If it is set to 0, the automatic download will be disabled. The valid range is 5 to 720 minutes.

How do you remove manually-edited entries when the XML phonebook is downloaded?

Set the “Remove Manually-edited Entries on Download” option to “Yes”.

How do you configure the method for importing groups?

You can choose “Replace” to replace existing groups or “Append” to add imported groups to the current ones.

How do you sort the phonebook?

Sort the phonebook based on the selection of the first name or last name.

How do you download the XML phonebook file to a local PC?

Click on “Download XML Phonebook”.

How do you upload a local XML phonebook file to the phone?

Click on “Upload XML Phonebook”.

How do you control the behavior of the phonebook key?

There are five options: Default, LDAP Search, Local Phonebook, Local Group, and Broadsoft Phonebook.

How do you configure the IP address or DNS name of the LDAP server?

Configure the address in the “Server Address” field.

How do you configure the LDAP server port?

Configure the port in the “Port” field. The default port number is “389”.

How do you configure the LDAP search base?

Configure the search base in the “Base” field.

How do you configure the bind “Password” for querying LDAP servers?

Configure the password in the “Password” field. The field can be left blank if the LDAP server allows anonymous binds.

How do you configure the filter used for number lookups in LDAP?

Configure the filter in the “LDAP Number Filter” field.

How do you configure the filter used for name lookups in LDAP?

Configure the filter in the “LDAP Name Filter” field.

How do you select the protocol version for the phone to send the bind requests in LDAP?

Select the protocol version in the “LDAP Version” field. The default setting is “Version 3”.

How do you specify the “name” attributes of each record returned in the LDAP search result?

Specify the attributes in the “LDAP Name Attributes” field, separated by spaces.

How do you specify the “number” attributes of each record returned in the LDAP search result?

Specify the attributes in the “LDAP Number Attributes” field, separated by spaces.

How do you configure the entry information to be shown on the phone’s LCD when using LDAP?

Configure the information in the “LDAP Display Name” field. Up to 3 fields can be displayed.

How do you specify the maximum number of results to be returned by the LDAP server?

Specify the maximum number in the “Max. Hits” field. If set to 0, the server will return all search results. The default setting is 50.

How do you specify the interval for the server to process a request and the client waits for the server to return in LDAP?

Specify the interval (in seconds) in the “Search Timeout” field. The default setting is 30 seconds.

How do you enable LDAP number searching when dialing and receiving calls?

Enable the “LDAP Lookup” option.

How do you configure the display name when LDAP looks up the name for incoming or outgoing calls?

Configure the display name in the “Lookup Display Name” field. This field must be a subset of the LDAP Name Attributes.

How do you set a STUN Server IP?

Enter the STUN Server IP (or FQDN) in the STUN Server field under Settings🡪General Settings. If using a public IP, keep this field blank.

When should you use random ports?

When multiple GXPs are behind the same NAT, you should set “Use Random Ports” to “Yes”. If using a public IP address, set this parameter to “No”.

How do you enable NAT traversal?

Enable the device to use NAT traversal in the “NAT Traversal” field under Accounts X🡪Network Settings. Select Keep-Alive, Auto, STUN (with STUN server path configured too) or other option according to the network setting.

How do you view contacts, edit contacts, or dial out with the Click-to-Dial feature?

From the Web GUI, go to the Phonebook🡪Contacts page, where you can view contacts, add a new contact, edit selected contacts, or dial a contact.

How do you enable the Click-To-Dial feature?

Make sure the option “Click-To-Dial Feature” under web GUI🡪Settings🡪Call Features is turned on.

How do you dial a number using the Click-To-Dial feature?

Click on the dialing icon on the top menu of the Web GUI, enter the number, and click Dial. The phone will go off hook and dial out the number from the selected account.

How do you send a command for the phone to dial out using a web browser?

Use the following URL format: http://ip_address/cgi-bin/api-make_call?phonenumber=1234&account=0&login=admin&password=admin, replacing the fields with your specific information.

How do you save configuration changes?

After making changes, press the “Save” button, then press the “Apply” button on the top of the web GUI page. Or, you can press the “Save and Apply” button directly. It is recommended to reboot the phone after applying changes.

How do you reboot the phone remotely?

Press the “Reboot” button on the top right corner of the web GUI page.

How do you start packet capture?

Choose the capture location (internal storage or connected USB) and then press the Start button.

How do you stop packet capture?

Press the Stop button.

How do you download a packet capture file to a local PC?

Press the Download button.

How do you include or exclude RTP packets in the packet capture?

Use the “With RTP Packets” option.

How do you define the capture filename when using USB storage?

Use the “USB Filename” option.

How do you send a multicast page?

Set the multicast address and port under Web UI, Settings 🡪 Programmable keys, selecting Multicast paging as the key mode for dial page call.

How do you listen to multicast pages?

Configure the listening settings under Web UI Settings 🡪 Multicast Paging.

How do you monitor the call status of users using Eventlist BLF?

The phone sends out a subscription request to the UCM, changing the indicator status of the Line keys, MPKs, or virtual MPKs associated with the monitored users.

How do you pick up calls to monitored extensions?

Use a predefined feature code called BLF- Call-pickup Prefix.

How do you use VPK keys?

VPK (“Virtual Multi-Purpose Keys”) can be assigned with multiple functions, and more VPKs can be displayed on different screens.

How do you set an Action URL for outbound notifications?

Set the “Action URL” under device web UI🡪Settings🡪Outbound Notifications.

What are the supported events for outbound notifications?

  • Setup Completed
  • Registered
  • Unregistered
  • Off Hook
  • On Hook
  • Incoming Call
  • Outgoing Call
  • Missed Call
  • Established Call
  • Terminated Call
  • Open DND
  • Close DND
  • Open Forward
  • Close Forward
  • Blind Transfer
  • Attended Transfer
  • Hold Call
  • UnHold Call

What are the dynamic variables that can be used in Action URLs?

  • $phone_ip
  • $mac
  • $product
  • $program_version
  • $hardware_version
  • $language
  • $local
  • $display_local
  • $remote
  • $display_remote
  • $active_user

When is the Action URL sent?

When a specific phone event occurs, 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.

How do you upgrade the phone via the keypad menu?

  1. Press the MENU button and navigate to select System.
  2. In the System options, select Upgrade.
  3. In the Upgrade options, select Firmware Upgrade Via.
  4. Select the upgrade method and press “OK”.
  5. In the Upgrade options, select Firmware Server.
  6. In the Firmware Server options, select Self-defined Firmware Server.
  7. Enter the firmware server path and press the “OK” softkey.
  8. The device may prompt for a reboot to upgrade.

How do you upgrade the phone via the Web GUI?

  1. Open a web browser on your PC and enter the IP address of the phone.
  2. Login with the administrator username and password.
  3. Go to Maintenance🡪Upgrade and Provisioning page.
  4. Enter the IP address or FQDN for the upgrade server in the “Firmware Server Path” field.
  5. Choose to upgrade via TFTP or HTTP/HTTPS.
  6. Update the changes by clicking the “Save and Apply” button.
  7. Reboot or power cycle the phone to update the new firmware.

What do you do when the upgrading process starts?

Do not interrupt or power cycle the phone when the upgrading process is on.

What is the recommended environment to complete firmware upgrades?

Complete firmware upgrades in a controlled LAN environment whenever possible.

What can you use if you do not have a local TFTP/HTTP server?

Grandstream offers a NAT-friendly HTTP server. Alternatively, you can download a free TFTP or HTTP server.

How do you perform a local firmware upgrade via TFTP?

  1. Unzip the firmware files and put all of them in the root directory of the TFTP server.
  2. Connect the PC running the TFTP server and the phone to the same LAN segment.
  3. Launch the TFTP server and go to the File menu🡪Configure🡪Security to change the TFTP server’s default setting from “Receive Only” to “Transmit Only”.
  4. Start the TFTP server and configure the TFTP server in the phone’s web configuration interface.
  5. Configure the Firmware Server Path to the IP address of the PC.
  6. Update the changes and reboot the phone.

What should you do to ensure proper connection with external devices during a local firmware upgrade via TFTP?

Disable any firewall on your PC.

How are Grandstream SIP Devices configured?

They can be configured via the Web Interface or via a Configuration File (binary or XML) through TFTP or HTTP/HTTPS.

What is the “Config Server Path”?

The “Config Server Path” is the TFTP or HTTP/HTTPS server path for the configuration file. It can be the same or different from the “Firmware Server Path”.

What is a parameter associated with each field in the web configuration page?

A parameter consists of a capital letter P and 2 to 3 (could be extended to 4) digit numeric numbers.

What is the format of an XML configuration file name?

An XML file named “cfgxxxxxxxxxxxx.xml” followed by a file named “cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the phone.

What happens if the download of “cfgxxxxxxxxxxxx.xml” file is not successful?

The provision program will download a generic cfg.xml file.

What happens when certain p-values are included in the first config file?

The phone will try to download a second cfg.xml file and apply it without rebooting (maximum 3 extra attempts).

Which P-values trigger an attempt to download a second config file?

  • 212 — Config upgrade via
  • 234 — Config prefix
  • 235 — Config postfix
  • 237 — Config upgrade Server
  • 240 – Authenticate Config File
  • 1359 – XML Config File Password
  • 8463 – Validate Server Certificate
  • 8467 – Download and process ALL Available Config Files
  • 20713 – Always authenticate before challenge
  • 22011 – Bypass Proxy For
  • 22030 – Enable SSL host verification for provision

What happens if the listed p-values are changed while managing configuration on the Web UI or LCD?

The provision process will be triggered.

Which p-values trigger auto-provision when changed on the Web UI or LCD?

  • 192 — Firmware upgrade server
  • 232 — Firmware prefix
  • 233 — Firmware postfix
  • 6767 — Firmware Upgrade Via
  • 6768 — Firmware HTTP/HTTPS Username
  • 6769 — Firmware HTTP/HTTPS Password
  • 237 — Config upgrade Server
  • 212 — Config upgrade via
  • 234 — Config prefix
  • 235 — Config postfix
  • 1360 — Config HTTP/HTTPS username
  • 1361 — Config HTTP/HTTPS password

How do you restore the phone to factory default settings?

  1. Press the MENU button to bring up the keypad configuration menu.
  2. Select “System” and enter.
  3. Select “Factory Reset”.
  4. Confirm the reset when the warning window appears.
  5. Press “OK” to confirm and the phone will reboot.

What should you do before restoring the phone to factory default settings?

Backup or print all the settings before restoring to factory defaults.

CLICK HERE TO DOWNLOAD GRANDSTREAM NETWORKS GXP1760 (01) PDF MANUAL

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