GXP17xx Series - Administration Guide

  • Updated on July 16, 2022

Thank you for purchasing Grandstream GXP1760/GXP1760W/GXP1780/GXP1782 mid-range IP phones.

The GXP1760W is a mid-range IP phone with integrated dual-band Wi-Fi (GXP1760W only), a sleek design and moderate call-volume features. It comes equipped with 6 lines, 3 SIP accounts, 6 dual-color line keys and 4 XML programmable context sensitive soft keys on a 200 x 80 pixel (3.3”) back-lit LCD display screen. For added personalization the GXP1760W features personalized ring tone/ring back tone music and integration with advanced web and enterprise applications. The GXP1760W features dual auto-sensing 10/100Mbps network ports and offers automated provisioning features with media access control. This mid-range IP phone delivers outstanding performance and design to users worldwide at a budget-friendly price point.

The GXP1780/1782 is a powerful mid-range IP phone with advanced telephony features. This mid-range IP phone comes equipped with 8 lines, 4 SIP accounts, 8 dual-color line keys and 4 XML programmable context sensitive soft keys on a 200 x 80 pixel (3.3”) back-lit LCD display screen. For added personalization the GXP1780/1782 features personalized ring tone/ring back tone music and integration with advanced web and enterprise applications. It is also one of the first Grandstream phones to come equipped with a Kensington Security Slot— one of the most popular anti-theft solutions on the market. The GXP1780/1782 supports the fastest possible connection speeds with dual auto-sensing Gigabit network ports (on GXP1782 only) as well as automated provisioning features with media access control. The GXP1780/1782 is the perfect fit for users looking for a featured-pack IP phone that delivers high performance and design at an affordable price point.

PRODUCT OVERVIEW

Feature Highlights

The following table contains the major features of the GXP1760W & GXP1780/82:

GXP1760/GXP1760W

  • 6 lines
  • 200*80-pixel backlit LCD display
  • 4 XML programmable Softkeys
  • 8 Dedicated function Keys
  • 5-way conference
  • Wi-Fi support (GXP1760W only)


GXP1780/GXP1782

  • 8 lines
  • 200*80-pixel backlit LCD display
  • 4 XML programmable Softkeys
  • 8 Dedicated function Keys
  • 5-way conference
Table 1: GXP1760W/GXP1780/GXP1780 Features at a Glance

GXP1760/GXP1760W Technical Specifications

The following table defines all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for GXP1760W.

Protocols/Standards

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

Network Interface

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

Wi-Fi

Yes, integrated dual-band Wi-Fi 802.11 a/b/g/n/ac (2.4Ghz & 5Ghz).(GXP1760W only)

Graphic Display

200×80-pixel (3.3”) backlit LCD display.

Features Keys

6 line keys with up to 3 SIP accounts, 4 XML programmable context sensitive Softkeys, 5 navigation/menu keys, 8 dedicated function keys for: PHONEBOOK, TRANSFER, CONFERENCE, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME.

Voice Codecs

Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), G.723, iLBC, OPUS, in-band, and out-of-band DTMF (in audio, RFC2833, SIP INFO).

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets), USB, and a separate EHS port reserved for external EHS adapter in the future.

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call-appearance (SCA) / bridged-line-appearance (BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 500 records), XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, Hot Desking, personalized music ringtones and music on hold, server redundancy and fail-over.

HD Audio

Yes, HD handset and speakerphone with support for wideband audio.

Base Stand

Yes, 2 angle positions available. Wall Mount stand sold separately.

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, Kensington Security Slot (Kensington Lock) support.

Multi-Language

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese.

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included, Input: 100-240V, Output +5V, 1A.

Integrated Power-over-Ethernet (802.3af).

Max power consumption: 5W.

Physical

Dimension: 231mm(W) x 167mm(L) x 86mm(H).

Unit weight: 0.925kg.

Package weight: 1.55kg.

Temperature and Humidity

Operation: 0°C to 40°C.

Storage: -10°C to 60°C.

Humidity: 10% to 90% Non-condensing.

Package Content

GXP1760W phone, handset with cord, base stand, universal power supply, network cable, Quick Installation Guide, GPL license.

Compliance

FCC: Part 15 (CFR 47) Class B.

CE: EN55022 Class B, EN55024 Class B, EN61000-3-2, EN61000-3-3, EN60950-1.

RCM: AS/ACIF S004, AS/NZS CISPR22/24, AS/NZS 60950.1

Table 2: GXP1760/GXP1760W Technical Specifications

GXP1780/GXP1782 Technical Specifications

The following table defines all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for GXP1780/GXP1782.

Protocols/Standards

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

Network Interface

Dual switched Ethernet ports with integrated PoE. Auto-sensing 10/100Mbps (GXP1780), 10/100/1000Mbps (GXP1782).

Graphic Display

200*80 pixel backlit LCD display.

Features Keys

8 line keys with up to 4 SIP accounts, 4 XML programmable context sensitive Softkeys, 5 navigation/menu keys, 8 dedicated function keys for: PHONEBOOK, TRANSFER, CONFERENCE, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME.

Voice Codecs

Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723, iLBC, in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO).

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets).

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call-appearance (SCA) / bridged-line-appearance (BLA), downloadable phonebook (XML, LDAP, up to 500 items), call waiting, call log (up to 500 records), XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, Hot Desking, personalized music ringtones and music on hold, server redundancy and fail-over.

HD Audio

Yes, HD handset and speakerphone with support for wideband audio.

Base Stand

Yes, 2 angle positions available. Wall Mount stand sold separately.

QoS

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

Security

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

Multi-Language

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese.

Upgrade/Provisioning

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

Power & Green Energy Efficiency

Universal power adapter included, Input: 100-240V, Output +5V, 1A.

Integrated Power-over-Ethernet (802.3af).

Max power consumption: 5W.

Physical

Dimension: 231mm(W) x 167mm(L) x 86mm(H).

Unit weight: 0.925kg.

Package weight: 1.55kg.

Temperature and Humidity

Operation: 0°C to 40°C.

Storage: -10°C to 60°C.

Humidity: 10% to 90% Non-condensing.

Package Content

GXP1780/GXP1782 phone, handset with cord, base stand, universal power supply, network cable, Quick Installation Guide, GPL license.

Compliance

FCC: Part 15 (CFR 47) Class B.

CE: EN55022 Class B, EN55024 Class B, EN61000-3-2, EN61000-3-3, EN60950-1.

RCM: AS/ACIF S004, AS/NZS CISPR22/24,  AS/NZS 60950.1

Table 3: GXP1780/GXP1782 Technical Specifications

CONFIGURATION GUIDE

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.

Configuration via Keypad

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

  • Enter MENU options: When the phone is in idle, press the round MENU button to enter the configuration menu.
  • Navigate in the menu options: Press the arrow keys up/down/left/right to navigate in the menu options.
  • 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.

Note: The phone automatically exits MENU mode in these situations: An incoming call when the phone is off hook or the MENU mode if left idle for more than 60 seconds.

The MENU options are listed in the following table.

Call History

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.

Status

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.

Phone Book

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.

Messages

Message sub menu include 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).

Preference

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.

  • 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.

  • Date Time

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

  • Security

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.

Direct IP Call

Makes direct IP call.

Phone

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.

System

System sub menu includes the following options:

  • Network
    • 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.

  • Upgrade
    • 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.

  • Password

Specifies the Password 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.

  • Authentication
    • Admin Password

Indicates the password of the admin account.

  • End User Password

Indicates the password of the user account.

  • Test Password Strength

Turns on/off Test Password Strength feature. This will allow only passwords with some constraints to ensure better security.

  • Factory Functions
    • 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.

Table 4: Configuration Menu

The following diagram shows the keypad MENU configuration flow:

Figure 1: Keypad MENU Configuration

Configuration via Web Browser

The GXP1760/GXP1760W/GXP1780/GXP1782 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Google Chrome, Mozilla Firefox, and Microsoft’s IE.

To access the Web GUI:

  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.

Notes:

  • 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.
  • If the phone is properly connected to a working Internet connection, the IP address of the phone will display in MENU🡪Status🡪Network Status. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. Users will need this number to access the Web GUI. For example, if the phone has IP address 192.168.40.154, please enter “http://192.168.40.154” in the address bar of the browser.
  • There are two default passwords for the login page:
User LevelPasswordWeb Pages Allowed

End User Level

123

Only Status and Maintenance

Administrator Level

admin

All pages

Table 5: Default Passwords

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.
  • When logged in to the Web GUI as End user, users cannot access the network settings (Both Basic and Advanced network settings).
Figure 2 : Change Password on first boot

  • 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. 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).

Definitions

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

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

Status Page Definitions

Status 🡪 Account Status

Account

Account index.

For GXP1760: up to 3 SIP accounts.

For GXP1760W: up to 3 SIP accounts.

For GXP1780: up to 4 SIP accounts.

For 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.

Status 🡪 Network Status

MAC Address

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

IP Setting

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.

Status 🡪 System Info

Product Model

Product model of the phone.

Part Number

Product part number.

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.

Table 6: Status Page Definitions

Accounts Page Definitions

Account x 🡪 General Settings

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

Configures user account information provided by your VoIP service provider (ITSP). It is 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.

Account x 🡪 Network Settings

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 SIP server is configured as domain name, phone will not send DNS query, but use “Primary IP” or “Backup IP x” to send SIP message if at least one of them are not empty. Phone will try to use “Primary IP” first. After 3 tries without any response, it will switch to “Backup IP x”, and then it will switch back to “Primary IP” after 3 re-tries.

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

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.

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.

Account x 🡪 SIP Settings 🡪 Basic Settings

TEL URI

Determines if the phone has an assigned PSTN telephone number. This field should be set to “User=Phone”, so that “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 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 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, 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

Determines the network protocol used for the SIP transport. Users can choose from TCP, UDP 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.

Account x 🡪 SIP Settings 🡪 Custom SIP Headers

Use Privacy Header

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

Use

X-Grandstream-PBX Header

Enables / disables the use of X-Grandstream-PBX header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is “No”.

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.

Account x 🡪 SIP Settings 🡪 Advanced Features

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

Different soft switch vendors have special requirements. Therefore, users may need select special features to meet these requirements. Users can choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, 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”.

Account x 🡪 SIP Settings 🡪 Session Timer

Enable Session Timer

Enable/Disable session timer support.

Session Expiration

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.

Min-SE

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

Caller Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it makes outbound calls. The default setting is “No”.

Callee Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it receives inbound calls. The default setting is “No”.

Force Timer

If 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.

Select “Yes” to use the INVITE method to refresh the session timer.

The default setting is “No”.

Account x 🡪 SIP Settings 🡪 Security Settings

Check Domain Certificates

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

Validate Certificate Chain

Validate certification chain when TCP/TLS is configured. The default setting is “No”.

Validate Incoming Messages

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

Check SIP User ID for incoming INVITE

If set to “Yes”, SIP User ID will be checked in the Request URI of the incoming INVITE. If it does not match the phone’s SIP User ID, the call will be rejected. The default setting is “No”.

Accept Incoming SIP from Proxy Only

When set to “Yes”, the SIP address of the Request URL in the incoming SIP message will be checked. If it does not match the SIP server address of the account, the call will be rejected. The default setting is “No”.

Authenticate Incoming INVITE

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

Account x 🡪 Audio Settings

Preferred Vocoder

Multiple vocoder types are supported on the phone, the vocoders in the list is a higher preference. Users can configure vocoders in a preference list that is included with the same preference order in SDP message.
Note: Using firmware version 1.0.0.43 or higher, users are now able to perform concurrent calls using G.729 codec as well.

Use First Matching Vocoder in 200OK SDP

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

Disable Multiple m line in SDP

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

SRTP Mode

Enable SRTP mode based on your selection from the drop-down menu.

The default setting is “Disabled”.

Crypto Life Time

Enable or disable the crypto life time when using SRTP. If users set to disable this option, phone does not add the crypto life time to SRTP header.

The default setting is “Yes”.

Symmetric RTP

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

Silence Suppression

Controls the silence suppression/VAD feature of the audio codecs except for

G.723 (pending) and G.729. If set to “Yes”, a small quantity of RTP packets

containing comfort noise will be sent during the periods of silence. If set to “No”,

this feature is disabled. Default setting is “No”

Jitter Buffer Type

Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The default setting is “Adaptive”.

Jitter Buffer Length

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

Voice Frames Per TX

Configures the number of voice frames transmitted per packet. When configuring this, it should be noted that the “ptime” value for the SDP will change with different configurations here. This value is related to the codec used and the actual frames transmitted in payload during the call. For end users, it is recommended to use the default setting, as incorrect settings may influence the audio quality.

The default setting is 2.

G723 Rate

Configure encoding rate for G723.1 codec.

G.726-32 Pack Mode

Selects “ITU” or “IETF” for G726-32 packing mode. The default setting is “ITU”.

iLBC Frame Size

Configure iLBC packet frame size.

iLBC Payload Type

Specify iLBC payload type.

Valid type is 96-127.

OPUS Payload Type

Specifies OPUS payload type. Valid range is 96 to 127. Cannot be the same as iLBC or DTMF Payload Type.

DTMF Payload Type

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

Send DTMF

This parameter specifies the mechanism to transmit DTMF digits. 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”.

Account x 🡪 Call Settings

Early Dial

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

Default setting is “No”.

Dial Plan Prefix

Configures the prefix to be added to each dialed number.

Dial Plan

A dial plan establishes the expected number and pattern of digits for a telephone number. This parameter configures the allowed dial plan for the phone. Default setting is “{ x+ | \+x+ | *x+ | *xx*x+ }”.

Dial Plan Rules:

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

x – any digit from 0-9

X – digits from 0-9, and letters from a-z, A-Z.

  1. xx+ – at least 2-digit numbers
  2. xx – only 2-digit numbers
  3. ^ – exclude
  4. [3-5] – any digit of 3, 4, or 5
  5. [147] – any digit of 1, 4, or 7
  6. <2=011> – replace digit 2 with 011 when dialing
  7. | – the OR operand
  8. {X123} – match Z123, e123, 5123, …
  9. Flag T when adding a “T” at the end of the dial plan, the phone will wait for 3 seconds before dialing out. This gives users more flexibility on their dial plan setup. E.g. with dial plan 1XXT, phone will wait for 3 seconds to let user dial more than just 3 digits if needed. Originally the phone will dial out immediately after dialing the third digit.
  10. Back slash “\” Character can be used to escape specific letters. e.g. if { \p\a\r\k\+60 } dial plan is configured, park+60 should be able to pass dial plan check. This also can be used to escape Mark and User-unreserved characters.

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

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

Allow 311, 611, and 911 or any 10-digit numbers with leading digits 1617;

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

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

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

Allows any number with leading digit 1 followed by a 3-digit number, followed by any number between 2 and 9, followed by any 7-digit number OR Allows any length of numbers with leading digit 2, replacing the 2 with 011 when dialed.

  • Example 4: If we set the dial plan with {\*123}, it should allow input *123 to pass dial plan check.
  • Example 5: If we set the dial plan with {\$123}, it should allow input $123 to pass dial plan check.
  • Example 6: If we set the dial plan with {12\_3}, it should allow input 12_3 to pass dial plan check.

Example of a simple dial plan used in a Home/Office in the US:

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

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

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

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

Call Log

Configures Call Log setting on the phone. 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”.

Send Anonymous

If set to “Yes”, 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”.

Anonymous Call Rejection

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 speaker phone to answer incoming calls after a short reminding beep.

Default setting is “No”.

Allow Auto Answer by Call-Info

If set to “Yes”, 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”.

Custom Call-Info for Auto Answer

Used in addition to match the contents of the info parameter in the Call-Info header for auto answer.

Refer-To Use Target Contact

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

The default setting is “No”.

Transfer on Conference Hangup

If set to “Yes”, when the phone hangs up as the conference initiator, the conference call will be transferred to the other parties so that other parties will remain in the conference call.

The default setting is “No”.

Disable Recovery on Blind Transfer

Disable recovery to the call to the transferee on failing blind transfer to the target.

The default setting is “No”.

No Key Entry Timeout (s)

Defines the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the digits will be sent out.

The default value is 4 seconds.

The valid range is from 1 to 15.

Use # as Dial Key

Allows users to configure the “#” key as the “Send” key. If set to “Yes”, the “#” key will immediately dial out the input digits. In this case, this key is essentially equivalent to the “Send” key. If set to “No”, the “#” key is included as part of the dialing string and please make sure the dial plan is properly configured to allow dialing # out.

The default setting is “Yes”.

On Hold Reminder Tone

Plays 3 beep tones as reminder when call is on hold. It plays the three beeps after 1 minute and then every 30 seconds. Default is “Disabled”.

Account Ring Tone

Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu.
– “Default Ringtone” supports up to 10 custom ringtones;

– Each “Account -Ringtone” supports up to 10 custom ringtones.

Match Incoming Caller ID

Specifies matching rules with number, pattern, or Alert Info text. When the incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive ringtone.

Matching rules:

  • Specific caller ID number. For example, 8321123.
  • A defined pattern with 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.

[345]xx: 3-digit number with the leading digit of 3, 4 or 5.

[6-9]xx: 3-digit number with the leading digit from 6 to 9.

  • Alert Info text

Users could configure the matching rule as certain text (e.g., priority) and select the custom ring tone mapped to it.

The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info: <http://127.0.0.1>; info=priority.

Selects the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info matches the rule, the phone will ring with the selected ring.

Ring Timeout

Defines the timeout (in seconds) for the rings on no answer.

The default setting is 60.

The valid range is from 30 to 3600.

Account x 🡪 Feature Codes

Enable Local Call Features

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

Do Not Disturb (DND)-On

Configures DND feature code to turn on DND.

Do Not Disturb (DND)-Off

Configures DND feature code to turn off DND.

Delayed Call Forward Wait Time

Defines the timeout (in seconds) before the call is forwarded on no answer. The default value is 20 seconds. The valid range is 1 to 120.

Table 7: Account Page Definitions

Settings Page Definitions

Settings 🡪 General Settings

Local RTP Port

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

Use Random Port

Forces random generation of both the local SIP and RTP ports. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is “Yes”.

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

Keep-alive Interval

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

Use NAT IP

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

STUN Server

The IP address or Domain name of the STUN server. STUN resolution results are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT routers work with STUN.

Public Mode

Configures to turn on/off the public mode for Hot Desking feature.

The default setting is “No”.

Test Password Strength

Only allow password with these constraints to ensure better security:

The password has to be more than 9 characters/digits and has to fulfill at least 3 options among 4 options below:

1) Numeric (0-9)

2) Capital letters (A-Z)

3) Lower case (a-z)

4) Special characters (!, @, #, $, %, ^, &, *, (, ), etc)

Default setting is “No”.

Settings 🡪 Call Features

Onhook Dial Barging

When the option is set to “Disabled”, onhook dialing will not be interrupted by an incoming call. Default is “Enabled”

Off-hook Auto Dial

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

Off-hook Timeout

If configured, 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.

Enable Automatic Redial

If enable, phone will automatically redial when called number is busy. Default is No.

Automatic Redial Times

The total times that need to be redial if Automatic Redial is enabled. Range between 1 and 200.

Automatic Redial Interval

The interval between each time redial if Automatic Redial is enabled. Range between 1 and 360.

Bypass Dial Plan Through Call History and Directories

Enable/Disable the dial plan check while dialing through the call history and any phonebook directories. The default setting is “No”.

Disable Call Waiting

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

Disable Call Waiting Tone

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

Ring for Call Waiting

Ringing instead of playing call-waiting tone when audio is on handset or headset

Disable Busy Tone on Remote Disconnect

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

Disable Direct IP Call

Disables Direct IP Call. The default setting is “No”.

Use Quick IP Call mode

When set to “Yes”, users can dial an IP address under the same LAN/VPN segment by entering the last octet in the IP address. To dial quick IP call, off hook the phone 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”.

Disable Conference

Disables the Conference function.

The default setting is “No”.

Disable in-call DTMF Display

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

Mute Key Functions While Idle

Specifies the function of mute key in idle. Default setting is “DND”. When select “Idle Mute” and press Mute key while idle, the future incoming call will be answered with mute. When select “Disabled”, Mute key will not take effect while idle.

Enable Auto Unmute

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

Disable Transfer

Disables the Transfer function. The default setting is “No”.

In-call dial number on pressing transfer key

Configures the number for the phone to dial as DTMF during the call using TRAN button.

Attended Transfer Mode

When Attended Transfer Mode is set to “Static”, only Blind Transfer softkey is available on LCD after the user presses TRANSFER button and enters the number for transferring call.

When Attended Transfer Mode is set to “Dynamic”, both Blind Transfer and Attended Transfer softkeys are available.

Do Not Escape # as %23 in SIP URI

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

Click-To-Dial Feature

Enables Click-To-Dial feature. If this feature is enabled, user could click the green dial button on left top corner of phone’s Web GUI, then choose the account and dial to the target number. The default setting is “Disabled”.

Call History Flash Writing: Write Timeout

Defines the interval (in seconds) to save the call history to phone’s flash. 0 means this option is disabled. The default value is 300 seconds.

Max Unsaved Log

Defines the number of unsaved logs before written to phone’s flash. 0 means this option is disabled. The default value is 200 entries.

Default call log type

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

Return Code When Refusing Incoming Call

Send selected type of SIP message to the call when refusing the incoming call.

  • Busy (486)
  • Temporarily Unavailable (480)
  • Not Found (404)
  • Decline (603)

Default is “Busy (486)”.

Return Code When Enable DND

Send selected type of SIP message when enabling DND.

  • Busy (486)
  • Temporarily Unavailable (480)
  • Not Found (404)
  • Decline (603)

Default is “Busy (486)”.

Local Call Recording Feature

Enables/Disables the ability to record calls locally while on the call screen. The default setting is “Disabled”

Saved Local Call Recording Location

Defines the location where the recordings will be stored, either on the internal storage or on the connected USB. Default setting is “Internal Storage”

Download Local Call Recordings

Allows users to download the available local call recording.

User-Agent Prefix

Configure the prefix in the “User-Agent” header.

Predictive Dialing Feature

Enables/Disables predictive dialing feature.

While dialing, the phone will predict and list candidates of the target number based on the entered digits. 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”.

Use Pound (#) For Redial

Enables/Disables Pound(#) key for redial, if set to “No” the pound key will not act as redial when it is pressed in idle screen.

Enable Diversion Information Display

When “Enable Diversion Information Display” is set to “Yes”, Diversion info will be displayed if the device receives INVITE with Diversion header.

Enable Enhanced Acoustic Echo Canceller

Used to choose to enable or disable the echo canceller on the phone in speaker mode.

Settings 🡪 Multicast paging

Paging Barge

During active call if incoming multicast page is higher priority (1 being the highest) than this value the call will be held and multicast page will be played. The default setting is “Disabled”.

Paging Priority Active

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

Multicast Paging Codec

The codec for sending multicast pages, there are 5 codecs could be used: PCMU, PCMA, G.726-32, G.729A/B, G.722 (wide band), G.723 and iLBC. The default setting is “PCMU”.

Multicast Listening

Defines multicast listening addresses and labels. For example:

“Listening Address” should match the sender’s Value such as

“237.11.10.11:6767”

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

For details, please check the “Multicast Paging User Guide” on our Website.

Settings 🡪 Ring Tone

Call Progress Tones:

  • System Ring Tone
  • Dial Tone
  • Second Dial Tone
  • Message Waiting
  • Ring Back Tone
  • Call-Waiting Tone
  • Busy Tone
  • Reorder Tone

Configures ring or tone frequencies based on parameters from local telecom. The default value is North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.

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

Call Waiting Tone Gain

This adjusts the call waiting tone volume.

Users can select “Low”, “Medium” or “High”. The default setting is “Low”.

Speaker Ring Vol

Configures speaker ring volume. Valid range is 0 to 7.

Notification Tone Volume

Configure notification Tone Volume. Valid range is 0 to 7.

Note: This option helps the user to adjust the volume for paging, intercom, auto answer call and call hold reminder beep notification.

Settings 🡪 Audio Control

Headset Key Mode

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

  • Default Mode:
  • When the phone is in idle, press HEADSET button to off hook the phone and make calls by using headset. Headset icon will display on the screen in dialing/talking status.
  • When there is an incoming call, press HEADSET button to pick up the call using headset.
  • When there is an active call using headset, press HEADSET button to hang up the call.
  • When Speaker/Handset is being used in dialing/talking status, press HEADSET button to switch to headset. Press it again to hang up the call. Or press speaker/Handset to switch back to the previous mode.
  • Toggle Headset/Speaker:
  • When the phone is in idle, press HEADSET button to switch to Headset mode. The headset icon will display on the left side of the screen. In this mode, if pressing Speaker button or Line key to off hook the phone, headset will be used.
  • When there is an active call, press HEADSET button to toggle between Headset and Speaker.

Headset Type

Selects whether the connected headset is normal RJ11 headset, or Plantronics EHS headset.

Always Ring Speaker

Configures to enable or disable the speaker to ring when headset is used on “Toggle Headset/Speaker” mode.

If set to “Yes”, when the phone is in Headset “Toggle Headset/Speaker” mode, both headset and speaker will ring on incoming call.

The default setting is “No”.

Headset TX gain

Configures the transmission gain of the headset. Available values are:

-15dB, -12dB, -9dB, -6dB, 0dB, 6dB, 9dB. The default value is 0dB.

Headset RX gain

Configures the reception gain of the headset. Available values are: -9dB, -6dB, 0dB, 6dB, 9dB. The default value is 0dB.

Handset TX gain

Configures the transmission gain of the handset. Available values are: -6dB, 0dB, 6dB. The default value is 0dB.

Settings 🡪 LCD Display

Backlight Brightness: Active

Configures the LCD brightness when the phone is active.

Valid range is 0 to 8 where 8 is the brightest. Default value is 6.

Backlight Brightness: Idle

Configures the LCD brightness when the phone is idle.

Valid range is 0 to 8 where 8 is the brightest. Default value is 2.

Active Backlight Timeout

Configures the minute of active backlight timeout. The valid range is 1 to 90. Default value is 1.

Disable Missed Call Backlight

If set to “Yes”, the screen will turn off the LCD backlight when there is a missed call on the phone. The default setting is “No”.

Hide System Softkey on Main Page

Check to hide the system generated Softkey on main page.

Disable Missed Call Backlight

Users can turn off LCD blacklight when having a missed call and also get missed call notification by flash MWI LED by set this option to “Yes, but flash MWI LED”. In this case when missed call exists, MWI will be flashing to notify users.

Settings 🡪 LED Control

Disable VM/MSG power light flash

Configures the ON/OFF of voicemail light flash. If select the choice yes, the VM/MSG light cannot flash even though there is an unread voice mail or message.

Settings 🡪 Date and Time

NTP Server

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

The default setting is us.pool.ntp.org.

NTP Update Interval

Time interval for updating time from the NTP server. Valid time value is in between 5 to 1440 minutes.

The default setting is “1440” minutes.

Allow DHCP Option 42 override NTP Server

Defines whether DHCP Option 42 should override NTP server or not. When enabled, DHCP Option 42 will override the NTP server if it is set up on the LAN. The default setting is “Yes”.

Time Zone

Configures the date/time used on the phone according to the specified time zone.

Allow DHCP Option 2 override Time Zone Setting

Defines whether DHCP Option 2 should override Time Zone setting or not. When enabled, DHCP Option 2 will override the Time Zone setting if it is set up on the LAN. The default setting is “Yes”.

Self-Defined Time Zone

This parameter allows the users to define their own time zone.

The syntax is: std offset dst [offset], start [/time], end [/time]

Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0

MTZ+6MDT+5

This indicates a time zone with 6 hours offset with 1 hour ahead (when daylight saving) which is U.S central time. If it is positive (+) if the local time zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian) and negative (-) if it is east.

M4.1.0,M11.1.0

The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)

The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd Tuesday…)

The 3rd number indicates weekday: 0,1,2,..,6 (for Sun, Mon, Tues, … ,Sat)

Therefore, this example is the DST which starts from the First Sunday of April to the 1st Sunday of November.

Date Display Format

Configures the date display format on the LCD. The following formats are supported and the default setting is yyyy-mm-dd:

  • yyyy-mm-dd: 2012-07-02
  • mm-dd-yyyy: 07-02-2012
  • dd-mm-yyyy: 02-07-2012
  • dddd, MMMM dd: Friday, October 12
  • MMMM dd, dddd: October 12, Friday

Time Display Format

Configures the time display in 12-hour or 24-hour format on the LCD.

The default setting is in 12-hour format.

Settings 🡪 Web Service

Use Auto Location Service

Enables/disables auto location services on the phone.

Default setting is “Yes”.

Settings 🡪 XML Application

Idle Screen XML Download

Configures to enable idle screen XML download. Users could select HTTP/HTTPS/TFTP to download the XML idle screen file.

Default setting is “No”.

Download Screen XML at Boot-up

If set to “Yes”, the idle screen XML file will be downloaded when the phone boots up. The default setting is “No”.

Use Custom Filename

Specifies the custom file for the idle screen XML file to be downloaded.

Idle Screen XML Server Path

Configures the server path to download the idle screen XML file.

This field could be IP address or URL, with up to 256 characters.

Settings 🡪 Programmable Keys

Virtual Multi-Purpose Keys Settings

Call Screen Settings 🡪 Key Mode: Defines the key mode used. Two modes are available:

  • Line Mode: Each configured line key can only hold 1 call, and users can switch to choose between accounts by pressing on a free line key.

Advantage: Easy to manage. Disadvantage: Limited line space.

  • Account Mode: Each configured line key is reserved for the corresponding account and can have up to 8 grouped lines. Users can press on “New call” softkey to initiate a new call.

Advantage: Multiple calls per key. Disadvantage: Can get a bit complicated

Call Screen Settings 🡪 Transfer Mode via VPK: Defines whether to perform blind transfer, attended transfer, or a new call with the specific in the Value field when a user presses “Transfer” virtual multiple-purpose key.

Call Screen Settings 🡪 Enable transfer via non-Transfer MPK: When set to Yes, MPK with type BLF, Speed dial, etc, will perform as transfer MPK under active call.

Use Long Label

The phone is able to display longer label in idle screen. Phone will support up to 16 characters in the label. If there are 17+ characters, 16th character will be replaced as “…”

Show Key Label

Users can choose to Show/Hide information of VPK when dialing.

If “Hide” is selected, while dialing, VPK will only display “1, 2, 3, 4”.

Virtual Multi-Purpose Keys (VPK)

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

  • Line

Regular line key to open up a line and switch line. The Value field can be left blank.

  • Shared Line

Share line for Shared Line Appearance feature. Select the Account registered as Shared line for the line key.

The Value field can be left blank.

  • Speed Dial

Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed or enter the IP address to set the Direct IP call as Speed Dial.

  • Busy Lamp Field (BLF)

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

  • Presence Watcher

This option has to be supported by a presence server and it is tied to the “Do Not Disturb” status of the phone’s extension.

  • Eventlist BLF

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

  • Speed Dial via active account

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

  • Dial DTMF

Enter a series of DTMF digits in the Value field to be dialed during the call. “Enable MPK Sending DTMF” has to be set to “Yes” first.

  • Voice Mail

Select Account and enter the Voice Mail access number in the Value field.

  • Call Return

The last answered calls can be dialed out by using Call Return. The Value field should be left blank.

Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.

  • Transfer

Select Account and enter the number in the Value field to be transferred (blind transfer) during the call.

  • Call Park

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

  • Monitored Call Park

Select account from Account field and enter the call park extension in the Value field to park/pick up the call, and also monitor the parked call via Line Key’s light.

  • Intercom

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

  • LDAP Search

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

The Base in LDAP configuration will be used if the Description field is left blank.

Enter the LDAP Name/Number filter in the Value field.

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

Description 🡪 ou=video,ou=SZ,dc=grandstream,dc=com

Value 🡪 sn=Li

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

  • Multicast Paging

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

  • Record

This option is for Recording calls. Enter Line key description in Description filed and the recorded extension number in Value field.

Please make sure whether your VOIP provider supports this feature before using it.

  • Call Log

Select Account and enter account number in the Value field to allow configuration of call log for other extension.

  • Menu

This option is to take users to the main menu screen directly.

  • Information

Display system information such as IP address, MAC address and software version.

  • Message

Display information about instant messages and voice mails received for the programmed account.

  • Provision

Used for VPK/Softkey to trigger provision process

Idle Screen Softkeys

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 be dialed.

  • Speed Dial via active account

Similar to Speed Dial but it will dial based on the current active account.

For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2.

  • Voice Mail

Select Account and enter the Voice Mail access number in the Value field.

  • Call Return

The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.

  • Intercom

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

  • LDAP Search

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

The Base in LDAP configuration will be used if the Description field is left blank.

Enter the LDAP Name/Number filter in the Value field. For example:

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

Description 🡪 ou=video,ou=SZ,dc=grandstream,dc=com

Value 🡪 sn=Li

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

  • Call Log

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

  • Menu

Shortcut for Menu button.

  • Information

Display system information such as IP address, MAC address and software version.

  • Message

Display information about instant messages and voice mails received for the programmed account.

  • Provision

Users can manually setup this feature to provision the phone by pressing either VPK keys or softkeys.

Settings 🡪 Broadsoft 🡪 Broadsoft XSI

XSI

Configures XSI Directory.

  • Server

Configure the BroadWorks Xsi server URI. If the server uses HTTPS, please add the header “HTTPS” ahead of the Server URI.

For instance, “https://SERVER_URI”.

  • Port

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

  • XSI Authentication Type

Select the authentication type to use to authenticate against the Broadsoft server.

User can choose the Login credentials, SIP Credentials, or use the existing Accounts to authenticate.

  • Login Credentials
    • Login Username

Configure the Username for the BroadWorks XSI feature.

  • Login Password

Configure the password for the BroadWorks XSI feature.

  • SIP Credentials
    • SIP User Name

Configure SIP Username for the BroadWorks XSI server.

  • SIP UserID

Configure SIP User ID for the BroadWorks XSI server.

  • SIP Password

Configure SIP Password for the BroadWorks XSI server.

Network Directories

Enable/Disable Broadsoft Network directories and defines the directory name.

The directory types are:

  • Group Directory

Enable/Disable and rename the BroadWorks XSI Group Directory features on the phone. If keep the Name box blank, the phone will use the default name “Group” for it.

  • Enterprise Directory

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

  • Group Common

Enable/Disable and rename the BroadWorks XSI Group Common Directory features on the phone. If keep the Name box blank, the phone will use the default name “Group Common” for it.

  • Enterprise Common

Enable/Disable and rename the BroadWorks XSI Enterprise Common Directory features on the phone.

If keep the Name box blank, the phone will use the default name “Enterprise Common” for it.

  • Personal Directory

Enable/Disable and rename the BroadWorks XSI Personal Directory features on the phone.

If keep the Name box blank, the phone will use the default name “Personal” for it.

  • Missed Call Log

Enable/Disable and rename the BroadWorks XSI Missed Call Log features on the phone. If keep the Name box blank, the phone will use the default name “Missed” for it.

  • Placed Call Log

Enable/Disable and rename the BroadWorks XSI Placed Call Log features on the phone. If keep the Name box blank, the phone will use the default name “Outgoing” for it.

  • Received Call Log

Enable/Disable and rename the BroadWorks XSI Placed Call Log features on the phone. If keep the Name box blank, the phone will use the default name “Incoming” for it.

Settings 🡪 External Service

Grandstream Door System

Allows GXP17xx to send open door command to GDS3710. Authentication is required to make commands to GDS3710 for some actions.

To perform authentication with GDS3710 and show Open Door softkey, the GXP17xx needs to be configured as follows:

  1. Service Type: GDS.
  2. System Identification: String to identify the GDS3710.
  3. System Number: The SIP number or the IP address of GDS3710.
  4. Access Password: The Remote PIN configured on GDS3710.

Notes:

  • If GDS3710 makes a call by SIP number, the number of GDS service on GXP phone should be the same as “SIP User ID” on SIP Basic Settings page on GDS3710.
  • If GDS3710 makes direct IP call, the number of GDS service on GXP phone should be the same as “IP Address” on Network Info page on GDS3710.
  • The password of GDS service on GXP phone should be the same as “Remote PIN to Open the door” on GDS3710.
  • Users can configure the phone to open 2 doors by adding two passwords for the same GDS number. After setting up, there would be two softkey OpenDoor1, OpenDoor2 for opening the two different doors and this is decided by order of the door system’s configuration entries of the same GDS number on the phone.
  • User can click on “Reset” to delete all the GDS Services Configuration on the GXP17XX.

Settings 🡪 Outbound Notification

Action URL

For detailed instructions, please refer to: [Outbound Notification Support]

  • 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

Settings 🡪 Affinity Settings

Affinity Support

Allows communication with GS Affinity CTI application to manage telephone calls from computer. If enabled, a reboot is required to establish the communication. Default is “Disabled”.

GS Affinity CTI Application is available HERE and its User Guide from HERE.

Preferred Account

Chooses the account on which CTI support is enabled.

Table 8: Settings Page Definitions

Network Page Definitions

Network 🡪 Basic Settings

Internet Protocol

Selects “IPv4 Only”, “IPv6 Only”, “Both, prefer IPv4” or “Both, prefer IPv6”. Default setting is “IPv4 Only”.

IPv4 Address

Allows users to configure the appropriate network settings on the phone to obtain IPv4 address from DHCP server. Users could select “DHCP”, “Static IP” or “PPPoE”. By default, it is set to “DHCP”.

DHCP Host name (Option 12)

Specifies the name of the client. This field is optional but may be required by some Internet Service Providers.

DHCP Vendor Class ID (Option 60)

Used by clients and servers to exchange vendor class ID. The default setting is “Grandstream GXP1760W” for GXP1760W, ”Grandstream GXP1780” for GXP1780, “Grandstream GXP1782” for GXP1782.

PPPoE Account ID

Enter the PPPoE account ID.

PPPoE Password

Enter the PPPoE Password.

PPPoE Service Name

Enter the PPPoE Service Name.

Statically configured as

Select to apply the static configuration set for the IP address.

IPv4 Address

Enter the IP address when static IP is used.

Subnet Mask

Enter the Subnet Mask when static IP is used for IPv4.

Gateway

Enter the Default Gateway when static IP is used for IPv4.

DNS Server 1

Enter the DNS Server 1 when static IP is used for IPv4.

DNS Server 2

Enter the DNS Server 2 when static IP is used for IPv4.

Preferred DNS Server

Enter the Preferred DNS Server for IPv4.

IPv6 Address

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

Static IPv6 Address

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

IPv6 Prefix Length

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

Prefix Static

Select to apply the prefix set on IPv6 Prefix when using “Statically configured” IPv6 address type.

IPv6 Prefix (64 bits)

Enter the IPv6 Prefix (64 bits) when Prefix Static is used in “Statically configured” IPv6 address type.

DNS Server 1

Enter the DNS Server 1 for IPv6.

DNS Server 2

Enter the DNS Server 2 for IPv6.

Preferred DNS server

Enter the Preferred DNS Server for IPv6.

Network 🡪 Advanced Settings

802.1X mode

Allows the user to enable/disable 802.1X mode on the phone. The default value is disabled. To enable 802.1X mode, this field should be set to EAP-MD5, users may also choose EAP-TLS, or EAP-PEAP.

802.1X Identity

Enter the Identity information for the 802.1x mode.

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

MD5 Password

Enter the MD5 Password for the 802.1X mode.

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

802.1X CA Certificate

Upload 802.1X CA certificate to the phone or delete existed 802.1X CA certificate from the phone.

802.1X Client Certificate

Upload 802.1X Client certificate to the phone or delete existed 802.1X Client certificate from the phone.

HTTP Proxy

Specifies the HTTP proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.

HTTPS Proxy

Specifies the HTTPS proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.

Layer 3 QoS for SIP

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

Layer 3 QoS for RTP

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

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0.

Layer 2 QoS 802.1p Priority Value

Assigns the priority value of the Layer2 QoS packets. The default value is 0.

PC Port Mode

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

PC Port VLAN Tag

Assigns the VLAN Tag of the PC port. The default value is “0”.

PC Port Priority Value

Assigns the priority value of the PC port. The default value is “0”.

Enable LLDP

Control LLDP (Link Layer Discovery Protocol) service. Default setting is “Enabled”.

Network 🡪 Remote Control

CSTA Control

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

Network 🡪 OpenVPN® Settings

OpenVPN® Enable

Enable/Disable OpenVPN® feature. Default is No.

OpenVPN® Server Address

Specifies the IP address or FQDN for the OpenVPN® Server.

OpenVPN® Port

Specifies the listening port of the OpenVPN® server.

Default is 1194.

OpenVPN® Transport

Specifies the Transport Type of OpenVPN® whether UDP or TCP.

Default is UDP.

OpenVPN® CA

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

OpenVPN® Certificate

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

OpenVPN® Client Key

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

Network 🡪 Wi-Fi Settings (Available on GXP1760W only)

Enable/Disable Wi-Fi

Enables / Disables the Wi-Fi on the phone. Default setting is “No”.

Access Point

(1 – 10)

SSID

Enters Wi-Fi SSID name to connect.

Password

Configures the authentication password to access Wi-Fi Network.

Security Type

Specifies the Wi-Fi encryption type. The available options are the following: None, WEP, WPA, WPA Enterprise and Auto. Default settings is None.

Table 9: Network Page Definitions

Maintenance Page Definitions

Maintenance 🡪 Web Access

User Password

New Password

Set new password for web GUI access as User. This field is case sensitive.

Confirm Password

Enter the new User password again to confirm.

Admin Password

Current Password

The current admin password is required for setting a new admin password.

New Password

Set new password for web GUI access as Admin. This field is case sensitive.

Confirm Password

Enter the new Admin password again to confirm.

Maintenance 🡪 Upgrade and Provisioning

Upgrade Firmware

Allows users to upload the firmware file locally by pressing Start, after selecting the correct firmware file from the local storage, the phone will start the firmware upgrade automatically.

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W pre/suffix changes, Always Skip the Firmware Check. The default setting is “Always Check for New Firmware”.

Always Authenticate Before Challenge

Only applies to HTTP/HTTPS. If enabled, the phone will send credentials before being challenged by the server.

The default setting is “No”.

Allow DHCP Option 43 and Option 66 Override Server

Default setting is “Yes”. DHCP option 66 originally was only designed for TFTP server. And then was extended to support an HTTP URL.

GXP phones support both TFTP and HTTP server via option 66. Users can also use DHCP option 43 vendor specific option to do this. DHCP option 43 approach has priorities.

Additional Override DHCP Option

When enabled, users could select Option 150 or Option 160 to override the firmware server instead of using the configured firmware server path or the server from option 43 and option 66 in the local network. Please note this option will be effective only when option “Allow DHCP Option 43 and Option 66 to Override Server” is enabled. The default setting is “None”.

Allow DHCP Option 120 to override SIP Server

Enables DHCP Option 120 from local server to override the SIP Server on the phone. The default setting is “No”.

3CX Auto Provision

Enables automatic provision feature on the phone when 3CX is used as the SIP server. The default setting is “Yes”.

Automatic Upgrade

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

Randomized Automatic Upgrade

Randomizes Automatic Upgrade within the range of hours of the day or postpone the upgrade every X minute(s) by random 1 to X minute(s)

Hour of the Day (0-23)

Defines the period of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes. The default value is starting at 1 and ending at 0.

Day of the Week

(0-6)

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

Disable SIP NOTIFY Authentication

Device will not challenge NOTIFY with 401 when set to “Yes”.

The default setting is “No”.

Config

Config Upgrade Via

Determine the config upgrade method via TFTP, HTTP or HTTPS.

The default setting is “HTTP”.

Config Server Path

Defines the server path for provisioning. It could be different from the firmware server for upgrading.

Default is “fm.grandstream.com/gs”.

Config HTTP/HTTPS User Name

Defines user name for the HTTP/HTTPS server.

Config HTTP/HTTPS Password

Defines password for HTTP/HTTPS server.

Config File Prefix

Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted prefix will be downloaded and flashed into the phone.

Config File Postfix

Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.

XML Config File Password

Defines the password for encrypting the XML configuration file using OpenSSL. This is required for the phone to decrypt the encrypted XML configuration file.

Authenticate Conf File

Sets the phone system to authenticate configuration file before applying it. When set to “Yes”, the configuration file must include value P1 with phone system’s administration password. If it is missed or does not match the password, the phone system will not apply it. Default setting is “No”.

Download Device Configuration

Click to download phone’s configuration file in .txt format.

Upload Device Configuration

Upload configuration file to phone.

Firmware

Firmware Upgrade Via

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

Firmware Server Path

Defines the server path for the firmware server.

It could be different from the configuration server for provisioning. Default is “fm.grandstream.com/gs”.

Firmware HTTP/ HTTPS User Name

Defines user name for the HTTP/HTTPS server.

Firmware HTTP/ HTTPS Password

Defines password for HTTP/HTTPS server.

Firmware File Prefix

Enables your ITSP to lock firmware updates. If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone.

Firmware File Postfix

Enables your ITSP to lock firmware updates. If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the phone.

Maintenance 🡪 Syslog

Syslog Protocol

If set to SSL/TLS, the syslog messages will be sent through secured TLS protocol to syslog server. Default setting is UDP.

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

Syslog Server

The URL or IP address of the syslog server for the phone to send syslog to.

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

Syslog Level

Selects the level of logging 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 on boot up (INFO level).
  • NAT related info (INFO level).
  • sent or received SIP message (DEBUG level).
  • SIP message summary (INFO level).
  • inbound and outbound 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).

Send SIP Log

Configures whether the SIP log will be included in the syslog messages.

The default setting is “No”.

Auto Recover From Abnormal

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

Maintenance 🡪 Language

Display Language

Selects display language on the phone. There are 21 languages that can be set as display language, user could also choose “Auto” or “Downloaded Language” as display language.

The default setting is “Auto”.

Maintenance 🡪 TR-069

ACS URL

URL for TR-069 Auto Configuration Servers (ACS).

TR-069 Username

ACS username for TR-069.

TR-069 Password

ACS password for TR-069.

Periodic Inform Enable

Enables periodic inform. If set to “Yes”, device will send inform packets to the ACS. The default setting is “No”.

Periodic Inform Interval

Sets up the periodic inform interval to send the inform packets to the ACS.

Connection Request Username

The user name for the ACS to connect to the phone.

Connection Request Password

The password for the ACS to connect to the phone.

Connection Request Port

The port for the ACS to connect to the phone.

CPE SSL Certificate

The Cert File for the phone to connect to the ACS via SSL.

CPE SSL Private Key

The Cert Key for the phone to connect to the ACS via SSL.

Maintenance 🡪 Security Settings🡪 Security

Configuration via Keypad Menu

Configures the access control for the users to configure from keypad Menu. There are three different options.

The default setting is “Unrestricted”:

  • Unrestricted

All the options can be accessed in keypad Menu.

  • Basic settings only

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

  • Constraint Mode

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

  • Locked Mode

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

Enable STAR key Keypad Locking

If set to “Yes”, the keypad can be locked by pressing and holding the STAR * key for about 4 seconds. A lock icon will show indicating the keypad is locked. The default setting is “Yes”.

Note: When the keypad is locked, users would need press and hold the STAR * key for about 4 seconds again and then enter the password to unlock it.

Password to Lock/ Unlock

Configures the password to lock/unlock the keypad.

Validate Server Certificates

Validates server certificates with our trusted list for TLS connections. If set to “No”, device will bypass certificate validation (not recommended).

SIP TLS Certificate

Upload or Delete SSL Certificate used for SIP Transport in TLS/TCP.

SIP TLS Private Key

Upload or Delete SSL Private key used for SIP Transport in TLS/TCP.

SIP TLS Private Key Password

SSL 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 GXP phone default certificate

Web Access Mode

Sets the protocol for Web interface “HTTPS” or “HTTP” or both.

If set to “Disabled”, the Web Access for the phone will be disabled.

The default setting is “HTTP”.

Disable SSH

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

Web/Keypad/Restrict mode Lockout Duration

Specifies the time in minutes that the web or LCD login interface will be locked out to user after five login failures. This lockout time is used for web login, STAR keypad unlock and LCD restrict mode admin login.

Range is 0-60 minutes.

Minimum TLS Version

The function allows users to choose minimum TLS version for HTTPS provisioning and SIP transport. This setting requires reboot to take effect on HTTPS web access. Provisioning and SIP transport do not need reboot.

Maximum TLS Version

The function allows users to choose maximum TLS version for HTTPS provisioning and SIP transport. This setting requires reboot to take effect for HTTPS web access. Provisioning and sip transport do not need reboot.

Enable/Disable Weak Ciphers

Allows users to enable Weak TLS Ciphers Suite or 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. This feature could force the TLS version/Cipher suites for HTTPS provisioning and the TLS version for sip transport (TLS/TCP) and HTTPS web access.

Maintenance 🡪 Security Settings🡪 Trusted CA Certificates

Trusted CA Certificates

Users can specify which CA certificate the phone needs to trust.

– Default Certificates: The phone will verify the server certificate based on the built-in trusted certificates list.

– Custom Certificates: The phone will verify the server certificate based on the custom trusted certificates list.

– All Certificates: The phone will verify the server certificate based on the trusted certificates list including build-in and custom trusted certificates.

Maintenance 🡪 Packet Capture

Capture Location

Choose location where the capture will be stored, either on the internal storage or on the connected USB.

Default is “Internal Storage”.

With RTP Packets

Defines whether packet capture file contains RTP or not.

Default setting is “No”.

USB File Name

Defines the filename of the capture. Only required for USB.

Start/Stop/Download

Click to Start/Stop and Download the packet capture.

Table 10: Maintenance Page Definitions

Phonebook Page Definitions

Phonebook 🡪 Contacts

Group

Specifies to which group the contact belong.

Add Contact

Specifies Contact’s First Name, Last Name, Phone Number, Accounts and Groups to add one new contact in phonebook.

Edit Contact

Edits selected contact.

Delete All Contacts

Deletes all contacts from phonebook.

Phonebook 🡪 Group Management

Add Group

Specifies Group’s name to add new group.

Edit Group

Edits selected group.

Phonebook 🡪 Phonebook Management

Enable Phonebook XML Download

Configures to enable phonebook XML download. Users could select HTTP/HTTPS/TFTP to download the phonebook file. Default setting is “Disabled”.

Note: Receiving a SIP NOTIFY with “Event: sync-contacts” header will trigger a phonebook download from configured server.

The phone will challenge received SIP NOTIFY to get authenticated either using admin credentials (if no SIP account is configured) or using SIP account credentials.

HTTP/HTTPS User Name

The user name for the HTTP/HTTPS server.

HTTP/HTTPS Password

The password for the HTTP/HTTPS server.

Phonebook XML Server Path

Configures the server path to download the phonebook XML. This field could be IP address or URL, with up to 256 characters.

Phonebook Download Interval

Configures the phonebook download interval (in minutes). If it is set to 0, the automatic download will be disabled. Default value is 0. Valid range is 5 to 720 minutes.

Remove Manually-edited Entries on Download

If set to “Yes”, when XML phonebook is downloaded, the entries added manually will be automatically removed. The default setting is “Yes”.

Import Group Method

When set to “Replace”, existing groups will be completely replaced by imported ones. When set to “Append”, the imported groups will be attended with the current ones.

Sort Phonebook by

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

The default setting is “Last Name”.

Download XML Phonebook

Click on “Download” to download the XML phonebook file to local PC.

Upload XML Phonebook

Click on “Upload” to upload local XML phonebook file to the phone.

Phonebook Key Function

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

Phonebook 🡪 LDAP

Server Address

Configures 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 begin.

Example:

dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com

User Name

Configures the bind “Username” for querying LDAP servers. Some LDAP servers allow anonymous binds in which case the setting can be left blank.

Password

Configures the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

LDAP Number Filter

Configures the filter used for number lookups.

Examples:

(|(telephoneNumber=%)(Mobile=%) returns all records which has the “telephoneNumber” or “Mobile” field starting with the entered prefix.

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

LDAP Name Filter

Configures the filter used for name lookups.

Examples:

(|(cn=%)(sn=%)) returns all records which has the “cn” or “sn” field starting with the entered prefix.

(!(sn=%)) returns all the records which do not have the “sn” field starting with the entered prefix.

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

LDAP Version

Selects the protocol version for the phone to send the bind requests.

The default setting is “Version 3”.

LDAP Name Attributes

Specify the “name” attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated name attributes.

Example:

gn

cn sn description

LDAP Number Attributes

Specifies the “number” attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated number attributes.

Example:

telephoneNumber

telephoneNumber Mobile

LDAP Display Name

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

Example:

%cn %sn %telephoneNumber

Max. Hits

Specifies the maximum number of results to be returned by the LDAP server. If set to 0, 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 client waits for server to return. The default setting is 30 seconds.

Sort Results

Specifies whether the searching result is sorted or not. Default setting is “No”.

LDAP Lookup

Configures to enable LDAP number searching when dialing and receiving calls.

Lookup Display Name

Configures the display name when LDAP looks up the name for incoming call or outgoing call. This field must be a subset of the LDAP Name Attributes.

Example:

gn

cn sn description

Table 11: Phonebook Page Definitions

NAT Settings

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

  • STUN Server

Under Settings🡪General Settings, enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the internet and enter it on this field. If using Public IP, keep this field blank.

  • Use Random Ports

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

  • NAT Traversal

It is under Accounts X🡪Network Settings. Default setting is “No”. Enable the device to use NAT traversal when it is behind firewall on a private network. Select Keep-Alive, Auto, STUN (with STUN server path configured too) or other option according to the network setting.

Editing Contacts and Click-To-Dial

From GXP1760/GXP1760W/GXP1780/GXP1782 Web GUI, users could view contacts, edit contacts, or dial out with Click-to-Dial feature on the top of the Web GUI. In the following figure, the Contact page shows all the added contacts (manually or downloaded via XML phonebook). Here users could add new contact, edit selected contact, or dial the contact/number.

Before using the Click-To-Dial feature, make sure the option “Click-To-Dial Feature” under web GUI🡪Settings🡪Call Features is turned on. By default, it is disabled and the dialing icon in web GUI is in grey .

When clicking on the icon on the top menu of the Web GUI, a new dialing window will show for you to enter the number. Once Dial is clicked, the phone will go off hook and dial out the number from selected account. Please see Figure 11 in the following pages for more details.

Additionally, users could directly send the command for the phone to dial out by specifying the following URL in PC’s web browser, or in the field as required in other call modules.

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

In the above link, replace the fields with:

  • ip_address:

Phone’s IP Address.

  • phonenumber=1234:

The number for the phone to dial out.

  • account=0:

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

  • password=admin/123:

The admin login password or user login password of phone’s Web GUI.

Figure 3: Web GUI 🡪 Phonebook🡪Contacts

Figure 4: Click-to-Dial

Saving Configuration Changes

After users makes changes to the configuration, press the “Save” button will save but not apply the changes until the “Apply” button on the top of web GUI page is clicked. Or users could directly press “Save and Apply” button. We recommend rebooting or powering cycle the phone after applying all the changes.

Rebooting from Remote Locations

Press the “Reboot” button on the top right corner of the web GUI page to reboot the phone remotely. The web browser will then display a reboot message. Wait for about 1 minute to log in again.

Packet Capture

GXP1760/GXP1760W/GXP1780/GXP1782 is embedded with packet capture function on firmware 1.0.0.38 or greater. The related options are under Maintenance 🡪 Packet Capture.

Figure 5: Packet Capture in Idle

Users need to choose first the capture location either the internal storage or the connected USB and then press Start button to start packet capture. Press Stop to end capture.

User can press Download button to download capture file to local PC.

Notes:

  • User can also define whether RTP packets will be captured or not using With RTP Packets option.
  • USB Filename is only required for USB and it defines the capture filename.

Multicast Paging

GXP1760/GXP1760W/GXP1780/GXP1782 supports multicast paging, including sending and listening. On the phone, users could send multicast page by setting the multicast address and port. Also, users can listen to at most 10-different multicast IP address.

Multicast sender related settings are under Web UI, Settings 🡪 Programmable keys. Select Multicast paging as the key mode for dial page call. Multicast paging listening related settings are under Web UI Settings 🡪 Multicast Paging.

Configuring Eventlist BLF

Grandstream GXP1760/GXP1760W/GXP1780/GXP1782 Enterprise IP Phones support both Grandstream UCM Busy Lamp Filed and Eventlist BLF features and allows end users, such as attendant, to monitor the call status of users in the list. GXP1760/GXP1760W/GXP1780/GXP1782 supports this feature by sending out the subscription request to the UCM and changing the indicator status of the Line keys, MPKs, or virtual MPKs that associated with the monitored users. Additionally, the phone is also able to pick up the calls to the monitored extensions by using a pre-defined feature code called BLF- Call-pickup Prefix.

For more details, refer to: https://documentation.grandstream.com/knowledge-base/how-to-configure-eventlist-blf/

Configuring VPK Keys

Grandstream GXP1760/GXP1760W/GXP1780/GXP1782 Enterprise IP Phones support VPK (“Virtual Multi-Purpose Keys”) feature to add more flexibility and provide better user experience using Multi-Purpose Keys modes. With the new VPK feature design, line keys can be assigned with multiple functions. Users have also the ability to add more VPKs which will be displayed on different screens.

For more details on VPK configuration, please refer to following guide: VPK Configuration Guide

Outbound Notification Support

Outbound notification option can be found under device web UI🡪Settings🡪Outbound Notifications. In the web UI, under Outbound Notifications: “Action URL” can be found.

To use Outbound Notification🡪Action URL, users need to know the supported events and the dynamic variables for the supported events. The dynamic variables for the supported events will be replaced by the actual values on the phone in order to notify the event to SIP server.

Figure 6: Action URL Settings Page

Supported EventsP-valuesValid Format

Setup Completed

P8304

String

Registered

P8305

Unregistered

P8306

Off Hook

P8308

On Hook

P8309

Incoming Call

P8310

Outgoing Call

P8311

Missed Call

P8312

Established Call

P8313

Terminated Call

P8314

Open DND

P8316

Close DND

P8317

Open Forward

P8318

Close Forward

P8319

Blind Transfer

P8320

Attended Transfer

P8321

Hold Call

P8324

UnHold Call

P8325

Table 12: Action URL – Supported Events


Dynamic VariableDescription

$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 called number on the phone.

$display_local

The display name of the called number on the phone.

$remote

The call number on the remote phone.

$display_remote

The display name of the call number on the remote phone.

$active_user

The account number during a call on the phone.

Table 13: Action URL – Supported Dynamic Variables

After the user finishes setting Action URL on phone’s web UI, when the specific phone event occurs on the phone, 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 on the phone’s web UI🡪Settings🡪Outbound Notification🡪Action URL:

Incoming Call: 192.168.5.250/mac=$mac&display_remote =$display_remote

Outgoing Call: 192.168.5.250/remote=$remote&phone_ip=$phone_ip

On Hold: 192.168.5.250/program_version=$program_version

During incoming call, outgoing call, and call hold, capture the trace on the phone and check the packets. We can see the phone send Action URL with actual values to SIP server to notify phone events. In the following screenshot, from top to bottom, the phone events for each HTTP message are: Incoming Call, hold call and Outgoing Call in the format of the defined action URL with the parameters replaced with actual values.

Figure 7: Action URL Packets

UPGRADING AND PROVISIONING

The GXP1760/GXP1760W/GXP1780/GXP1782 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP or HTTP/HTTPS, the server name can be FQDN or IP address.

Examples of valid URLs:

firmware.grandstream.com

fw.ipvideotalk.com/gs

There are two ways to setup a software upgrade server: The LCD Keypad Menu or the Web Configuration Interface.

Upgrade via Keypad Menu

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

  1. Press MENU button and navigate using Up/Down arrow 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 reboot to upgrade.

Note: If not using DHCP option 66/43, please set the setting to “No”.

When upgrading starts, the screen will show upgrading progress. When done, you will see the phone restarts again. Please do not interrupt or power cycle the phone when the upgrading process is on.

Upgrade via Web GUI

Open a web browser on PC and enter the IP address of the phone. Then, login with the administrator username and password. Go to Maintenance🡪Upgrade and Provisioning page, enter the IP address or the FQDN for the upgrade server in “Firmware Server Path” field and choose to upgrade via TFTP or HTTP/HTTPS. Update the change by clicking the “Save and Apply” button. Then “Reboot” or power cycle the phone to update the new firmware.

When upgrading starts, the screen will show upgrading progress. When done, you will see the phone restart again. Please do not interrupt or power cycle the phone when the upgrading process is on.

Firmware upgrading takes around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. We recommend completing firmware upgrades in a controlled LAN environment whenever possible.

No Local TFTP/HTTP Servers

For users that would like to use remote upgrading without a local TFTP/HTTP server, Grandstream offers a NAT-friendly HTTP server. This enables users to download the latest software upgrades for their phone via this server. Please refer to the webpage:

https://www.grandstream.com/support/firmware

Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A free windows version TFTP server 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:

  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” for the firmware upgrade.
  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.

Note: Please disable any firewall on your PC to ensure proper connection with external devices.

End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server.

Configuration File Download

Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File (binary or XML) through TFTP or HTTP/HTTPS. The “Config Server Path” is the TFTP or HTTP/HTTPS server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server Path” can be the same or different from the “Firmware Server Path”.

A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with the “New Password” in the Web GUI🡪Maintenance🡪Web Access page🡪Admin Password. For a detailed parameter list, please refer to the corresponding firmware release configuration template.

When the GXP1760/GXP1760W/GXP1780/GXP1782 boots up or reboots, it will issue a request to download a configuration XML file named “cfgxxxxxxxxxxxx.xml” followed by a file named “cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the phone, i.e., “cfg000b820102ab.xml” and “cfg000b820102ab”. If the download of “cfgxxxxxxxxxxxx.xml” file is not successful, the provision program will download a generic cfg.xml file. The configuration file name should be in lower case letters.

Note: (Attempt to download Config File again)

When doing provision on the phone, if your first config file contains p-values listed below, phone will try to download the potential second cfg.xml file and apply the second file without rebooting. Maximum 3 extra attempts.

Those P-values are:

*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

Note: (P-values that trigger Auto-Provision)

If the p-values listed below are changed while managing configuration on web UI or LCD, the provision process will be triggered:

* 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.

For more details on XML provisioning, please refer to:

https://www.grandstream.com/hubfs/Product_Documentation/gs_provisioning_guide.pdf?hsLang=en

RESTORE FACTORY DEFAULT SETTINGS

Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup or print all the settings before you restore to the factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.

Please follow the instructions below to reset the phone:

  1. Press MENU button to bring up the keypad configuration menu.
  2. Select “System” and enter.
  3. Select “Factory Reset”.
  4. A warning window will pop out to make sure a reset is requested and confirmed.
  5. Press “OK” softkey to confirm and the phone will reboot. To cancel the Reset, press “Back” softkey instead.

CHANGE LOG

This section documents significant changes from previous versions of administration guide for GXP1760/GXP1760W/GXP1780/GXP1782. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.

Firmware Version 1.0.1.123

  • No major changes.

Firmware Version 1.0.1.119

  • No major changes.

Firmware Version 1.0.1.118

  • No major changes.

Firmware Version 1.0.1.116

  • No major changes.

Firmware Version 1.0.1.106

  • Added ability to configure Minimum TLS version. [Minimum TLS Version]
  • Added ability to configure Maximum TLS Version. [Maximum TLS Version]
  • Add ability to Enable/Disable Weak TLS Ciphers suites. [Enable/Disable Weak Ciphers]
  • Added Support of Central Africa Time zone “CAT”. [Time Zone]

Firmware Version 1.0.1.98

  • Add Callee ID display. [Callee ID Display]
  • Added Ability to specify which certificate to trust. [Trusted CA Certificates]
  • Added Support of more custom ringtones. [Account Ring Tone]
  • Added Support of Pound (#) key acting as redial. [Use Pound (#) For Redial]
  • Added Support to upload SIP TLS Private Key from web UI. [SIP TLS Certificate]
  • Added Support to flash MWI LED for missed call notification. [Disable Missed Call Backlight]
  • Added Support of locked mode for configuration via keypad. [Locked Mode]
  • Added Ability to display Diversion info. [Enable Diversion Information Display]
  • Increased the size of display name on LCD. [Use Long Label]
  • Added Support of Auth Header on Initial REGISTER. [Add Auth Header On Initial Register]
  • Added Support to hide VPK information when dialing. [Show Key Label]
  • Added Support to enable or disable Acoustic Echo Cancellation. [Enable Enhanced Acoustic Echo Canceller]
  • Added Support of “Provision” mode for VPK/Softkey to trigger provisioning. [Provision]
  • Added Ability to download certificate files during provisioning.
  • Added Support to allow more special characters to be used in SIP account via keypad. [SIP User ID]
  • Added Ability to allow @ character in SIP user ID. [SIP User ID]
  • Added Support to configure two Passwords for the same GDS. [Grandstream Door System]

Firmware Version 1.0.1.89

  • Added Support to edit VPKs using device’s keypad [Configuring VPK Keys]

Firmware Version 1.0.1.82

  • Added Support for the new OUI (C0:74:AD). (Note: GXP1760W with new MAC address will not be able to downgrade to previous versions)

Firmware Version 1.0.1.79

  • Added Support to prevent users from accessing the network settings. [Configuration via Web Browser]
  • Added Support to synchronize phonebook through SIP Notify. [SIP NOTIFY]

Firmware Version 1.0.1.74

  • Added Support to auto redial if callee is busy. [Enable Automatic Redial]
  • Added Support to configure VPK/Softkey to launch provisioning process. [Provision]
  • Added Headset gain options. [Headset TX gain] [Headset RX gain]
  • Improved to display more Cyrillic characters on local phonebook of phone’s LCD screen.
  • Added Option to choose either to override or accumulate groups when uploading a new XML phonebook. [Import Group Method]
  • Added Support to configure device with custom certificate signed by custom CA certificate. [Custom Certificate]
  • Added More fields for Distinctive Ring Tone. [Match Incoming Caller ID]
  • Added Support to unmute device manually if it is muted manually. [Enable Auto Unmute]
  • Improved Device to support 802.1x with special symbols such as “-” and “@”. [802.1X Identity] [MD5 Password]
  • Added Support to dial an abbreviated number instead of the whole number for BLF call pick-up.
  • Added Support more characters for password input from LCD: “<,>,?, {, }, [, ], ^” using key “1”.

Firmware Version 1.0.1.64

  • Enhanced syslog to run on other ports instead of default port. [Syslog Server]
  • Added Auto provision starts when certain p-values are changed. [P-values that trigger Auto-Provision]
  • Added Attempt to download config files. [Attempt to download Config File]
  • Added support for 3CX Auto Provision. [3CX Auto Provision]
  • Added Support for TLS version 1, 1.1, and 1.2 for SIP.
  • Add HTTP/HTTPS Username/Password fields for firmware and provisioning paths in LCD menu. [HTTP/HTTPS Server]
  • Added Dial plan flag T support. [Flag T]
  • Add Country code modification.
  • Added option in LCD menu to test Wi-Fi Signal Strength for GXP1760W. [Wi-Fi Signal Strength]

Firmware Version 1.0.1.58

  • Added popup at first web GUI access to notify the admin to change the default password. [Administrator Password]
  • Added Let’s Encrypt Root CA to Trusted CA.
  • Added SIP support for 3CX hot desking feature.
  • Added Host name of the phone on the DHCP INFORM using DHCP option 12.
  • Added automatically log in web UI from 3CX server interface.
  • Added “Account Display” option. [Account Display]
  • Added MAC header support in SIP header. [Use MAC Header]
  • Added Test Password Strength. [Test Password Strength]
  • Added ability to enable/disable Ring for Call Waiting. [Ring for Call Waiting]
  • Added a timeout for the Active Backlight. [Active Backlight Timeout]
  • Added Set Config upgrade via HTTPS & Firmware upgrade via HTTP as default values.
  • Added ability to enable/disable VM/MSG power light flash. [Disable VM/MSG power light flash]
  • Added option to specify the transfer mode via VPK. [Transfer Mode via VPK]
  • Added Enable/Disable for transfer via non-transfer MPK. [transfer via non-Transfer MPK]
  • Added option to enable/disable CSTA Control. [CSTA Control]
  • Added option to randomize the Automatic Upgrade. [Randomized Automatic Upgrade]
  • Added option to schedule the Automatic Upgrade for a period instead of a fixed hour of the day. [Hour of the Day (0-23)]
  • Added ability to validate server certificates. [Validate Server Certificates]
  • Added option to get web access via both HTTP and HTTPS. [Web Access Mode]
  • Added new “use SBC” account networking configuration option to control whether phone sends out BYE SIP message right after it receives INVITE with Replaces header in call queue scenario.
  • Added Send SIP unregister for all accounts when factory reset the phone.

Firmware Version 1.0.1.30

  • This is the initial version for GXP1760W.

Firmware Version 1.0.1.28

  • Added Call Hold reminder option. [On Hold Reminder Tone]
  • Added notification tone volume configuration in web UI. [Notification Tone Volume]
  • Added support to provisioning “Trusted CA Certificates”. [Trusted CA Certificates]
  • Added GDS Surveillance Action Key Support Integration. [Grandstream Door System]
  • Added Static/Dynamic auto-attended transfer support. [Attended Transfer Mode]
  • Added option to enable/disable web UI access. [Web Access Mode]

Firmware Version 1.0.1.8

  • Added “Account Mode”. [Account Mode]
  • Added ability to disable PANI and PEI header. [Custom SIP Headers]
  • Added “Subscribe Expiration” option. [Subscribe Expiration]
  • Added support for provisioning “Trust CA Certificate”. [Trusted CA Certificates]
  • Added Pvalue for VPK mode in string format support in XML configuration file.

Firmware Version 1.0.0.48

  • Allow certificate upload for OpenVPN®. [OpenVPN® Settings]
  • Added OpenVPN® copyright and trademark.
  • Added support for concurrent G.729 calls. [Preferred Vocoder].
  • Added option for Predictive Dial Feature. [Predictive Dialing Feature].

Firmware Version 1.0.0.38

  • No major changes.

Firmware Version 1.0.0.37

  • No major changes.

Firmware Version 1.0.0.13

  • This is the initial version for GXP1780/GXP1782.

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