Grandstream Networks, Inc.
GXP1610/GXP1615
GXP1620/GXP1625
GXP1628/GXP1630
Small Business IP Phone
Administration Guide
GXP16XX Administration Guide
Version 1.0.7.13
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COPYRIGHT
©2021 Grandstream Networks, Inc. http://www.grandstream.com
All rights reserved. Information in this document is subject to change without notice. Reproduction or
transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without
the express written permission of Grandstream Networks, Inc. is not permitted.
The latest electronic version of this guide is available for download here:
http://www.grandstream.com/support
Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc.
in the United States, Europe, and other countries.
CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this
product in any way other than as detailed by this guide, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with devices as it may cause damage to the products and void
the manufacturer warranty.
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GNU GPL INFORMATION
GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 firmware contains third-party software
licensed under the GNU General Public License (GPL). Grandstream uses software under the specific
terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of
the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
http://www.grandstream.com/support/faq/gnu-general-public-license/gnu-gpl-information-download
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DOCUMENT PURPOSE
This document describes how to configure GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630
features via phone’s LCD menu and Web GUI menu. The intended audiences of this document are phone
administrators.
To learn the basic functions of GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630, please visit
http://www.grandstream.com/support to download the latest “GXP16XX series User Guide”.
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Table of Content
DOCUMENT PURPOSE ................................................................................................. 3
CHANGE LOG ................................................................................................................ 9
Firmware Version 1.0.7.13 ..................................................................................................................... 9
Firmware Version 1.0.7.6 ....................................................................................................................... 9
Firmware Version 1.0.7.3 ....................................................................................................................... 9
Firmware Version 1.0.5.3 ....................................................................................................................... 9
Firmware Version 1.0.4.152 ................................................................................................................... 9
Firmware Version 1.0.4.140 ................................................................................................................. 10
Firmware Version 1.0.4.132 ................................................................................................................. 10
Firmware Version 1.0.4.128 ................................................................................................................. 10
Firmware Version 1.0.4.106 ................................................................................................................. 11
Firmware Version 1.0.4.100 ................................................................................................................. 11
Firmware Version 1.0.4.88 ................................................................................................................... 12
Firmware Version 1.0.4.82 ................................................................................................................... 12
Firmware Version 1.0.4.65 ................................................................................................................... 12
Firmware Version 1.0.4.55 ................................................................................................................... 12
Firmware Version 1.0.4.46 ................................................................................................................... 13
Firmware Version 1.0.4.33 ................................................................................................................... 13
Firmware Version 1.0.4.22 ................................................................................................................... 13
Firmware Version 1.0.4.17 ................................................................................................................... 13
Firmware Version 1.0.4.6 ..................................................................................................................... 14
Firmware Version 1.0.3.28 ................................................................................................................... 14
Firmware Version 1.0.3.6 ..................................................................................................................... 14
Firmware Version 1.0.2.29 ................................................................................................................... 14
Firmware Version 1.0.2.26 ................................................................................................................... 14
Firmware Version 1.0.2.24 ................................................................................................................... 14
Firmware Version 1.0.2.22 ................................................................................................................... 14
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Firmware Version 1.0.2.6 ..................................................................................................................... 15
Firmware Version 1.0.2.4 ..................................................................................................................... 15
Firmware Version 1.0.1.14 ................................................................................................................... 15
Firmware Version 1.0.1.10 ................................................................................................................... 15
GUI INTERFACE EXAMPLES ...................................................................................... 16
WELCOME ................................................................................................................... 17
PRODUCT OVERVIEW ................................................................................................ 19
Feature Highlights (GXP1610/GXP1615/GXP1620/GXP1625/GXP1628) .......................................... 19
Feature Highlights (GXP1630) ............................................................................................................. 19
GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 Technical Specifications .................. 20
CONFIGURATION GUIDE ............................................................................................ 26
Configuration via Keypad ..................................................................................................................... 26
Configuration via Web Browser ........................................................................................................... 32
Definitions ............................................................................................................................................ 33
Status Page Definitions ................................................................................................................ 33
Account Page Definitions ............................................................................................................. 35
Settings Page Definitions ............................................................................................................. 51
Network Page Definitions ............................................................................................................. 68
Maintenance Page Definitions ...................................................................................................... 73
Contacts Page Definitions ............................................................................................................ 83
NAT Settings ........................................................................................................................................ 87
Public Mode ......................................................................................................................................... 87
Lock Settings on the Phone ................................................................................................................. 87
DHCP VLAN ........................................................................................................................................ 88
Ability to Configure Record Button ....................................................................................................... 89
Editing Contacts and Click-To -Dial ...................................................................................................... 89
Saving the Configuration Changes ...................................................................................................... 91
Action URL ........................................................................................................................................... 91
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Rebooting from Remote Locations ...................................................................................................... 97
UPGRADING AND PROVISIONING ............................................................................ 98
Upgrade via Keypad Menu .................................................................................................................. 98
Upgrade via Web GUI .......................................................................................................................... 98
No Local TFTP/HTTP Servers ............................................................................................................. 99
Configuration File Download ................................................................................................................ 99
Certificates and Keys Provisioning .................................................................................................... 101
RESTORE FACTORY DEFAULT SETTINGS ............................................................. 102
Restore to Factory Default via the LCD Menu ................................................................................... 102
Restore to Factory Default via the Web GUI ..................................................................................... 102
EXPERIENCING THE GXP16XX SERIES .................................................................. 103
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Table of Tables
Table 1: GXP1610/GXP1615 Technical Specifications ............................................................................... 20
Table 2: GXP1620/GXP1625 Technical Specifications ............................................................................... 21
Table 3: GXP1628 Technical Specifications ................................................................................................ 23
Table 4: GXP1630 Technical Specifications ................................................................................................ 24
Table 5: GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 Menu Options ............................ 26
Table 6: User / Admin Levels Information ................................................................................................... 32
Table 7: Status Page Definitions ................................................................................................................. 33
Table 8: Account Page Definitions .............................................................................................................. 35
Table 9: Settings Page Definitions .............................................................................................................. 51
Table 10: Network Page Definitions ............................................................................................................ 68
Table 11: Maintenance Page Definitions ..................................................................................................... 73
Table 12: Contacts Page Definitions ........................................................................................................... 83
Table 13: Action URL - Supported Events ................................................................................................... 94
Table 14: Action URL - Supported Dynamic Variables ................................................................................ 94
Table 15: Action URL Parameters P-values ................................................................................................ 95
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Table of Figures
Figure 1: Keypad Menu Flow ...................................................................................................................... 31
Figure 2: Change Password on first boot .................................................................................................... 33
Figure 3: Lock/Unlock Configuration via Keypad Menu .............................................................................. 88
Figure 4: Enable DHCP VLAN .................................................................................................................... 88
Figure 5: DHCP Option 132 & 133 .............................................................................................................. 89
Figure 6: Call Recording Feature Codes .................................................................................................... 89
Figure 7: Web GUI→Phonebook→Contacts ............................................................................................... 90
Figure 8: Click-to-Dial .................................................................................................................................. 91
Figure 9: Action URL Settings Page ............................................................................................................ 93
Figure 10: Action URL Packet ..................................................................................................................... 95
Figure 11: Rebooting the phone via web GUI ............................................................................................. 97
Figure 12: Certificates Files Download ..................................................................................................... 101
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CHANGE LOG
This section documents significant changes from previous versions of user manuals for
GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630. Only major new features or major
document updates are listed here.
Firmware Version 1.0.7.13
• No major changes
Firmware Version 1.0.7.6
• Added Ability to configure variables in the provisioning server URL [Config Server Path][Firmware
Server Path]
Firmware Version 1.0.7.3
• Added Ability to auto answer pre-defined numbers. [Auto Answer Numbers]
• Added Two accounts support on GXP1610/1615. [Table 1: GXP1610/GXP1615 Technical
Specifications]
Firmware Version 1.0.5.3
• Added support of Beirut Time zone. [Time Zone]
• Added Support of Central Africa Time zone “CAT”. [Time Zone]
• Added ability to choose minimum TLS version. [Minimum TLS Version]
• Added ability to choose maximum TLS version. [Maximum TLS Version]
• Added ability to Enable/Disable weak ciphers suites. [Enable/Disable Weak Ciphers]
Firmware Version 1.0.4.152
• Added GNU GPL License information display on LCD and Web UI [GNU GPL License]
• Added Ability to scroll long names on the screen to display the full name. [Caller ID Display Mode]
• Added Support to input username and password via keypad for BroadSoft’s directories. [Login
Credentials]
• Added Support of “Provision” mode for VPK/MPK to trigger provisioning [Provision]
• Added Ability to allow @ character in SIP user ID [SIP User ID]
• Added Support of Auth Header on Initial REGISTER [Add Auth Header On Initial REGISTER]
• Add Support to include DHCP Option 12 into DHCP Inform message [Host name (Option 12)]
• Increased SIP Certificate length limitation from 2048 characters to 8192 [SIP TLS Certificate]
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Firmware Version 1.0.4.140
• Added Support for Emergency call under functional lock [functional lock]
• Added Support for callee ID display [callee id display]
• Added ability to create Custom Call Screen Softkey Layout [Custom Call Screen Softkey Layout]
• Added ability to customize the feature code [Feature Codes]
• Added option to specify which certificate to trust [load ca certificates]
• Added show both call session timer and hold duration timer on LCD during call hold. [Show on Hold
Duration]
• Added more custom ringtone choices. [Default Ringtone]
• Added option to disable # key from acting as redial key. [Use Pound (#) For Redial]
• Added support for variable $active_url in Action URL. [action URL]
• Added upload button to upload SIP TLS Private Key. [SIP TLS Certificate] [SIP TLS PRIVATE KEY]
• Added option to hide reject softkey on ringing state [Hide/Disable reject key]
• Added option "Disable Busy Tone on Remote Disconnect" [disable busy tone]
• Added option to specify DNS SRV query frequency. [DNS Cache Refresh Time]
• Added ability to open door in idle stage for GDS integration. [GDS OpenDoor]
• Added ability to use special characters on 802.1X MD5 password. [MD5 Password]
• Added option to choose CID display behavior (static or dynamic). [Caller ID Display Mode]
• Added option to enable/disable Diversion info display. [Enable Diversion Information Display]
• Changed default value for “Enable CDP” to be “Enabled” instead of “Disabled”. [Enable CDP]
• Added options to disable hook switch. [disable hook switch]
Firmware Version 1.0.4.132
• Increased phonebook limit to 1000 entries. [PRODUCT OVERVIEW]
Firmware Version 1.0.4.128
• Added Support for SNMPv3 [SNMP]
• Added FTP/FTPS support for provisioning and firmware upgrade [Config Upgrade Via] [Firmware
Upgrade Via] [UPGRADING AND PROVISIONING]
• Added support to configure device with custom certificate signed by custom CA certificate [custom
CA certificate]
• Added support for local packet capture. [Packet Capture]
• Added support for Group Listening with Speaker [Group Listening with Speaker]
• Added support for more Action URL notification [Action URL]
• Added support for XML Application [XML Application]
• Added support for RTCP-XR. [GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630
Technical Specifications]
• Added Broadsoft call park notification feature [Broadsoft Call Park]
• Added Codec Negotiation Priority feature [Codec Negotiation Priority]
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• Added support for "tftp://" in DHCP Option 66 [Allow DHCP Option 43 and Option 66 to Override
Server]
• Added support to customize idle time to logout the web access [Web Session Timeout]
• Added support to customize number of failed attempts to access web GUI [Web Access Attempt
Limit]
• Added support for BLF call pick-up to go through dial plan [Bypass Dial Plan]
• Added support of on hook dialing with Live DialPad [Enable Live DialPad]
• Added support for CID and Image from SIP NOTIFY [Caller ID Display]
• Added support to randomize TR069 Inform Interval [Randomized TR069 Startup]
• Added support to configure off-hook auto dial delay [Off-hook Auto Dial Delay]
• Added support for auto answer delay [Auto Answer Delay]
• Added more fields for Distinctive Ring Tone. [Match Incoming Caller ID]
• Add support to enable or disable Acoustic Echo Cancellation. [Enable Enhanced Acoustic Echo
Canceller]
• Add support to download certificate files during provisioning. [Certificates and Keys Provisioning]
• Added support of on hold reminder tone [On Hold Reminder Tone]
• Updated phonebook string in LCD and Web UI to “Contacts” [Contacts] [Contacts Page Definitions]
Firmware Version 1.0.4.106
• Enhanced Syslog to run on other ports instead of default port. [Syslog Server]
• Enhanced Intercom options (including Intercom Mute, Intercom Tone). [Intercom]
• 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 again]
• Added “Provision” button to trigger Provision Requests. [Provision]
Firmware Version 1.0.4.100
• Added “Import Group Method” to append or replace existing groups. [Import Group Method]
• Added option to validate server certificates with our trusted list for TLS connections. [Validate server
certificates]
• Added option to enable or disable user web access. [Enable User Web Access]
• Added support for Cisco Discovery Protocol. [Enable CDP]
• Added option to enable CSTA Control. [CSTA Control]
• Added option to test strength of password. [Test Password Strength]
• Added option to configure the timeout of the active backlight. [Active Backlight Timeout]
• Added option to specify the transfer mode via VPK. [Transfer Mode via MPK].
• Added support for transfer when using a non-transfer MPK. [Enable Transfer via non-Transfer MPK]
• Added support for SBC. [Use SBC]
• Added option to include the MAC address on the SIP message header. [Use MAC Header]
• Force admin to change default password upon first time login. [Password on first boot]
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• Added support for HTTPS server based on TLSv1.2.
• Added monitored call park MPK option and supported to be provisioned by 3CX.
• Added support to automatically log in web UI from server interface for 3CX.
• Added “Let's Encrypt” Validation for 3CX.
Firmware Version 1.0.4.88
• No major changes.
Firmware Version 1.0.4.82
• Added “Account Display” option to configure SIP account display label on LCD. [Account Display]
• Added “Disable VM/MSG Power Light Flash” option to enable/disable voicemail/message
indication. [Enable / Disable VM/MSG Power Light Flash]
• Added web UI option to upload SSH public key for SSH access. [SSH Public Key]
• Added “Ring for Call Waiting” option to enable/disable ringing the speaker phone on call waiting.
[Ring For Call Waiting]
• Added call forwarding configuration on web UI. [Call Forward]
• Added support for static/dynamic auto-attended transfer. [Attended Transfer Mode]
• Added support for BLF server. [BLF Server]
• Added automatic redial support. [Automatic Redial]
• Added blacklist/whitelist support in phonebook. [Blacklist, Whitelist]
• Added option to disable hook switch. [Disable Hook Switch]
• Added notification tone value configuration [Notification tone volume]
• Added SNMP support. [SNMP]
• Added support for Genesys Agent Login/Logout and status update. [Enable User Presence
Subscription]
• Improved slow performance issue after phone is used for some time.
• Added Let’s Encrypt Root CA to Trusted CA.
• Increased max length of dial plan from 512 to 1024 characters.
Firmware Version 1.0.4.65
• Added support for Syslog over TLS [Syslog Protocol]
• Added support of Provisioning "Trusted CA Certificates" [Trusted CA certificates]
• Added option to select the method for caller information display [Remote Info Display]
• Added support to enable/disable IPv4/IPv6 [Internet Protocol]
• GDS37XX OpenDoor Key support integration [Grandstream Door System]
Firmware Version 1.0.4.55
• Allow certificate upload for OpenVPN. [Open VPN® Settings]
• Added option to sort Broadsoft phonebook by Last/First name. [Sort Phonebook]
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• Added ability to lock the phone ringing volume. [Lock Speaker Volume]
• Added ability to customize the domain name on the XSI request. [XSI Actions Path]
• Changed OpenVPN to OpenVPN® for web page and added the trademark. [Open VPN® Settings]
• Added HTTP/HTTPS Authentication for phonebook XML file download. [HTTP/HTTPS ;
HTTP/HTTPS Password]
• Updated syslog message to include MAC address and firmware version. [Syslog]
Firmware Version 1.0.4.46
• Added support for concurrent G.729 calls.
• Added option to enable/disable web UI access. [Web Access Mode]
• Added option to hide the password on the display after dialing a specific prefix. [Hide Dialing
Password]
• Added LDAPS support. [LDAP protocol]
• Added support for certificate DN validation for LDAPS.
• Added option to remove SIP error on LCD. [Show SIP Error Response]
Firmware Version 1.0.4.33
• Added support of intercom button to send multicast paging. [Intercom Key Mode]
• Added option to disable custom SIP header. [Custom SIP Headers]
• Added support to send SIP log without enabling debug level. [Send SIP Log]
• Added option to bypass firmware upgrade confirmation. [Firmware Upgrade Confirmation]
Firmware Version 1.0.4.22
• Added ability to lock settings on phone. [Maintenance → Security] [Lock Settings on the Phone]
• Added support for DHCP option 132 and 133 tunneled through DHCP option 43. [DHCP VLAN]
• Added support for DHCP option 132 for (802.1Q VLAN ID) and 133 (QOS priority level).
• Added ability to adjust ringtone level from web GUI. [Default Ringtone]
• Added support for separated QoS setting for SIP and RTP.
• Added configurable web access port. [Maintenance → Security]
• Added ability to configure record button. [Ability to Configure Record Button]
Firmware Version 1.0.4.17
• Added ability to set forward on busy and on no answer from the LCD.
• Added ability to display more characters for account name on LCD display. [Wide Idle Screen View]
• Added option for predictive dial feature. [Predictive Dialing Feature]
• Added option to adjust the Maximum Transmission Unit (MTU) of IP packets. [Maximum
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Transmission Unit (MTU)]
• Spited register and subscription expiration settings. [Subscribe Expiration]
• Added ability to hide web access mode and disable SSH from LCD menu under constraint mode.
[Configuration via Keypad Menu]
• Add support for Latvian Language.
• Update Polish language.
Firmware Version 1.0.4.6
• Added ability to customize the system softkey. [Softkeys]
• Added ability to hide system softkey on main screen.
• Added support of configurable User Agent field. [User-Agent Prefix]
Firmware Version 1.0.3.28
• Added option to disable recovery on blind transfer. [Disable Recovery on Blind Transfer]
• Added call-pickup barge-in code. [Call Pickup Barge-in Code]
• Added options to hide remote user ID based on both call type and length. [Hide Remote User ID]
• Added option to set BroadSoft’s contacts download interval. [XSI]
Firmware Version 1.0.3.6
• Added G723.1 and iLBC codecs support. [Preferred Vocoder]
• Added Event Notification. [Maintenance]
• Added options to specify Broadsoft XSI authentication method. [XSI]
Firmware Version 1.0.2.29
• Added DHCP option 150/160 support. [Additional Override DHCP option]
Firmware Version 1.0.2.26
• Added ability to filter characters from dial numbers.
Firmware Version 1.0.2.24
• Web logo and style changed.
Firmware Version 1.0.2.22
• Added DHCP Option 125 support.
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Firmware Version 1.0.2.6
• Enabled/Disabled Broadsoft Call Park subscribe. [Broadsoft Call Park]
• Separate Firmware/Config’s upgrade via, HTTP/HTTPS username, and HTTP/HTTPS password.
[Maintenance → Upgrade and Provisioning]
Firmware Version 1.0.2.4
• Ignore Alert-Info Header. [Ignore Alert-Info Header]
• Validate Certification Chain. [Validate Certification Chain]
• Bypass Dial Plan through Call History and Directories. [Bypass Dial Plan through Call History and
Directories]
Firmware Version 1.0.1.14
• Backup Outbound Proxy. [Backup Outbound Proxy]
• Enable OPTIONS Keep Alive. [Enable OPTIONS Keep Alive]
• OPTIONS Keep Alive Interval. [OPTIONS Keep Alive Interval]
• OPTIONS Keep Alive Max Lost. [OPTIONS Keep Alive Max Lost]
Firmware Version 1.0.1.10
This is the initial version.
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GUI INTERFACE EXAMPLES
http://www.grandstream.com/sites/default/files/Resources/gxp16xx_web_gui.zip
1. Screenshot of Login Page.
2. Screenshots of Status Pages.
3. Screenshots of Accounts Pages.
4. Screenshots of Settings Pages.
5. Screenshots of Network Pages.
6. Screenshots of Maintenance Pages.
7. Screenshots of Contacts Pages.
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WELCOME
Thank you for purchasing Grandstream Small Business IP phone GXP1610/GXP1615/GXP1620/
GXP1625/GXP1628/GXP1630. GXP16xx series is a next generation small-to-medium business IP phone:
the GXP1620/GXP1625 features 2 lines with 2 SIP accounts, superior HD wideband audio, 132 x 48 backlit
graphical LCD; the GXP1610/GXP1615 supports 2 lines with 2 SIP accounts, 132 x 48 graphical LCD;
GXP1630 supports 3 lines with 3 SIP accounts, 132x64 backlit graphical LCD. The GXP1610/GXP1615/
GXP1620/GXP1625/GXP1628/GXP1630 delivers rich features of 3 XML programmable context-sensitive
softkeys, dual network ports with PoE (GXP1625/GXP1628/GXP1630 only), EHS (Electronic Hook-Switch)
with Plantronics headsets, 3-ways conference on GXP1610/GXP1615/GXP1620/GXP1625/GXP1628 and
up to 4-ways conference on GXP1630, superb full-duplex hands-free speakerphone with advanced acoustic
echo cancellation, advanced security protection for privacy, and compatible with Grandstream UCM
Features. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP
phone with affordable cost.
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Caution:
Changes or modifications to this product not expressly approved by Grandstream, or operation of this
product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
Warning:
Please do not use a different power adaptor with the GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/
GXP1630 as it may cause damage to the products and void the manufacturer warranty.
This document is subject to change without notice. The latest electronic version of this user manual is
available for download here:
http://www.grandstream.com/support
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for
any purpose without the express written permission of Grandstream Networks, Inc is not permitted.
FCC Caution:
Any Changes or modifications not expressly approved by the party responsible for compliance could void
the user’s authority to operate the equipment. This device complies with part 15 of the FCC Rules.
Operation is subject to the following two conditions: (1) This device may not cause harmful interference,
and (2) this device must accept any interference received, including interference that may cause undesired
operation.
Note: This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against
harmful interference in a residential installation. This equipment generates uses and can radiate radio
frequency energy and, if not installed and used in accordance with the instructions, may cause harmful
interference to radio communications. However, there is no guarantee that interference will not occur in an
installation. If this equipment does cause harmful interference to radio or television reception, which can be
determined by turning the equipment off and on, the user is encouraged to try to correct the interference by
one or more of the following measures:
- Reorient or relocate the receiving antenna.
- Increase the separation between the equipment and receiver.
- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
- Consult the dealer or an experienced radio/TV technician for help.
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PRODUCT OVERVIEW
Feature Highlights (GXP1610/GXP1615/GXP1620/GXP1625/GXP1628)
• 132*48 pixels backlit graphical LCD display (GXP1610/GXP1615 do not support backlit)
• 2 dual-color line keys, 3 XML programmable context-sensitive soft keys, 8 BLF keys (GXP1628
only), 3-way conference, multi-language support
• HD wideband audio (GXP1620/GXP1625/GXP1628 only), superb full-duplex hands-free
speakerphone with advanced acoustic echo cancellation and excellent double-talk performance
• Large phonebook (up to 1000 contacts) and call history (up to 200 records)
• Automated personal information service (e.g., local weather, etc.), personalized music ring
tone/ring back tone, flexible customizable screen content & format using XML, and advanced Web
and enterprise applications, local weather service
• Dual switched 10/100 Mbps ports on GXP1610/GXP1615/GXP1620/GXP1625, dual switched autosensing 10/100/1000 Mbps Ethernet ports on GXP1628, integrated PoE on
GXP1615/GXP1625/GXP1628
• Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for
advanced security protection, 802.1x for media access control
• Perfectly compatible with Grandstream UCM Features
• Innovative UI design (easy and useable multi-language support)
• Use with Grandstream’s UCM6xxx series IP PBX appliance for Zero-Config provisioning, 1-touch
call recording and more
Feature Highlights (GXP1630)
• 132*64 pixels backlit graphical LCD display
• 3 dual-color line keys, 3 XML programmable context-sensitive soft keys, 8 BLF keys, 4-way
conference, multi-language support
• HD wideband audio, superb full-duplex hands-free speakerphone with advanced acoustic echo
cancellation and excellent double-talk performance
• Large phonebook (up to 1000 contacts) and call history (up to 200 records)
• Automated personal information service (e.g., local weather, etc.), personalized music ring
tone/ring back tone, flexible customizable screen content & format using XML, and advanced Web
and enterprise applications, local weather service
• Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports with integrated PoE
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• Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for
advanced security protection, 802.1x for media access control
• Perfectly compatible with Grandstream UCM Features
• Innovative UI design (easy and useable multi-language support)
• Use with UCM6xxx IP PBX appliance for Zero-Config provisioning, 1-touch call recording and more
GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 Technical Specifications
Table 1: GXP1610/GXP1615 Technical Specifications
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS,
FTP/FTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE,
SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS,
SRTP
Network Interfaces
Dual switched 10/100 Mbps ports, integrated PoE (GXP1615 only)
Graphic Display
132*48 LCD display
Feature Keys
2 line keys with dual-color LED and 2 SIP accounts. 3 XML programmable
context sensitive soft keys. 5 (navigation, menu) keys. 13 dedicated function
keys for PAGE/INTERCOM, PHONEBOOK, MESSAGE, HOME, HOLD,
RECORD, MUTE, HEADSET, TRANSFER, CONFERENCE, SEND and
REDIAL, SPEAKERPHONE, VOLUME
Voice Codecs
Support for G.711µ/a, G.722 (wide-band), G.723.1, G.726-32, G.729 A/B,
iLBC, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO)
Telephony Features
Hold, transfer, forward (unconditional/no-answer/busy), 3-way conference,
call park/pickup, downloadable phone book (XML, LDAP, up to 1000 items),
call waiting, call history (up to 200 records), off-hook auto dial, auto answer,
click-to-dial, flexible dial plan, Hot Desking, personalized music ringtones,
server redundancy & fail-over
Headset Jack
RJ9 headset jack (allowing EHS with Plantronics headsets)
Base Stand
Yes, allow 2 angle positions available, Wall Mountable
QoS
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level access control, MD5 and MD5-sess based
authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS,
802.1x media access control
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Multi-language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
simplified and traditional Chinese, Korean, Japanese, and more
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS/ FTP/FTPS, mass provisioning
using TR-069 or AES encrypted XML configuration file
Power & Green Energy
Efficiency
Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA
Temperature and
Humidity
Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity: 10% to 90% Non-condensing
Physical
Dimension: 209mm（L）x 184.5mm（W）x 76.2mm（H） (with handset)
Unit Weight: 0.74kg Package Weight: 1.1kg
Package Content
GXP1610/GXP1615 phone, handset with cord, base stand, universal power
supply, network cable, Quick Installation Guide, brochure, GPL license
Compliance
FCC: Part 15 (CFR 47) Class B
CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1
Table 2: GXP1620/GXP1625 Technical Specifications
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS,
FTP/FTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE,
SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS,
SRTP
Network Interfaces
Dual switched 10/100 Mbps ports, integrated PoE (GXP1625 only)
Graphical Display
132*48 pixel backlit graphical LCD display
Feature Keys
2 line keys with dual-color LED and 2 SIP accounts. 3 XML programmable
context sensitive soft keys. 5 (navigation, menu) keys. 13 dedicated function
keys for MUTE, HEADSET, TRANSFER, CONFERENCE, SEND and
REDIAL, SPEAKERPHONE, VOLUME, PHONEBOOK, MESSAGE, HOLD,
PAGE/INTERCOM, RECORD, HOME
Voice Codecs
Support for G.711µ/a, G.722 (wide-band), G.723.1, G.726-32, G.729 A/B,
iLBC, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO)
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Telephony Features
Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup, 3way conference, shared-call-appearance (SCA) / bridged-line-appearance
(BLA), downloadable phone book (XML, LDAP, up to 1000 items), call waiting,
call history (up to 200 records), off-hook auto dial, auto answer, click-to-dial,
flexible dial plan, Hot Desking, personalized music ringtones, server
redundancy & fail-over.
Headset Jack
RJ9 headset jack (allowing EHS with Plantronics headsets)
HD Audio
Yes, HD handset and speakerphone with support for wideband audio
Base Stand
Yes, 2 angle positions available, Wall Mountable
QoS
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level access control, MD5 and MD5-sess based
authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS,
802.1x media access control
Multi-language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
simplified and traditional Chinese, Korean, Japanese, and more
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS /FTP/FTPS, mass provisioning
using TR-069 or AES encrypted XML configuration file
Power & Green Energy
Efficiency
Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA
PoE IEEE802.3af Class 2, 3.84W-6.49W (GXP1625 only)
Temperature and
Humidity
Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity: 10% to 90% Non-condensing
Physical
Dimension: 209mm（L）x 184.5mm（W）x 76.2mm（H） (with handset)
Unit Weight: 0.73kg Package Weight: 1.1kg
Package Content
GXP1620/GXP1625 phone, handset with cord, base stand, universal power
supply, network cable, Quick Installation Guide, brochure, GPL license
Compliance
FCC: Part 15 (CFR 47) Class B
CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1
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Table 3: GXP1628 Technical Specifications
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS,
FTP/FTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE,
SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS,
SRTP
Network Interfaces
Dual switched 10/100/1000 Mbps ports, integrated PoE
Graphical Display
132 x 48 backlit graphical LCD display
Feature Keys
2 line keys with dual-color LED and 2 SIP accounts. 3 XML programmable
context sensitive soft keys. 5 (navigation, menu) keys. 8 BLF keys. 13
dedicated function keys for PAGE/INTERCOM, PHONEBOOK, MESSAGE,
HOME, HOLD, RECORD, MUTE, HEADSET, TRANSFER, CONFERENCE,
SEND and REDIAL, SPEAKERPHONE, VOLUME
Voice Codecs
Support for G.711µ/a, G.722 (wide-band), G.723.1, G.726-32, G.729 A/B,
iLBC, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO)
Telephony Features
Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup, 3way conference, shared-call-appearance (SCA) / bridged-line-appearance
(BLA), downloadable phone book (XML, LDAP, up to 1000 items), call waiting,
call history (up to 200 records), off-hook auto dial, auto answer, click-to-dial,
flexible dial plan, Hot Desking, personalized music ringtones, server
redundancy & fail-over.
Headset Jack
RJ9 headset jack (allowing EHS with Plantronics headsets)
HD Audio
Yes, HD handset and speakerphone with support for wideband audio
Base Stand
Yes, 2 angle positions available, Wall Mountable
QoS
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level access control, MD5 and MD5-sess based
authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS,
802.1x media access control
Multi-language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
simplified and traditional Chinese, Korean, Japanese, and more
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTP/ FTP/FTPS, mass provisioning
using TR-069 or AES encrypted XML configuration file
Power & Green Energy
Efficiency
Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA
PoE IEEE802.3af Class 2, 3.84W-6.49W
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Temperature and
Humidity
Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity: 10% to 90% Non-condensing
Physical
Dimension: 220.5mm (L) x 192.5mm (W) x 76.0mm (H) (with handset)
Unit weight: 0.8kg; Package weight: 1.2kg
Package Content
GXP1628, handset with cord, base stand, universal power supply, network
cable, Quick Installation Guide, brochure, GPL license, BLF label cards
Compliance
FCC: Part 15 (CFR 47) Class B
CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1
Table 4: GXP1630 Technical Specifications
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS,
FTP/FTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE,
SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS,
SRTP
Network Interfaces
Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports, integrated
PoE
Graphical Display
132 x 64 backlit graphical LCD display
Feature Keys
3 line keys with dual-color LED and 3 SIP accounts. 3 XML programmable
context sensitive soft keys. 5 (navigation, menu) keys. 8 BLF keys. 13
dedicated function keys for PAGE/INTERCOM, PHONEBOOK, MESSAGE,
HOME, HOLD, RECORD, MUTE, HEADSET, TRANSFER, CONFERENCE,
SEND and REDIAL, SPEAKERPHONE, VOLUME
Voice Codecs
Support for G.711µ/a, G.722 (wide-band), G.723.1, G.726-32, G.729 A/B,
iLBC, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO)
Telephony Features
Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup, 4ways conference, shared-call-appearance (SCA) / bridged-line-appearance
(BLA), downloadable phone book (XML, LDAP, up to 1000 items), call waiting,
call history (up to 200 records), off-hook auto dial, auto answer, click-to-dial,
flexible dial plan, Hot Desking, personalized music ringtones, server
redundancy & fail-over.
Headset Jack
RJ9 headset jack (allowing EHS with Plantronics headsets)
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HD Audio
Yes, HD handset and speakerphone with support for wideband audio
Base Stand
Yes, 2 angle positions available, Wall Mountable
QoS
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level access control, MD5 and MD5-sess based
authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS,
802.1x media access control
Multi-language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
simplified and traditional Chinese, Korean, Japanese, and more
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS/ FTP/FTPS, mass provisioning
using TR-069 or AES encrypted XML configuration file
Power & Green Energy
Efficiency
Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA
PoE IEEE802.3af Class 2, 3.84W-6.49W
Temperature and
Humidity
Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity: 10% to 90% Non-condensing
Physical
Dimension: 220.5mm (L) x 192.5mm (W) x 76.0mm (H) (with handset)
Unit weight: 0.8kg; Package weight: 1.2kg
Package Content
GXP1630 phone, handset with cord, base stand, universal power supply,
network cable, Quick Installation Guide, brochure, GPL license, BLF label
cards
Compliance
FCC : Part 15 (CFR 47) Class B
CE : EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1
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CONFIGURATION GUIDE
The GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 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 UP/DOWN/LEFT/RIGHT arrow keys to navigate in the
menu options.
• Enter/Confirm selection. Press the round MENU button to enter the selected option.
• Exit. Press LEFT arrow key to exit to the previous menu.
• The phone automatically exits MENU mode with an incoming call, when the phone is off hook or
the MENU mode if left idle for more than 60 seconds.
• When the phone is in idle, pressing the navigation keys UP/DOWN/RIGHT can access the call
history entries:
UP – Missed Calls
DOWN – Dialed Calls
RIGHT – Answered Calls
The MENU options are listed in the following table.
Table 5: GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 Menu Options
Call History
Call History sub menu includes the following options:
• Local Call Log:
Displays answered calls, dialed calls, missed calls, transferred calls, forwarded calls.
• Broadsoft Call Log:
Displays Broadsoft call log.
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Status
Status sub menu includes the following options:
• Network Status.
Press to enter the sub menu for IP setting information (DHCP/Static IP/PPPoE), IPv4
address, IPv6 address, Subnet Mask, Gateway，DNS server and NTP server.
• Account Status.
Shows account registration status.
• System status
• Hardware menu: hardware version and P/N number.
• Software Version menu: boot, core, base, prog, locale version, recovery version
and MAC address.
• IP Geographic Information menu: city, language, and time zone.
• Special feature menu: Display the special features enabled.
Contacts
Contacts sub menu includes the following options:
• Local Phonebook
Displays phonebook and users could add, edit, search, and delete contacts, or
download phonebook XML to the phone. When doing phonebook search, user can
only search ASCII characters.
• Local Group
Displays phonebook group.
Note: Besides 3 embedded groups: Family, Friends and Work, user can create, edit,
and delete your own new groups. GXP phone allows at most 7 customized groups.
• Broadsoft Phonebook
• LDAP Directory
Configures LDAP directory options, displays LDAP directory by searching. LDAP
search does not support entering Non-ASCII characters.
Messages
• Instant Messages
Displays received instant messages.
• Voice Mails
Displays voice mail messages.
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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 LCD contrast by pressing left/right arrow key.
• LCD Brightness (Not applicable to GXP1610/GXP1615)
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 could select
Automatic for local language based on IP location if available.
• Date Time
Configures time zone and NTP server on the phone.
• Security
Configures the access control for the users to configure from keypad Menu, select
the web access mode, and enable / disable the SSH access.
• Headset type
Select Normal or Plantronics EHS headset type.
• Star Key Lock
Turns on/off keypad lock feature and configures keypad lock password.
Direct IP Call
Makes direct IP call.
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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
• Selects preferred internet protocol (IP v4 or IP v6);
• Configures IP mode (DHCP/Static IP/PPPoE);
• Configures DHCP option 12 (Host Name) and option 60 (Vendor Class ID);
• Configures PPPoE account ID and password;
• Configures IP address, Netmask, Gateway, DNS Server 1, and DNS Server 2;
• Configures 802.1X mode;
• Configures Layer 2 QoS;
• Configures PC Port Settings;
• Configures OpenVPN Settings;
• Configures IPv6 Settings
• Upgrade
Configures firmware server and config server for upgrading and provisioning the
phone.
• UCM Detect
Displays connected UCM server List and specifies Protocol, IP, and Port of the target
UCM server.
• Factory Functions
Factory Functions sub menu includes the following options:
• Audio Loopback
Speak to the phone using speaker/handset/headset. If you can hear your voice, your
audio is working fine. Press Menu button to exit audio loopback mode.
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• Diagnostic Mode
All LEDs will light up. Press any key (except MENU key) on the keypad to display
the button name in the LCD. Lift and put back the handset or press Menu button to
exit diagnostic mode.
• Keyboard Diagnostic
Press all the available keys on the phone. The LCD will display the name for the keys
to be pressed to finish the keyboard diagnostic mode.
• Certification Verification
Verify the certificate loaded on the phone.
• Factory Reset
Resets the phone to factory default settings.
Reboot
Reboots the phone
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The following picture shows the keypad MENU configuration flow.
Figure 1: Keypad Menu Flow
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Configuration via Web Browser
The GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 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 Microsoft’s IE, Mozilla Firefox, and Google Chrome.
To access the GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 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 on LCD.
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 should 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:
Table 6: User / Admin Levels Information
User Level
User
Password
Web Pages Allowed
End User Level
user
123
Only Status page, Basic Settings in
Advanced Settings page, and some
settings in maintenance page.
Administrator Level
admin
admin
Browse all pages
• When accessing the GXP16xx for the first time or after factory reset, users will be asked to
change the default administrator password before accessing GXP16xx 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.
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Figure 2: Change Password on first boot
• When changing any settings, always SUBMIT them by pressing the SAVE button on the bottom of the
page. 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 Basic Setting and Account Setting, and most of the options
under Advanced Settings do not require reboot after submitting the changes. Under Advanced Setting,
the parameters on network configuration require reboot after update.
Definitions
This section describes the options in theGXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630
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.
• Advanced Settings: To configure Call features, Time settings, Web services, Programmable keys
and etc.
• Maintenance: To configure Network settings, Web/SSH Access, Upgrading and provisioning,
Language, Contacts and etc.
Status Page Definitions
Table 7: Status Page Definitions
Status → Account Status
Account x
Account index.
SIP User ID
Displays the configured SIP User ID.
SIP Server
Displays the configured SIP Server address.
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SIP Registration
Displays SIP registration status YES/NO.
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
DHCP, Static IP or PPPoE.
IPv4 Address
The IPv4 address obtained on the phone.
IPv6 Address
The IPv6 address obtained on the phone.
OpenVPN® IP
The IP address of the phone obtained from the OpenVPN server
Subnet Mask
The subnet mask obtained on the phone.
Gateway
The gateway address obtained on the phone.
DNS Server 1
The DNS server address 1.
DNS Server 2
The DNS server address 2.
PPPoE Link Up
PPPoE connection status.
NAT Type
NAT Type
NAT Traversal
Display the NAT traversal status of Account 1 and Account 2.
Status → System Info
Product Model
Product model of the phone.
Part Number
Product part number.
Software Version
• Boot Version: boot version number.
• Core Version: core version number.
• Base Version: base version number.
• Program Version. This is the main firmware release number, which is
always used for identifying the software system of the phone.
• Locale Version. This is the version of localization language package.
• Recovery Version.
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IP Geographic
Information
• City: Displays the city name.
• Language: Displays current language been used.
• Time Zone: Displays current time zone.
Special Feature
• OpenVPN Support: Display the status of the OpenVPN feature.
System Time
• System up time: Displays system up time since the last reboot.
• System time: Displays current system time on the phone system.
Service Status
GUI and Phone service status.
Core Dump
Core dump file that could be downloaded for troubleshooting purpose.
Status → GNU GPL License
GNU GPL License
Shows the GNU GPL License information.
Account Page Definitions
The GXP1610/GXP1615 support two SIP accounts, GXP1620/GXP1625/GXP1628 support two SIP
accounts, and GXP1630 supports three SIP accounts that can be configured to accommodate independent
SIP accounts. Every SIP account has an individual configuration page.
Table 8: Account Page Definitions
Account x → General Settings
Account Active
This field indicates whether the account is active. The default setting is “Yes”.
Account Name
The name associated with each account to be displayed on the LCD.
SIP Server
The URL or IP address, and port of the SIP server. This is provided by your
VoIP service provider (ITSP).
Secondary SIP Server
The URL or IP address, and port of the SIP server. This will be used when the
primary SIP server fails.
Outbound Proxy
IP address or 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
Secondary outbound proxy which will be used when the primary proxy cannot
be connected.
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BLF Server
The URL or IP address, and port of the BLF Server. When “BLF Server” is
configured, the SUBSCRIBE request will be sent to this address for BLF.
SIP User ID
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.
Users can also register an account with a SIP user ID that carries "@". For
example, you can put "111@test.com" in your SIP User ID field. When
registering, the phone will register your account as "111@test.com" instead of
111.
Note: The server domain will not be included in the SIP from header.
Authentication ID
SIP service subscriber’s Authenticate ID used for authentication. It can be
identical to or different from the SIP User ID. This field supports up to 8192
characters.
Authentication
Password
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
The SIP server subscriber’s name (optional) that will be used for Caller ID
display.
Voice Mail Access
Number
This parameter allows you to access voice messages by pressing the
MESSAGE button on the phone.
This ID is usually the VM portal access number. For example, in Asterisk server,
8500 could be used.
Account Display
This option allows you to configure how your SIP account label will be displayed
on the phone’s screen. If set to “Username”, LCD account label will display the
Account Name configured for this SIP account. If set to “User ID”, it will then
display the SIP User ID configured for this SIP account.
Account x → Network Settings
DNS Mode
This parameter controls how the Search Appliance looks up IP addresses for
hostnames. There are four modes: A Record, SRV, NATPTR/SRV, and 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:
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• Primary IP: Primary IP address where the phone sends DNS query to;
• 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
This parameter configures whether the NAT traversal mechanism is activated.
Users could select the mechanism from No, STUN, Keep-Alive, UPnP, Auto or
VPN. If set to “STUN” and STUN server is configured, the phone will route
according to the STUN server. If NAT type is Full Cone, Restricted Cone or
Port-Restricted Cone, the phone will try to use public IP addresses and port
number in all the SIP&SDP messages. The phone will send empty SDP packet
to the SIP server periodically to keep the NAT port open if it is configured to be
“Keep-Alive”. Configure this to be “No” if an outbound proxy is used. “STUN”
cannot be used if the detected NAT is symmetric NAT.
Proxy-Require
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.
Use SBC
Indicate whether an SBC server is used.
Account x → SIP Settings → Basic Settings
TEL URI
If the phone has an assigned PSTN telephone number, this field should be set
to “User=Phone”. Then a “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 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
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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 frequency (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 default is 60, and maximum value
is 64800 (about 45 days).
Reregister Before
Expiration
Specifies the time frequency (in seconds) that the phone sends re-registration
request before the Register Expiration. The default value is 0.
Enable OPTIONS Keep
Alive
Select whether the phone will keep sending a message to check the connection
with the server.
OPTIONS Keep Alive
Interval
Specifies the frequency (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, and 5064 for Account 3.
SIP Registration
Failure Retry Wait Time
Specifies the interval to retry registration if the process is failed. 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.
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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.
SIP URI Scheme when
using TLS
Specifies if “sip:” or “sips:” will be used when TLS/TCP is selected for SIP
Transport. The default setting is “sips:”.
Use Actual Ephemeral
Port in Contact with
TCP/TLS
This option is used to control the port information in the Via header and Contact
header. If set to No, these port numbers will use the permanent listening port
on the phone. Otherwise, they will use the ephemeral port for the connection.
Outbound Proxy Mode
Configures the outbound proxy mode, to place in route header in sending SIP
messages, or always sent to outbound proxy. Default setting is “in route”
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.
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.
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Caller ID Display
The phone supports Caller ID display as follows:
• When set to “Auto” (Default), 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.
Note: During call, if the phone receives NOTIFY message with type
header:"application/cid-info+xml" and event header:"cid-info-update", the
phone will take the message and update CID information and CID image.
Ignore Alert-Info
Header
This option is used to configure default ringtone. If set to “Yes”, configured
default ringtone will be played even if Alert-info header is present.
The default setting is “No”.
Add Auth Header On
Initial REGISTER
When enabled, Device will send Authentication Header on the first Register
instead of second Register.
Default is No.
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. When set to "Default", the
Privacy Header will show in INVITE only when "Huawei IMS" special feature is
on. 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-PreferredIdentity Header
Controls whether the P-Preferred-Identity Header will present in the SIP INVITE
message or not. The default setting is “default”: The P-Preferred-Identity
Header will show in INVITE unless “Huawei IMS” special feature is on. If set to
“Yes”, the P-Preferred-Identity Header will always show in INVITE. If set to
“No”, the P-Preferred-Identity Header will not show in INVITE.
Use P-Access
Network-Info Header
Enables/disables the use of P-Access-Network-Info header in SIP request.
When disabled, the SIP message sent from the phone will not include the
selected header. Default setting is “No”.
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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
If Yes except REGISTER, the sip message for register or unregister will
contains MAC address in the header, and all the outgoing SIP messages
except REGISTER message will attach the MAC address to the User-Agent
header; If Yes to ALL, the sip message for register or unregister will contains
MAC address in the header, and all the outgoing SIP message including
REGISTER will attach the MAC address to the User-Agent header; If No,
neither will the MAC header be included in the register or unregister message
nor the MAC address be attached to the User-Agent header for any outgoing
SIP message.
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 Web GUI
→Settings→Programmable Keys page, please select "Eventlist BLF" in the
key mode, choose account, enter the value of each extension in the list.
(Supported only on GXP1628/GXP1630)
Auto Provision
Eventlist BLFs
When enabled, empty multi-purpose keys will be automatically provisioned to
the monitored extensions in the Eventlist BLF. Default setting is “Disabled”.
(Supported only on GXP1628/GXP1630)
Conference URI
Configures the conference URI when using Broadsoft N-way calling feature.
Music On Hold URI
Music On Hold URI to call when a call is on hold if server supports it.
BLF Call-pickup Prefix
Configures the prefix prepended to the BLF extension when the phone picks
up a call with BLF key. (Supported only on GXP1628/GXP1630)
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Call Pickup Barge-in
Code
Set feature code of call pickup barge-in feature.
(Supported only on GXP1628/GXP1630)
PUBLISH for Presence
Enables Presence feature on the phone.
(Supported only on GXP1628/GXP1630)
Enable User Presence
Subscription
If the user presence subscription is enabled, the phone will subscribe and notify
about user presence to the sip server.
Note: To enable this feature, register Genesys account on your phone’s SIP
account first. The SIP server should support this feature.
Hide Login Soft Key
On Auto User
Presence Subscription
This option allows you to hide the soft key to login or logout on idle screen if
auto user ID and password are configured.
Auto User Presence ID
With this ID, the phone will subscribe user presence automatically once the
account is registered.
Auto User Presence
password
With this password, the phone will subscribe user presence automatically once
the account is registered.
Special Feature
Different soft switch vendors have special requirements. Therefore, users may
need select special features to meet these requirements. Users can choose
from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, Huawei IMS,
PhonePower and iPageOn depending on the server type. Default is “Standard”.
Broadsoft
Broadsoft Call Center
Default setting is “Disabled”. When set to “Enabled”, a soft key “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
With “Hoteling Event” enabled, user can access the Hoteling feature option by
pressing the “BSCCenter” softkey. Default setting is “No”.
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. Default is “Disabled”.
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Broadsoft Call Park
When enabled, phone will send the SUBSCRIBE to Broadsoft server to obtain
Broadsoft Call Park notification.
Notes: The phone shows parked call notification on its LCD.
If parked member is a monitored extension on BLF MPK key, the MPK will be
flashing red.
Account x → SIP Settings → Session Timer
Enable Session Timer
This option is used to enable or disable session timer on the phone side when
server side can provide both session timer UPDATE or session audit UPDATE.
The default setting is “Yes”.
Session Expiration
The SIP Session Timer extension (in seconds) that enables SIP sessions to be
periodically "refreshed" via a SIP request (UPDATE, or re-INVITE).
If there is no refresh via an UPDATE or re-INVITE message, the session will
be terminated once the session interval expires.
Session Expiration is the time (in seconds) where the session is considered
timed out, provided no successful session refresh transaction occurs
beforehand. The default setting is 180. The valid range is from 90 to 64800.
Min-SE
The minimum session expiration (in seconds). The default value is 90 seconds.
The valid range is from 90 to 64800.
Caller Request Timer
If set to “Yes” and the remote party supports session timers, the phone will use
a session timer when it makes outbound calls.
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.
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”.
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.
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.
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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 → Security
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 Certification
Chain
Choose whether certification chain will be validated when TLS/TCP mode is
used.
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
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 use 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. Default setting is
"RFC2833"
DTMF Payload Type
Configures the payload type for DTMF using RFC2833. Default value is 101.
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Preferred Vocoder
7 different vocoder types are supported on the phone, including G.711 U-law
(PCMU), G.711 A-law (PCMA), G.723.1, G.729A/B, G.722 (wide band), iLBC,
and G726-32. Users can configure vocoders in a preference list that is included
with the same preference order in SDP message.
Use First Matching
Vocoder in 200OK SDP
When set to “Yes”, the device will use the first matching vocoder in the received
200OK SDP as the codec.
The default setting is “No”.
Codec Negotiation
Priority
This option allows user to choose which Codec sequence to use.
If set to “Caller”, the phone will negotiate the codecs based on the SDP from
the received SIP INVITE. If set to “Callee”, the phone will negotiate the audio
codecs based on the codec priority on the phone.
Disable Multiple m line
in SDP
If enabled, the phone always responses 1 m line in SDP regardless multiple m
lines are offered.
SRTP Mode
Enables the SRTP mode based on your selection.
Default setting is “Disabled”.
Symmetric RTP
Defines whether symmetric RTP is supported or not.
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”
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
This option determines the encoding rate for G723 codec. Users can choose
from 6.3kbps encoding rate and 5.3kbps encoding rate. The default setting is
“5.3kbps encoding rate”.
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G.726-32 Packing
Mode
Selects “ITU” or “IETF” for G.726-32 packing mode.
The default setting is “IETF”.
iLBC Frame Size
This option determines the iLBC packet frame size. Users can choose from
20ms and 30ms. The default setting is “30ms”.
iLBC payload Type
This option is used to specify iLBC payload type. Valid type is 96-127.
Jitter Buffer Type
Selects either Fixed or Adaptive based on network conditions. The default
setting is “Adaptive”.
Jitter Buffer Length
Defines jitter buffer length based on network conditions.
Hide Vocoder
Hides vocoder information on call screen if set to “yes”. Default settings is “No”.
Account x → Call Settings
Early Dial
Selects whether to enable early dial. If it is set to “Yes”, the SIP proxy must
support 484 responses. The default setting is “No”.
Dial Plan Prefix
Sets the prefix 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.
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;
3. Grammar: X – any character from 0-9, a-z, A-Z.
a) xx+ - at least 2-digit numbers
b) xx – only 2-digit numbers
c) XX – two characters ( AA, Ab, 1C, f5, 68,…)
d) \t\e\s\t : only string “test” will pass the dial plan check
e) ^ - exclude
f) [3-5] – any digit of 3, 4, or 5
g) [147] – any digit of 1, 4, or 7
h) <2=011> - replace digit 2 with 011 when dialing
i) | - the OR operand
• Example 1: {[369]11 | 1617xxxxxxx}
Allow 311, 611, and 911 or any 11-digit numbers with leading digits 1617;
• Example 2: {^1900x+ | <=1617>xxxxxxx}
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Block any number of leading digits 1900 or add prefix 1617 for any dialed 7digit 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 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
dialing7 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.
Max length of dial plan is up to 1024 characters.
Bypass Dial Plan
Bypass the dial plan when dialing from one of the available items:
Contacts ; Call History Incoming Call ; Call History Outgoing Call ; Dialing
Page ; MPK ; API
The default settings bypass the dial plan for the MPK only.
Call Log
Configures Call Log setting on the phone. You can log all calls, only log
incoming/outgoing calls 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, blocking the Caller ID to be displayed.
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.
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Auto Answer Numbers
The function allows users to have the phone configured with a pre-defined list
of numbers that will perform auto answer.
For “Auto Answer Numbers”, it accepts:
Digits :1,2,3,4,5,6,7,8,9; x – any digit from 0-9; xx – any two digits from 0-9;
[1-5] – any digit from 1 to 5; +: it matches the previous character as many times
as needed like regular expression.
Auto Answer Numbers can be split with ";", for example: 1x;2xxx;3x+
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
Defines whether the call is transferred to the other party if the initiator of the
conference hangs up.
The default setting is “No”.
Disable Recovery on
Blind Transfer
Defines whether to enable/disable recovery to the call to the transferee on
failing blind transfer to the target.
Blind Transfer Wait
Timeout
Defines the timeout (in seconds) for waiting SIP frag response in blind transfer.
Valid range is 30 to 300.
No Key Entry Timeout
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.
Key As Send
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 equivalent to the “Send” key. If set to “No”, the “#” key
is included as part of the dialing string.
On Hold Reminder
Tone
If enabled, the user will hear 3 beeps as hold reminder, the first on hold
reminder occurs at 1 min and the subsequent on hold reminder occurs every
30 sec. Default is “Disabled”.
Call Recording On
Configures the DTMF sequence sent when pressing the Record key during a
call on this account.
Call Recording Off
Configures the DTMF sequence sent when pressing the Record key during a
call on this account when turning recording off.
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Hide Dialing Password
Allows user to hide the password when the dialing number matches the
configured prefix.
Remote Info Display
Choose which caller information to display once receiving a call:
• Name and Number: Display the caller ID name and number.
• Number: Display only the caller number.
Name: Display only the caller ID name.
Hide Remote User ID
Settings to allow user to hide and display name of remote user ID that has a
length more than specific in the incoming and/or outgoing calls
Ring Tone
Account Ring Tone
Configures ring tone for the account.
Match Incoming Caller
ID
Specifies matching rules with number, pattern or Alert Info text. (Up to 10 rules).
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.
Default is 60.
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Account x → Intercom Settings
Allow Auto Answer by
Call-Info/Alert-Info
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep, based on the SIP CallInfo/Alert-Info header sent from the server/proxy.
Mute on answer
Intercom call
When enabled, phone will mute the incoming intercom call by Call-Info/AlertInfo
Play warning tone for
Auto Answer Intercom
When enabled, phone will play warning tone when auto answer Intercom
Custom Alert-Info for
Auto Answer
Used exclusively to match the contents of the Alert-Info header for auto
answer.
The default auto answer headers will not be matched if this is defined.
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.
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.
Call Forward
Unconditionally(All)—
On
Configures Call Forward All feature code to activate unconditional call
forwarding.
Call Forward
Unconditionally(All)—
Off
Configures Call Forward All feature code to deactivate unconditional call
Forwarding.
Target
Configures the extension that the call will be forwarded to.
Call Forward Busy—
ON
Configures Call Forward Busy feature code to activate busy call forwarding.
Call Forward Busy—
Off
Configures Call Forward Busy feature code to deactivate busy call forwarding.
Target
Configures the extension that the call will be forwarded to.
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Call Forward Delayed
(No Answer)—On
Configures Call Forward Delayed feature code to activate no answer call
forwarding.
Call Forward Delayed
(No Answer)—Off
Configures Call Forward Delayed feature code to deactivate no answer call
forwarding.
Target
Configures the extension that the call will be forwarded to.
Delayed Call Forward
Wait Time
Defines the timeout (in seconds) before the call is forwarded on no answer.
Valid range is 1 to 120. Default is 20
Settings Page Definitions
Table 9: Settings Page Definitions
Settings → General Settings
Local RTP Port
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.
Local RTP Port Range
This parameter defines the range of local RTP port from 48 to 10000.
Use Random Port
When set to “Yes”, this parameter will force random generation of both the local
SIP and RTP ports. This is usually necessary when multiple phones are behind
the same full cone NAT. The default setting is “Yes”
Note: This parameter must be set to “No” for Direct IP Calling to work
Keep-alive Interval
Specifies how often the phone sends a blank UDP packet to the SIP server to
keep the “ping hole” on the NAT router to open. The default setting is 20
seconds.
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 public mode for hot Desking feature on the phone.
If set to “Yes”, users would need fill in the SIP Server address for account 1 as
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well. Then reboot the phone. When the phone boots up, users will require
entering SIP User ID and Password on the LCD to login and use the phone.
Note: When the phone is in public mode login screen, press CONF button will
have the IP address of the phone displayed.
Enable Fix For RTP
Timestamp Jump
Make RTP timestamps be continuous if there is audio loss caused by
timestamp jump
Test Password
Strength
Only Allow password with some constraints to ensure better security.
DNS Cache Refresh
Time
The phone will send out DNS queries based on the configured DNS Cache
Refresh Time after responding response to refresh IP addresses for SIP. Valid
Range is [5 -1440] minutes. Default is 30.
Settings → Call Features
Onhook Dial Barging
When the option is set to "Disabled", onhook dialing will not be interrupted by
an incoming call.
Off-hook Auto Dial
Configures a User ID/extension to dial automatically when the phone is
Offhook. The phone will use the first account to dial out.
The default setting is “No”.
Off-hook Auto Dial
Delay
Defines the timeout (in seconds) for off-hook auto dial. Valid range is 0-30. If
set to 0, it will be dialed out immediately; If set to other values, it will be dialed
out after the delay.
Off-hook Timeout
If configured, when the phone is Onhook, it will go Offhook after the timeout (in
seconds). The default value is 30 seconds.
Enable Live DialPad
If the Live DialPad is enabled, after dialing the numbers, the phone will wait for
the time set in the expire time (in seconds) and dial out automatically after using
handsfree mode.
Live DialPad Expire
Time
Sets the Live DialPad expire time, interval is between 2s and 15s.
The default value is 5s.
Busy Tone Expiration
Busy Tone Expiration (in seconds). The default value is 30 seconds.
Enable Automatic
Redial
When the phone dials a number and no answer, if automatic redial is enabled,
the phone will automatically redial this number.
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Automatic Redial
Times
The total times that need to be redial if Automatic Redial is enabled (in
seconds). The default value is 10 seconds.
Automatic Redial
Interval
The interval between each time redial if Automatic Redial is enabled (in
seconds). The default value is 20 seconds.
Intercom Key Mode
Controls the behavior of the intercom hard key. Default setting is “Intercom”.
Intercom User ID
Configures the intercom extension number for account 1 to dial out. This User
ID is mapped to the INTERCOM button on the phone.
Intercom Multicast
Paging Address
Defines the multicast address that will be broadcasted to when the key is set to
multicast paging.
Intercom Multicast
Paging Label
Determines the optional name to be displayed during the page.
Bypass Dial Plan
through Call History
and Directories
Enable/disable dial plan check while dialing through the call history and any
phonebook directories.
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
This option allows you to configure whether the speaker phone will ring when
there is a new incoming call during an active call. If set to No, the user will hear
call waiting tone. If set to Yes, the phone will ring using the speakerphone.
Disable Busy Tone on
Remote Disconnect
Disable the busy tone heard in the handset when the call is disconnected
remotely. The default setting is No.
Disable Direct IP Call
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,
Offhook 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”.
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Disable in-call DTMF
Display
When it is set to “Yes”, the DTMF digits entered during the call will not display.
The default setting is “No”.
Enable Sending DTMF
via specific MPKs
Allows certain MPKs to send DTMF in-call. This option does not affect Dial
DTMF.
Mute Key Functions
While Idle
When set to “DND”, the DND will be enabled for future incoming call if pressing
MUTE key in idle state; If this feature is set to “Idle Mute”, MUTE key will take
effect in idle state and future incoming call will be answered with mute;
Otherwise, MUTE key will not take effect in idle state.
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 the call screen. If 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”.
Use Pound (#) For
Redial
If set to “Yes”, the “#” key will immediately redial out the previous number.
Default is Yes.
Click-To-Dial Feature
Enables Click-To -Dial feature. 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.
The default value is 300 seconds.
Max Unsaved Log
Defines the number of unsaved logs before written to phone’s flash.
The default value is 200 entries.
User-Agent Prefix
Add a prefix in the User Agent field on the SIP header.
Predictive Dialing
Feature
Allows the phone to use your dialed number and search the compatible
numbers from history and contacts and will “Predict” the number you are going
to dial when this feature is enabled.
This saves time for the user, so they will not need to dial the entire number
every time they wanted to call the number. Default settings is Enabled.
Disable Hook Switch
User can Enable/Disable the Hook Switch as follows:
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When set to "Yes", disable hook switch completely; When set to "For Answer
Call", hook switch cannot be used for answering call. Default is No.
Show SIP Error
Response
Allows users to enable or disable SIP error response to be displayed on LCD.
Default setting is Yes.
Auto Answer Delay
Configures the delay for the incoming call automatic answer.
This option takes effect only if [Auto Answer] is enabled on the phone.
Valid range is 0 to 10 (in seconds). Default is 0.
Enable Enhanced
Acoustic Echo
Canceller
Allows user to choose to enable or disable the echo canceller on the phone in
speaker mode. Default is Yes.
Show on Hold Duration
Show the duration of holding a call on the LCD.
Default is No.
Hide/Disable reject key
When “Hide/Disable REJECT Key” is set to “Yes”, the device will not display
REJECT softkey upon incoming call.
Default is No.
Enable Diversion
Information Display
When set to “Yes” (Default), Diversion info will be displayed if the device
receives INVITE with Diversion header.
Settings → Multicast paging
Paging Barge
During active call if incoming multicast page’s priority is higher than this value,
the call will be held, and multicast page will be played.
Paging Priority Active
If enabled, during a multicast page if another multicast is received with higher
priority, that one will be played instead.
Multicast Paging
Codec
The codec for sending multicast pages, there are 7 codecs could be used:
PCMU, PCMA, G.723.1, G.726-32, G.729A/B, G.722(wide band), and iLBC.
The default setting is “PCMU”.
Multicast Channel
Number
Multicast Channel Number (0-50). 0 for normal RTP packets, 1-50 for Polycom
multicast format packets.
Multicast Sender ID
Outgoing caller ID that displays to your page group recipients (for multicast
channel 1 - 50).
Multicast Listening
Defines multicast listening addresses and labels. GXP16xx phone can listen to
at most 10 multicasts addresses with different priorities.
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Settings → Preferences → 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, 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".
Enable Group Listening
Allow handset to be able to listen when picked up during a call with headset
Group Listening with
Speaker
If enabled, during a call, the phone will display softkey to enable speaker
listening along with handset or headset.
Headset TX gain
Configures the transmission gain of the headset. The default value is 0dB.
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Headset RX gain
Configures the receiving gain of the headset. The default value is 0dB.
Handset TX gain
Configures the transmission gain of the handset. The default value is 0dB.
Settings → Preferences → 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.
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 date/time used on the phone according to the specified time zone.
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 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,
3
rd
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:
• 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
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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 → Preferences → LCD Display
Backlight Brightness:
Active
Configures the LCD backlight brightness level (from 0 to 8) for phone's active
status. The default value is 6.
Note: This option is not applicable to GXP1610/GXP1615.
Backlight Brightness:
Idle
Configures the LCD backlight brightness level (from 0 to 8) for phone's idle
status. The default value is 2.
Note: This option is not applicable to GXP1610/GXP1615.
Active Backlight
Timeout
Configure the minute of active backlight timeout. The valid range is 0 to 90.
When Active Backlight Timeout is set to 0, the backlight will be constantly on.
LCD Contrast
Configures the LCD contrast level (from 1 to 5).
The default value is 3.
Disable Missed Call
Backlight
When it is set to "Yes", the LCD backlight will not be turned on when there is a
new missed call. The default setting is "No".
Note: This option is not applicable to GXP1610/GXP1615.
Wide Idle Screen View
Allows to remove the date and moves the notification icons to the right for a
longer account name display when enabled. The default setting is "No".
Note: This feature needs a reboot to take effect.
Caller ID Display Mode
Users can configure “Caller ID Display Mode” to control the caller ID display
preference:
• Dynamic (Default): If the caller ID is long, it will keep scrolling until it is
displayed completed.
• Static: The caller ID will not scroll.
Settings → Preferences → LED Control
Enable / Disable
VM/MSG Power Light
Flash
This option controls the LED behavior upon new/unread Voicemail and
Message.
If set to “No”, the LED light will be flashing when there is unread Voicemail and
Message. Otherwise, it will be off.
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Settings → Preferences → Ring Tone
Call Progresses 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
Configures the call waiting tone gain to adjust call waiting tone volume. The
default is "Low".
Speaker Ring Volume
Configures speaker ring volume. The valid range is 0 to 7.
Notification tone
volume
This option allows you to adjust the volume for paging, intercom, auto answer
call and call hold reminder beep. The valid range is 0 to 7.
Lock Speaker Volume
Locks the volume of the phone’s ring tone. When set to “Yes”, the volume
buttons on the phone will be disabled during ringing.
Default settings is “No”.
Default Ringtone
Allows to set the Default Ringtone to be used globally. Up to 10 custom
ringtones are supported.
Note: The ring tone set in individual accounts has a higher priority than this
setting. If the user wants the default ring tone to be used globally, he needs to
set the ring tone of each account to Default Ring Tone; Otherwise, the one
specified for each account will be used instead.
Total Number of
Custom Ringtone
Update
Configures the total number of custom ringtone update that can be
downloaded during provisioning process. Valid range 0-10.
Default value is 3.
Settings → Web Service
Use Auto Location
Service
To enable or disable auto location services on the phone (Reboot required).
Default settings is Yes.
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Settings → XML Applications
Idle Screen
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.
XML Application
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.
Softkey Label
Specifies the Softkey name displayed on the idle screen for the users to
enter XML application.
The default Softkey Label is “XML Service”.
Settings → Programmable Keys → Multi-Purpose Keys Settings
Transfer Mode via
MPK
Perform blind transfer, attended transfer, or a new call with the specific in the
Value field when a user presses "Transfer" multiple-purpose key.
Enable Transfer via
non-Transfer MPK
MPK with type BLF, Speed dial, etc, will perform similar to “transfer MPK”
under active call.
Settings → Programmable Keys → Programmable Keys
Line Key X
Assigns a function to the corresponding line key. 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.
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• 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 off hook and account 2 is active, it will call the
configured Speed Dial number using account 2.
• Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension number
in the Value field to be monitored. (Supported only on GXP1628/GXP1630)
• 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. (Supported only on
GXP1628/GXP1630)
• 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. (Supported only on
GXP1628/GXP1630)
• 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.
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• Call Park
Select Account and enter the call park extension in the Value field to
park/pick up the call. (Supported only on GXP1628/GXP1630)
• 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 Name 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 Name field is left blank. Enter the LDAP
Name/Number filter in the Value field.
LDAP search does not support entering Non-ASCII characters
• Multicast Paging
This option is for multicast sending.
Enter Line key description in Description field and multicast sending
address in Value field.
• GDS OpenDoor
This option is for opening the door via GDS when the phone is in idle stage
and MPK configured is pressed.
Notes : The device needs to have GDS open door setting configured under
device web UI→ Settings →External Service
- the GDS device must be properly configured such as enabling “Remote PIN
To Open Door” under GDS web UI → Door System Settings → Basic
Settings.
• Provision
Select this feature to make the phone trigger an instant provisioning.
Softkeys
Assigns a function to the corresponding Softkey. 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.
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• 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 off hook 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 Name 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 Name field is left blank. Enter the LDAP
Name/Number filter in the Value field.
LDAP search does not support entering Non-ASCII characters
Multi-Purpose Keys
(GXP1628/GXP1630
only)
Assigns a function to the corresponding Multi-Purpose Key.
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.
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• 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 off hook and account 2 is active, it will call the
configured Speed Dial number using account 2.
• Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension number
in the Value field to be monitored.
• 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.
• 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.
• 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.
• Intercom
Select Account and enter extension number in Value field to intercom.
• LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP search
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base in the Name 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 Name field is left blank. Enter the LDAP
Name/Number filter in the Value field. LDAP search does not support
entering Non-ASCII characters
• Multicast Paging
This option is for multicast sending. Enter Line key description in
Description field and multicast sending address in Value field.
• Provision
Select this feature in order to make the phone trigger an instant
provisioning.
Settings → Programmable Keys → Softkeys Settings
Softkey Layout
Custom Softkey Layout
Enables/ Disables the custom Softkey layout. Default is Disabled
Idle Screen Softkey
Layout
Hide System Softkey on Idle Page
Users can choose a system generated softkey to hide on idle page. Forward
is checked by default.
Custom Call Screen
Softkey Layout
• Dialing State
Custom softkey layout when device is under DIALING state.
Available softkeys: EndCall, Redial, Backspace, Dial, Clear.
• Onhook Dialing State
Custom softkey layout when device is under ONHOOK DIALING state.
Available softkeys: Answer, Reject, Forward, Dial, Backspace, Cancel.
• Ringing State
Custom softkey layout when device is under RINGING state.
Available softkeys: Answer, Reject, Forward.
• Calling State
Custom softkey layout when device is under CALLING state.
Available softkeys: Answer, Reject, Forward, EndCall.
• Call Connected State
Custom softkey layout when device is under CALL CONNECTED state.
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Available softkeys: Record On/Off(UCM), CallPark (UCM), ParkedCalls,
Answer, Answer, Reject, Forward, EndCall, PrivateHold.
• Conference Connected State
Custom softkey layout when device is under CONFERENCE CONNECTED
state.
Available softkeys: Record On/Off(UCM), ParkedCalls, Answer, Reject,
EndCall.
• On Hold State
Custom softkey layout when device is under ON HOLD state.
Available softkeys: ParkedCalls, ReConf.
• Call Failed State
Custom softkey layout when device is under CALL FAILED state.
Available softkeys: EndCalls, ReConf
• Transfer State
Custom softkey layout when device is under TRANSFER state.
Available softkeys: Cancel, Backspace, BlindTrnf, AttTrnf
• Conference State
Custom softkey layout when device is under CONFERENCE state.
Available softkeys: Cancel, Backspace.
Settings → Broadsoft XSI
XSI
XSI
• Server
Defines Broadsoft XSI server address with protocol.
• Port
Defines the port of XSI Server.
• XSI Actions Path
Configures the deployment path for Broadsoft XSI Actions. If it is
empty, the path “com.broadsoft.xsi-actions” will be used.
• Broadsoft Contact Download Interval
Defines the interval for update Broadsoft contacts.
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• XSI Authentication Type.
Specify preferred XSI authentication type: Login credentials, SIP
credentials, or using credentials from registered accounts.
Login Credentials
• Login Username
Username for Broadsoft XSI server
• Login Password
Password for Broadsoft XSI server
Note : The users can enter login credentials through LCD for Broadsoft
Directories.
SIP Credentials
• SIP Username
SIP username for Broadsoft XSI server
• SIP User ID
SIP user ID for Broadsoft XSI server
• SIP Password
SIP password for Broadsoft XSI server
Sort Phonebook by: Allows users to sort the Broadsoft Phone search based
on the selection of first name or last name. Default setting is “Last Name”.
Network Directories
Enable/Disable Broadsoft Network directories and defines the directory name.
The directory types are:
• Group Directory
• Enterprise Directory
• Group Common
• Enterprise Common
• Personal Directory
• Missed Call Log
• Placed Call Log
• Received Call Log
Settings → External Service
Grandstream Door
System
Connect to a GDS37XX and send OpenDoor request.
• Account
The account to be used on the phone to interact with the GDS37XX.
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• System Identification
A name or a number to identify the GDS37XX.
• System Number
The SIP extension or the IP address of the GDS37XX depending on
the deployed scenario, Peering or Registration.
• Access Password
The password set on the GDS37XX to unlock the door.
Notes:
- When using Peering scenario, on “System Number” field of the
GXP16XX specify the IP address of the peered GDS37XX.
- When using Registration scenario and both GXP16XX and GDS37XX
are registered on the same SIP server, specify the SIP extension of the
GDS37XX on “System Number” field on GXP16XX.
- The “Access Password” on GXP16XX should be matching “Remote
PIN to Open the door” on GDS37XX.
Network Page Definitions
Table 10: Network Page Definitions
Network → Basic Settings
Internet Protocol
Selects one of the following IP protocol modes:
• IPv4 Only: Enforce IPv4 protocol only.
• IPv6 Only: Enforce IPv6 protocol only.
• Both, Prefer IPv4: Enable both IPv4 and IPv6 and prefer IPv4.
• Both, prefer IPv6: Enable both IPv4 and IPv6 and prefer IPv6.
Note: Make sure to reboot the phone for the changes to take effect.
IPv4 Address
Allows users to configure the appropriate network settings on the phone to
obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE".
By default, it is set to "DHCP".
Host name (Option 12)
Specifies the name of the client, added for the DHCP INFORM using Option
12. This field is optional but may be required by some Internet Service
Providers.
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Vendor Class ID
(Option 60)
Used by clients and servers to exchange vendor class ID.
PPPoE Account ID
Enter the PPPoE account ID.
PPPoE Password
Enter the PPPoE Password.
PPPoE Service Name
Enter the PPPoE Service Name.
IPv4 Address
Enter the IP address when 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.
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. It can be set to EAP-MD5, EAP-TLS or EAPPEAPv0/MSCHAPv2.
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802.1X Identity
Enter the Identity for the 802.1X mode.
MD5 Password
Enter the MD5 Password for the 802.1X mode.
Note: the use of special characters is supported.
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.
Enable DHCP VLAN
Enables VLAN settings auto-configuration using DHCP options 132 & 133
tunneled through DHCP Option 43.
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
Configures the PC port mode. When set to "Mirrored", the traffic in the LAN
port will go through PC port as well and packets can be captured by connecting
a PC to the PC port. The default setting is "Enable".
PC Port VLAN Tag
Assigns the VLAN Tag of the PC port.
PC Port Priority Value
Assigns the priority value of the PC port.
Enable CDP
Allows users to enable/disable CDP feature which will allow the phone to
receive CDP packets from the uplink. Default is Enabled.
Note: When CDP feature is enabled, make sure VLAN is also configured
accordingly. It will need reboot to take effect for both enabling/disabling.
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Enable LLDP
Users can Enable/Disable the LLDP (Link Layer Discovery Protocol) service to
advertise information about the phone to other devices on the network. Default
is Enabled.
Maximum Transmission
Unit (MTU)
Defines the Maximum Transmission Unit value (MTU) in bytes.
Default is 1500.
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 → Open VPN® Settings
OpenVPN® Enable
Enables/Disables the OpenVPN® feature. Default settings is No.
OpenVPN® Server
Address
Configures the address of the OpenVPN® server.
OpenVPN® Port
Defines the port of the OpenVPN® server. Default is 1194.
OpenVPN® Transport
Determines network protocol UDP or TCP used for OpenVPN®.
Default is UDP.
OpenVPN® CA
Uploads the OpenVPN® CA.
OpenVPN® Certificate
Uploads the OpenVPN® Certificate.
OpenVPN® Client Key
Uploads the OpenVPN® Client Key.
OpenVPN® Cipher
Method
Must be the same cipher method used by the OpenVPN® server
OpenVPN® Username
OpenVPN® authentication username (optional)
OpenVPN® Password
OpenVPN® authentication password (optional)
Network → SNMP Settings
Enable SNMP
Enable/Disable SNMP service. Default is No.
Version
Choose between (Version 1, Version 2, or Version 3).
Port
Listening Port of SNMP daemon (Default 161).
SNMP Trap IP
IP address of trap destination.
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SNMP Trap port
Port of Trap destination (Default 162)
SNMP Trap Interval
Time interval between traps (Default is 5).
SNMP Trap Version
Choose between (Version 1, Version 2).
Community
Name of SNMPv1/v2c community.
SNMP Trap Community
Name of SNMPv1/v2c trap community.
SNMP Username
Username for SNMPv3.
Security Level
noAuthUser: Users with security level noAuthnoPriv and context name as
noAuth.
AuthUser: Users with security level authNoPriv and context name as auth.
privUser : Users with security level authPriv and context name as priv.
Authentication Protocol
Select the Authentication Protocol: “None” or “MD5” or “SHA.”
Privacy Protocol
Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
Authentication Key
Enter the Authentication Key.
Privacy Key
Enter the Privacy Key.
SNMP Trap Username
Username for SNMPv3 Trap.
Trap Security Level
noAuthUser: Users with security level noAuthnoPriv and context name as
noAuth.
authUser: Users with security level authNoPriv and context name as auth.
privUser: Users with security level authPriv and context name as priv.
Trap Authentication
Protocol
Select the Authentication Protocol: “None” or “MD5” or “SHA”.
Trap Privacy Protocol
Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
Trap Authentication Key
Enter the Trap Authentication Key.
Trap Privacy Key
Enter the Trap Privacy Key.
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Maintenance Page Definitions
Table 11: Maintenance Page Definitions
Maintenance → Web Access
New Password
Allows the administrator to set the password for user-level web GUI access.
This field is case sensitive with a maximum length of 30 characters.
Confirm Password
Confirms the end user password field to be the same as above.
Current Password
Determines the current password which is required to set a new admin
password.
New Password
Allows users to change the admin password. The password field is purposely
hidden for security purpose. This field is case sensitive with a maximum length
of 30 characters.
Confirm Password
Confirms the admin password field to be the same as above.
Maintenance → Upgrade and Provisioning
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.
Always Authenticate
Before Challenge
Only applies to HTTP/HTTPS. If enabled, the phone will send credentials before
being challenged by the server.
Validate Hostname in
Certificate
To validate the hostname in the SSL certificate.
Allow DHCP Option 43
and Option 66 to
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.
Note:
Users should include http:// on the IP address or the URL of DHCP option 66
on the server if they want to use an http server. If the URL/IP is without a scheme
(protocol) the phone adds tftp:// by default. If the URL/IP includes tftp://, the
phone will not add tftp:// and use the URL/IP as it is.
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Additional Override
DHCP option
Additional DHCP Option that will be used as a firmware server instead of the
setting one or name server from option 43 and 66. However, this option will be
effective only when option 'Allow DHCP Option 43 and Option 66 to Override
Server' is enabled, the options are 150 and 160.
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 provisioning using PnP. If enabled, the phone will start
sending a SIP SUBSCRIBE message to a multicast IP address 224.0.1.75 at
boot up.
The phone should receive response message 200 or 500 from the server,
followed by a SIP NOTIFY including URL from where to download configuration
file to get provisioned. The default setting is “Yes” (Enabled).
Automatic Upgrade
Enables automatic upgrade and provisioning.
The default setting is "No".
Hour of the Day (0-23)
Defines the hour of the day to check the HTTP/TFTP server for firmware
upgrades or configuration files changes. The default value is 1.
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.
Firmware Upgrade
Confirmation
Allow users to automatically upgrade to the version that the server has provided
without need to confirm the upgrade to avoid end users manually select no when
doing mass upgrade. When set to “Yes”, the phone will ask the user to upgrade.
Default setting is “Yes”.
Provision
Click Start to trigger the Provision request.
Config Upgrade Via
Selects firmware upgrade/provisioning method: TFTP, HTTP, HTTPS, FTP, or
FTPS. Default is HTTPS.
Config Server Path
Defines the server path for provisioning. It could be different from the firmware
server for upgrading.
Support added to configure variables in the provisioning server URL. Currently,
the following variables are supported in the provisioning server URL:
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• $PN: it is used to identify the directory name of the provisioning server directory
where the corresponding boot files and configuration files are located.
• $MAC: it is used to identify the MAC address of the IP phone.
Variables $PN and $MAC can be embedded in server URL setting in Web UI
and also in DHCP Option 66. Example (Web UI): /192.168.0.2/$PN_$MAC
Example (DHCP Option 66): tftp://192.168.0.2/$PN_$MAC $PN will be replaced
with phone model, e.g., GXP1625 $MAC will be replaced with phone's MAC
address, e.g., 000b829a8ffe
Config HTTP/HTTPS
Username
The username for the HTTP/HTTPS server (configuration file).
Config HTTP/HTTPS
Password
The password for the HTTP/HTTPS server (configuration file).
Config File Prefix
This field enables user to store different configuration files in one single directory
on the configuration server. If configured, only the configuration file with the
matching prefix will be downloaded.
Config File Postfix
This field enables user to store different configuration files in one single directory
on the configuration server. If configured, only the configuration file with the
matching postfix will be downloaded.
XML Config File
Password
The password for encrypt/decrypt the XML configuration file using OpenSSL.
Authentication 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
User can press the “Download” button to export the device’s configuration file.
The configuration file is named as “config.txt”.
Upload Device
Configuration
User can press the “Upload” button to upload configuration files to the phone.
Firmware Upgrade Via
Allows users to choose the firmware upgrade method: TFTP, HTTP, FTP/FTPS
or HTTPS for firmware file download.
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Firmware Server Path
Defines the server path for the firmware server. It could be different from the
configuration server for provisioning.
Support added to configure variables in the provisioning server URL. Currently,
the following variables are supported in the provisioning server URL:
• $PN: it is used to identify the directory name of the provisioning server directory
where the corresponding boot files and configuration files are located.
• $MAC: it is used to identify the MAC address of the IP phone.
Variables $PN and $MAC can be embedded in server URL setting in Web UI
and also in DHCP Option 66. Example (Web UI): /192.168.0.2/$PN_$MAC
Example (DHCP Option 66): tftp://192.168.0.2/$PN_$MAC $PN will be replaced
with phone model, e.g., GXP1625 $MAC will be replaced with phone's MAC
address, e.g., 000b829a8ffe
Firmware HTTP/HTTPS
Username
The username for the HTTP/HTTPS server (Firmware file).
Firmware HTTP/HTTPS
Password
The password for the HTTP/HTTPS server (Firmware file).
Firmware File Prefix
This field enables user to store different firmware files in one single directory on
the firmware server. If configured, only the firmware file with the matching prefix
will be downloaded.
Firmware File Postfix
This field enables user to store different firmware files in one single directory on
the firmware server.
If configured, only the firmware file with the matching postfix will be downloaded.
Maintenance → Syslog
Syslog Protocol
Choose whether to send syslog messages over UDP or secured SSL/TLS
connection.
Default is UDP.
Notes:
• User can upload the CA file under Maintenance→Security settings.
• A reboot is required in order for the changes to take effect.
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.
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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).
Note: Syslog messages also includes MAC address and firmware version.
Syslog Keyword
Filtering
Configures syslog message to be filtered based on keywords provided.
If multiple keywords are filled, it should be separated by “,”.
Note: No spaces allowed.
Send SIP Log
Configures whether the SIP log will be included in the syslog messages or not.
The default setting is "No".
Maintenance → Language
Display Language
Selects display language on the phone.
Language File Postfix
Specifies the language file postfix for downloaded language.
Maintenance → Action URL
Setup Completed
Specifies action URL for signal to be sending out when phone finished setting
up processes.
Registered
Specifies action URL for signal to be sending out when phone successfully
registered.
Unregistered
Specifies action URL for signal to be sending out when phone successfully
unregistered SIP account.
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Register Failed
Specifies action URL for signal to be sending out when the IP phone fails to
register a SIP account.
Off Hook
Specifies action URL for signal to be sending out when phone is in off-hook
state.
On Hook
Specifies action URL for signal to be sending out when phone is in on-hook
state.
Incoming Call
Specifies action URL for signal to be sending out when phone received the
incoming call.
Outgoing Call
Specifies action URL for signal to be sending out when phone make an
outbound call.
Missed Call
Specifies action URL for signal to be sending out when phone missed a call.
Answered Call
Specifies action URL for signal to be sending out when phone answered a call.
Rejected Call
Specifies action URL for signal to be sending out when phone rejected a call.
Forwarded Call
Specifies action URL for signal to be sending out when phone forwarded a call.
Established Call
Specifies action URL for signal to be sending out when phone established a call.
Terminated Call
Specifies action URL for signal to be sending out when phone terminated a call.
Idle to Busy
Specifies action URL for signal to be sending out when phone changes from idle
to busy.
Busy to Idle
Specifies action URL for signal to be sending out when phone changes from
busy to idle.
Open DND
Specifies action URL for signal to be sending out when Do Not Disturb mode is
enabled.
Close DND
Specifies action URL for signal to be sending out when Do Not Disturb mode is
disabled.
Open Forward
Specifies action URL for signal to be sending out when forwarding is enabled.
Close Forward
Specifies action URL for signal to be sending out when forwarding is disabled.
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Open Unconditional
Forward
Specifies action URL for signal to be sending out when phone disables call
unconditional forward option.
Close Unconditional
Forward
Specifies action URL for signal to be sending out when phone enables call
unconditional forward option.
Open Busy Forward
Specifies action URL for signal to be sending out when phone enables call busy
forward option.
Close Busy Forward
Specifies action URL for signal to be sending out when phone disables call busy
forward option.
Open No Answer
Forward
Specifies action URL for signal to be sending out when phone enables call no
answer forward option.
Close No Answer
Forward
Specifies action URL for signal to be sending out when phone disables call no
answer forward option.
Blind Transfer
Specifies action URL for signal to be sending out when phone performed a blind
transfer.
Attended Transfer
Specifies action URL for signal to be sending out when phone performed an
attended transfer.
Transfer Finished
Specifies action URL for signal to be sending out when phone successfully
transferred a call.
Transfer Failed
Specifies action URL for signal to be sending out when phone fails to transfer a
call.
Hold Call
Specifies action URL for signal to be sending out when phone placed a call on
hold.
UnHold call
Specifies action URL for signal to be sending out when phone placed retrieved
a held call.
Mute call
Specifies action URL for signal to be sending out when mutes a call.
UnMute call
Specifies action URL for signal to be sending out when un-mutes a call.
Open Syslog
Specifies action URL for signal to be sending out when syslog is enabled on the
phone.
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Close Syslog
Specifies action URL for signal to be sending out when syslog is disabled on the
phone.
IP Change
Specifies action URL for signal to be sending out when IP address change.
Auto-Provisioning
Finish
Specifies action URL for signal to be sending out when phone completed auto
provisioning.
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 username 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.
Randomized TR069
Startup
When enabled, the phone will send out first TR-069 INFORM message to server
on randomized timing between 1 to 3600 seconds after phone boots up.
Maintenance → Security Settings → Security
Configuration via
Keypad Menu
Configures the access control for the users to configure from keypad Menu.
• Unrestricted. All the options can be accessed in keypad Menu.
• Basic settings only. CONFIG option will not be available in LCD Menu.
• Constraint Mode. CONFIG, FACTORY FUNCTIONS and NETWORK
options will not be available in LCD menu.
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• Locked Mode. Allows disabling the MENU. When this option is selected, it
will restrict user access to any settings on the LCD: allowing only making
and receiving phone calls and basic functionality provided by feature keys.
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 and then enter the password to unlock it.
If the Star Key Lock is enabled without specifying password, user can press and
hold the STAR * key for 4 seconds and press OK to unlock the phone.
Keypad Lock Type
Users can set Keypad Functional Lock to their phone instead of completely
locking the phone.
• If set to "Functional", only Functional Keys will be locked but you are still
allowed to dial out emergency calls;
• If set to "Full", all keys will be locked, and no emergency calls are allowed
Password to
lock/unlock
Configures the password to lock/unlock the keypad. The password field allows
number with up to 32 characters.
Emergency numbers
Emergency numbers, separated by comma (,).
Keypad Lock Timer
Defines the timeout (in seconds) of idle screen for locking keypad. Valid range
is 0 to 3600.
Validate Server
Certificates
Validate server certificates with our trusted list for TLS connections. If set to
"No", device will bypass certificate validation (not recommended).
SIP TLS Certificate
SSL Certificate used for SIP TLS Transport. Managed through “Upload” and
“Delete” buttons.
SIP TLS Private Key
SSL Private key used for SIP TLS Transport. Managed through “Upload” and
“Delete” buttons.
SIP TLS Private Key
Password
SSL Private key password used for SIP TLS Transport.
Custom Certificate
This feature allows users to update to the device their own certificate signed
by custom CA certificate to manage client authentication.
Web Access Mode
Allows users to enable web access for the phone by specifying which protocol
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(HTTP/HTTPS) will be used for web interface access or disable the web access
by selecting “Disabled”. Default setting is HTTP.
Enable User Web
Access
Administrator can disable or enable user web access.
HTTP Web Port
Configures the HTTP port under the HTTP web access mode.
HTTPS Web Port
Configures the HTTPS port under the HTTPS web access mode.
Disable SSH
Disable SSH access. SSH is enabled by default.
SSH Public Key
Allows to use authentication keys for SSH access. The public key should be
loaded to phone’s web UI while the private key should be used in the SSH tool
side. Note: This will allow upcoming SSH access without password.
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 logout time is used for web login, STAR
keypad unlock, and LCD restrict mode admin login. Range is 0-60 minutes.
Web Session Timeout
Configures how long the WEB GUI can remain idle before logout, the accepted
range of values is 2-60 minutes. The default setting is 10.
Web Access Attempt
Limit
Configures the number of logins attempts before lockout. The accepted range
is 1-10. The default value is 5
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 on
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 could upload up to 6 CA certificates files to be trusted by the phone and
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establish secure connections over SSL/TLS severs.
The certificates can either be uploaded directly via Web GUI or provisioned
using Pvalues [8433 – 8438].
Load CA Certificates
The phone will verify the server certificate based on the built-in, custom or both
trusted certificates list. Users can specify which CA certificates 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
Packet Capture
Now if users want to capture their phone’s traffic, they can use the packet
capture function on the phone.
• Press Start to capture packets on the phone. “Running” will be
displayed on the web UI indicating that the phone started the capture.
• Press Stop to stop the current running capture.
Press Download to save the captured trace locally.
Contacts Page Definitions
Table 12: Contacts Page Definitions
Contacts → Local Phonebook
Add Contact
Specifies Contact’s First Name, Last Name, Phone Number, Accounts and
Groups (Blacklist, Whitelist, Work, Friends and Family) to add one new contact
in phonebook.
Note: If the contact number belongs to Blacklist group, the call from this number
will be blocked. If the contact number belongs to Whitelist group, when the
phone is on DND mode, the call from whitelist number will be allowed.
Edit Contact
Edits selected contact.
Delete All Contacts
Deletes all contacts from phonebook.
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Contacts → Group Management
Add Group
Adds new groups. User can also edit and delete existing groups on this page.
Edit Group
Edits selected group.
Contacts → Phonebook Management
Enable Phonebook
XML Download
Configures to enable phonebook XML download. Users could select
HTTP/HTTPS/TFTP to download the phonebook file. The default setting is
"Disabled".
HTTP/HTTPS
Username
Defines the username for authentication within the HTTP/HTTPS server when
downloading the phonebook file.
HTTP/HTTPS
Password
Defines the password for authentication within the HTTP/HTTPS server when
downloading the phonebook file.
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. The default value is 0. The valid range is
5 to 720 minutes.
Remove Manuallyedited 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
one; When set to "Append", the imported groups will be attended with the
current one.
Sort Phonebook by
Sort the phonebook on the selection of first name or 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
Controls the behavior of phonebook key. There are four options: Default, LDAP
Search, Local Phonebook, Local Group.
Contacts → LDAP
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LDAP protocol
Allows users to select LDAP or LDAPS protocol. Default setting is LDAP.
Server Address
Configures the IP address or DNS name of the LDAP server.
Port
Configures the LDAP server port.
Base
Configures the LDAP search base. This is the location in the directory where
the search is requested to begin.
Examples:
dc=grandstream, dc=com
ou=Boston, dc=grandstream, dc=com
Username
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".
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LDAP Name Attributes
Specifies 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.
Examples:
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.
Examples:
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.
The 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
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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.
Public Mode
The GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 supports Hot Desking using public
mode. Under public mode, users could login the phone with the SIP account User ID and password. Please
follow the steps below to configure the phone for public mode:
1. Fill up the SIP server address for account 1 first.
2. Under Web GUI→Settings→General Settings, set "Public Mode" option to "Yes". Click "Save"
and reboot the phone.
3. When the phone boots up, SIP User ID and Password to register to the configured SIP server in
account 1 will be required. Enter the correct account information to log in to the phone. When
entering the account information, press softkey "123/abc" to toggle input method.
4. In login page, pressing CONF button on the phone will show phone's IP address.
5. After using the phone, go to LCD MENU→Log Out to log off the public mode.
Lock Settings on the Phone
The GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 support the “Locked Mode”, when this
feature is selected from Web GUI→Maintenance→Security→Configuration Via Keypad Menu, it will
restrict user access to any settings on the LCD: menu is disabled, only allow making and receiving phone
calls and basic functionality provided by the feature keys (transfer / conference /hold / mute / volume / send
/ speaker).
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Figure 3: Lock/Unlock Configuration via Keypad Menu
DHCP VLAN
This option can be enabled from web UI→Network→Advanced Settings, it adds support for DHCP option
132 & 133 tunneled through DHCP Option 43 on the phone to deploy is on the network. By enabling this
option, the phone will try to read VLAN setting from DHCP option 132 and 133. If both of values in these
two options are empty, phone will then read from option 43.
Figure 4: Enable DHCP VLAN
After user enables this feature, the phone will disable LLDP VLAN setting and try to read VLAN setting from
DHCP to override manual setting. Firstly, the phone will send discover package and receive offer package
for option 132 (VLAN ID) and option 133 (VLAN priority) via DHCP server.
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Figure 5: DHCP Option 132 & 133
If one of Option 132 or Option 133 is not empty, the phone will read values in these two options. However,
if Option 132 and Option 133 are both empty, the phone will retrieve the value of Option 43 for VLAN ID
and priority. If VLAN ID and priority are both valid, VLAN setting will take effect after phone rebooting; if one
of the values is invalid, DHCP VLAN setting will not override manual setting.
Ability to Configure Record Button
This option can be found under web UI→Account→Call Settings, it allows users to configure the DTMF
that will be sent once record button is pressed during the call.
Figure 6: Call Recording Feature Codes
Editing Contacts and Click-To -Dial
From GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 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.
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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 4 and Figure 5 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:
The IP address of the phone.
• 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 and 1 for account 2.
• password=admin/123:
The admin or user login passwords of phone's Web GUI.
Figure 7: Web GUI→Phonebook→Contacts
Add contacts.
Edit contact.
Click to call this contact
from the phone.
Click to dial from
available lines.
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Figure 8: Click-to-Dial
Saving the 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.
Action URL
Action URL options can be found under device web UI→Maintenance→Action URL.
To use 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
to notify the event to SIP server.
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Figure 9: Action URL Settings Page
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Table 13: Action URL - Supported Events
Supported Events
Setup Completed
Terminated Call
Blind Transfer
Registered
Idle to Busy
Attended Transfer
Unregistered
Busy to Idle
Transfer Failed
Register Failed
Open DND
Hold Call
Off Hook
Close DND
UnHold Call
On Hook
Open Forward
Mute Call
Incoming Call
Close Forward
UnMute Call
Outgoing Call
Open Unconditional Forward
Open Syslog
Missed Call
Close Unconditional Forward
Close Syslog
Answered Call
Open Busy Forward
IP Change
Rejected Call
Close Busy Forward
Auto-Provision Finish
Forwarded Call
Open No Answer Forward
Established Call
Close No Answer Forward
Table 14: Action URL - Supported Dynamic Variables
Supported Dynamic Variables
Dynamic Variable
Description
$phone_ip
The IP address of the phone
$mac
The MAC address of the phone
$product
The product name of the phone
$program_version
The software version of the phone
$hardware_version
The hardware version of the phone
$language
The display language of the phone
$local
The 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
$active_user
The account number during a call on the phone
After the user finishes setting Action URL on phone’s web UI, when the specific phone event occurs on the
phone, phone will send the Action URL to the specified SIP server. The dynamic variables in the Action URL
will be replaced by the actual values.
Here is an example:
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Configure the following Action URL on the phone’s web UI→Settings→Outbound Notification→Action URL:
Incoming Call: 172.18.24.103/mac=$mac&local=$local
Outgoing Call: 172.18.24.103/remote=$remote&phone_ip=$phone_ip
On hold: 172.18.24.103/program_version=$program_version
During incoming call, outgoing call, and call hold, capture the trace on the phone and exam 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: Outgoing Call,
Incoming Call and On Hold in the format of the defined action URL with the parameters replaced with actual
values.
Figure 10: Action URL Packet
The P values listed in below table are for the Action URL options.
Table 15: Action URL Parameters P-values
P Value
Web UI Option
Value Format
P8304
Setup Completed
String
P8305
Registered
P8306
Unregistered
P8307
Register Failed
P8308
Off Hook
P8309
On Hook
P8310
Incoming Call
P8311
Outgoing Call
P8312
Missed Call
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P22171
Answered Call
P8315
Rejected Call
P22172
Forwarded Call
P8313
Established Call
P8314
Terminated Call
P8328
Idle to Busy
P8329
Busy to Idle
P8316
Open DND
P8317
Close DND
P8318
Open Forward
P8319
Close Forward
P22164
Open Unconditional
Forward
P22165
Close Unconditional
Forward
P22166
Open Busy Forward
P22167
Close Busy Forward
P22168
Open No Answer
Forward
P22169
Close No Answer
Forward
P8320
Blind Transfer
P8321
Attended Transfer
P8322
Transfer Failed
P8324
Hold Call
P8325
UnHold Call
P8326
Mute Call
P8327
UnMute Call
P8330
Open Syslog
P8331
Close Syslog
P22170
IP Change
P8332
Auto-Provision Finish
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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.
Figure 11: Rebooting the phone via web GUI
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UPGRADING AND PROVISIONING
The GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630 can be upgraded via TFTP,
HTTP/HTTPS, FTP or FTPS by configuring the URL/IP Address for the TFTP/HTTP/HTTPS/FTP/FTPS
server and selecting a download method. Configure a valid URL for TFTP or HTTP/HTTPS or FTP/FTPS;
the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com/BETA
fw.mycompany.com
There are two ways to 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 Config.
2. In the Config options, select Upgrade.
3. Enter the firmware server path and select upgrade method. The server path could be in IP address
format or FQDN format.
4. Press the "OK" softkey. A reboot message window will be prompt.
5. Reboot the phone to have the change take effect.
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/FTP/FTPS. 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 3 minutes in a controlled LAN or 5-10 minutes over the Internet. We
recommend completing firmware upgrades in a controlled LAN environment whenever possible.
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Note:
Please do not interrupt or power cycle the GXP1610/GXP1615/GXP1620/GXP1625/GXP1628/GXP1630
during upgrading process.
No Local TFTP/HTTP Servers
Service providers should maintain their own firmware upgrade servers. For users who do not have a
TFTP/HTTP/HTTPS/FTP/FTPS server, some free Windows version TFTP servers are available for
download from:
http://tftpd32.jounin.net/tftpd32_download.html.
Please check our web site at http://www.grandstream.com/support/firmware for latest firmware.
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.
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 or FTP/FTPS. The "Config Server Path" is the TFTP or
HTTP/HTTPS or FTP/FTPS 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 field in the web configuration page. A parameter consists
of a Capital letter P and 2 to 5-digit numeric numbers. i.e., P2 is associated with the "Admin Password" in
the Web GUI→Maintenance→Web/Telnet Access page. For a detailed parameter list, please refer to the
corresponding firmware release configuration template.
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