Grandstream Networks GXP1782 IP Phone User Manual
Grandstream Networks, Inc. IP Phone
User Manual
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| Document Description | User Manual |
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| Date Submitted | 2016-07-12 00:00:00 |
| Date Available | 2016-07-12 00:00:00 |
| Creation Date | 2016-07-01 13:13:54 |
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| Document Title | User Manual |
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| Document Author: | Grandstream |
Grandstream Networks, Inc.
GXP1760 & GXP1780/82 - Small Business IP Phones
Administration Guide
COPYRIGHT
©2016 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.
GXP17xx Administration Guide
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GNU GPL INFORMATION
GXP1760/GXP1780/GXP1782 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.
GXP17xx Administration Guide
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Table of Contents
CHANGE LOG ................................................................................................................ 6
Firmware Version 1.0.0.1 ....................................................................................................................... 6
GUI INTERFACE EXAMPLES ........................................................................................ 7
WELCOME ..................................................................................................................... 8
PRODUCT OVERVIEW .................................................................................................. 9
Feature Highlights .................................................................................................................................. 9
GXP1760 Technical Specifications ........................................................................................................ 9
GXP1780/GXP1782 Technical Specifications ..................................................................................... 11
CONFIGURATION GUIDE ............................................................................................ 13
Configuration Via Keypad .................................................................................................................... 13
Configuration Via Web Browser ........................................................................................................... 19
Definitions ............................................................................................................................................ 20
Status Page Definitions ................................................................................................................ 20
Accounts Page Definitions ............................................................................................................ 21
Settings Page Definitions ............................................................................................................. 32
Network Page Definitions ............................................................................................................. 47
Maintenance Page Definitions ...................................................................................................... 49
Phonebook Page Definitions ........................................................................................................ 53
Blf Led Patterns ................................................................................................................................... 56
NAT Settings ........................................................................................................................................ 57
Weather Update ................................................................................................................................... 58
Editing Contacts and Click-To-Dial ...................................................................................................... 59
Wallpaper ............................................................................................................................................. 60
Saving The Configuration Changes ..................................................................................................... 61
Rebooting from Remote Locations ...................................................................................................... 62
Packet Capture .................................................................................................................................... 62
Multicast Paging ................................................................................................................................... 64
Configuring event list Blf ...................................................................................................................... 64
UPGRADING AND PROVISIONING ............................................................................ 65
Upgrade Via Keypad Menu .................................................................................................................. 65
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Shortcut of Upgrade and Provision Via Keypad Menu ........................................................................ 65
Upgrade Via Web GUI ......................................................................................................................... 66
No Local TFTP/HTTP Servers ............................................................................................................. 66
Configuration File Download ................................................................................................................ 67
RESTORE FACTORY DEFAULT SETTINGS ............................................................... 69
EXPERIENCING GXP1760/GXP1780/GXP1782 .......................................................... 71
GXP17xx Administration Guide
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Table of Tables
Table 1: GXP1760/GXP1780/GXP1780 Features in a Glance ..................................................................... 9
Table 2: GXP1760 Technical Specifications .................................................................................................. 9
Table 3: GXP1780/GXP1782 Technical Specifications ............................................................................... 11
Table 4: CONFIGURATION MENU ............................................................................................................. 13
Table 5: Status Page Definitions ................................................................................................................. 20
Table 6: Account Page Definitions .............................................................................................................. 21
Table 7: Settings Page Definitions .............................................................................................................. 32
Table 8: Network Page Definitions .............................................................................................................. 47
Table 9: Maintenance Page Definitions ....................................................................................................... 49
Table 10: Phonebook Page Definitions ....................................................................................................... 53
Table 11: BLF LED Patterns ........................................................................................................................ 56
Table of Figures
Figure 1: Keypad MENU Configuration ....................................................................................................... 18
Figure 2: Web Service ................................................................................................................................. 58
Figure 3: Web GUI - Phonebook->Contacts ............................................................................................... 60
Figure 4: Click-to-Dial.................................................................................................................................. 60
Figure 5: Download Wallpaper from Server ................................................................................................ 61
Figure 6: Upload Selected Wallpaper to Phone .......................................................................................... 61
Figure 7: Packet Capture in Idle ................................................................................................................. 62
Figure 8: Capture to USB Drive .................................................................................................................. 63
Figure 9: Capture to Internal Storage ......................................................................................................... 63
GXP17xx Administration Guide
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CHANGE LOG
This section documents significant changes from previous versions of administration guide for
GXP1760/GXP1780/GXP1782. Only major new features or major document updates are listed here. Minor
updates for corrections or editing are not documented here.
Firmware Version 1.0.0.1
This is the initial version for GXP1760/GXP1780/GXP1782
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GUI INTERFACE EXAMPLES
http://www.grandstream.com/sites/default/files/Resources/17xx_web_gui.zip
1. Screenshot of Login Page
2. Screenshots of Status Pages
3. Screenshots of Profiles Pages
4. Screenshots of DECT Pages
5. Screenshots of Phonebook Pages
6. Screenshots of Settings Pages
7. Screenshots of Maintenance Pages
GXP17xx Administration Guide
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WELCOME
Thank you for purchasing Grandstream Small Business IP phone GXP1760/GXP1780/GXP1782
Enterprise IP Phone. GXP17xx series is a next generation small –to –medium business IP phone. The
GXP1760 features 6 dual-color line keys with 3 SIP accounts, 4 XML programmable context-sensitive soft
keys, 200 x 80-pixel backlit LCD display; the GXP1780/GXP1782 supports 8 dual-color lines with 4 SIP
account,
4XML
programmable
keys,
200
80-pixel
backlight
LCD
display;
The
GXP1760/GXP1780/GXP1782 contains dual network ports with PoE, EHS (Electronic Hook-Switch) with
Plantronics headsets, 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
GXP17xx Administration Guide
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PRODUCT OVERVIEW
Feature Highlights
The following table contains the major features of the GXP1760 & GXP1780/82:
Table 1: GXP1760/GXP1780/GXP1780 Features in a Glance
GXP1760
GXP1780
GXP1782
6 lines
200*80-pixel backlit LCD display
4 XML programmable soft keys
9 Dedicated function Keys
4-way conference
8 lines
200*80-pixel backlit LCD display
4 XML programmable soft keys
9 Dedicated function Keys
5-way conference
GXP1760 Technical Specifications
The following table resumes all the technical specifications including the protocols / standards supported,
voice codecs, telephony features, languages and upgrade/provisioning settings for GXP1760
Table 2: GXP1760 Technical Specifications
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record,
SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP,
TR-069, 802.1x, TLS, SRTP, IPV6
Network Interface
Graphic Display
Dual switched auto-sensing 10/100 Mbps Ethernet ports with integrated PoE
200*80 pixel backlit LCD display
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Features Keys
6 line keys with up to 3 SIP accounts, 4 XML programmable context sensitive
softkeys, 5 navigation/menu keys, 11 dedicated function keys for : MESSAGE(with
LED indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET,
MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-
Voice Codecs
Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723, ILBC,
in-band and out-of-band DTMF(in audio, RFC2833, SIP INFO)
Auxiliary Ports
Telephony Features
RJ9 headset jack (allowing EHS with Plantronics headsets),
Hold,
transfer,
forward,
4-way
conference,
call
park,
shared-call-appearance(SCA)/bridged-line-appearance(BLA),
call
pickup,
downloadable
phonebook(XML, LDAP, up to 2000 items), call waiting, call log(up to 500 records),
XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible
dial plan, hot desking, personalized music ringtones and music on hold, server
redundancy and fail-over
HD Audio
Base Stand
QoS
Security
Yes, HD handset and speakerphone with support for wideband audio
Yes, 2 angle positions available, Wall Mountable(optional)
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
User and administrator level passwords, MD5 and MD5-sess based authentication,
256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control
Multi-Language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
Chinese, Korean, Japanese
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS, mass provisioning using TR-069 or
AES encrypted XML configuration file.
Power & Green
Universal power adapter included: Input:100-240V; Output: +5V, 1A ;
Energy Efficiency
Integrated Power-over-Ethernet(802.3af)
Max power consumption : 5W
Dimension: 231mm(W) x 167mm(L) x 86mm(H)
Unit weight: 0.925kg.
Package weight: 1.55kg
Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity:
Physical
Temperature and
Humidity
Package Content
10％ to 90% Non-condensing
GXP1760 phone, handset with cord, base stand, universal power supply, network
cable, Quick Installation Guide, GPL license
Compliance
FCC: Part 15 (CFR 47) Class B
CE: EN55022 Class B; EN55024 Class B; EN61000-3-2; EN61000-3-3;
EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; ASs/NZS 60950.1
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GXP1780/GXP1782 Technical Specifications
The following table resumes all the technical specifications including the protocols / standards supported,
voice codecs, telephony features, languages and upgrade/provisioning settings for GXP1780/GXP1782
Table 3: GXP1780/GXP1782 Technical Specifications
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record,
SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP,
TR-069, 802.1x, TLS, SRTP, IPV6
Network Interface
Dual switched Ethernet ports with integrated PoE. Auto-sensing 10/100Mbps
(GXP1780), 10/100/1000Mbps (GXP1782)
Graphic Display
Features Keys
200*80 pixel backlit LCD display
8 line keys with up to 4 SIP accounts, 4 XML programmable context sensitive
softkeys, 5 navigation/menu keys, 11 dedicated function keys for : MESSAGE(with
LED indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET,
MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-
Voice Codecs
Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723, ILBC,
in-band and out-of-band DTMF(in audio, RFC2833, SIP INFO)
Auxiliary Ports
Telephony Features
RJ9 headset jack (allowing EHS with Plantronics headsets),
Hold,
transfer,
forward,
5-way
conference,
call
park,
shared-call-appearance(SCA)/bridged-line-appearance(BLA),
call
pickup,
downloadable
phonebook(XML, LDAP, up to 2000 items), call waiting, call log(up to 500 records),
XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible
dial plan, hot desking, personalized music ringtones and music on hold, server
redundancy and fail-over
HD Audio
Base Stand
QoS
Security
Yes, HD handset and speakerphone with support for wideband audio
Yes, 2 angle positions available, Wall Mountable(optional)
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
User and administrator level passwords, MD5 and MD5-sess based authentication,
256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control
Multi-Language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
Chinese, Korean, Japanese
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS, mass provisioning using TR-069 or
AES encrypted XML configuration file.
Power & Green
Universal power adapter included: Input:100-240V; Output: +5V, 1A ;
Energy Efficiency
Integrated Power-over-Ethernet(802.3af)
Max power consumption : 5W
Dimension: 231mm(W) x 167mm(L) x 86mm(H)
Unit weight: 0.925kg.
Package weight: 1.55kg
Physical
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Temperature and
Humidity
Package Content
Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity:
10％ to 90% Non-condensing
GXP1780/GXP1782 phone, handset with cord, base stand, universal power
supply, network cable, Quick Installation Guide, GPL license
Compliance
FCC: Part 15 (CFR 47) Class B
CE: EN55022 Class B; EN55024 Class B; EN61000-3-2; EN61000-3-3;
EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; ASs/NZS 60950.1
GXP17xx Administration Guide
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CONFIGURATION GUIDE
The GXP1760/ GXP1780/ GXP1782 can be configured via two ways:
LCD Configuration Menu using the phone's keypad;
Web GUI embedded on the phone using PC's web browser.
Configuration Via Keypad
To configure the LCD menu using phone's keypad, follow the instructions below:
Enter MENU options. When the phone is in idle, press the round MENU button to enter the
configuration menu;
Navigate in the menu options. Press the arrow keys up/down/left/right to navigate in the menu
options;
Enter/Confirm selection. Press the round MENU button or “Select” soft key to enter the selected
option;
Exit. Press “Exit” soft key to exit to the previous menu;
Return to Home page.
In the Main menu, press Home soft key to return home screen;
In sub menu, press and hold “Exit” soft key until the Exit soft key changes to Home soft key, then
release the soft key.
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 UP navigation key can see phone’s IP address.
The MENU options are listed in the following table.
Table 4: CONFIGURATION MENU
Call History
Displays Local call logs:
All Calls/Answered Calls/Dialed Calls/Missed Calls/Transferred Calls
Status
Displays account status, network status, software version number and Hardware
Account status.
Network status.
Press to enter the sub menu for MAC address, IP setting information
(DHCP/Static IP/PPPoE), IPv4 address, IPv6 address, Subnet Mask, Gateway
and DNS server.
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System Information
Press to enter the sub menu for Hardware version, P/N number. Boot, Core,
Base, Prog version and IP Geographic Information.
Contacts
Contacts sub menu includes the following options:
Local Phonebook
Local Group
LDAP Directory
Contacts sub menu is for Local Phonebook, Local Group, LDAP Directory and
Broadsoft Phonebooks. User could configure phonebooks/groups/LDAP options
here, download phonebook XML to the phone and search phonebook/LDAP
directory.
Messages
Message sub menu include the following options:
Instant Message
Displays received instant messages;
Voice Mails
Displays
voicemail message information in the format below:
new
messages/all messages (urgent messages/all urgent messages).
Preference
Preference sub menu includes the following options:
Do Not Disturb
Enables/disables Do Not Disturb on the phone.
Star Key Lock
Turns on/off keypad lock feature and configures keypad lock password. The
default keypad lock password is null. If user enabled Star Key lock without
configuring password, user can unlock keypad by holding * key 4 seconds and
pressing “OK” button.
Sounds
Ring Tone
Configures different ring tones for incoming call.
Ring Volume
Adjusts ring volume by pressing left/right arrow key.
Appearance
Active LCD Brightness
Adjusts active LCD brightness by pressing left/right arrow key
Idle LCD Brightness
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Adjusts idle LCD brightness by pressing left/right arrow key
Screensaver
Enable/Disable Screensaver
Screensaver Timeout
Configures the minutes of idle before the screensaver activates.
Valid range is 3 to 6.
Language and Input
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. By default, it is Auto.
Default Input Selection
Selects the Input mode from Multi-Tap and Shiftable. By default,
it is Multi-Tap.
Multi-Tap: User may tap the same key multiple times to switch to
the desired character.
Shiftable: After pressing the number button, user will see the IDs
of the characters that matching to the button. User can select the
desired character by entering the corresponding ID on keypad.
Date Time
Allow DHCP Option 42 to override NTP server
Allow DHCP Option 2 to override Time Zone setting
Time Settings
It is used to configure date and time on the phone.
Search Mode
Specifies the phonebook search mode to QuickMatch or ExactMatch. By
default, it is QuickMatch.
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.
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System
System sub menu includes the following options:
Network
IP Setting
Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE
account ID and password; Configures static IP address, Netmask,
Gateway, DNS Server 1 and DNS Server 2.
802.1X
Enables/Disables 802.1X mode; Configures 802.1x identity and
MD5 password.
Layer 2 QoS
Configures 802.1Q/VLAN Tag and priority value. Select “Reset
Vlan Config” to reset VLAN configuration.
Upgrade
Firmware Server
Configures firmware server for upgrading the phone.
Config Server
Configures config server for provisioning the phone.
Upgrade Via
Specifies upgrade/provisioning via TFTP/HTTP/HTTPS.
Start Provison
Starts Provision immediately.
Language Download
Auto Language Download
Language Download
Factory Functions
Diagnostic Mode
All LEDs will light up. All keys’ name will display in red on LCD
screen before diagnosing. Press any key on the keypad to
diagnose the key’s function. When done, the key’s name will
display in blue on LCD. Lift and put back the handset to exit
diagnostic mode.
Audio Loopback
Speak to the phone using speaker/handset/headset. If you can
hear your voice, your audio is working fine. Press “Exit” soft key to
exit audio loopback mode.
LCD on/off
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Selects this option to turn off LCD. Press any button to turn on
LCD.
LCD Diagnostic
Enters this option and press Left/Right Navigation key to do LCD
Diagnostic. Press “Exit” soft key to quite.
Certificate Verification
This is used to validate certificate chain for the server’s certificate.
UCM Detect
Detect/connect UCM server to process auto-provision. Manually input the
IP and port of the UCM server phone wants to bind with; Or select from the
available UCM server in network.
Operations
Factory Reset
It is used to restore the phone to factory default settings.
Reboot
Reboots the phone.
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The following picture shows the keypad MENU configuration flow:
Call History
Status
Contacts
Messages
Preference
Phone
Local Call Log
Broadsoft Call Log
Account Status
Network Status
System Status
Local Phonebook
Local Group
LDAP Directory
Broadsoft Phonebook
Ring Tone
Ring Volume
Instant Messages
Voice Mails
Active LCD Brightness
Idle LCD Brightness
Screensaver
Screensaver Timeout
Do Not Disturb
Forward Call
Ring Tone
Ring Volume
LCD Contrast
LCD Brightness
Download SCR XML
Erase Custom SCR
Display Language
Date and Time
Star key Lock
SIP
Call Features
System
Enable/Disable
Password
Network
Upgrade
Language Download
Factory Functions
UCM Detect
Factory Reset
Reboot
Display Language
Default Input Selection
Internet Protocol
Ipv4 settings
DHCP settings
PPPoE Settings
Static IP Settings
802.1x
Layer 2 QoS
PC Port Settings
IPv6 settings
Firmware Server
Config Server
Upgrade via
Start Provision
Diagnostic Mode
Audio Loopback
LCD on/off
LCD Diagnostic
Certificate Verification
Figure 1: Keypad MENU Configuration
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Configuration Via Web Browser
The GXP1760/GXP1780/GXP1782 embedded Web server responds to HTTP/HTTPS GET/POST
requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as
Google Chrome, Mozilla Firefox and Microsoft’s IE.
To access the Web GUI:
1. Connect the computer to the same network as the phone;
2. Make sure the phone is turned on and shows its IP address. You may check the IP address by
pressing Up arrow button when phone is at idle state;
3. Open a Web browser on your computer;
4. Enter the phone’s IP address in the address bar of the browser;
5. Enter the administrator’s login and password to access the Web Configuration Menu.
Note:
The computer has to be connected to the same sub-network as the phone. This can be easily done by
connecting the computer to the same hub or switch as the phone connected to. In absence of a
hub/switch (or free ports on the hub/switch), please connect the computer directly to the PC port on the
back of the phone;
If the phone is properly connected to a working Internet connection, the IP address of the phone will
display in MENU->Status->Network Status. This address has the format: xxx.xxx.xxx.xxx, where xxx
stands for a number from 0-255. Users will need this number to access the Web GUI. For example, if
the phone has IP address 192.168.40.154, please enter "http://192.168.40.154" in the address bar of
the browser;
There are two default passwords for the login page:
User Level
Password
Web Pages Allowed
End User Level
123
Only Status and Maintenance
Administrator Level
admin
All pages
The password is case sensitive with maximum length of 25 characters.
When changing any settings, always SUBMIT them by pressing the "Save" or "Save and Apply" button
on the bottom of the page. If the change is saved only but not applied, after making all the changes,
click on the "APPLY" button on top of the page to submit. After submitting the changes in all the Web
GUI pages, reboot the phone to have the changes take effect if necessary (All the options under
"Accounts" page and "Phonebook" page do not require reboot. Most of the options under "Settings"
page do not require reboot).
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Definitions
This section describes the options in the phone's Web GUI. As mentioned, you can log in as an
administrator or an end user.
Status: Displays the Account status, Network status, and System Info of the phone;
Account: To configure the SIP account;
Settings: To configure call features, ring tone, audio control, LCD display, date and time, Web
services, XML applications, programmable keys and etc.;
Network: To configure network settings;
Maintenance: To configure web access, upgrading and provisioning, syslog, language settings,
TR-069, security and etc.;
Phonebook: To manage Phonebook and LDAP.
Status Page Definitions
Table 5: Status Page Definitions
Status Account Status
Account index.
Account
For GXP1760: up to 3 SIP accounts
For GXP1780: up to 4 SIP accounts
For GXP1782: up to 4 SIP accounts
SIP User ID
SIP Server
SIP Registration
Displays the configured SIP User ID for the account.
Displays the configured SIP Server address, URL or IP address, and port of
the SIP server.
Displays SIP registration status for the SIP account, it will display Yes/No with
Green/Red background.
Status Network Status
Global unique ID of device, in HEX format. The MAC address will be used for
MAC Address
provisioning and can be found on the label coming with original box and on the
label located on the back of the device.
IP Setting
Configured address type: DHCP, Static IP or PPPoE.
IPv4 Address
Displays the IPv4 address obtained on the phone.
IPv6 Address
Displays the IPv6 address obtained on the phone.
Subnet Mask
Displays the subnet mask obtained on the phone.
Gateway
Displays the gateway address obtained on the phone.
DNS Server 1
Displays the DNS server address 1 obtained on the phone.
DNS Server 2
Displays the DNS server address 2 obtained on the phone.
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PPPoE Link Up
PPPoE connection status.
NAT Type
Displays the type of NAT connection used by the phone.
NAT Traversal
Displays the status of NAT connection for each account on the phone.
Status System Info
Product Model
Product model of the phone.
Part Number
Product part number.
Software Version
Boot: boot version number;
Core: core version number;
Base: base version number;
Prog: program version number. This is the main firmware release number,
which is always used for identifying the software system of the phone;
IP Geographic
Information
Locale: locale version number;
Recovery: recovery version number.
City: displaying phone location;
Language: displaying language;
Time Zone: displaying time zone;
System Up Time
System up time since the last reboot.
System Time
Current system time on the phone system.
Service Status
GUI and Phone service status.
Core Dump
Core dump file that could be downloaded for troubleshooting purpose.
Accounts Page Definitions
Table 6: Account Page Definitions
Account x General Settings
Account Active
Account Name
SIP Server
Secondary SIP Server
Indicates whether the account is active. The default setting is "Yes".
Configures the name associated with each account to be displayed on the
LCD.
Specifies the URL or IP address, and port of the SIP server. This should be
provided by VoIP service providers (ITSP).
Specifies the URL or IP address, and port of the SIP server. This will be used
when the primary SIP server fails.
Configures the IP address or the domain name of the primary outbound
proxy, media gateway or session border controller. It's used by the phone for
Outbound Proxy
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
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BLF Server
Configures the optional server used for SUBSCRIBE requests to indicate
other extensions status on the SIP server.
Configures user account information provided by your VoIP service provider
SIP User ID
(ITSP). It's usually in the form of digits similar to phone number or actually a
phone number.
Authenticate ID
Configures
the SIP service subscriber's
Authenticate ID used for
authentication. It can be identical to or different from the SIP User ID.
Configures the account password required for the phone to authenticate with
Authenticate Password
the ITSP (SIP) server before the account can be registered. After it is saved,
this will appear as hidden for security purpose.
Name
Specifies SIP server subscriber's name (optional) that will be used for Caller
ID display.
Sets if the phone system allows users to access the voice messages by
Voice Mail User ID
pressing the MESSAGE key on the phone. This ID is usually the VM portal
access number. For example, in UCM6100 IPPBX, *97 could be used.
Show Account Name
Enables to display the account name; if it is checked, the user ID will be
Only
hidden and the account name will be displayed only.
Account x Network Settings
Defines which parameter will control how the Search Appliance looks up IP
addresses for hostnames. There are four modes: A Record, SRV,
NATPTR/SRV, Use Configured IP. The default setting is "A Record". If the
user wishes to locate the server by DNS SRV, the user may select "SRV" or
"NATPTR/SRV".
If "Use Configured IP" is selected, please fill in the three fields below:
DNS Mode
Primary IP:
Backup IP 1;
Backup IP 2.
If SIP server is configured as domain name, phone will not send DNS query,
but use “Primary IP” or “Backup IP x” to send SIP message if at least one of
them are not empty. Phone will try to use “Primary IP” first. After 3 tries
without any response, it will switch to “Backup IP x”, and then it will switch
back to “Primary IP” after 3 re-tries.
If SIP server is already an IP address, phone will use it directly even “User
Configured IP” is selected.
Specifies which NAT traversal mechanism will be enabled on the phone
system. Users could select the mechanism from No, STUN, Keep-alive,
NAT Traversal
UPnP, Auto or VPN. The default setting is "No".
If set to "STUN" and STUN server is configured, the phone will route
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according to the STUN server. If NAT type is Full Cone, Restricted Cone or
Port-Restricted Cone, the phone will try to use public IP addresses and port
number in all the SIP&SDP messages. The phone will send empty SDP
packet to the SIP server periodically to keep the NAT port open if it is
configured to be "Keep-alive". Configure this to be "No" if an outbound proxy
is used. "STUN" cannot be used if the detected NAT is symmetric NAT. Set
this to "VPN" if OpenVPN is used.
Determines a SIP Extension to notify the SIP server that the phone is behind
Proxy-Require
a NAT/Firewall. Do not configure this parameter unless this feature is
supported on the SIP server.
Account x SIP Settings Basic Settings
Determines if the phone has an assigned PSTN telephone number, this field
should be set to "User=Phone". Then a "User=Phone" parameter will be
TEL URI
attached to the Request-Line and "TO" header in the SIP request to indicate
the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in
the SIP request. The default setting is "Disable".
SIP Registration
Selects whether or not the phone will send SIP Register messages to the
proxy/server. The default setting is "Yes".
If set to "Yes", the SIP user's registration information will be cleared when the
Unregister On Reboot
phone reboots. The SIP Contact header will contain "*" to notify the server to
unbind the connection. The default setting is "No".
Specifies the frequency (in minutes) in which the phone refreshes its
Register Expiration
registration with the specified registrar. The default value is 60 minutes. The
maximum value is 64800 minutes (about 45 days).
Reregister Before
Specifies the time frequency (in seconds) that the phone sends
Expiration
re-registration request before the Register Expiration. The default value is 0.
Defines the local SIP port used to listen and transmit. The default value is
Local SIP Port
5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066 for Account
4, 5068 for Account 5, 5070 for Account 6. The valid range is from 1 to 65535.
SIP Registration Failure
Specifies the interval to retry registration if the process is failed. The valid
Retry Wait Time
range is 1 to 3600. The default value is 20 seconds.
SIP T1 Timeout is an estimate of the round trip time of transactions between
SIP T1 Timeout
a client and server. If no response is received the timeout is increased, and
request re-transmit retries would continue until a maximum amount of time
define by T2. The default setting is 0.5 seconds.
SIP T2 Timeout is the maximum retransmit time of any SIP request
SIP T2 Timeout
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.
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SIP Transport
Determines the network protocol used for the SIP transport. Users can
choose from TCP, UDP and TLS. The default setting is “UDP”.
SIP URI Scheme when
Specifies if "sip" or "sips" will be used when TLS/TCP is selected for SIP
using TLS
Transport. The default setting is "sips".
Use Actual Ephemeral
Port in Contact with
TCP/TLS
This option is used to control the port information in the Via header and
Contact header. If set to No, these port numbers will use the permanent
listening port on the phone. Otherwise, they will use the ephemeral port for the
particular connection. The default setting is “No”.
Configures to remove outbound proxy from route. This is used for the SIP
Remove OBP from route
Extension to notify the SIP server that the device is behind a NAT/Firewall.
The default setting is "No".
Support SIP Instance ID
Defines whether SIP Instance ID is supported or not. The default setting is
"Yes".
When set to "Yes", a SUBSCRIBE for Message Waiting Indication will be sent
SUBSCRIBE for MWI
periodically. The phone supports synchronized and non-synchronized MWI.
The default setting is "No".
SUBSCRIBE for
When set to "Yes", a SUBSCRIBE for Registration will be sent out
Registration
periodically. The default setting is "No".
The use of the PRACK (Provisional Acknowledgment) method enables
reliability to SIP provisional responses (1xx series). This is very important in
Enable 100rel
order to support PSTN internetworking. To invoke a reliable provisional
response, the 100rel tag is appended to the value of the required header of
the initial signaling messages. The default setting is "No".
When set to "Auto", the phone will look for the caller ID in the order of
P-Asserted Identity Header, Remote-Party-ID Header and From Header in
Caller ID Display
the incoming SIP INVITE. When set to "Disabled", all incoming calls are
displayed with "Unavailable". When set to "From Header", the phone will
display the caller ID based on the From Header in the incoming SIP INVITE.
The default setting is "Auto".
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
Use Privacy Header
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. The default setting is
"Default".
Controls whether the P-Preferred-Identity Header will present in the SIP
Use P-Preferred-Identity
INVITE message. The default setting is "default": the P-Preferred-Identity
Header
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
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"No", the P-Preferred-Identity Header will not show in INVITE.
Add Auth Header on
To define whether authorization Header will be added on initial REGISTER
Initial REGISTER
from the first REGISTER. The default setting is “No”.
This is used to perform a factory reset through SIP NOTIFY. When the phone
Allow SIP Reset
receives the NOTIFY with event: RESET, the phone should perform a factory
reset after the authentication. The default setting is “No”.
Account x SIP Settings Advanced Features
For Shared Call Appearance, phone must send a SUBSCRIBE-request for
Line Seize Timeout
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.
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.,
Eventlist BLF URI
BLF1006@myserver.com) with a list of extension included. On the phone, in
this "eventlist BLF URI" field, fill in the URI without the domain (i.e.,
BLF1006).
To
monitor
the
extensions
in
the
list,
under
Web
GUI->Settings->Programmable Keys page, please select "eventlist BLF" in
the key mode, choose account, enter the value of each extension in the list.
Auto Provision Eventlist
BLFs
Conference URI
Music On Hold URI
When option is enabled, empty multi-purpose keys will be automatically
provisioned to the monitored extensions in the Eventlist BLF. The default
setting is “Disabled”.
Configures Conference URI for N-way conference (Broadsoft Standard).
Configures Music On Hold URI to call when a call is on hold. This feature has
to be supported on the server side.
Force BLF Call-pickup
Configures to always use the prefix for BLF Call-pickup. The default setting is
by prefix
“No”.
BLF Call-pickup Prefix
PUBLISH for Presence
Omit charset=UTF-8 in
MESSAGE
Configures the prefix prepended to the BLF extension when the phone picks
up a call with BLF key. The default setting is **.
Enables presence feature on the phone. The default setting is "No".
Omit charset=UTF-8 in MESSAGE content-type
Different soft switch vendors have special requirements. Therefore, users
may need select special features to meet these requirements. Users can
Special Feature
choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro,
Huawei IMS and PhonePower depending on the server type. The default
setting is "Standard".
Broadsoft Call Center
Default setting is "No". When set to "Yes", a soft key “BSCCenter” is
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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.
Broadsoft Hoteling event feature. Default setting is "No". With “Hoteling
Hoteling Event
Event” enabled, user can access the Hoteling feature option by pressing the
“BSCCenter” sof tkey.
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".
Broadsoft Executive
When enabled, Feature Key Synchronization will be enabled regardless of
Assistant
web settings.
This feature is used for Broadsoft call feature synchronization. When it's
Feature Key
enabled, DND, Call Forward features and Call Center Agent status can be
Synchronization
synchronized between Broadsoft server and phone. The default setting is
"Disabled".
Broadsoft Call Park
When enabled, it will send SUBSCRIBE to Broadsoft server to obtain Call
Park notifications. The default setting is “Disabled”.
Account x SIP Settings Session Timer
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
Session Expiration
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
Caller Request Timer
The minimum session expiration (in seconds). The default value is 90
seconds. The valid range is from 90 to 64800.
If set to "Yes" and the remote party supports session timers, the phone will
use a session timer when it makes outbound calls. The default setting is "No".
If set to "Yes" and the remote party supports session timers, the phone will
Callee Request Timer
use a session timer when it receives inbound calls. The default setting is
"No".
If Force Timer is set to "Yes", the phone will use the session timer even if the
Force Timer
remote party does not support this feature. If Force Timer is set to "No", the
phone will enable the session timer only when the remote party supports this
feature. To turn off the session timer, select "No". The default setting is "No".
UAC Specify Refresher
UAS Specify Refresher
As a Caller, select UAC to use the phone as the refresher; or select UAS to
use the Callee or proxy server as the refresher. The default setting is "Omit".
As a Callee, select UAC to use caller or proxy server as the refresher; or
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select UAS to use the phone as the refresher. The default setting is "UAC".
The Session Timer can be refreshed using the INVITE method or the
Force INVITE
UPDATE method. Select "Yes" to use the INVITE method to refresh the
session timer. The default setting is "No".
Account x SIP Settings Security Settings
Check Domain
Choose whether the domain certificates will be checked or not when
Certificates
TLS/TCP is used for SIP Transport. The default setting is "No".
Validate Certificate
Validate certification chain when TCP/TLS is configured. The default setting
Chain
is “No”.
Validate Incoming
Choose whether the incoming messages will be validated or not. The default
Messages
setting is "No".
Check SIP User ID for
incoming INVITE
Accept Incoming SIP
from Proxy Only
Authenticate Incoming
INVITE
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming
INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected.
The default setting is "No".
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
account, the call will be rejected. The default setting is "No".
If set to "Yes", the phone will challenge the incoming INVITE for
authentication with SIP 401 Unauthorized response. The default setting is
"No".
Account x Audio Settings
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), via RTP (RFC2833), or via
SIP INFO.
• In audio, which means DTMF is combined in the audio signal (not very
Send DTMF
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's easily to cause desynchronized of DTMF and media packet for the reason
the SIP and RTP are transmitted respectively. The default setting is
"RFC2833".
DTMF Payload Type
This parameter sets the payload type for DTMF using RFC2833. Default is
101. The valid range is from 96 to 127.
Multiple vocoder types are supported on the phone, the vocoders in the list is
Preferred Vocoder
a higher preference. Users can configure vocoders in a preference list that is
included with the same preference order in SDP message.
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Use First Matching
When it is set to "Yes", the device will use the first matching vocoder in the
Vocoder in 200OK SDP
received 200OK SDP as the codec. The default setting is "No".
Disable Multiple m line
When it is set to “No”, the device will reply with multiple m lines; Otherwise, it
in SDP
will reply 1 m line. The default setting is “No”.
SRTP Mode
Enable SRTP mode based on your selection from the drop-down menu. The
default setting is "Disabled".
Enable or disable the crypto life time when using SRTP. If users set to disable
Crypto Life Time
this option, phone does not add the crypto life time to SRTP header. The
default setting is “Yes”.
Symmetric RTP
Defines whether symmetric RTP is supported or not. The default setting is
"No".
Controls the silence suppression/VAD feature of the audio codec G.729. If set
Silence Suppression
to "Yes", when silence is detected, a small quantity of VAD packets (instead
of audio packets) will be sent during the period of no talking. If set to "No", this
feature is disabled. The default setting is "No".
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the "ptime" value for the SDP will
Voice Frames Per TX
change with different configurations here. This value is related to the codec
used and the actual frames transmitted during the in payload call. For end
users, it is recommended to use the default setting, as incorrect settings may
influence the audio quality. The default setting is 2.
G.726-32 Packing Mode
Jitter Buffer Type
Jitter Buffer Length
Selects "ITU" or "IETF" for G726-32 packing mode. The default setting is
“ITU”.
Selects either Fixed or Adaptive for jitter buffer type, based on network
conditions. The default setting is "Adaptive".
Selects jitter buffer length from 100ms to 800ms, based on network
conditions. The default setting is "300ms".
Account x Call Settings
Selects whether or not to enable early dial. If it's set to "Yes", the SIP proxy
must support 484 responses. Early Dial means that the phone sends for each
Early Dial
pressed digit a SIP INVITE message to SIP server. SIP server looks into its
extensions and, if no match happened yet, it sends back a "484 Address
Incomplete" message. Otherwise, it executes the action. The default setting is
"No".
Dial Plan Prefix
Configures the prefix to be added to each dialed number.
A dial plan establishes the expected number and pattern of digits for a
Dial Plan
telephone number. This parameter configures the allowed dial plan for the
phone.
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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;
a) xx+ - at least 2 digit numbers
b) xx - only 2 digit numbers
c) ^ - exclude
d) [3-5] - any digit of 3, 4, or 5
e) [147] - any digit of 1, 4, or 7
f)
<2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
Example 1: {[369]11 | 1617xxxxxxx}
Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617;
Example 2: {^1900x+ | <=1617>xxxxxxx}
Block any number of leading digits 1900 or add prefix 1617 for any dialed 7
digit numbers;
Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}
Allows any number with leading digit 1 followed by a 3 digit number, followed
by any number between 2 and 9, followed by any 7 digit number OR Allows
any length of numbers with leading digit 2, replacing the 2 with 011 when
dialed.
Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right):
^1900x. - prevents dialing any number started with 1900;
<=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by
dialing 7 numbers and 1617 area code will be added automatically;
1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11
digits length;
011[2-9]x - allows international calls starting with 011;
[3469]11 - allows dialing special and emergency numbers 311, 411, 611
and 911.
Note:
In some cases where the user wishes to dial strings such as *123 to activate
voice mail or other applications provided by their service provider, the *
should be predefined inside the dial plan feature. An example dial plan will be:
{ *x+ } which allows the user to dial * followed by any length of numbers.
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Configures Call Log setting on the phone. You can log all calls, only log
Call Log
incoming/outgoing calls (missed calls will not be logged), or disable call log.
The default setting is "Log All Calls".
Account Ring Tone
Allows users to configure the ringtone for the account. Users can choose from
different ringtones from the dropdown menu.
Specifies matching rules with number, pattern or Alert Info text. When the
incoming caller ID or Alert Info matches the rule, the phone will ring with
selected distinctive ringtone. Matching rules:
Specific caller ID number. For example, 8321123;
A defined pattern with certain length using x and + to specify, where x
could be any digit from 0 to 9. Samples:
xx+ : at least 2-digit number;
xx : only 2-digit number;
[345]xx: 3-digit number with the leading digit of 3, 4 or 5;
Match Incoming Caller
ID
[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: ; info=priority
Selects the distinctive ring tone for the matching rule. When the incoming
caller ID or Alert Info matches the rule, the phone will ring with the selected
ring.
Ring Timeout
Defines the timeout (in seconds) for the rings on no answer. The default
setting is 60. The valid range is from 10 to 300.
If set to "Yes", the "From" header in outgoing INVITE messages will be set to
Send Anonymous
anonymous, essentially blocking the Caller ID to be displayed. The default
setting is "No".
Anonymous Call
Rejection
Auto Answer
Allow Auto Answer by
Call-Info
If set to "Yes", anonymous calls will be rejected. The default setting is "No".
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep. Default setting is “No”.
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls, based on the SIP info header sent from the
server/proxy. The default setting is "No".
Allow Barging by
When it is enabled, the phone will put the current call on hold automatically
Call-Info
and answer the incoming calls based on the SIP Call-Info header sent from
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the server/proxy. However, if the current call was answered based on the SIP
Call-Info header, then all other incoming calls with SIP Call-Info headers will
be rejected automatically. The default setting is “No”.
Custom Call-Info for
Used in addition to match the contents of the info parameter in the Call-Info
Auto Answer
header for auto answer.
Refer-To Use Target
If set to "Yes", the "Refer-To" header uses the transferred target's Contact
Contact
header information for attended transfer. The default setting is "No".
Transfer on Conference
Hangup
If set to "Yes", when the phone hangs up as the conference initiator, the
conference call will be transferred to the other parties so that other parties will
remain in the conference call. The default setting is "No".
Disable Recovery on
Disable recovery to the call to the transferee on failing blind transfer to the
Blind Transfer
target. The default setting is “No”.
No Key Entry Timeout
(s)
Defines the timeout (in seconds) for no key entry. If no key is pressed after
the timeout, the digits will be sent out. The default value is 4 seconds. The
valid range is from 1 to 15.
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
Use # as Dial Key
essentially equivalent to the "Send" key. If set to "No", the "#" key is included
as part of the dialing string and please make sure the dial plan is properly
configured to allow dialing # out. The default setting is “Yes”.
On Hold Reminder Tone
If set to “Enabled”, phone will play a reminder tone when it has a call on hold.
The default setting is “Disabled”.
Account x Feature Codes
When enabled, Do No Disturb, Call Forwarding and other call features can be
used via the local feature codes on the phone. Otherwise, the provisioned
Enable Local Call
feature codes from the server will be used. User configured feature codes will
Features
be used only if server provisioned feature codes are not provided. And once
feature codes are configured, either via server provisioning or local setting, a
softkey named “Features” will show on the LCD screen.
Do Not Disturb
(DND)--On
Do Not Disturb
(DND)--Off
Call Forward
Unconditionally
(All)--On
Configures DND feature code to turn on DND.
Configures DND feature code to turn off DND.
Configures Call Forward All feature code to activate unconditional call
forwarding.
Call Forward
Configures Call Forward All feature code to deactivate unconditional call
Unconditionally
forwarding
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(All)--Off
Call Forward Busy--On
Call Forward Busy--Off
Configures Call Forward Busy feature code to activate busy call forwarding.
Configures Call Forward Busy feature code to deactivate busy call
forwarding.
Call Forward Delayed
Configures Call Forward Delayed feature code to activate no answer call
(No Answer)--On
forwarding.
Call Forward Delayed
Configures Call Forward Delayed feature code to activate no answer call
(No Answer)--Off
forwarding.
Delayed
Defines the timeout (in seconds) before the call is forwarded on no answer.
Call Forward
Wait Time
The default value is 20 seconds. The valid range is 1 to 120.
Settings Page Definitions
Table 7: Settings Page Definitions
Settings General Settings
This parameter defines the local RTP port used to listen and transmit. It
is the base RTP port for channel 0. When configured, channel 0 will use
Local RTP Port
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.
When set to "Yes", this parameter will force random generation of both
the local SIP and RTP ports. This is usually necessary when multiple
Use Random Port
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.
Specifies how often the phone sends a blank UDP packet to the SIP
Keep-alive Interval
server in order to keep the "ping hole" on the NAT router to open. The
default setting is 20 seconds. The valid range is from 10 to 160.
Use NAT IP
The NAT IP address used in SIP/SDP messages. This field is blank at
the default settings. It should ONLY be used if it's required by your ITSP.
The IP address or Domain name of the STUN server. STUN resolution
STUN Server
results are displayed in the STATUS page of the Web GUI. Only
non-symmetric NAT routers work with STUN.
Public Mode
Configures to turn on/off the public mode for hot desking feature. The
default setting is “No”.
Settings Call Features
Off-hook Auto Dial
Configures a User ID/extension to dial automatically when the phone is
off hook. The phone will use the first account to dial out. The default
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setting is "No".
If configured, when the phone is on hook, it will go off hook after the
Off-hook Timeout
timeout (in seconds). The default value is 30 seconds. The valid range is
from 10 to 60.
Bypass Dial Plan Through
Enable/Disable the dial plan check while dialing through the call history
Call History and Directories
and any phonebook directories. The default setting is “No”.
Disable Call Waiting
Disables the call waiting feature. The default setting is "No".
Disable Call Waiting Tone
Disables the call waiting tone when call waiting is on. The default setting
is "No".
Disable Busy Tone on
Disable the busy tone heard in the handset when call is disconnected
Remote Disconnect
remotely. The default setting is “No”.
Disable Direct IP Call
Disables Direct IP Call. The default setting is "No".
When set to "Yes", users can dial an IP address under the same
LAN/VPN segment by entering the last octet in the IP address. To dial
quick IP call, off hook the phone and dial #XXX (X is 0-9 and XXX
Use Quick IP Call mode
<=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
Disable in-call DTMF Display
Disables the Conference function. The default setting is "No".
When it's set to "Yes", the DTMF digits entered during the call will not be
displayed on phone LCD. The default setting is "No".
Enable sending DTMF via
Enables certain MPKs to send DTMF during the call. This option does
specific MPKs
not affect Dial DTMF. The default setting is "No".
Disable Active MPK Page
When option is enabled, Active MPK Page on the extension boards will
be disabled. The default setting is "No".
Specifies the function of mute key in idle. Default setting is “DND”. When
Mute Key Functions While
select “Idle Mute” and press Mute key while idle, the future incoming call
Idle
will be answered with mute. When select “Disabled”, Mute key will not
take effect while idle. The default setting is “No”.
Disable Transfer
Disables the Transfer function. The default setting is "No".
In-call dial number on
Configures the number for the phone to dial as DTMF during the call
pressing transfer key
using TRAN button.
Auto-Attended Transfer
If set to "Yes", the phone will use attended transfer by default. The default
setting is "No".
Do Not Escape # as %23 in
Specifies whether to replace # by %23 or not for some special situations.
SIP URI
The default setting is "No".
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Enables Click-To-Dial feature. If this feature is enabled, user could click
Click-To-Dial Feature
the green dial button on left top corner of phone’s Web GUI, then choose
the account and dial to the target number. The default setting is
"Disabled".
Call History Flash Writing:
Defines the interval (in seconds) to save the call history to phone's flash.
Write Timeout
0 means this option is disabled. The default value is 300 seconds.
Call History Flash Writing:
Defines the number of unsaved logs before written to phone's flash. 0
Max Unsaved Log
means this option is disabled. The default value is 200 entries.
This option is used for users to set the default call log list after select
Default call log type
MENUCALL HISTORY. Broadsoft Call Log or Local Call Log option will
only show its own list. Default option will keep both call log lists.
Local Call Recording
Enables/Disables the ability to record calls locally while on the call
Feature
screen. The default setting is "Disabled".
Saved Local Call Recording
This option is used for users to set the location where the recordings will
Location
be stored.
Download Local Call
Recordings
When there are recordings presented, you may download them here.
Settings Multicast paging
During active call if incoming multicast page is higher priority (1 being the
Paging Barge
highest) than this value the call will be held and multicast page will be
played. The default setting is “Disabled”.
If enabled, during a multicast page if another multicast is received with
Paging Priority Active
higher priority (1 being the highest) that one will be played instead. The
default setting is “Disabled”.
The codec for sending multicast pages, there are 5 codecs could be
Multicast Paging Codec
used: PCMU, PCMA, G.726-32, G.729A/B, G.722 (wide band). The
default setting is “PCMU”.
Defines multicast listening addresses and labels. For example:
“Listening Address” should match the sender’s Value such as
Multicast Listening
“237.11.10.11:6767”
“Label” could be the description you want to use.
For details, please check the “Multicast Paging User Guide” on our
Website.
Settings Ring Tone
Call Progresses Tones:
Configures ring or tone frequencies based on parameters from local
System Ring Tone
telecom. The default value is North American standard. Frequencies
Dial Tone
should be configured with known values to avoid uncomfortable high
Second Dial Tone
pitch sounds.
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Message Waiting
Ring Back Tone
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
Call-Waiting Tone
(Frequencies are in Hz and cadence on and off are in 10ms)
Busy Tone
ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of
Reorder Tone
silence. In order to set a continuous ring, OFF should be zero. Otherwise
it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up
to three cadences are supported.
Call Waiting Tone Gain
This adjusts the call waiting tone volume. Users can select "Low",
"Medium" or "High". The default setting is "Low".
Settings Audio Control
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.
Headset Key Mode
When Speaker/Handset is being used in dialing/talking status,
press EADSET 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.
Configures to enable or disable the speaker to ring when headset is used
Always Ring Speaker
on "Toggle Headset/Speaker" mode. If set to "Yes", when the phone is in
Headset "Toggle Headset/Speaker" mode, both headset and speaker will
ring on incoming call. The default setting is "No".
Headset TX gain
Configures the transmission gain of the headset. The default value is
0dB.
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Headset RX gain
Handset TX gain
Configures the receiving gain of the headset. The default value is 0dB.
Configures the transmission gain of the handset. The default value is 0
dB.
Settings LCD Display
Backlight Brightness: Active
Backlight Brightness: Idle
Configures the LCD brightness when the phone is active. Valid range is
10 to 100 where 100 is the brightest. Default value is 100.
Configures the LCD brightness when the phone is idle. Valid range is 10
to 100 where 0 is off and 100 is the brightest. Default value is 60.
Disable Missed Call
If set to "Yes", the screen will turn off the LCD backlight when there is a
Backlight
missed call on the phone. The default setting is "No".
Specifies the wallpaper source mode: Default, Download, USB and
Wallpaper Source
Uploaded. User could upload a wallpaper source into your phone, or
download it from file server with the server path, or plug your USB drive
with wallpaper source into ph to upload the wallpaper.
Wallpaper Server Path
Upload Wallpaper
Screensaver
Specifies the wallpaper server path. This option will take effect when
wallpaper source is “Download”.
Click on the “Upload” button to browse and upload the desired wallpaper
file. This option will take effect when wallpaper source is “Uploaded”.
Disable/Enable Screensaver. By default, screensaver is enabled.
This option is used for users to set the location where screensaver is
Screensaver Source
loaded from. If from USB, please have a folder named “screensavers”
containing your pictures.
Screensaver Timeout
Configures the minutes of idle before the screensaver activates. Valid
range is 3 to 6. The default time is 3 minutes.
Settings LCD Control
This is used to configure the color and pattern of the LED based on
BLF LED Pattern
status updates. The default setting is “Default”.
The BLF LED Patterns are listed in the following Table 13.
Settings Date and Time
NTP Server
NTP Update Interval
Allow DHCP Option 42
Override NTP Server
Time Zone
Defines the URL or IP address of the NTP server. The phone may obtain
the date and time from the server. The default setting is us.pool.ntp.org.
Time interval for updating time from the NTP server. Valid time value is in
between 5 to 1440 minutes. The default setting is “1440” minutes.
Defines whether DHCP Option 42 should override NTP server or not.
When enabled, DHCP Option 42 will override the NTP server if it's set up
on the LAN. The default setting is "Yes".
Configures the date/time used on the phone according to the specified
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time zone.
This parameter allows the users to define their own time zone.
The syntax is: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0
MTZ+6MDT+5
This indicates a time zone with 6 hours offset with 1 hour ahead (when
daylight saving) which is U.S central time. If it is positive (+) if the local
time zone is west of the Prime Meridian (A.K.A: International or
Self-Defined Time Zone
Greenwich Meridian) and negative (-) if it is east.
M4.1.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3rd Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon,
Tues, ... ,Sat)
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
Configures the date display format on the LCD. The following formats are
supported:
Date Display Format
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
The default setting is yyyy-mm-dd.
Time Display Format
Configures the time display in 12-hour or 24-hour format on the LCD. The
default setting is in 12-hour format.
Settings Web Service
Configures to enable or disable weather update on the phone. The
Enable Weather Update
default setting is "Yes". If set to "No", the weather information screen will
not show.
Configures weather city code for the phone to look up the weather
information. The default setting is "Automatic" and the weather
City Code
information will be obtained based on the IP location of the phone if
available. Otherwise, specify the self-defined city code. For example,
USCA0638 is the city code for Los Angeles, CA, United States.
Update Interval
Degree Unit
Specifies the weather update interval (in minutes). The default value is
15 minutes.
Specifies the degree unit for the weather information to display on the
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phone. User could chose Fahrenheit, Celsius, or Auto to display the
degree unit. The default setting is “Auto”.
Configures to enable or disable currency update on the phone. The
Enable Currency Update
default setting is "Yes". If set to "No", the currency information screen will
not show.
Configures currency code for the phone to look up and display the
Currency Code
currency information. The default setting is:
“EUR/USD;GBP/USD;CAD/USD;AUD/USD;CNY/USD;JPY/USD”
Settings XML Application
Server Path
Softkey Label
Configures the server path to download the idle screen XML file. This
field could be IP address or URL, with up to 256 characters.
Specifies the softkey name displayed on the idle screen for the users to
enter XML application. The default Softkey Label is “XMLApp”.
Settings Programmable Keys
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.
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
the Value field to be dialed, or enter the IP address to set the Direct
Line Key
IP call as Speed Dial.
Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension
number in the Value field to be monitored.
Presence Watcher
This option has to be supported by a presence server and it is tied to
the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX
collects the information from the phones and sends it out in one
single notify message. PBX server has to support this feature.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active
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account. For example, if the phone is offhook and account 2 is active,
it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the Value field to be dialed during
the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value
field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Value field should be left blank. Also, this option is not binding to the
account and the call will be returned based on the account with the
last answered call.
Transfer
Select Account, and enter the number in the Value field to be
transferred (blind transfer) during the call.
Call Park
Select Account, and enter the call park extension in the Value field to
park/pick up the call.
Monitored Call Park
Select account from Account field, and enter the call park extension
in the Value field to park/pick up the call, and also monitor the parked
call via Line Key’s light.
Intercom
Select Account, and enter the extension number in the Value field to
do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP
search base in the Description field. It could be the same or different
from the Base in LDAP configuration under Advanced Settings. The
Base in LDAP configuration will be used if the Description field is left
blank. Enter the LDAP Name/Number filter in the Value field.
For example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since
the
Base
for
LDAP
server
configuration
is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Multicast Paging
This option is for multicast sending. Enter Line key description in
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Description field and multicast sending address in Value field.
Record
This option is for Recording calls. Enter Line key description in
Description filed and the recorded extension number in Value field.
Please make sure whether your VOIP provider supports this feature
before using it.
Call Log
Select Account and enter account number in the Value field to allow
configuration of call log for other extension.
Assigns a function to the corresponding softkeys.
The key mode options are:
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
the Value field to be dialed.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active
account. For example, if the phone is offhook and account 2 is active,
it will call the configured Speed Dial number using account 2.
Voice Mail
Select Account and enter the Voice Mail access number in the Value
field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Value field should be left blank. Also, this option is not binding to the
Softkeys
account and the call will be returned based on the account with the
last answered call.
Intercom
Select Account, and enter the extension number in the Value field to
do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP
search base in the Description field. It could be the same or different
from the Base in LDAP configuration under Advanced Settings. The
Base in LDAP configuration will be used if the Description field is left
blank. Enter the LDAP Name/Number filter in the Value field.
For example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
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Since
the
Base
for
LDAP
server
configuration
is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Call Log
Select Account and enter account number in the Value field to
access to the Call Log of that selected account.
Assigns a function to the corresponding Multi-Purpose Key. 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.
BLF (Busy Lamp Field)
This option has to be supported on the PBX and it indicates the
status of the extension. The three possible states are idle (green),
busy (red), ringing (blinking red).
Presence Watcher
This option has to be supported by a presence server and it is tied to
the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX
collects the information from the phones and sends it out in one
single notify message. PBX server has to support this feature.
Multi-Purpose Key
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active
account. For example, if the phone is offhook and account 2 is active,
it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the Value field to be dialed during
the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value
field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Value field should be left blank. Also, this option is not binding to the
account and the call will be returned based on the account with the
last answered call.
Transfer
Select Account, and enter the number in the Value field to be
transferred (blind transfer) during the call.
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Call Park
Select Account, and enter the call park extension in the Value field to
park/pick up the call.
Monitored Call Park
Select account from Account field, and enter the call park extension
in the Value field to park/pick up the call, and also monitor the parked
call via MPK button’s light.
Intercom
Select Account, and enter the extension number in the Value field to
do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP
search base in the Description field. It could be the same or different
from the Base in LDAP configuration under Advanced Settings. The
Base in LDAP configuration will be used if the Description field is left
blank. Enter the LDAP Name/Number filter in the Value field.
For example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since
the
Base
for
LDAP
server
configuration
is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Multicast Paging
This option is for multicast sending. Enter Line key description in
Description filed and multicast sending address in Value field.
Record
This option is for Recording calls. Enter Line key description in
Description filed and the recorded extension number in Value field.
Please make sure whether your VOIP provider supports this feature
before using it.
Call Log
Select Account and enter account number in the Value field to
access to the Call Log of that selected account.
Settings Programmable Keys Programmable Keys
Assigns a function to the corresponding softkeys.
Softkeys
The key mode options are:
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
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the Value field to be dialed.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active
account. For example, if the phone is offhook and account 2 is active,
it will call the configured Speed Dial number using account 2.
Voice Mail
Select Account and enter the Voice Mail access number in the Value
field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Value field should be left blank. Also, this option is not binding to the
account and the call will be returned based on the account with the
last answered call.
Intercom
Select Account, and enter the extension number in the Value field to
do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP
search base in the Description field. It could be the same or different
from the Base in LDAP configuration under Advanced Settings. The
Base in LDAP configuration will be used if the Description field is left
blank. Enter the LDAP Name/Number filter in the Value field.
For example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since
the
Base
for
LDAP
server
configuration
is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Call Log
Select Account and enter account number in the Value field to
access to the Call Log of that selected account.
Menu
This option is to take users to the main menu screen directly.
Settings Programmable Keys Virtual Multi-Purpose Keys (VPK)
Assigns a function to the corresponding VPK. The key mode options are:
VPK
Line
Regular line key to open up a line and switch line. The Value field
can be left blank.
Shared Line
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Share line for Shared Line Appearance feature. Select the Account
registered as Shared line for the line key. The Value field can be left
blank.
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
the Value field to be dialed, or enter the IP address to set the Direct
IP call as Speed Dial.
Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension
number in the Value field to be monitored.
Presence Watcher
This option has to be supported by a presence server and it is tied to
the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX
collects the information from the phones and sends it out in one
single notify message. PBX server has to support this feature.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active
account. For example, if the phone is offhook and account 2 is active,
it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the Value field to be dialed during
the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value
field.
Call Return
The last answered calls can be dialed out by using Call Return. The
Value field should be left blank. Also, this option is not binding to the
account and the call will be returned based on the account with the
last answered call.
Transfer
Select Account, and enter the number in the Value field to be
transferred (blind transfer) during the call.
Call Park
Select Account, and enter the call park extension in the Value field to
park/pick up the call.
Monitored Call Park
Select account from Account field, and enter the call park extension
in the Value field to park/pick up the call, and also monitor the parked
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call via Line Key’s light.
Intercom
Select Account, and enter the extension number in the Value field to
do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP
search base in the Description field. It could be the same or different
from the Base in LDAP configuration under Advanced Settings. The
Base in LDAP configuration will be used if the Description field is left
blank. Enter the LDAP Name/Number filter in the Value field.
For example:
If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com
Value -> sn=Li
Since
the
Base
for
LDAP
server
configuration
is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Multicast Paging
This option is for multicast sending. Enter Line key description in
Description field and multicast sending address in Value field.
Record
This option is for Recording calls. Enter Line key description in
Description filed and the recorded extension number in Value field.
Please make sure whether your VOIP provider supports this feature
before using it.
Call Log
Select Account and enter account number in the Value field to allow
configuration of call log for other extension.
Menu
This option is to take users to the main menu screen directly.
Settings Broadsoft Broadsoft Directories
Configures XSI Directory.
Server
Configure the BroadWorks Xsi server URI. If the server uses
HTTPS, please add the header “HTTPS” ahead of the Server
XSI
URI.
For instance, “https://SERVER_URI”.
Port.
Configure the BroadWorks Xsi server port. The default port is 80.
If the server uses HTTPS, please configure 443.
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User name.
Configure the Username for the BroadWorks Xsi feature.
Password.
Configure the password for the BroadWorks Xsi feature.
Enable/Disable Broadsoft Network directories and defines the directory
name. The directory types are:
Group Directory
Enable/Disable and rename the BroadWorks Xsi Group Direcory
features on the phone. If keep the Name box blank, the phone
will use the default name “Group” for it.
Enterprise Directory
Enable/Disable and rename the BroadWorks Xsi Enterprise
Directory features on the phone. If keep the Name box blank, the
phone will use the default name “Enterprise” for it.
Group Common
Enable/Disable and rename the BroadWorks Xsi Group
Common Directory features on the phone. If keep the Name box
blank, the phone will use the default name “Group Common” for
it.
Enterprise Common
Enable/Disable and rename the BroadWorks Xsi Enterprise
Common Directory features on the phone. If keep the Name box
Network Directories
blank, the phone will use the default name “Enterprise Common”
for it.
Personal Directory
Enable/Disable and rename the BroadWorks Xsi Personal
Directory features on the phone. If keep the Name box blank, the
phone will use the default name “Personal” for it.
Missed Call Log
Enable/Disable and rename the BroadWorks Xsi Missed Call
Log features on the phone. If keep the Name box blank, the
phone will use the default name “Missed” for it.
Placed Call Log
Enable/Disable and rename the BroadWorks Xsi Placed Call Log
features on the phone. If keep the Name box blank, the phone
will use the default name “Outgoing” for it.
Received Call Log
Enable/Disable and rename the BroadWorks Xsi Placed Call Log
features on the phone. If keep the Name box blank, the phone
will use the default name “Incoming” for it.
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Settings Broadsoft Broadsoft IM&P
Server
Broadsoft IM&P server address. Usually not necessary to
configure and can already be found in the Broadsoft IM&P
username.
LogIn Credentials
Port
Port for the Broadsoft IM&P server. Default port is 5222.
Username
Broadsoft IM&P username, not the Broadsoft account username.
Password
Broadsoft IM&P password, not the Broadsoft account password.
Enables Broadsoft Instant Message and Presence feature. The default
Broadsoft IM&P
setting is “Disabled”.
Associated Broadsoft
Specifies the associated account. User could choose each account on
Account
the phone.
Choose to whether or not login to the Broadsoft IM&P account at
Auto Login
boot-up. The default setting is “No”.
Choose whether or not to display non-xmpp contacts associated with the
Display Non XMPP Contacts
Broadsoft IM&P user. Non-xmpp contacts will not display a presence or
status message. The default setting is “No”.
Network Page Definitions
Table 8: Network Page Definitions
Network Basic Settings
Internet Protocol
Selects Prefer IPv4 or Prefer IPv6. The default setting is “Prefer IPv4”.
Allows users to configure the appropriate network settings on the phone to
IPv4 Address Type
obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE". By
default, it is set to "DHCP".
DHCP Host name
Specifies the name of the client. This field is optional but may be required by
(Option 12)
some Internet Service Providers.
DHCP Vendor Class ID
(Option 60)
Used by clients and servers to exchange vendor class ID. The default setting
is “Grandstream GXP1760” for GXP1760, ”Grandstream GXP1780” for
GXP1780, “Grandstream GXP1782” for GXP1780.
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.
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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.
Allows users to configure the appropriate network settings on the phone to
IPv6 Address Type
obtain IPv6 address. Users could select "Auto-configured" or "Statically
configured" for the IPv6 address type.
Static IPv6 Address
IPv6 Prefix Length
IPv6 Prefix
Enter the static IPv6 address when Full Static is used in "Statically configured"
IPv6 address type.
Enter the IPv6 prefix length when Full Static is used in "Statically configured"
IPv6 address type.
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
Allows the user to enable/disable 802.1X mode on the phone. The default
802.1X mode
value is disabled. To enable 802.1X mode, this field should be set to
EAP-MD5, users may also choose EAP-TLS, or EAP-PEAP.
802.1X Identity
Enter the Identity information for the 802.1x mode.
MD5 Password
Enter the MD5 Password for the 802.1X mode.
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
Upload 802.1X Client certificate to the phone; or delete existed 802.1X Client
Certificate
certificate from the phone.
HTTP Proxy
HTTPS Proxy
Layer 3 QoS for SIP
Layer 3 QoS for RTP
Layer 2 QoS
802.1Q/VLAN Tag
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.
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.
Defines the Layer 3 QoS parameter for SIP. This value is used for IP
Precedence, Diff-Serv or MPLS. The default value is 26.
Defines the Layer 3 QoS parameter for RTP. This value is used for IP
Precedence, Diff-Serv or MPLS. The default value is 46.
Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0.
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Layer 2 QoS 802.1p
Priority Value
Assigns the priority value of the Layer2 QoS packets. The default value is 0.
Configures the PC port mode. When set to "Mirrored", the traffic in the LAN
PC Port Mode
port will go through PC port as well and packets can be captured by
connecting a PC to the PC port. The default setting is "Enabled".
PC Port VLAN Tag
Assigns the VLAN Tag of the PC port. The default value is “0”.
PC Port Priority Value
Assigns the priority value of the PC port. The default value is “0”.
Enable LLDP
Control the LLDP (Link Layer Discovery Protocol) service. The default setting
is “Enabled”.
Maintenance Page Definitions
Table 9: Maintenance Page Definitions
Maintenance Web Access
User Password
Set new password for web GUI access as User. This field is case sensitive.
Confirm Password
Enter the new User password again to confirm.
Current Admin
Password
New Password
Confirm Password
The current admin password is required for setting a new admin password.
Set new password for web GUI access as Admin. This field is case
sensitive.
Enter the new Admin password again to confirm.
Maintenance Upgrade and Provisioning
Specifies how firmware upgrading and provisioning request to be sent:
Firmware Upgrade and
Always Check for New Firmware, Check New Firmware only when F/W
Provisioning
pre/suffix changes, Always Skip the Firmware Check. The default setting is
“Always Check for New Firmware”.
XML Config File
Password
HTTP/HTTPS User
Name
HTTP/HTTPS
Password
The password for encrypting the XML configuration file using OpenSSL.
This is required for the phone to decrypt the encrypted XML configuration
file.
The user name for the HTTP/HTTPS server.
The password for the HTTP/HTTPS server.
Always Authenticate
Only applies to HTTP/HTTPS. If enabled, the phone will send credentials
Before Challenge
before being challenged by the server. The default setting is “No”.
Upgrade Via
Allows users to choose the firmware upgrade method: TFTP, HTTP or
HTTPS. The default setting is “HTTP”.
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Firmware Server Path
Config Server Path
Defines the server path for the firmware server. It could be different from the
configuration server for provisioning.
Defines the server path for provisioning. It could be different from the
firmware server for upgrading.
Enables your ITSP to lock firmware updates. If configured, only the
Firmware File Prefix
firmware with the matching encrypted prefix will be downloaded and flashed
into the phone.
Enables your ITSP to lock firmware updates. If configured, only the
Firmware File Postfix
firmware with the matching encrypted postfix will be downloaded and
flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
Config File Prefix
configuration file with the matching encrypted prefix will be downloaded and
flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
Config File Postfix
configuration file with the matching encrypted postfix will be downloaded
and flashed into the phone.
Default setting is “Yes”. DHCP option 66 originally was only designed for
Allow DHCP Option 43
TFTP server. Later on it was extended to support an HTTP URL. GXP
and Option 66 Override
phones support both TFTP and HTTP server via option 66. Users can also
Server
use DHCP option 43 vendor specific option to do this. DHCP option 43
approach has priorities.
Allow DHCP Option
120 to override SIP
Server
3CX Auto Provision
Automatic Upgrade
Hour of the Day (0-23)
Day of the Week (0-6)
Enables DHCP Option 120 from local server to override the SIP Server on
the phone. The default setting is "No".
Enables automatic provision feature on the phone when 3CX is used as the
SIP server. The default setting is "Yes".
Enables automatic upgrade and provisioning. The default setting is "No".
Defines the hour of the day to check the HTTP/TFTP server for firmware
upgrades or configuration files changes. The default value is 1.
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
Device will not challenge NOTIFY with 401 when set to “Yes”. The default
Authentication
setting is “No”.
Authenticate Conf File
Download Device
Configuration
Upload Device
Authenticates configuration file before acceptance. The default setting is
"No".
Click to download phone’s configuration file in .txt format.
Upload configuration file to phone.
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Configuration
Maintenance Syslog
Syslog Server
The URL or IP address of the syslog server for the phone to send syslog to.
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:
Syslog Level
Send SIP Log
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).
Configures whether the SIP log will be included in the syslog messages.
The default setting is "No".
Auto Recover From
If set to “Yes”, the phone will automatically recover when running abnormal.
Abnormal
The default setting is “Yes”.
Maintenance Language
Selects display language on the phone. There are 21 languages can be set
Display Language
as display language, user could also choose “Auto” or “Downloaded
Language” as display language. The default setting is “Auto”.
Configure the default input selection.
Default Input Selection
Multi-Tap: multi-tap to switch character;
Shiftable: select input from available characters.
The default setting is “Multi-Tap”.
Auto language
This is used to configure the device to download language files
download
automatically from server. The default setting is “No”.
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
Periodic Inform
Enables periodic inform. If set to "Yes", device will send inform packets to
the ACS. The default setting is "No".
Sets up the periodic inform interval to send the inform packets to the ACS.
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Interval
Connection Request
Username
Connection Request
Password
Connection Request
Port
The user name for the ACS to connect to the phone.
The password for the ACS to connect to the phone.
The port for the ACS to connect to the phone.
CPE SSL Certificate
The Cert File for the phone to connect to the ACS via SSL.
CPE SSL Private Key
The Cert Key for the phone to connect to the ACS via SSL.
Maintenance Security Settings Security
Configures the access control for the users to configure from keypad Menu.
There are three different options:
Configuration via
Unrestricted. All the options can be accessed in keypad Menu.
Basic settings only. The SIP option under Phone submenu, and
Network, Upgrade, UCM Detect and Factory Reset options under
Keypad Menu
System submenu will not be available in LCD Menu.
Constraint Mode. The phone will require administration password to
change the Network, Upgrade and Factory Reset options under System
submenu, and SIP option under Phone submenu as well.
The default setting is “Unrestricted”.
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
Enable STAR key
locked. The default setting is "Yes".
Keypad Locking
Note:
When the keypad is locked, users would need press and hold the STAR *
key for about 4 seconds again and then enter the password to unlock it.
Password to
Configures the password to lock/unlock the keypad.
Lock/Unlock
SIP TLS Certificate
SSL Certificate used for SIP Transport in TLS/TCP.
SIP TLS Private Key
SSL Private key used for SIP Transport in TLS/TCP.
SIP TLS Private Key
SSL Private key password used for SIP Transport in TLS/TCP.
Password
Web Access Mode
Sets the protocol for web interface. The default setting is “HTTP”.
Disable SSH
Disables SSH access. The default setting is “No”.
Web/Keypad/Restrict
Specifies the time in minutes that the web or LCD login interface will be
mode Lockout
locked out to user after five login failures. This lockout time is used for web
Duration
login, STAR keypad unlock and LCD restrict mode admin login. Range is
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0-60 minutes.
Maintenance Security Settings Trusted CA Certificates
Trusted CA Certificates
Upload CA Certificate file to phone.
Maintenance Packet Capture
Status
Capture Location
With RTP Packets
USB Filename
Displays packet capture status. When user starts to capture trace file, it will
show “RUNNING” status, otherwise, it will show “STOPPED”.
Selects the location where capture file will be stored: internal storage or
USB. The default setting is “Internal Storage”.
Defines whether the packet capture file contains RTP or not. The default
setting is “No”.
Defines the filename of the capture. Only required when capture location is
USB.
Phonebook Page Definitions
Table 10: Phonebook Page Definitions
Phonebook Contacts
Add Contact
Specifies Contact’s First Name, Last Name, Phone Number, Accounts and
Groups to add one new contact in phonebook.
Edit Contact
Edits selected contact.
Delete All Contacts
Deletes all contacts from phonebook.
Phonebook Group Management
Add Group
Specifies Group’s name to add new group.
Edit Group
Edits selected group.
Phonebook Phonebook Management
Enable Phonebook
XML Download
HTTP/HTTPS User
Name
Configures to enable phonebook XML download. Users could select
HTTP/HTTPS/TFTP to download the phonebook file. The default setting is
"Disabled".
The user name for the HTTP/HTTPS server.
HTTP/HTTPS Password
The password for the HTTP/HTTPS server.
Phonebook XML Server
Configures the server path to download the phonebook XML. This field could
Path
be IP address or URL, with up to 256 characters.
Phonebook Download
Interval
Configures the phonebook download interval (in minutes). If it's set to 0, the
automatic download will be disabled. The default value is 0. The valid range is
5 to 720 minutes.
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Remove
Manually-edited
Entries on Download
Sort Phonebook by
Download XML
Phonebook
Upload XML
Phonebook
If set to "Yes", when XML phonebook is downloaded, the entries added
manually will be automatically removed. The default setting is "Yes".
Sort phonebook based on the selection of first name or last name. The default
setting is “Last Name”.
Click on "Download" to download the XML phonebook file to local PC.
Click on "Upload" to upload local XML phonebook file to the phone.
Control the behavior of phonebook key. There are five options: Default, LDAP
Phonebook Key
Search, Local Phonebook, Local Group, and Broadsoft Phonebook. The
Function
default setting is “Default”, when user presses it, phone LCD will show the five
options.
Default search mode
Configures the default phonebook search mode. The default setting is “Qick
match”.
Phonebook LDAP
LDAP Protocol
Configures the LDAP protocol to LDAP or LDAPS. The default setting is
“LDAP”. LDAPS is a feature to support LDAP over TLS.
Server Address
Configures the IP address or DNS name of the LDAP server.
Port
Configures the LDAP server port. The default port number is “389”.
Configures the LDAP search base. This is the location in the directory where
the search is requested to begin.
Base
Example:
dc=grandstream, dc=com
ou=Boston, dc=grandstream, dc=com
User Name
Password
Configures the bind "Username" for querying LDAP servers. Some LDAP
servers allow anonymous binds in which case the setting can be left blank.
Configures the bind "Password" for querying LDAP servers. The field can be
left blank if the LDAP server allows anonymous binds.
Configures the filter used for number lookups.
Examples:
LDAP Number Filter
(|(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.
Configures the filter used for name lookups.
LDAP Name Filter
Examples:
(|(cn=%)(sn=%)) returns all records which has the "cn" or "sn" field starting
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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".
Specify the "name" attributes of each record which are returned in the LDAP
search result. This field allows the users to configure multiple space separated
LDAP Name Attributes
name attributes.
Example:
gn
cn sn description
Specifies the "number" attributes of each record which are returned in the
LDAP search result. This field allows the users to configure multiple space
LDAP Number
separated number attributes.
Attributes
Example:
telephoneNumber
telephoneNumber Mobile
Configures the entry information to be shown on phone's LCD. Up to 3 fields
LDAP Display Name
can be displayed.
Example:
%cn %sn %telephoneNumber
Max. Hits
Search Timeout
Sort Results
LDAP Lookup
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.
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.
Specifies whether the searching result is sorted or not. Default setting is "No".
Configures to enable LDAP number searching when dialing and receiving
calls.
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.
Lookup Display Name
Example:
gn
cn sn description
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Blf Led Patterns
Table 11: BLF LED Patterns
Pattern: Default
Pattern: Analog
Call’s state
Light Indication
Call’s state
Light Indication
Offline
Off
Offline
Off
Idle
Solid Green
Idle
Solid Green
Trying
Solid Red
Trying
Solid Red
Talking
Solid Red
Talking
Solid Red
Proceeding
Flashing Red
Proceeding
Solid Red
Incoming call
Flashing Red
Incoming call
Flashing Red
Pattern: Directional
Mode: Inverse
Call’s state
Light Indication
Call’s state
Light Indication
Offline
Off
Offline
Off
Idle
Solid Green
Idle
Solid Red
Trying
Flashing Green
Trying
Solid Green
Talking
Solid Red
Talking
Solid Green
Flashing Green
Proceeding
Flashing Green
Flashing Red
Incoming call
Flashing Green
Proceeding
(Initiator)
Proceeding
(Receiver)
Incoming call
Flashing Red
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Mode: Reversed (Red)
Call’s state
Offline
Idle
Mode: Reversed (Green)
Light Indication
Off (Extension Board Icon:
Off)
Off (Extension Board Icon:
Idle)
Call’s state
Offline
Idle
Light Indication
Off (Extension Board Icon:
Off)
Off (Extension Board Icon:
Idle)
Trying
Solid Red
Trying
Solid Green
Talking
Solid Red
Talking
Solid Green
Proceeding
Solid Red
Proceeding
Solid Green
Incoming call
Flashing Red
Incoming call
Flashing Green
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.
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Weather Update
To customize GXP1760/GXP1780/GXP1782 to display weather information for the preferred city, users
could go to web GUI->Settings->Web Service page and enter the city code in the following options:
Figure 2: Web Service
By default, the City Code is set to "Automatic", which allows the phone to obtain weather information
based on the IP location detected. To use "Self-Defined City Code" option, please follow the steps below
to obtain the correct city code:
In a web browser, go to www.weather.com;
Enter the city name in the search field. For example, Boston, MA. And click on "SEARCH";
The searching result will show in a new window with URL in the browser's address bar. For example,
http://www.weather.com/weather/today/Boston+MA+USMA0046:1:US
In the above link, USMA0046 is the city code to be filled in "Self-Defined City Code" option.
Users could then further configure the "Update Interval" and "Degree Unit" for weather information
display.
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Editing Contacts and Click-To-Dial
From GXP1760/GXP1780/GXP1782 Web GUI, users could view contacts, edit contacts, or dial out with
Click-to-Dial feature
on the top of the Web GUI. In the following figure, the Contact page shows all
the added contacts (manually or downloaded via XML phonebook). Here users could add new contact, edit
selected contact, or dial the contact/number.
Before using the Click-To-Dial feature, make sure the option "Click-To-Dial Feature" under web
GUI->Settings->Call Features is turned on. By default it's disabled and the dialing icon in web GUI is in
grey
When clicking on the
icon on the top menu of the Web GUI, a new dialing window will show for you
to enter the number. Once Dial is clicked, the phone will go off hook and dial out the number from selected
account. Please see Figure 11 in the following pages for more details.
Additionally, users could directly send the command for the phone to dial out by specifying the following
URL in PC's web browser, or in the field as required in other call modules.
http://ip_address/cgi-bin/api-make_call?phonenumber=1234&account=0&password=admin/123
In the above link, replace the fields with
ip_address:
Phone's IP Address.
phonenumber=1234:
The number for the phone to dial out
account=0:
The account index for the phone to make call. The index is 0 for account 1, 1 for account 2, 2 for
account 3, and etc.
password=admin/123:
The admin login password or user login password of phone's Web GUI.
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Click to dial from
available lines.
Edit contact.
Add contacts. Click to call this
contact from the phone.
Figure 3: Web GUI - Phonebook->Contacts
Figure 4: Click-to-Dial
Wallpaper
GXP1760/GXP1780/GXP1782 support 3 approaches for wallpaper configurations: “Default”, “Download”,
and “UploadDefault Mode
Under Default mode, the phone will display the wallpaper supplied by firmware.
Download Mode
Under Download mode, the phone will download the wallpaper from the specified server path under
“Wallpaper Server Path” option. The Wallpaper Server Path option will take effect only when Download
mode specified. See Figure 5 Download wallpaper from server. The server path must begin with tftp:// or
http:// or https://, otherwise, phone will assume HTTP mode.
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Figure 5: Download Wallpaper from Server
Uploaded Mode
Under Uploaded mode, user can browse and upload a .jpg or .jpeg format wallpaper file. The image must
be smaller 500 KB. See Figure 6 Upload selected wallpaper to phone.
Figure 6: Upload Selected Wallpaper to Phone
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.
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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.
Packet Capture
GXP1760/GXP1780/GXP1782 is embedded with packet capture function on firmware 1.0.0.1 or greater.
The related options are under Maintenance -> Packet Capture.
Figure 7: Packet Capture in Idle
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In the option Capture Location, User can select to store the capture file in phone’s internal storage or
extended USB storage. When the Capture Location is specified as USB, a USB drive must be connected.
When USB is selected as the capture location, user can define the capture file name in USB Filename
field. User can also define whether RTP packets will be captured or not from With RTP Packets option.
Note: Only GXP17xx supports USB connection.
Figure 8: Capture to USB Drive
Figure 9: Capture to Internal Storage
When the capture configuration is set, press Start button to start packet capture. The Status will become
RUNNING while capturing, as showed in Figure 8 and Figure 9. Press Stop button to end capture.
If the capture location is internal storage, user can press Download button to download capture file to local
PC. If capture location is USB, the file will be saved on USB automatically with the defined name. The
capture file is in .pcap format.
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Multicast Paging
GXP1760/GXP1780/GXP1782 supports multicast paging, including sending and listening. On the phone,
users could send multicast page by setting the multicast address and port. Also, users can listen to at most
10 different multicast IP address.
Multicast sender related settings are under Web UI, Settings -> Programmable keys. Select Multicast
paging as the key mode for dial page call. Multicast paging listening related settings are under Web UI
Settings -> Multicast Paging.
For more details on Multicast paging features, please visit http://www.grandstream.com/support to
download the latest “ Multicast Paging User Guide”.
Configuring event list Blf
Grandstream GXP1760/GXP1780/GXP1782 Enterprise IP Phones support both Grandstream UCM Busy
Lamp Filed and Event List BLF features and allows end users, such as attendant, to monitor the call status
of users in the list. GXP1760/GXP1780/GXP1782 supports this feature by sending out the subscription
request to the UCM and changing the indicator status of the Line keys, MPKs, or virtual MPKs that
associated with the monitored users. Additionally, the phone is also able to pick up the calls to the
monitored extensions by using a pre-defined feature code called BLF- Call-pickup Prefix.
For more details on Eventlist BLFconfiguration guide, please refer to:
http://www.grandstream.com/sites/default/files/Resources/GXP21x0_Eventlist_BLF_Guide.pdf
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UPGRADING AND PROVISIONING
The GXP1760/GXP1780/GXP1782 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP
Address for the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for
TFTP or HTTP/HTTPS; the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com
fw.ipvideotalk.com/gs
There are two ways to setup a software upgrade server: The LCD Keypad Menu or the Web Configuration
Interface.
Upgrade Via Keypad Menu
Follow the steps below to configure the upgrade server path via phone's keypad menu:
Press MENU button and navigate using Up/Down arrow to select System;
In the System options, select Upgrade;
Enter the firmware server path and select upgrade method. The server path could be in IP address
format or FQDN format;
Select Start Provision option, and press the "Select" soft key.
A warning window will be prompt for provision confirmation. Press “YES” soft key to start
upgrading/provisioning immediately.
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.
Shortcut of Upgrade and Provision Via Keypad Menu
When GXP phone is in idle state, user could press HOLD key and RIGHT navigation key together to
trigger provision functions. Similarly, phone will pop up reboot banner while idle, if user presses HOLD key
and LEFT navigation key together. After the provision or reboot banner pops up on LCD screen, user could
press YES/NO soft key to confirm/cancel the action.
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Upgrade Via Web GUI
Open a web browser on PC and enter the IP address of the phone. Then, login with the administrator
username and password. Go to Maintenance->Upgrade and Provisioning page, enter the IP address or
the FQDN for the upgrade server in "Firmware Server Path" field and choose to upgrade via TFTP or
HTTP/HTTPS. Update the change by clicking the "Save and Apply" button. Then "Reboot" or power cycle
the phone to update the new firmware.
When upgrading starts, the screen will show upgrading progress. When done you will see the phone
restart again. Please do not interrupt or power cycle the phone when the upgrading process is on.
Firmware upgrading takes around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. We
recommend completing firmware upgrades in a controlled LAN environment whenever possible.
No Local TFTP/HTTP Servers
For users that would like to use remote upgrading without a local TFTP/HTTP server, Grandstream offers a
NAT-friendly HTTP server. This enables users to download the latest software upgrades for their phone via
this server. Please refer to the webpage:
http://www.grandstream.com/support/firmware
Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A
free windows version TFTP server is available for download from :
http://www.solarwinds.com/products/freetools/free_tftp_server.aspx
http://tftpd32.jounin.net/.
Instructions for local firmware upgrade via TFTP:
1. Unzip the firmware files and put all of them in the root directory of the TFTP server;
2. Connect the PC running the TFTP server and the phone to the same LAN segment;
3. Launch the TFTP server and go to the File menu->Configure->Security to change the TFTP server's
default setting from "Receive Only" to "Transmit Only" for the firmware upgrade;
4. Start the TFTP server and configure the TFTP server in the phone’s web configuration interface;
5. Configure the Firmware Server Path to the IP address of the PC;
6. Update the changes and reboot the phone.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use
Microsoft IIS web server.
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Configuration File Download
Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File
(binary or XML) through TFTP or HTTP/HTTPS. The "Config Server Path" is the TFTP or HTTP/HTTPS
server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format.
The "Config Server Path" can be the same or different from the "Firmware Server Path".
A configuration parameter is associated with each particular field in the web configuration page. A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with the "New Password" in the Web GUI->Maintenance->Web Access
page->Admin Password. For a detailed parameter list, please refer to the corresponding firmware release
configuration template.
When the GXP1760/GXP1780/GXP1782 boots up or reboots, it will issue a request to download a
configuration XML file named "cfgxxxxxxxxxxxx.xml" followed by a file named "cfgxxxxxxxxxxxx", where
"xxxxxxxxxxxx" is the MAC address of the phone, i.e., "cfg000b820102ab.xml" and "cfg000b820102ab". If
the download of "cfgxxxxxxxxxxxx.xml" file is not successful, the provision program will download a
generic cfg.xml file. The configuration file name should be in lower case letters.
For more details on XML provisioning, please refer to:
http://www.grandstream.com/sites/default/files/Resources/gs-provisioning_guide_public.pdf
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RESTORE FACTORY DEFAULT SETTINGS
Warning:
Restoring the Factory Default Settings will delete all configuration information on the phone. Please
backup or print all the settings before you restore to the factory default settings. Grandstream is not
responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
Please follow the instructions below to reset the phone:
Press MENU button to bring up the keypad configuration menu;
Select "System" and enter;
Select "Operations - Factory Reset";
A warning window will pop out to make sure a reset is requested and confirmed;
Press the "Yes" soft key to confirm and the phone will reboot. To cancel the Reset, press “No” soft key
instead.
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EXPERIENCING GXP1760/GXP1780/GXP1782
Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware
releases, additional features, FAQs, documentation and news on new products.
We encourage you to browse our product related documentation, FAQs and User and Developer Forum
for answers to your general questions. If you have purchased our products through a Grandstream
Certified Partner or Reseller, please contact them directly for immediate support.
Our technical support staff is trained and ready to answer all of your questions. Contact a technical support
member or submit a trouble ticket online to receive in-depth support.
Thank you again for purchasing Grandstream IP phone, it will be sure to bring convenience and color to
both your business and personal life.
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