Grandstream Networks, Inc.
GXV3370
IP Multimedia Phone for Android
TM
Administration Guide
GXV3370 Administration Guide
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COPYRIGHT
©2022 Grandstream Networks, Inc. https://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:
https://www.grandstream.com/support
Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc.
in the United States, Europe, and other countries.
CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this
product in any way other than as detailed by this guide, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with devices as it may cause damage to the products and void
the manufacturer warranty.
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FCC Statement
This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant
to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful
interference in a residential installation. This equipment generates, uses, and can radiate radio frequency
energy and, if not installed and used in accordance with the instructions, may cause harmful interference
to radio communications. However, there is no guarantee that interference will not occur in a particular
installation.
If this equipment does cause harmful interference to radio or television reception, which can be determined
by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more
of the following measures:
Reorient or relocate the receiving antenna.
Increase the separation between the equipment and receiver.
Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
Consult the dealer or an experienced radio/TV technician for help.
The device complies with FCC radiation exposure limits set forth an uncontrolled environment and it also
complies with Part 15 of the FCC RF Rules. This equipment must be installed and operated in accordance
with provided instructions and the antenna(s) used for this transmitter must be installed to provide a
separation distance of at least 20 cm from all persons and must not be co-located or operating in conjunction
with any other antenna or transmitter. End-users and installers must be provided with antenna installation
instructions and consider removing the no-collocation statement.
This device complies with Part 15 of the FCC Rules. Operation is subject to the following two conditions:
(1) this device may not cause harmful interference, and (2) this device must accept any interference
received, including interference that may cause undesired operation.
Caution
Any changes or modification not expressly approved by the party responsible for compliance could void the
user’s authority to operate the equipment.
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CE DECLARATION OF CONFORMITY
This transmitter complies with the essential requirements and provisions of directives 2014/53/EU,
2014/30/EU, 2015/35/EU, and subsequent amendments, according to standards
ETSI EN 300 328 V2.1.1 (2016-11); EN 301 893 V2.1.1 (2017-05)
ETSI EN 301 489-1 V2.1.1 (2017-02); ETSI EN 301 489-17 V3.1.1 (2017-02)
EN 60950-1:2006+A11:2009+A1:2010+A12:2011+A2:2013
EN 62311: 2008; EN62479: 2010
Manufacturer:
Grandstream Networks, Inc.
126 Brookline Ave, 3
rd
Floor Boston, MA 02215, USA
EUT Feature
Tx/Rx Frequency
Range
2402~2480 MHz
2412~2472 MHz
5150~5250 MHz
5250~5350 MHz
5470~5725 MHz
Number of Channels
79
13
UNII Band I:
802.11a/n-HT20-VHT20: 4 channels
802.11n-HT40: 2 channels
UNII Band II:
802.11a/n-HT20: 4 channels
802.11n-HT40: 2 channels
UNII Band III:
802.11a/n-HT20: 11 channels
802.11n-HT40: 5 channels
Carrier Frequency of
Each Channel
f=2402+k MHz
(k=0,1,2….,78)
-
-
Antenna Type/Gain
Internal PCB Antenna /
gain 3 dBi
Internal PCB Antenna /
gain 3 dBi
Internal PCB Antenna /
gain 4 dBi
Type of Modulation
Bluetooth BR 1Mbps: GFSK
Bluetooth EDR 2Mbps: π/4-
DQPSK
Bluetooth EDR 3Mbps:
8DPSK
802.11b: DSSS (DBPSK /
DQPSK / CCK)
802.11g/n: OFDM (BPSK /
QPSK / 16QAM / 64 QAM)
802.11a/n: OFDM (BPSK / QPSK /
16QAM / 64QAM)
Operation
temperature
0 °C ~ +40 °C
Storage temperature
-10 °C ~ +60 °C
Humidity
10 ~ 90% non-condensing
Domestic use
Industrial use Class B
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Caution: Exposure to Radio Frequency Radiation
This equipment complies with EU radiation exposure limits set forth for an uncontrolled environment. This
equipment should be installed and operated with minimum distance of 20 cm between the radiator and your
body.
CE Authentication
Hereby, Grandstream Networks, Inc. declares that the radio equipment GXV3370 is in compliance with
Directive 2014/53/EU.
The full text of the EU declaration of conformity is available at the following internet address:
https://www.grandstream.com/support/resources/
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GNU GPL INFORMATION
GXV3370 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:
https://www.grandstream.com/sites/default/files/Resources/gxv3370_gpl.zip
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Table of Contents
DOCUMENT PURPOSE ............................................................................................... 12
CHANGE LOG .............................................................................................................. 13
Firmware Version 1.0.3.36 ...................................................................................................................... 13
Firmware Version 1.0.3.30 ...................................................................................................................... 13
Firmware Version 1.0.3.28 ...................................................................................................................... 13
Firmware Version 1.0.3.27 ...................................................................................................................... 13
Firmware Version 1.0.3.25 ...................................................................................................................... 13
Firmware Version 1.0.3.19 ...................................................................................................................... 14
Firmware Version 1.0.3.9......................................................................................................................... 14
Firmware Version 1.0.3.8......................................................................................................................... 14
Firmware Version 1.0.3.3......................................................................................................................... 14
Firmware Version 1.0.1.55 ...................................................................................................................... 15
Firmware Version 1.0.1.54 ...................................................................................................................... 15
Firmware Version 1.0.1.43 ...................................................................................................................... 15
Firmware Version 1.0.1.33 ...................................................................................................................... 15
Firmware Version 1.0.1.28 ...................................................................................................................... 16
Firmware Version 1.0.1.12 ...................................................................................................................... 16
WELCOME ................................................................................................................... 17
PRODUCT OVERVIEW ................................................................................................ 18
Feature Highlights ................................................................................................................................... 18
GXV3370 Technical Specifications .......................................................................................................... 19
GETTING STARTED ..................................................................................................... 21
Equipment Packaging.............................................................................................................................. 21
GXV3370 Phone Setup ........................................................................................................................... 22
Using the Phone Stand ........................................................................................................................ 22
Using the Slots for Wall Mounting ........................................................................................................ 22
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Connecting the GXV3370 ........................................................................................................................ 23
Cleaning the Phone ................................................................................................................................. 23
GXV3370 LCD SETTINGS ........................................................................................... 24
Access LCD Settings ............................................................................................................................... 25
Features .................................................................................................................................................. 25
Network ................................................................................................................................................... 27
Ethernet Settings ................................................................................................................................. 27
Wi-Fi ..................................................................................................................................................... 28
VPN ...................................................................................................................................................... 28
General Network Settings .................................................................................................................... 29
Proxy Settings ...................................................................................................................................... 29
Tethering & Portable Hotspot ............................................................................................................... 29
Basic ........................................................................................................................................................ 30
Sound ................................................................................................................................................... 30
Display ................................................................................................................................................. 31
Language & keyboard .......................................................................................................................... 31
Date & Time ......................................................................................................................................... 32
System ..................................................................................................................................................... 32
Security Settings .................................................................................................................................. 32
Peripherals ........................................................................................................................................... 33
Accounts .............................................................................................................................................. 33
Power Information ................................................................................................................................ 33
Reboot the Phone ................................................................................................................................ 34
Apps......................................................................................................................................................... 34
Application Management ..................................................................................................................... 34
Default Application ............................................................................................................................... 36
Notification Center ............................................................................................................................... 36
Advanced ................................................................................................................................................. 36
Account Settings .................................................................................................................................. 36
System Update .................................................................................................................................... 37
Syslog .................................................................................................................................................. 37
System Security ................................................................................................................................... 37
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Status....................................................................................................................................................... 38
Account Status ..................................................................................................................................... 38
Network Status ..................................................................................................................................... 38
System Info .......................................................................................................................................... 38
Storage Status ..................................................................................................................................... 38
GXV3370 WEB GUI SETTINGS ................................................................................... 39
Status Page Definitions ........................................................................................................................... 39
Status/Account Status ......................................................................................................................... 39
Status/Network Status ......................................................................................................................... 39
Status/System Info ............................................................................................................................... 40
Account Page Definitions ........................................................................................................................ 40
Account/General Settings .................................................................................................................... 40
Account/SIP Settings ........................................................................................................................... 43
Account/Codec Settings ...................................................................................................................... 46
Account/RTP Settings .......................................................................................................................... 50
Account/Call Settings ........................................................................................................................... 51
Account/Advanced Settings ................................................................................................................. 57
Account/Special Features .................................................................................................................... 58
Phone Settings Page Definitions ............................................................................................................. 59
Phone Settings/General Settings ......................................................................................................... 59
Phone Settings/Call Settings ............................................................................................................... 61
Phone Settings/Ring Tone ................................................................................................................... 65
Phone Settings/Video Settings ............................................................................................................ 66
Phone Settings/Multicast Paging ......................................................................................................... 66
Network Settings Page Definitions .......................................................................................................... 67
Network Settings/Ethernet Settings ..................................................................................................... 67
Network Settings/Wi-Fi Settings .......................................................................................................... 70
Network Settings/OpenVPN® Settings ................................................................................................ 70
Network Settings/Advanced Network Settings .................................................................................... 71
Network Settings/Affinity Settings ........................................................................................................ 73
System Settings Page Definitions ........................................................................................................... 73
System Settings/Time and Language .................................................................................................. 73
System Settings/Security Settings ....................................................................................................... 74
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System Settings/Preferences .............................................................................................................. 76
System Settings/TR-069 ...................................................................................................................... 78
Maintenance Page Definitions ................................................................................................................. 80
Maintenance/Upgrade ......................................................................................................................... 80
Maintenance/System Diagnosis .......................................................................................................... 85
Maintenance/Event Notification ........................................................................................................... 87
Application Page Definitions .................................................................................................................... 88
Applications/Programmable key .......................................................................................................... 88
System Application/Contacts ............................................................................................................... 90
System Application/LDAP Phonebook ................................................................................................. 93
System Application/Recording ............................................................................................................. 95
Value-added Service Page Definitions .................................................................................................... 95
Value-added Service/Value-added Service (0/10) ............................................................................... 95
Value-added Service/General Settings ................................................................................................ 96
Value-added Service/Broadsoft Settings ............................................................................................. 96
Value-added Service/Broadsoft Directories ......................................................................................... 98
Value-added Service/BroadSoft IM&P ................................................................................................. 99
UPGRADING AND PROVISIONING .......................................................................... 100
Upgrade and Provisioning Configuration............................................................................................... 100
Upload Firmware Locally ....................................................................................................................... 102
Upgrade via SD Card ............................................................................................................................ 102
No Local Firmware Servers ................................................................................................................... 103
Provisioning and Configuration File Download ..................................................................................... 103
FACTORY RESET ...................................................................................................... 105
Restore to Factory Default via LCD Menu............................................................................................. 105
Restore to Factory Default via the Web GUI ......................................................................................... 105
Restore to Factory Default via Hard Keys ............................................................................................. 106
SAFE MODE ............................................................................................................... 107
SDK FRAMEWORK SERVICE ................................................................................... 108
EXPERIENCING THE GXV3370 APPLICATION PHONE .......................................... 109
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Table of Tables
Table 1: GXV3370 Features in a Glance .................................................................................................... 18
Table 2: GXV3370 Technical Specifications ................................................................................................ 19
Table 3: Equipment Packaging ................................................................................................................... 21
Table 4: GXV3370 LCD Settings ................................................................................................................. 24
Table of Figures
Figure 1: GXV3370 Package Content ......................................................................................................... 21
Figure 2: Built in Stand and Mounting Slots on The GXV3370. .................................................................. 22
Figure 3: Tab on the Handset Cradle .......................................................................................................... 22
Figure 4: GXV3370 Back / Side View ......................................................................................................... 23
Figure 5: GXV3370 System Settings .......................................................................................................... 25
Figure 6: GXV3370 Wi-Fi Hotspot .............................................................................................................. 30
Figure 7: GXV3370 Upgrade – Detect New Version – Start Provision ..................................................... 100
Figure 8: GXV3370 Upgrade Configuration via LCD ................................................................................ 101
Figure 9: GXV3370 Upgrade Configuration via Web GUI ........................................................................ 101
Figure 10: Upload Firmware File to Update .............................................................................................. 102
Figure 11: Config File Upgrade Verification .............................................................................................. 104
Figure 12: GXV3370 LCD - Confirm Factory Reset .................................................................................. 105
Figure 13: GXV3370 Web GUI - Factory Reset ........................................................................................ 105
Figure 14: GXV3370 Web GUI - Confirm Factory Reset .......................................................................... 106
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DOCUMENT PURPOSE
This document describes how to configure the GXV3370 via phone's LCD menu and web UI menu to fully
manipulate phone's features. The intended audiences of this document are VoIP administrators. To learn
the basic functions of GXV3370, please visit https://www.grandstream.com/support to download the latest
“GXV3370 User Guide”.
This guide covers following topics:
Product Overview
Getting Started
GXV3370 LCD Settings
GXV3370 Web GUI Settings
Upgrading and provisioning
Restore factory default settings
Safe Mode
SDK Framework Service
Experiencing the GXV3370 applications
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CHANGE LOG
This section documents significant changes from previous versions of administration guide for GXV3370.
Only major new features or major document updates are listed here. Minor updates for corrections or editing
are not documented here.
Firmware Version 1.0.3.36
Added P22038 to restrict Wi-Fi settings on web UI. [Network Settings/Wi-Fi Settings]
Supported GS Affinity. [Network Settings/Affinity Settings]
Firmware Version 1.0.3.30
Added “Maximum Number of SIP Request Retries” option when DNS mode is set to SRV.
[Account/General Settings]
Added “Failback Expiration” settings when DNS SRV Failover Mode is Failback. [Account/General
Settings]
Added maximum transmission unit (MTU) settings. [Network Settings/Advanced Network Settings]
Added "Screen Timeout" and "Screensaver Timeout" settings. [System Settings/Preferences]
Added the option to upload GUI customization file from web UI. [Maintenance/Upgrade]
Added Separate Ethernet and Wi-Fi traffic setting. [Network Settings/Advanced Network Settings]
Firmware Version 1.0.3.28
Added 'Keep Associated Account' option under Contacts page. [Keep Associated Account][Keep
Associated Account]
Firmware Version 1.0.3.27
Improved Special Features settings layout on web UI. [Special Features]
Added a “Mute” option to account ring tone list. [Account Ring Tone]
Supported "Do not prompt Missed calls" for Ring Group missed call. [Call Log]
Firmware Version 1.0.3.25
Added support for $variable in provisioning link (such as $MAC or $PN). [Firmware Server Path][Config
Server Path]
Added “Start provision” button to trigger the device to fetch configuration file from server on both web
UI and LCD. [Start Provision]
Differentiated rejected call logs from missed calls. [Rejected Call Notification]
Added Lagos, Nigeria GMT+1:00 to time zone list. [Time Zone]
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Added file format verification when uploading OpenVPN® certificate files. [OpenVPN® CA][OpenVPN®
Client Certificate][OpenVPN® Client Key]
Added file format verification when uploading configuration files. [Upload Device Configuration]
Added Call Function Button Display Timeout (s). [Call Function Button Display Timeout (s)]
Added Self-defined time zone under web UI Time Settings. [Self-defined Time Zone]
Supported Handsfree TX Gain settings. [Handsfree TX Gain]
Added more Handset TX Gain (dB) options. [Handset TX Gain ]
Supported voice monitoring VQ RTCP-XR session report. [VQ RTCP-XR Collector Name][VQ RTCP-
XR Collector Address][VQ RTCP-XR Collector Port]
Supported web UI control list. [Web Access Control][Web Access Control List]
Supported LCD to be turned on when BLF/SCA status updated. [Enable LCD Turn on Automatically
when BLF/SCA status changes]
Supported send URL via programmable key. [Send URL]
Firmware Version 1.0.3.19
Implemented Slovak language to system. [GXV3370 Technical Specifications]
Added reset button for each account to restore to default values.
Firmware Version 1.0.3.9
Support Enable/Disable Call Waiting from each account. [Enable Call Waiting]
Support Enable/Disable preview in Value-added Service. [Enable Preview]
Enabled Offhook Auto Dial to be configured with special characters as ‘#’ or ‘* [Offhook Auto Dial]
Firmware Version 1.0.3.8
Added Berlin, Dublin to Time Zone list. [Time Zone]
Supported letters in dial plan, such as {\p\a\r\k\*x+}. [Dial Plan]
Added display condition for DTMF button display during a call. [Display Condition]
Supported each account can disable video call separately. [Enable Video Call]
Added door button control in the incoming call interface. [Display Open Door Button]
Supported volume adjustment on web UI. [Media volume]
Supported callee ID display feature. [Callee ID Display]
Firmware Version 1.0.3.3
Added support for guest login feature. [Phone Settings/General Settings]
Added support for code selection for DND and rejecting calls. [Phone Settings/Call Settings]
Added support for Group Listen with Speaker feature. [Phone Settings/Call Settings]
Added support for Different Networks for Data and VoIP Calls configuration. [Network Settings/Ethernet
Settings]
Supported random upgrade check. [Maintenance/Upgrade]
Supported video codec H.263. [Account/Codec Settings]
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Firmware Version 1.0.1.55
No major changes.
Firmware Version 1.0.1.54
Added Support for configuring 2 Access Passwords to control the same GDS System. [Value-added
Service Page Definitions]
Added support for Zoom Mode feature. [Zoom mode]
Added support for Wi-Fi band selection from web UI. [Wi-Fi]
Added support for choosing “#” button as dial or redial function separately. [Use # as Dial Key]
Added support for Broadsoft IM&P service. [Value-added Service/BroadSoft IM&P]
Added support for MAC header in SIP REGISTER. [Use MAC Header]
Added support for direct or quick IP call, and paging call mode selection in Dialpad. [Use Direct IP Call
Mode] [Use Paging Call Mode] [Use Quick IP-Call mode]
Added support for apply all available config files. [Download and Process All Available Config Files]
Supported EEE (Energy-Efficient Ethernet) mode. [Enable EEE Mode]
Supported HDMI audio enable/disable switch. [Sound]
Firmware Version 1.0.1.43
Added option to disable recording during call. [Record Mode]
Added support for Czech language [GXV3370 Technical Specifications]
Added support to boot up in safe mode and option to boot the phone in safe mode from web GUI. [Safe
Mode]
Added RTP timeout support. [RTP Timeout Timer]
Added mute support for paging/intercom. [Mute on Answer Intercom Call]
Added SIP NOTIFY support to sync phone book [Allow Sync Phonebook via SIP Notify]
Added support for second dial tone. [Call Progress Tones] [Dial Plan]
Added option to support SUBSCRIBE expiration. [Subscribe Expiration]
Added support for Baudisch Door System feature. [Door System Type]
Added support for disabling handset. [Handset Option]
Added ringtone selection support for door system settings. [Value-added Service Page Definitions]
Added support to transfer a call via MPK BLF. [Enable transfer via non-Transfer Programmable Key]
Removed codec Jitter Buffer Type settings from web UI account settings since GXV3370 is using a
fixed value.
Removed Hide Local Call History from web UI phone settings due to local Contacts app would not
include Broadsoft contacts.
Firmware Version 1.0.1.33
Supported TLS version 1.2. [GXV3370 Technical Specifications]
Updated to SDK package version 1.1.5. [SDK FRAMEWORK SERVICE]
Hide LCD/DDR/Factory Serial Number in the web UI under System Info. [System Info]
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Added more options in web UI Contacts Automatic Download Interval. [Automatic Download Interval]
Supported choosing call function buttons for single call or conference. [Phone Settings/Call Settings].
Firmware Version 1.0.1.28
Added different service type to web UI Value-added Service. [Value-added Service]
Supported SDK. [SDK FRAMEWORK SERVICE]
Added LDAP page configuration. [LDAP Phonebook]
Added Wi-Fi country code configuration. [Country Code]
Removed LDAP page temporarily. It will be added later.
Added a new option “Basic settings & Network setting” for Configuration via Keypad Menu.
[Configuration via Keypad Menu]
Removed “Prefer Non-Interleaved Mode” in account code settings.
Added “Speed conference” in MPK. [Applications/Programmable key]
Added LCD Settings Network VPN, to add or edit VPN profile. [**Note: Since firmware version
1.0.3.36 this setting can be restricted from WEB UI as well as from LCD screen. This can only be configured
via config template. When P22038=0, the Wi-Fi settings are neither accessible from web UI nor from LCD.
To enable Wi-Fi on web UI and LCD screen, use P22038=1.
VPN]
Firmware Version 1.0.1.12
This is the initial version.
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WELCOME
Thank you for purchasing Grandstream GXV3370 IP Multimedia Phone for Android
TM
. The GXV3370 IP
Video Phone for Android combines a 16-line IP video phone with a multi-platform video conferencing
solution and the functionality of an Android tablet to offer an all-in-one communications solution. The phone
features a 7” 1024x600 capacitive touch screen TFT LCD, Mega pixel camera, dual Gigabit ports with
PoE/PoE+, HD audio and video, integrated Wi-Fi (802.11a/b/g/n) & Bluetooth, rich peripheral interfaces,
and Android 7.0. By combining a state-of-the-art IP video phone, an advanced video conferencing solution,
and the functionality of a tablet, businesses throughout the world can now use the GXV3370 for all
communication and productivity needs.
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PRODUCT OVERVIEW
Feature Highlights
The following table contains the major features of the GXV3370:
Table 1: GXV3370 Features in a Glance
GXV3370
16 lines with up to 16 SIP, up to 7-way audio conference and 3-way
720p 30fps HD video conference, phonebook with up to 1000
contacts, call history with up to 1000 records
Dual switched 10/100/1000Mbps network ports, Dual band 2.4 &
5GHz Wi-Fi (802.11a/b/g/n), PoE/PoE+, Bluetooth 4.0+EDR, USB,
SD, HDMI, EHS with Plantronics headsets support
7” (1024x600) capacitive (5 points) touch screen TFT LCD, mega
pixel CMOS sensor camera with privacy shutter
HD wideband audio, full-duplex Hands-free speakerphone with HD
acoustic chamber, advanced acoustic echo cancellation and excellent
double-talk performance
Runs the Android™ Operating System 7.0.
Built-in support for GXV series of video surveillance cameras, and
GDS370x series of access control devices.
Create Android applications for any business need to run on
GXV3370 using Google’s API and Grandstreams SDK toolkit.
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GXV3370 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 the phone GXV3370.
Table 2: GXV3370 Technical Specifications
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A
record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP-
MED, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6, OpenVPN
®
.
Network Interfaces
Dual switched 10/ 100/ 1000 Mbps ports with integrated PoE/PoE+
Graphic Display
7” 1024x600 capacitive touch screen (5 points) TFT LCD
Camera
Tiltable mega-pixel CMOS camera with privacy shutter, 720P@30fps
Bluetooth
Yes, integrated. Bluetooth 4.0 + EDR.
Wi-Fi
Yes, dual-band 2.4 & 5GHz with 802.11 a/b/g/n
Auxiliary Ports
RJ9 headset jack (allowing EHS with Plantronics headsets), 3.5mm stereo headset
with microphone, USB port, SD, HDMI-out (1.4 up to 720p30fps)
Feature Keys
2 function touch keys VOLUME +/-, 3 dedicated Android touch keys HOME, MENU,
and BACK
Voice Codec
G.711µ/a, G.722 (wide-band), G.726-32, iLBC, Opus, G.729A/B in-band and out-
of-band DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC,
ANS
Video Codec and
Capabilities
H.263,H.264 BP/MP/HP, video resolution up to 720p, frame rate up to 30 fps, bit
rate up to 2Mbps, 3-way video conference (720p@30fps), anti-flickering, auto
focus and auto exposure
Telephony Features
Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup, 7-way
audio conference(including the host), shared-call-appearance (SCA) / bridged-
line-appearance (BLA), virtual MPK, downloadable contacts (XML, LDAP, up to
1000 items), call record, call log (up to 1000 records), call waiting, auto answer,
XML customization of screen, click-to-dial, flexible dial plan, hot desking,
personalized music ringtones and music on hold, server redundancy & fail-over
Sample Applications
Local apps: Contacts, Call History, File Manager, MPK, Settings, Browser,
Voicemail, Clock, Recorder, SMS, Backup and Restore, etc.
Support 3
rd
party apps: Skype, Google Hangouts, Skype for Business, etc.
API/SDK available for advanced custom application development.
Applications
Deployment
Allow Android 7.0 compliant applications to be developed, downloaded, and run in
the embedded device with provisioning control
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HD Audio
Yes, HD handset and speakerphone with support for wideband audio
Base Stand
Yes, integrated stand with multiple adjustable angles. Wall mountable
QoS
Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level passwords, MD5 and MD5-sess based authentication,
256-bit AES encrypted configuration file, TLS, SRTP, HTTPS, 802.1x media access
control
Multi-language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
Chinese, Korean, Japanese, Czech, Slovak and more
Upgrade/ Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS or local HTTP upload, mass
provisioning using TR-069 or AES encrypted XML configuration file
Power and Green
Energy Efficiency
Universal power adapter included: Input: 100-240VAC 50-60Hz; Output 12VDC
1.5A
Integrated PoE* 802.3af Class 3, PoE+ 802.3at, Class 4
*When using PoE to power up phone, the USB port will not work.
Physical
Dimension : 252mm (W) x 211mm (L) x 84mm (H)
Unit weight: 1.08kg
Package weight: 1.77kg
Temperature and
Humidity
Operation: 0
o
C to 40
o
C
Storage: -10
o
C to 60
o
C
Humidity: 10% to 90% Non-condensing
Package Content
GXV3370 phone, handset with cord, base stand, universal power supply, network
cable, screen cleaning cloth, quick installation guide, brochure, GPL license
Compliance
FCC: Part 15 (CFR 47) Class B; UL 60950 (power adapter); Part68 (HAC)
CE : EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1,
EN62479, RoHS
RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 4268
IC: ICES 003, RSS 247, CS 03, RSS 102
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GETTING STARTED
This chapter provides basic installation instructions including the list of the packaging contents and also
information for obtaining the best performance with the GXV3370.
Equipment Packaging
Table 3: Equipment Packaging
Figure 1: GXV3370 Package Content
Note: Check the package before installation. If you find anything missing, contact your system administrator.
GXV3370
1 x GXV3370 Main Case.
1 x Handset.
1 x Phone Cord.
1 x Ethernet Cable.
1 x 12V Power Adapter.
1 x Wall Mount.
1 x Screen Cleaning Cloth.
1 x Quick Installation Guide.
1 x GPL License.
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GXV3370 Phone Setup
The GXV3370 can be installed on the desktop using the built-in stand or attached on the wall using the
slots for wall mounting.
Figure 2: Built in Stand and Mounting Slots on The GXV3370.
Using the Phone Stand
The GXV3370 has a built-in phone stand. To use it, pull out the phone stand handle on the back of the
phone. Adjust the angle as preferred and make sure the phone stands still on the desktop
Using the Slots for Wall Mounting
1. Attach the wall mount to the slots on the back of the phone;
2. Attach the phone to the wall via the wall mount hole;
3. Pull out the tab from the handset cradle (see figure below);
4. Rotate the tab and plug it back into the slot with the extension up to hold the handset while the
phone is mounted on the wall.
Figure 3: Tab on the Handset Cradle
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Connecting the GXV3370
To setup your GXV3370, please follow the steps below:
1. Connect the handset and main phone case with the phone cord;
2. Connect the LAN port of the phone to the RJ-45 socket of a hub/switch or a router (LAN side of the
router) using the Ethernet cable;
3. Connect the 12V DC output plug to the power jack on the phone; plug the power adapter into an
electrical outlet. If PoE switch is used in step 2, this step could be skipped;
4. The LCD will display booting up or firmware upgrading information. Before continuing, please wait
for the main screen display to show up;
5. Using the web configuration interface or from the menu of the touch screen, you can further
configure network connection using static IP, DHCP etc.
Figure 4: GXV3370 Back / Side View
Cleaning the Phone
For daily dust removal and fingerprint removal, please use the screen cleaning cloth in the factory package
to wipe the phone. For some special cases like medical environment, you can use medical alcohol or
isopropanol. The steps are as followed:
1. Before cleaning the phone, stop using it and disconnect it from the power supply.
2. Spray a small amount of disinfectant on screen, camera, handle, and other places that are easily
touched by users.
3. Wipe the phone with screen cleaning cloth.
4. Power on until the disinfectant is completely volatilized.
Notes:
Keep the power plug clean and dry or may lead to electric shock or other perils.
DO NOT use disinfectant too frequently.
DO NOT use high degree or even pure disinfectant. It could damage the phone.
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GXV3370 LCD SETTINGS
The GXV3370 LCD MENU provides an easy access to the settings on the phone. Some of the settings
from Web GUI could be configured via the LCD as well. The following table shows the LCD setting menu
options.
Table 4: GXV3370 LCD Settings
Features
Call Forwarding
Auto-Answer
Harassment interception
Account Ringtones
Shared Call Appearance (SCA)
Bluetooth
Network
Ethernet settings
Wi-Fi
General Network Settings
Proxy Settings
Tethering & Portable Hotspot
Basic
Voice
Display
Language & Keyboard
Date & Time
System
Security Settings
Peripherals
Accounts
Power Information
Reboot the Phone
Apps
Application Management
Default Application
Notification Center
Advanced
Account Settings
System Update
Syslog
System Security
About
Account Status
Network Status
System Info
Storage Status
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Access LCD Settings
To open the settings menu, you should:
Tap on Settings app on the screen. Or;
Swipe down from the top of the home screen to open the notifications panel and hit the
Settings icon in the top right corner.
Figure 5: GXV3370 System Settings
Features
In this menu, users can configure different features related to each account of the active accounts:
Call Forwarding
The incoming call to this SIP account can be forwarded to another account using different rules as
configured here.
- Disabled: Call forwarding feature is disabled. This is the default setting.
- Unconditional: Forward all calls to a number.
- Time based: Set the time range and number to be forwarded the calls to. In this time range, calls
are forwarded to the number specified in In Time Forward To; out of this time range, calls are
forwarded to the number specified in Out Time Forward To.
- Others: Configure Call forward when the phone is Busy or on DND or based on No Answer Out.
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Auto-Answer
- IF Enabled and set to “Always”, the phone will automatically turn on the speaker phone to answer
all incoming calls.
- If enabled and set to “Enable Intercom/Paging”, the phone will answer the call based on the SIP
info header sent from the server/proxy.
- By default, it is turned off.
Harassment interception
- Blacklist: This menu allows configuring a blacklist of number that will be blocked from calling the
phone, users can either enter the numbers to block manually, from contacts or from call history
- Intercept anonymous calls: when enabled the phone rejects all the anonymous calls, users can
choose on which account this setting is to be applied
Account Ringtones
Select a ringtone for the incoming call to the SIP account chosen. The system ringtone is set by default.
Shared Call Appearance (SCA)
- Shared Call Appearance (SCA): Enable or disable SCA on the account.
- Enable Barge-in: If set to Yes, the user could barge into an active call on a shared line.
- Auto Fill Call Park Code: If set to Yes, the configured Call Park Service Code will be
automatically filled in on the phones dial pad when picking up the parked call. This is used when
Special Mode is set to BroadSoft (from web UI or provisioning) and Enable SCA is set to Yes.
- Call Park Service Code: Configure the retrieving feature code for call parking. If Auto Fill Call
Park Code is set to Yes, this call park service code will be automatically filled in on the phones
dial pad when picking up the parked call. This is used when Special Mode is set to BroadSoft
(from web UI or provisioning) and Enable SCA is set to Yes.
- Seize Line Timeout (s): Configure the time for the line can be seized (in seconds) when using
shared line. The default setting is 15 seconds. For Shared Call Appearance, phone will send a
SUBSCRIBE-request for the line-seize event package whenever a user attempts to take the shared
line off hook. “Line Seize Timeout” is the line-seize event expiration timer.
Bluetooth
- Bluetooth: Tap on Bluetooth to turn on/off Bluetooth connection. By default, it is turned off.
- Enable handsfree mode: Tap on “Enable handsfree mode” to activate it.
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- Show received files: Shows the Transfer history of Bluetooth files
- Additional Settings: This menu is available only when the Bluetooth is enabled:
Device Name. Tap to change the name of the GXV3370, which is displayed on other
Bluetooth devices when discovered. By default, it is GXV3370_XXXXXXWhere XXXXXX
are the last 6 digits of the phone’s MAC address.
Visibility timeout. Tap to select the timeout interval among “2 minutes”, “5 minutes”, “1 hour”
or “never”. By default, the visibility timeout is 2 minutes.
Visibility to nearby Bluetooth devices. Sets the visibility of the phone to other Bluetooth
devices. Normally this option is enabled during pairing process so that other Bluetooth devices
can discover the GXV3370.
- Available devices: This section will show the available devices for pairing. Tap on to initiate
scan process on the GXV3370 to discover the Bluetooth devices within the range.
Network
Users can configure Ethernet settings, Wi-Fi, VPN, PPPoE and other advanced network settings here.
Ethernet Settings
Preferred Internet Protocol: Selects which Internet protocol to use. When both Ipv4 and Ipv6 are
enabled, phone attempts to use preferred protocol first and switches to the other choice if it fails.
Ipv4 Settings: Here user can configure the IPv4 address Type. If DHCP is selected, the phone will
get an IP address automatically from the DHCP server in the network. This is the default mode. If Static
IP is selected, manually enter the information for IP Address, Subnet Mask, Default Gateway, DNS
Server, and Alternative DNS server. If PPPoE is selected, type PPPoE Account ID and PPPoE
Password provided from the PPPoE server to get authenticated for network access.
Ipv6 Settings: Here user can configure the Ipv6 address Type. If DHCP is selected, the phone will get
an IP address automatically from the DHCP server in the network. This is the default mode. If Static IP
is selected, manually enter the information for IP Address, Prefix Length, DNS Server, and Alternative
DNS server.
802.1x mode: This option allows the user to enable/disable 802.1x mode on the phone. The default
setting is disabled. To enable 802.1x mode, select the 802.1x mode and enter the required configuration
depending on the 802.1x mode chosen. The available modes are EAP-MD5, EAP-TLS and EAP-PEAP
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Wi-Fi
Tap on “Wi-Fi” to turn on/off Wi-Fi connection. By default, it is turned off.
Tap on “Wi-Fi Band” to set the type of Wi-Fi Band (“2.4G”, “5G” or “2.4G&5G”). The default setting is
2.4G&5G.
Tap on Wi-Fi Settingsto set up and manage wireless access points. This option is available only
when Wi-Fi is turned on.
- Add Network. If the Wi-Fi network SSID does not show up in the list, or users would like to set up
advanced options for the Wi-Fi network, roll to the end of the Wi-Fi list and select “Add Network”.
Then Enter SSID, Security type, password and set up address type (DHCP/Static IP/PPPoE) in the
prompt dialog. The phone will reboot with Wi-Fi network connected.
- Refresh. Press MENU button and select “Refresh” to initiate scan for the Wi-Fi network within
the range.
- Advanced Settings. Press MENU button and select “Advanced”.
o Install Certificates. This is to install certificates (previously download/uploaded to the
phone) when connecting to Wi-Fi network requesting SSL certificate.
o Wi-Fi Direct. This is to set up peer-to-peer connection between two Wi-Fi Direct devices
so that they can share data and sync files. Press MENU button and select Advanced”.
Then tap on “Wi-Fi Direct”, the Available devices will be displayed.
- Wi-Fi Configuration
o Saved Networks: this is to show all the Wi-Fi Networks that are registered on the phone.
o Network notification: If enabled, the phone will show notification on the top status bar
indicating an open network is available. By default, it is enabled.
o MAC address: This shows the MAC address of the Wi-Fi.
o IP address: This shows the IP address of the phone from Wi-Fi network.
**Note: Since firmware version 1.0.3.36 this setting can be restricted from WEB UI as well as from LCD
screen. This can only be configured via config template. When P22038=0, the Wi-Fi settings are neither
accessible from web UI nor from LCD. To enable Wi-Fi on web UI and LCD screen, use P22038=1.
VPN
Enable / Disable OpenVPN®
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General Network Settings
LLDP
Turn on/off LLDP on the GXV3370. If turned on, the phone will be able to discover the LAN polices as
set up in the switch side to obtain network settings such as VLAN tag, Layer 2 QoS 802.1p priority and
Layer 3 QoS in a plug-and-play manner.
LLDP TX Interval
Specifies the time interval, in seconds, between successive LLDP-MED transmission cycles
Layer 3 SIP QoS
This field defines the layer 3 QoS parameter for SIP packets.
This is the value used for IP Precedence, Diff-Serv or MPLS. The Default value is 26.
Layer 3 Audio QoS
This field defines the layer 3 QoS parameter for audio packets. This is the value used for IP Precedence,
Diff-Serv or MPLS. The Default value is 46.
Layer 3 Video QoS
This field defines the layer 3 QoS parameter for video packets. This is the value used for IP Precedence,
Diff-Serv or MPLS. The Default value is 34.
Second Layer QoS 802.1Q/VLAN tag (Ethernet)
This field contains the value used for layer 2 VLAN tagging for the Ethernet network.
The Default value is 0.
Second Layer QoS 802.1p/Priority Value (Ethernet)
This assigns the priority value of the Layer 2 QoS packets on the Ethernet Network.
The Default value is 0.
Layer 2 QoS 802.1p priority (Wi-Fi)
This assigns the priority value of the Layer 2 QoS packets on the Wi-Fi Network. Default value is 0.
Proxy Settings
For some network setup, it is required to connect to the Internet via proxy server. Manually configure “Proxy
hostname”, “Proxy port” and “Bypass proxy for” in proxy settings for the phone to get Internet connection
successfully.
Tethering & Portable Hotspot
The GXV3370 can serve as a Wi-Fi access point for other devices to provide wireless access to the network
if the Portable Wi-Fi hotspot is turned on.
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1. Turn on hotspot by tapping on “Portable Wi-Fi hotspot”. Icon will show on the top status bar.
2. Tap on Set up Wi-Fi hotspotto configure network SSID, security type and password. Please make
sure the password has at least 8 characters. Otherwise, users will not be able to save the setting.
3. Tap on “Hotspot & tethering settings” and choose either to set up the hotspot for IPV4 only or for IPV4
& IPV6
4. On the other device that needs Wi-Fi access, turn on Wi-Fi, look for the SSID of the GXV3370 hotspot
and enter authentication information to get connected.
Figure 6: GXV3370 Wi-Fi Hotspot
Basic
Sound
Use the Voice settings to configure the phone’s sound mode, volume, ring tone and notification tone.
Silent mode. Tap on it to turn on/off the sound from speaker when there is an incoming call.
HDMI. Enable/disable audio switch between the phone and the HDMI output connected device (e.g.
TV). When enabled, the TV will be used for audio output.
Media Volume. Adjust the sound volume for media audio
Alarm Volume. Adjust the alarm ring volume
Ring Volume. Adjust the phone ringing volume
Notification Volume. Adjust the notification sound volume
Ringtone. Select phone’s ringtone for incoming call.
Default Notification Ringtone. Select notification ringtone.
Default Alarm Ringtone. Select the alarm ringtone
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Other Sounds. Enable/disable Dial pad Tones, Screen locking sounds, Touch sounds and Button
tones.
Display
Brightness. Tap on Brightness and scroll left/right to adjust the LCD brightness.
Screen timeout. Tap to open the dialog to set the screen timeout interval.
Screensaver timeout. Tap to set the screensaver timeout interval.
Screensaver. Enable/disable the screensaver. Two options are available:
- Clock: If the clock is set as screensaver, tap on and set the clock style and the Night Mode
- Screensaver: If screensaver is set, please tap on to set use a network images or use local
images as screensaver and set the Animation Interval between the images.
Font size. Tap on it to adjust the font size for LCD screen.
Zoom mode. Enable/Disable Zoom mode feature. If enabled, the font and icon become larger.
Language & keyboard
Language. Tap to open the list of chosen languages, Language Number 1 is the language used on the
phone. Tap on Add a Language to add more languages to the list.
Spell checker. Configure whether to check spellings and select the language to check.
Personal dictionary. Add new words to user’s dictionary so that they will not be displayed as error in
the text.
Keyboard & input methods
Set up default input method for virtual and physical keyboard and the different parameters of the related
to the Keyboard use. The default input method is Android Keyboard.
- Virtual Keyboard:
Android keyboard (AOSP): Set up the language used on Android keyboard and configure
its different parameters including sound, auto-correction, word suggestion and so on.
Manage Keyboards. Tap on the + sign to choose which keyboard to use on the phone.
- Physical Keyboard: When the physical keyboard is connected to the phone, users will have the
possibility to choose a keyboard among the available ones on the virtual keyboard
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Show virtual Keyboard: this option gives the possibility if keep showing the virtual
keyboard even if the physical one is connected to the phone.
Keyboard shortcuts helper: Display available shortcuts.
Speech-Text to Speech Input.
- Prefer engine. Select the engine type. The default engine is Pico TTS.
- General:
- Speech Rate. Adjust the speech rate.
- Pitch: Sets the speech pitch for the Text-To-Speech engine.
- Reset Speech Rate: Reset the speed at which the text is spoken to normal.
- Reset speech pitch: Reset the pitch at which the text is spoken to default.
- Listen to an example: Play and listen the example of speech synthesis.
- Default Language Status: Indicates if the language used is supported or not.
Pointer Speed. Adjust the sensitivity of the mouse pointer.
Date & Time
Enable and use specified NTP server address. Assign the URL or IP Address of NTP Server. The
default NTP Server used is pool.ntp.org
Set date. Set the current date for the GXV3370.
Set time. Set the time on the GXV3370 manually.
Select time zone. Select the time zone for the GXV3370.
Use 24-hour format. Check/uncheck to display the time using 24-hour time format or not. For example,
in 24-hour format, 13:00 will be displayed instead of 1:00 p.m.
Select date format. Select the format of year, month, and day for the date to be displayed.
System
Security Settings
Device Security-Screen lock. Set up pattern or password for screen lock. Wizard will be provided to
set up the pattern. The screen will be locked after booting up or the screen is off (i.e., screensaver
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screen activated, or manually slide down Status BarScreen Off to turn off LCD). Users will
then be required to enter password or pattern to login. When the screen is locked, users can still be
able to answer or reject incoming call.
Passwords-Make passwords visible. Check/uncheck to show/hide letters when user’s type screen
lock password instantly.
Device Administration
- Device administrators. View or deactivate device administrators.
- Unknown sources. Check/uncheck to enable/disable permission to install applications that you
obtain from web sites and email.
Credentials Storage
- Storage Type: Shows the type of the storage which is “Hardware-Backed by default
- Trusted Credentials. Display trusted CA certificates for system or user. Users can tap on the
certificate to check the credential details or disable it.
- User Credentials. View and modify stored credentials
- Install from SD card. Install encrypted certificates from SD card.
- Clear credentials. Clear credential storage of all contents and reset its password.
Advanced
- Trust Agents. View or deactivate trust agents
- Apps with usage access. Manage what apps have access to app-usage data on your device.
Peripherals
Plug in RJ9/UHS Headset. Switch the media channel to RJ9 headset after plugging.
Accounts
Add a system account to synchronize contacts calendars and other information.
Power Information
PoE Power Supply notification. If enabled, the phone’s system will display a notification of disabling USB
socket when using PoE power supply. If disabled, the notification will not be shown.
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Reboot the Phone
Press to reboot the phone. A confirmation window will pop up to Cancel or go on with the reboot.
Apps
Application Management
Tap on an application, process, or service to open it. The Application Info screen for each application lists
its name, version, size, etc. Depending on the app, it may also include options for managing the
application’s data, forcing the application to stop, and disabling the application. Usually the options are:
Tap the Force stopsoftkey to stop an application forcefully. This setting might not be valid for some
applications.
Tap the Stop softkey to stop an application gracefully. This setting might not be valid for some
applications.
Tap the Disablesoftkey to disable the application. Users could tap on “Enable” to turn it back on again.
This usually applies to the built-in applications.
Tap the “Uninstall” softkey to uninstall the applications.
Storage provides storage information that an application uses on the phone. Tap “Clear data” to delete
an application setting and other data. This setting might valid for some applications. If the application
stores data in a temporary space of the phone’s memory, “Cache” lists how much information is stored.
Tap on “Clear cache to clear the cache.
Permissionslists information of the data that the app has access to. For example, the application
might access the location information, storage, phone calls etc.
Open by default. If the application is configured to launch certain file type by default, tap on “Clear
defaults” to reset this.
If an application is misbehaving, tap on “Reportsoftkey (if available) to send the developer information
for the application.
Memory will show the memories used on the phone by the applications
Modify System settings gives the application the permission to modify the system settings
Store provides Information about the Install source of the App
Note:
Stopping a built-in application, operating system processes or services might disable one or more
dependent functions on the phone. Users may need restart the phone to restore full functionality again.
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Default Application
This page allows to set default applications to launch with certain actions. Default applications can be set
for following actions:
Opening Links. Select which application to use as default when clicking on a web link (browser);
when opening a picture (gallery) or when opening a music file (music).
Assist & Voice Input. Select a default application if previously installed.
Home app. Select default launcher application if already installed.
Browser app. Select default browser if more than one is installed.
Emergency app. Select emergency default application if already installed.
Notification Center
Tap on an application, process, or service to open it. The notification Info screen for each application lists
supported actions and allow user to activate/deactivate each notification. Following notifications can be
configured (supported notifications depend on the applications):
Block all.
Show silently.
Advanced
Account Settings
Account Settings page allows to configure SIP settings for each account. Tap on Account# to access the
settings, when configured press sign (on the top right corner) to confirm the changes or press back button
to cancel them. Users can press Empty configuration on the bottom of the page to clear all the settings.
Following settings can be configured for each account. Refer to [Account/General Settings] for description
of each option.
Account Activation.
Account Name.
SIP Server.
SIP User ID.
SIP Authentication ID.
SIP Authentication Password.
Outgoing Proxy Server.
Voicemail User ID.
Display Name.
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System Update
This page allows to initiate upgrade process by checking if a new firmware is available in the configured
firmware server path, and then upgrading if available. Users can press Settings to configure
Firmware/Provisioning settings directly from the phone’s LCD. Following settings can be configured from
this screen:
Firmware upgrade and configuration file detection. This will send a request to firmware and
provisioning server to upgrade/provision the phone if the files are available on the servers.
Firmware:
o Upgrade Mode: This field allows the user to choose the firmware upgrade method: TFTP, HTTP
or HTTPS.
o HTTP/HTTPS username: The username for the HTTP/HTTPS server if set up on the server.
o HTTP/HTTPS password: The password for the HTTP/HTTPS server if set up on the server.
o Firmware Server Path: This defines the server path for the firmware server. It can be different
from the configuration server for provisioning.
Config:
o Upgrade mode: This field allows the user to choose the provisioning method: TFTP, HTTP or
HTTPS.
o HTTP/HTTPS username: The username for the HTTP/HTTPS server if set up on the server.
o HTTP/HTTPS password: The password for the HTTP/HTTPS server if set up on the server.
o Config Server Path: This defines the server path for the provisioning server. It can be different
from the firmware server.
Syslog
Syslog level: Select the level of logging for syslog. The default setting is None”. There are 4 levels:
DEBUG, INFO, WARNING and ERROR.
System log protocol: Select the protocol of syslog (UDP or SSL/TLS).
Syslog server address: The URL/IP address for the syslog server. If the GXV3370 has network
connection, the phone will send the syslog packets to this server address.
System log keyword filtering: Only send the syslog with keyword, multiple keywords are
separated by comma. Example: set the filter keyword to “SIP” to filter SIP log.
System Security
Disable Web Login: This disables web GUI access.
Developer Mode. To enable/disable developer mode.
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Revoke debugging authorization. To Revoke access to debugging from all computers previously
authorized
Factory Reset. Restore default settings.
Status
Account Status
This page displays all available accounts on the phone with respective status (Configured/Not Configured
and Registered/Unregistered).
Network Status
This page displays Network status including Ipv4/v6 address, subnet mask, gateway, DNS server…
System Info
This page shows system info including Total Memory, Available Memory, Android Version, System Version,
Hardware version, CPU temperature…
Storage Status
This page shows device storage status.
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GXV3370 WEB GUI SETTINGS
The GXV3370 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow users to configure the application phone through a Web browser such as Microsoft’s IE, Mozilla
Firefox, Google Chrome and etc.
Status Page Definitions
Status/Account Status
Account
16 SIP accounts on the phone.
Number
SIP User ID for the account.
SIP Server
URL or IP address, and port of the SIP server.
Status
Registration status for the SIP account.
Status/Network Status
MAC Address
Global unique ID of device, in HEX format. The MAC address will be used
for provisioning and can be found on the label coming with original box and
on the label located on the back of the device.
NAT Type
Type of NAT connection used by the phone.
Ipv4
Address Type
Configured address type: DHCP, Static IP or PPPoE.
Ipv4 Address
IP address of the phone.
Subnet Mask
Subnet mask of the phone.
Default Gateway
Default gateway of the phone.
DNS Server 1
DNS Server 1 of the phone.
DNS Server 2
DNS Server 2 of the phone.
Ipv6
Ipv6 Address Type
Configured address type: DHCP, Static IP or PPPoE.
Ipv6 Address
IPv6 address of the phone.
Ipv6 DNS Server 1
IPv6 DNS Server 1 of the phone.
Ipv6 DNS Server 2
IPv6 DNS Server 2 of the phone.
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Status/System Info
Account Page Definitions
GXV3370 has 16 lines that can be configured to accommodate 16 independent SIP accounts. Each SIP
account has an individual configuration page.
Account/General Settings
On Register
Account Active
Indicates whether the account is active. The default value for the first
account is Yes.
Account Name
Configures the name associated with each account to be displayed on the
LCD.
SIP Server
Specifies the URL or IP address, and port of the SIP server. This should
be provided by VoIP service provider (ITSP).
SIP User ID
Configures user account information provided by your VoIP service
provider (ITSP). It is usually in the form of digits similar to phone number
or actually a phone number.
SIP Authentication ID
Configures the SIP service subscribers Authenticate ID used for
authentication. It can be identical to or different from the SIP User ID.
SIP Authentication
Password
Configures the account password required for the phone to authenticate
with the ITSP (SIP) server before the account can be registered. After
saving, it will appear as hidden for security purpose.
Display Name
Specifies the SIP server subscribers name (optional) that will be used for
Caller ID display. The configured content will be included in the From,
Contact and P-Preferred-Identity headers of SIP INVITE message.
Product Model
Product model of the phone: GXV3370.
Hardware Revision
Hardware version number.
Part Number
Product part number.
Serial Number
Product serial number.
System Version
Firmware version ID. This is the main software release version, which is
used to identify the software system of the phone.
Recovery Version
Recovery image version.
Boot Version
Booting code version.
Kernel Version
The kernel version.
System Up Time
System up time since the last reboot.
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Tel URI
Indicates E.164 number in “From” header by adding “User=Phone”
parameter or using “Tel:” in SIP packets, if the phone has an assigned
PSTN Number.
Disabled: Will use “SIP User ID” information in the Request-Line and
“From” header.
User=Phone: “User=Phone” parameter will be attached to the
Request-Line and “From” header in the SIP request to indicate the
E.164 number. If set to “Enable”.
Enabled: “Tel:” will be used instead of “sip:” in the SIP request.
Please consult your carrier before changing this parameter. Default is
“Disabled”.
Voice Mail Access
Number
Sets if the phone system allows users to access the voice messages by
pressing the MESSAGE key on the phone. This ID is usually the VM portal
access number. For example, in UCM6xxx IPPBX, *97 could be used.
Network Settings
Outbound Proxy
Configures the IP address or the domain name of the primary outbound
proxy, media gateway or session border controller. It is used by the phone
for firewall or NAT penetration in different network environments. If a
symmetric NAT is detected, STUN will not work and only an outbound
proxy can provide a solution
Secondary Outbound
Proxy
Sets IP address or domain name of the secondary outbound proxy, media
gateway or session border controller. The phone system will try to connect
the Secondary outbound proxy only if the primary outbound proxy fails.
DNS Mode
Defines which DNS service will be used to lookup IP address for SIP
server’s hostname. There are three modes:
A Record
SRV
NATPTR/SRV
To locate the server by DNS SRV set this option to SRV or
NATPTR/SRV. Default setting is A Record.
Maximum Number of SIP
Request Retries
Sets the maximum number of retries for the device to send requests to the
server. In DNS SRV configuration, if the destination address does not
respond, all request messages are resent to the same address according
to the configured retry times. The default setting is 4. The range is from 1
to 10.
DNS SRV Fail-over Mode
The option will decide which IP is going to be used in sending subsequent
SIP packets (ex: Register refresh requests) after the list of Ips for SIP
server host is resolved with DNS SRV.
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Default (prefer server with lowest SRV priority):
The phone will always prefer to send SIP requests to the available server
having the lowest priority, and in case it’s down it contacts the next one,
but once the server having lowest priority is UP again, the phone will switch
over to this one.
Saved one until DNS TTL (Stay on responding IP until DNS
timeout):
On this mode, the phone will resolve DNS SRV records and tries to send
the request to the server having lowest priority and if it doesn’t respond, it
will move on to the next IP until one of the servers responds, once this
happen the phone will keep contacting this responding IP until DNS timeout
(30 minutes) before starting over.
Saved one until no response (Stay on responding IP until its
failure):
On this mode, the phone will send SIP requests to the last responding IP,
and it does not failover/switchover to the next one until this responding
server is down.
Saved Failback follows failback expiration timer:
On this mode, the phone will send all SIP requests to the current failover
SIP server or Outbound Proxy until the failback timer expires.
Failback Expiration can specify the duration (in minutes) since failover to
the current SIP server or Outbound Proxy before making failback attempts
to the primary SIP server or Outbound Proxy. The default setting is 60. The
range is from 1 to 64800.
NAT Traversal
Specifies which NAT traversal mechanism will be enabled on the phone
system. It can be selected from the dropdown list:
NAT NO
STUN
Keep-alive
UPnP
Auto
VPN
If the outbound proxy is configured and used, it can be set to “NAT NO”.
If set to STUN and STUN server is configured, the phone system will
periodically send STUN message to the SUTN server to get the public IP
address of its NAT environment and keep the NAT port open. STUN will
not work if the NAT is symmetric type.
If set to “Keep-alive”, the phone system will send the STUN packets to
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maintain the connection that is first established during registration of the
phone. The “Keep-alive” packets will fool the NAT device into keeping the
connection open and this allows the host server to send SIP requests
directly to the registered phone.
If it needs to use OpenVPN to connect host server, it needs to set it to
“VPN”. If the firewall and the SIP device behind the firewall are both able
to use UPNP, it can be set to “UPNP”. Both parties will negotiate to use
which port to allow SIP through. The default setting is Keep-alive.
Proxy-Require
Adds the Proxy-Required header in the SIP message. It is used to indicate
proxy-sensitive features that must be supported by the proxy. Do not
configure this parameter unless this feature is supported on the SIP server.
Account/SIP Settings
SIP Basic Settings
SIP Registration
Allows the phone system to send SIP REGISTER messages to the
proxy/server. The default setting is Yes.
Unregister before New
Registration
Controls whether to clear SIP user’s information by sending un-register
request to the proxy server. When set to All, the un-registration is
performed by sending a REGISTER message with “Contact” header set to
* and Expires=0 parameters to the SIP server when the phone starts pre-
registration after rebooting. If set to Instance, the phone only cleans the
current SIP users info by sending REGISTER message with “Contact”
header set to concerned SIP user’s info and Expires=0 parameters to the
SIP server. The default setting is Instance.
Register
Expiration (m)
Configures the time period (in minutes) in which the phone refreshes its
registration with the specified registrar. The default setting is 60. The
maximum value is 64800 (about 45 days).
Subscribe Expiration
(m)
Specifies the frequency (in minutes) in which the phone refreshes its
subscription with the specified registrar. The maximum value is 64800
(about 45 days). Default value is 60.
Re-register before
Expiration (s)
Specifies the time frequency (in seconds) that the phone sends re-
registration request before the Register Expiration. The default setting is 0.
The range is from 0 to 64,800.
Registration Retry Wait
Time (s)
Configures the time period (in seconds) in which the phone will retry the
registration process in the event that is failed. The default setting is 20. The
maximum value is 3600 (1 hour).
Add Auth Header on RE-
Configure if the SIP account needs to add Auth header in RE-REGISTER.
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REGISTER
If the option is checked, device will always add authentication
header in REGISTER.
If the option is unchecked, device will only send authentication for
the first REGISTER.
Enable SIP OPTIONS
Keep Alive
Enables SIP OPTIONS to track account registration status so the phone
will send periodic OPTIONS message to server to track the connection
status with the server. The default setting is No.
SIP OPTIONS Keep Alive
Interval (s)
Configures the time interval when the phone sends OPTIONS message to
SIP server. The default setting is 30 seconds, which means the phone will
send an OPTIONS message to the server every 30 seconds. The default
range is 1-64800.
SIP OPTIONS Keep Alive
Maximum Tries
Configures the maximum times of sending OPTIONS message
consistently from the phone to server. Phone will keep sending OPTIONS
messages until it receives response from SIP server. The default setting is
“3”, which means when the phone sends OPTIONS message for 3 times,
and SIP server does not respond this message, the phone will send RE-
REGISTER message to register again. The valid range is 3-10.
Subscribe for MWI
Configures the phone system to subscribe voice message service. If it is
set to “Yes”, the phone system will periodically send SIP SUBSCRIBE
message for Message Waiting Indication service. GXV3370 phone system
supports both synchronized and non-synchronized MWI. Default is “No”.
Use Privacy Header
Determines if the Privacy header will be presented in the SIP INVITE
message and if it includes the caller info in this header. If it is set to
“Default”, the Privacy Header will be omitted in INVITE when Huawei IMS”
special feature is active. If set to Yes”, it will always be presented. If set to
“No”, it will always be omitted. The default setting is “Default”.
Use P-Preferred-Identity
Header
Specifies if the P-Preferred-Identity Header will be presented in the SIP
INVITE message. If set to “default”, the P-Preferred-Identity Header will be
omitted in SIP INVITE message when “Huawei IMS” special feature is
active. If set to “Yes”, the P-Preferred-Identity Header will always be
presented. If set to “No”, it will be omitted. The default setting is “Default”.
Use P-Access-Network-
Info Header
Use P-Access-Network-Info header in SIP INVITE message.
Default setting is Yes.
Use P-Emergency-Info
Header
Use P-Emergency-Info header in SIP INVITE message.
Default setting is Yes.
Use MAC Header
Use MAC Header in SIP register request and add MAC address in User-
Agent header in SIP request. Default Setting is NO.
SIP Transport
Determines which network protocol will be used to transport the SIP
message. It can be selected from TCP/UDP/TLS. Default setting is “UDP”.
Local SIP Port
Determines the local SIP port used to listen and transmit. The default
setting is 5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066
for Account 4, 5068 for Account 5, and 5070 for Account 6. The valid range
is from 5 to 65535.
SIP URI Scheme When
Using TLS
Defines which SIP header, sip” or “sips”, will be used if TLS is selected for
SIP Transport. The default setting is “sip”.
Use Actual Ephemeral
Port in Contact with
TCP/TLS
Determines the port information in the Via header and Contact header of
SIP message when the phone system use TCP or TLS. 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”.
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Support SIP Instance ID
Determines if the phone system will send SIP Instance ID. The SIP
instance ID is used to uniquely identify the device. If set to “Yes”, the SIP
Register message Contact header will include +sip.instance tag.
The default setting is Yes.
SIP T1 Timeout
Defines an estimate of the round-trip time of transactions between a client
and server. If no response is received in T1, the figure will increase to 2*T1
and then 4*T1. The request re-transmit retries would continue until a
maximum amount of time define by T2.
The default setting is 0.5 sec.
SIP T2 Interval
Specifies the maximum retransmit time of any SIP request messages
(excluding the SIP INVITE message). The re-transmitting and doubling of
T1 continues until it reaches the T2 value. The default setting is 4 sec.
SIP Timer D Interval
Defines the amount of time that the server transaction can remain when
unreliable response (3xx-6xx) received. The valid value is 0-64 seconds.
The default value is 0.
Remove OBP from
Route
Configures the phone system to remove the outbound proxy URI from the
Route header. This is used for the SIP Extension to notify the SIP server
that the device is behind a NAT/Firewall. If it is set to “Yes”, it will remove
the Route header from SIP requests. The default setting is No.
Enable 100rel
Actives PRACK (Provisional Acknowledgment) method. PRACK improves
the network reliability by adding an acknowledgement system to the
provisional Responses (1xx). It is set to “Yes”, the phone system will
response to the 1xx response from the remote party.
The default setting is No.
Session Timing
Enable Session Timer
Allows the phone system to use the session timer, when set to “Yes”, it will
be added in the SIP INVITE message to notify the server.
Session Expiration (s)
Configures the phone system’s SIP session timer. It enables SIP sessions
to be periodically refreshed via a SIP request (UPDATE, or re-INVITE). If
there is no refresh via an UPDATE or re-INVITE message, the session will
be terminated once the session interval expires. Session Expiration is the
time (in seconds) where the session is considered timed out, provided no
successful session refresh transaction occurs beforehand.
The default setting is 180. The valid range is from 90 to 64800.
Min-SE (s)
Determines the minimum session expiration timer (in seconds) if the phone
act as a timer refresher.
The default setting is 90. The valid range is from 90 to 64800.
UAC Specify Refresher
Sets which party will refresh the active session if the phone makes
outbound calls.
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Account/Codec Settings
Preferred Vocoder
Preferred Vocoder
Lists the available and enabled Audio codecs for this account. Users can
enable the specific audio codecs by moving them to the selected box and
set them with a priority order from top to bottom. This configuration will be
included with the same preference order in the SIP SDP message.
If it is set to “UAC” and the remote party does not support Refresher
feature, the phone system will refresh the active session.
If it is set to “UAS”, the remote party will refresh it. If it is set to “Omit”, the
header will be omitted so that it can be selected by the negotiation
mechanism. The default setting is “Omit”.
UAS Specify Refresher
Specifies which party will refresh the active session if the phone receives
inbound calls. If it is set to “UAC”, the remote party will refresh the active
session. If it is set to “UAS” and the remote party does not support refresh
feature, the phone system will refresh it.
The default setting is “UAC”.
Caller Request Timer
Sets the caller party to act as refresher by force. If set to “Yes” and both
party support session timers, the phone will enable the session timer
feature when it makes outbound calls. The SIP INVITE will include the
content “refresher=uac”. The default setting is “No”.
Callee Request Timer
Sets the callee party to act as refresher by force. If set to “Yes” and both
parties support session timers, the phone will enable the session timer
feature when it receives inbound calls. The SIP 200 OK will include the
content “refresher=uas”. The default setting is “No”.
Force Timer
Configures the session timer feature on the phone system by force.
If it is set to “Yes”, the phone will use the session timer even if the remote
party does not support this feature.
If 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”.
Force INVITE
Sets the SIP message type for refresh the session. If it is set to “Yes”, the
Session Timer will be refreshed by using the SIP INVITE message.
Otherwise, the phone system will use the SIP UPDATE or SIP OPTIONS
message. Default is “No”.
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Codec Negotiation
Priority
Configures the phone to use which codec sequence to negotiate as the
callee. When set to “Caller”, the phone negotiates by SDP codec sequence
from received SIP Invite; When set to “Callee”, the phone negotiates by
audio codec sequence on the phone.
The default setting is “Callee”.
Use First Matching
Vocoder in 200OK SDP
Configures the phone to use the first matching codec in the 200OK
message.
The default value is 0.
ILBC Frame Size
Sets the ILBC (Internet Low Bitrate Codec) frame size if ILBC is used.
Users can select it from 20ms or 30ms. The default setting is 30ms.
G726-32 ITU Payload
Configures G726-32 payload type for ITU packing mode. Payload 2 is static
and payload dynamic is dynamic.
The default setting is “2”.
G726-32 Dynamic PT
Specifies the G726-32 payload type, and the valid range is 96 to 126. The
default setting is “126”.
Opus Payload Type
Defines the desired value (96-126) for the payload type of the Opus codec.
The default value is 123.
DTMF
Specifies the mechanism to transmit DTMF (Dual Tone Multi-Frequency)
signals.
There are 3 supported modes: in audio, RFC2833, or SIP INFO.
In audio, which means DTMF is combined in the audio signal (not very
reliable with low-bit-rate codecs);
RFC2833, which means to specify DTMF with RTP packet. Users could
know the packet is DTMF in the RTP header as well as the type of DTMF.
SIP INFO, which uses SIP INFO to carry DTMF. The defect of this mode
is that it is easily to cause desynchronized of DTMF and media packet if
the SIP and RTP messages are required to transmitted, respectively.
The default setting is RFC2833.
DTMF Payload Type
Configures the RTP payload type that indicates the transmitted packet
contains DTMF digits. Valid range is from 96 to 126. Default value is 101.
Enable Audio RED with
FEC
If set to Yes, FEC will be enabled for audio call.
The default setting is No.
Audio FEC Payload Type
Configures audio FEC payload type. The valid range is from 96 to 126.
The default value is 121.
Audio RED Payload
Type
Configures audio RED payload type. The valid range is from 96 to 126.
The default value is 124.
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Silence Suppression
Enables the silence suppression/VAD feature. If it is set 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”.
Voice Frames Per TX
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the “ptime” value for the SDP will
change with different configurations here. This value is related to the codec
used and the actual frames transmitted during the call. For end users, it is
recommended to use the default setting, as incorrect settings may
influence the audio quality.
The default setting is 2.
Preferred Video Codec
Preferred Video Codec
This parameter allows user to select preferred video codec from the
available list. The phone supports H.264, H.263.This configuration will be
included with the same preference order in the SIP SDP message.
Enable Video FEC
When enabled, the video sender will temporarily allocate part of the
bandwidth to one data channel to send FEC data to system, thus, to
improve the video quality the receiver gets. Enabling this function will take
up part of bandwidth and reduce call rate. The default setting is “Yes”.
Enable RFC5168
Support
Enables/disables RFC5168 mechanism for video calls. RFC5168 allows
SIP party to request the sender to refresh its video frame in H.264 or
refresh the full picture in VP8.The default setting is “Yes”.
Video FEC Mode
If set to 0, FEC is not sent by separate port. If set 1, FEC is sent by separate
port. Default setting is 0.
FEC Payload Type
Configures FEC payload type. The range is 96-127. Default setting is 120.
Packetization Mode
Configures video packetization mode. If set to Single NAL Unit Mode, the
packetization mode will be negotiated as single NAL unit mode when dial
video calls, if the other party does not support the negotiation, then single
NAL unit mode will be used for video encoding by default. If set to Non-
Interleaved Mode, the packetization mode will be negotiated as Non-
interleaved mode when dial video calls, If the other party does not support
negotiation, then the Non-interleaved mode will be used for video encoding
by default. The default setting is Non-Interleaved Mode.
H.264 Image Size
Sets the H.264 image size. It can be selected from the dropdown list.
720P
4CIF
VGA
CIF
QVGA
QCIF
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Note: For some network environment, the default setting “720P” might be
too high that causes no video or video quality issue during video call. In
this case, please change “H.264 Image Size” to “VGA” or “CIF” and change
“Video Bit Rate” to “384kbps” or lower. The default setting is 720P.
Use H.264 Constrained
Profiles
Configures that whether to set H.264 constrained profiles.
The default setting is “No”.
H.264 Profile Type
Selects the H.264 profile type from the dropdown list.
Baseline Profile
Main Profile
High Profile
BP/MP/HP (Default Setting)
Note: Lower levels are easier to decode, but higher levels offer better
compression. Usually, for the best compression quality, choose “High
Profile”; for playback on low-CPU machines or mobile devices, choose
“Baseline Profile”. If BP/MP/HP” is selected, all three profiles “Baseline
Profile” “Main Profile” and “High Profile” will be used for negotiation during
video decoding to achieve the best result. This is usually used in video
conference when there is higher requirement on the video.
Video Bit Rate
Configures the bit rate for video call. It can be selected from the dropdown
list. The default setting is 2048 kbps. The valid range is from 32 2048
kbps.
Note: The video bit rate can be adjusted based on the network
environment. Increasing the video bit rate may improve video quality if the
bandwidth is permitted. If the bandwidth is not permitted, the video quality
will decrease due to packet loss. For some network environment, the
default setting 720P” might be too high that causes no video or video
quality issue during video call. In this case, please change “H.264 Image
Size” to VGA” or “CIFand change “Video Bit Rate” to “384kbps” or lower.
SDP Bandwidth
Attribute
Sets the SDP bandwidth attribute. It can be selected from the drop-down
list. The default setting is “Media Level”.
Standard: use AS format in session level; use TIAS format in media
level
Media Level: use AS format in media level.
Session Level: use AS format in session level.
None: no modifications in the session format.
Note: Please do not modify this setting without knowing the session format
supported by the server. Otherwise, it might cause video decoding failure.
H.264 Payload Type
Specifies the H.264 codec message payload type format. The default
setting is 99. The valid range is from 96 to 127.
H.263 Encoder
Resolution
Selects the resolution (CIF/QCIF) used for H.263 codec.
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Presentation Settings
Enable BFCP
If set to Yes, the device will be able to receive the presentation stream in
video calls and video meetings.
Initial INVITE with Media
Info
Initial INVITE SDP contains presentation media.
Presentation H.264
Image Size
Selects the H.264 image size. Users can select 1080P or 720P.
Presentation H.264
Profile Type
Select the Presentation H.264 Profile Type from Baseline Profile, Main
Profile, High Profile and BP&MP&HP. The default setting is
BP&MP&HP.
The lower the profile type is, the easier the packet can be decoded.
However, higher level has high compression ratio. For device with low
CPU, select Baseline Profile to play record; Baseline Profile is more
likely to be used in a video conference that has high demanding for the
video quality. Select among the three types to achieve best video effect.
Presentation Video Bit
Rate
Configures the bit rate of the video. The video bit rate can be adjusted
based on the network environment. Increasing the video bit rate may
improve video quality if the bandwidth is permitted. If the bandwidth is not
permitted, the video quality will decrease due to packet loss. Video Bit Rate
can be set to integer value from 512kbps to 2048kbps.
Presentation Video
Frame Rate
Configure the video frame rate for presentation.
BFCP Transport
Protocol
Defines the transport protocol used for BFCP. Users can choose from
Auto/UDP/TCP. The default setting is UDP first, if not supported, then
choose TCP. If choose Auto, automatically switches between UDP
and TCP.
Account/RTP Settings
SRTP Mode
Sets if the phone system will enable the SRTP (Secured RTP) mode. It can
be selected from dropdown list:
Disable
Enabled but not forced
Enabled and forced
SRTP uses encryption and authentication to minimize the risk of denial of
service. (DoS). If the server allows to use both RTP and SRTP, it should
be configured as “Enabled but not forced”. The default setting is “Disable”.
SRTP Key Length
Configures all the AES (Advanced Encryption Standard) key size within
SRTP. It can be selected from dropdown list:
AES128&256 bit
AES 128 bit
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AES 256 bit
If it is set to “AES 128&256 bit”, the phone system will provide both AES
128 and 256 cipher suite for SRTP. If set to “AES 128 bit”, it only provides
128-bit cipher suite; if set to AES 256 bit”, it only provides 256-bit cipher
suite. The default setting is “AES128&256 bit”.
Enable SRTP Key
Lifetime
Defines the SRTP key lifetime. When this option is set to be enabled, during
the SRTP call, the SRTP key will be valid within 2
31
SIP packets, and phone
will renew the SRTP key after this limitation. Default is “Yes”.
RTCP Destination
Configures a remote server URI where the RTCP messages will be sent to
during an active call.
Symmetric RTP
Configures if the phone system enables the symmetric RTP mechanism.
If it is set to “Yes”, the phone system will use the same socket/port for
sending and receiving the RTP messages. The default setting is No.
RTP IP Filter
Receives the RTP packets from the specified IP address and Port by
communication protocol. If it is set to “IP Only”, the phone only receives the
RTP packets from the specified IP address based on the communication
protocol; If it is set to “IP and Port”, the phone will receive the RTP packets
from the specified IP address with the specified port based on the
communication protocol. The default setting is “Disable”.
RTP Timeout Timer (s)
Disconnects the call automatically when there is no RTP stream for a
specific timeout. Default is 30 seconds.
VQ RTCP-XR Collector
Name
Configures the host name of the RTCP server that accepts voice quality
reports contained in SIP PUBLISH messages.
VQ RTCP-XR Collector
Address
Configures IP address of the RTCP server that accepts voice quality
reports contained in SIP PUBLISH messages.
VQ RTCP-XR Collector
Port
Configures the port of the RTCP server that accepts voice quality reports
contained in SIP PUBLISH messages.
Account/Call Settings
Call Features
Enable Video Call
Configures the video call function for this account. If set to Default, it will
be configured according to global video call function.
Start Video
Automatically
Permits the phone system to enable the video feature automatically when
it makes an outbound call. If set to “Yes”, the video codec attributes will be
included in the SIP INVITE message. Or the attributes will not be included.
The default setting is “Yes”.
Remote Video Request
Configures the preference to handle video request from the remote party
during an audio call. The default is “Prompt”.
Prompt”: A message will be prompted if a video request is received.
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Users can select “Yes” to establish video or “No” to reject the request.
Accept”: Video request will be accepted automatically, and video will be
established.
Deny”: Video request will be rejected automatically.
Video Layout
Defines whether to enter full screen when incoming video call is answered.
Fullscreen: GXV3370 will show the remote video feed in full screen.
Only display Remote Screen: GXV3370 only displays the remote
screen in full screen mode.
Default”: GXV3370 will show both remote and local video feed.
Auto Answer
Sets the phone system to allow to answer an incoming call automatically
when idle. If it is set to “Yes”, the phone will automatically enable the
speaker phone to answer all the incoming calls after a short reminding
beep. If set to “Enable Intercom/Paging”, it will automatically answer the
incoming calls whose SIP INVITE includes auto-answer tag in the info
header. The default setting is “No”.
Play Warning Tone for
Auto Answer Intercom
When this option is enabled, the phone will play a warning tone When auto-
answering intercom. The default setting is “Yes”.
Intercom Barging
Configures whether to answer the incoming intercom call when there is
already an active call on the phone. When” Intercom Barging” is enabled,
and if the current active call is an intercom call, the incoming intercom call
will be automatically rejected; otherwise if the current active call is not an
intercom call, the current active call will be put on hold and the incoming
intercom will be automatically answered. When “Intercom Barging” is
disabled, a prompt will show up indicating the incoming intercom call without
interrupting the current active call. Default setting is disabled.
Auto Preview
Configures whether to turn on video to preview the video of the caller. If set
to Yes, the user can view the video and hear the caller on the incoming
page when there is an incoming call. If set to Yes with Ringing, the caller
can view the video of the caller and hear the ringtone on the incoming page
but cannot hear the caller. Note: If Auto Answer function has been enabled,
this function does not take effect. Default Value: No.
Send Anonymous
Sets the phone system to make an anonymous outgoing call. If set to Yes”,
the “From” header in the SIP INVITE messages will be set to anonymous,
blocking the Caller ID to be displayed. Default is “No”.
Intercept Anonymous
Calls
If set to Yes, anonymous calls will be automatically blocked. The default
setting is “No”.
Call Log
Categorizes the call logs saved for this account. If it is set to “Log All”, all
the call logs of this account will be saved. If set to “Log Incoming/Outgoing
Calls (Missed Calls Not Record)”, the whole call history will be saved other
than missed call. If it is set to “Disable Call All”, none of the call history will
be saved.
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If it is set to “Don’t Prompt Missed Call”, the phone will log the missed call
histories, but there is no prompt to indicate the missed calls on phone LCD.
The default setting is “Log All”.
Note: Call log will not show for Ring Group missed call if Do not prompt
Missed calls” is selected.
Enable Call Features
Configures the local start command feature. If it is set to “Yes”, the feature
will be enabled to recognize the local star code command. Otherwise, it will
be disabled. The default setting is “No”.
Enable Call Waiting
This feature allows user to set call waiting availability under each account
rather than globally.
Mute on Answer
Intercom Call
When enabled, phone will mute the incoming intercom call based on Call-
Info/Alert Info Headers. Default is disabled.
Transfer on 3-way
Conference Hang up
Transfers conference from hosted party when hang up, thus other parties
can continue the conference without interruption. Default is unchecked.
Use # as Dial Key
Treats “#” as the “Send” (or “Dial”) key when set to “Yes”. If set to No”, this
“#” key can be included as part of the dialed number. Please make sure the
dial plan is properly configured to allow dialing # out. Default is “Yes”.
Use # as Redial Key
Allows users to configure the "#" key as the "Redial" key. If set to "Yes", the
"#" key will immediately redial the last call. In this case, this key is equivalent
to the "Redial" key. If set to "No", the "#" key is treated as part of the dialed
string. Default is “Yes”.
DND Call Feature On
Configures the feature code to enable the DND (Do Not Disturb) feature
for this account. If it is configured, the phone will dial the feature code
automatically when the DND feature is enabled.
DND Call Feature Off
Configures the feature code to disable the DND (Do Not Disturb) feature
for this account. If it is configured, the phone will dial the feature code
automatically when the DND feature is disabled.
No Key Entry Timeout
(s)
Determines the expiration timer (in seconds) for no key entry. The dialed
digit will be sent out if no other digits entered within the set period. The
default value is 4 seconds. The valid range is from 1 to 15. This feature
does not work if the dialer page is entered via the Account Widget on the
phone.
Ring Timeout (s)
Defines the expiration timer (in seconds) for the rings with no answer. The
default setting is 60. The valid range is from 10 to 300.
Refer-To Use Target
Contact
Sets the phone system to use the target’s Contact header tag to the Refer-
To header in the SIP REFER message during an attended transfer.
The default setting is “No”.
RFC2543 Hold
If yes, c=0.0.0.0 will be used in INVITE SDP for hold.
Call Forward
Call Forward Type
Sets the Call Forwarding feature for this account.
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None: Disable call forwarding feature.
Unconditional: Set to forward all calls to a specified account.
Time based: Set the call forwarding rule based on time. The system
can forward incoming calls to the accounts of In Time Forward to and
Out Time Forward to.
Others: Set the call forwarding rule based on following account status.
Forward when Busy: the call will be forward to number set
under “Busy To” when the account is busy.
Forward when No Answer: The call will be forwarded to the
number set under “No Answer To” after the configured timeout.
(range 1- 120s)
Forward when DND: When the phone is on DND mode the
call will be forwarded to number configured under “DND To”.
Dial Plan
Dial Plan Prefix
Configures the digits prepended to the dialed number.
Disable Dial Plan
Enables/disables the Dial plan mechanism for different cases. If the
specific case is checked, the Dial plan mechanism will be disabled.
Dial Page: It controls the pattern of dialing numbers from the keypad,
phone app and account widget.
Contact: It controls the pattern of dialing numbers from local or LDAP.
Incoming call history: It controls the pattern of dialing numbers from
inbound call logs.
Outgoing Call History: It controls the pattern of dialing numbers from
outbound call logs.
MPK&Click2Dial: It controls the pattern of dialing numbers from MPK
app and the link on the webpage.
The default setting is unchecking all the cases.
Dial Plan
Configures the dial plan to establish the expected number and pattern of
digits for a telephone number. This parameter configures the allowed dial-
plan for the phone.
Dial Plan Rules:
1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d,+
2. Grammar: x any digit from 0-9;
a) xx+ or 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
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f) <2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
h) \+ - add + to the dialing number
i) , - play second dial tone.
j) Back slash \”—can be used to escape specific letters.
E.g. if { \p\a\r\k\+60 } dial plan is configured, park+60
should be able to pass dial plan check. This also can be
used to escape Mark and User-unreserved characters.
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+}
Allow 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 allow any length of numbers with leading digit 2, replacing the 2 with
011 when dialed.
Example 4: {0,x+}
If user dials 0 second dial tone will be played, and users can continue
entering digits. For instance, if pressing 01234, after pressing 0 second
dial tone is played, full number is sent in INVITE.
3. Default: Outgoing { x+ | \+x+ | *x+ | *xx*x+ }
Allow any number of digits, OR any number with a leading +, OR any
number with a leading *, OR any number with a leading * followed by a 2
digits number and a *. To dial + from keypad, press on 0 until + appears
on LCD.
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 allow 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 |- allow dialing to any US/Canada Number with 11
digits length
011[2-9]x. allow international calls starting with 011
[3469]11 allow dialing special and emergency numbers 311, 411,
611 and 911
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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.
Caller IDs
Caller ID Display
Specifies which header tag will be used from the SIP INVITE message for
the Caller ID display. If it is set to Auto, the phone system will use the one
of the available headers in the priority hierarchy of P-Asserted Identify
Header, Remote-Party-ID Header and FROM Header. If it is set to “From
Header”, it will use the FROM header information for the Caller ID. If it is
set to “Disabled”, all the incoming calls Caller ID will be displayed with
“Unavailable”. The default setting is “Auto”.
Callee ID Display
If set to "Auto", the phone will update the callee ID in the order of P-
Asserted Identity Header, Remote-Party-ID Header and To Header in the
180 Ringing. If set to "Disabled", the callee ID will be displayed as
"Unavailable". If set to "To Header", the callee ID will not be updated and
displayed as To Header. The default setting is “Auto”.
Ring Tones
Account Ring Tone
Configures the ringtone for the account. Users can set ringtones from the
dropdown list. User can also import customized ringtone from LCD Setting
menu. The customized ringtone file name will also be showed up in the
dropdown list that allows user to select. A “silent” option has been added
to account ring tone list.
Ignore Alert-Info header
Configures the default ringtone. If set to “yes”, the incoming alert info
header from the SIP server will be ignored and default configured ringtone
will be played. The default setting is “No”.
Match Incoming Caller
ID
Specifies the rules for the incoming calls. If the incoming caller ID or Alert
Info matches the number, pattern or Alert Info text rules, the phone will play
the selected distinctive ringtone. The rule policy:
Specific caller ID number. For example, 8321123;
A defined pattern with certain length using x and + to specify, where x
could be any digit from 0 to 9. Samples:
xx+ : at least 2-digit number;
xx : only 2-digit number;
[345]xx: 3-digit number with the leading digit of 3, 4 or 5;
[6-9]xx: 3-digit number with the leading digit from 6 to 9.
Alert Info text
Users could configure the matching rule as certain text (e.g., priority)
and select the custom ring tone mapped to it.
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The custom ring tone will be used if the phone receives SIP INVITE
with Alert-Info header in the following format:
Alert-Info: <http://127.0.0.1>; info=priority
Distinctive Ring Tone
Selects the distinctive ring tone if the incoming caller ID matched the
specified Match Incoming Caller ID rule. If so, the phone will play the
selected ringtone.
Account/Advanced Settings
Security Settings
Check Domain
Certificate
Sets the phone system to check the domain certificates if TLS/TCP is used
for SIP Transport.
The default setting is "No".
Validate Certification
Chain
Configures whether to validate certification chain when TLS/TCP is
configured for SIP Transport. If this is set to “Yes”, phone will validate
server against the new certificate list. The default setting is “No”.
Validate Incoming SIP
Messages
Specifies if the phone system will check the incoming SIP messages caller
ID and CSeq headers. If the message does not include the headers, it will
be rejected.
The default setting is "No".
Allow Unsolicited
REFER
It is used to configure whether to dial the number carried by Refer-to after
receiving SIP REFER request actively. If it is set to "Disabled", the phone
will send error warning and stop dialing. If it is set to "Enabled/Force Auth",
the phone will dial the number after sending authentication, if the
authentication failed, then the dialing will be stopped. If it is set to
"Enabled", the phone will dial up all numbers carried by SIP REFER. The
default is "Disabled".
Only Accept SIP
Requests from Known
Servers
Answers the SIP request from saved servers when set to Yes”, only the
SIP requests from saved servers will be accepted; and the SIP requests
from the unregistered server will be rejected. The default setting is “No”.
Check SIP User ID for
Incoming INVITE
Configures the phone system to check the SIP User ID in the Request URI
of the SIP INVITE message from the remote party. If it does not match the
phone's SIP User ID, the call will be rejected. The default setting is “No”.
Allow SIP Reset
It is used to configure whether to allow SIP Notification message to perform
factory reset on the phone.
The default setting is "No".
Authenticate Incoming
INVITE
Configures the phone system to authenticate the SIP INVITE message
from the remote party. If set to "Yes", the phone will challenge the incoming
INVITE for authentication with SIP 401 Unauthorized response.
The default setting is "No".
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SIP Realm used for
Challenge INVITE &
NOTIFY
Configure this item to validate incoming INVITE, but you must enable
authenticate incoming INVITE first to make it take effect. You can verify the
NOTIFY information for the provision, including check- sync, resync and
reboot, but only when SIP NOTIFY authentication enabled first to make it
take effect.
MOH
MOH Mode
Configures MOH mode. If set to "Local MOH", a local MOH audio file needs
to be uploaded for this mode to work.
Upload Local MOH
Audio File
Loads the MOH (Music on Hold) file to the phone. Click on Browse” button
to upload the music file from local PC. The MOH audio file has to be in .wav
or .mp3 format.
Note: Please be patient while the audio file is being uploaded. It could take
more than 3 minutes to finish the uploading especially the file size is large.
The button will show as “Processingduring the uploading. Once done, it
will show as Browse” again. Click on “Save” on the bottom of the web
page and “Apply” on the top of the web page to save the change.
Advanced Features
Virtual Account Group
It is used to set to categorize accounts in server mode groups, the accounts
in the same group will be combined as one and the account widget will
display the Caller ID in the account with lowest ID. The phone can answer
any incoming calls to each account in groups.
If user makes an outbound call, the phone system will use the lowest ID
account by default. If the account fails or SIP INVITE message is timeout,
the phone system will failover to the next account in the group with higher
account ID. If all the accounts are not available in the group, the phone
system will traverse all the accounts in the group and notify the end users
the session is failed.
Allow Sync Phonebook
via SIP Notify
Allows users to synchronize XML phonebook upon receiving SIP NOTIFY
message with header Event: sync-contacts.
Note: Received SIP NOTIFY will be first challenged for authentication
purpose before contacting configured server to download XML phonebook.
The parameters used are the ones configured at [Download Contacts].
The authentication can be done either using admin credentials (if no SIP
account is configured) or using SIP account credentials. The default setting
is “Yes”.
Account/Special Features
Special Feature
Configures phone’s settings to meet different vendors’ server
requirements.
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Users can choose from Standard, CBCOM, RNK, China Mobile, ZTE IMS,
Mobotix, ZTE NGN, or Huawei IMS depending on the server type.
Call Settings
Feature Key
Synchronization
This feature is used for BroadSoft / Metaswitch call feature
synchronization. When it's set to BroadSoft / Metaswitch, DND and Call
Forward features can be synchronized with BroadSoft / Metaswitch server.
The call forward function will take effect on the server side while the local
call forward function is not effective.
Enable BroadSoft Call
Park
If enabled, phone will send SUBSCRIBE to BroadSoft server to obtain Call
Park notifications.
Conference URI
Configures the conference URI for BroadSoft Network Conference feature.
BroadSoft Call Center
Enables BroadSoft Call Center feature. When enabled, Feature Key
Synchronization will be enabled regardless of web UI settings.
Hoteling Event
Enables BroadSoft Hoteling feature.
Call Center Status
When enabled", the phone will send SUBSCRIBE to the server to obtain
call center status.
SCA
Enable SCA
(Shared Call
Appearance)
If enabled, the Shared Call Appearance (BroadSoft Standard) feature will
be applied for the registered account.
Enable Barge-in
If enabled, users can barge into an active call on a shared line.
Auto-filling Call Park
Feature Code
If enabled, the Call Park feature code will be automatically filled in the
dialing interface when the phone is in call pickup.
Call Park Feature Code
Configures the pickup feature code for Call Park.
Line-seize Timeout (s)
Configures the interval (in seconds) when the line seize is considered
timed out when Shared Line feature is used. Valid range is 15-60.
Phone Settings Page Definitions
Phone Settings/General Settings
Basic Settings
Local RTP Port
Defines the local RTP-RTCP port pair used to listen and transmit.
(The default value is 50040, Max is 65000). The following rule is applied:
N>=0, the default value of Port_Value is 5004.
- Audio RTP port: Port_Value+10*N
- Audio RTCP port: Port_Value+10*N+1
- Video RTP port: Port_Value+10*N+2
- Video RTCP port: Port_Value+10*N+3
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- FEC RTP port: Port_Value+10*N+4
- FEC RTCP port: Port_Value+10*N+5
- BFCP Protocol port: Port_Value+10*N+6
- BFCP RTP port: Port_Value+10*N+8
- BFCP RTCP port: Port_Value+10*N+9
Use Random Port
Forces the phone system to use random ports for both SIP and RTP
messages. This is usually necessary when multiple phones are behind the
same full cone NAT. The default setting is “No”.
Note: This parameter must be set to “No” for Direct IP Calling to work.
Disable in-call DTMF
display
Enables/disables the phone system to omit the DTMF digits displaying
from the LCD screen. The default setting is “No”.
Enable Enterprise
Contacts Timeout Auto
Search
Configures whether to display the matched content automatically in search
of the LDAP contacts when timeout. If set to “No”, users need to click the
“Search” button to search the matched contacts mentioned above. The
default setting is “Yes”.
Keep-alive Interval (s)
Specifies how the phone system will send a Binding Request packet to the
SIP server in order to keep the “ping hole” on the NAT router to open. The
default setting is 20 seconds. The valid range is from 10 to 160.
STUN Server
Configures the URI of STUN (Simple Traversal of UDP for NAT) server.
The phone system will send STUN Binding Request packet to the STUN
server to learn the public IP address of its network. Only non-symmetric
NAT routers work with STUN.
The default setting is “stun.ipvideotalk.com”.
TURN Server Username
Fill in the username to validate TURN server.
TURN Server Password
Fill in the password to validate TURN server.
Use NAT IP
Configures the IP address for the Contact header and Connection
Information in the SIP/SDP message. It should ONLY be used if it is
required by your ITSP.
The default setting is keep the box blank.
Guest Features
Guest Settings
Guest Login
Enable/disable guest login. Users should configure SIP domain name
when using this function. Users should input SIP username and password
on LCD after rebooting the phone. Setting is disabled by default and
requires reboot to take effect.
Guest Login Timeout
Configures guest login timeout. If user have logged in as guest, the web
page will log out and exit automatically if there is no operation upon timeout
Guest Login PIN Code
When set "Guest Login" option to "Yes" and "Guest Login Timeout" to
"Never", users should input the PIN code to login.
SIP Domain
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Server Alias
The server alias is used to customize the server name.
Server List
This is the list of SIP domain names under the selected severs. The
registration priority of addresses in the list is from top to bottom. Up to 16
SIP domains are supported.
Phone Settings/Call Settings
Call Settings
Enable Video Call
Feature
Enables the video call feature on the phone. The default setting is “Yes”.
Use Direct IP Call Mode
Configures enable/disable direct IP call mode of the phone. If set to "Yes",
the feature of direct IP call will be enabled.
Use Paging Call Mode
Configures enable/disable paging call mode of the phone. If set to "Yes",
the feature of paging call will be enabled.
Use Quick IP-Call mode
Sets the phone system to automatically fills in the first three octets to make
an outbound IP call. If it is set to “Yes”, users can dial an IP address under
the same LAN/VPN segment by entering the last octet in the IP address.
To dial quick IP call, offhook the phone and dial XXX (X is 0-9 and XXX
<=255), the 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. This setting can be configured after
enabling ‘Use Direct IP Call Mode option. The default setting is “No”.
Enable Call-Waiting
Enables the call waiting feature. If it is not checked, the phone system will
reject the second incoming call during an active session without user’s
knowledge. But this missed call record will be saved to remind users. The
default setting is “Yes”.
Enable Call-Waiting
Tone
Sets the phone system to play the call waiting tone if there is another
incoming call. If it is set to “No”, the phone will only display the indicator on
the LCD screen for another incoming call. The default setting is “Yes”.
Enable DND Reminder
Ring
Enables the DND reminder ring. If set to "Yes", the ring splash that
indicates an incoming call when DND is enabled will be played. Default
setting is “Yes
Enable Transfer
Enables the Transfer function. Default settings is “Yes”
Hold Call Before
Completing Transfer
When set to "Yes" the phone holds the second call before completing the
attended transfer (it sends the INVITE method to hold the call before
sending the REFER method).
Default Transfer Mode
Sets the default transfer mode for the phone system. If the Blind Transfer
or Attended Transfer mode is set, the phone system will use the specific
mode to transfer an active call.
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The users still have privilege to switch the mode on the LCD screen when
they tap the transfer key. The default setting is “Blind Transfer”.
Enable transfer via non-
Transfer Programmable
Key
Programmable Key with the type speed dial, BLF and speed dial via active
account will perform as transfer programmable keys under active call. The
transfer mode during the call depends on the “Default Transfer Mode”
mentioned above. MPK can also be selected as forward/transfer
destination on the ringing screen when [Enable Function for Incoming
Call] is set to “Call Transfer”.
Enable Function for
Incoming Call
Enables the preview feature for the incoming video calls. Defines the
function for the incoming video call.
If it is set to Preview”, the phone system will pop up the PREVIEW
key on the LCD screen when there is an incoming video call, and
users could tap on it to check video caller without answering the
incoming video call (the call will keep playing ringback on the caller
side).
Note: By pressing the preview button, the phone will send the SIP 183
message to the caller’s camera, based on SIP RFC3261; the callers
camera should start sending the stream to the phone upon receiving
the SIP 183.
At any time, the GXV3370 user can press the "Answer" button. If
done, then the phone will send the SIP 200OK and call will be fully
established.
If set to "Call Transfer", the phone system will pop up the
"TRANSFER" key on the LCD screen when there is an incoming call,
and users could tap on it to show up the dialer without answering the
incoming call, then, users could transfer this incoming call to others.
The default setting is “None”.
Enable Conference
Enables the conference. When set to “No”, the phone will block the
conference application.
The default setting is “Yes”.
Auto Conference
Allows the phone system to invite all call parties into a conference by
pressing the Conf key. If it is disabled, the end user has to add each call
party to conference manually.
The default setting is “No”.
Auto Mute on Entry
Configures whether to mute the call on entry automatically.
If set to "Disable", then do not use auto mute function.
If set to "Auto Mute on Outgoing Call", then mute automatically
when the other party answers the outgoing call.
If set to "Auto Mute on Incoming Call", then mute automatically
when answers the incoming call; If set to "Mute on Incoming &
Outgoing Call", then mute automatically when the call gets through.
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Note: This function only take effect when the phone is from the idle status
to call status. Users could click the Mute button on call interface to cancel
the current mute status.
Always Ring Speaker
Determines if the speaker will play the ringtone if the speaker channel is
not set as default channel. If set to Yes”, the phone will force to play the
ring speaker in speaker channel. The default setting is “No”.
End user might need this feature when the headset is connected.
Offhook Auto Dial
Configures the User ID/extension to dial automatically when the phone is
off hook. The phone will use the first account to dial the configured numbers
out.
The default setting is “No”.
Offhook Auto Dial Delay
(s)
Defines the timer for warm line dialing. After the timer expires, the phone
system will dial the configured number in Off-hook Auto Dial automatically.
If it is not configured, the configured number will be dialed immediately.
Offhook/OnHook
Timeout (s)
If configured, the phone will exit the dial-up screen when timeout after
Offhook or Onhook. default is 30s.
The valid range is 10-60s.
Handset Option
Configures the Handset options and can be set to:
If set to Enable: when off hook, the phone will enter into the dial
screen, and the media channel will switch to the handset; no matter if
any third app is opening.
If set to “Disable”: The Handset will be disabled.
If set to Auto”: when the 3rd-party application is calling, only the media
channel will switch to the handset when off hook; otherwise, the phone
will enter into the dial screen, and the media channel will switch to the
handset.
Default setting is “Auto”.
Auto UnHold When
Press the Line Key
Configures when there are multiple lines, whether to UnHold the line
automatically when click the line being held and hold the line in the primary
call. Note: the hold situation which is set manually will not be put on hold
automatically. The default setting is "No".
Virtual Account Group
Avaya Mode
If set to "Yes", when processing SIP Register 3XX Response, it will parse
the address site in 3XX, modify the account server info "SIP Server: port"
& "SIP Transaction" in virtual account group and initiate registration again.
This feature is designed for the Avaya customers.
Number of Accounts The
Virtual Account Group
Register
Configures the number of concurrent the main server and standby server
register, each account supports 1 SIP main server and 4 standby server
registration.
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Filter Characters
Sets the characters for filter when dial out numbers. Users could set up
multiple characters. For example, if set to “[()-]”, when dial (0571)-8800-
8888, the character “()-“ will be automatically filtered and dial
057188008888 directly.
Escape # as %23 in SIP
URI
Determines which characters will be included in the SIP INVITE URI if end
users input #. If it is set to “Yes”, the phone system will replace the #
by %23. Otherwise, it will include # in the SIP INVITE message.
The default setting is “Yes”.
Use 3
rd
Party App as
Basic Phone
Enter the package and activity of the 3rd party app that substitutes GS
phone application, separated by "/". e.g.:
com.broadsoft.ucone.androidtablet/com.broadsoft.android.common.activit
y.LauncherActivity.
After input, when the user offhook or click the phone application, it will
automatically enable the configured app to enter the corresponding
interface.
Default setting is blank, which means use GS phone application.
Record Mode
Configures phone recording mode.
If set to "Record locally", then the phone will use the local tape recorder
for call recording, and the audio file will be saved in accordance with the
tape recorder setup.
If set to "Record on PortaOne", then will send the specified SIP
messages to the corresponding server.
If set to "Record on UCM", then will send the recording feature code to
the UCM server to request for recording, and the recording function will
be executed by the server.
If set to "Record on Broadsoft", then will send the recording feature
code to the Broadsoft server to request for recording, and the recording
function will be executed by the server.
If set to “Disable”, recording functionality will be disabled during calls.
Enable Auto Record
When Call Established
Configures whether to auto record a call. If enabled, all recordings will start
automatically after the call is established.
Rejected Call
Notification
Specifies whether to enable rejected call notification. Once enabled, a
missed call will prompt on LCD when rejecting an incoming call, and the
rejected call will be saved in Call History Missed numbers and missed call
prompt will be displayed.
Return Code When
Refusing Incoming Call
When refusing the incoming call, the phone will send the selected type of
SIP message of the call.
Return Code When
Enable DND
When DND is enabled, the phone will send the selected type of SIP
message.
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Group Listen with
Speake
Configures whether to enable speaker sound when in a call with
handset/headset. If set to "Yes", the speaker sound is enabled, but you
cannot talk through the speaker.
One-way Call Function
Buttons
Users can configure and customize the call functions in the call interface
during a normal call. The available call functions are: Mute, Hold, Video,
Record, Transfer, Call Details, New Call, Keyboard, Media Channel,
Conference.
Note: Users can choose only 3 of these functions to be shown on the call
interface during a one-way call.
Conference Call
Function Buttons
Users can configure and customize the call functions in the call interface
during a conference. The available call functions are: Invite, Mute, Hold,
Record, Call Details, New Call, Keyboard, Media Channel.
Note: Users can choose only 3 of these functions to be shown on the call
interface during a conference.
Call Function Button
Display Timeout (s)
Configures the timeout period for call function buttons display. If set to "0",
the buttons will be always displayed. The value range is 0-30 seconds.
Phone Settings/Ring Tone
Auto Config CPT by
Region
Configures whether to choose Call Progress Tone automatically by region.
If set to "Yes", the phone will configure CPT (Call Progress Tone)
according to different regions automatically. If set to "No", you can
manually configure CPT parameters. The default setting is "No".
Call Progress Tones:
Dial Tone
Second Dial Tone
Ring Back Tone
Busy Tone
Reorder Tone
Confirmation Tone
Call-Waiting Tone
Configures tone frequencies according to user preference. By default, the
tones are set to North American frequencies. Frequencies should be
configured with known values to avoid uncomfortable high pitch sounds.
Syntax: f1=val,f2=val [,c=on1/off1[-on2/off2[-on3/off3]]];
(Frequencies are in Hz and cadence on and off are in 10ms)
ON is the period of ringing ("On time" in "ms") while OFF is the period of
silence. 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 repeats the pattern.
Please refer to the document below to determine your local call progress
tones: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
Call-Waiting Tone Gain
Adjusts the call waiting tone volume. Users can select "Low", "Medium" or
"High". The default setting is "Low".
Default Ring Cadence
Defines the ring cadence for the phone. The default setting is
c=2000/4000.
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Phone Settings/Video Settings
Video Frame Rate
Configures video frame rate for SIP video call from “5 frames/second”, “15
frames/second”, “25 frames/second” and “30 frames/second”. The default
setting is 15 frames/second. The video frame rate is adjustable based on
network conditions. Increasing the frame rate will significantly increase the
amount of data transmitted, therefore consuming more bandwidth. The
video quality will be affected due to packet loss if extra bandwidth is not
allocated.
Video Display Mode
Sets the video display mode to "Original proportion", "Cut proportionally"
or "Add black margin proportionally”.
If set to "Original proportion", the phone displays video in its
original proportion. If the video display proportion is different from
the one of the phones, the phone will stretch or compress video to
display it.
If set to "Cut proportionallythe phone will cut video to meet its
own display proportion.
If set to "Add black margin proportionally", the phone will
display video in its original proportion. If it still exists spare space,
the phone will add black edge on it.
Enable Frame Skipping
in Video Decoder
Enables the phone system for frame skipping in video decoder. If it is
enabled, the video decoder will skip the P frame and start decoding from
the next I frame. Enabling this option will help reduce flickering in the video
when the bandwidth is limited in the network environment. The default
setting is "Yes".
Phone Settings/Multicast Paging
Multicast Paging
Paging Barge
Sets the threshold of paging calls. If the paging call’s priority is higher than
the threshold, the existing call will be hold and the paging call will be
answered. Otherwise, the existing call does not be affected. If it is set to
Disable, any paging call will not be answered. The default setting is
“Disable”.
Paging Priority Active
Determines if a new paging call whose priority is higher than the existing
paging call will be answered. If it is checked, this feature will be enabled.
The default setting is disabled.
Multicast Paging Codec
Selects the codec type for the multicast paging call. This list includes
PCMU, PCMA, G726-32, G722, and G729A/B, iLBC, Opus.
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Enable Multicast Paging
Video
Enables the video feature to establish a multicast paging call. The default
setting is disabled.
Multicast Paging Video
Codec
Sets the video codec for the multicast paging call. The default setting is
“H.264”.
Multicast Paging Image
Size
Sets the video image size for the multicast paging call. This list includes
720P, 4CIF, VGA, CIF, QVGA, and QCIF.
The default setting is VGA.
Multicast Paging Video
Bit Rate
Determines the video bit rate for the multicast paging call. The default
setting is “256 kbps”.
Multicast Paging Video
Frame Rate
Configures the video frame rate for the multicast paging call. This list
includes “15 frames/second”, 25 frames/second”, “30 frames/second”, and
“Variable frames rate”.
Multicast Paging H.264
Profile Type
Specifies the H.264 codec profile type for the multicast paging call. This list
includes “Baseline Profile”, “Main Profile”, and “High Profile”.
Note: Lower profile is easier to decode, while higher profile has higher
compress rate. Usually, use Baseline profile for low CPU Performance
device, and choose high profile for video conference.
Multicast Paging H.264
Payload Type
Determines the H.264 codec payload type for the multicast paging call.
The default setting is “99”.
Multicast Listening
Priority
Listening Address
Label
Configures the IP address and port number for monitoring multicast paging
call. When the initiator initiates a call, answer the call, and display listening
address and tag of the monitoring target. This feature supports Video
Multicast, when the initiator initiates Video Multicast, it will automatically
add the number 2 on the port of the listening address. Reboot the phone
to make changes take effect.
The valid IP address range is from 224.0.0.0 to 239.255.255.255. Users
may also fill the label for each listening address corresponding to priority.
Network Settings Page Definitions
Network Settings/Ethernet Settings
Ethernet Settings
Preferred Internet
Protocol
If IPv4 is selected, the phone will be using IPv4 addressing, otherwise, it
will be using IPv6 addressing.
Default is Prefer IPv4.
Different Networks for
Data and VoIP Calls
Configures whether to set up different networks for the phone data and the
call. If set to "Yes", you need to configure the data network and VoIP
network, respectively.
Note: This setting needs reboot to take effect.
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Network Configuration of Data
IPv4
IPv4 Address Type
Users could select "DHCP", "Static IP" or "PPPoE".
DHCP: Obtain IP address via a DHCP server in the LAN. All domain
values for static IP/PPPoE are unavailable, even though the values have
been saved in the flash.
PPPoE: Configures PPPoE account/password. Obtain the IP address
from the PPPoE server via dialing.
Static IP: Manually configures IP Address, Subnet Mask, Default
Router's IP Address, DNS Server 1, and DNS Server 2.
By default, it is set to "DHCP".
DHCP VLAN Override
Selects the DHCP Option VLAN mode. When set to "DHCP Option 132
and DHCP option 133", the phone will get DHCP option 132 and 133 as
VLAN ID and VLAN priority. When set to "Encapsulated in DHCP Option
43", the phone will get values from Option 43 which encapsulate VLAN ID
and VLAN priority.
Note: Please make sure the "Allow DHCP Option 43 and Option 66 to
Override Server" setting under maintenanceupgrade is checked. The
default setting is " DHCP Option 132 and DHCP option 133 ".
Host name (Option 12)
Sets the name of the client in the DHCP request. It is optional but may be
required by some Internet Service Providers.
Vendor Class ID
(Option 60)
Configures the vendor class ID header in the DHCP request.
The default setting is Grandstream GXV3370.
IP Address
Defines the phone’s static IP address if the static IP is used.
Subnet Mask
Determines the network’s subnet mask if the static IP is used.
Default Gateway
Defines the network’s gateway address if the static IP is used.
DNS Server 1
Configures the primary DNS IP address if the static IP is used.
DNS Server 2
Configures the secondary DNS IP address if the static IP is used.
PPPoE Account ID
Configures the PPPoE account ID if the PPPoE is used.
PPPoE Password
Sets the PPPoE password if the PPPoE is used.
Layer 2 QoS
802.1Q/VLAN Tag
(Ethernet)
Assigns the VLAN Tag of the Layer 2 QoS packets for Ethernet.
The Default value is 0.
Note: When Different Networks for Data and VoIP Calls set to Yes, then
this option will be applied for Data.
Layer 2 QoS 802.1p
Priority Value (Ethernet)
Assigns the priority value of the Layer 2 QoS packets for Ethernet.
The Default value is 0. The setting needs reboot to take effect.
Note: When Different Networks for Data and VoIP Calls set to Yes, then
this option will be applied for Data.
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IPv6
IPv6 Address
Configures the appropriate network settings on the phone. Users could
select from "Auto-configured" or “Statically configured”.
Static IPv6 Address
Enter the static IPv6 address in "Statically configured" IPv6 address type.
IPv6 Prefix Length
Enter the IPv6 prefix length in "Statically configured" IPv6 address type.
Default is 64.
DNS Server 1
Enter DNS Server 1 when static IP is used.
DNS Server 2
Enter DNS Server 2 when static IP is used.
Preferred DNS Server
Enter the Preferred DNS server.
Network Configuration of VoIP Calls
IPv4
IPv4 Address Type
Users could select "DHCP" or "Static IP".
DHCP: Obtain IP address via a DHCP server in the LAN. All domain
values for static IP/PPPoE are unavailable, even though the values have
been saved in the flash.
Static IP: Manually configures IP Address, Subnet Mask, Default
Router's IP Address, DNS Server 1 and DNS Server 2.
By default, it is set to "DHCP".
IP Address
Defines the phone’s static IP address if the static IP is used.
Subnet Mask
Determines the network’s subnet mask if the static IP is used.
Default Gateway
Defines the network’s gateway address if the static IP is used.
DNS Server 1
Configures the primary DNS IP address if the static IP is used.
DNS Server 2
Configures the secondary DNS IP address if the static IP is used.
Layer 2 QoS
802.1Q/VLAN Tag
(Ethernet) for VoIP Calls
Assigns VLAN Tag of Layer 2 QoS packets (Ethernet) for VoIP. The default
value is 0.
Layer 2 QoS 802.1p
Priority Value (Ethernet)
for VoIP Calls
Assigns the priority value of Layer 2 QoS packets (Ethernet) for VoIP Calls.
The default value is 0. The setting needs reboot to take effect.
802.1x Mode
802.1x mode
Enables and selects the 802.1x mode for the phone system. The supported
802.1x modes are: EAP-MD5, EAP-TLS, EAP-PEAP
The default setting is "Disable".
802.1x Identity
Enters the identity information for the selected 802.1x mode. (This setting
will be displayed only if 802.1 X mode is enabled).
802.1x Secret
Enters the secret for the 802.1x mode. This option will appear when 802.1x
mode is EAP-MD5 or EAP-PEAP.
CA Certificate
Uploads the CA Certificate file to the phone. (This setting will be displayed
only if the 802.1 X mode is enabled)
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Client Certificate
Loads the Client Certificate file to the phone. (This setting will be displayed
only if the 802.1 X TLS mode is enabled)
Private Key
Loads the private key file to the phone. (This setting will be displayed only
if the 802.1 X TLS mode is enabled)
Network Settings/Wi-Fi Settings
Wi-Fi Basics
Wi-Fi Function
Enables/disables the Wi-Fi feature.
The default setting is "Disable".
Wi-Fi Band
Configures the Wi-Fi frequency band from the list:
2.4G
5G
2.4 G & 5G
Default setting is 2.4G & 5G.
ESSID
Permits to scan and select the available Wi-Fi networks within the range if
the Wi-Fi feature is enabled. Click on "Select" to select the Wi-Fi network
to connect to. The ESSID will be auto filled in the ESSID filed.
Add Network
ESSID
Configure the hidden ESSID name.
Security Mode for
Hidden SSID
Defines the security mode used for the wireless network when the SSID is
hidden. Default is “None”.
Password
Determines the password for the selected Wi-Fi network.
Advanced Settings
Layer 2 QoS 802.1p
Priority Value (Wi-Fi)
Assigns the priority value of the Layer 2 QoS packets for Wi-Fi.
The Default value is 0.
Country Code
Configure Wi-Fi country code. The default value is "United States of
America".
Note: Reboot is required to take effect.
Note: Since firmware version 1.0.3.36, Wi-Fi settings can be restricted from WEB UI as well as from LCD
screen. This can only be configured via config template. When P22038=0, the Wi-Fi settings are neither
accessible from web UI nor from LCD. To enable Wi-Fi on web UI and LCD screen, use P22038=1.
Network Settings/OpenVPN® Settings
Enable OpenVPN®
This enables/disables OpenVPN® functionality and requires the user to
have access to an OpenVPN® server. The default setting is No. NOTE: To
use OpenVPN® functionalities, users must enable OpenVPN® and
configure all the settings related to OpenVPN®, including server address,
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port, OpenVPN® CA, certificate, and key. Additionally, the user must also
set the SIP account to use "VPN" for the "Nat Traversal" (under Account->
Network Settings).
Enable OpenVPN®
Comp-lzo
Enables OpenVPN® LZO compression. When the LZO compression is
enabled on the OpenVPN® server, you must turn on it at the same time.
Otherwise, the network will fail to connect.
OpenVPN® Server
Address
The URL/IP address for the OpenVPN® server.
OpenVPN® Port
The network port for the OpenVPN® server. By default, it is set to 1194.
OpenVPN® Transport
Determines network protocol (UDP or TCP) used for OpenVPN®.
Default is UDP.
OpenVPN® CA
OpenVPN® CA file (ca.crt) required by the OpenVPN® server for
authentication purposes. Press "Upload" to upload the corresponding file
to the device. The file will not be uploaded if it is not in the correct format.
OpenVPN® Client
Certificate
OpenVPN® Client certificate file (*.crt) required by the OpenVPN® server
for authentication purposes. Press "Upload" to upload the corresponding
file to the device. The file will not be uploaded if it is not in the correct
format.
OpenVPN® Client Key
The OpenVPN® Client key (*.key) required by the OpenVPN® server for
authentication purposes. Press "Upload" to upload the corresponding file
to the device. The file will not be uploaded if it is not in the correct format.
OpenVPN® Cipher
Method
The cipher method of OpenVPN®, must be the same cipher method used
by the OpenVPN® server. Available options are: “Blowfish”, “AES-128” or
“AES-256”. Default is “Blowfish”.
OpenVPN® Username
OpenVPN® authentication username (optional).
OpenVPN® Password
OpenVPN® authentication username (optional).
Network Settings/Advanced Network Settings
Advanced Network Settings
DNS Refresh Timer (m)
Configures the refresh time (in minutes) for DNS query. If set to "0", the
phone will use the DNS query TTL from DNS server response.
Default value is 0.
DNS Failure Cache
Duration (m)
Configures the duration (in minutes) of previous DNS cache when DNS
query fails. If set to "0", the feature will be disabled. Note: Only valid for
SIP registration.
Default value is 0.
IPv4 Preferred DNS1
Server
This fields sets the preferred DNS server for the user.
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IPv4 Preferred DNS2
Server
This fields sets the second preferred DNS server for the user.
Enable LLDP
Enables the LLDP (Link Layer Discovery Protocol) feature on the phone
system. If it is set to “Yes”, the phone system will broadcast LLDP PDU to
advertise its identity and capabilities and receive same from a physical
adjacent layer 2 peer.
The default setting is "Yes".
LLDP TX Interval (s)
Configures the interval the phone sends LLLD-MED packet.
The default setting is 30s.
Enable CDP
Configures whether to enable CDP to receive and/or transmit information
from/to CDP-enabled devices.
The default setting is "Yes".
Layer 3 QoS for SIP
Defines the Layer 3 packet’s QoS parameter for SIP messages in decimal
pattern. This value is used for IP Precedence, Diff-Serv or MPLS.
The default setting is 26 it is equivalent to the DSCP name constant AF31.
Layer 3 QoS for Audio
Defines the Layer 3 packet’s QoS parameter for RTP messages in decimal
pattern. This value is used for IP Precedence, Diff-Serv or MPLS.
The default setting is 46 it is equivalent to the DSCP name constant EF.
Layer 3 QoS for Video
Defines the Layer 3 packet’s QoS parameters for H.264 messages in
decimal pattern. This value is used for IP Precedence, Diff-Serv or MPLS.
The default setting is 34 it is equivalent to the DSCP name constant AF41.
HTTP/HTTPS
User-Agent
Sets the user-agent for phonebook and screen saver.
SIP User-Agent
Sets the user-agent for SIP.
Separate Ethernet & Wi-
Fi Traffic
Configures whether to separate Ethernet & Wi-Fi traffic. If enabled, VoIP
calls are forced to use Ethernet, and other application data preferentially
use Wi-Fi.
Maximum Transmission
Unit (MTU)
Configures the MTU in bytes. The MTU must be set according to the
needs. The range is form 576 to 1500.
Note: If MTU is set to less than 1280, IPv6 may not take effect.
PC Port Mode
PC Port Mode
Enables and defines the PC port mode. If it is set to “Mirrored”, the traffic
in the LAN port will go through PC port as well and packets can be captured
by connecting a PC to the PC port. The default setting is "Enabled".
PC Port VLAN Tag
Defines the VLAN Identifier of the Layer 2 frame for PC port. This adds the
VLAN tag value on the target address received from the LAN port of the
phone then sends the value to the device connected to the PC port.
Note: VLAN tag value on the device connected to the PC port should be
the same as the VLAN tag value assigned to the PC port here.
PC Port Priority Value
Determines the Priority Code Point within a Layer 2 frame header for PC
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port.
Proxy
HTTP/HTTPS Proxy
Hostname
Configures the HTTP/HTTPS proxy URI of the network. Some of networks
requires going through a proxy to access to the Internet. The default setting
is keeping this field blank.
HTTP/HTTPS Proxy Port
Configures the HTTP/HTTPS proxy port number of the network. Some of
networks requires going through a proxy to access to the Internet. The
default setting is keeping this field blank.
Bypass Proxy For
Defines the specific URI that the phone can directly access to without
HTTP/HTTPS proxy. If it is filled, the phone will bypass the proxy to send
the packets to the specific URI. The default setting is filed blank.
Network Settings/Affinity Settings
Affinity Support
Configures whether to enable the Affinity feature. Note: The Affinity CTI
function can only be used when the phone has a registered account.
This option is disabled by default.
Preferred Account
Selects SIP account for Affinity.
System Settings Page Definitions
System Settings/Time and Language
Time Settings
Assign NTP Server
Address
Defines the URL or IP address of the NTP server. The phone may obtain
the current date and time information from the server. The default setting
is “pool.ntp.org”.
DHCP Option 42
override NTP server
Obtains NTP server address from a DHCP server using DHCP Option 42;
it will override configured NTP Server. If set to No”, the phone will use
configured NTP server to synchronize time and date even if an NTP server
is provided by DHCP server. The default setting is "Yes".
DHCP Option 2 to
override Time Zone
setting
Obtains time zone setting (offset) from a DHCP server using DHCP Option
2; it will override selected time zone.
If set to “No”, the phone will use selected time zone even if provided by
DHCP server. The default setting is Yes.
Time Zone
Specifies the local time zone for the phone. It covers the global time zones
and user can selected the specific one from the drop-down list.
Self-defined Time Zone
This parameter allows the users to define their own time zone.
The syntax is: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0
MTZ+6MDT+5
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This indicates a time zone with 6 hours offset with 1 hour ahead (when
daylight saving) which is U.S central time. If it is positive (+) if the local time
zone is west of the Prime Meridian (A.K.A: International or Greenwich
Meridian) and negative (-) if it is east.
M4.1.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)
The 2nd number indicates the nth iteration of the weekday: (1
st
Sunday, 3
rd
Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, ,Sat)
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
Time Display Format
Specifies which format will be used to display the time. It can be selected
from 12-hour and 24-hour format.
Date Display Format
Determines which format will be used to display the date.
It can be selected from the drop-down list.
Normal (M/DD/YYYY): 1/31/2012
YYYY/MM/DD: 2012/01/31
MM/DD/YYYY: 01/31/2012
DD/MM/YYYY: 31/01/2012
The default setting is MM/DD/YYYY.
Language
Language
Sets the language to display on the phone's LCD.
System Settings/Security Settings
Web/SSH Access
Enable SSH
Enables SSH access to the phone. Default setting is “Yes”.
SSH Port
Customizes SSH port. Valid range: 1 65535. Default is 22.
Access Method
Determines which protocol will be used to access the phone ‘s Web GUI.
It can be selected from HTTP and HTTPS. The default setting is HTTP.
Port
Specifies which port to use to access the phone ‘s Web UI. By default, if
HTTP, the port number will be 80; if HTTPS is selected, the port number
will be 443.
Web Access Control
Configures Web access control by using Whitelist or Blacklist on incoming
IP addresses.
Default setting is None.
Web Access Control List
Only allow the list of IP addresses as Whitelist, or restrict the list of IP
addresses as Blacklist to access Web.
Configuration via
Keypad Menu
Configures access control for keypad Menu settings on the Settings
interface of the phone.
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Unrestricted: configure all settings on the Settings interface;
Basic Settings Only: The Advanced Settings option will not be
displayed;
Basic Settings & Network Settings: Only the Advanced Settings
option will not be displayed
Constraint Mode: users need to input admin user password to
configure Wireless & Network and Advanced Settings.
Note: When access control for keypad is limited to “Basic Settings Only”
or “Constraint Mode”, the Admin authentication will be mandatory to start
Factory Reset process.
Permission to
Install/Uninstall Apps
Configures the permissions for users to install/uninstall the applications.
If set to "Allow", the user is free to install/uninstall third-party apps.
If set to "Require admin password", the user need to input the
correct administrator password to install/uninstall third-party apps.
If set to "Require admin password if the app source is unknown",
the user need to input admin password only when install apps from
unknown source, administrator password authentication is required
when the user uninstall third-party apps.
If set to "Not allow", the user cannot install/uninstall third-party apps.
User Info Management
Current Admin
Password
Enter current logged-in user's password. This field is case sensitive. The
default password is "admin".
New Admin Password
Allows the user to change the admin password. The password field is
purposely blank after clicking the “Save” button for security purpose. This
field is case sensitive with a maximum length of 32 characters.
Confirm Admin
Password
Enter the new Admin password again to confirm.
New User Password
Allows the administrator to set the password for user-level web GUI
access. This field is case sensitive with a maximum length of 32
characters. The default password is “123”.
Confirm New User
Password
Enter the new User password again to confirm.
TLS
Minimum TLS Version
Configures the minimum TLS version supported by the phone.
Maximum TLS Version
Configures the maximum TLS version supported by the phone.
Enable Weak TLS Cipher
Suite
Defines the function for weak TLS cipher suites. If set to "Enable Weak
TLS Cipher Suites", allow users to encrypt data by weak TLS cipher suites.
If set to "Disable Symmetric Encryption RC4/DES/3DES", allow users to
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disable weak cipher DES/3DES and RC4.
SIP TLS
SIP TLS Certificate
Defines the SSL certificate used for SIP over TLS.
SIP TLS Private Key
Defines the SSL Private key used for SIP over TLS.
SIP TLS Private Key
Password
Defines the SSL Private key password used for SIP over TLS.
Certificate Management
CA Certificate
Import Trusted CA
Certificates
Allows to upload the CA Certificate file to phone.
Trusted CA Certificates
Lists trusted CA certificates previously uploaded. Administrator can delete
a certificate from here.
User Certificate
Add Certificate
Allows to upload & Install User Certificate file to phone.
Custom Certificate
Import Custom
Certificate
Allows to upload a Custom Certificate file to phone.
Custom Certificate
Lists trusted Custom Certificate previously uploaded. Administrator can
delete a certificate from here.
System Settings/Preferences
LCD & LED Management
Enable Missed Call
Backlight
If set to "Yes", LCD backlight will be turned on when there is a missed call
on the phone. The default setting is "Yes".
Note: Reboot is required for the setting to take effect
Enable Missed Call
Indicator
If set to "Yes", the LED indicator on the upper right corner of the phone will
light up when there is missed call on the phone. Default setting is "Yes".
Enable MWI Indicator
If set to "Yes", the LED indicator on the upper right corner of the phone will
light up when there is new voicemail on the phone. If it set to No”, the LED
indicator will keep off if the phone system receives SIP NOTIFY message
about unread voice mail. The default setting is "Yes".
Enable New Message
Indicator
If set to "Yes", the LED indicator on the upper right corner of the phone will
light up when there is new message on the phone. Default setting is "Yes".
Enable Contact Full
Indicator
If set to "Yes", the LED indicator on the upper right corner of the phone will
light up when the contact storage or message storage is full. The default
setting is "Yes".
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Enable Indicator When
LCD is Off
If set to "Yes", the LED indicator on the upper right corner of the phone will
light up when the LCD screen is off. If it set to No”, the LED indicator will
keep off when the LCD screen is off. The default setting is "Yes".
Screen Timeout
Configures the timeout interval of the LCD backlight. If set to "never", the
screen will always stay on. The default value is 3 minutes.
Screensaver Timeout
Configures the screensaver timeout. The default value is 2 minutes
Audio Control
RJ9 Headset TX Gain
(dB)
Configures the Transmission Gain in RJ9 headset channel. It can be
selected from the dropdown list. The default setting is 0dB:
-24
-18
-12
-6
0
+6
+12
+18
+24
RJ9 Headset RX Gain
(dB)
Configures the Receive Gain in RJ9 headset channel. It can be selected
from the dropdown list. The default setting is 0dB:
-9
-6
0
+6
+9
3.5mm Earphone TX
Gain (dB)
Set the Transmission Gain in 3.5mm earphone headset channel. It can be
selected from the dropdown list. The default setting is 0dB:
-24
-18
-12
-6
0
+6
+12
+18
+24
Headset Type
Specifies which type of headset will be connected to the phone system. It
can be selected from the dropdown list:
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Normal Headset
Plantronics EHS
If a normal RJ11 headset is connected, it should set to “Normal Headset”.
If a Plantronics EHS headset is used, it should set to “Plantronics EHS”.
Enable 3.5mm Headset
Control
If set to "Yes", the headset can control the Onhook and Offhook. The
default setting is "No".
Handset TX Gain (dB)
Configures the transmission gain of the handset. Default setting is “0dB”.
0
-4
-2
+2
+4
+6
Handsfree TX Gain (dB)
This feature configures the transmission gain for handsfree mode. Default
setting is “0dB”.
0
-16
-12
-8
-6
-4
0
+4
+6
+8
+12
+16
Adjust volume
Configures the volume of ringtone.
Media volume
Configures the volume of media.
Alarm volume
Configures the volume of Alarm.
System Settings/TR-069
Enable TR-069
Sets the phone system to enable the “CPE WAN Management Protocol”
(TR-069).
The default setting is "No".
ACS URL
Specifies URL of TR-069 ACS (e.g., http://acs.mycompany.com), or IP
address.
ACS Username
Enters username to authenticate to ACS.
ACS Password
Enters password to authenticate to ACS.
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Periodic Inform Enable
Sends periodic inform packets to ACS. Default is “No”.
Periodic Inform Interval
(s)
Configures to sends periodic “Inform” packets to ACS based on specified
interval.
Connection Request
Username
Enters username for the ACS to connect to the phone.
Connection Request
Password
Enters password for the ACS to connect to the phone.
Connection Request
Port
Enters the port for the ACS to connect to the phone.
CPE Cert File
Uploads Cert File for the phone to connect to the ACS via SSL.
CPE Cert Key
Uploads Cert Key for the phone to connect to the ACS via SSL.
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Maintenance Page Definitions
Maintenance/Upgrade
Firmware
Upgrade via Manual Upload
Complete Upgrade
If enabled, all files will be replaced except user data. Default is disabled.
Upload Firmware File to
Update
Allows users to load the local firmware to the phone to update the firmware.
Upgrade via Network
Firmware Upgrade Mode
Allows users to choose the firmware upgrade method: TFTP, HTTP,
HTTPS, or Manual Upload. The default setting is “HTTP”.
Firmware Server Path
Sets IP address or domain name of firmware server. The URL of the server
that hosts the firmware release. Default is “fm.grandstream.com/gs”.
Administrators are now able to configure variables in the provisioning
server URL. Currently, the following variables are supported in the
provisioning server URL:
$PN: is used to identify the directory name of the provisioning server
directory where the corresponding boot files and configuration files are
located.
$MAC: is used to identify the MAC address of the IP phone.
Variables $PN and $MAC can be embedded in server URL setting in Web
UI and in DHCP Option 66.
Example (Web UI): 192.168.0.2/$PN/$MAC
Example (DHCP Option 66): tftp://192.168.0.2/$PN/$MAC
$PN will be replaced with phone model, e.g., GXV3370 $MAC will be
replaced with phone's MAC address, e.g., 000b829a8ffe
HTTP/HTTPS Username
Enters the username for the firmware HTTP/HTTPS server.
HTTP/HTTPS Password
Enters the password for the firmware HTTP/HTTPS server.
Firmware File Prefix
Checks if firmware file is with matching prefix before downloading it.
This field enables user to store different versions of firmware files in one
directory on the firmware server.
Firmware File Postfix
Checks if firmware file is with matching postfix before downloading it. This
field enables user to store different versions of firmware files in one
directory on the firmware server.
Firmware Upgrade
Click the "Update" button to check whether the firmware in the firmware
server has an updated version, if so, update immediately.
Config File
Configure Manually
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Download Device
Configuration
Click to download the device configuration file in .txt format.
Upload Device
Configuration
Upload configuration file to the phone.
Configure via Network
Use Grandstream GAPS
It is used to configure the download path and update mode for the
configuration file server.
If set to "Yes", the device will set the download path of the
configuration file to “fm.grandstream.com/gs” by default and use
HTTPS protocol to connect to the server.
If set to "No", then users can manually configure the path and update
mode for the configuration file server.
Config Upgrade Via
Selects provisioning method: TFTP, HTTP or HTTPS.
Default setting is “HTTPS”.
Config Server Path
Sets IP address or domain name of configuration server. The server hosts
a copy of the configuration file to be installed on the phone. Default is
“fm.grandstream.com/gs”.
Administrators are now able to configure variables in the provisioning
server URL. Currently, the following variables are supported in the
provisioning server URL:
$PN: is used to identify the directory name of the provisioning server
directory where the corresponding boot files and configuration files are
located.
• $MAC: is used to identify the MAC address of the IP phone.
Variables $PN and $MAC can be embedded in server URL setting in Web
UI and in DHCP Option 66.
Example (Web UI): 192.168.0.2/$PN/$MAC
Example (DHCP Option 66): tftp://192.168.0.2/$PN/$MAC
$PN will be replaced with phone model, e.g., GXV3370 $MAC will be
replaced with phone's MAC address, e.g., 000b829a8ffe
HTTP/HTTPS Username
Configures the username for the config HTTP/HTTPS server.
HTTP/HTTPS Password
Configures the password for the config HTTP/HTTPS server.
Always Perform HTTP
Basic Authentication
Configures whether to send basic HTTP authentication information to the
server when downloading firmware, config file or GUI customization file. If
enabled, the phone will always send HTTP/HTTPS username and
password even if the server didn't request authentication. If disabled, the
phone will only send HTTP/HTTPS username and password when the
server requires authentication.
Default is No.
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Config File Prefix
Checks if configuration files are with matching prefix before downloading
them.
This field enables user to store different configuration files in one directory
on the provisioning server.
Config File Postfix
Checks if configuration files are with matching postfix before downloading
them. This field enables user to store different configuration files in one
directory on the provisioning server.
Authenticate Conf File
Sets the phone system to authenticate configuration file before applying it.
When set to “Yes”, the configuration file must include value P1 with phone
system’s administration password. If it is missed or does not match the
password, the phone system will not apply it. The default setting is “No”.
XML Config File
Password
Decrypts XML configuration file when encrypted. The password used for
encrypting the XML configuration file is using OpenSSL.
Download Device
Configuration
Downloads the phone's configuration file in text format. The config file
includes all the P value parameters for phone's current settings except
password for security purpose. Users can use the Grandstream
configuration file generator to generate binary config file from this text file.
Upload Device
Configuration
Uploads configuration file to the phone. The file will not be uploaded if it is
not in the correct format.
Note: The GXV3370 supports only txt format for config file upload.
Start Provision
Press to trigger the device to fetch configuration file from server configured
under the Config server Path. The same button is located on LCD screen
under Settings System Update.
GUI Customization File
Configure Manually
Upload GUI
Customization File
Uploads GUI customization file from web UI manually.
Configure via Network
GUI Customization File
Download Mode
Selects download method: TFTP, HTTP or HTTPS.
Default setting is “HTTPS”.
GUI Customization File
URL
Sets IP address or domain name of the GUI customization file server. The
server hosts a copy of the file to be installed on the phone. The Default
setting is fm.grandstream.com/gs.
GUI Customization File
HTTP/HTTPS Username
Enters the username for the firmware HTTP/HTTPS server.
GUI Customization File
HTTP/HTTPS Password
Enters the password for the firmware HTTP/HTTPS server.
Use Configurations of
Retrieve and download customization file with the configuration of the
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Config File Server
config file.
Provision
Automatic Upgrade
Automatic Upgrade
Specifies when the firmware upgrade process will be initiated; there are 4
options:
No: The phone will only do upgrade once at boot up.
Check every day: User needs to specify “Hour of the day (0-23)”.
Check every week: User needs to specify “Hour of the day (0-23)”
and “Day of the week (0-6)”.
Check at a period Time: User needs to specify “Hour of the day
(0-23)”
Note: Day of week is starting from Sunday.
The default setting is “No”.
Enable Randomized
Automatic Upgrade
Configures whether the phone will upgrade automatically at random time
point within the configured period. This option is used for multiple phones
upgrade at the same time.
Automatic Upgrade
Check Interval (m)
Configures how the phone system will check the server for new firmware
and configuration file downloading.
It only valid if the user selects “Check at a period of time” in the “Automatic
Upgrade”. Default setting is 10080 (namely 7 days).
Starting Ending Hour of
the Day (0-23)
Defines at which hour of the day they phone system will check the
HTTP/HTTPS/TFTP server for firmware upgrades or configuration files
changes.
Day of the Week
Defines which day of the week the phone system will check the
HTTP/HTTPS/TFTP server for firmware upgrades or configuration files
changes.
Firmware Upgrade and
Provisioning
Defines the phone system’s rules for automatic upgrade. It can be selected
from:
Always Check at bootup
Always Check at bootup, when F/W pre/suffix changes,
Skip the Firmware Check.
The default setting is “Always Check at bootup”.
Upgrade with Prompt
If set to "No", the phone will automatically start upgrading after
downloading the firmware file. Otherwise, users would need to confirm in
the prompted message on the LCD screen to start upgrading process.
The default setting is "Yes".
Config Provision
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Download and Process
All Available Config
Files
By default, the device will provision the first available config in the order of
cfgMAC, cfgMAC.xml, cfgMODEL.xml, and cfg.xml (corresponding to
device specific, model specific, and global configs). If set to "Yes", the
device will download and apply (override) all available configs in the order
of cfgMAC, cfg.xml, cfgMODEL.xml, cfgMAC.xml.
Default Setting is “No”.
Config Provision
Device will download the configuration files and provision by the configured
order.
DHCP Option
Allow DHCP Option 43,
160 and 66 Override
Server
If DHCP option 43, 160 and 66 is enabled on the LAN side, the device will
reset the CPE, upgrade, network VLAN tag, and priority configuration
according to option 43 sent by the server. At the same time, the update
mode and server path of the configuration upgrade mode will be reset
according to the option 160 and 66 sent by the server.
Default is "Yes".
DHCP Option 120
Override SIP Server
Configures the phone system to allow the DHCP offer message to override
the Config Server Path via the Option 120 header.
Default setting is "Yes".
Allow DHCP Option 242
(Avaya IP Phones)
Enables DHCP Option 242. Once enabled, the phone will use the
configuration info issued by the local DHCP in Option 242 to configure
proxy, transport protocol and server path.
The default setting is “Yes”.
PNP Feature
Enable PNP Feature
Enables the PNP (Plug and Play) feature on the device. If it is enabled, the
device will be set as a provision server to send SIP NOTIFY message
including the provision URL to response the client phone’s SIP
SUBSCRIBE request.
This feature will be enabled if the PNP URL is configured. If this setting is
enabled, the 3CX Auto provision will be disabled automatically.
PNP URL
Configures the URL to provision another client phone’s config server path.
The URL will be included in the SIP NOTIFY message.
PnP(3CX) Auto
Provision
Sets the phone system to broadcast the SIP SUBSCRIBE message during
booting up to allow itself to be discovered and be configured by the SIP
platform. The default setting is "Yes".
Advanced Settings
Enable SIP NOTIFY
Authentication
Enables the phone to challenge SIP NOTIFY with 401.
The default setting is “Yes”.
Validate Certification
Chain
Configures whether to validate the server certificate when download the
firmware/config file. If it is set to "Yes", the phone will download the
firmware/config file only from the legitimate server. Default setting is "No".
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Enable EEE Mode
Enable/disable EEE mode. If set to "Yes", the phone will turn on the EEE
mode. Note: Regardless of whether the EEE mode is turned on or off, the
network will reconnect.
mDNS Override Server
Sets the phone system to broadcast the Multicast DNS (mDNS) message
during booting up to allow itself to be discovered and be configured by the
SIP platform.
If it is set to “User Type A’, the phone system will broadcast the MDNS
message “A_grandstream-cfg.local”;
if it is set to “Use Type SRV”, the MDNS message will be
“SRV_grandstream-cfg.local”.
The default setting is “Use Type A”.
Factory Reset
Resets the phone system to the default factory setting mode.
If the “Clear the SD card” is checked, the SD card storage mounted on the
phone will be format as well.
Safe Mode
Configures enable/disable safe mode. If enabled, the phone will enter safe
mode after rebooting, which will help remote troubleshooting when an
abnormal situation occurs. Note: Once entering safe mode, only the
system applications will be up and running, all widgets and 3
rd
party
applications will be disabled
Maintenance/System Diagnosis
Syslog
Syslog Protocol
Select the transport protocol over which log messages will be carried.
UDP: Syslog messages will be sent over UDP.
SSL/TLS: Syslog messages will be sent securely over TLS connection.
To upload server CA certificate, follow below steps:
Copy CA file in SD card and plug it to the phone.
Go to LCD menu SettingsSecurity SettingsInstall from SD
card to install the CA file.
Syslog Server
Configures the URI which the phone system will send the syslog messages
to. The default setting is "log.ipvideotalk.com".
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Syslog Level
Selects the level of logging for syslog. The default setting is "None". There
are 4 levels from the dropdown list: DEBUG, INFO, WARNING and
ERROR. The following information will be included in the syslog packet:
DEBUG (Sent or received SIP messages).
INFO (Product model/version on boot up, NAT related info, SIP
message summary, Inbound and outbound calls, Registration
status change, negotiated codec, Ethernet link up).
WARNING (SLIC chip exception).
ERROR (SLIC chip exception, Memory exception).
Note: Changing syslog level does not require a reboot to take effect.
Syslog Keyword Filter
Only send the syslog with keyword, multiple keywords are separated by
comma.
Example: set the filter keyword to “SIP” to filter SIP log.
Logcat
Clear Log
Clears the log files saved in the phone system.
Log Tag
Configures the filter to display the specified process log file.
Log Priority
Selects the log priority to display. It can be selected from list below:
Verbose (Default Setting)
Debug
Info
Warning
Error
Fatal
Silent (suppress all output)
Get Log
Displays the log file on the web page.
Debug
One-click Debugging
One-click Debugging
Capture the checked info in the debugging list, click "Start" to debug if
including "Capture trace" item and click "Stop" to end.
Click "Capture" in another situation. All retrieved files will be generated to
a package, and the last package will be overwritten, while the trace file will
stay remain.
Debug Info Menu
Display a list of info items that can be debugged, currently supports system
logs, info log, capture package, tombstones and ANR log. The captured
data can be viewed in "Debug information list". The default is all selected.
Debug Info List
You can select the existing debugging info package or grab package. Click
the "Delete" button on the right to delete the file.
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View Debug Info
You can select the existing debugging info package or grab package. Click
the "Delete" button on the right to delete the file.
Core Dump
Enable Core Dump
Generation
Configures whether to generate and save the core dump file when the
program crashes. The default setting is “No”.
Core Dump List
Selects the existing core dump file in the drop-down box. Users could
delete the file by pressing on “Delete” button.
View Core Dump
Press “List” button to view all existing core dump files. The files are listed
in chronological order, users could click the file name to download the file
to the local computer.
Record
Record
Click to start capturing audio data, click the "Stop" button to end. To capture
the audio data of the device can help to locate audio issues. The default is
not enabled. You can record up to 1-minute audio data.
Recording List
Choose the existing audio file. Click the "Delete" button on the right to
delete this file.
View Recording
Click on the "List" button to view. The captured audio data will be sorted by
time. Click to download the data to the computer for analysis. Note: The
audio data file will be saved under FileManager Internal Storage
Recfiles folder. Users can also delete files under this folder.
Traceroute
Target Host
The IP address or URL for the Target Host of the Traceroute.
Press Start to send traceroute request to configured target host.
Press Stop to end traceroute running process.
Maintenance/Event Notification
Set the URL for events on phone web GUI, and when the corresponding event occurs on the phone, the
phone will send the configured URL to SIP server. The dynamic variables in the URL will be replaced by
the actual values of the phone before sending to SIP server, in order to achieve the purpose of events
notification. Here are the standards:
1. The IP address of the SIP server needs to be added at the beginning and separate the dynamic
variables with a "/".
2. The dynamic variables need to have a "$" at the beginning. For example: local=$local
3. If users need to add multiple dynamic variables in the same event, users could use "&" to connect with
different dynamic variables. For example: 192.168.40.207/mac=$mac&local=$local
4. When the corresponding event occurs on the phone, the phone will send the MAC address and phone
number to server address 192.168.40.207.
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Bootup Completed
Configures the event URL when phone boots up.
Incoming Call
Configures the event URL when phone has an incoming call.
Outgoing Call
Configures the event URL when phone has an outgoing call.
Offhook
Configures the event URL when the phone is off hook.
Onhook
Configures the event URL when the phone is on-hook.
Missed Call
Configures the event URL when the phone has new a missed call.
Connected
Configures the event URL when a call is established.
Disconnected
Configures the event URL when a call is disconnected.
DND On
Configures the event URL when DND is enabled.
DND Off
Configures the event URL when DND is disabled.
Forward On
Configures the event URL when the forward feature is enabled on the
phone.
Forward Off
Configures the event URL when the forward feature is disabled on the
phone.
Blind Transfer
Configures the event URL when users transfer a call with blind transfer on
the phone.
Attended Transfer
Configures the event URL when users transfer a call with attended transfer
on the phone.
On Hold
Configures the event URL when users hold a call on the phone.
UnHold
Configures the event URL when users resume a call on the phone.
Log On
Configures the event URL when users log on the phone successfully.
Log Off
Configures the event URL when users log off the phone.
Register
Configures the event URL when an account in the phone is registered
successfully.
Unregister
Configures the event URL when an account in the phone is unregistered.
Application Page Definitions
Applications/Programmable key
Programmable key
Format
Display Format
Configures the display format for the MPK. Users could select "Name",
"User ID" or "Name(User ID)". "Name" is the one saved in phone contacts.
The default setting is "Name, User ID, Key mode”.
Show Display Name
from Server
If selected, the display name on the server will replace the name users
configured.
BLF
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Key Mode
The key modes are:
Speed Dial: Press to dial the UserID when the accounts being
configured as VPK.
Busy Lamp Field: Monitor the UserID status when the accounts
being configured as VPK.
Call Transfer: Transfer the current active call to UserID when the
accounts being configured as VPK.
Call Intercom: Intercom/paging to the UserID when the accounts
being configured as VPK.
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 UserID when the
accounts being configured as VPK.
Dial DTMF: Dial the DTMF digits of the UserID when the accounts
being configured as VPK during the call.
Call Park: Configure the call park feature code to park or retrieve the
call.
Multicast Paging: For multicast sending, please fill in the display
name in the Settings and fill in the sending address in the multicast
address.
Speed Conference: Quickly dial up multiple numbers to set up a
meeting.
Dial Prefix: After configured, once pressed this key, all numbers use
this account will automatically add the prefix promptly
Send URL: After configured and click on the button, the device will
send the entered URL automatically.
Account
Configures the SIP account when the accounts being configured as VPK.
Display Name
Configures the display name when the accounts being configured as VPK.
User ID
Configures the UserID for the corresponding VPK mode when the accounts
being configured as VPK.
DTMF Content
When key mode is set to Dial DTMF it configures the dialed DTMF content.
Address
When key mode is set to Multicast Paging, it configures the multiple
broadcast address.
Conference name
Set the name of speed meeting when the key mode is set to Quick
Conference.
Mute all members
Enable or disable mute all meeting members when key mode is set to
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Quick Conference.
Configure
Configures the Number list when key mode is set to Quick Conference.
Programmable key General Settings
Enable LCD Turn on
Automatically when
BLF/SCA status
changes
Configures whether to enable LCD turn on automatically when BLF/SCA
status changes.
Account
Displays account name if configured, otherwise, displays “Account X”
where X is the account number.
BLF Call-pick Prefix
Configures the prefix prepended to the BLF extension if the phone answers
a call to the monitored party by the BLF key. Default setting is ** for each
account.
Event List URI
Determines the event list BLF URI on the phone to monitor the extensions
in the list with MPK keys. This feature is based on BroadSoft standard. It
requires filling in the BLF ID to the box.
For example, if the server provides the URI: BLF123@myserver.com, this
field should be filled with BLF123. Then the monitored extensions will be
populated in the MPK app or Extension Board (if supported).
Force BLF Call-pickup
by Prefix
Uses the prefix for BLF Call-pickup. The default setting is “No”.
System Application/Contacts
General Settings
Sort Phonebook by
Sets which part of name, first name or last name, will be sorted in
alphabetical order to display.
Default Contacts Tab
Controls the behaviors of the phonebook key. It could be set to:
Default
LDAP Search
Local Phonebook
Local Group
Broadsoft Phonebook
Favorites.
The default setting is “Default”, which set the phonebook key to the
Contacts menu.
Emergency Call
Numbers
Configures the emergency contact in logout mode. If the system is logout,
guest users can dial the configured emergency contacts.
Input the number in the input box and click "Add" to add the number to the
contacts list.
To delete the existing ICE number, select the number in the contacts list
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and click "Delete".
Import/Export Contacts
Import
Clear The Old List
Determines if the phone system will delete the previous contacts when a
new contact file is imported. If set to "Yes", the previous contacts will be
removed. The default setting is "No".
Clear Old History Mode
If set to "Clear all", the phone will delete all previous records before
importing the new records.
If set to "Keep Local Contacts", the new-added local new contacts will
not be deleted when importing new records.
Replace Duplicate Items
Configures the phone system to keep the original contact entries when
duplicated contact entries are included in the contact file. If set to "Yes",
the phone will replace the original entries to the new one. Otherwise, the
phone system will save both contact entries. The default setting is "No".
Replace Duplicate
Entries Mode
If set to "Replace by name", replace the records of the same name
automatically when importing new records.
If set to "Replace by number", replace the records of the same number
automatically when importing new records.
Keep Associated
Account
The feature configures whether to keep the associated account when
importing contacts. If enabled, the associated account information in the
contact file will be synchronized to the phone
File Encoding
Specifies the encoding format for phonebook file importing. The default
setting is UTF-8. It can be selected from the dropdown list:
UTF-8
GBK
UTF-16
UTF-32
Big5
Big5-HKSCS
Shift-JIS
ISO8859-1
ISO8859-15
Windows-1251
EUC-KR
File Type
Sets the type format for phonebook file importing. It can be selected from
the dropdown list. The default setting is "XML".
XML
vCard
Import Local File
Uploads the contact file from PC to the phone.
Export
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File Encoding
Specifies the encoding format for phonebook file exporting.
The default setting is UTF-8. It can be selected from the dropdown list:
UTF-8
GBK
UTF-16
UTF-32
Big5
Big5-HKSCS
Shift-JIS
ISO8859-1
ISO8859-15
Windows-1251
EUC-KR
File Type
Sets the type format for phonebook file importing. It can be selected from
the dropdown list.
XML
VCard
The default setting is "XML".
Export
Downloads the phonebook file from the phone to PC.
Download Contacts
Clear The Old List
Sets the phone system to delete the previous contacts when a new contact
file is downloaded.
If "Yes", the previous contacts will be removed.
The default setting is "No".
Clear Old History Mode
If set to "Clear all", the phone will delete all previous records before
downloading the new records.
If set to "Keep Local Contacts", the new-added local new contacts will
not be deleted when downloading new records.
Replace Duplicate Items
Keeps the original contact entries when duplicated contact entries are
included in the contact file. If set to "Yes", the phone will replace the original
entries to the new one. Otherwise, the phone system will save both contact
entries. The default setting is "Yes".
Replace Duplicate
Entries Mode
If set to "Replace by name", replace the records of the same name
automatically when importing new records.
If set to "Replace by number", replace the records of the same number
automatically when importing new records.
Download Mode
Enables the phone system to download phonebook file and select the
server and protocol to download the phonebook file. It can be selected from
TFTP, HTTP, and HTTPS. The default setting is “OFF”.
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Keep Associated
Account
The feature configures whether to keep the associated
account when importing contacts. If enabled, the associated
account information in the contact file will be synchronized to
the phone
File Encoding
Selects the encoding format for phonebook file download. The default
setting is UTF-8. It can be selected from the dropdown list:
UTF-8
GBK
UTF-16
UTF-32
Big5
Big5-HKSCS
Shift-JIS
ISO8859-1
ISO8859-15
Windows-1251
EUC-KR
Download Server
Configures the server URL to download the phonebook file.
The phone system will send a request to the server to download the
phonebook file with filename phonebook.xml.
HTTP/HTTPS Username
Configures username for HTTP/HTTPS server to download the phonebook
file.
HTTP/HTTPS Password
Specifies password for HTTP/HTTPS server to download phonebook file.
Automatic Download
Interval
Determines how the phone system to send the request to the server to
download the phonebook file.
It can be selected from the dropdown list:
None
5 Minutes
30 Minutes
1 Hour
2 Hour
4 Hour
6 Hour
8 Hour
12 Hour
Download Now
Starts downloading the XML phonebook to the phone immediately.
System Application/LDAP Phonebook
Connection Mode
Selects which protocol will be used for LDAP searching, LDAP or LDAPS.
Server Address
Configures the URI of the LDAP (Lightweight Directory Access Protocol)
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server.
Port
Configures the LDAP server port. The default LDAP port number is 389.
Base DN
Determines the LDAP search base. This is the location in the directory
where the search is requested to begin.
Example:
dc=grandstream, dc=com
ou=Boston, dc=grandstream, dc=com
Username
Configures the bind "Username" for querying LDAP servers. Some LDAP
servers allow anonymous binds in which case the setting can be left blank.
Password
Specifies the bind "Password" for querying LDAP servers. The field can be
left blank if the LDAP server allows anonymous binds.
LDAP Name Attributes
Configures the "name" attributes of each record which are returned in the
LDAP search result. This field allows the users to configure multiple space
separated name attributes.
Example:
cn sn description
LDAP Number
Attributes
Configures the "number" attributes of each record which are returned in the
LDAP search result. This field allows the users to configure multiple space
separated number attributes.
Example:
telephoneNumber
telephoneNumber Mobile
LDAP Mail Attributes
Determines the "mail" attributes of each record which are returned in the
LDAP search result.
Example: mail
LDAP Name Filter
Configures the filter used for name lookups.
Examples:
(|(cn=%)(sn=%)) returns all records which has the "cn" or "sn" field starting
with the entered prefix;
(!(sn=%)) returns all the records which do not have the "sn" field starting
with the entered prefix;
(&(cn=%) (telephoneNumber=*)) returns all the records with the "cn" field
starting with the entered prefix and "telephoneNumber" field set.
LDAP Number Filter
Defines the filter used for number lookups.
Examples:
(|(telephoneNumber=%)(Mobile=%) returns all records which has the
"telephoneNumber" or "Mobile" field starting with the entered prefix;
(&(telephoneNumber=%) (cn=*)) returns all the records with the
"telephoneNumber" field starting with the entered prefix and "cn" field set.
LDAP Mail Filter
Determines the filter used for mail lookups. Example: (mail=%)
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Search Field Filter
Configures to filter according to which fields when searching on LDAP.
Users can choose between ‘Name Filter’, ‘Number Filter’, ‘Mail Filter’ or All
Filter’. The default setting is "All Filter".
LDAP Displaying Name
Attributes
Configures the entry information to be shown on phone's LCD. Up to 3
fields can be displayed.
Example: %cn %sn %telephoneNumber
Max Hits
Specifies the maximum number of results to be returned by the LDAP
server. If set to 0, server will return all search results. Default setting is 50.
Search Timeout (s)
Configures the interval (in seconds) for the server to process the request
and client waits for server to return. The default setting is 4 seconds.
LDAP Lookup For Dial
Sets the phone system to do the LDAP number searching when making
outgoing calls. The default setting is "No".
LDAP Lookup For
Incoming Call
Sets the phone system to do LDAP number searching for incoming calls.
The default setting is "No".
LDAP Dialing Default
Account
Configures the default account that being used when dialing LDAP contact.
Users may choose the Account 1-6; the default setting is “Default”.
System Application/Recording
File name
Displays the name of the recording file
Duration
Displays the duration of the phone call
Date
Displays the date the call was recorded on
Operation
Delete, Modify, or download the recording file
Value-added Service Page Definitions
Value-added Service/Value-added Service (0/10)
Service Type
Users can set the service type to “Door System” to configure the door
system options, or to “DTMF” to set DTMF content to send it during calls.
Door System Type
Set the door system type to “GDS” if the GDS door system is used or set
it to “Baudisch” if another door system brand is used.
Note: Each GDS door system has 2 different access passwords to control
2 doors separately named as [Related Display Name1] & [Related
Display Name2] below for door 1 and 2 respectively.
System Number
Specifies the door system number which is the SIP user ID configured on
door system or its IP address, if the door system is using IP call. It enables
to show open door button when caller number or IP address matches with
this setting. e.g:"36311" or "192.168.124.81".
Note: When set “Door System Type” to “Baudisch” a “configure” button will
appear to allow user to configure groups of door system URL and User ID
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for 100 entries.
System Number: This is used to configure the User ID of door system.
Once configured, only the call from this User ID would use door system
while other calls use the default mode.
System Address: This is used to input the IP address or URL of the
system to identify the call from door system.
Users can set HTTP authentication credentials on the URL for Door
Systems that require authentication to send HTTP stream. The URL
format will be similar to the following:
http://username:p[email protected].150/goform/stream?cmd=get&channel=4
Display Name
Configures the display name of the door system. When the call matches
the configured system number, the name will be displayed on LCD.
Related Display Name1
Configures the name that will be displayed on LCD for door 1 when the call
matches the configured GDS door system number.
Access password
Determines the door system password which should match the one
configured on the used door system settings. In case the GDS is set as
‘Door System type’ parameter, the password should match the one
configured on the GDS to open door 1.
Related Display Name2
Indicates the name that will be displayed on LCD for door 2 when the call
matches the configured system number.
Access password
The configured password should match the one configured on the GDS to
open door 2.
System Ringtone
Allows users to configure the ringtone for the door System. Users can
choose different ringtones from the dropdown list.
DTMF Content
Set the DTMF content that is going to be sent when the DTMF button is
pressed under Call screen“Keypad”.
Display Condition
Configures whether the DTMF button display on incoming call interface or
outgoing call interface.
Value-added Service/General Settings
Display Open Door
Button when Calling
Configures whether display Open Door button when there is an incoming
call.
Enable Preview
This feature allows user to enable/disable video preview but keep the
ringing.
Value-added Service/Broadsoft Settings
Broadsoft Call Features
Feature Key
Synchronization
Synchronizes the BroadSoft standard call feature.
If it is enabled, the phone will send SIP SUBSCRIBE message to the
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server and receive SIP NOTIFY message from the server to synchronize
the DND, Call Forwarding and Call Center features.
The default setting is “Disable”.
Enable BroadSoft Call
Park
Configures whether to send SUBSRCIRBE message to BroadSoft server
to obtain Call Park notifications.
The default setting is “No”.
Conference URI
Configures the network-based conference URI (the BroadSoft Standard).
If it is configured, end user needs to tap the N-way key during the
conference to transfer the host to the remote media server.
Broadsoft Call Center
When enabled, Feature Key Synchronization will be enabled regardless
of web settings. Default is Disabled.
Hoteling Event
Enables BroadSoft Hoteling Event feature.
Default is Disabled.
Call Center Status
When set to "Yes", the phone will send SUBSCRIBE to the server to obtain
call center status.
Default is Disabled.
SCA
Enable SCA (Shared Call
Appearance)
Enables/disables the Shared Call Appearance (the Broadsoft Standard)
feature for this account. If it is set to “Yes”, the phone system can update
and share account status with another device. The default setting is "No".
Enable Barge-In
Enables/disables the Barge-In feature. If it is set to "Yes", the user could
tap the SCA account to barge into an active session with another shared
line. The default setting is "No".
Auto-filling CallPark
Feature Code
If it is set to "Yes", the configured "Call Park Service Code" will be
automatically filled in on the phone's dial pad when picking up the parked
call. This option will be active only if "Special Mode" is set to "Broadsoft"
and "Enable SCA" is set to "Yes". The default setting is "Yes".
CallPark Feature Code
Configures the pickup feature code for call park. If " Auto-filling CallPark
Feature Code " is set to "Yes", this call park service code will be
automatically filled in on the phone's dial pad when picking up the parked
call. This is used when "Special Mode" is set to "BroadSoft" (from web UI
or provisioning) and "Enable SCA" is set to "Yes"
Line Seize Timeout (s)
Defines line-seize expiration timer. For Shared Call Appearance, the
phone must send a SUBSCRIBE request for the line-seize event package
whenever a user attempts to take the shared line off hook. The default
value is 15 seconds. The valid range is from 15 to 60.
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Value-added Service/Broadsoft Directories
XSI Service Settings
Authentication Type
Defines the authentication type in way of login or use SIP. If set to “Login
Credentials”, please fill in Username and Password in the following
options; if set to “SIP Credentials”, please fill in Username, User ID, and
Password.
The default setting is “SIP Credentials”.
Server
Configures the BroadWorks XSI server URI. If the server uses HTTPS,
please add the header “HTTPS” ahead of the Server URI. For instance,
“https://SERVER_URI”.
Port
Configures the BroadWorks XSI server port. The default port is 80. If the
server uses HTTPS, set to 443.
Action Path
Configures Action path for BroadSoft XSI server.
BroadWorks User ID
Determines the User ID for Broadsoft XSI server.
SIP Authentication ID
Determines username for Broadsoft XSI server.
SIP Authentication
Password
Determines password for Broadsoft XSI server.
Login Password
Specifies the password for the BroadWorks XSI query.
BroadSoft Directory &
Call Logs Update Interval
(s)
Configures the interval to obtain BroadSoft call log and directory data thus
to update the matching number data on the dialing interface.
Default is 1800 seconds.
BroadSoft Directory Hits
Configures the maximum hits returned from the BroadSoft XSI server
directory. The valid range is from 1 to 1000. If set to blank, server’s default
value will be used.
BroadSoft Directory
Order
Defines the BroadSoft directory order displayed on LCD. Select one item
and click the Up/Down arrow on the right to adjust the order.
Network Directories
Group Directory
Enables/disables the BroadWorks XSI Group Directory features on the
phone. The name filed is used to name the directory. If it keeps blank, the
phone system will use the default name “Group” for it
Enterprise Directory
Enables/disables the BroadWorks XSI Enterprise Directory features on
the phone. The name filed is used to name the directory. If it keeps blank,
the phone system will use the default name “Enterprise” for it.
Group Common
Enables/disables the BroadWorks XSI Group Common features on the
phone. The name filed is used to name the directory. If it keeps blank, the
phone system will use the default name “Group Common” for it.
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Enterprise Common
Enables/disables the BroadWorks XSI Enterprise Common features on
the phone. The name filed is used to name the directory. If it keeps blank,
the phone system will use the default name “Enterprise Common” for it.
Personal Directory
Enables/disables the BroadWorks XSI Personal Directory features on the
phone. The name filed is used to name the directory. If it keeps blank, the
phone system will use the default name “Personal Directory” for it.
Polycom Phonebook
Enables/disables the BroadWorks XSI Polycom Phonebook features on
the phone. The name filed is used to name the directory. If it keeps blank,
the phone system will use the default name “Polycom Phonebook” for it
Missed Call Log
Enables/disables the BroadWorks XSI Missed Call Log features on the
phone. The name filed is used to name the directory. If it keeps blank, the
phone system will use the default name “Missed” for it.
Placed Call Log
Enables/disables the BroadWorks XSI Placed Call Log features on the
phone. The name filed used to name the directory. If it keeps blank, the
phone system will use the default name “Outgoing” for it.
Received Call Log
Enables/disables the BroadWorks XSI Received Call Log features on the
phone. The name filed used to name the directory. If it keeps blank, the
phone system will use the default name “Incoming” for it.
Value-added Service/BroadSoft IM&P
Login Credentials
Server
BroadSoft IM&P server address. Usually it is not necessary to configure,
and it can already be found in the BroadSoft IM&P username
Port
Specifies the BroadSoft IM&P server port. The default port is 5222.
Username
Determines the Username for the BroadSoft IM&P query. This is usually
not the same as BroadSoft account username.
Password
Determines the password for the BroadSoft IM&P query. This is usually
not the same as BroadSoft account password.
IM&P Settings
Enable BroadSoft IM&P
Enables the BroadSoft XMPP feature.
Associated BroadSoft
Account
Binds the SIP account for dialing the XMPP client contacts.
Auto Login
Sets the client to be logged automatically if the application is started.
Display Non XMPP
Contacts
Determines if the client app will show the non-XMPP contacts.
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UPGRADING AND PROVISIONING
The GXV3370 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, HTTP or
HTTPS; the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com/BETA
fw.mycompany.com
Upgrade and Provisioning Configuration
There are two ways to setup upgrade and provisioning on GXV3370. They are Keypad Menu and Web GUI.
Configure via keypad Menu
In GXV3370 Settings, select Advanced System Update.
1. Press Detect New Version to initiate upgrade process.
Figure 7: GXV3370 Upgrade – Detect New Version – Start Provision
2. Press settings icon to configure upgrade settings. Users may then select the upgrade mode
and enter the IP address or FQDN for the Firmware server and the Config server. After making
the changes, tap Save button to save the change. Then reboot the phone or go back and press
Detect New Version.
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Figure 8: GXV3370 Upgrade Configuration via LCD
Configure via Web GUI
Open a web browser on PC and enter the IP address for the GXV3370. Then login with the administrator
username and password (that needs to be at least 6 characters). Go to Maintenance Upgrade. In
the Upgrade web page, enter the IP address or the FQDN for the upgrade server and choose to upgrade
via TFTP, HTTP or HTTPS (The default setting is HTTPS). Save and apply the changes, press Upgrade
button, or reboot the phone to initiate firmware upgrade process.
Figure 9: GXV3370 Upgrade Configuration via Web GUI
Note: Please do not power off or unplug the GXV3370 when the upgrading process is on.
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Upload Firmware Locally
If there is no HTTP/TFTP server, users could also upload the firmware to the GXV3370 directly via Web
GUI. Please follow the steps below to upload firmware to GXV3370 locally.
1. Download the latest GXV3370 firmware file from the following link and save it in your PC.
http://www.grandstream.com/support/firmware
2. Log in the Web GUI as administrator in the PC.
3. Go to Web GUIMaintenanceUpgrade.
4. Click the "Upload" button, a window will be prompted to select firmware file to upload.
5. Select the firmware file from your PC. Then uploading progress will show at the button where it was
"Upload" in the above step.
6. When uploading is done, users can see the upgrading process starts on the GXV3370 LCD.
7. The phone will reboot again with the new firmware version upgraded.
Figure 10: Upload Firmware File to Update
Upgrade via SD Card
For users that could not use remote upgrade or could not access the phone's Web GUI to upload firmware,
upgrading via external SD card is an alternative. Follow the steps below to upgrade GXV3370 via SD card.
1. Download the firmware file to PC and save it in SD card.
2. Insert the SD card to GXV3370.
3. Power cycle the GXV3370.
4. Wait for the LED in the upper right of the phone to light up in Green, then about 2 seconds later press
the both keys (the first LCD key from the left : Volume Down) and (the third LCD key from
the left : Home) at the same time, it will go into the upgrading process.
5. The GXV3370 will start upgrading and display the upgrading process in the screen.
6. Wait until the upgrading is done.
7. The GXV3370 will reboot itself.
8. Check the firmware status and remove the SD card.
Note: Upgrading via USB storage device is not supported on the GXV3370.
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No Local Firmware Servers
Service providers should maintain their own firmware upgrade servers. For users who do not have a
TFTP/HTTP/HTTPS server, some free Windows version TFTP servers are available for download from:
http://www.solarwinds.com/free-tools/free-tftp-server and http://tftpd32.jounin.net/.
Please check our web site at https://www.grandstream.com/support/firmware for latest firmware.
Instructions for local firmware upgrade via TFTP:
1. Unzip the firmware files and put all of them in the root directory of the TFTP server;
2. Connect the PC running the TFTP server and the GXV3370 device to the same LAN segment;
3. Launch the TFTP server and go to the File menuConfigureSecurity 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 GXV3370.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use Microsoft
IIS web server.
Provisioning and 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, HTTP or 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 “Admin Password” in the Web GUISystem SettingsSecurity
SettingsUser Info Management page. For a detailed parameter list, please refer to the corresponding
firmware release configuration template in the following link:
https://www.grandstream.com/support/tools
When the GXV3370 boots up, it will issue TFTP or HTTP request to download a configuration XML file
named "cfgxxxxxxxxxxxx" followed by "cfgxxxxxxxxxxxx.xml", where "xxxxxxxxxxxx" is the MAC address of
the phone, i.e., "cfg000b820102ab" and "cfg000b820102ab.xml". If downloading "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 the following document:
https://www.grandstream.com/sites/default/files/Resources/gs_provisioning_guide.pdf
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Note:
When the prompt in the figure below shows up, it means the firmware/config authentication failed. So the
user will be required to check the username/password on device web UI Maintenance Upgrade:
Firmware HTTP/HTTPS username
Firmware HTTP/HTTPS password
Config HTTP/HTTPS username
Config HTTP/HTTPS password
Figure 11: Config File Upgrade Verification
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FACTORY RESET
Restore to Factory Default via LCD Menu
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.
In order to restore the GXV3370 unit to factory reset via the LCD Menu, please, refer to the following steps:
1. On GXV3370 idle screen, go to Settings Advanced System Security Factory reset.
2. Tap on Ok to confirm.
Figure 12: GXV3370 LCD - Confirm Factory Reset
Restore to Factory Default via the Web GUI
1. Login GXV3370 Web GUI and go to Maintenance Upgrade Advanced Settings.
2. At the bottom of the page, click on the Reset button for Factory reset.
Figure 13: GXV3370 Web GUI - Factory Reset
3. A dialog box will pop up to confirm factory reset.
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4. Click OK to restore the phone to factory settings.
Figure 14: GXV3370 Web GUI - Confirm Factory Reset
Restore to Factory Default via Hard Keys
For users that could not restore the GXV3370 to factory reset via LCD Menu or the Web GUI, restoring the
unit via Hard keys is an alternative. Please, follow the steps below to restore the GXV3370 via Hard Keys:
1. Power cycle the GXV3370.
2. Wait for the LED in the upper right of the phone to light up in Green, then about 2 seconds, press the
both keys (the second LCD key from the left: Volume Up) and (the 4th LCD key from the left :
Menu) at the same time, it will go into the factory reset process.
3. The LCD screen will display “Factory reset, please wait”.
4. The GXV3370 will reboot with factory default settings.
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SAFE MODE
The GXV3370 allows users to enter safe mode by enabling safe mode option from WEB UI under
MaintenanceUpgrade/Advanced Settingssafe mode. If enabled, the phone will enter safe mode
after rebooting. Users can alternatively enter safe mode by pressing the Menu button during bootup.
Before entering the safe mode, please power cycle the phone and when the plain text “GRANDSTREAM”
shows up, immediately press, and hold the Menu button when the five buttons light up again after the
top right LED flash ends. Users will see the phone boot up in safe mode.
Under safe mode, only the system applications will be up and running. Normally safe mode is not needed
unless the phone cannot function anymore caused by incompatible 3rd party applications. For example, if
a 3rd party application is downloaded and installed on the phone that cause the phone keep crashing or
freezing and users cannot operate on the phone's settings, users can enter safe mode to remove the 3rd
party application and boot up in normal mode again.
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SDK FRAMEWORK SERVICE
The GXV3370 Software Development Kit (SDK) is making it possible for developers and system
administrators to run and use API to manage the LCD screen display of the phone and its configuration.
Besides inheriting the Android interface functions, it has added other interfaces according to user’s
requirements for developing customized services and bring them into effective action on the GXV3370.
Note: SDK framework service is supported starting from firmware 1.0.1.21.
For more information about “SDK Framework Service”, please refer to the documentation included in the
following SDK package:
https://www.grandstream.com/sites/default/files/Resources/GXV33xx_WP820_SDK_Framework_Service
_Guide.zip
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EXPERIENCING THE GXV3370 APPLICATION PHONE
Please visit our website: https://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 Enterprise Application phone, it will be sure to bring
convenience and color to both your business and personal life.
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