Grandstream Networks GXV3175-P IP MULTIMEDIA PHONE User Manual GXV3175 User Manual

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Grandstream Networks, Inc.
GXV3175 V2
IP Multimedia Phone
Grandstream Networks, Inc.
GXV3175 USER MANUAL
Updated : 3/2012
GXV3175 User Manual
Index
WELCOME ..................................................................................................6
GXV3175 OVERVIEW.................................................................................................................... 6
PRODUCT OVERVIEW ...............................................................................8
INSTALLATION ......................................................................................... 11
EQUIPMENT PACKAGING ................................................................................................................. 11
SAFETY COMPLIANCES............................................................................................................. 11
WARRANTY ................................................................................................................................. 12
CONNECTING THE GXV3175 IP MULTIMEDIA PHONE................................................................... 12
WALL MOUNT .............................................................................................................................. 13
PHONE STAND ............................................................................................................................ 13
CONNECTING THE GXV3175 IP MULTIMEDIA PHONE............................................................ 14
WELCOME SETUP GUIDE.......................................................................................................... 14
MAKING THE FIRST CALL WITH IPVIDEOTALK SERVICE ...................................................... 17
USING THE GXV3175 IP MULTIMEDIA PHONE...................................... 18
LCD ...................................................................................................................................................... 18
MULTIPLE SIP ACCOUNTS AND LINES ............................................................................................ 23
HANDSET, SPEAKER AND HEADSET MODE................................................................................... 24
MAKING A CALL.................................................................................................................................. 24
DURING CALLS .................................................................................................................................. 27
ANSWERING CALLS ................................................................................................................... 27
CALL FUNCTION OPTIONS ........................................................................................................ 29
CALL HOLD.................................................................................................................................. 30
CALL TRANSFER ........................................................................................................................ 31
3-WAY CONFERENCE ................................................................................................................ 33
VOICEMAIL（MESSAGE WAITING INDICATION） ................................................................... 34
MUTE............................................................................................................................................ 34
KEYPAD INPUT ................................................................................................................................... 35
CALL FEATURES ..................................................................................... 36
GXV3175 WEB CONFIGURATION INTERFACE...................................... 37
ACCESSING THE WEB CONFIGURATION PAGES .......................................................................... 38
Grandstream Networks, Inc.
GXV3175 USER MANUAL
FIRMWARE VERSION 1.0.1.1
Updated : 3/2012
DEFINITIONS ...................................................................................................................................... 38
STATUS PAGE DEFINTIONS ...................................................................................................... 39
ADVANCED SETTINGS PAGE DEFINITIONS ............................................................................ 40
MAINTENANCE PAGE DEFINITIONS......................................................................................... 42
SIP ACCOUNT CONFIGURATION PAGE DEFINITIONS............................................................ 49
SAVING THE CONFIGURATION CHANGES...................................................................................... 54
REBOOTING FROM REMOTE LOCATIONS...................................................................................... 54
CUSTOMIZING THE GXV3175 IP MULTIMEDIA PHONE ........................ 55
XML GUI CUSTOMIZATION GUIDE ................................................................................................... 55
GMI GUIDE .......................................................................................................................................... 55
SOFTWARE UPGRADE............................................................................ 56
UPGRADE VIA MENU ......................................................................................................................... 56
UPGRADE VIA WEB BROWSER CONFIGURATION PAGE.............................................................. 56
NO LOCAL TFTP SERVERS............................................................................................................... 57
RESTORE FACTORY DEFAULT SETTINGS............................................ 57
RESTORE TO FACTORY DEFAULT SETTINGS VIA LCD MENU ..................................................... 57
RESTORE TO FACTORY DEFAULT SETTINGS VIA THE WEB INTERFACE .................................. 58
EXPERIENCING THE GXV3175 MULTIMEDIA PHONE .......................... 59
TABLE OF TABLES
GXV3175 USER MANUAL
Table 1 :Key Features ................................................................................................................................... 8
Table 2: GXV3175 Hardware Specification ................................................................................................... 8
Table 3: GXV3175 Technical Specifications.................................................................................................. 8
Table 4: GXV3175 Ports and Interface ....................................................................................................... 12
Table 5: LCD Icon Definition........................................................................................................................ 21
Table 6: GXV3175 Dialpad Definition.......................................................................................................... 23
Table 7: GXV3175 Call Function Description.............................................................................................. 30
Table 8: GXV3175 Keypad Definition.......................................................................................................... 35
Table 9 :GXV3175 Call Features ................................................................................................................ 36
Table 10: GXV3175 Audio/Video Codec Feature Code .............................................................................. 36
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WELCOME
Thank you for purchasing Grandstream GXV3175 IP Multimedia Phone. GXV3175 IP Multimedia Phone
represents the future in personal IP multimedia communication as the ultimate desktop multimedia phone.
You will experience a brand-new exploration with GXV3175’s 7” touch screen color LCD, plug-and-play
access to free real-time video/voice call, always-on Internet web entertainment and social networking
applications.
This user manual is designed to help you configure and manage GXV3175 IP Multimedia Phone. Besides
demonstrating how to install the unit with ease, this manual will explain how to fully utilize the entire
phone’s voice and video call features like conference call, direct IP call as well as explore all the built-in
feature-rich multimedia applications such as YouTube, Google Voice, Flickr and more.
GXV3175 OVERVIEW
The GXV3175 is an ideal desktop IP multimedia phone highlighting 7” touch screen color LCD, tilt capable
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1.3M CMOS camera, integrated Wifi, comprehensive auxiliary ports, feature-rich web applications with
novel streamline design and modeling. It supports a high quality selection of audio codecs and is
compatible with the H.264/H.263/H.263+ video codec. The advanced telephony and video features,
interoperability with 3rd party SIP products and ease of use make it an innovative and unique IP multimedia
device for both enterprise and consumer users.
Once the GXV3175 is plugged into the broadband Internet connection, you can begin to make video calls
with impressive clarity and enjoy other features of the multimedia phone including:
•
Integrated web browser for one-touch access to personalized RSS feeds of real-time online
information services (news videos, stock and currency updates, weather forecasts and etc.)
•
Access to thousands of Internet radio stations and popular online music networks such as Last.fm
•
Access to social networking sites such as Google Voice and web photo albums like Yahoo Flickr
•
Access to streaming Internet entertainment sites including YouTube and Movie Trailer
•
Access to more than 6 built-in games featuring the touch screen operation
•
Digital Photo Frame
•
Full duplex speakerphone
•
Ability to project video onto TV
CAUTION: Changes or modifications to this product not expressly approved by Grandstream, or operation of
this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
•
This document is subject to change without notice. The latest electronic version of this user manual is
available for downloading here:
GXV3175 User Manual(Download)
•
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any
purpose without the written permission of Grandstream Networks, Inc. is not permitted.
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GXV3175 USER MANUAL
Page 5 of 60
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PRODUCT OVERVIEW
Table 1: Key Features
Open Compatible Standards
SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS
(A record and SRV), DHCP, PPPoE, TFTP, NTP.
Interfaces
2 x 10/100Mbps Ethernet Ports with integrated PoE，2 x USB（2.0）port，
3.5mm stereo headset port, SD Card slot, built-in WiFi(b/g/n).
Extraordinary Audio Quality
Advanced DSP for audio, Silence Suppression, VAD, CNG, AEC and AGC.
Extraordinary Video Quality
Real-time video H.263/H.263+ and H.264 codec even when operating
under low bandwidth (32kbps-1Mbps).
Advanced Video Features
7” resistive touch screen LCD with 800 x 480 resolution, 45 degree rotating
lens (perpendicular), Advanced VGA lens, auto focus, auto exposure,
zoom in and zoom out (2x optical 2x digital) , Camera Block,
PIP(Picture-in-Picture) and still picture capture/store.
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Application Features
- Traditional audio features: Caller ID, Call Waiting, Call Holding, Call
Transfer, Three-way Conference, Do-Not-Disturb, Mute, Automatic dial.
- Application features: Built-in Web Browser, RSS News, Weather
Forecast, Internet Radio, Media Player, Picture Viewer, Games,
Calculator, Alarm clock, File Manager, Youtube, Google Voice, Facebook,
Twitter and etc.
Table 2: GXV3175 Hardware Specification
Ethernet Port
Dual switched 10M/100M auto-sensing Ethernet ports.
LCD
800 x 480 resistive touch screen LCD.
Camera
Tilt Capable 2.0M pixel CMOS camera with privacy shutter (VGA).
Auxiliary Ports
RCA Type Stereo and Media output port, 3.5mm stereo headset port, 2 x
USB port, SD card slot.
Exterior
Black ABS plastic, Flat Panel with 1 x home button, two-angle stand.
Universal Power Supply
Input: 100-240V AC
Output: 12V DC, 1.5A
US/Euro/UK/Australian style plug available.
Dimension
243.5mm x 168mm x 36mm.
Weight
1.04kg (phone with handset and stand), 1.95kg (package weight).
Operating Temperature
32°F - 104° F/0°C - 40°C.
Humidity
10%-90% Non-Condensing.
Compliance
FCC/CE/C-Tick.
Table 3: GXV3175 Technical Specifications
Lines
Three individual SIP accounts.
Protocol Support
SIP 2.0, UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP,
ICMP, DNS, DHCP, NTP/SNTP, TFTP.
Display
7” resistive touch screen LCD with 800 x 480 resolution.
Network Interfaces
Dual switched 10M/100M auto-sensing Ethernet ports, 2 x USB 2.0 port,
1 x audio/video output port (to synchronize output video to TV), stereo
headset jack.
Device Management
Layer 2 QoS (IEEE 802.1p/Q tagging-VLAN) and Layer 3 QoS (ToS,
DiffServ, MPLS), web interface or via secure (AES encrypted) central
configuration file for mass deployment, auto/manual provisioning system,
remote software (TFTP/HTTP/HTTPS) upgrade for deployed devices
including those behind NAT/firewall.
Provisioning
- Support for automatic NAT/firewall traversal and remote automatic
software
upgrade
and
security,
providing
end-users
with
“zero
configuration” and true “plug-and-play” functionality.
- Support for remote configuration monitoring and syslog.
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- Configuration through LCD, web browser or an external configuration file.
- Support for IETF STUN NAT traversal and symmetric RTP, Static IP or
DHCP.
Audio Features
- HD handset and full duplex speaker, advanced DSP.
- Supports G.723.1, G.729A/B,G.711μ/A (PCMU/PCMA), G.726-32,G.722,
GSM-FR, L15-256 DTMF (In-audio, RFC2833, SIP Info); Silence
suppression, VAD, CNG, AGC, masking for packet loss/delay; AEC, AGC
for Speaker; jitter buffer protocol.
Video Features
- Support for Jitter Buffer delay and packet loss concealment to enhance
audio and video quality.
- Support for H.263/H.263+ and H.264 real-time video codecs (WQVGA,
QQVGA, QCIF, QVGA, WQVGA, CIF, VGA, 4CIF) up to 30frames/sec,
which ensures high quality video transmission even under low bandwidth
conditions (32kbps-1Mbps).
- 7” resistive touch screen LCD with 800 x 480 resolution, 45 degree
rotating lens (perpendicular), advanced VGA CMOS camera and sensor.
- Anti-flickering of images, auto focus, auto exposure, zoom (2x optical 2x
digital ), PIP(Picture-in-Picture) , audio mute, camera block, call log, video
phone book, screensaver, still picture capture/store (VGA) and visual voice
message indicator.
Call Handling Features
Caller ID, call waiting ID, call waiting/flash, call transfer, call holding, call
forwarding, Do-Not-Disturb, three-way conference, redial, automatic
dialing on off-hook, automatic answering, call records, volume control,
voice message waiting indicator, downloadable custom ring tone,
switchable handset/headset/speaker during call.
Caller ID（Privacy）
Private header support for anonymous calls.
Firmware Upgrade
- Configuration file authentication (before accepting changes).
- TFTP/HTTP/HTTPS upgrade support, allows users to specify different
URLs for the server to download from.
Advanced Server Features
DNS SRV support, SIP server failure transfer, message waiting indicator
and custom screensaver.
Security
- MD5 and MD5-sess DIGEST encoding and authentication.
- Security protection: SIP over TLS and SRTP.
- Support for TR-069.
- Support for OpenVPN for increased security and control.
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INSTALLATION
EQUIPMENT PACKAGING
This GXV3175 package contains:
1. One (1) GXV3175 Phone Main Case with One (1) Stylus inserted in the back
2. One (1) Handset
3. One (1) Quick Start Guide
4. One (1) Headset Dongle
5. One (1) Ethernet Cable
6. One (1) Phone Cord
7. One (1) 12V Power Adapter
8. One (1) Phone Stand
9. One (1) Wall Mount
10. One (1) TV VGA Adapter Cable
Figure 1: GXV3175 Equipment Packaging
SAFETY COMPLIANCES
GXV3175 complies with FCC/CE/C-Tick safety standards. The power adapter of GXV3175 is compliant
with the UL standard. Please use the universal power adapter provided with the GXV3175 package only.
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FIRMWARE VERSION 1.0.3.40
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The manufacturer’s warranty does not cover damages to the phone caused by unsupported power
adapters.
WARRANTY
If GXV3175 was purchased from a reseller, please contact the company the phone was purchased for
replacement, repair or refund. If the phone was purchased directly from Grandstream, please contact
Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number before
the product is returned. Grandstream reserves the right to remedy warranty policy without prior notification.
WARNING: Use the power adapter provided with the phone. Do not use a different power adapter as this
may damage the phone. This type of damage is not covered under warranty.
CONNECTING THE GXV3175 IP MULTIMEDIA PHONE
Figure 2: GXV3175 Ports and Interface
Table 4: GXV3175 Ports and Interface
Item
Name
Description
Home Button
Press Home button to navigate to main screen
Camera
1.3M pixel adjustable CMOS camera with privacy shutter
Ethernet Port
Network Ethernet Port
10/100Mbps RJ-45 port connecting to Ethernet
Power Jack
12V DC Power connector port
RJ11 Jack
Phone handset connector port
Camera Adjust Wheel
Scroll the wheel to turn on/off the camera;
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10/100Mbps RJ-45 port connecting to PC
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Adjust the camera position and lens angle
Stylus
Stylus for touch screen (optional)
USB Port
USB devices may be connected via the USB port. For example, you can
connect a USB flash drive to save captured pictures and plug in USB
keyboard or mouse for the built-in web browser
10
SD Card Slot
SD card could be inserted in for picture/music/video files storage
11
HDMI
High-Definition Multimedia Interface
12
Headset Jack
3.5mm stereo headset connector port
WALL MOUNT
GXV3175 has two slots on the back of the phone for wall mounting convenience. (See Figure 3)
Figure 3: Wall Mount for GXV3175
PHONE STAND
GXV3175 can also be placed on the table or desk via the phone stand. (See Figure 4) There are two
positions for the stand on the back of the phone with each position supporting a different angle.
Figure 4: Phone Stand for GXV3175
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CONNECTING GXV3175 IP MULTIMEDIA PHONE
Figure 5: Connecting the GXV3175
WELCOME SETUP GUIDE
After the GXV3175 completes the touch screen collaboration a welcome screen will appear that will allow
the user to access the Setup Guide. In this Setup Guide, a user can configure account information,
network settings, and upgrade settings.
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Figure 6: Setup Guide Welcome Screen
After hitting “Next”, the user will reach the Account Settings screen. By default, the IPVideoTalk account is
set to Account 1 slot. Users may choose to remove this account by inputting their own account
information, or keep this account and select “Create one more account” to create a new account. The
GXV3175 allows up to 3 accounts.
Figure 7: Setup Guide Account Settings
The following screen that appears is the Network settings. Here the user can select between DHCP,
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PPPoE, or Static IP.
Figure 8: Setup Guide Network Settings
After pressing “Next”, the user will be brought to the Upgrade Settings. This section allows the user to
choose where the phone will send upgrade requests to as well as configuration requests. The Automatic
Upgrade field allows the user to specify how often the phone should check for firmware/configuration.
Figure 9: Setup Guide Upgrade Settings
Once completed with all the Setup Guide steps, the user will be prompted to reboot the phone so that
changes can take effect immediately.
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Figure 10: Quick Setup Guide Completion
MAKING THE FIRST CALL WITH IPVIDEOTALK SERVICE
GXV3175 offers users the ability to send and receive free lifetime video calling to other GXV3175/GXV3140
on the IPVideoTalk network. Once the GXV3175 is connected, video calls can be made using
Grandstream’s configuration peer-to-peer SIP technology and IPVideoTalk service. To place the first video
call, complete the following three simple steps:
1) Connect the handset, network cable and power adapter to the GXV3175 using the methods
described above. After the phone boots up and obtains an IP address through DHCP, it will attempt
to register to the IPVideoTalk network. When the user boots the phone for the first time, Account 1
will be registered to the IPVideoTalk server with a provisioned extension number 810xxxx.
2) If the registration is successful, the “IPVideoTalk” text on the LCD will turn green and an assigned
IPVideoTalk number (810xxxx) specific to this phone will be displayed. At this point, the phone is
ready for making and receiving video calls over the IPVideoTalk network.
3) Tap on the account (See Figure 11) in the touch screen then the dial pad will show up for on-hook
dialing. Dial the number of any other phone registered on the IPVideoTalk network and press
“Video Call” or “Audio Call” button in the dial pad. If the user has purchased more
GXV3175/GXV3140 IP Multimedia phones, these phones are able to establish video calls
between each other and the user will immediately experience the plug and play nature of the
GXV3175.
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Figure 11: Select IPVideoTalk account in the marked area to make first call
USING THE GXV3175 IP MULTIMEDIA PHONE
LCD
When the phone is idle, the LCD screen will look similar to Figure 9. (The account and IP address may be
different from case to case, depending on the network environment and the settings.)
Figure 12: GXV3175 LCD Idle Screen
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Users could hold and drag on the center area of the main screen then slide from right to left to switch the
screen to view the different desktop widgets in the four screens. In the left hand side of the status bar, there
are four (4) green tabs indicating the current page of the screen.
Figure 13 shows the screen that displays weather and currency information.
Figure 13: LCD IDLE screen displaying Weather and Currency Information
Move around all the way to left to switch to screen that displays News Videos and Stock information. See
Figure 14.
Figure 14: LCD Idle screen displaying News Videos and Stock information
Switch again to the last screen that shows Calendar and RSS News. See figure 15.
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Figure 15: LCD Idle screen displaying Calendar and RSS News
In Figure 15, on the button menu of each screen, users could select “Menu” to access the full menu, select
“Phone” icon to make call, select PIP/News Videos/Youtube/Internet Radio/Brower to access the
corresponding applications. If the users press
button on the left, new widgets which include Info,
Weather, RSS News, Contacts and Internet Radio will be available to open and display in the main screen.
(See Figure 16)
Figure 16: Open new widget in Idle screen
Users may close/configure the widget by holding the finger on the widget area until close/setting options
show up on the right (Marked in Figure 17).
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Figure 17: Close/Configure the widget in Idle screen
By selecting the right hand side area of the status bar, users will be able to set the phone’s status in the
popped out dropdown list. See Figure 18.
Figure 18: LCD Idle screen displaying status bar settings
Table 5: LCD Icon Definition
Type
Icon
Network
Definition
Note
Connected to the network
Failed to connect to the network
Call Related
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Account 1 Auto Answer
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Account 2 Auto Answer
Account 3 Auto Answer
Account 1 Call Forward
If call forward is on: unconditional
Account 2 Call Forward
forward, forward when busy or
Account 3 Call Forward
forward on no answer.
Ring Volume
Ring volume, ranges from 0-9
Phone Status
Camera disabled.
Do-Not-Disturb (DND)
Missed Call.
Audio Output
The headset is in use and is off hook.
The headset is plugged in.
The phone is on speaker.
The handset is off hook.
WiFi
WiFi signal strength, ranging from 0 to 4
Message
Unread message
The icon is displayed at the right
Message box is full
hand side of the status bar
Applications
Unread text message.
Alarm clock
Background running program
The icon is displayed at the
center of the status bar
External devices
SD card
USB flash drive.
USB keyboard
USB mouse
MULTIPLE SIP ACCOUNTS AND LINES
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Figure 19: GXV3175 Dial Pad
Table 6: GXV3175 Dial Pad Definition
Item
Definition
Line/Account selection. Three independent SIP accounts may be configured, and the Line options
can be used to choose the line/account for audio/video call.
Account option. Dropdown list will show all the registered account for selection.
Call option. Dropdown list will show three types of call: Call, Paging and IP Call for direct IP Cal.
Audio output. Three types of audio output are available: speaker, handset and headset.
Call volume. Press
Standard keypad.
Delete input number. Press to delete the input number when dialing out.
Contacts. Press to open contact list.
Call History. Press to open call history.
10
Message. Press to open message box.
11
Voice Mail. Press to access voice mail box.
12
BLF. Press to access BLF.
to turn down or press
to turn up the volume.
The GXV3175 supports up to 3 independent SIP accounts. Each account may have separate SIP servers,
usernames and NAT configurations. The registered account will display in the idle screen by default. When
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the phone is off-hook, users could press Line1/Line 2/Line 3 or the account option in the dial pad shown in
Figure 19 to switch between different SIP accounts.
Incoming calls through the three accounts will try to use the corresponding line. If this line is busy, the
green icon for the corresponding line will become red. When a call comes in and the line is busy, the next
idle line will be used.
HANDSET, SPEAKER AND HEADSET MODE
The GXV3175 allows users to switch from handset to speaker or headset by pressing corresponding icons
shown in Dial Pad (Table 6, Item 5). If headset is plugged in, the headset icon will be activated to select.
MAKING A CALL
There are several ways to make a call.
1. By selecting account displayed in idle screen
Users could press the account area displayed in idle screen to make a call via the selected
account.
• In idle screen, select the registered account by touching the marked area in Figure.
• The dial pad will show up with the selected account.
• Dial the number and press “Audio Call” or ”Video Call” to dial out. Or press “Redial” for the
last dialed number.
Figure 20: Making a call by select account displayed in idle screen
2. By “Phone” in idle screen
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• In idle screen, select the “Phone” icon
in the bottom menu.
• The dial pad will show up with dial tone.
• Dial the number and press “Audio Call” or “Video Call” to dial out. Or press “Redial” for the
last dialed number.
3. By taking handset off hook
Users will be able to make a call when the phone is idle or running other applications by taking the
handset off hook.
• Take the handset off hook then the dial pad will show up with dial tone.
• Select line/account.
• Press “Redial” or dial the number and press “Audio Call”/”Video Call” to dial out.
4. By Call History
• Access the Phone Menu by pressing “MENU” icon in idle screen, and then select “Call
history”.
• The LCD monitor will display “All” calls, “Received” calls, “Dialed” calls and “Missed calls”.
Select the call history that you wish to view.
• Press
to dial the selected number. (See figure 21)
Figure 21: Dial by Call History
5. By Contacts
• Access the Phone Menu by pressing “MENU” icon in idle screen, and then select “Contacts”.
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Or choose “Contacts” icon in dial pad to access contact list.
• The LCD monitor will display the contact list. Scroll to the contact to be dialed.
• Press
to call the select contact. The primary number of the contact will be dialed out.
See Figure 22.
Figure 22: Dial by Contact List
• If selecting the contact for full contact information, there will be four numbers available to dial
out: Home, Work, Mobile and Fax. Press
to dial the corresponding number. See
Figure 23.
Figure 23: Dial by Contact List – more contact information
6. By Contact Favorite in idle screen
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In idle screen, press
and select
to open the “Contacts” widget. Users could add the
contact to the favorite list. Select the contact and press
to dial out. (See Figure 24)
Figure 24: Dial by Contact Favorite in idle screen
7. By Messages
Users could access the Phone Menu by pressing “MENU” icon in idle screen, and then select
“Message”. Similar to dialing by Call History, press
to dial the selected contact in message
history.
DURING CALLS
ANSWERING CALLS
1. Incoming Video Call: When the phone rings, select “Accept Audio”, “Accept Video” or “Reject”.
(See Figure 25) Users could toggle among handset/speaker/headset to answer the call and adjust
the call volume by pressing the speaker icon as well.
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Figure 25: Answer incoming video call
2. Incoming Audio Call: When an audio call is coming in, select “Accept Audio” or “Reject” in the
screen shown as Figure 26.
Figure 26: Answer incoming video call
3. Missed Call: if the call is unanswered, a missed call message will show up in idle screen. (See
Figure 27). Users could press View to access the detail of the missed call.
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Figure 27: Missed Call Notification
CALL FUNCTION OPTIONS
Figure 28 shows the screen after the users answer the call. Press “Options” button
to
access all the call functions.
Figure 28: Use Call Options during a video call
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Figure 29: Call Function Options
Table 7: GXV3175 Call Function Description
OPTION MENU
FUNCTION
DESCRIPTION
PIP
Press to switch videos between the caller and callee
Hold
Press to hold the call
Mute
Press to mute/unmute the call
Volume
Press to increase or decrease volume
Conference
Press to make 3-way conference call
Snapshot
Press to take a snapshot of the current video. The
snapshot will be automatically saved to folder “screenshot”
in Tools->File Manager
Video off
Press to switch between Video off and Video on. In the
screen, the default picture will show instead of the video
when video is turned off
Turn up and turn down the video brightness
Video zoom in and turn out
DTMF Dial Pad
Press the number and # or * for DTMF
CALL HOLD
1. Call Hold：During a call, press “Hold” button to place the call on hold. The line icon will become
yellow after call hold. The following will be displayed on the screen. (See Figure 30)
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Figure 30: Call Hold
2. Call Resume: In Figure 30, press “Unhold” button to resume to the call.
3. Multiple Calls: After call hold, users could select another line to make calls. If there is another call
coming in, the user will be able to select “Accept” or “Reject” in the right hand side of the screen.
Accept new incoming call will put the previous call on hold. To toggling between several calls,
users may need to turn off the video first by tapping on “Video off” in “Option” (See Figure 29) if it is
a video call. Then the following figure will show up for users to select between multiple lines. See
Figure 31.
Figure 31: Multiple Calls
CALL TRANSFER
1. Blind Transfer: During a call, press the “Transfer” button to place the other party on hold. The
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phone will display the following message: “Dial Number (Blind) OR Select Line (Attended)”. (See
Figure 32). Dial the extension number and press “Send”. This will transfer the call to the other
party immediately.
Figure 32: Blind Transfer
2. Attended Transfer: During a call, select another line to establish call with the third party using the
same account. This will put the previous line (Line 1) on hold. Now, press the “Transfer” button and
the message: “Dial Number (Blind) OR Select Line (Attended)” will show up in the screen. Then
select Line 1 to finish attended transfer. (See Figure 33)
Figure 33: Attended Transfer
NOTE:
•
To transfer calls across SIP domains, SIP service providers must support transfer across SIP
domains.
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•
If users enter a wrong number and wish to cancel the transfer, just press the “Cancel” in the screen
shown in Figure 32. Then the call with the first party will resume automatically.
•
If the user is on a video call with the first party, the users may need to turn off the video by pressing
“Video off” or pressing “Hold” to activate the screen to select another line.
3-WAY CONFERENCE
The GXV-3175 supports 3-way video conferencing. See Figure 34.
Figure 34: 3-Way Conference
1. Initiate a Conference Call: During a call, select another line by pressing “Line” button to call the
second party using the same account. This will place the first call on hold. Once the user has
established the second call, press the “Conference” button then select the line on hold by pressing
the corresponding “Line” button. This would bring the three parties together in a 3 way conference.
2. Cancel the Conference: If after pressing the “Conference” button, a user decides not to
conference anyone, press “Cancel” to cancel the conference.
3. End the Conference: There are two ways to end a conference: The first way is to press “END” in
the conference call. The second way to end a conference is to simply hang up and terminate the
call.
NOTE:
•
•
During the conference, users can see three-way videos if it is video call.
To activate the call options during the conference, users could simply tap on the screen then the
options will show up. Press PIP button to change the layout of the video display for the three
parties.
•
During the three way conference, if the initiator of the conference hangs up, the conference will
end. If users wish to allow the remaining two parties to stay in conversation after the initiator hangs
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up, the conference initiator should set “Transfer on Conference Hangup” to “Yes” in the web
configuration interface. This would allow the transfer of call to the remaining party after the initiator
exits the conference.
•
If the user is on a video call with the first party, the users may need to turn off the video by pressing
“Video off” or pressing “Hold” to activate the screen to select another line.
VOICEMAIL（MESSAGE WAITING INDICATION）
If the blue Message Waiting Indication (MWI) LED icon is flickering in the HOME button, a new voice mail
has been received and is waiting to be retrieved. To retrieve the voice mail, activate dial pad first. Then dial
the voicemail box number or press Voice Mail button
(the button needs to be configured in web
configuration interface first). The Interactive Voice Response (IVR) prompts the user through the message
retrieval process. The users may need to navigate to call function option
and use the DTMF
keypad for voicemail options.
NOTE: Each of the 3 accounts has its own voicemail. The Voicemail access number may be set up in the
“Voicemail ID” configuration under “Accounts” in the web configuration interface.
MUTE
1． During a call, the LCD screen will display “Mute”. Select the button to mute the call.
2． After the “Mute” button is pressed, the LCD screen will display “Unmute”. When this button is
pressed, the mute feature is cancelled.
KEYPAD INPUT
The build-in soft keypad in GXV3175 supports English and Chinese for text input. See Figure 35.
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Figure 35: GXV3175 Soft Keypad
Table 8: GXV3175 Keypad Definition
Item
Function
Caps lock.
Hide keypad
Switch between English/Chinese input.
Switch to characters with accents
Delete
Switch between Letter/Number/Symbol
NOTE: External keyboard could also be used by plugging via USB port on the phone.
CALL FEATURES
The following table shows the codes for call features. To use the code, select the line you wish to use by
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pressing the corresponding Line button and enter the feature code.
Table 9: GXV3175 Call Features
Code
Feature
*01
Select the preferred video/audio codec used for the call. Dial *01 + codec feature code +
Phone/Ext. Number (See Table 8 for codec feature code)
*02
Force the video/audio codec used for the call. Dial *02 + codec feature code + Phone/Ext.
Number (See Table 8 for codec feature code)
*30
Block Caller ID (for all subsequent calls)
*31
Send Caller ID (for all subsequent calls)
*50
Disable Call waiting (for all subsequent calls)
*51
Enable Call Waiting (for all subsequent calls)
*67
Block Caller ID (per call): Dial *67 + Phone/Ext. Number (no dial-tone in between).
*82
Send Caller ID (per call): Dial *82 + Phone/Ext. Number (no dial-tone in between).
*83
Send audio only: Dial *83 + Phone/Ext. Number (no dial-tone in between)
*84
Send audio and video: Dial *84 + Phone/Ext. Number (no dial-tone in between)
*70
Disable Call Waiting (per call): Dial *70 + Phone/Ext. Number (no dial-tone in between).
*71
Enable Call Waiting (per call): Dial *71 + Phone/Ext. Number (no dial-tone in between).
*72
Unconditional Call Forward: Dial *72 + Phone/Ext. Number followed by the # key. Wait for a
dial-tone and then hang up (dial-tone means input is successful).
*73
Cancel Unconditional Call Forward: Dial *73 and wait for a dial-tone before hanging up.
*90
Busy Call Forward: Dial *90 + Phone/Ext. Number followed by the # key. Wait for a dial- tone
and then hang up.
*91
Cancel Busy Call Forward: dial *91 and wait for a dial-tone before hanging up.
*92
Delayed Call Forward: Dial *92 + Phone/Ext. Number followed by the # key. Wait for a dial-tone
and then hang up.
*93
Cancel Delayed Call Forward: Dial *93 and wait for a dial-tone before hanging up.
Table 10: GXV3175 Audio/Video Codec Feature Code
Codec
Feature Code
PCMU
7110
PCMA
7111
G723
723
G726-32
72632
G729
729
G722
722
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GSM
7200
L16-256
7202
H.264
264
H.263
263
H.263+
2631998
GXV3175 WEB CONFIGURATION INTERFACE
The GXV-3175’s embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded
HTML pages allows users to configure the IP Multimedia Phone through a Web browser such as
Microsoft’s IE or Mozilla Firefox (Java Script must be enabled).
Figure 36: Web Browser Interface for GXV3175
ACCESSING THE WEB CONFIGURATION PAGES
The GXV3175 Web Configuration Interface URL is http://Phone-IP-Address, where the Phone-IP Address
is the IP address displayed on the phone’s LCD screen.
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To access the phone’s Web Configuration Menu:
•
Connect the computer to the same network as the phone.
•
Make sure the phone is turned on and shows its IP-address on the LCD display.
•
Open a Web-browser on your computer.
•
Enter the phone’s IP-address in the address bar of the browser.
•
Enter the administrator’s login and password to access the Web Configuration Menu.
1. The computer has to be connected to the same sub-network as the phone. This is easily done by
connecting the computer to the same hub or switch as the phone it is connected to. In absence of
a hub/switch (or free ports on the hub/switch), please connect the computer directly to the phone
using the PC-port on the phone.
2. If the phone is properly connected to a working Internet connection, the phone will display its IP
address. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255.
Users will need this number to access the Web Configuration Menu. For example, if the phone
shows 192.168.0.60, please enter “http://192.168.0.60” in the address bar of the browser.
3. The default login name for the administrator is “admin”. The default administrator password is set
to “admin”. The default login name for end-users is “user” while the default user password is set to
“123”.
NOTE: When changing any settings, always SUBMIT them by pressing the SAVE button on the bottom of
the page. For those settings that are shown in the web user interface (UI) with a star “*” next to it, users
must reboot the phone for the changes to take effect.
DEFINITIONS
This section describes the options in the Web configuration user interface. As mentioned, you can log in as
an administrator or an end-user.
•
Status: Displays the Account status, Network status, and System info of the phone.
•
Account (1~3): To configure each of the SIP accounts.
•
Advanced Settings: To configure General settings, Call Features, Video Settings, and Ring
Tones.
•
Maintenance: To configure Network Settings, WiFi Settings, 3G Settings, Time Settings,
Web/Telnet Access, Upgrade, Syslog, Debug, Language, Network Manager, OpenVPN Settings
and Device Manager.
STATUS PAGE DEFINTIONS
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Status/Account Status
Account
Shows the status of the 3 accounts.
Number
Shows the extension number of the SIP account.
SIP Server
Shows the URL/IP address and port of the SIP server.
Status
Shows the status of the account.
Status/Network Status
MAC Address
The device ID, in HEX format. This is a very important ID for ISP
troubleshooting.
Address Type
This field shows the type of address configured: DHCP, Static IP or PPPoE.
IP Address
This field shows the IP address of the GXV-3175.
Subnet Mask
This field shows the subnet mask of the GXV-3175.
Default Gateway
This field shows the Gateway of the GXV-3175.
DNS Server
This field shows the DNS Server of the GXV-3175.
NAT Type
This field shows the type of NAT configured.
Status/System Info
Product Model
Defines the product model: GXV3175.
Hardware Revision
Hardware version number: Main Board, Interface Board.
Part Number
This field contains the product part number.
Program Version
• Program: This is the main firmware release number, which is always used
for identifying the software (or firmware) system of the phone.
System Up Time
This field shows system up time since the last reboot.
ADVANCED SETTINGS PAGE DEFINITIONS
Advanced Settings/General Settings
Local RTP Port
This parameter defines the local RTP-RTCP port pair used to listen and
transmit. It is the base RTP port for channel 0. When configured, for audio,
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channel 0 will use this port _value for RTP and the port_value+1 for its
RTCP; channel 1 will use port_value+4 for RTP and port_value+5 for its
RTCP. For video, channel 0 will use port_value+2 for RTP and port_value+3
for its RTCP; channel 1 will use port_value+6 for RTP and port_value+7 for
RTCP. The default value is 5004.
Use Random Port
When set to YES, this parameter will force random generation of both the
local SIP and RTP ports. This is usually necessary when multiple
GXV-3175s are behind the same full cone NAT. The Default setting is YES.
(This parameter must be set to NO before Direct IP Calling will work)
Keep-alive Intervals (s)
This parameter specifies how often the GXV-3175 sends a blank UDP
packet to the SIP server in order to keep the “ping hole” on the NAT router to
open. The default setting is 20 seconds.
STUN Server
The IP address or Domain name of the STUN server. STUN resolution
results are displayed in the STATUS page of the Web UI. Only
non-symmetric NAT routers work with STUN.
Default STUN server: stun.ipvideotalk.com
Use NAT IP
The NAT IP address used in SIP/SDP messages. This field is blank at the
default settings. This should ONLY be used if your ITSP requires it.
SSL Certificate
This defines the SSL certificate needed to access certain websites.
SSL Private Key
This defines the SSL Private key.
SSL Private key Password
This defines the SSL private key password.
Advanced Settings/Call Features
Disable Call-Waiting
The default setting is No. If set to Yes, the call waiting feature will be
disabled.
Disable Call-Waiting Tone
The default setting is No. If set to Yes, the call waiting tone will be disabled.
Disable Direct IP Call
The default setting is No. If set to Yes, Direct IP calling will be disabled.
Offhook Auto Dial
Configure a User ID/extension to dial automatically when the phone is taken
offhook. By default, the phone will use the first account to dial out.
Advanced Settings/Video Settings
Enable Motion Detection
The phone will exit idle/screensaver mode if motion is detected. The default
setting is Yes.
Video Frame rate
The Default value is 15 frames/second. The video frame rate is adjustable
based on network conditions.
Increasing the frame rate will increase the amount of transferred data
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significantly therefore consuming more bandwidth. Lack of bandwidth will
impair the video due to packet loss.
Video Packet Size
The Default value is 1400, range from 100 to 1400.
It is recommended to use 600~800 if you have an Outbound Proxy or Media
Gateway.
Video Rate Control
Frame, TMN8 or GOP. The Default setting is Frame.
TMN8 is good for bandwidths larger than 384kbps
Video Frame Skipping
Skips bad video frames as they are received
The Default setting is No.
I-Frame Reference Only
When enabled, all P frames will only have reference to previous I frames.
This method may improve video quality when the network has heavy packet
loss. It will have modest increase in bandwidth usage. The Default setting is
No.
Packetization-Mode
The packetization mode (0 or 1) for the H.264 video packets. The default
setting is 0.
Redundant P-frame
This setting allows the phone to send redundant P-frames of I frames. When
an I frame is delayed or has packet loss, the redundant P-frames will be
used instead. It may make video phone call less prone to errors and thus
make the video stream more robust towards error propagation, but may
increase bandwidth usage. The default setting is set to No.
Adaptive MB Intra Refresh
This parameter specifies whether Adaptive MB Intra Refresh is used or not.
It may protect most important MBs (macro blocks) subjected to packet loss
and to avoid error propagation. The intra-refresh rate is adapted according
to the video packet loss rate. It may increase bandwidth usage. The default
setting is set to No.
Video Packet Loss Rate
Specifies the video packet loss rate for the Adaptive MB Intra Refresh.
Users can choose from: Less than 5%, Less than 10%, Less than 15%,
Less than 20%, Less than 25%. The default setting is set to Less than 10%.
Advanced Settings/Ring Tone
Call Progress Tones:
Using these settings, users can configure ring or tone frequencies based on
Dial Tone
parameters from the local telecom provider. By default, they are set to the
Ring Back Tone
Busy Tone
Reorder Tone
North American standard.
Frequencies
should
be
configured
with
known
values
avoid
uncomfortable high pitch sounds.
Confirmation Tone
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
Call-Waiting Tone
(Frequencies are in Hz and cadence on and off are in 10ms)
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PSTN disconnect Tone
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 repeat the pattern. Up to
three cadences are supported.
Default Ring Cadence
This defines the ring cadence for the phone. The default setting is:
c=2000/4000;
MAINTENANCE PAGE DEFINITIONS
Maintenance/Network Settings
Address Type
This selects the type of IP address assigned: DHCP, PPPoE or Static IP.
PPPoE
When PPPoE is selected, the user needs to enter the following: PPPoE
Account ID, PPPoE Password, PPPoE Service Name.
Static IP
When Static IP is selected, the user needs to enter IP address, Subnet
Mask, Default Gateway, DNS Server 1 and DNS Server 2.
Alternate DNS Server
This field sets the alternate DNS server for the user.
Layer 3 Qos
This field defines the layer 3 QoS parameter. It is the value used for IP
Precedence, Diff-Serv or MPLS. The Default value is 0.
Layer 2 QoS
This field contains the value used for layer 2 VLAN tagging. The Default
802.1Q/VLAN Tag
value is 0.
Layer 2 QoS
This assigns the priority value of the Layer2 QoS packets. The Default value
802.1p Priority Value
is 0.
802.1x Mode
This field sets 802.1x Mode. The default value is disabled. If EAD-MD5 is
selected, users will be required to enter Identity and MD5 Password.
Proxy
This field sets HTTP Proxy, HTTPS Proxy, FTP Proxy or No Proxy. Using
HTTP Proxy, other protocols can be applied by checking the box “Apply to
Other Protocol”.
Maintenance/Wifi Settings
WiFi Functions
This parameter enables/disables the WiFi function. The default setting is set
to “Disable”.
Wireless Mode
This parameter defines the wireless mode used. The GXV3175 supports
802.11b/g/n
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Network Type
This parameter defines the network type of the wireless network: Ad-hoc or
AP/Managed. By default, it is set to “AP or Managed”.
Channel
This parameter defines the channel (1-10) for the wireless network. The
default setting is set to “Auto”.
ESSID
This parameter sets the ESSID for the Wireless network. Press “Scan” to
scan for the available wireless network and add it. Users can also press
“Add” to add the ESSID directly.
Security Mode
This parameter defines the security mode used for the wireless network.
The following are supported: WEP/Shared, WEP/OPEN, WPA PSK TKIP,
WPA PSK AES, WPA2 PSK TKIP, WPA2 PSK AES. The default setting is
set to “Disabled”.
Users will need to enter the corresponding authentication password for the
security mode.
Maintenance/3G Settings
3G
USB
Configuration
Click Here to download Config file and Script
File
3G USB Connect Script
Click Here to download Config file and Script
Note:
The user is required to have the config file for the modem and a script.
Please make sure that the used 3G Modem is support by the GXV3175(list
of supported 3G modems below)
WCDMA
•
Huawei E1750
•
Huawei E180
•
Sierra Wireless 306
CDMA2000
•
ZTE AC581
•
Huawei EC1260
EVDO
•
Novatel Ovation MC760
Maintenance/Time Settings
NTP Server
This parameter defines the URL or IP address of the NTP (Network Time
Protocol) server. The GXV3175 phone may obtain the date and time from
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the server.
The default setting is ntp.ipvideotalk.com.
DHCP Option 42 override
Select Yes for the user to allow DHCP Option 2 to override the NTP server if
NTP server
there is one on the LAN. The default setting is NO.
Time Zone
This parameter controls the date/time display according to the specified time
zone.
Time Display Format
This parameter sets time display format to 12 Hour or 24 Hour
Date Display Format
This parameter sets date display format to YY-MM-DD or MM-DD-YY or
DD-MM-YY.
Self-Defined Time Zone
This parameter allows the users to define their own time zone.
Syntax: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M3.2.0,M11.1.0, MTZ+6MDT+5
This indicates a time zone with 6 hours offset with 1 hour ahead which is
U.S central time. If it is positive (+) if the local time zone is west of the Prime
Meridian (A.K.A: International or Greenwich Meridian) and negative (-) if it is
east.
M3.2.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)
The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd
Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues,..,Sat)
Therefore, this example is the DST which starts from the second Sunday of
March to the 1st Sunday of November.
Maintenance/Web and Telnet Access
Disable Telnet
The default value is No. If set to YES, the GXV3175 will not allow any telnet
access to the phone.
Access Method
This defines the access method for web access: HTTP or secure HTTPS
protocol.
Port
This defines the port for web access. By default, HTTP uses port 80 and
HTTPS uses port 443. This field is for customizable web ports.
Admin Password
This defines the administrator password for web interface access. Only the
administrator can configure the “Advanced Settings” and “Account x” pages.
The password is case sensitive and the maximum password length is 25
characters.
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User Password
This defines the user password for web interface access.
Maintenance/Upgrade
Lock Keypad for Update
The default value is NO. If set to YES, the keypad will be disabled from
making any updates.
XML Config File Password
The password used for encrypting the XML configuration file using
OpenSSL. This is required for the phone to decrypt the encrypted XML
configuration file.
HTTP/HTTPS User Name
The user name for the HTTP server.
HTTP/HTTPS Password
The password for the HTTP server.
Upgrade Via
This field allows the user to choose the firmware upgrade method: TFTP,
HTTP or HTTPS.
Defines the server path for the firmware server. It can be different from the
Firmware Server Path
Configuration server which is used for provisioning.
Config Server Path
This is the server path for provisioning; it can be different from the firmware
server.
Firmware File Prefix
This field enables your ITSP to lock firmware updates. If configured, only the
firmware with the matching encrypted prefix will be downloaded and flashed
into the phone.
Firmware File Postfix
This field enables your ITSP to lock firmware updates. If configured, only the
firmware with the matching encrypted postfix will be downloaded and
flashed into the phone.
Config File Prefix
This field enables your ITSP to lock configuration updates. If configured,
only the configuration file with the matching encrypted prefix will be
downloaded and flashed into the phone.
Config File Postfix
This field enables your ITSP to lock configuration updates. If configured,
only the configuration file with the matching encrypted postfix will be
downloaded and flashed into the phone.
DHCP Option 66 override
The Default setting is YES. If DHCP option 66 is enabled on the LAN side,
Server
the TFTP server can be redirected. Please be very careful when configuring
this as the redirection could break the phone if this happens during the
firmware upgrade.
DHCP
Option
120
Override SIP Server
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Automatic Upgrade
The default value is NO. Choose “YES” to enable automatic HTTP upgrade
and provisioning.
Period time of upgrade
Specifies the time period to check for firmware upgrade. The default setting
checking
is 10080 minutes ( 7 days)
Hour of the day (0-23)
Defines the hour of the day to check the HTTP/TFTP server for firmware
upgrades or configuration files changes.
Day of the week (0-6)
Defines the day of the week to check the HTTP/TFTP server for firmware
upgrades or configuration files changes.
Automatic Upgrade Rule
Defines the rules for automatic upgrade: Always Check, when F/W
suffix/prefix changes, Skip the Firmware Check.
Authenticate Conf File
Authenticate configuration file before acceptance. The Default setting is NO.
Factory Reset
Restore to factory default settings. Users can choose from: Full Reset, VoIP
Data Reset, and User Data Reset. After the selection is made, press the
“Reset” button to factory reset the phone.
NOTE: Users can use the backup/restore functionality of the phone prior to
the factory reset.
Maintenance/Syslog
Syslog Server
The URL/IP address for the syslog server.
Syslog Level
Select the level of logging for syslog. The default setting is NONE.
There are 4 levels: INFO, DEBUG, ERROR and WARNING.
Syslog messages are sent based on the following events:
Product model/version on boot up (INFO level)
NAT related info (INFO level)
Sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
Inbound and outbound calls (INFO level)
Registration status change (INFO level)
Negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
Memory exception (ERROR level).
Maintenance/Debug
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Capture Trace
Press START to start capturing a trace, and press STOP to abort the
capture process.
View Trace
Press LIST to view the list of captured traces. It will show the time the trace
was captured. Click on the name of the file to download and view the trace.
Enable Application Debug
When set to YES, the trace captured will contain debug messages.
The default setting is NO.
Show All Version
Default is set to NO.
When set to yes, all debug messages including
syslog will be captured in the trace.
Maintenance/Language
Language
Selects the language setting for the phone. The current version supports
English, Traditional Chinese and Simplified Chinese.
Maintenance/Network Manager
ACS URL
Enter URL for TR-069 Auto Configuration Servers (ACS)
ACS Username
Enter ACS username
ACS Password
Enter ACS password
Periodic Inform Enable
Enable periodic inform. By default, it is disabled
Periodic Inform Interval(s)
Enter periodic inform interval(s). By default, it is set to 300s.
Connection
Request
Enter connection request username
Request
Enter connection request password
Username
Connection
Password
Connection Request Auth
Select the authentication method among “No authentication”, “Basic”, or
Mode
“Digest”.
Connection Request Port
Enter the connection request port
Maintenance/OpenVPN Settings
OpenVPN Enable
This enables/disables OpenVPN functionality. The default setting is No. To
use OpenVPN functionality, it requires user to have access to an OpenVPN
server.
OpenVPN Server Address
The URL/IP address for the OpenVPN server.
OpenVPN Server Port
The network port for the OpenVPN server. By default, it is set to 1194.
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OpenVPN CA
OpenVPN CA file (ca.crt) required by the OpenVPN server for
authentication purposes. Press “Upload” to upload the corresponding file
to the phone.
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 phone.
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 phone.
NOTE: To use OpenVPN functionalities, users must enable OpenVPN and configure all of the settings
related to OpenVPN, including server address, 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). Reboot the phone after these settings are saved for the changes to take effect. Once the phone
connects to the OpenVPN server and the authentication is correct, the phone will show the VPN IP
address under the Status-> Network Status, but it will still display the LAN IP on the LCD of the phone.
Maintenance/Device Manager (LCD)
Screen Saver Timeout
This sets the lapsed idle time needed for the screen saver to appear. The
(m)
default setting is 5 minutes.
Screen Saver Interval
The screen saver is a picture slide show. This defines the frequency to
change the picture on the screen for the screen saver.
(s)xd
The default setting is 10 seconds.
LCD Auto Power Off Time
This defines the lapsed idle time needed for the LCD to power off
(m)
automatically in order to save power. The default setting is 10 minutes.
Maintenance/Device Manager (Headset)
Headset TX Gain (dB)
Transmission gain. It’s a headset setting to control the voice intensity. The
default setting is 0dB.
Headset RX Gain (dB)
Receive gain. It’s a headset setting to control the voice intensity. The default
setting is 0dB.
Maintenance/Device Manager (Camera)
Zoom mode
Select the desired camera zoom mode (Tele, optical, digital). Can adjust
local video zoom either in mirror mode or during the call (on the fly) by
pressing the Left or Right Arrow Key.
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Exposure Setting
Select the desired Camera Exposure mode based on the surroundings. The
default setting is set to Auto. The default setting is 8.
Color Mode
Select the desired Camera Color Mode (Color or Monochromatic). The
default setting is set to Color.
White Balance
Select the desired Camera White Balance to be automatic or Fixed. The
default setting is set to Auto.
Lens Correction
Correct Camera Lens noise. The default setting is set to YES.
It is recommended to set this parameter to YES to reduce camera lens noise
and improve video quality.
Flicker Control
Reduces flicker of image shot with a video camera caused by luminance
fluctuation of an illuminating light source. Users can select 50Hz/60Hz
depending on the frequency of the commercial electrical power.
The default setting is set to Auto.
Maintenance/Device Manager (TV Port)
TV Output
Select the desired TV Output type (PAL or NTSC). NTSC 4:3 by default.
SIP ACCOUNT CONFIGURATION PAGE DEFINITIONS
The GXV3175 has three lines that can be configured to accommodate independent SIP accounts. Every
SIP account has an individual configuration page.
Account/General Settings
Account Active
Select YES to enable and use the Line.
Account Name
The account name for the user that is displayed on the LCD screen.
SIP Server
The IP address of the SIP server or the given domain name of the VoIP
service provider.
SIP User ID
User account information provided by the VoIP service provider; normally
similar to a telephone number or an actual telephone number.
Authenticate ID
The authenticate ID for the SIP user. It can differ or be the same as the SIP
user ID.
Authenticate password
The password that the GXV uses to authenticate with the ITSP (SIP) server.
After it is saved, this will appear as blank for security reasons. The
maximum length is 25 characters.
Voice Mail User ID
When this is configured, the user can dial to the voicemail server using the
MESSAGE button. This ID is normally the feature code for Voice Mail.
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Name
The Caller ID that will be displayed for the account.
Tel URI
The default setting is “Disable”. If the Video phone has an assigned PSTN
number, this field should be set to “Enable”. If “User=Phone” is set, a
“User=Phone” parameter will be attached to the “From header” in the SIP
request to indicate the E.164 number.
Account/Network Settings
Outbound Proxy
IP address or Domain name of the Outbound Proxy, or Media Gateway, or
Session Border Controller. Used by the GXV3175 for firewall or NAT
penetration in different network environments. If a symmetric NAT is
detected, STUN will not work and ONLY an Outbound Proxy will work.
DNS Mode
The default is set to A Record. If the user wishes to locate the server by DNS
SRV, the user may select SRV or NATPTR/SRV.
NAT Traversal
This setting decides whether the NAT traversal mechanism is activated. If it
is set to “STUN” and STUN server is configured, the GXV3175 will route
according the STUN server. In this mode, the STUN client embedded in the
phone will communicate with the appointed STUN server to examine which
type of Firewall/NAT setting is employed. If the type of NAT detected is Full
Cone, Restricted Cone or Port-Restricted cone, the phone will try to use
public IP addresses and port in all the SIP and SDP messages.
If the “NAT Traversal” is configured to be “Keep-alive”, the phone will send
an empty SDP packet (without payload data) to the SIP server once in 20
seconds to keep the NAT port open.
If VPN is used, users should select “VPN” for NAT Traversal.
Users could also set “Auto” or “UPnP” according to the network
configuration.
If an outbound proxy server is used, please configure this to be “NAT NO”.
Proxy Require
Notifies the SIP server that the unit is behind a NAT/Firewall.
Account/SIP Settings
SIP Registration
Selects whether or not the GXV3175 will send a SIP Register message to
the proxy; The default setting is YES.
Unregister On Reboot
If set to YES, the SIP user registration messages will be cleared on reboot.
The default setting is YES.
Register Expiration (m)
This defines the expiration time of the registration in minutes. The default
value is 50 minutes; the maximum value is 65535 minutes (approximately
45 days) and the minimum value is 5 minutes.
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Wait
Time
Retry
The wait time for registration retry if registration fails. The default is set to 20
Registration (s)
Seconds.
Local SIP Port
The local SIP port for sending and receiving data. The default settings for
Account 1/2/3 are 5060/5062/5064.
Subscribe for MWI
Default is set to No. When selected, a SUBSCRIBE for Message Waiting
Indication will be sent out periodically.
Session Expiration (s)
Session Expiration is the time (in seconds) at which the session is
considered timed out if no successful session refresh transaction occurs
beforehand. The default value is 180 seconds.
MIN-SE (s)
The minimum session expiration (in seconds). The default value is 90
seconds.
UAC Specify Refresher
As a Caller, select UAC to use the phone as the refresher, or select UAS to
use the Callee or proxy server as the refresher.
UAS Specify Refresher
As a Callee, select UAC to use the caller or proxy server as the refresher, or
select UAS to use the phone as the refresher.
Force INVITE
The Session Timer can be refreshed using the INVITE or UPDATE
methods. Select “Yes” to use the INVITE method to refresh the session
timer
Caller Request Timer
The default setting is Yes. If Yes is selected, the phone will use the session
timer when it makes outbound calls if remote party supports session timer.
Callee Request Timer
If Yes is selected, the phone will use the session timer when it receives
inbound calls with session timer request.
Force Timer
If Yes is selected, the timer will be activated even if the other party does not
support session timers. If No is selected, the timer will be activated only if
the other party supports the timer. To disable the timer, Caller Request timer,
Callee request timer and Force Timer should all be set to No.
Enable 100rel
The use of the PRACK (Provisional Acknowledgment) method enables
reliability to SIP provisional responses (1xx series). This is very important in
order to support PSTN internetworking. To invoke a reliable provisional
response, the 100rel tag is appended to the value of the required header of
the initial signaling messages. The default setting is No.
SIP Transport
Selects the SIP transport method. There are three types: TCP, UDP and
TLS.
Symmetric RTP
Selects whether or not symmetric RTP is supported.
Support SIP Instance ID
Selects whether or not SIP Instance ID is supported.
Validate
This configuration selects whether or not the incoming messages should be
Incoming
Messages
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SIP T1 Timeout
This configures the timeout period for SIP T1. The default setting is 0.5
Seconds.
SIP T2 Interval
This configures the time interval for SIP T2. The default setting is 4 seconds.
Remove OBP from route
The SIP Extension notifies the SIP server that it is behind a NAT/firewall.
Account/Codec Settings
Start Video Automatically
--YES: Accepts all incoming video, this is the default setting;
--NO: video is disabled.
--Enable/Disable during dialing: Dialing time choice to enable/disable video
DTMF
Assigns the mechanism to send DTMF. There are three supported modes:
in audio (with DTMF as audio signals), through RTP and through SIP INFO.
DTMF Payload Type
Sends DTMF using RFC2833. The default is 101.
Preferred Vocoder
This
products
supports
up
to
different
Vocoders:
G.711(a/μ)(PCMU/PCMA),GSM,G726-32,G.723.1,G.729A/B,
including
G.722
and
L16-256.
When this has been configured, the system will select the preferred vocoder
from the list and the same order is sent with the SDP messages.
The user can change the preferred vocoder priority order by toggling with
the up and down buttons. It is recommended to use the default setting
“PCMU”; if other vocoders are chosen, there could be an audio transmission
error.
Preferred Video Codec
Selects the preferred video codec from the available list. It is recommended
to use H.264.
Enable RFC5168 support
Select YES to enable RFC5168 support. The default is set to NO.
Video Bit Rate
Rate of information content in a video stream. (Ranges from 32 – 1024
kbps)
H.264 Image Size
Select the type of H.264 compression (4CIF, VGA, WQVGA, CIF, QVGA,
QCIF, QQVGA)
H.264Payload type
Selects the dynamic H.264 codec payload type (ranges from 96-127). The
default setting is 99.
H.263+ Payload type
Selects the H.263+ codec payload type (ranges from 96-127). The default
setting is 103.
L16-256 Payload type
Selects the L16-256 codec payload type (ranges from 96-127). The default
setting is 103.
H.263 Encoder Resolution
Selects the resolution (CIF/QCIF) used for H.263 codec. The default
resolution is CIF.
SRTP Mode
Grandstream Networks, Inc.
This is option to secure the RTP audio
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Silence Suppression
This controls the G723/G729 silence suppression/VAD. If set to YES, when
no audio is detected, the phone will send a small number of VAD packets
(instead of audio packets). If Set to NO, this feature is disabled. The default
setting is NO.
Voice Frame per TX
This configures the number of voice frames transmitted per packet. (It is
recommended to set this based on the Ethernet IS packet limitation – 1500
Byte or 120kb/s).
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 in payload call.
For example: if the codec is configured to be G.723 and “Voice Frames per
TX” is set to 2. Then in the INVITE request, the “ptime” value for SDP will be
60 ms. This is because the G723 voice frame is 30 ms. Similarly, if the
setting is 2 and the codec is G729 or G711 or G726, the “ptime” value for
SDP in the INVITE request will be 20 ms.
If the voice frames per TX configuration exceeds the maximum value
allowed, the phone will choose the preferred codec and use the maximum
value allowed.
The maximum value is 64 frames.
Adjusting these values will also change the jitter buffer. The phone has an
algorithm for setting the jitter buffer and the jitter buffer rate is around
20-200ms.
For end-users, it is recommended to use the default setting, as incorrect
settings may influence the quality of audio.
G723 Rate
This defines the encoding rate for G723 vocoder, and can be 6.3Kb/s or
5.3Kb/s. Please confirm with your service provider.
Jitter Buffer Type
Users can select either Fixed or Adaptive based on network conditions. The
default is Adaptive.
Jitter Buffer Length
User can select Low, Medium, or High based on network conditions. The
default is Medium.
Account/Call Settings
Dial Plan Prefix
Sets the prefix added to each dialed number.
Dial Plan
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;
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a) xx+ - at least 2 digit numbers
b) xx. - only 2 digit numbers
c) ^ - exclude
d) [3-5] - any digit of 3, 4, or 5
e) [147] - any digit of 1, 4, or 7
f)
<2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
• Example 1: {[369]11 | 1617xxxxxxx}
Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617
• Example 2: {^1900x+ | <=1617>xxxxxxx}
Block any number of leading digits 1900 or add prefix 1617 for any dialed 7
digit
numbers
• Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}
Allows any number with leading digit 1 followed by a 3 digit number, followed
by any number between 2 and 9, followed by any 7 digit number OR Allows
any length of numbers with leading digit 2, replacing the 2 with 011 when
dialed.
3. Default: Outgoing – {x+}
Allow any length of numbers.
Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right):
• ^1900x. - prevents dialing any number started with 1900
• <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by
dialing
7 numbers and 1617 area code will be added automatically
• 1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11
digits
length
• 011[2-9]x. - allows international calls starting with 011
• [3469]11 - allow dialing special and emergency numbers 311, 411, 611 and
911
NOTE: In some cases where the user wishes to dial strings such as *123 to
activate voice mail or other applications provided by their service provider,
the * should be predefined inside the dial plan feature. An example dial plan
will be: { *x+ } which allows the user to dial * followed by any length of
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numbers.
Early Dial
The default setting is NO. Use only if the proxy supports 484 response.
Refer to Use Target
The default is NO. If set to YES, then for Attended Transfer, the “Refer-To”
Contact
header uses the transferred target’s contact header information.
Auto Answer
The default setting is NO. If it is set to YES, the GXV3175 will play a “beep”
sound and answer the phone automatically on the speaker.
Send Anonymous
If set to YES, the “From” header in the outgoing SIP messages will be set to
anonymous to block the caller ID from displaying. The default setting is NO.
Anonymous Call Rejection
The default setting is NO；If set to YES，any anonymous call will be rejected
（with 486 Busy message）
Call Log
Options to log calls are Log All, Log Incoming/Outgoing Only(missed calls
will not be recorded), Disable All
Special Feature
Different soft switch vendors have special requirements; therefore users
may need to select special features to meet these requirements. Users can
choose from Standard, CBCOM, RNK, China Mobile, ZTE IMS and Mobotix
depending on their need.
The default setting is Standard.
Feature Key Sync
Default is “No”. This option is to synchronize DND/Call Forward features
with Broadsoft. When set to “Yes”, a SUBSCRIBE will be sent out
periodically to the server. Then when DND/Call Forward features Call
Forward No Answer, Unconditional Call Forward and Call Forward on Busy)
are configured or changed on the phone and the Broadsoft server side,
those features will be synchronized on the phone side and the Broadsoft
server side.
Enable Call Features
The default is set to YES. If it is set to NO, call features are disabled.
Call
Sets the number to forward to unconditionally.
Forward
Unconditional
Call Forward When Busy
Sets the number to forward to when busy.
Call Forward When No
Sets the number to forward to when no answer.
Answer
Delayed
Call
Forward
The wait time before transferring the call to another extension or to
Wait Time (s)
voicemail; the default setting is 20 seconds.
No Key Entry Timeout (s)
Defines the time out duration on no key entry.
Default setting is 4 seconds.
Ring Timeout (s)
Defines the time out duration on no answer.
Default setting is 60 second.
Transfer on Conference
Grandstream Networks, Inc.
Defines whether or not the call is transferred to the other party if the initiator
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Hangup
of the conference hangs up.
Default setting is set to No.
Use # as Dial Key
Selects whether or not # is used as Send.
Account ring tone
This allows the user to configure the general ringtone used for this account.
Users can choose from different ringtones from the dropdown menu. They
can also upload mp3 files in Ringtone manager for their own ringtone.
Ring Tone
Users can select up to 3 distinctive ringtones to match incoming caller ID.
Users can choose from different ringtones from the dropdown menu. They
can also upload mp3 files in Ringtone manager for their own ringtone.
SAVING THE CONFIGURATION CHANGES
After users makes changes to the configuration, press the “Save” button in the Configuration Menu. The
web browser will then display a message window to confirm the saved changes. Reboot the phone for the
new settings to take effect if the settings that you have changed have a * next to it.
REBOOTING FROM REMOTE LOCATIONS
Press the “Reboot” button at the top right-hand corner of the configuration menu to reboot the phone
remotely. The web browser will then display a message window to confirm with the user if they wish to
reboot the phone or not. Click on “Yes” and the phone will reboot. Wait approximately 2 minutes to log in
again.
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CUSTOMIZING THE GXV3175 IP MULTIMEDIA PHONE
XML GUI CUSTOMIZATION GUIDE
The GXV3140 IP Multimedia Phone allows users to customize the GUI desktop layout as well as GUI
configurations on the phone, offering users with flexibility and control. This includes display/hide certain
applications, configure parameters on the phone with specific configuration files, control the display
appearance and enable/disable some applications and much more.
For more details on the XML Based GUI Configuration on the GXV3175, please refer to:
XML GUI Customization Guide
GXV3175 XML GUI Customization Config Tool and Examples
GMI GUIDE
GMI (Grandstream Manager Interface) is a management API developed by Grandstream
Networks. Designed for our IP Multimedia phones, it allows partners to develop customized
applications on the phone.
GMI supports standard HTML/CSS/JavaScript, users can use these dynamic web page
development languages to develop their customized application. GMI will display the
application on the phone based on the web pages generated.
Additionally, GMI provides several basic API functions (packaged in JavaScript) to facilitate
users calling the existing applications on the multimedia phone or to obtain the phone status
etc. This allows users with basic web application programming skills to develop their
customized application on the multimedia phone, without the effort of learning a new
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programming language. All that is required for the user is to understand how the GMI works
and how to use these simple APIs to interact with the phone.
For more details on the GXV3175 GMI Configuration, please refer to:
GXV3175 GMI Guide
GXV3175 GMI Example 1
GXV3175 GMI Example 2
SOFTWARE UPGRADE
The GXV3175 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the
TFTP/HTTP/HTTPS server and selecting a download method. To configure a valid URL for TFTP or HTTP,
the server name can be the FQDN or IP address format. Examples of valid URLs：
1. fw.ipvideotalk.com/gs
2. 72.172.83.110
There are two ways to setup a software upgrade server: Via Menu or the Web Configuration Interface.
UPGRADE VIA MENU
In “Menu”->“System Settings”->“Maintenance”, users could configure the software upgrade server through
the menu in the phone. In the “Upgrade” subpage, users may then select the upgrade mode and enter the
IP address or FQDN of the Firmware server and the System Config server. After making the changes,
press save to apply the new settings and reboot the device for upgrading.
UPGRADE VIA WEB BROWSER CONFIGURATION PAGE
Open a web browser and enter the IP address for the GXV3175. Next, log in with the administrator
username and password. Under Maintenance, select “Upgrade and provisioning”, enter the IP address
or the FQDN (URL) of the upgrade server and choose to upgrade via TFTP or HTTP/HTTPS. Save the
changes and reboot the unit.
At this moment, the LCD will display the progress of the software download/upgrade. Please do not disrupt
or power down the unit.
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NOTE: The software upgrade will last up to 5 minutes (through the LAN side) and may take longer when
upgrading through the Internet (limited by the network bandwidth). Grandstream recommends
downloading and upgrading the software in a controlled LAN environment.
NO LOCAL TFTP SERVERS
For users without a local TFTP server, Grandstream offers a NAT-friendly TFTP server. This enables users
to download the latest software upgrades for their phone via this server. Please refer to the webpage:
http://www.grandstream.com/firmware.html.
Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A
free windows version TFTP server is available for download from
http://support.solarwinds.net/updates/New-customerFree.cfm or http://tftpd32.jounin.net/.
Instructions for local firmware upgrade:
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 GXV3175 device to the same LAN segment.
3. Launch the TFTP server and go to the File menu -> Configure -> Security to change the TFTP server's
default setting from "Receive Only" to "Transmit Only" for the firmware upgrade.
4. Start the TFTP server and configure the TFTP server in the phone’s web configuration interface.
5. Configure the Firmware Server Path to the IP address of the PC.
6. Update the changes and reboot the GXV3175.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use
Microsoft IIS web server.
RESTORE FACTORY DEFAULT SETTINGS
WARNING: Restoring the Factory Default Settings will delete all configuration information on the phone.
Please backup or print all the settings before you restore to the factory default settings. Grandstream is not
responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
There are two methods to restore the GXV3175 to the factory default settings.
RESTORE TO FACTORY DEFAULT SETTINGS VIA LCD MENU
The steps to restore the phone to factory default settings through the LCD menu are as follows:
Step 1: Select MENU to access the phone menu and select System Settings.
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Step 2: Select Maintenance and navigate to Upgrade page.
Step 3: Select the Factory Reset option.
Step 4: Press “OK”. The phone will reboot and perform a factory reset.
After factory reset, users will need to do calibration before the phone’s main screen show up.
RESTORE TO FACTORY DEFAULT SETTINGS VIA THE WEB INTERFACE
Step 1: Login to the system configuration page through a web browser and select “Maintenance”. From
the menu on the left, select “Upgrade”.
Step 2: At the bottom of the “Upgrade” page, click on the “Reset” button for Factory reset after selecting
the degree of factory reset (Full Reset/User Data Reset/VoIP Data Reset).
Step 3: Click on OK to restore the phone to factory settings.
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FCC
Warning
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.
Any Changes or modifications
not expressly approved by the party responsible for compliance could void the user's authority to operate
the equipment.
RF
This equipment complies with FCC’s and IC's RF radiation exposure limits set forth for an uncontrolled
environment. The antenna(s) used for this transmitter must be installed and operated to provide a
separation distance of at least 20 cm from all persons and must not be collocated or operating in
conjunction with any other antenna or transmitter. Installers must ensure that 20cm separation distance will
be maintained between the device (excluding its handset) and users.
15.105B
For a Class B digital device or peripheral, the instructions furnished the user shall include the following
or similar statement, placed in a prominent location in the text of the manual: Note: This equipment has
been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC
Rules. These limits are designed to provide reasonable protection against harmful interference in a
residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not
installed and used in accordance with the instructions, may cause harmful interference to radio
communications. However, there is no guarantee that interference will not occur in 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.
15.21 Statement:
Information to user: the user’s manual or instruction manual for an intentional or unintentional radiator
shall caution the user that changes or modifications not expressly approved by the party responsible for
ompliance could void a sample statement is below: Note: The manufacturer is not responsible for any
radio or TV interference caused by unauthorized modifications to this equipment. Such modifications could
void the user’s authority to operate the equipment.
Grandstream Networks, Inc.
GXV3175 USER MANUAL
FIRMWARE VERSION 1.0.1.1
Page 59 of 60
Updated : 3/2012
EXPERIENCING THE GXV3175 MULTIMEDIA PHONE
Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware
releases, additional features, FAQs, documentation and news on new products.
We encourage you to browse our product related documentation, FAQs and User and Developer Forum
for answers to your general questions. If you have purchased our products through a Grandstream
Certified Partner or Reseller, please contact them directly for immediate support.
Our technical support staff is trained and ready to answer all of your questions. Contact a technical
support member or submit a trouble ticket online to receive in-depth support.
Thank you again for purchasing Grandstream GXV3175 Touchscreen IP Multimedia phone, it is sure to
bring convenience and color to both your business and personal life.
Grandstream Networks, Inc.
GXV3175 USER MANUAL
FIRMWARE VERSION 1.0.1.1
Page 60 of 60
Updated : 3/2012

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