CALL FEATURES .................................................................................................................................................... 18
VIDEO SURVEILLANCE ......................................................................................................................................... 51
Thank you for purchasing the Grandstream GXV300x IP Video Phone. The GXV300x is a cost effective,
easy to use and easy to configure IP video communications solution. The GXV300x supports popular
voice codecs and H.264, H.263/H.263+ video codec. It is based on SIP industry standard and interoperable with 3
rd
P
P
party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology,
while visually commuting with anyone, anywhere in world.
This manual will help you learn how to operate and manage your GXV IP Video Phone and make the best
use of its many upgraded features including simple and quick installation, multi-party conferencing, and
direct IP-IP Calling.
GXV3000X S
ERIES OVERVIEW
Currently, the GXV300x IP Video Phone series consists of the following three models: GXP3000,
GXV3005 and GXV3006.
The GXV300x series is a next generation IP video phone based on SIP standard and H.264 video codec,
the codec of choice for high-quality video. It also supports H.263/H.263+ legacy video codec just for compatibility. The GXV300x supports real-time (up to 30fps) high-quality video at very modest bandwidth level
(as low as 32kbps, up to 1Mbps). The phone allows nearly all viewing angles via a large 5.6” TFTP adjustable LCD screen and a VGA camera, creating high-quality videoconferences from your desk.
The GXV300x combines great design and technology with excellent audio and picture quality, ease of
use and broad interoperability with 3rd party SIP products. It is ideal for any multi-media communication
environment, the virtual office and all IP video communications for business or residential users.
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 (one year) manufacturer warranty.
•This document is contains links to Grandstream GUI Interfaces. Please remember to download these
examples
•This document is subject to change without notice. The latest electronic version of this user manual
is available for download from the following location:
•Reproduction or transmittal of 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.
Grandstream Networks, Inc. GXV300x User ManualPage 2 of 54
Firmware 1.2.3.7 Updated: 12/2010
I
NSTALLATION
Unpack and check all accessories. Equipment included in the package:
The GXV300x IP Video Phone package contains:
1. One GXV300x Main Case
2. One Handset
3. One Phone Cord
4. One Universal Power Adaptor
5. One Ethernet Cable
6. Two Wall Mount Sustaining Brackets
T
S
AFETY COMPLIANCES
The GXV300x phone complies with FCC/CE and various safety standards. The GXV300X power adaptor
is compliant with UL standard. Only use the universal power adapter provided with the GXV package.
The manufacturer’s warranty does not cover damages to the phone caused by unsupported power adaptors.
T
W
ARRANTY
If you purchased your GXV300x from a reseller, please contact the company where you purchased your
phone for replacement, repair or refund. If you purchased the product directly from Grandstream, contact
your Grandstream Sales and Service Representative for a RMA (Return Merchandise Authorization)
number before you return the product. Grandstream reserves the right to remedy warranty policy without
prior notification.
Warning: Please do not attempt to use a different power adaptor. Using other power adaptor may dam-
age the GXV300x and will void the manufacturer warranty.
Grandstream Networks, Inc. GXV300x User ManualPage 3 of 54
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C
Video/Audio
GXV3006
GXV3005
3 5 1 4 8 7 6 2
ONNECTING YOUR
GXV IP V
F
IGURE 1: CONNECTING THE
IDEO PHONE
GXV300X
1 PC
2 LAN
3 USB
4
Jacks
5 Headset Jack
6 Power Jack
7
PHONE (RJ11)
T
ABLE 1: DEFINITIONS OF THE
10/100Mbps RJ-45 port for connecting PC as switch mode
10/100Mbps RJ-45 port for uplink to Ethernet.
USB Interface for external USB device like flash to store captured snap-
shot image; USB keyboard and mouse.
RCA Output Interface to external display like TV set
2.5 mm headset jack
12V DC power port
FXS port should be connected to analog phones. (Only applicable on
)
GXV-300X C
ONNECTORS
8
LINE (RJ-11)
FXO port should be connected to the PSTN line (Only applicable on
)
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W
Tab
Handset
Tab with exte
n-
Tab with
exte
n-
ALL MOUNT
GXV300x can be wall mounted. The two (2) wall mount sustaining brackets can be attached to the bottom
of the main body:
To position the phone on the wall, place two fixed hangers on the wall, hang the back of the phone on the
fixed hangers. The two sustaining brackets will balance the bottom of the phone and position the phone
on the wall.
To use the handset, pull out the tab (extension downward) from the handset cradle, rotate the tab and plug it
back into the slot with the extension up for to hold the handset.
Rest
sion down
sion up
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P
Desktop View
Side View
Back View
Closed
Model
Overview
P/N
001
-
One LINE(FXO/PSTN) port
007
010
Open Standards Compatible
record and SRV
),
DHCP
, PPPoE, TFTP, NTP.
Network Interfaces
Superb Audio Quality
Superb Video Quality
baseline
eo
bandwidth from
32kbps
– 1Mbps.
Advanced Video features
Feature Rich
ward,
DND
, mute, autodial
/
off-hook dial
.
Advanced Functionality
Ethernet Port
FXO telephone port (
LINE Port)
FXS telephone port (PHONE port)
LCD
Camera
Auxiliary Port
RODUCT OVERVIEW
T
ABLE 2:
GXV300X P
GXV3000
GXV3005
GXV3006
RODUCT MODELS
-Three lines
-Three lines
-Three lines
-One PHONE(FXS) port
xxx-xx
xxx-xx
xxx-xx
Note: Digits of P/N which are marked in red indentify the specific model.
For example: The device is a GXV3000 if the P/N number is 963-00001-42B
T
ABLE 3:
GXV300X K
EY FEATURES IN A GLANCE
SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A
Dual 10/100mbps Ethernet ports
Advanced Digital Signal Processing (DSP), Silence suppression, VAD, CNG,
AEC, AGC.
Supports H.263/H.263+ and H.264 (
5.6in TFT color LCD, 180 degrees vertical rotation, 300 degrees horizontal
rotation, advanced VGA resolution CMOS camera sensor, auto focus and
exposure, auto zoom, camera block, PIP, still picture capture/store.
Traditional voice features including caller ID, call waiting, hold, transfer, for-
-xxx
-xxx
-xxx
) real-time video codec with vid-
T
ABLE 4: HARDWARE SPECIFICATIONS
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Dual 10M/100M auto-sensing Ethernet ports function as a mini-switch
1 FXO PSTN line port (RJ11) (This is only applicable on GXV3005)
1 FXS PHONE port (RJ11) (This is only applicable on GXV3006)
5.6 inch TFT color LCD
Advanced CMOS sensor (VGA resolution)
RCA style stereo audio & composite video output jack,
one 2.5mm headset jack, two 2.0 USB ports
Exterior
Universal Power Supply
Dimension
Weight
Operating Temperature
Humidity
Compliance
Lines
Protocol Support
Display
Feature Keys
5 display/menu navigation keys
Net
work Interface
video to an external TV simultaneously), headset jack
Device Management
ment
, Keypad & adjustable 5.6 inch TFTP color LCD, Auto/manual provisioning system
,
Provisioning
Support automated NAT traversal without manual manipulation of firewall/NAT
Support remote device monitoring and events reporting using Syslog
Support device configuration via LCD, Web browser or central secure config
u
ration file
Video Features
Advanced VGA resolution CMOS camera sensor (view angle adjustable)
mute and camera block (for privacy), call log, video phone book
Black or Silver ABS plastic, 30 buttons (6 of which are LED buttons)
100-240V input, 12VDC/1.2A output, US/Euro/UK/Australian style plug availa-
ble
6.5cm x 18cm x 16cm
1.2kg
0º-40º C
10 - 95% non-condensing
FCC/CE/C-Tick
ECHNICAL SPECIFICATIONS
3 individual SIP accounts; 3 line buttons
Support SIP 2.0, UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP,
ICMP, DNS, DHCP, NTP/SNTP, TFTP.
5.6 inch TFT color LCD
8 dedicated keys: Message Button, Hold, Transfer, Conference, Speakerphone, Send,
Mute/Del, Camera Block
Web interface or via secure (AES encrypted) central configuration file for mass deploy GUI Interface, Address Book, Remote software upgrade (via TFTP/HTTP) for deployed
Dual (2) 10/100 auto-sensing Ethernet ports (switched)
Support 2 USB (2.0) host ports, 1 audio and 1 video output jack (capable of outputting
Support Layer 2 (IEEE 802.1p/Q tagging (VLAN) and Layer 3 QoS (DiffServ)
devices including behind firewall/NAT
Support remote automated and secure provisioning and software upgrade through fire-
wall/NAT to enable “zero configuration” and “plug-and-dial” for end users.
Support NAT traversal using IETF STUN and Symmetric RTP
Manual or dynamic host configuration protocol (DHCP) network setup
Powerful video DSP with advanced adaptive jitter control and packet loss concealment
technology to ensure superb audio and video quality
Support H.263/H.263+ (for legacy compatible) and H.264 (baseline) real-time video
codec (at CIF or QVGA resolution and up to 30 frames/second) to ensure highest quality video delivery at 32kbps -- 1Mbps bandwidth level (bit rate/frame rate configurable)
High quality 5.6-inch TFTP color LCD (allowing 2-dimensional view angle adjustment
including 180 degrees vertical rotation and nearly 300 degrees horizontal rotation)
Anti-flickering, auto focus and auto exposure, zoom, PIP (Picture-in-Picture), audio
Configurable screen-saver pictures, still picture capture/store (VGA resolution), visual
voice message indicator
Intuitive graphic user interface enabled by 5 navigation buttons
Support for G.723.1, G.729A/B, G.711 µ/A (PCMU/PCMA), G.726 with dynamic negotiation of codec and packet time
Silence Suppression, VAD (voice activity detection), CNG (comfort noise generation),
AGC (automatic gain control), Packet delay & loss concealment
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for speakerphone
Call hold, call transfer (attended/blind), Do-Not-Disturb(DND), call forward (no answer,
Grandstream Networks, Inc. GXV300x User ManualPage 7 of 54
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Caller ID (Privacy)
Firmware Upgrades
Advanced Server
Features
Security
DIGES
T authentication and encryption using MD5 and MD5
-
sess
caller ID, 3-way conferencing, redial, dial plans, off-hook auto dial, auto answer, early
dial, speed dial , call log, and volume control, Visual voice mail indicator, audio mute &
camera block (for privacy), downloadable custom ring-tones
Support for anonymous call using privacy header
Support for Authenticating configuration file before accepting changes;
Support firmware upgrade via TFTP or HTTP, Allow user to specify different URL for
configuration file and firmware files
Support DNS SRV Look up and SIP Server Fail Over, Visual message waiting indication, Support customizable idle screensaver
Secure signaling (SIP over TLS, pending) and secure voice/video communication
(SRTP).
Grandstream Networks, Inc. GXV300x User ManualPage 8 of 54
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U
Icon LCD Icon Definitions
Connectivity Status / SIP Proxy/Server Icon:
Phone Status Icon:
Audio Mute:
DND:
Speaker Phone Status Icon:
Handset, Speakerphone and Ring Volume Icon:
SING
GXV300X IP V
IDEO PHONE
G
ETTING FAMILIAR WITH
F
IGURE 2:
GXV LCD S
The LCD display disappears when the screensaver is on. If you do not want the LCD display, you can
turn it off.
CREEN
LCD
T
ABLE 6:
LCD I
CON DEFINITIONS
Color – connected to SIP Server/IP address received
Gray – SIP Proxy/Server not registered
OFF when the handset is on-hook
ON when the handset is off-hook
ON when audio mute is enabled
ON when DND is enabled
FLASH when phone rings or a call is pending
OFF when the speakerphone is off
ON when the speakerphone is on
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Each icon appears next to the volume icon
To adjust volume, use the up/down arrow button
Real-time Clock:
Missed Call Icon:
Video Mute:
Voice/
Video Message:
Forward:
F
IGURE 3: GETTING FAMILIAR WITH KEYPAD
Synchronized to Internet time server (Internet required)
Time zone configurable via web browser
AM/PM indicator
Icon appears when you miss a call
Number indicates how many missed calls
ON when video is disabled
ON when there is unheard message
ON when call forwarding is enabled
Line 1-3 keys
Mute/Delete
Camera Block
Message
Standard keypad
Menu Keys
Conference
Transfer
Phonebook
OSD/Camera
Loopback Display
On/Off key
Hold
Speaker Send/Re-Dial
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Table 7: GXV Button Definitions
Key Button
Key Button Definitions
LINE1
-
LINE
3
3 Line keys with LED
, can be configured to different SIP profile
s or providers
increase
ring vo
l
ume when phone is in IDLE mode
Or decrease ring volume when is in IDLE mode
LEFT
shortcut
Call History
TRNF
TRANSFER key:
Transfer
an ACTIVE
call to another number
CONF
M
ESSAGE
tions
as a no
r
mal IP phone
user has a voicemail
)
A
DDRESS BOOK
HOLD
Place
ACTIVE call
on hold
S
END
ReD
ial
UP ↑
DOWN ↓
RIGHT
OK
CAMERA BLOCK
MUTE/DEL
Use button to select next “Menu Item” when phone is in keypad configuration
mode; Or increase handset/speakerphone volume when phone is ACTIVE; Or
Use button to select previous “Menu Item” when phone is in keypad configure
mode; Or decrease handset/speakerphone volume when phone is ACTIVE;
Shift cursor to left
Shift cursor to right. Also function as
phone is in IDLE state.
Enter Keypad Configuration “MENU” mode when phone is in IDLE mode. Use
as ENTER key when in Keypad Configuration.
Press CONF button to connect Calling/Called party into conference
Enter to retrieve (video) voice mails or other messages
Enable/Disable Camera for privacy. Block camera to make call without video.
When button is activated, a red icon will flash on the LCD and the phone func-
Enable/Disable Mute.
On an active call, press button MUTE the call;
When using in keypad configuration, function as DELETE key;
Activate/De-activate do-not-disturb (DND) when idle;
Reject an incoming call during ring stage (send caller directly to voicemail if
to bring up
when
CAMERA LOOPBACK DIS-
PLAY and OSD ON/OFF
SPEAKER
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Enable/Disable camera local loopback display; On Screen Display (OSD),
function like loop switch when pressed.
XML based Video Phone Address Book
Enable/Disable hands-free speaker phone mode
Press SEND to dial a new number or
the last number dialed.
M
ULTIPLE
The GXV300x can support up to 3 independent SIP accounts. Each account is capable of independent
SIP server, user and NAT settings. Each of the 3 line buttons (LINE1-LINE3) is “virtually” mapped to an
individual SIP account. In off-hook state, select an idle line and the name of the account (as configured in
the web interface) is displayed on the LCD and a dial tone is heard. Or, press the selected line continuously until the related SIP profile showed up on the LCD.
For example: Configure the three SIP accounts as “FreeWorldDialup”, “BroadSoft” and “Asterisk PBX”
respectively and ensure each is active and registered. When LINE1 is pressed, you will hear a dial tone
and see “FreeWorldDialup” on the LCD display; when LINE2 is pressed, you will hear a dial tone and see
“BroadSoft” on the LCD display; when LINE3 is pressed, you will hear a dial tone and see “Asterisk PBX”
on the LCD display. Or press LINE1 twice you will hear dial tone and see “BroadSoft” also.
To make a call, select the line you wish to use. The LED will light up solid red. If you continue to press
one LINE, it will circulate among the registered accounts.
For example: when LINE1 is pressed, the LCD displays “FreeWorldDialup”; If LINE1 is pressed twice, the
LCD displays “BroadSoft” and the subsequent call will be made through SIP account 2 - BroadSoft.
Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When
the “virtually” mapped line is in use, the GXV300x will flash the next available LINE (from left to right) in
red. A line is ACTIVE when it is in use and the corresponding LED is solid red.
SIP A
CCOUNTS AND LINES
Handset, Speakerphone and Headset Mode
Handset can be toggled between Speaker/Headset. To switch between Handset and Speaker/Headset,
press the Hook Flash in the handset cradle or press the SPEAKER button.
Completing Calls
There are four ways to complete a call:
1. D
IAL:
To make a phone call
•Take Handset/SPEAKER/Headset off-hook
Or press an available LINE key (activates speakerphone).
•The line will have a dial tone and the primary line (LINE1) LED is red.
If you wish, select another LINE key (alternative SIP account).
• Enter the phone number.
• press the SEND key.
2. R
EDIAL
: To redial the last dialed phone number.
When redialing the phone will use the same SIP account as was used for the last call. Thus, when
the third SIP account was made for the last call/call attempt, the phone will use the third account to
redial.
•Take Handset/SPEAKER/Headset off-hook or
press an available LINE key (activates speakerphone), the corresponding LED will be red.
•Press the SEND button
3. U
SE THE MENU:
phone number. Press OK to select and press OK again to dial. The call will dial out in speakermode with related SIP profile.
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Press the OK button to bring up the Main Menu. Select Phone Book and select
4. U
5. U
6. P
NOTE:
SING THE CALL HISTORY:
To call a phone number in the phone’s history
When using the call history, the phone will use the same SIP account as was used for the last
call/call attempt. Thus, when returning a call made to the third SIP account, the phone will use the
third SIP account return the call.
• Press the MENU button to bring up the Main Menu.
• Select Call History and then “Received Calls”, “Missed Calls” or “Dialed Calls” depending
on your needs
• Select phone number using the arrow keys
• Press OK to select
• Press OK again to dial.
SING THE PHONEBOOK
: Calling a phone from the phone’s phonebook.
Each entry in the phonebook can be attached to an individual SIP account. The phone will use
that SIP account to make the phone call.
•Go to the phonebook by:
i. Pressing the phonebook button (bottom, left-hand side of phone), or
ii. Pressing the DOWN arrow key, or
iii. Pressing the menu button and
Selecting “Phone book” and
Press MENU
• Select the phone number by using the arrow keys
• Press OK to select
• Press OK again to dial.
AGING/INTERCOM
:
The paging/intercom function can only be used if the SERVER/PBX supports this feature and both
the phones and PBX are correctly configured.
• Take the Handset/SPEAKER/Headset off-hook,
• Select the LINE key associated with account
• Press OK key to display LCD: LINEx: PAGE USING.
• Dial the phone number you want to Page/Intercom
• Press SEND key.
• Dial-tone and dialed number display occurs after the handset is off-hook and the line key is se-
lected.
•The phone waits 4 seconds (by default; no key entry timeout) before sending and initiating the
call. Press the “SEND” button (or # key if configured) to override the 4 second delay.
Making Calls using IP Addresses
Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy. VoIP
calls can be made between two phones if:
• Both phones have public IP addresses, or
• Both phones are on a same LAN/VPN using private or public IP addresses, or
Grandstream Networks, Inc. GXV300x User ManualPage 13 of 54
Firmware 1.2.3.7 Updated: 12/2010
•Both phones can be connected through a router using public or private IP addresses (with neces-
sary port forwarding or DMZ)
To make a direct IP call, please follow these steps:
1. Press MENU button to bring up MAIN MENU.
2. Select “Direct IP Call” using the arrow-keys.
3. Press OK to select.
4. Input the 12-digits target IP address. (Please see example below).
5. Press OK key to initiate call.
For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input
the following: 192*168*1*60#5062 - “ * ” key represent “.” and “#” key represent “:”. Press OK to dial out.
Quick IP Call Mode
Dial an IP address under the same LAN/VPN by keying in the last octet in the IP address. This partially
simulates a PBX function using the CMSA/CD without a SIP server. Static IP usage recommended.
In the “Advanced Settings” page, set the "Use Quick IP-call mode” to “YES”. When #xxx# is dialed,
where x is 0-9 and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. “aaa.bbb.ccc” is from the
local IP address regardless of subnet mask.
#xx or #x are also valid. The leading 0 is not required (but OK).
For example:
• 192.168.0.2 calling 192.168.0.3 -- dial #3 follow by SEND or #
• 192.168.0.2 calling 192.168.0.23 -- dial #23 follow by SEND or #
• 192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or #
• 192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3
NOTE: If you have a SIP Server configured (not recommended for IP calling), a Direct IP-IP still works. If
you are using STUN, the Direct IP-IP call will also use STUN. Configure the “Use Random Port” to “NO”
when completing Direct IP calls.
Grandstream Networks, Inc. GXV300x User ManualPage 14 of 54
Firmware 1.2.3.7 Updated: 12/2010
M
AKING PHONE CALLS
Receiving Calls
1. Incoming single call: Phone rings with selected ring-tone. The corresponding account LINE
flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER.
2. Incoming multiple calls: Call Waiting tone audible. Next available lines will flash red .
3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call via
SPEAKER. (PBX or Server must support this feature first)
Call Waiting/ Call Hold
1. Hold: Place a call on ‘hold’ by pressing the “HOLD” button.
2. Resume: Resume call by pressing the corresponding blinking LINE key.
3. Multiple Calls: Automatically place ACTIVE call on ‘HOLD’ by selecting another available LINE
key to place or receive another call. Call Waiting tone audible when line is in use and there is a
call waiting to be answered.
Call Transfer
GXV300x supports both
1. Blind Transfer: Press “TRNF” button, then dial the number and press the “SEND” button (or # if
configured) to complete transfer of an active call.
2. Attended (or Supervised) Transfer: Press “Line x” button to make a call and automatically
place the ACTIVE LINE on HOLD. Once the call is established, press “TRNF” key then the key
with blinking line to transfer the call and hang up.
NOTE:
•To transfer calls across SIP domains, SIP service providers must support transfer across SIP
domains. Blind transfer will usually use the primary account SIP profile.
•If pressed wrong number and want to cancel the transfer, just press the “TRNF” again to cancel
the transfer. Or press HOLD key to hold the line and redo again.
blind
and
attended
(or supervised) transfer:
3-Way Conferencing
GXV-3000 supports 3-way conferencing.
1. Initiate a Conference Call: Place first call on hold by press LINE key to call second party and
establish the call, press the CONF button then press the line on hold (blinking), bring three parties
together.
2. Toggle Video Conference: Once the 3-way conference is established, press the CONF key to
toggle between two video users for video.
Grandstream Networks, Inc. GXV300x User ManualPage 15 of 54
Firmware 1.2.3.7 Updated: 12/2010
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