Grandstream Networks, Inc.
Analog IP Gateway GXW-410xv
4 or 8 FXO Ports w/Video Surveillance
GXW-410xv User’s Manual
Firmware Version 1.0.0.36
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TABLE OF FIGURES
GXW-410XV USER MANUAL
FIGURE 1: DIAGRAM OF GXW-410XV BACK PANEL ..........................................................................................5
FIGURE 2: DIAGRAM OF GXW-410XV DISPLAY PANEL......................................................................................6
FIGURE 3: FUNCTIONAL DIAGRAM OF IP-PBX & GXW-410XV ..........................................................................7
FIGURE 4: SCREEN-SHOT OF GXW-410XV LOG-IN SCREEN............................................................................12
FIGURE 5: SCREENSHOT OF ADVANCED USER CONFIGURATION......................................................................13
FIGURE 6: SCREEN-SHOT OF GXW-410XV CHANNEL DIALING ........................................................................16
FIGURE 7: SCREEN-SHOT OF SAVE CONFIGURATION ......................................................................................20
FIGURE 8: SCREEN-SHOT OF REBOOTING ......................................................................................................20
FIGURE 9: SCREEN-SHOT OF VIDEO SURVEILLANCE*......................................................................................22
APPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER........................................................25
APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIO ............................................................25
APPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATION.................................26
TABLE OF TABLES
GXW-410XV USER MANUAL
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS.........................................................................................5
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL ......................................................................................6
TABLE 3: GXW-410XV SOFTWARE FEATURES ..............................................................................................10
TABLE 4: HARDWARE SPECIFICATION OF GXW-410XV ...................................................................................11
TABLE 6: WEB LOG-IN DEFINITIONS...............................................................................................................12
TABLE 7: STATUS PAGE DEFINITIONS ............................................................................................................13
TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONS ..............................................................................14
TABLE 9: FXO LINES CONFIGURATION DEFINITIONS .......................................................................................16
TABLE 10: CHANNELS PAGE DEFINITIONS ......................................................................................................17
TABLE 11: GRANDSTREAM CONFIGURATION PAGE DEFINITIONS......................................................................18
GUI INTERFACES
GXW-410XV USER MANUAL
1. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
3. SCREENSHOT OF CHANNELS CONFIGURATION PAGE
4. SCREENSHOT OF FXO LINES CONFIGURATION PAGE
5. SCREENSHOT OF PROFILE 1 CONFIGURATION PAGE
6. SCREENSHOT OF STATUS CONFIGURATION PAGE
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WELCOME
Thank you for purchasing the Grandstream GXW–410xv IP Analog FXO Gateway. The GXW–410xv is a
cost effective, easy to use and easy to configure IP communications solution for any business. The
GXW–410xv supports popular voice codecs and is designed for full SIP compatibility and interoperability
with 3rd party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology, integrate
a traditional phone system into a VoIP network, and efficiently manage communication costs.
This manual will help you learn how to operate and manage your GXW FXOAnalog IP Gateway and
make the best use of its many upgraded features including simple and quick installation, multi-party
conferencing, and direct IP-IP Calling. This IP Analog Gateway is very easy to manage and scalable,
specifically designed to be an easy to use and affordable VoIP solution for the small – medium business
or enterprise. Enable the video surveillance port to give piece of mind while you are away from your
business.
Gateway GXW-410xv Overview
The GXW410x offers an easy to manage, feature rich IP communications solution for any small business
or businesses with virtual and/or branch locations who want to leverage their broadband network and/or
add new IP Technology to their current phone system. The Grandstream Enterprise Analog VoIP
Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls and vice versa. There are
two models - the GXW-4104v and GXW-4108v, which have either 4 and 8 FXO ports respectively. The
installation is the same for either model.
A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-410xv
series. In this environment, the SIP server handles SIP registration and call control and the GXW-410xv
processes media conversion between IP and PSTN calls. By design, the system supports the call
progress tones and PSTN signaling standards for North America, Europe, Latin America, Asia, and
various other countries/regions.
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.
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•
•
This document is contains links to Grandstream GUI Ingerfaces. Please remember to download
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 the entire or any part, in any form or by any means, electronic or print,
for any purpose without the express written permission of Grandstream Networks, Inc. is not
permitted.
PACKAGING
Unpack and check all accessories. Equipment included in the package:
1) One GXW-410xv Unit
2) One universal power adaptor
3) One Ethernet cable
SAFETY COMPLIANCES
The GXW-410xv is compliant with various safety standards including FCC/CE. Its power adaptor is
compliant with UL standard. Warning: use only the power adapter included in the GXW-410xv
package. Using an alternative power adapter may permanently damage the unit.
WARRANTY
Grandstream has a reseller agreement with our reseller customer. End users should contact the company
from whom you purchased the product for replacement, repair or refund.
If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service
Representative for a RMA (Return Materials Authorization) number. Grandstream reserves the right to
remedy warranty policy without prior notification.
CONNECTING THE GXW-410XV
FIGURE 1: DIAGRAM OF GXW-410XV BACK PANEL
GXW-410xv
Power Supply
FXO Ports
LAN/WAN RJ-45
Ethernet Ports
On/Off Switch
VIDEO IN Jack
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS
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LAN (or PC)
Connect your PC to the LAN to find IP address from your Router/DHCP
Server. The GXW-410xv acts as a switch.
WAN (or LAN) Connect to the internal LAN network or router.
VIDEO IN
RESET
Connection for Analog based Video Surveillance Camera (RCA)
Factory Reset button. Press for 7 seconds to reset factory default settings.
POWER IN
OFF/ON
Power adapter connection
Off/On switch
FXO1 - FXO8
FXO ports to be connected to physical PSTN lines from a traditional PSTN
PBX or PSTN Central Office.
FIGURE 2: DIAGRAM OF GXW-410XV DISPLAY PANEL
GXW- 410xv
Display
FXO port
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL
Power LED
Indicates Power. Remains ON when Power is connected and unit is
turned ON.
Ready LED
LAN LED
PC LED
Remains ON after boot-up.
Indicates LAN (or WAN) port activity
Indicates PC (or LAN) port activity
Video LED
Remains solid green on boot-up. If Video IN terminal is connected,
indicates video activity.
LEDs 1 - 8
Indicate status of the respective FXO Ports on the back panel
Busy - ON (Solid Green)
Available - OFF
NOTE: All LEDs display green when ON.
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Application Description
A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-410xv
series. In this environment, the SIP server handles SIP registration and call control and the GXW-410xv
processes media conversion between IP and PSTN calls. By design, the system supports the call
progress tones and PSTN signaling standards for North America, Europe, Latin America, Asia, and
various other countries/regions.
FIGURE 3: FUNCTIONAL DIAGRAM OF IP-PBX & GXW-410XV
Anywhere in the world
IPPBX or
SIP Server
FXO Lines
4 or 8 Ports
GXW-410xv
PSTN
Cloud
IP/LAN
IP/WAN
Grandstream IP Phones
PSTN Analog
SIMPLE CONFIGURATION: MEDIA GATEWAY TO ACCESS PSTN NETWORKS
GXW-410xv can be configured to work with any leading SIP server, for a pure media gateway to access
PSTN networks. In such applications, the user only needs to configure GXW-410xvgateway Stage Dialing
field and Sip Server field.
For a simple set-up, users only need to configure a SIP server field for default SIP Profile 1. This field
should be configured to point to the SIP server to be used with the GXW-410xv.
For advanced applications, the user is required to choose at least one SIP server field from the SIP
profiles and one stage dialing under system Channel configuration table. On SIP server sides, the SIP
server must be configured to forward user PSTN calls to the GXW-410xv.
Please be aware that by default, the system uses North American PSTN settings and TWO STAGE
dialing to access PSTN networks for VOIP to PSTN calls and PSTN to VOIP calls. Two stage dialing
means the end-user will hear dial-tone twice. First dial-tone is used to let users to input destination
number in the same network of the calling networks. Second dial-tone is used to let users to input final
destination number.
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The GXW-410xv also supports ONE STAGE dialing, which means users only need to input one final
PSTN number when first dial-tone is heard for calls from VOIP to PSTN. This requires configuring both
SIP server and GXW-410xv to one stage dialing (see last section of quick guide and user manual for one
stage dialing). For one stage PSTN to VOIP calls, user needs to configure off-hook auto dial field (see
last section on sample configuration
EXTENSIVE CONFIGURATION: MEDIA GATEWAY CONFIGURATION FOR MULTIPLE USERS
The GXW-410xv can be configured to work with a variety of SIP server features and traditional PBX on
PSTN networks, with a different SIP server on each physical port. Each port may have its own voice
setting, dialing settings, PSTN termination setting, and DTMF transmission settings.
OFF-HOOK AUTO DIAL
The FXO interface currently does not support direct inward dialing (DID). The GXW-410xv implements an
off-hook auto dial feature for each physical port. Configure off-hook auto dial to forward PSTN incoming
call to a specific SIP number, call center or hunt group.
SAMPLE CONFIGURATIONS - IP PBX PEERS WITH GXW-410XV
There are 2 methods to configure GXW to work with IP PBX:
1) Configure GXW with SIP Accounts in IP PBX, this will enable you to put GXW behind a
NAT/Firewall.
2) Configure GXW without SIP Accounts in IP PBX, this makes GXW function as a PEER gateway.
more information for quick installation.
Note: In regions other than North American, the user is also required to configure call progress tones
and PSTN line termination fields. Check with local PSTN service carriers on values service providers use
on the lines. If service provider doesn’t provide these values and users don’t know what the correct
values are, please use the default values. Contact product support for additional help in configuring your
GXW-410xv.
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Features
GXW-410xv is a next generation IP voice and video gateway that features full interoperability with
leading IP-PBXs, SoftSwitches and SIP platforms. The Gateway series offers superb voice and video
quality, traditional telephony functionality, simple configuration, feature rich functionality and an additional
video port that enables the gateway to act like a video surveillance gateway.
SOFTWARE FEATURES OVERVIEW
•
•
•
•
•
•
•
•
•
•
•
•
4 and 8 FXO port media gateways
Video surveillance port
External power supply
Two RJ-45 ports (switched or routed)
TFTP and HTTP firmware upgrade support
Multiple SIP accounts, multiple SIP profiles (3 profiles per system)
Supports Audio Codecs: G711, G723, G729 and GSM
Supports Video Codecs: H.264
G.168 – echo cancellation
Flexible DTMF transmission: In Audio, RFC2833, SIP Info or any combination of the 3
Selectable, multiple LBR coders per channel
T.38 compliant
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HARDWARE SPECIFICATION
TABLE 4: HARDWARE SPECIFICATION OF GXW-410XV
LAN interface
LED
2xRJ45 10/100Mbps
8 LEDs (GREEN)
Universal Switching
Power Adaptor
Input: 100-240V AC, 50/60Hz, 0.5A Max
Output: 12V DC, 1.25A
UL certified
Dimension
Weight
225mm (L) x 172mm (W) x 42mm (H)
0.29 lbs (3.5 oz)
Temperature
32~104°F
0~40°C
Humidity
10% - 90% (non-condensing)
FCC, CE
Compliance
CONFIGURATION GUIDE
CONFIGURATION WITH WEB BROWSER
The GXW-410xv has an embedded Web server that will respond to HTTP GET/POST requests. It also
has embedded HTML pages that allow a user to configure the IP phone through any common web
browser. Examples of GUI interfaces can be downloaded @
ACCESSING THE WEB CONFIGURATION MENU
1. Connect the Power to the GXW-410xv unit.
2. Connect an Ethernet cable between the LAN port on GXW-410xv to your PC.
3. You will have to assign a dummy IP with the same subnet as the GXW IP Address, which is
192.168.0.160 by default. So, set an IP address like 192.168.0.x for your PC.
to the GXW-410xv web server.
You may choose to use DHCP or PPPoE connection or another static IP address according to your local
network environment.
The Gateway Web Configuration Menu can be then accessed by the following URI: http://Gateway-IP-
NOTE: To access the configuration page, type the GXW IP address into the browser, stripping out the
leading “0” because the browser will parse in octet. e.g. if the IP address is: 192.168.001.014, please
type in: 192.168.1.14.
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END USER CONFIGURATION
Once this HTTP request is entered and sent from a Web browser, the GXW-410xv will respond with a
login screen. There are two default passwords for the login page:
User Level:
Password: Webpages allowed:
End User Level
Administrator Level
123
Only Status and Basic Settings
All pages can be browsed.
admin
FIGURE 4: SCREEN-SHOT OF GXW-410XV LOG-IN SCREEN
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2005-2006
After login, the next configuration page is the Basic Configuration page, explained in detail in Table 6:
Web Log-in Definition.
TABLE 6: WEB LOG-IN DEFINITIONS
Web Access
Web Port
Select HTTP or secure HTTPS protocol for Web Access
By default, HTTP uses port 80 and HTTPS uses port 443. This field is
for customizable web port.
End User Password This contains the password to access the Web Configuration Menu.
This field is case sensitive with a maximum length of 25 characters.
IP Address
There are two modes to operate the GXW-410xv:
DHCP mode: all the field values for the Static IP mode are not used
(even though they are still saved in the Flash memory.) The GXW-
410xv acquires its IP address from the first DHCP server it discovers
from the LAN it is connected.
Using the PPPoE feature: set the PPPoE account settings. The GXW-
410xv will establish a PPPoE session if any of the PPPoE fields is set.
Static IP mode: configure the IP address, Subnet Mask, Default
Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary)
fields. These fields are set to zero by default.
Time Zone
Controls how the date/time is displayed according to the specified time
zone.
You may also access the Device Status page which provides details of the GXW product. The Device
Status page terms are defined in Table 7: Status Page Definitions.
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TABLE 7: STATUS PAGE DEFINITIONS
Hardware Revision
MAC Address
Hardware version number: Main Board, Interface Board
The device ID in HEX format. This is a very important ID for ISP
troubleshooting.
IP Address
This field shows LAN IP address of GXW-410xv
This field contains the product model info.
Product Model
Software Version
Program: This is the main software release. Boot and Loader are not changed
often.
System Up Time
Registered
This field shows system up time since the last reboot.
This field indicates whether the different Channels are registered to the SIP
server(s).
FXO Line Connected
This field will give the status of each physical FXO Line connected to the
Gateway. It will update the status regularly.
Yes - Connected and Idle
Busy - Connected and Busy
No - Not connected
PPPoE Link Up
This field shows whether the PPPoE connection is running if connected to DSL
modem.
Detected NAT Type
This field shows what kind NAT the GXW-410xv is connected to via its LAN
port. It is based on STUN protocol.
ADVANCED USER SETTINGS
ADVANCED USER CONFIGURATION
The end-user needs to login to the advanced user configuration page the same way as for the basic
configuration page.
FIGURE 5: SCREENSHOT OF ADVANCED USER CONFIGURATION
Grandstream Device Configuration
Password
Login
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Advanced User configuration includes the end user configuration and advanced configurations including:
SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.
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TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONS
Admin
Administrator password. Only the administrator can configure the “Advanced Settings”
Password
page. Password field is purposely left blank for security reasons. The maximum
password length is 25 characters.
g723 Rate
G723 encoding rate (6.3kbps or 5.3kbps)
Layer 3 QoS
This field defines the layer 3 QoS parameter which can be the value used for IP
Precedence or Diff-Serv or MPLS. Default value is 48.
Layer 2 QoS
This contains the value used for layer 2 VLAN tag. Default setting is
blank.
Inter Digit
Timeout
Default is 4 seconds.
Local RTP port This parameter defines the local RTP-RTCP port pair the GXW-410xv will listen and
transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this
port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2
for RTP and port_value+3 for its RTCP. The default value is 5004.
Use Random
Port
This parameter, when set to Yes, will force random generation of both the local SIP and
RTP ports. This is usually necessary when multiple GXW-410xvs are behind the same
NAT.
Keep-alive
interval
This parameter specifies how often the GXW-410xv sends a blank UDP packet to the
SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
IP address or Domain name of the STUN server.
STUN Server
Firmware
Upgrade &
Provisioning
This radio button will enable GXW-410xv to download firmware or configuration file
through either TFTP or HTTP.
Via TFTP
Server
This is the IP address of the configured TFTP server. If selected and it is non-zero or
not blank, the GXW410x will attempt to retrieve new configuration file or new code
image from the specified TFTP server at boot time. It will make up to 5 attempts before
timeout and then it will start the boot process using the existing code image in the Flash
memory. If a TFTP server is configured and a new code image is retrieved, the new
downloaded image will be verified and then saved into the Flash memory.
Note: Please do NOT interrupt the TFTP upgrade process (especially the power supply)
as this will damage the device. Depending on the network environment this process
can take up to 15 or 20 minutes.
Via HTTP
Server
The URL for the HTTP server used for firmware upgrade and configuration via HTTP.
For example, ttp://provisioning.mycompany.com:6688/Grandstream/1.0.0.36
Here “:6688” is the specific TCP port that the HTTP server is listening at, it can be
omitted if using default port 80.
Note: If Auto Upgrade is set to No, GXW-410xv will only do HTTP download once at
boot up.
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Automatic
Upgrade
Choose Yes to enable automatic upgrade and provisioning. In “Check for new firmware
every” field, enter the number of days to enable GXW-410xv to check the server for
firmware upgrade or configuration in the defined period of days. When set to No, GXW-
410xv will only do upgrade once at boot up.
“Always check for New Firmware.” Check New Firmware only when F/W pre/suffix
changes”
Syslog Server The IP address or URL of System log server. This feature is especially useful for ITSP
(Internet Telephone Service Provider)
Syslog Level
Select the ATA to report the log level. Default is NONE. The level is one of DEBUG,
INFO, WARNING or ERROR. Syslog messages are sent based on the following
events:
1. product model/version on boot up (INFO level)
2. NAT related info (INFO level)
3. sent or received SIP message (DEBUG level)
4. SIP message summary (INFO level)
5. inbound and outbound calls (INFO level)
6. registration status change (INFO level)
7. negotiated codec (INFO level)
8. Ethernet link up (INFO level)
9. SLIC chip exception (WARNING and ERROR levels)
10. memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains the
following components:
GS_LOG: [device MAC address][error code] error message
Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000]
Ethernet link is up
NTP server
URI or IP address of the NTP (Network Time Protocol) server, which will be used by the
phone to synchronize the date and time.
Enable Video
Surveillance
Work in progress.
RTSP Port
By default it is 554.
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Configuring the FXO channels on the GXW – 410xv is an easy process. Follow the GUI interfaces. The
Device Status page terms are defined in Table 9: FXO Lines Configuration Definitions. An example
of the Channel Dialing Configuration is shown in Figure 6. Please note the default is always configured.
The user has the option to change the default settings as described in the Table 9.
TABLE 9: FXO LINES CONFIGURATION DEFINITIONS
Enable Current
Disconnect
When set to Y, Current Disconnect is enabled. Certain PSTN Cos require this to be
enabled, in order to realize correct disconnect for PSTN side. Default it Y.
AC Termination
Impedance
Selects the impedance of the analog Line connected to the FXO port on the GXW-
410xv.
Silence Timeout Terminate call after long silence detected. Default is 60 seconds, max 65536.
DTMF Digit
Length
Default value is 100ms.
DTMF Digit
Volume
Default value is -11dB.
DTMF Dial Pause Default value is 100ms.
Wait Dial-tone
Dialing Stage
It is recommended to set this to Y in case Dialing stage is set to 1.
Dialing stage can be set to 1 or 2. Note: When set to 1, the Server needs to be
configured to allow forwarding and receipt of SIP messages from GXW IP address
directly.
Off-hook Auto
Dial
This parameter allows users to configure a User ID or extension number to be
automatically dialed upon off-hook. Please note only the user part of the SIP address
needs to be entered here. The GXW-410xv will automatically append the ‘@’ and host
portion of the corresponding SIP address.
FIGURE 6: SCREEN-SHOT OF GXW-410XV CHANNEL DIALING
Channel Dialing
1. DTMF Digit
Length(X10ms):
ch1-8:10;
(1-200, default 10)
(-31-0, default -11)
(1-200, default 10)
(default Yes)
ch1-8:-11;
ch1-8:10;
ch1-8:Y;
ch1-8:2;
2. DTMF Digit Volume(dB):
3. DTMF Dial Pause(X10ms):
4. Wait Dial-Tone(Y/N):
5. Dialing Stage(1/2):
(default 2)
ch1-8:605;
6. Off-hook Auto Dial(VoIP):
Update
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TABLE 10: CHANNELS PAGE DEFINITIONS
SIP User ID
User account information, provided by VoIP service provider (ITSP).
Usually in the form of digit similar to phone number or actually a phone
number.
Authentication ID
SIP service subscriber’s Authenticate ID used for authentication. Can be
identical to or different from SIP User ID.
Authentication Password SIP service subscriber’s account password for GXW-410xv to register to
(SIP) servers of ITSP.
Profile ID
Select the corresponding Profile ID (1/2/3)
Call Progress Tones
Using these settings, user can configure tone frequencies according to user
preference. By default, the tones are set to North American frequencies.
Frequencies should be configured with known values to avoid
uncomfortable high pitch sounds. 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.
•
•
•
•
“Dial tone”
“Ringback tone”
“Busy/Re order tone”
“Confirmation tone”
Channel Voice Settings
Channel voice settings mentioned below.
Tx to PSTN Audio Gain
(dB)
Allows user to set a value in dB for transmission to PSTN Audio Gain.
Rx from PSTN Audio
Gain (dB)
Allows user to set a value in dB for receive from PSTN Audio Gain.
Silence Suppression
This controls the silence suppression/VAD feature of G723 and G729. If set
to “Yes”, when a silence is detected, small quantity of VAD packets (instead
of audio packets) will be sent during the period of no talking. If set to “No”,
this feature is disabled.
Echo Cancellation
When set to Y, Echo cancellation is enabled.
Channel specific Setting Channel specific settings mentioned below.
DTMF Method
This parameter specifies the mechanism to transmit DTMF digits. There
are7 modes supported: in audio which means DTMF is combined in audio
signal (not very reliable with low bit-rate codec), via RTP (RFC2833), or via
SIP INFO. Multiple DTMF transmission schemas can be selected.
1 – in-audio
2 – RFC2833
3 – in-audio and RFC2833
4 – SIP Info
5 – in-audio and RFC2833
6 – SIP Info and RFC2833
7 – in-audio, RFC2833, and SIP Info
No Key Entry Timeout
Default is 4 seconds.
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TABLE 11: GRANDSTREAM CONFIGURATION PAGE DEFINITIONS
Activate Profile
Profile Name
SIP Server
When set to Yes the SIP Profile is activated.
A name to identify a Profile.
SIP Server’s IP address or Domain name provided by VoIP service provider.
Outbound Proxy IP address or Domain name of Outbound Proxy, or Media Gateway, or Session
Border Controller. Used by GXW-410xv for firewall or NAT penetration in different
network environments. If symmetric NAT is detected, STUN will not work and
ONLY outbound proxy can correct the problem.
Use DNS SRV:
Default is No. If set to Yes the client will use DNS SRV to look up server.
User ID is Phone If the GXW-410xv has an assigned PSTN telephone number, this field should be
Number
set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will
be attached to the “From” header in SIP request.
SIP Registration This parameter controls whether the GXW-410xv needs to send REGISTER
messages to the proxy server. The default setting is “Yes”.
Unregister on
Reboot
Default is No. If set to yes, the SIP user’s registration information will be cleared on
reboot.
Register
Expiration
This parameter allows the user to specify the time frequency (in minutes) for the
GXW-410xv to refresh its registration with the specified registrar. The default
interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about
45 days).
Local SIP port
NAT Traversal
This parameter defines the local SIP port the GXW-410xv will listen and transmit.
The default value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2,
Account 3 and Account 4 respectively.
This parameter defines whether the GXW-410xv NAT traversal mechanism will be
activated or not. If activated (by choosing “Yes”) and a STUN server is also
specified, then the GXW-410xv will behave according to the STUN client
specification. Under this mode, the embedded STUN client inside the GXW-410xv
will attempt to detect if and what type of firewall/NAT it is sitting behind through
communication with the specified STUN server. If the detected NAT is a Full Cone,
Restricted Cone, or a Port-Restricted Cone, the GXW-410xv will attempt to use its
mapped public IP address and port in all of its SIP and SDP messages. If the NAT
Traversal field is set to “Yes” with no specified STUN server, the GXW-410xv will
periodically (every 20 seconds or so) send a blank UDP packet (with no payload
data) to the SIP server to keep the “hole” on the NAT open.
Proxy-Require
Early Dial
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Default is No. Use only if proxy supports 484 response.
Session
Expiration
Grandstream implemented SIP Session Timer. The session timer extension
enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or
re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or
re-INVITE message, the session will be terminated. 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.
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Min-SE
The minimum session expiration (in seconds). The default value is 90 seconds.
Caller Request
Timer
If selecting “Yes” the phone will use session timer when it makes outbound calls if
remote party supports session timer.
Callee Request
Timer
If selecting “Yes” the phone will use session timer when it receives inbound calls
with session timer request.
Force Timer
If selecting “Yes” the phone will use session timer even if the remote party does not
support this feature. Selecting “No” will allow the phone to enable session timer only
when the remote party support this feature. To turn off Session Timer, select “No”
for Caller Request Timer, Callee Request Timer, and Force Timer.
UAC Specify
Refresher
As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee
or proxy server as the refresher.
UAS Specify
Refresher
As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.
Force INVITE
Session Timer can be refreshed using INVITE method or UPDATE method. Select
“Yes” to use INVITE method to refresh the session timer.
Enable 100rel
The use of the PRACK (Provisional Acknowledgment) method enables reliability to
be offered to SIP provisional responses (1xx series). This is very important if PSTN
inter-networking is to be supported. A user’s request to use reliable provisional
responses is invoked by the 100rel tag which is appended to the value of the
required header of initial signalling messages.
Send
Anonymous
If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will
be set to anonymous, essentially blocking the Caller ID from displaying.
Preferred
Vocoder
The GXW-410xv supports up to 5 different Vocoder types including G.711 A-/U-law,
GSM, G.723.1, G.729A/B. The user can configure Vocoders in a preference list
that will be included with the same preference order in SDP message. The first
Vocoder in this list can be entered by choosing the appropriate option in “Choice 1”.
Similarly, the last Vocoder in this list can be entered by choosing the appropriate
option in “Choice 8”.
Special Feature
Default is Standard. Choose the selection to meet some special requirements from
Soft Switch vendors like Nortel, Broadsoft, etc.
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SAVING THE CONFIGURATION CHANGES
Once a change is made, press the “Update” button in the Configuration Menu. The GXW-410xv will
display the following screen to confirm that the changes have been saved. Reboot or power cycle the
GXW-410xv after all the changes are made so that those changes can take effect.
FIGURE 7: SCREEN-SHOT OF SAVE CONFIGURATION
REBOOTING FROM REMOTE
The administrator can remotely reboot the unit by pressing the “Reboot” button at the bottom of the
configuration menu. The following screen will indicate that rebooting is underway.
FIGURE 8: SCREEN-SHOT OF REBOOTING
Grandstream Device Configuration
The device is rebooting now...
You may re-login by clicking on the link below in 30 seconds.
All Rights Reserved Grandstream Networks, Inc. 2005
The user can re-login to the unit after waiting for about 30 seconds.
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VIDEO SURVEILLANCE
The GXW-410xv can be used with an Analog Surveillance Camera to perform video surveillance function.
This application should be used in a LAN environment or when both sides have public IP address.
VIDEO SURVEILLANCE PROCEDURES
¾
Gateway side:
1. In the ADVANCED SETTING page, find the following field and change from default setting
NO to YES, reboot the device.
2. Connect an analog based surveillance camera to the VIDEOIN connection at the back panel
of the unit.
¾
PC side (Monitor Device):
support RFC 3984.
2. Launch VLC.
3. Go to Preferences->Input/Codecs->Demuxers->H264, check “Advanced options” in the
bottom. The option “Frames per Second” will show. Change that value to 5 and then save.
4. Go to Preferences->Input/Codecs->Access modules->Real RTSP, check “Advanced options”
in the bottom. The option “Caching value (ms) will show. Change that value to 1000 and then
save. You may change it to a smaller value to reduce the delay.
5. If the viewer is under NAT, go to Preferences->Demuxers->Access modules->RTP/RTSP,
check “Advanced options” in the bottom. The option “Use RTP over RTSP (TCP)” will show.
Check that option box. (Grandstream does NOT recommend this network environment)
6. Close the Preferences window and go to File->Open Network Stream:
a) Select RTSP as the protocol
b) Enter the URL in the format of rtsp://admin:
ADMIN_PASSWORD@DEVICE_IP_ADDRESS:DEVICE_RTSP_PORT.
Change the blue text according to your configuration:
•
•
•
ADMIN_PASSWORD is the device’s web configuration password for admin.
DEVICE_IP_ADDRESS is the device IP.
DEVICE_RTSP_PORT is the RTSP port setting of the device.
If the port uses default value 554, the port portion can be omitted from the URL
c) Click OK to start the video.
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FIGURE 9: SCREEN-SHOT OF VIDEO SURVEILLANCE*
* PC client side running VLC as monitoring station
FIRMWARE UPGRADE
Firmware (or software) upgrades can be done either via TFTP or HTTP. The corresponding configuration
settings are on the configuration page. End users should NOT touch the configuration settings that are
useful for ITSPs. To upgrade your unit firmware, follow these steps:
1. Under Advanced Settings webpage, enter your TFTP or HTTP Server IP address (or FQDN) next
to the “Firmware Upgrade: Upgrade Server” field.
2. Select via TFTP or HTTP accordingly.
3. If you plan to use Automatic Upgrade, set it to “Yes”, otherwise No (this will make it check for
upgrade every time you reboot).
UPGRADE THROUGH HTTP
To upgrade firmware via HTTP, the field “Firmware Upgrade and Provisioning: Upgrade Via” needs to be
set to HTTP. The “Firmware Server Path” should be set to where the firmware files are located.
For example, the user can use the following URL in the Firmware Server Path:
firmware.mycompany.com: 6688/Grandstream/1.0.0.29 where firmware.mycompany.com is the FQDN of
the HTTP server. It can also be in IP address format. “:6688” is the TCP port the HTTP server listening to,
default http server listens to port 80. “/Grandstream/1.0.0.29” is the RELATIVE directory to the root dir on
HTTP web server.
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UPGRADE THROUGH TFTP
To upgrade firmware via TFTP, set the field “Firmware Upgrade and Provisioning: Upgrade Via” to TFTP.
The TFTP server can be configured in either IP address format or FQDN.
To configure the TFTP server via the Web configuration interface, follow these five steps:
1. Open your browser to input the IP address of the GXW-410xv.
2. Enter the admin password to enter the configuration screen.
3. Enter the TFTP server address or URL in the “Firmware Server Path” field near the
bottom of the configuration screen.
4. Once the “Firmware Server Path” is set, update the change by clicking the “Update”
button.
5. Reboot or power cycle the unit.
If the configured updating server is found and a new code image is available, the GXW-410xv will retrieve
the new image files by downloading them into the GXW-410xv’s SRAM. During this stage, the GXW-
410xv’s LED will blink until the checking/downloading process is completed. Upon verification of
checksum, the new code image will be saved into the Flash. If TFTP fails for any reason (e.g., TFTP
server is not responding, there are no code image files available for upgrade, or checksum test fails, etc),
the GXW-410xv will stop the TFTP process and simply boot using the existing code image in the flash.
Firmware upgrading may take as long as 20 minutes over the Internet, or just 20+ seconds if it is
performed on a LAN. Grandstream recommends conducting firmware upgrades in a controlled LAN
environment if possible.
NO LOCAL TFTP SERVER
For users who do not have a local TFTP server, Grandstream provides a NAT-friendly TFTP server on
the public Internet for users to download the latest firmware upgrade automatically. Please check the
Services section of Grandstream’s Web site to obtain this TFTP server IP address. Alternatively, user
can download and install a free TFTP or HTTP server in his LAN for a firmware upgrade.
A free Windows version TFTP server can be downloaded from:
Directions For Downloading TFTP Server:
1. Unzip the file and put all of the files under the root directory of the TFTP server.
2. Put the PC running the TFTP server and the GXW–410x in the same LAN segment.
3. Go to File -> 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, in the phone’s web configuration page.
5. Configure the Firmware Server Path with the IP address of the PC.
6. Update the change and reboot the unit.
You can also download the free HTTP server from http://httpd.apache.org/ or just use Microsoft IIS web.
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RESTORE FACTORY DEFAULT SETTING
WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone.
Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will
not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP
service provider.
FACTORY RESET
The ONLY way to restore default factory settings is as follows:
1. Unplug the Ethernet cable.
2. Locate a needle sized hole on the back panel of the gateway unit next to the Power connection.
3. Enter a needle like object in this hole and keep it pressed for about 7 seconds.
4. You will see the LAN port LEDs (green and orange) go off and on simultaneously; this indicates
the reset went through.
5. All settings have been erased and the gateway is back to factory settings.
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Examples of GXW-410xv Configurations
APPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER
Scenario: A business with a traditional phone system (with or without broadband access) and an IP
PBX or SIP Servers connecting to an Internet Telephone Service Provider (ITSP).
Anywhere in the world
IPPBX or
SIP Server
FXO Lines
4 or 8 Ports
PSTN
IP/LAN
IP/WAN
Cloud
GXW-410xv
Grandstream IP Phones
PSTN Analog
APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIO
Scenario: a small business with traditional analog PBX lines and broadband access who want to extend
their traditional PBX to virtually anywhere in the world, using the internet. (Any SIP End point, such as
Grandstream BugeTone, HandyTone, GXP-2000 or GXV-3000 are needed in this scenario)
Company A - Boston, MA
6 employees
IPPBX or
SIP Server
Internet
Cloud
IP/LAN
IP/LAN
Optional
FXO Lines
GXW-
FXO Lines
Traditional
PBX
PSTN
Cloud
GXW-410xv
FXS | IPPBX | SIP Platform
Grandstream IP Phones
Anywhere in the world
Any SIP endpoint
Any branch, anywhere
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APPLICATION THREE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER AND VIDEO SURVEILLANCE
Scenario: The GXW-410xv offers an additional video surveillance port which can be configured for
surveillance. It is the only small business analog gateway that offers this security feature.
Branch A – Boston, MA
6 employees
IPPBX or
SIP Server
IPPBX or
SIP Server
Internet
Cloud
IP/LAN
IP/LAN
PSTN
Cloud
GXW 410x
GXW 410x
Grandstream IP
Grandstream IP
Branch B – Denver, CO
4 employees
Anywhere in the world
APPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATION
Scenario Four: The GXW-410xv offers an IP to IP pure IP Communications System configuration,
where all locations use IP phones.
Branch A - Boston, MA
6 employees
IPPBX or
SIP Server
IPPBX or
SIP Server
Internet
Cloud
IP/LAN
IP/LAN
PSTN
Cloud
GXW 410x
GXW 410x
Grandstream IP Phones
Grandstream IP Phones
Branch B – Denver, CO
4 employees
Anywhere in the world
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GLOSSARY OF TERMS
ADSL Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit
from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800 kbps upstream,
depending on line distance.
AGC Automatic Gain Control is an electronic system found in many types of devices. Its purpose is to
control the gain of a system in order to maintain some measure of performance over a changing range of
real world conditions.
ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically
IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The protocol
operates below the network layer as a part of the interface between the OSI network and OSI link layer. It
is used when IPv4 is used over Ethernet
ATA Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP, like
Grandstream HT series products.
CODEC Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A)
converter for translating the signals from the outside world to digital, and back again.
CNG Comfort Noise Generator, generate artificial background noise used in radio and wireless
communications to fill the silent time in a transmission resulting from voice activity detection.
DATAGRAM A data packet carrying its own address information so it can be independently routed from
its source to the destination computer
DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or
compressed. Lossy compression algorithms ordinarily decimate while sub-sampling.
DECT Digital Enhanced Cordless Telecommunications: A standard developed by the European
Telecommunication Standard Institute from 1988, governing pan-European digital mobile telephony.
DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public
switched telephone network, Closed User Groups (CUGs), Local Area Networks, and wireless local loop.
The DECT Common Interface radio standard is a multi-carrier time division multiple access, time division
duplex (MC-TDMA-TDD) radio transmission technique using ten radio frequency channels from 1880 to
1930 MHz, each divided into 24 time slots of 10ms, and twelve full-duplex accesses per carrier, for a total
of 120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all 12
possible accesses (time slots) simultaneously by using different frequencies or using only one frequency.
All signaling information is transmitted from the RFP within a multi-frame (16 frames). Voice signals are
digitally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation.
DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain
names into IP addresses
DID Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going
through an attendant or auto-attendant.
DSP Digital Signal Processor. A specialized CPU used for digital signal processing. Grandstream
products all have DSP chips built inside.
DTMF Dual Tone Multi Frequency. The standard tone-pairs used on telephone terminals for dialing
using in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals
support only 12 of them (0-9, * and #).
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FQDN Fully Qualified Domain Name. A FQDN consists of a host and domain name, including top-level
Grandstream is the second-level domain, and and.com is the top level domain.
FXO Foreign eXchange Office. An FXO device can be an analog phone, answering machine, fax, or
anything that handles a call from the telephone company like AT&T. They should also operate the same
way when connected to an FXS interface.
•
An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions have
their own standards.
•
FXO is complimentary to FXS (and the PSTN).
FXS Foreign eXchange Station. An FXS device has hardware to generate the ring signal to the FXO
extension (usually an analog phone).
•
An FXS device will allow any FXO device to operate as if it were connected to the phone
company. This makes your PBX the POTS+PSTN for the phone.
•
The FXS Interface connects to FXO devices (by an FXO interface, of course).
DHCP The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the
configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses, to
deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to provide
other configuration information such as the addresses for printer, time and news servers.
echo from a voice communication in order to improve voice quality on a telephone call. In addition to
preventing echo from traveling across a network. There are two types of echo of relevance in telephony:
acoustic echo and hybrid echo. Speech compression techniques and digital processing delay often
contribute to echo generation in telephone networks.
H.323 A suite of standards for multimedia conferences on traditional packet-switched networks.
HTTP Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and retrieve
functions of a server
IP Internet Protocol. A packet-based protocol for delivering data across networks.
IP-PBX IP-based Private Branch Exchange
IP Telephony (Internet Protocol telephony, also known as Voice over IP Telephony) A general term for
the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and
other forms of information that have traditionally been carried over the dedicated circuit-switched
connections of the public switched telephone network (PSTN). The basic steps involved in originating an
IP Telephony call are conversion of the analog voice signal to digital format and compression/translation
of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched
networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony
are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is
only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP
Telephony software essentially provides free telephone calls anywhere in the world. However, the
challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border
controllers resolve this issue by providing quality assurance comparable to legacy telephone systems.
IVR IVR is a software application that accepts a combination of voice telephone input and touch-tone
keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and
perhaps other media.
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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-
bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for
Ethernet is 1500 byte.
NAT Network Address Translation
NTP Network Time Protocol, a protocol to exchange and synchronize time over networks The port used
is UDP 123 Grandstream products using NTP to get time from Internet
OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.
OBP/SBCs are put into the signaling and media path between calling and called party. The OBP/SBC
acts as if it was the called VoIP phone and places a second call to the called party. The effect of this
behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the
OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private
OBP/SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network.
Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks
with internet connections using NAT.
PPPoE Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in
Ethernet frames. It is used mainly with cable modem and DSL services.
PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or
called POT (Plain Old Telephone), or circuit switched network.
RTCP Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time
Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does not
transport any data itself. It is used periodically to transmit control packets to participants in a streaming
multimedia session. The primary function of RTCP is to provide feedback on the quality of service being
provided by RTP.
RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video
over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first
published in 1996 as RFC 1889
SDP Session Description Protocol is a format for describing streaming media initialization parameters. It
SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261).
SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice
transmission and uses fewer resources and is considerably less complex than H.323. All Grandstream
products are SIP based
STUN Simple Traversal of UDP over NATs is a network protocol allowing clients behind NAT (or multiple
NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by
the NAT with a particular local port. This information is used to set up UDP communication between two
hosts that are both behind NAT routers. The protocol is defined in RFC 3489. STUN will usually work well
with non-symmetric NAT routers.
TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP,
applications on networked hosts can create connections to one another, over which they can exchange
TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very
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UDP, programs on networked computers can send short messages known as datagrams to one another.
UDP does not provide the reliability and ordering guarantees that TCP does; datagrams may arrive out of
order or go missing without notice. However, as a result, UDP is faster and more efficient for many
lightweight or time-sensitive purposes.
VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing
wherein, the presence or absence of human speech is detected from the audio samples.
VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of
signaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc.
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