Manual FSX GrandStream
Manual FSX GrandStream
WELCOME ................................................................................................. 7
GATEWAY GXW400X OVERVIEW ....................................................................................................... 7
SAFETY COMPLIANCE ........................................................................................................................ 7
WARRANTY........................................................................................................................................... 7
Page 1 of 60
Page 2 of 60
TABLE OF FIGURES
GXW400X User Manual
TABLE OF TABLES
GXW400X User Manual
http://www.grandstream.com/sites/default/files/Resources/gxw400x_gui.zip
Page 3 of 60
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
http://www.grandstream.com/sites/default/files/Resources/ht5xx_gpl.tar.gz.
Page 4 of 60
CHANGE Log
This section documents significant changes from previous versions of GXW400X user manuals. Only
major new features or major document updates are listed here. Minor updates for corrections or editing
are not documented here.
Added feature [Hold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target
Before ReferHold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target
Before ReferHold Target Before Refer] in profile settings, which allows user to hold or not hold the phone
call before refer.
Added feature [Crypto Life Time] in profile settings, which allows user to enable or disable Crypto life
time when using SRTP.
Added feature [Play busy/reorder tone before Loop Current Disconnect] in profile settings, which allows
user to configure if it will play busy/reorder tone before loop current disconnect upon call fail.
Added option [SIP Timer D] to configure RFC 3261 timer D in Advanced Settings.
Changed option name from [Allow DHCP Option 66 to override server] to [Allow DHCP Option 66 or 160
to override server]. Now option160 will be accepted by unit along with option 66 when enabled.
Added feature [Download Device XML Configuration] in advanced settings, which allows user to
download device configuration to local directory in xml format.
Added feature [Upload firmware] in advanced settings, which allows user to upload firmware file from
local directory.
Added feature [Upload configuration] in advanced settings, which allows user to upload configuration
file from local directory.
Added the options to enable/disable [Always send HTTP Basic Authentication Information]
Added the options to restrict the m field sent in SDP [Disable Multiple m line in SDPDisable Multiple m
line in SDP]
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Added the options to enable/disable [Do Not Escape '#' as %23 in SIP URI]
Added network whist/black list function on WAN port [White list for WAN side] [Black list for WAN side]
Added the options to enable/disable [Use P-Preferred-Identity Header] and [Use Privacy Header]
Added option to enable/disable SIP NOTIFY Authentication. [Disable SIP NOTIFY Authentication]
Added option [Use Configured IP ] in DNS mode. Added configurable parameter [Primary IP][Backup
IP1][Backup IP2]
Added option [Use Request Routing ID in SIP INVITE Header] to allow user to replace From and Contact
headers for outgoing calls by [Request URI Routing ID]
Added field [Request URI Routing ID] to allow device to route the calls to individual fxs ports based on
the Request URI user ID in the incoming INVITE.
Added two CPE SSL configuration [CPE SSL Certificate][CPE SSL Private Key]
Changed the SSL Web UI description to [SIP TLS Certificate][SIP TLS Private Key] and [SIP TLS Private
Key Password]
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WELCOME
Thank you for purchasing the Grandstream GXW400X Analog FXS IP Gateway. The GXW400X offers an
easy to manage, easy to configure IP communications solution for any business with virtual and/or branch
locations. The GXW400X supports popular voice codecs and is designed for full SIP compatibility and
interoperability with third 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 FXS Analog 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.
The new GXW400X series has a compact and quiet design (no fans) and offers superb audio quality, rich
feature functionality, strong security protection, and good manageability. It is auto-configurable, remotely
manageable and scalable.
The GXW400X features 4 or 8 port FXS interface for analog telephones, dual 10M/100Mbps network
ports with integrated router, PSTN life line in case of power failure, and an RS232 serial for administration.
In addition, it supports the option of 3 SIP Server profiles, caller ID for various countries/regions, T.38 fax,
flexible dialing plans, security protection (SIPS/TLS), comprehensive voice codec including G.711 (a/ulaw), G.723.1, G.726(16/24/32/48 bit rates), G.729A/B/E and iLBC.
SAFETY COMPLIANCE
The GXW400X is compliant with various safety standards including FCC/CE. Its power adapter is
compliant with UL standard. Warning: use only the power adapter included in the GXW400X package.
Use of alternative power adapter may permanently damage the unit.
WARRANTY
Grandstream has a reseller agreement with our reseller customers. End users should contact the
FIRMWARE VERSION 1.0.14.100
Page 7 of 60
company from whom the product was purchased, for replacement, repair or refund.
If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service
Representative for an RMA (Return Materials Authorization) number. Grandstream reserves the right to
change the warranty policy without prior notification.
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 located at
http://www.grandstream.com/sites/default/files/Resources/gxw400x_usermanual_english.pdf
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.
Page 8 of 60
GXW400X package.
EQUIPMENT PACKAGING
Follow the instructions from the topic Configuring GXW 400X with Web Browser for initial configuration.
The GUI pages will guide you through the remaining steps to set-up your gateway. Examples of the GUI
Interfaces can be downloaded from:
http://www.grandstream.com/sites/default/files/Resources/gxw400x_gui.zip.
Page 9 of 60
GXW 400x
PSTN
Console Line
Reset Power
Supply Connection
LAN/WAN
RJ-45 Ethernet
(PC
Ports
connection)
FXS
Ports
LAN
WAN
RESET
Factory Reset button. Press for 7 seconds to reset factory default settings.
DC 12V
CONSOLE
LINE
When the unit loses power or unit became unregistered, FXS port 1 will be able to
make/receive calls from the PSTN line connected to this port.
FXS1 FXS8
Note: Once the GXW400X is turned on and configured, the front display panel indicates the status of the
unit.
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GXW4008
Display
LEDs
FXS
status
port
Power LED
Ready LED
Console
LAN LED
WAN LED
LED 18
NOTE:
Slow blinking of READY, WAN and LAN LED together indicates a firmware upgrade or
provisioning state.
LED POWER, READY and WAN lights are ON when device is up and running and successfully
registered to the SIP Server.
Slow blinking of READY LED indicates that device has not registered with any SIP Service
provider.
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Page 12 of 60
GXW400X FEATURES
The GXW4000 series is a next generation IP voice gateway that is interoperable and compatible with
leading IP-PBXs, Softswitches and SIP platforms. The GXW4000 series FXS gateway is autoconfigurable, remotely manageable and scalable. The GXW4000 series gateways come in two models the GXW-4004 and GXW-4008, each offering superb voice quality, traditional telephony functionality, easy
deployment, and 4 and 8 FXS ports respectively. Each model features flexible dialing plans, integrated
call routing to support a pure IP network call and an external power supply.
4 or 8 FXS ports
T.38 Fax
Voice Activation Detection (VAD), Comfort Noise Generation (CNG), and Packet Loss
Concealment (PLC)
GXW4004
GXW4008
Telephone Interfaces
4 FXS ports
8 FXS ports
SIP Provisioning
Network Interface
4 Concurrent Calls
8 Concurrent Calls
Capabilities
Page 13 of 60
Unit
(supports
protocol),
RTP
G.168
and
AAL2
compliant
Echo
G.168
compliant
Echo
Cancellation,
G.711
PSTN Fail-over
Voice Compression
(VAD/CNG
(VAD/CNG
decoder,
with
format)
encoder
G.723.1A,
and
G.726(ADPCM
16/24/32/40
bit
rates),
format)
encoder
and
G.729A/B/E, iLBC
adaptive
jitter
buffer,
clock
control),
Fax over IP
Fax Pass-through,
QoS
Diffserv,
TOS,
802.1
P/Q
VLAN
tagging
Transport Protocol
RTP
RTP
DTMF Method
IP Signaling
Provisioning
Control
TLS/SIPS
TLS/SIPS
Management
access
Dial Plan
Yes
Yes
UPnP Support
Yes
Yes
Caller ID
Page 14 of 60
DTMF-based CID
Yes
Yes
EMC
Class B
Class B
HARDWARE SPECIFICATION
The hardware specifications of the GXW FXS series are detailed in Table 4.
TABLE 4: HARDWARE SPECIFICATION OF GXW400X SERIES GATEWAYS
GXW4004
GXW4008
Ports
4 FXS Ports
8 FXS Ports
Network interface
RJ45
Full Duplex
RJ45
Duplex
for WAN,10/100Base-TX,
Full Duplex
Duplex
PSTN Port
Console
Universal Switching
Power Adaptor
Max
Max
UL certified
UL certified
Mounting
Dimension
Weight
Temperature
32~104F / 0~40C
32~104F / 0~40C
Humidity
Safety
UL
UL
Compliance
FCC, CE
FCC, CE
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BASIC OPERATIONS
UNDERSTANDING GXW VOICE PROMPTS
GXW400X has a built-in voice prompt menu for simple device configuration. To enter the voice prompt
menu, press *** on the standard analog phone connected to any FXS port.
Menu
Voice Prompt
Users Options
Main Menu
DHCP
Mode,
PPPoE
02
IP Address + IP address
03
Subnet + IP address
04
Gateway + IP address
05
07
Preferred Vocoder
PCM U
PCM A
iLBC
G-726
G-723
G-729
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10
MAC Address
12
13
Firmware
Server
IP
Address
14
new IP address.
Configuration
15
Server
IP
Address
new IP address.
Upgrade Protocol
16
Firmware Version
17
Firmware Upgrade
Direct IP Calling
47
1.
always check
2.
3.
never upgrade
71-78
Phone
calls
between
GW 400x
86
Voice Mail
99
RESET
Invalid Entry
Page 18 of 60
Examples:
1. Dial a number (e.g. (626) 666-7890), first enter the prefix number (usually 1+ or international
code) followed by the phone number. Press # or wait for 4 seconds. Check with your VoIP
service provider for further details on prefix numbers.
DIRECT IP CALLS
Direct IP calling allows two parties, that is, a FXS Port with an analog phone and another VoIP Device, to
talk to each other in an ad hoc fashion without a SIP proxy.
Using IVR
1. Pick up the analog phone then access the voice menu prompt by dial ***
2. Dial 47 to access the direct IP call menu
3. Enter the IP address using format ex. 192*168*0*160 after the dial tone.
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Destination ports can be specified by using * (encoding for :) followed by the port number.
Examples:
a) If the target IP address is 192.168.0.160, the dialing convention is
*47 or Voice Prompt with option 47, then 192*168*0*160.
Followed by pressing the # key if it is configured as a send key or wait 4 seconds. In this case,
the default destination port 5060 is used if no port is specified.
b) If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:
*47 or Voice Prompt with option 47, then 192*168*0*160*5062 followed by pressing the # key
if it is configured as a send key or wait for 4 seconds.
NOTE: When completing direct IP call, the Use Random Port should set to NO. You can not make
direct IP calls between FXS ports since they are using same IP.
CALL HOLD
Place a call on hold by pressing the flash button on the analog phone (if the phone has that button).
Press the flash button again to release the previously held Caller and resume conversation. If no flash
button is available, use hook flash (toggle on-off hook quickly). You may drop a call using hook flash.
CALL WAITING
Call waiting tone (2 short beeps) indicates an incoming call, if the call waiting feature is enabled. Toggle
between incoming call and current call by pressing the flash button. First call is placed on hold. Press
the flash button to toggle between two active calls.
CALL TRANSFER
BLIND TRANSFER
Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C:
3. Caller A presses FLASH on the analog phone to hear the dial tone.
4. Caller A dials *87 then dials caller Cs number, and then # (or wait for 4 seconds).
5. Caller A will hear the dial tone. Then, A can hang up.
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NOTE: Enable Call Feature must be set to Yes in web configuration page.
ATTENDED TRANSFER
Assume that Caller A and B are in conversation. Caller A wants to Attend Transfer B to C:
1. Caller A presses FLASH on the analog phone for dial tone.
2. Caller A then dials Caller Cs number followed by # (or wait for 4 seconds).
3. If Caller C answers the call, Caller A and Caller C are in conversation. Then A can hang up to
complete transfer.
4. If Caller C does not answer the call, Caller A can press flash to resume call with Caller B.
NOTE: When Attended Transfer fails and A hangs up, the GXW400X will ring back user A to remind A
that B is still on the call. A can pick up the phone to resume conversation with B.
3-WAY CONFERENCING
Assuming that call party A and B are in conversation. A (GXW400X) wants to bring C in a conference:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial
tone.
2. A dials Cs number then # (or wait for 4 seconds).
3. If C answers the call, then A presses FLASH to bring B, C in the conference.
4. If C does not answer the call, A can press FLASH back to talk to B.
5. If A presses FLASH during conference, C will be dropped out.
6. If A hangs up, the conference will be terminated for all three parties when configuration
Transfer on Conference Hangup is set to No. If the configuration is set to Yes, A will
transfer B to C so that B and C can continue the conversation.
HUNTING GROUP
This feature allows the user to setup a single SIP account on the gateway and have the ability to use all
FXS ports to make/receive calls. Using this feature, all ports active in same Hunting Group will have the
same phone number and incoming calls will be distributed in a Linear or Circular manner among the ports
FIRMWARE VERSION 1.0.14.100
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active in that Hunting Group. The number of hunting groups is limited by the number of ports each GXW
model has - i.e. each port can be its own Hunting Group. The most practical and efficient way to use
Hunting Groups is to assign 2 or 3 ports to separate Hunting Groups.
One additional and popular way to use the Hunting Group feature is called multiplexed analog lines. In
this configuration, a legacy PBX system with 8 FXO trunks can be connected to 8 GXW 4008 ports
configured as Hunting Group. The GXW can be registered to a SIP server provider using only one phone
number. If the SIP service provider allows multiple calls to the same number, the GXW will allow 8
concurrent calls to the same SIP number. All office members can be reached remotely using the same
phone number in a round-robin fashion.
1. Configure the SIP account from your VoIP Service Provider on FXS port 1 under FXS Ports
webpage.
2. Select Active under the Hunting Group drop box for FXS port 1.
3. For the remaining ports (say 2, 3 and 4) select 1 for Hunting Group. Ports 2, 3 and 4 are now
active members of the hunting group associated with port 1.
This configuration will route all calls directed to FXS port 1 to ports 2, 3 and/or 4 in round robin fashion
respectively if port 1 is busy or times out. You can configure the ring timeout on the Profile page.
FXS Port #1: SIP UserID and Authenticate ID entered, Hunting group set to "Active"
FXS Port #2: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"
FXS Port #3: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"
FXS Port #4: SIP UserID and Authenticate ID entered, Hunting group set to "Active"
FXS Port #5: SIP UserID and Authenticate ID left blank, Hunting Group set to "4"
FXS Port #6: SIP UserID and Authenticate ID left blank, Hunting Group set to "4"
FXS Port #7: SIP UserID and Authenticate ID entered, Hunting group set to "Active"
FXS Port #8: SIP UserID and Authenticate ID left blank, Hunting Group set to "7"
Please be aware, the choice of 1 for ports 2 and 3, the choice of 4 for ports 5 and 6, the choice 7 for port
FIRMWARE VERSION 1.0.14.100
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8 is required to indicate that the SIP account tied to port marked as Active will be used for all members
of the same Hunting group. Needless to say, those members of the same Hunting group may not be
sequential ports. In following example ports 3, 5 and 7 tied to SIP Account configured in Port #1 marked
as Active, and ports 4,6,8 tied to SIP Account configured in Port #2 marked as Active as well.
FXS Port #1: SIP UserID and Authenticate ID entered, Hunting group set to "Active"
FXS Port #2: SIP UserID and Authenticate ID entered, Hunting Group set to "Active"
FXS Port #3: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"
FXS Port #4: SIP UserID and Authenticate ID left blank, Hunting group set to "2"
FXS Port #5: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"
FXS Port #6: SIP UserID and Authenticate ID left blank, Hunting Group set to "2"
FXS Port #7: SIP UserID and Authenticate ID left blank, Hunting group set to "1"
FXS Port #8: SIP UserID and Authenticate ID left blank, Hunting Group set to "2"
Note: A single call directed to the SIP account will NOT result in all ports ringing at the same time. They
will ring in the hunting group only. This feature is applicable to incoming calls only.
There are two types of hunting groups, Linear and Circular. Linear style will sort the call to the lowestnumbered available line, this is also called serial hunting. Circular style will distribute the calls "roundrobin". If a call is assigned to line 1, the next call goes to 2 and the next to 3. The succession throughout
each of the lines continues even if one of the previous lines becomes available. When the end of the hunt
group is reached, the hunting starts over at the first line. Lines are skipped if they are still busy on a
previous call. These two hunting styles can be configured from the Profile_x page.
INTER-PORT CALLING
In some cases a user may want to make phone calls between the phones connected to multiple ports of
the same gateway when it is used as a stand alone unit, without the use of a SIP server. This feature will
also be applicable when the gateway is used with Hunting Groups and is registered to SIP server only
with one master number. In such cases users still will be able to make inter-port calls by using the IVR
feature.
For example on the GX4004 and GXW4008 the user connected to port 1 can reach the user connected to
port 3 by dialing *** and 73. Digit 7 indicated using inter-port calling feature, digit 3 indicates the port
number which should be reached. At the same manner the user connected to port 4 can reach the user
connected to port 8 by dialing *** and 78.
FIRMWARE VERSION 1.0.14.100
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The RJ-11 line jack on the GXW400X side connected to a legacy PSTN line, functions as a pass through
jack when the GXW400X loses power or becomes unregistered. In this case, analog phone connected to
FXS port 1 will be directly connected to RJ-11 jack marked as LINE. The pass through/life line mode
enables the user to use the analog phone for PSTN calls directly without using an access code.
GXW400X supports fax in two modes: 1) T.38 (Fax over IP) and 2) Fax Pass through. T.38 is the
preferred method because it is more reliable and works well in most network conditions. If the service
provider supports T.38, please use this method by selecting T.38 as fax mode (default). If the service
provider does not support T.38, pass-through mode may be used. If you have problems with sending or
receiving Fax, toggle the Fax Tone Detection Mode setting.
GXW400X supports RADIUS for authentication, authorization and billing purposes. Primary and
secondary RADIUS server configurations are available to provide redundancy to this feature. In case
Primary Radius server becomes unusable, RADIUS requests will be automatically sent to the secondary
server. When at least one RADUIS server was configured, the device will allow users to make phone calls
only after authorization from RADIUS server has been received. CDR (Call Detail Record) is also sent to
the RADIUS server for billing purposes. RAIDUS server can send requests to terminate calls when run
out of pre-paid credit.
The GXW400X will be able to work in VoIP billing environment using redundant double server
configuration. User will be able to configure primary and secondary RADUIS server IP Addresses or
FQDNs. Once at least one RADUIS server was configured, the device will allow users to make phone
calls only after permission from RADIUS server has been received. In case Primary Radius server
becomes unusable, secondary will take role of primary and will manage credit recourses in the network.
Imbedded RADIUS client also supports request generated by Radius server to terminate calls when run out of
pre-paid credit.
Page 24 of 60
CALL FEATURES
GXW400X supports the traditional telephony features available in a PBX as well as additional advanced
telephony features.
Key
Call Features
*02
Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729
(G729), *0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726r40), *027201 (iLBC)
*03
Disable LEC (pe call) Dial *03 + number . No dial tone is played in the middle.
*16
Enable SRTP
*17
Disable SRTP
*30
*31
*67
*82
*47
Direct IP Calling. Dial *47 + IP address. No dial tone will be played in the middle. Detail see
Direct IP Calling section on page 12.
*50
*51
*69
Call Return Service: Dial *69 and the phone will dial the last incoming phone number
received.
*70
*71
*72
Unconditional Call Forward: Dial *72 and then the forwarding number followed by #. Wait
for dial tone and hang up. (dial tone indicates successful forward)
*73
Cancel Unconditional Call Forward: Dial *73 and wait for dial tone, then hang up.
*74
Enable Paging Call: Dial *74 and then the destination phone number you want to activate in
Paging mode.
*78
Enable Do Not Disturb (DND): When enabled all incoming calls will be rejected.
*79
Disable Do Not Disturb (DND): When disabled, incoming calls will be accepted.
Page 25 of 60
*87
Blind Transfer
*90
Busy Call Forward: Dial *90 and then the forwarding number followed by #. Wait for dial
tone then hang up.
*91
Cancel Busy Call Forward: dial *91. Wait for dial tone. Hang up.
*92
Delayed Call Forward: Dial *92 and then the forwarding number followed by #. Wait for
dial tone then hang up.
*93
Cancel Delayed Call Forward: Dial *93 for a dial tone, then hang up.
Flash/Hook
If user hears call waiting beep, flash/hook will switch to the new incoming call. Also used to
switch to a new channel for a new call.
Page 26 of 60
CONFIGURATION GUIDE
CONFIGURING GXW400X VIA VOICE PROMPT
DHCP MODE
Select voice menu option 01 to enable GXW400X to use DHCP.
STATIC IP MODE
Select voice menu option 01 to enable GXW400X to use STATIC IP mode, then use option 02, 03, 04, 05
to set up IP address, Subnet Mask, Gateway and DNS server respectively.
UPGRADE PROTOCOL
Select voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose
between TFTP and HTTP.
The GXW400X series gateway has an embedded Web server that allows users to configure the
GXW400X through a web browser.
Page 27 of 60
NOTE:
WAN side HTTP access is disabled by default for security reasons. You can enable HTTP access on
the configuration page by setting WAN side HTTP access to be YES.
Initial access to the configuration pages is always from the LAN port. The instructions are listed above.
The IVR announces 12 digits IP address, you need to strip out the leading 0 in the IP address. For ex.
IP address: 192.168.001.014, you need to type in http://192.168.1.14 in the web browser.
Once the HTTP request is entered and sent from a web browser, the user will see a log-in screen. There are
two default passwords for the login page:
User Level:
Password:
123
Administrator Level
Admin
The password is case sensitive with maximum length of 25 characters. The factory default password for End
User and administrator is 123 and admin respectively. Only an administrator can access the ADVANCED
SETTING, Profile 1, Profile 2 and FXS PORTs configuration pages. Please reference the GUI pages using
the following link:
http://www.grandstream.com/sites/default/files/Resources/gxw400x_gui.zip.
Page 28 of 60
IMPORTANT SETTINGS
The end-user must configure the following settings according to the local environment.
NOTE: Most settings on the web configuration pages are set to the default values.
NAT SETTINGS
If you plan to keep the gateway within a private network behind a firewall, we recommend using STUN
Server. The following three (3) settings are useful in the STUN Server scenario:
1. STUN Server (under Advanced Settings webpage)
Enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the
internet and enter it on this field. If using Public IP, keep this field blank.
2. Use Random Ports (under Advanced Settings webpage)
It really depends on your network settings, so set this parameter to Yes or No, whichever works.
Generally if you have multiple IP devices under the same network, it should be set to Yes. If using a
Public IP address, set this parameter to No.
3. NAT Traversal (under the Profile web pages)
Set this to Yes when gateway is behind firewall on a private network.
DTMF METHODS
DTMF Settings are in Profile pages.
DTMF in-audio
You can enable set priority of DTMF methods according to your preference, from Priority 1 to 3. This
setting should be based on your server DTMF setting.
G729 A/B/E
G723
G726 (16/24/32/40)
Page 29 of 60
iLBC
DEFINITIONS
This section will describe the options in the Web configuration user interface. As mentioned, a user can
log in as an administrator or end-user.
STATUS: Displays the network status, account status, software version and MAC-address of the
phone
BASIC SETTINGS: Basic preferences such as date and time settings, multi-purpose keys and
LCD settings can be set here.
ADVANCED SETTINGS: To set advanced network settings, codec settings and XML
configuration settings.
FXS PORTS: To configure each of the FXS ports and Hunting Groups etc.
TABLE 7: BASIC SETTINGS
Password to access the Web Configuration Menu. This field is case sensitive with a
maximum length of 25 characters.
Web Port
By default, HTTP uses port 80. This field is for customizable web port.
Telnet Server
Default is Yes.
IP Address
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DHCP hostname
This option specifies the name of the client. This field is optional but may be
required by some Internet Service Providers. Default is blank.
PPPoE account ID
PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point
Protocol over Ethernet) connection.
PPPoE password
This field is optional. If your ISP uses a service name for the PPPoE connection,
enter the service name here. Default is blank.
Time Zone
Controls how the date/time is displayed according to the specified time zone.
Language
Device Mode
This parameter controls whether the device is working in NAT router mode or Bridge
mode. Save the setting and reboot prior to configuring the GXW.
Uplink Bandwidth
Downlink Bandwidth
Page 31 of 60
If set to Yes, the GXW400X would act as an UPnP gateway for your UPnP enable
applications. UPnP - Universal Plug n Play
If set to Yes, the GXW400X will respond to the PING. Default is No.
Yes setting may make the gateway vulnerable to a Denial of Service attack.
If set to Yes, user can access the configuration page through the WAN port,
Access
instead of through the PC port. WARNING: this configuration is less secure than
default option. Default is No.
HTTP/telnet:
Case 1: WAN side telnet/HTTP access enabled
- If white list exists, then ONLY these IP addresses are ALLOWED to web and telnet
access.
- If black list exists and white list is empty, then ONLY these IP addresses are NOT
ALLOWED to web and telnet access.
Case 2: WAN side telnet/HTTP access is not allowed:
-All addresses are NOT ALLOWED http and telnet access.
Black list for WAN side
List the IP address or IP range in the White list. The same rules as white list.
This allows you to change/set the MAC address on the WAN interface.
Default is Yes.
Base IP for the LAN port which functions as a Gateway for the subnet.
Default value is 192.168.2.1.
When the device detects WAN IP is conflicting with LAN IP, the LAN base IP
address will be changed based on the network mask -- the effective subnet will be
increased by 1. For example; 192.168.2.1 will be changed to 192.168.3.1 if net
mask is 255.255.255.0. Then the device will reboot
Default is 100
Default is 199
Value is set in units of hours. Default value is 120 hrs (5 Days.) The time IP
address is assigned to the LAN clients.
DMZ IP
Page 32 of 60
Port Forwarding
In addition to the Basic Settings configuration page, end users also have access to the Device Status
page.
TABLE 8: STATUS PAGE
MAC Address
The device ID in hexadecimal format. This is needed for Internet Service Provider
troubleshooting. Note: there are separate MAC addresses for the WAN side and the
LAN side. The LAN MAC address will be used for provisioning and can be found on the
label on original box and on the label located on the bottom panel of the device.
WAN IP Address
Product Model
Software Version
Program: This is the main software release. Boot and Loader are not changed often.
System Up Time
PPPoE Link Up
NAT
Shows type of NAT the GXW400X is connected to via its WAN port. It is based on
STUN protocol.
Port Status
Port
Hook
Registration
DND
FXS1
On Hook
Registered
No
FXS2
Off Hook
Registered
No
FXS3
On Hook
Not
No
Forward
Busy
Delayed
Forward
Forward
613
614
Registered
FXS8
On Hook
Registered
Yes
615
Advanced User configuration includes not only the end user configuration, but also advanced
configurations such as:
miscellaneous configuration.
FIRMWARE VERSION 1.0.14.100
Page 33 of 60
Page 34 of 60
Admin Password
Administrator password.
Settings page.
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
STUN Server
For a sample list of public free STUN servers please refer to:
http://www.voip-info.org/wiki-STUN
Keep-alive interval
This parameter specifies the number of seconds after which a new blank UDP
packet will be sent out to the proxy/server port in order to have the port stay open
and the device remain reachable. Default is 20 seconds.
Use STUN keep-alive to detect WAN side network problems. If keep-alive request
network connectivity
does
not yield any response for configured number of times, the device will restart the
TCP/IP
stack. If the STUN server does not respond when the device boots up, the
feature is
disabled.
Enables GXW400X to download firmware or configuration file using either the TFTP
Provisioning
or HTTP/S protocols.
This is the IP address of the configured TFTP server. If selected and it is non-zero
or not blank, the GXW400X retrieves the new configuration file or new code image
from the specified TFTP server at boot time.
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 is saved into the Flash memory.
Note: Do NOT interrupt the firmware upgrade process (especially the power
supply) as this will damage the device. Depending on the network environment
this process may take up to 15 or 20 minutes.
Via HTTP or HTTPS Server
The URL for the HTTP server used for firmware upgrade and configuration via HTTP.
Note: If Auto Upgrade is set to No, GXW400X will only do HTTP download once at
boot up.
FIRMWARE VERSION 1.0.14.100
Page 35 of 60
IP address or domain name of firmware server. That URL of the server that hosts the
firmware release. The default server is: fm.grandstream.com/gs
IP address or domain name of configuration server. The server hosts a copy of the
configuration file to be installed on the gateway. The default server is:
fm.grandstream.com/gs
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 Password
This field enables user to store different versions of firmware files in one single
directory on the firmware server. If configured, only the firmware file with the
matching prefix will be downloaded.
This field enables user to store different versions of firmware files in one single
directory on the firmware server. If configured, only the firmware file with the
matching postfix will be downloaded.
This field enables user to store different configuration files in one single directory on
the configuration server. If configured, only the configuration file with the matching
prefix will be downloaded.
This field enables user to store different configuration files in one single directory on
the configuration server. If configured, only the configuration file with the matching
postfix will be downloaded.
If set to Yes, configuration and upgrade server information can be obtained using
DHCP option 66 or option 160 from DHCP server located in customers environment.
Note: If DHCP Option 66 is enabled, the gateway will attempt downloading a
configuration file from the server URL provided by DHCP, even though Config
Server Path is left blank.
Automatic Upgrade
Page 36 of 60
Device will not challenge NOTIFY with 401 when set to Yes.
Authentication
Authenticate Conf File
If set to Yes, configuration file is authenticated before being accepted. This protects
the configuration from unauthorized modifications.
Firmware Key
The GXW400X series supports SIP over TLS. It has built-in private key and SSL
certificate. The user specified SSL certificate used for SIP over TLS is in X.509
format.
You may also customize the SSL Private Key. The user specified SSL private key
used for SIP over TLS is in X.509 format.
Password
ACS URL
ACS Username
ACS Password
Default is No. If set to YES, device will send inform packets to the ACS
Frequency that the inform packets will be sent out to the ACS
Connection Request
Username
Connection Request
Password
Connection Request Port
Set a port number for the ACS to connect to this device, default is 7547
Configuration option for all FXS ports ring cadence for all incoming calls.
(Syntax: c=on1/off1-on2/off2-on3/off3; [...]) Default is set to c=2000/4000; (US
standards)
Page 37 of 60
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. ONis 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
Please refer the document below to determine your local call progress tones:
http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
Disables the voice prompt configuration. Default is No. If set to Yes accessing
integrated voice menu will be impossible.
Disables the Direct IP Call function. Default is No. If set to Yes direct IP-to-IP
calling will not be supported.
Lifeline Mode
Life line feature ensures user can place/receive a PSTN call in an emergency
situation.
1.
If set to Auto, in case of power loss or loss of SIP registration, the PSTN
line will be seamlessly connected to analog phone connected to FXS port.
2.
3.
If set to Always Disconnected, user can only place VoIP calls, regardless
of any power loss and/or SIP registration problems. User will be unable to
place/receive any PSTN calls.
This feature allows users to place an outbound PSTN call in case there is a loss of
an active registration (SIP server unreachable) of all FXS profiles. If set to YES,
when GXW400X recognizes a loss of registration, all outgoing calls will be routed to
an FXO gateway.
The use of this option presumes a configured GXW410x or another FXO gateway
with an active PSTN line connection.
FXO Gateway
Page 38 of 60
NTP server
URI or IP address of the NTP (Network Time Protocol) server. Used by the phone to
synchronize the date and time. An extensive list of public NTP servers can be found
at http://www.ntp.org
Default is 1440. Updates the Network Time Protocol (Values range from 5 1440
minutes)
Syslog Server
The IP address or URL of System log server. The server collects system log
information from the device.
Syslog Level
Select the GXW400X 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.
2.
3.
4.
5.
6.
7.
8.
9.
Example:
Ethernet link is up
Send SIP Log
If Syslog is enabled and Send SIP Log is set to YES, then SIP messages will also be
delivered via Syslog. Default is set to NO.
Secret
configuration
Page 39 of 60
Port
Secondary RADIUS Acct
Port
Secondary Radius Server
Secret
configuration
RADIUS Timeout
Default value is 2 seconds. The time between retries the GXW will send AccessRequest message to RADIUS server in purpose to authenticate it.
RADIUS Retry
Default value is 3 times. Number of times the device will try to authenticate itself with
preconfigured RADIUS server during initialization process.
Download Device
This setting allows user to download a text file containing all the P values of each
Configuration
setting as configured on the unit. (Note: For Security Reasons, any Password will not
be Downloaded)
Allows user to download and save a XML file containing all the P values of each
Configuration
setting as configured at that point on the unit. (Note: For Security Reasons, all
Passwords wont be Downloaded)
Upload firmware
Upload configuration
FXS Port
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.
Authenticate ID
Password
SIP service subscribers account password for GXW400X to register to (SIP) servers
of ITSP.
Name
Profile ID
Page 40 of 60
Hunting Group
This feature enables the gateway to register all existing FXS ports with the same
phone number. Each incoming call will be routed to first available port in Linear or
Circular mode. User may configure all ports as members of the same Hunting Group
or it may configure different port combinations for more than one Hunting Group.
For example: Ports 1, 3 and 5 are members of the same Hunting Group, the rest of
the ports may have separate numbers and may be reached independently.
Any port, member of a Hunting Group that is not registered with a SIP account, will be
able to place outbound calls using the SIP credentials of the primary Hunting Group
port.
For example: Port 1, 3 and 5 are members of the same Hunting Group. Port 1 is
registered with a SIP account. Ports 3 and 5 are not registered. Ports 3 and 5 will be
able to place outbound calls using the SIP account of port 1.
Select appropriate value for Hunting Group feature. The original SIP account should
be set to Active while the group members should be set to the port number of the
Active Port.
FXS Port #1: SIP UserID and Authenticate ID entered, Hunting group set to "Active"
FXS Port #2: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"
FXS Port #3: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"
FXS Port #4: SIP UserID and Authenticate ID entered, Hunting group set to "Active"
FXS Port #5: SIP UserID and Authenticate ID left blank, Hunting Group set to "4"
FXS Port #6: SIP UserID and Authenticate ID left blank, Hunting Group set to "4"
FXS Port #7: SIP UserID and Authenticate ID entered, Hunting group set to "Active"
FXS Port #8: SIP UserID and Authenticate ID left blank, Hunting Group set to "7"
Hunting Group 1 contains ports 1, 2, 3. Hunting Group 4 contains ports 4, 5, 6.
Hunting Group 7 contains ports 7, 8.
Request URI Routing ID If configured, device will route the incoming call to designated port by request URI
user ID in SIP INVITE.
Enable Port
If set to No, FXS port will become inactive (Default is set to Yes)
Port#
Offhook Auto-dial
This feature allows you to automatically dial the number specified in this field as soon
as the port is offhooked, i.e. when the receiver on the phone connected to Port# is
picked up.
Page 41 of 60
Offhook Auto-Dial
Configure the delay time for offhook auto-dial function. Range is 0-60 seconds,
Delay
default is 0.
Map to FXO Gateway IP This is used when peering with an FXO gateway of any brand. You have to specifically
mention the IP and sip port where the call will be sent to.
and Port
Profile Active
Primary SIP Servers IP address or Domain name provided by VoIP service provider.
Failover SIP Servers IP address or Domain name provided by VoIP Service provider.
This server will be used if the Primary SIP server becomes unavailable.
Default is no. If set to yes it will register to Primary Server if registration with Failover
Server
server expires
Outbound Proxy
SIP transport
User can select UDP or TCP or TLS. Please make sure youre SIP Server or network
environment supports SIP over the selected transport method. Default is UDP.
NAT Traversal
This parameter defines whether the GXW400X NAT traversal mechanism is activated or
not. If activated (by choosing Yes) and a STUN server is also specified, then the
GXW400X performs according to the STUN client specification. Under this mode, the
embedded STUN client will detect if and what type of firewall/NAT is being used. If the
detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the GXW400X
will 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 GXW400X
will periodically (every 20 seconds) send a blank UDP packet (with no payload data) to
the SIP server to keep the hole on the NAT open.
Page 42 of 60
DNS Mode
One from the 3 modes are available for DNS Mode configuration:
-A Record (for resolving IP Address of target according to domain name)
-SRV (DNS SRV resource records indicates how to find services for various protocols)
-NAPTR/SRV (Naming Authority Pointer according to RFC 2915)
-Use Configured IP (Use the three configured IP address instead of any DNS query)
One mode can be chosen for the client to look up server.
The default value is A Record
Primary IP
Backup IP1
Backup IP2
Tel URI
If set to Yes, device will use Request URI Routing ID defined in FXS ports settings to
ID in SIP INVITE
Header
SIP Registration
This parameter controls whether the GXW400X 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 users registration information is cleared on reboot.
Default is No. If set to Yes, user can place outgoing calls even when not registered (if
Registration
allowed by Internet Telephone Service Provider) but is unable to receive incoming calls.
Any port, member of a Hunting Group that is not registered with a SIP account, will be
able to place outbound calls using the SIP credentials of the primary Hunting Group port.
For example: Port 1, 3 and 5 are members of the same Hunting Group. Port 1 is
registered with a SIP account. Ports 3 and 5 are not registered. Ports 3 and 5 will be
able to place outbound calls using the SIP account of port 1, even if Outgoing Call
without Registration is set to No
Register Expiration
Allows the user to specify the time frequency (in minutes) for the GXW400X 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).
Page 43 of 60
Reregister before
This parameter allows the user to specify the reregistration time before expiration.
Expiration
Local SIP port
Defines the local SIP port the GXW400X will listen and transmit. The default value for
Profile 1 is 5060 and 6060 for Profile 2.
Defines the local RTP port pair the GXW400X will listen and transmit. It is the base RTP
port for channel 0. When configured, channel 0 will use this port _value for; channel 1
will use port_value+2 for RTP. The default value for Profile 1 is 5004 and 6004 for Profile
2.
Default is No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.
This is usually necessary when multiple GXW400X/HT50X are behind the same NAT.
Default is Yes. Allows user to hold the phone call before refer it. If set to No, the call will
Refer
Default is No. If set to YES, then for Attended Transfer, the Refer-To header uses the
Contact
Transfer on
Default is No. In which case if conference originator hangs up the conference will be
Conference
terminated. When option YES is chosen, originator will transfer other parties to each
Hang up
other so that B and C can choose either to continue the conversation or hang up.
Default is No. you can make a Conference by pressing Flash key. If set to Yes, you
3-Way Conference
Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.
Route Header:
Support SIP Instance
Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP
ID
Default is No. If set to yes all incoming SIP messages will be strictly validated according
message
to RFC rules. If message does not pass validation process, call will be rejected.
Default is No. Check the SIP User ID in Request URI. If they dont match, the call will be
incoming INVITE
rejected.
Authenticate incoming Default is No. If set to Yes, device will challenge the incoming INVITE for the
INVITE
Default is No. If incoming SIP message does not match with SIP Server, it will be
rejected.
Proxy Only
Page 44 of 60
If set to Default, it will only add Privacy or PPI header when special feature is not
Telkom SA or CBCOM.
Use P-Preferred-
If set to Default, it will only add Privacy or PPI header when special feature is not
Identity Header
Telkom SA or CBCOM.
SIP T1 Timeout
T1 is an estimate of the round-trip time between the client and server transactions.
If the network latency is high, select larger value for more reliable usage.
SIP T2 Interval
SIP Timer D
Preferred DTMF
The GXW400X supports up to 3 different DTMF methods including in-audio, via RTP
(RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.
order)
Disable DTMF
Default is No. If set to yes, use above DTMF order without negotiation
Negotiation
Send Hook Flash
Event
Enable Call Features
Default is Yes. (If Yes, call features using star codes will be supported locally)
Proxy Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Use NAT IP
Header
Page 45 of 60
Distinctive Ringtone
Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is
configured, then the device will ONLY uses this ring tone when the incoming call is from
the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller
ID is configured, the selected ring tone will be used for all incoming calls. Distinctive ring
tones can be configured not only for matching whole number, but also for matching
prefixes. In this case symbol * (star) will be used.
If server supports Alert-Info header and standard ring tone set (Bellcore) or distinctive
ring tone 1-10 is specified, then the ring tone in the Alert-Info header from server will be
used.
For example:
If configured as *617, Ring Tone 1 will be used in case of call arrived from
Massachusetts. Any other incoming call will ring using cadence defined in parameter
System Ring Cadence located under Advanced Settings Configuration page.
Disable Call Waiting
Default is No. If set to YES Call Waiting indication information will not be provided to
analog phone connected to this FXS port.
Default is No. If set to YES Call Waiting caller ID will not be provided to analog phone
Caller ID
Default is No. This is to disable the stutter Call Waiting Tone when a Call Waiting call
Tone
Disable Reminder
Default is No. This is to disable the Reminder Ring that is played when a call is waiting
Visual message indicator is a special on-hook caller ID type message that enables and
disables the message waiting light on certain phones. GXW400X has this feature
enabled by default. However, certain phones (rare) that do not support it may mistakenly
treat this CID signal as an incoming call. A configuration option is needed to turn on MWI
in this case.
If set to Yes, device will use # instead of %23 in the send URI.
Incoming call will stop ringing when not picked up given a specific period of time.
Default is 20 seconds. If call is not answered within this designated time period, the call
Timeout
Page 46 of 60
Linear and Circular. Linear style will sort the call to the lowest-numbered available line,
this is also called serial hunting. Circular style will distribute the calls "round-robin". If a
call is assigned to line 1, the next call goes to 2 and the next to 3. The succession
throughout each of the lines continues even if one of the previous lines becomes
available. When the end of the hunt group is reached, the hunting starts over at the first
line. Lines are skipped if they are still busy on a previous call. These two hunting styles
can be configured from the Profile_x page.
Default value is 20 seconds. In case this feature activated using * codes (*92 code), the
Wait Timeout
Default is 4 seconds. Call will be completed within this time interval if no additional key
entry occurs.
Early Dial
Default is No.
whether the phone will send an early INVITE each time a key is pressed when a user
dials a number. If set to Yes, an INVITE is sent using the dial-number collected thus
far; Otherwise, no INVITE is sent until the (Re-)Dial button is pressed or after about 5
seconds have elapsed if the user forgets to press the Re-Dial button. The Yes option
should be used ONLY if there is a SIP proxy configured and the proxy server supports
484 Incomplete Address response. Otherwise, the call will likely be rejected by the proxy
(with a 404 Not Found error).
This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP
calling.
Dial Plan Prefix
Allows users to configure the # key as the Send (or Dial) key. If set to Yes, # will
send the number. In this case, this key is essentially equivalent to the Dial key. If set
to No, this # key can be included as part of number.
Page 47 of 60
Dial Plan
2.
b.
c.
^ - exclude;
d.
e.
f.
g.
< =1> - add a leading 1 to all numbers dialed, vice versa will remove a
1 from the number dialed
h.
| - or
Note: In some cases user wishes to dial strings such as *123 to activate voice mail or
other application provided by service provider. In this case * should be predefined inside
dial plan feature and the Dial Plan will be: { [x*]+ }.
Page 48 of 60
Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indication will be
sent periodically.
Send Anonymous
If this parameter is set to Yes, the From header along with Privacy and
P_Asserted_Identity headers in outgoing INVITE message will be set to anonymous,
blocking Caller ID.
Anonymous Call
Default is No. If set to Yes, incoming calls with anonymous Caller ID will be rejected with
Rejection
Special Feature
Default is Standard. Choose the selection to meet some special requirements from
Softswitch vendors. Example of vendors - CBCOM, RNK.
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. When
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.
Min-SE
The minimum session expiration (in seconds). The default value is 90 seconds.
If selecting Yes the phone will use session timer when it makes outbound calls if
remote party supports session 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
As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or
Refresher
UAS Specify
As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use 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.
Default is No, If set to Yes, device will send an INVITE with audio vocoders upon
Fax
Page 49 of 60
Enable Silence
For fax machines that do not send a Disconnect when fax is done. This option
Enables/Disables the detection of silence in order to know the fax has finished. The
Disconnect
Enable 100rel
If set to Yes, device will include authorization header in the Register request.
Initial REGISTER
Use First Matching
Default is No. If set to Yes, device will include only the first match vocoder in its 200OK
Vocoder in 200OK
response, otherwise it will include all match vocoders in same order received in INVITE.
SDP
Preferred Vocoder
The GXW400X supports up to 5 different Vocoder types including G.711 A-/U-law, G.726
(Supports bit rates 16, 24, 32 and 40), G.723.1, G.729A/B/E and iLBC. The user can
configure Vocoders in a preference list that will be included with the same preference
order in SDP message. The first Vocoder is entered by choosing the appropriate option
in Choice 1. The last Vocoder is entered by choosing the appropriate option in Choice
8.
G723 Rate
Defines the encoding rate for G.723 vocoder. By default, 6.3kbps rate is chosen.
Default value is 97. Defines payload type for iLBC. The valid range is between 96 and
127.
AAL2-G726-16
Payload type
AAL2-G726-24
Payload type
AAL2-G726-32
Payload type
AAL2-G726-40
Payload type
G729E payload type
VAD
Default is No. VAD allows detecting the absence of audio and conserve bandwidth by
preventing the transmission of "silent packets" over the network.
Symmetric RTP
Default is No. When set to Yes the device will change the destination to send RTP
packets to the source IP address and port of the inbound RTP packet last received by
the device.
Page 50 of 60
Fax Mode
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)
Default is Callee. This decides whether Caller or Callee sends out the re-INVITE for
Mode
High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the
high requirement
SRTP Mode
Default is Disabled. Other options are Enabled but not forced, and Enabled and
forced.
SDES: http://www.rfc-base.org/rfc-4568.html
SRTP: http://www.rfc-base.org/rfc-3711.html
Crypto Life Time
Default is Enabled. Allows user to enable or disable Crypto life time when using SRTP.
SLIC Setting
Caller ID Scheme
Select the value according to the local Telco standard where the GXW400X is installed.
Please refer to the pull down list to select.
Polarity Reversal
Default is No. If set to Yes, polarity will be reversed upon call establishment and
termination.
Loop Current
Set to Yes if the traditional PBX you are using with GXW400X uses this method for
Disconnect
Play busy/reorder tone Default is No. If set to Yes, it will play busy/reorder tone before loop current disconnect
before Loop Current
Disconnect
Loop Current
Disconnect duration
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Default is Yes. If set to No, FLASH button could only be used for terminating calls.
Page 51 of 60
Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent
unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time
value.
On Hook Timing
Gain
phone), TX is for transmission volume (Analog phone to FXS). Default values are 0dB
for both parameters. Loudest volume: +6dB Lowest volume: -6dB.
Ring Tones
Grandstream recommends reboot or power cycle the IP phone after saving changes.
When GXW400X boots up, it will send TFTP or HTTP/HTTPS requests to download configuration files,
cfg000b82xxxxxx and cfg00082xxxxxx.xml, where 000b82xxxxxx is the LAN MAC address of the
GXW400X. If the download of cfgxxxxxxxxxxxx.xml is not successful, the provision program will issue
request a generic configuration file cfg.xml. Configuration file name should be in lower case letters.
The configuration data can be downloaded via TFTP or HTTP/HTTPS from the central server. A service
provider or an enterprise with large deployment of GXW400X can easily manage the configuration and
service provisioning of individual devices remotely from a central server.
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friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with
each individual Grandstream device for firmware upgrade, remote reboot, etc.
Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection
or with certain special provisioning settings.
Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS
provision the devices with redirection settings so that they will be redirected to customers TFTP or
HTTP/HTTPS server for further provisioning.
Grandstream also provides configuration tools (Windows and Linux/Unix version) to facilitate the task of
generating device configuration files. The Grandstream configuration tools are free to end users. The
configuration
tools
and
configuration
templates
are
available
for
download
from
http://www.grandstream.com/support/tools .
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SOFTWARE UPGRADE
Software upgrade can be done via either TFTP or HTTP/HTTPS. The corresponding configuration
settings are in the ADVANCED SETTINGS configuration page.
NOTES:
Firmware upgrade server in IP address format can be configured via IVR. Please refer to the
CONFIGURATION GUIDE section for instructions. If the server is in FQDN format, it must be set
via the web configuration interface.
Grandstream recommends end-user use the Grandstream HTTP server. Its address can be found
at http://www.grandstream.com/support/firmware . Currently the HTTP firmware server URL is
firmware.grandstream.com. For large companies, we recommend to maintain their own TFTP/
HTTP/HTTPS server for upgrade and provisioning procedures.
Once a Firmware Server Path is set, user needs to update the settings and reboot the device. If
the configured firmware server is found and a new code image is available, the GXW will attempt
to retrieve the new image files by downloading them into the GXW400x s SRAM. During this
stage, the GXWs LEDs will blink until the checking/downloading process is completed. Upon
verification of checksum, the new code image will then be saved into the Flash. If
TFTP/HTTP/HTTPS fails for any reason (e.g. TFTP/HTTP/HTTPS server is not responding, there
are no code image files available for upgrade, or checksum test fails, etc), the GXW will stop the
TFTP/HTTP/HTTPS process and simply boot using the existing code image in the flash.
Firmware upgrade may take as long as 15 to 30 minutes over Internet, or just 5 minutes if it is
performed on a LAN.
environment if possible. For users who do not have a local firmware upgrade server,
Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware upgrade.
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Oversea users are strongly recommended to download the binary files and upgrade firmware
locally in a controlled LAN environment.
Alternatively, user can download a free TFTP or HTTP server and conduct local firmware upgrade.
A free windows version TFTP server is available for download from
http://www.solarwinds.com/register/?Program=52&c=70150000000CcH2.
A configuration parameter is associated with each particular field in the web configuration page.
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with Admin Password in the ADVANCED SETTINGS page. For a
detailed parameter list, please refer to the corresponding firmware release configuration template.
When a Grandstream device boots up or reboots, it will issue a request for a configuration file
cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the MAC address of the device, i.e., cfg000b820102ab. In
addition, device will also requests a XML configuration file cfgxxxxxxxxxxxx.xml. If the download of
cfgxxxxxxxxxxxx.xml is not successful, the provision program will issue a request for a generic
configuration file cfg.xml. Configuration file name should be in lower case letters.
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Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and
Postfix. This makes it possible to store ALL of the firmware with different version in one single directory.
Similarly, Config File Prefix and Postfix allows device to download the configuration file with the matching
Prefix and Postfix. Thus multiple configuration files for the same device can be stored in one directory.
In addition, when the field Check New Firmware only when F/W pre/suffix changes is selected, the
device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix.
Automatic Upgrade:
No
Yes, every
10080
minutes (60-5256000).
(0-23).
(0-6).
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FACTORY RESET
There are two (2) methods for resetting your unit:
RESET BUTTON
Reset default factory settings following these four (4) steps:
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. Insert a pin in this hole, and press for about 7 seconds.
4. Take out the pin. All unit settings are restored to factory settings.
IVR COMMAND
Reset default factory settings using the IVR Prompt (Table 5):
1. Locate the MAC address of the device. It is the 12 digit HEX number on the bottom of the
unit.
2. Key in the MAC address. Use the following mapping:
0-9: 0-9
FIRMWARE VERSION 1.0.14.100
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A:
B:
222
C:
2222
D:
E:
333
F:
3333
NOTE:
1. Factory Reset will be disabled if the Lock keypad update is set to Yes.
2. Please be aware by default the GXW400X WAN side HTTP access is disabled. After a factory reset,
the devices web configuration page can be accessed only from its LAN port.
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