Grandstream Networks, Inc.
Analog IP Gateway GXW40XX
4, 8 or 24 FXS Ports
GXW40XX User Manual
Firmware Version 1.0.4.2
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TABLE OF FIGURES
GXW40XX User Manual
Figure 1: Diagram of GXW4004-8 and GXW4024 Back Panel...................................................................................5
Figure 2: Diagram of GXW4004-8 and GXW4024 Display Panel ..............................................................................6
TABLE OF TABLES
GXW40XX User Manual
Table 1: Definitions of the GXW Connectors ..............................................................................................................6
Table 2: Definitions of the GXW Display Panel ..........................................................................................................7
Table 3: GXW40XX Software Features.......................................................................................................................8
Table 4: Hardware Specification of GXW40XX series gateways..............................................................................10
Table 5: Definitions of the GXW Voice Prompts.......................................................................................................11
Table 6: Call Features Table (Star Code) ...................................................................................................................17
Table 7: Basic Settings ...............................................................................................................................................20
Table 8: Status Page ...................................................................................................................................................22
Table 9: Advanced Settings.........................................................................................................................................23
Table 10: FXS Ports Settings......................................................................................................................................26
Table 11: Profile settings............................................................................................................................................27
CONFIGURATION GUI INTERFACE EXAMPLES
GXW40XX User Manual
1. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
3. SCREENSHOT OF FXS PORTS CONFIGURATION LOGIN PAGE
4. SCREENSHOT OF PROFILE 1 CONFIGURATION PAGE
5. SCREENSHOT OF PROFILE 2 CONFIGURATION PAGE
6. SCREENSHOT OF STATUS PAGE
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WELCOME
Thank you for purchasing the Grandstream GXW40XX Analog FXS IP Gateway. The GXW40XX offers
an easy to manage, easy to configure IP communications solution for any business with virtual and/or
branch locations. The GXW40XX 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.
Gateway GXW40XX Overview
The new GXW40XX 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 GXW40XX features 4,8 or 24-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 port (except
GXW4024) 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/u-law), G.723.1, G.726(16/24/32/48 bit rates), G.729A/B/E and iLBC.
Safety Compliance
The GXW40XX 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 GXW40XX 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
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 contains links to Grandstream GUI Interfaces. Please download these examples
reference.
•
This document is subject to change without notice. The latest electronic version of this user manual
is located at
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.
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CONNECT YOUR GXW40XX GATEWAY
Connecting the GXW40XX gateway is easy. Before you begin, please verify the contents of the
GXW40XX package.
EQUIPMENT PACKAGING
Unpack and check all accessories. Equipment includes:
•
•
•
one device unit
one RJ45 Ethernet cable
one 12V universal power adapter
CONNECT THE GXW40XX
Follow these four (4) steps to connect your GXW40XX gateway to the Internet and access the unit’s
configuration pages.
1. Connect standard touch-tone analog phones to the FXS ports.
2. Insert an RJ45 Ethernet cable into the WAN port of GXW400X and connect the other end to an
uplink port (a router or a modem, etc.) Note: If the gateway is used in router mode, connect a PC
to the LAN port of GXW400X for initial configuration. In case of GXW4024, there is only one
Ethernet port.
3. Plug the power adapter into the GXW40XX gateway into a power outlet.
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:
FIGURE 1: DIAGRAM OF GXW4004-8 AND GXW4024 BACK PANEL
GXW 400x
PSTN
Line
Power
Supply
FXS
Ports
Reset
Console
Connection
(PC
LAN/WAN
RJ-45 Ethernet
Ports
connection)
GXW 4024
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50 PIN Telco
Connector
LAN Port
FXS
Ports
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS
LAN
Connect the LAN port with an Ethernet cable to your PC.
Connect to the internal LAN network or router.
Factory Reset button. Press for 7 seconds to reset factory default settings.
Power adapter connection
WAN
RESET
DC 12V
CONSOLE
LINE
Console connection for a PC.
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 – FXS24
FXS port to be connected to analog phones / fax machines.
Once the GXW40xx is turned on and configured, the front display panel indicates the status of the unit.
FIGURE 2: DIAGRAM OF GXW4004-8 AND GXW4024 DISPLAY PANEL
GXW4008
Display
LEDs
FXS port
status
GXW4024
Display LEDs
FXS port status
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL
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Power LED
Ready LED
Console
Indicates Power. Remains ON when Power is connected and turned ON.
Remains ON after boot-up.
Turns ON when Console port is being used.
Indicates LAN (or WAN) port activity
LAN LED
WAN LED
LED 1–24
Indicates PC (or LAN) port activity
Indicate status of the respective FXS Ports on the back panel
Busy - ON (Solid Green)
Available - OFF
Slow blinking FXS LEDs indicates Voice Mail for that port.
NOTE:
All LED lights display green when ON.
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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GXW40XX 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 auto-
configurable, remotely manageable and scalable. The GXW4000 series gateways come in three models
- the GXW-4004,GXW-4008 and GXW4024, each offering superb voice quality, traditional telephony
functionality, easy deployment, and 4,8 and 24 FXS ports respectively. Each model features flexible
dialing plans, integrated call routing to support a pure IP network call and an external power supply.
SOFTWARE FEATURES OVERVIEW
•
•
•
•
4, 8 or 24 FXS ports
Two RJ-45 ports (switched or routed) (one RJ-45 port in GXW4024)
Multiple SIP accounts & profiles (4,8 or 24 accounts / choice of 3 profiles per account)
Supports Voice Codecs:
G711(a/µ, Annex I & II), G723.1A, G726 (ADPCM with 16/24/32/40 bit rates), G729 A/B/E, iLBC
T.38 Fax
•
•
•
•
Comprehensive Dial Plan support for Outgoing calls.
G.168 Echo Cancellation
Voice Activation Detection (VAD), Comfort Noise Generation (CNG), and Packet Loss
Concealment (PLC)
•
Supports PSTN/PBX analog telephone sets or analog trunks
TABLE 3: GXW40XX SOFTWARE FEATURES
GXW4004
GXW4008
GXW4024
Telephone Interfaces
SIP Provisioning
4 FXS ports
8 FXS ports
24 FXS ports
4 SIP accounts, 2 profiles
8 SIP accounts, 2 profiles
24 SIP accounts, 3 profiles
Network Interface
Two (2) 10M/100 Mbps, Two (2) 10M/100 Mbps, RJ- One 10M/100 Mbps LAN,
RJ-45 45 RJ-45
Voice over Packet
Capabilities
Voice Activity Detection Voice Activity Detection Voice Activity Detection
(VAD) with CNG (comfort (VAD) with CNG (comfort (VAD) with CNG (comfort
noise generation) and PLC noise generation) and PLC noise generation) and PLC
(packet loss concealment), (packet loss concealment), (packet loss concealment),
AEC with NLP Packetized AEC with NLP Packetized AEC with NLP Packetized
Voice
(supports RTP/RTCP and (supports RTP/RTCP and (supports
AAL2 protocol), G.168 AAL2 protocol), G.168 G.168 compliant Echo
Echo compliant Echo Cancellation, Adaptive
Protocol
Unit Voice
Protocol
Unit Voice
Protocol
Unit
RTP/RTCP)
compliant
Cancellation,
Dynamic Cancellation, Dynamic Jitter Jitter Buffer, Fax/Modem
Jitter
Buffer,
Modem Buffer, Modem detection & detection & auto-switch to
detection & auto-switch to auto-switch to G.711
G.711
G.711
PSTN Fail-over
PSTN failover port on PSTN failover port on power N/A
power failure failure
G.711 + Annex I (PLC), G.711 + Annex I (PLC), G.711 + Annex I (PLC),
Annex
format)
decoder,
II
encoder
G.723.1A, G.723.1A,
with with 16/24/32/40 bit rates), G.726(ADPCM
(VAD/CNG Annex II (VAD/CNG format) Annex
and encoder and decoder, format)
G.726(ADPCM decoder,
II
(VAD/CNG
Voice Compression
encoder
and
G.723.1,
with
G.726(ADPCM
16/24/32/40 bit rates), G.729A/B/E, iLBC
16/24/32/40 bit rates),
G.729A/B/E, iLBC
G.729A/B/E, iLBC
G.726 provides proprietary
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G.726 provides proprietary VAD, CNG, and signal G.726 provides proprietary
VAD, CNG, and signal power estimation, Voice VAD, CNG, and signal
power estimation, Voice Play Out unit (reordering, power estimation, Voice
Play Out unit (reordering, fixed and adaptive jitter Play Out unit (reordering,
fixed and adaptive jitter buffer,
clock fixed and adaptive jitter
buffer,
synchronization),
clock synchronization),
AGC (automatic gain control),
AGC buffer,
synchronization),
clock
AGC
(automatic gain control),
(automatic gain control),
Status
output,
Decoder
Status output, Decoder controlling via voice packet Status output, Decoder
controlling via voice packet header
header
controlling via voice packet
header
DHCP Server/Client
Fax over IP
Yes, NAT Router or Yes,
NAT
Router
or Yes, DHCP Client only
Switched Mode
Switched Mode
T.38 compliant Group 3 T.38 compliant Group 3 Fax T.38 compliant Group 3
Fax Relay up to 14.4kpbs Relay up to 14.4kpbs and Fax Relay up to 14.4kpbs
and auto-switch to G.711 auto-switch to G.711 for Fax and auto-switch to G.711
for Fax Pass-through,
Pass-through,
for Fax Pass-through,
Fax Datapump V.17, V.19, Fax Datapump V.17, V.19, Fax Datapump V.17, V.19,
V.27ter, V.29 for T.38 fax V.27ter, V.29 for T.38 fax V.27ter, V.29 for T.38 fax
relay
relay
relay
QoS
Diffserv, TOS, 802.1 P/Q Diffserv, TOS, 802.1 P/Q Diffserv, ToS, 802.1 P/Q
VLAN tagging
RTP/RTCP
VLAN tagging
RTP/RTCP
VLAN tagging
RTP/RTCP
Transport Protocol
DTMF Method
In-audio, RFC2833, and/or In-audio, RFC2833, and/or In-Audio, RFC2833, and/or
SIP Info
SIP Info
SIP Info
IP Signaling
Provisioning
Control
SIP (RFC 3261)
TFTP, HTTP, HTTPS
TLS/SIPS
SIP (RFC 3261)
TFTP, HTTP, HTTPS
TLS/SIPS
SIP (RFC 3261)
TFTP, HTTP, HTTPS
TLS/SIPS
Management
Syslog support, HTTP and Syslog support, HTTP and
Syslog support, HTTP and
Telnet access
Telnet access
Telnet access
Dial Plan
Yes
Yes
Yes
Yes
Yes
N/A
UPnP Support
Caller ID
Bellcore Type 1 & 2, ETSI, Bellcore Type 1 & 2, ETSI, Bellcore Type 1 & 2, ETSI,
BT, NTT, and DTMF-based BT, NTT, and DTMF-based BT, NTT, and DTMF-based
CID
CID
CID
Polarity Reversal
Wink
/
Yes
Yes
Yes
EMC
EN55022/EN55024
FCC part15 Class B
and EN55022/EN55024
FCC part15 Class B
and EN55022/EN55024
FCC part15 Class B
and
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HARDWARE SPECIFICATION
The hardware specifications of the GXW FXS series are detailed in Table 4.
TABLE 4: HARDWARE SPECIFICATION OF GXW40XX SERIES GATEWAYS
GXW4004
GXW4008
GXW4024
Ports
4 FXS Ports
8 FXS Ports
24 FXS Ports
Network interface
1 RJ45 for LAN, 10/100
Base-TX, Full Duplex
1 RJ45 for LAN, 10/100
Base-TX, Full Duplex
1 RJ45 for LAN, 10/100 Base-
TX, Full Duplex
1 RJ45 for
1 RJ45 for
WAN,10/100Base-TX,
Full Duplex
WAN,10/100Base-TX, Full
Duplex
PSTN Port
Console
PSTN fail-over port
DB9 Serial port
PSTN fail-over port
DB9 Serial port
n/a
n/a
Universal
Switching
Input: 100-240V AC, Input:
100-240V
AC, Input: 100-240V AC, 50/60Hz
50/60Hz, 0.3A Max
Output: 12V DC, 1A
UL certified
50/60Hz, 0.3A Max
Output: 12V DC, 1.5A
UL certified
Output: 12V DC, 3.0A,
certified
UL
Power Adaptor
Short Haul Loop
2REN: Up to 1km on 24 2REN: Up to 1km on 24 2REN: Up to 1km on 24 AWG
AWG wire
AWG wire
wire
Mounting
Wall mount, Desktop
Wall mount, Desktop
Rack, Wall mount
Dimension
225mm (L) x 135mm (W) 225mm (L) x 135mm (W) x 445mm (L) x 269mm (W) x
x 35mm (H)
35mm (H)
44.5mm (H), or 1U
Weight
0.29 lbs (3.5 oz)
32~104°F / 0~40°C
0.29 lbs (3.5 oz)
32~104°F / 0~40°C
6.3 lbs (2.8 kg)
Temperature
Humidity
32~104°F / 0~40°C
10%
-
90%
(non- 10%
-
90%
(non- 10% - 90% (non-condensing)
condensing)
condensing)
Safety
UL
UL
UL
Compliance
FCC, CE
FCC, CE
FCC, CE, C-Tick
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BASIC OPERATIONS
UNDERSTANDING GXW VOICE PROMPTS
GXW40XX 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.
TABLE 5: DEFINITIONS OF THE GXW VOICE PROMPTS
Menu
Voice Prompt
User’s Options
Main Menu
“Enter a Menu Option”
Enter “*” for the next menu option
Enter “#” to return to the main menu
Enter 01 – 05, 07,10 - 17, 47, 86 or 99 Menu option
01
“DHCP Mode”, “PPPoE
Enter ‘9’ to toggle the selection
Mode ”or “Static IP Mode”
If user selects “Static IP Mode”, user need configure all the IP
address information through menu 02 to 05. If user selects
“Dynamic IP Mode”, the device will retrieve all IP address
information from DHCP server automatically when user reboots
the device.
02
“IP Address “ + IP address
The current WAN IP address is announced
Enter 12-digit new IP address if in Static IP Mode.
03
04
05
07
“Subnet “ + IP address
“Gateway “ + IP address
Same as Menu option 02
Same as Menu option 02
“DNS Server “ + IP address Same as Menu option 02
Preferred Vocoder
Enter “9” to go to the next selection in the list:
•
•
•
•
•
•
PCM U
PCM A
iLBC
G-726
G-723
G-729
10
12
13
“MAC Address”
Announces the Mac address of the unit.
Enter “9” to toggle between enabled and disabled
WAN Port Web Access
Firmware Server IP
Address
Announces current Firmware Server IP address. Enter 12 digit
new IP address.
14
15
Configuration Server IP
Address
Announces current Config Server Path IP address. Enter 12 digit
new IP address.
Upgrade Protocol
Upgrade protocol for firmware and configuration update. Enter
“9” to toggle between TFTP and HTTP
16
17
Firmware Version
Firmware Upgrade
Firmware version information.
Firmware upgrade mode. Enter “9” to rotate among the following
three options:
1. always check
2. check when pre/suffix changes
3. never upgrade
47
“Direct IP Calling”
Enter the target IP address to make a direct IP call, after dial
tone. (See “Make a Direct IP Call”.)
71-78
Phone calls between
different ports of the same
GW 400x
Applies to GXW4004/GXW4008. User can make internal calls
between different ports inside the same GW even without being
registered to SIP server. By dialing *** and 7x user can reach
colleague connected to port “x”.
701-724
Phone calls between
different ports of the same
GW4024
Applies to GXW4024. User can make internal calls between
different ports inside the same GW even without being registered
to SIP server. By dialing *** and 7xx user can reach colleague
connected to port “xx”.
86
99
Voice Mail
“RESET”
Number of voice mails
Enter “9” to reboot the device; or
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Enter MAC address to restore factory default setting
(See Restore Factory Default Setting section)
“Invalid Entry”
Automatically returns to Main Menu
Five Success Tips when using the Voice Prompt
1. “*” shifts down to the next menu option
2. “#” returns to the main menu
3. “9” functions as the ENTER key in many cases to confirm an option
4. All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP
address. For IP address, add 0 before the digits if the digits are less than 3 (i.e. - 192.168.0.26
should be key in like 192168000026. No decimal is needed).
5. Key entry can not be deleted but the phone may prompt error once it is detected
PLACING A PHONE CALL
PHONE OR EXTENSION NUMBERS
1. Dial the number directly and wait for 4 seconds (To change the default value, modify the following
setting - “No Key Entry Timeout”)
or
2. Dial the number directly and press # (Use # as dial key” must be configured in web configuration).
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.
Elements necessary to completing a Direct IP Call:
1. Both GXW40XX and other VoIP Device, have public IP addresses, or
2. Both GXW40XX and other VoIP Device are on the same LAN using private IP addresses, or
3. Both GXW40XX and other VoIP Device can be connected through a router using public or private
IP addresses (with necessary port forwarding or DMZ).
GXW40XX supports two ways to make Direct IP Calling:
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.
Using Star Code
1. Pick up the analog phone then dial “*47”
2. Enter the target IP address using same format as above.
Note: NO dial tone will be played between step 1 and 2.
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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 C’s number, and then # (or wait for 4 seconds).
5. Caller A will hear the confirm tone. Then, A can hang up.
NOTE: “Enable Call Feature” must be set to “Yes” in web configuration page.
Caller A can place a call on hold and wait for one of three situations:
1. A quick confirmation tone (similar to call waiting tone) followed by a dial tone. This indicates the
transfer is successful (transferee has received a 200 OK from transfer target). At this point,
Caller A can either hang up or make another call.
2. A quick busy tone followed by a restored call (on supported platforms only). This means the
transferee has received a 4xx response for the INVITE and we will try to recover the call. The
busy tone is just to indicate to the transferor that the transfer has failed.
3. Continuous busy tone. The phone has timed out. Note: continuous busy tone does not indicate
the transfer has been successful, nor does it indicate the transfer has failed. It often means there
was a failure to receive second NOTIFY – check firmware for most recent release.
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.
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2. Caller A then dials Caller C’s 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 GXW40XX 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
The GXW40XX supports Bellcore style 3-way Conference.
Instructions for 3-way conference:
Assuming that call party A and B are in conversation. A (GXW40XX) 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 C’s 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 round robin manner among the ports
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 a 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.
Example Configuration of a typical Hunting Group:
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.
Example configuration of a multiple Hunting Group:
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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"
…
FXS Port #24: SIP UserID and Authenticate ID left blank, Hunting Group set to "2"
Hunting Group 1 contains ports 1, 2, 3. Hunting Group 4 contains ports 4, 5, 6.
Hunting Group 7 contains ports 7, 8.
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
8 is required to indicate that the SIP account tied to port market 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.
Example of not sequential configuration of a multiple Hunting Group:
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"
…
FXS Port #24: 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.
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.
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On the GXW4024 inter-port calling is achieved by dialing *** and 7 plus two extra digits corresponding to
the port number. For example the user connected to port 1 can be reached by dialing *** and 701; the
user connected to port 24 can be reached by dialing *** 724.
PSTN PASS THROUGH/LIFE LINE
This feature is only applicable to GXW4004 and GXW4008 models. 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.
SENDING AND RECEIVING FAX
GXW40XX 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.
SUPPORT RADIUS PROTOCOL
GXW40XX 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.
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CALL FEATURES
GXW40XX supports the traditional telephony features available in a PBX as well as additional advanced
telephony features.
TABLE 6: CALL FEATURES TABLE (STAR CODE)
Key
*02
Call Features
Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729
(G729), *0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-
r40), *027201 (iLBC)
*03
*16
*17
*30
*31
*67
Disable LEC (pe call) Dial “*03” + ” number ”. No dial tone is played in the middle.
Enable SRTP
Disable SRTP
Block CallerID (for all-config change)
Send CallerID (for all-config change)
Block CallerID (per call)
*82
*47
Send CallerID (per call)
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
Disable Call Waiting (for all-config change)
Enable Call Waiting (for all-config change)
Call Return Service: Dial *69 and the phone will dial the last incoming phone number
received.
*70
*71
*72
Disable Call Waiting (Per Call)
Enable Call Waiting (Per Call)
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
*74
Cancel Unconditional Call Forward: Dial “*73” and wait for dial tone, then hang up.
Enable Paging Call: Dial “*74” and then the destination phone number you want to activate in
Paging mode.
*78
*79
*87
*90
Enable Do Not Disturb (DND): When enabled all incoming calls will be rejected.
Disable Do Not Disturb (DND): When disabled, incoming calls will be accepted.
Blind Transfer
Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial
tone then hang up.
*91
*92
Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.
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.
Pressing pound sign will serve as Re-Dial key.
#
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CONFIGURATION GUIDE
CONFIGURING GXW40XX VIA VOICE PROMPT
DHCP MODE
Select voice menu option 01 to enable GXW40XX to use DHCP.
STATIC IP MODE
Select voice menu option 01 to enable GXW40XX to use STATIC IP mode, then use option 02, 03, 04, 05
to set up IP address, Subnet Mask, Gateway and DNS server respectively.
FIRMWARE SERVER IP ADDRESS
Select voice menu option 13 to configure the IP address of the firmware server.
CONFIGURATION SERVER IP ADDRESS
Select voice menu option 14 to configure the IP address of the configuration server.
UPGRADE PROTOCOL
Select voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose
between TFTP and HTTP.
FIRMWARE UPGRADE MODE
Select voice menu option 17 to choose firmware upgrade mode among the following three options:
1) always check, 2) check when pre/suffix changes, and 3) never upgrade
WAN PORT WEB ACCESS (Note: This option is not present on GXW4024 gateway)
Select voice menu option 12 to enable WAN Port Wed Access of the device configuration pages.
CONFIGURING GXW40XX WITH WEB BROWSER
The GXW40XX series gateway has an embedded Web server that allows users to configure the
GXW40XX through a web browser.
ACCESS THE WEB CONFIGURATION MENU
The GXW40XX HTML configuration menu can be accessed via LAN or WAN port:
From the LAN port (For GXW4004/8 only)::
1. Directly connect a computer to the LAN port.
2. In Windows, open a ‘Command Prompt’ window
3. Type in “ipconfig /release”, the IP address etc. becomes 0.
4. Type in “ipconfig /renew”, the computer gets an IP address in 192.168.2.x segment by default
login page of the device.
From the WAN port (For GXW4004/8 only):
The WAN port HTML configuration option is disabled by default from factory. To access the HTML
configuration menu from the WAN port:
1. Follow table 4 to find the WAN side IP address.
Address (the GXW40XX IP-Address is the WAN IP address for the GXW40XX).
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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
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: Web pages allowed:
End User Level
Administrator Level
123
Only Status and Basic Settings
Browse all pages
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:
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
DTMF via RTP (RFC2833)
DTMF via SIP INFO
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.
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Preferred VOCODER (Codec)
The GXW40XX supports a broad range of voice codecs. Under Profile web pages, choose your preferred
order of different codecs:
•
•
•
•
•
•
PCMU/A (or G711µ/a)
G729 A/B/E
G723
G726 (16/24/32/40)
iLBC
AAL2 (all G726)
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.
Functions available for the end-user are:
•
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.
Additional functions available to administrators are:
•
ADVANCED SETTINGS: To set advanced network settings, codec settings and XML
configuration settings.
•
•
PROFILE X: To configure each of the SIP accounts.
FXS PORTS: To configure each of the FXS ports and Hunting Groups etc.
TABLE 7: BASIC SETTINGS
End User Password
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.
Default is Yes.
Telnet Server
IP Address
There are two modes to operate the GXW40XX :
DHCP mode: The GXW40XX acquires its IP address from the first DHCP server it
discovers from the LAN It is on.
PPoE: Set the PPPoE account settings. The GXW40XX 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.
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DHCP hostname
DHCP domain
This option specifies the name of the client. This field is optional but may be
required by some Internet Service Providers. Default is blank.
This option specifies the domain name that client should use when resolving
hostnames via the Domain Name System. Default is blank.
DHCP vendor class ID
PPPoE account ID
Used by clients and servers to exchange vendor-specific information. Default is
HT500.
PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point
Protocol over Ethernet) connection.
PPPoE password
PPPoE account password.
PPPoE Service Name
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.
Self Defined Time Zone
The syntax is std offset dst [offset],start[/time],end[/time]
Default is set to : MTZ+6MDT+5,M3.2.0,M11.1.0
MTZ+6MDT+5,
It indicates a time zone with 6 hours offset with 1 hour ahead which is the US
central time.
It is positive (+) if the local time zone is west of the Prime Meridian and negative
(-) if it is east.
Prime Meridian (International or Greenwich Meridian)
M3.2.0,M11.1.0
The 1st number indicates Month: 1,2,3,..,12 (for Jan, Feb, .., Dec)
The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd
Tuesday etc)
The 3rd number indicates Weekday: 0,1, 2, ..,6(for Sun, Mon, Tue, .., Sat)
Therefore, this example is the DST which starts from the second Sunday of
March to the 1st Sunday of November.
Language
Default is English. Languages supported with voice prompt.
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.
Note: This option is not present on GXW4024
NAT maximum ports
NAT TCP timeout
Maximum NAT ports, typically one connection per port.
Note: This option is not present on GXW4024
NAT TCP timeout in seconds
Note: This option is not present on GXW4024
NAT UDP timeout
Uplink Bandwidth
Downlink Bandwidth
Enable UPnP support
NAT UDP timeout in seconds.
Note: This option is not present on GXW4024
ISP specified WAN uplink bandwidth
Note: This option is not present on GXW4024
ISP specified WAN downlink bandwidth
Note: This option is not present on GXW4024
If set to “Yes”, the GXW40XX would act as an UPnP gateway for your UPnP enable
applications. UPnP - Universal Plug n Play
Note: This option is not present on GXW4024
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Reply to ICMP on WAN port If set to “Yes”, the GXW40XX will respond to the PING. Default is No.
Note: This option is not present on GXW4024
WAN side HTTP/Telnet
Access
If set to “Yes”, user can access the configuration page through the WAN port,
instead of through the “PC” port. WARNING: this configuration is less secure than
default option. Default is No.
Note: This option is not present on GXW4024
Cloned WAN MAC Address
Enable LAN DHCP
This allows you to change/set the MAC address on the WAN interface.
Note: This option is not present on GXW4024
Default is Yes.
Note: This option is not present on GXW4024
LAN Subnet Mask
Sets the LAN subnet mask. Default value is 255.255.255.0
Note: This option is not present on GXW4024
LAN DHCP Base IP
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
Note: This option is not present on GXW4024
LAN DHCP Start IP
LAN DHCP End IP
DHCP IP Lease Time
Default is 100
Note: This option is not present on GXW4024
Default is 199
Note: This option is not present on GXW4024
Value is set in units of hours. Default value is 120 hrs (5 Days.) The time IP
address is assigned to the LAN clients.
Note: This option is not present on GXW4024
DMZ IP
Forward all WAN IP traffic to a specific IP address if no matching port is used by
GXW40XX or defined in port forwarding.
Note: This option is not present on GXW4024
Port Forwarding
Forwards a matching (TCP/UDP) port to a specific LAN IP address with a specific
(TCP/UDP) port.
Note: This option is not present on GXW4024
In addition to the Basic Settings configuration page, end users also has 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
Shows WAN IP address of GXW40XX
Contains the product model info.
Software Version
Program: This is the main software release. Boot and Loader are not changed often.
Shows system uptime since the last reboot.
System Up Time
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PPPoE Link Up
NAT
Shows whether the PPPoE connection is running if connected to DSL modem.
Shows type of NAT the GXW40XX is connected to via its WAN port. It is based on
STUN protocol.
Port Status
Displays relevant information regarding the individual FXS ports. Example:
Port
Hook
Registration
DND
Forward
Busy
Delayed
Forward Forward
FXS1
FXS2
FXS3
On Hook
Off Hook
On Hook
Registered
Registered
Not
No
No
No
613
614
Registered
Registered
FXS24
On Hook
Yes
615
** FXS port 24 user has set Do Not Disturb.
FXS port 1 user has set his calls to be forwarded unconditionally to ext 613
FXS port 2 user has set his calls to be forwarded to 614 when his phone is busy.
FXS port 3 user is not registered with his SIP Server.
Advanced User configuration includes not only the end user configuration, but also advanced
configurations such as:
SIP configuration, Codec selection, NAT Traversal Setting and other
miscellaneous configuration.
Table 9: ADVANCED SETTINGS
Admin Password
Administrator password. Only the administrator can configure the “Advanced
Settings” page. Password field is purposely left blank for security reasons after
clicking update and saved. The maximum password length is 25 characters.
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
STUN Server
Value used for layer 2 VLAN tag. Default setting is blank.
IP address or Domain name of the STUN server.
STUN server assists devices behind a NAT firewall or a router to set up calls via
VoIP.
STUN server is often provided by the VoIP service provider or a public STUN server
can be used. For a sample list of public free STUN servers please refer to:
Keep-alive interval
Use STUN to detect
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.
Firmware Upgrade &
Provisioning
Enables GXW40XX to download firmware or configuration file using either the TFTP
or HTTP/S protocols.
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Via TFTP Server
This is the IP address of the configured TFTP server. If selected and it is non-zero
or not blank, the GXW40XX retrieves the new configuration file or new code image
from the specified TFTP server at boot time. After 5 attempts, the system will
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, GXW40XX will only do HTTP download once at
boot up.
Firmware Server Path
Config Server Path
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
XML Config File Password
The password used for encrypting the XML configuration file using OpenSSL.
This is required for the phone to decrypt the encrypted XML configuration file.
HTTP/HTTPS User Name
HTTP/HTTPS Password
Firmware File Prefix
The user name for the HTTP/HTTPS server.
The password for the HTTP/HTTPS server.
Default is blank. If configured, GXW40XX will request firmware file with the prefix.
This setting is useful for Internet Telephone Service Providers. End users should
keep it blank.
Firmware File Postfix
Config File Prefix
Config File Postfix
Default is blank. This setting is useful for Internet Telephone Service Providers. End
users should keep it blank.
Default is blank. This setting is useful for Internet Telephone Service Providers. End
users should keep it blank.
Default is blank. This setting is useful for Internet Telephone Service Providers. End
users should keep it blank.
Allow DHCP Option 66 to
override server
If set to “Yes”, configuration and upgrade server’s information can be obtained using
DHCP option 66 from DHCP server. This option specifies the URL of the tftp server.
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
Choose “Yes” to enable automatic upgrade and provisioning. When set to No,
GXW40XX will only do upgrade once at boot up.
When “Check every day” or “Check every week” is checked, user can specify “Hour
of the day (0-23)” or “Day of the week (0-6)”. Default time is Monday 1AM.
There are three options to choose from: “Always check for New Firmware at Boot
up”, “Check New Firmware only when F/W pre/suffix changes”, and “Always Skip the
Firmware Check”.
Authenticate Conf File
If set to Yes, configuration file is authenticated before being accepted. This protects
the configuration from unauthorized modifications.
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Firmware Key
SSL Certificate
For firmware encryption. It should be 32-digit in Hexadecimal Representation. End
user should keep it blank.
The GXW40XX 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.
SSL Private Key
You may also customize the SSL Private Key. The user specified SSL private key
used for SIP over TLS is in X.509 format.
SSL Private Key Password
ACS URL
Enter SSL Private Key password here.
User specify the Auto Configuration Server’s URL (TR-069 protocol)
User specify the ACS Username
ACS Username
ACS Password
User specify the ACS password
Periodic Inform Enable
Periodic Inform Interval
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
Set a user name for the ACS to connect to this device
Connection Request
Username
Connection Request
Password
Set a password for the ACS to connect to this device
System Ring Cadence
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)
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. 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:
Lock Keypad Update
Disable Voice Prompt
If set to “Yes”, the configuration update via keypad is disabled.
Disables the voice prompt configuration. Default is “No”. If set to “Yes” accessing
integrated voice menu will be impossible.
Disable Direct IP Call
Disables the Direct IP Call function. Default is “No”. If set to “Yes” direct IP-to-IP
calling will not be supported.
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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. If set to “Always Connected” the PSTN line will be always connected to
the phone connected to FXS port. VoIP calls will not be allowed in this
configuration.
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.
Failover to FXO Gateway
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 GXW40XX 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
NTP server
IP Address or URI of the FXO gateway.
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
Syslog Server
Syslog Level
The IP address or URL of System log server. The server collects system log
information from the device.
Select the GXW40XX 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
Example:
May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000]
Ethernet link is up
Primary RADIUS Server
Primary RADIUS Auth Port
Primary RADIUS Acct Port
IP Address or FQDN of the primary RADIUS Server
Primary Radius server authentication port. Default value is 1812
Primary Radius server accounting port. Default value is 1813
Primary Radius Server
Secret
Special secret string should be preconfigured according to RADIUS Server
configuration
Secondary RADIUS Server
IP Address or FQDN of the secondary RADIUS Server
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Secondary RADIUS Auth
Port
Secondary Radius server authentication port. Default value is 1812
Secondary Radius server accounting port. Default value is 1813
Secondary RADIUS Acct
Port
Secondary Radius Server
Secret
Special secret string should be preconfigured according to RADIUS Server
configuration
RADIUS Timeout
RADIUS Retry
Default value is 2 seconds. The time between retries the GXW will send “Access-
Request” message to RADIUS server in purpose to authenticate it.
Default value is 3 times. Number of times the device will try to authenticate itself with
preconfigured RADIUS server during initialization process.
Download Device
Configuration
This setting allows user to download a text file containing all the P values of each
setting as configured on the unit.
TABLE 10: FXS PORTS SETTINGS
FXS Port
FXS Port Number
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 subscriber’s Authenticate ID used for authentication. Can be
identical to or different from SIP User ID.
SIP service subscriber’s account password for GXW40XX to register to (SIP) servers
of ITSP.
Name
Any name to identify this specific user.
Select the corresponding Profile ID (1/2)
Profile ID
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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 round robin
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.
Example configuration of a multiple Hunting group:
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.
Port#
FXS Port Number
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.
Map to FXO Port#
This is used only when peering with a Grandstream GXW410x. Default is 1,
Supported values are 1-8, meaning line 1 to line 8 of the GXW410x device where the
port will be mapped to.
Map to FXO Gateway IP
and Port
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.
Sip port that will be annexed to the IP address above.
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TABLE 11: PROFILE SETTINGS
Profile Active
When set to Yes the SIP Profile is activated.
Primary SIP Server
Failover SIP Server
Primary SIP Server’s IP address or Domain name provided by VoIP service provider.
Failover SIP Server’s IP address or Domain name provided by VoIP Service provider.
This server will be used if the Primary SIP server becomes unavailable.
Prefer Primary SIP
Server
Default is no. If set to yes it will register to Primary Server if registration with Failover
server expires
Outbound Proxy
IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border
Controller. Used by GXW40XX 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.
SIP transport
NAT Traversal
User can select UDP or TCP or TLS. Please make sure you’re SIP Server or network
environment supports SIP over the selected transport method. Default is UDP.
This parameter defines whether the GXW40XX NAT traversal mechanism is activated or
not. If activated (by choosing “Yes”) and a STUN server is also specified, then the
GXW40XX 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 GXW40XX
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 GXW40XX
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.
DNS Mode
One from the 3 modes 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)
One mode can be chosen for the client to look up server.
The default value is “A Record”
User ID is Phone
Number
If the GXW40XX has an assigned PSTN telephone number, this field should be 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 GXW40XX 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 is cleared on reboot.
Outgoing Call without 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 GXW40XX 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).
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Local SIP port
Local RTP Port
Defines the local SIP port the GXW40XX will listen and transmit. The default value for
Profile 1 is 5060 and 6060 for Profile 2.
Defines the local RTP-RTCP port pair the GXW40XX 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,
port_value+3 for its RTCP and so on. The default value for Profile 1 is 5004 and 6004 for
Profile 2.
Use random port
Default is No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.
This is usually necessary when multiple GXW40XX/HT50X are behind the same NAT.
Refer to Use Target
Contact
Default is No. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
transferred target’s Contact header information.
Transfer on
Conference
Hang up
Default is No. In which case if conference originator hangs up the conference will be
terminated. When option YES is chosen, originator will transfer other parties to each
other so that B and C can choose either to continue the conversation or hang up.
Enable Ring-Transfer
Default is No, this will create a Semi-Attendant Transfer. When set to Yes, device can
transfer the call upon receiving ring back tone.
Disable Bellcore Style
3-Way Conference
Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you
need to dial *23 + second callee number.
Remove OBP from
Route Header:
Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.
Support SIP Instance
ID
Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP
Instance ID as defined in IETF SIP Outbound draft.
Validate incoming SIP 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.
Check SIP user ID for
incoming INVITE
Default is No. Check the SIP User ID in Request URI. If they don’t match, the call will
be rejected.
Allow Incoming SIP
Messages from SIP
Proxy Only
Default is No. If incoming SIP message does not match with SIP Server, it will be
rejected.
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
Maximum retransmission interval for non-INVITE requests and INVITE responses.
Sets the payload type for DTMF using RFC2833.
DTMF Payload Type
Preferred DTMF
method (in listed
order)
The GXW40xx 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.
Disable DTMF
Negotiation
Default is No. If set to yes, use above DTMF order without negotiation
Default is No. If set to yes, flash will be sent as a DTMF event.
Send Hook Flash
Event
Enable Call Features
Proxy Require
Default is Yes. (If Yes, call features using star codes will be supported locally)
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
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Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
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.
Disable Call Waiting
Caller ID
Default is No. If set to YES Call Waiting caller ID will not be provided to analog phone
connected to this FXS port.
Disable Call Waiting
Tone
Default is No. This is to disable the stutter Call Waiting Tone when a Call Waiting call
arrives. The CWCID will still be displayed.
Disable Reminder
Ring for On-Hold Call
Default is No. This is to disable the Reminder Ring that is played when a call is waiting
on hold and the analog phone goes on-hook.
Disable Visual MWI
Visual message indicator is a special on-hook caller ID type message that enables and
disables the message waiting light on certain phones. GXW40XX 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.
Visual MWI Type
Ring Timeout
This is the type of signal sent to the analog phone to make it turn the lamp ON upon
receiving a Voice mail. Check the phone’s manual to find out what signal is supported,
FSK (default) or Neon.
Note: This is only supported in model GXW4024 ver 4.A or newer.
Incoming call will stop ringing when not picked up given a specific period of time.
Hunting Group Ring
Timeout
Default is 20 seconds. If call is not answered within this designated time period, the call
will be forwarded to the next member of a Hunt Group.
Delayed Call Forward Default value is 20 seconds. In case this feature activated using * codes (*92 code), the
Wait Timeout
call will be forwarded after this preconfigured amount of time.
No Key Entry Timeout Default is 4 seconds. Call will be completed within this time interval if no additional key
entry occurs.
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Early Dial
Default is No. Use only if proxy supports 484 response. This parameter controls
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
Sets the prefix added to each dialed number.
Use # as Dial Key
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.
Dial Plan
Dial Plan Rules:
1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d
2. Grammar: x - any digit from 0-9;
a. xx+ - at least 2 digits number;
b. xx. – at least 2 digits number;
c. ^ - exclude;
d. [3-5] - any digit of 3, 4, or 5;
e. [147] - any digit 1, 4, or 7;
f. <2=011> - replace digit 2 with 011 when dialing
g. < =1> - add a leading 1 to all numbers dialed, vice versa will remove a
1 from the number dialed
h. | - or
•
•
Example 1: {[369]11 | 1617xxxxxxx} –
Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617
Example 2: {^1900x+ | <=1617>xxxxxxx} –
Block any number of leading digits 1900 and add prefix 1617 for any dialed 7
digit numbers
•
Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –
Allow any length of number with leading digit 2 and 10 digit-numbers of leading
digit 1 and leading exchange number between 2 and 9; If leading digit is 2,
replace leading digit 2 with 011 before dialing.
3. Default: Outgoing - {x+}
Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right):
• ^1900x. - prevents dialing any number started with 1900
• <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing 7
numbers and 1617 area code will be added automatically
• 1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits
length
• 011[2-9]x. - allows international calls starting with 011
[3469]11 - allow dialing special and emergency numbers 311, 411, 611 and 911
Note: In some cases 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*]+ }.
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Subscribe for MWI
Send Anonymous
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication will be
sent periodically.
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
Rejection
Default is No. If set to Yes, incoming calls with anonymous Caller ID will be rejected with
486 Busy message.
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.
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
Force Timer
If selecting “Yes” the phone will use session timer when it receives inbound calls with
session timer request.
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.
Send Re-INVITE After Default is No, If set to “Yes”, device will send an INVITE with audio vocoders upon
Fax
completition of Fax to continue session in audio only.
Use First Matching
Vocoder in 200OK
SDP
Default is No. If set to “Yes”, device will include only the first match vocoder in its 200OK
response, otherwise it will include all match vocoders in same order received in INVITE.
Preferred Vocoder
The GXW40XX 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.
Sets the iLBC frame size in 20ms or 30ms
iLBC Frame Size
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iLBC Payload type
Default value is 97. Defines payload type for iLBC. The valid range is between 96 and
127.
AAL2-G726-16
Payload type
Default value is 100. Range is from 96 to 127.
Default value is 99. Range is from 96 to 127.
Default value is 104. Range is from 96 to 127.
Default value is 103. Range is from 96 to 127.
Default value is 102. Range is from 96 to 127.
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.
Fax Mode
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)
Fax Tone Detection
Mode
Default is Callee. This decides whether Caller or Callee sends out the re-INVITE for
T.38 or Fax Pass Through.
Jitter Buffer Type
Select either Fixed or Adaptive based on network conditions.
Select Low, Medium or High based on network conditions.
Jitter Buffer Length
•
High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the
high requirement
•
•
Medium (initial 100ms, min 20ms, max 200ms)
Low (initial 50ms, min 10ms, max 100ms)
SRTP Mode
Default is Disabled. Other options are Enabled but not forced, and Enabled and
forced.
It uses SDP Security Description to exchange key. Please refer
Depends on standard phone type (and location)
SLIC Setting
Caller ID Scheme
Select the value according to the local Telco standard where the GXW40XX 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
Disconnect
Set to Yes if the traditional PBX you are using with GXW40XX uses this method for
signaling call termination. Default is No.
Loop Current
Default is 200. In 100 - 1 0000 milliseconds range.
Disconnect duration
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Hook Flash timing
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
On-hook timing is the minimum time for an on-hook event to be validated.
Handset volume adjustment. RX is for receiving volume (direction FXSto analog
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
Configure ring cadences according to preference.
SAVING THE CONFIGURATION CHANGES
After user makes a change to the configuration, press the “Update” button in the Configuration Menu. The
web browser will then display a message window to confirm saved changes.
Grandstream recommends reboot or power cycle the IP phone after saving changes.
REBOOTING FROM REMOTE
Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely. The web
browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in
again.
CONFIGURATION THROUGH A CENTRAL SERVER
Grandstream GXW40XX can be automatically configured from a central provisioning system.
When GXW40XX boots up, it will send TFTP or HTTP request to download configuration file,
“cfg000b82xxxxxx” or “cfg000b82xxxxxx.xml”, where “000b82xxxxxx” is the LAN MAC address of the
GXW40XX. It will first request “cfg000b82xxxxxx” then “cfg000b82xxxxxx.xml”
A service provider or an enterprise with large deployment of Grandstream devices can easily manage the
configuration and service provisioning of individual devices remotely from a central server.
Grandstream has a central provisioning system called GAPS (Grandstream Automated Provisioning
System). GAPS supports automatic configuration of Grandstream devices. GAPS uses enhanced (NAT
friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with
each individual Grandstream device.
Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection
or with certain special provisioning settings. At boot-up, Grandstream devices by default point to
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 customer’s TFTP or HTTP
server for further provisioning. Grandstream also provide GAPSLITE software package which contains
our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device
configuration files.
The GAPSLITE configuration tool is now free to end users. The tool and configuration template are
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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.
Firmware Upgrade through TFTP/HTTP/HTTPS
To upgrade via TFTP or HTTP/HTTPS, the “Firmware Upgrade and Provisioning upgrade via” field needs
to be set to TFTP, HTTP or HTTPS, respectively. “Firmware Server Path” needs to be set to a valid URL
of a TFTP or HTTP/HTTPS server, server name can be in either FQDN or IP address format. Here are
examples of some valid URL.
firmware.mycompany.com:6688/Grandstream/1.0.4.2
or
72.172.83.110
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
address is 72.172.83.110. 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 GXW’s 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. It is recommended to conduct firmware upgrade in a controlled 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.
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
Instructions for local firmware upgrade:
1. Unzip the file and put all of them under the root directory of the TFTP server.
2. Put the PC running the TFTP server and the GXW40XX device in the same LAN segment.
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3. TFTP server's security settings should be changed from "Receive Only" to "Transmit Only" for the
firmware upgrade.
4. Configure the Firmware Server Path with the IP address of the PC
5. Update the change and reboot the unit.
CONFIGURATION FILE DOWNLOAD
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through
TFTP or HTTP/HTTPS. “Config Server Path” is the TFTP or HTTP/HTTPS server path for configuration
file. It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server Path” can
be same or different from the “Firmware Server Path”.
A configuration parameter is associated with each particular field in the web configuration page.
A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page. For a
detailed parameter list, please refer to the corresponding firmware release configuration template.
When Grandstream Device boots up or reboots, it will issue request for configuration file named
“cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the LAN side MAC address of the device, i.e.,
“cfg000b820102ab”. The configuration file name should be in lower cases.
FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX
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 firmwares 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.
MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD
When “Automatic Upgrade” is set to “Yes”, Service Provider can use P193 to have the devices
periodically check with either Firmware Server or Config Server, whenever they are defined. This allows
the device periodically check whether there is any new changes need to be taken, similar to the AntiVirus
Software to upgrade the Virus Definition files. Screenshot is below:
Automatic Upgrade:
No
Check every day
Check every week
1
1
Hour of the day(0-23):
Day of the week(0-6):
If automatic upgrade is enabled, service provider can further customize the behavior and distribute server
load by setting hour of the day and/or day of the week for upgrade.
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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
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. Dial “***” for voice prompt.
2. Enter “99” and wait for “reset” voice prompt.
3. Enter the encoded MAC address (Look below on how to encode MAC address).
4. Wait 15 seconds and device will automatically reboot and restore factory settings.
Encode the MAC Address
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
A:
B:
C:
D:
E:
F:
22 (press the “2” key twice, “A” will show on the LCD)
222
2222
33 (press the “3” key twice, “D” will show on the LCD)
333
3333
For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”.
NOTE:
1. Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.
2. Please be aware by default the GXW40XX WAN side HTTP access is disabled. After a factory reset,
the device’s web configuration page can be accessed only from its LAN port.
Grandstream Networks, Inc.
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