Grandstream Networks Network Card GXW400x User Manual

Grandstream Networks, Inc.  
Analog IP Gateway GXW–400x  
4 or 8 FXS Ports  
GXW – 400x User’s Manual  
Firmware Version 1.0.0.17  
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TABLE OF FIGURES  
GXW-400X USER MANUAL  
TABLE OF TABLES  
GXW-400X USER MANUAL  
GUI INTERFACES  
GXW-400X USER MANUAL  
1. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE  
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE  
3. SCREENSHOT OF FXS PORTS CONFIGURATION PAGE  
4. SCREENSHOT OF PROFILE 1CONFIGURATION  
5. SCREENSHOT OF PROFILE 2 CONFIGURATION  
6. SCREENSHOT OF STATUS CONFIGURATION LOGIN PAGE  
Grandstream Networks, Inc.  
GXW-400x User Manual  
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WELCOME  
Thank you for purchasing the Grandstream GXW–400x IP Analog FXS Gateway. The GXW–400x is a  
cost effective, easy to use and easy to configure IP communications solution for any business. The  
GXW–400x supports popular voice codecs and is designed for full SIP compatibility and interoperability  
with 3rd party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology, integrate  
a traditional phone system into a VoIP network, and efficiently manage communication costs.  
This manual will help you learn how to operate and manage your GXW FXS Analog IP Gateway and  
make the best use of its many upgraded features including simple and quick installation, 3-way  
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 GXW–400x Overview  
The new GXW–400x 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 GXW–400x 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 port for  
administration. In addition, it supports 2 SIP account profiles, caller ID for various countries/regions, T.38  
fax, 1 or 2-stage dialing, flexible dialing plans, security protection (SIPS/TLS), comprehensive voice  
codecs including G.711 (a/u-law), G.723.1, G.726, G.728, G.729A/B/E and iLBC.  
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 remember to download these  
examples http://www.grandstream.com/GUI/GUI_GXW–400x for your reference.  
This document is subject to change without notice. The latest electronic version of this user manual  
is available for download from the following location:  
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,  
for any purpose without the express written permission of Grandstream Networks, Inc. is not  
permitted.  
Grandstream Networks, Inc.  
GXW-400x User Manual  
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CONNECTING YOUR GXW–400x  
Connecting your GXW–400x is easy. Before you begin, please verify the contents of the GXW–400x  
package.  
EQUIPMENT PACKAGING  
Unpack and check all accessories. Equipment included in the package:  
one 12V universal power adapter  
one Ethernet cable  
one device unit  
CONNECT THE GXW–400X  
Managing the GXW and connecting the unit to the VoIP network is very simple. Follow these four (4)  
steps to connect your GXW to the Internet and access the unit’s configuration pages. The GUI pages will  
guide you through the remaining steps to set-up your gateway. Examples of the GUI Interfaces can be  
1. Connect the Power to the GXW-400x unit.  
2. Connect an Ethernet cable between the LAN port on GXW-400x to your PC.  
3. Your PC will be assigned an IP address like 192.168.2.100  
4. Launch web browser and type http://192.168.2.1 at the address of web browser. This will give you  
access to the GXW-400x web server.  
FIGURE 1: DIAGRAM OF GXW–400X BACK PANEL  
GXW- 400x  
PSTN  
Power  
Supply  
FXS  
Ports  
Line  
Console  
Connection  
(PC  
LAN/WAN  
RJ-45 Ethernet  
Ports  
connection)  
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS  
LAN (or PC)  
Connect your PC to the LAN to find IP address from your Router/DHCP  
Server.  
WAN (or LAN) Connect to the internal LAN network or router.  
VIDEO IN  
RESET  
Connection for Analog based Video Surveillance Camera (RCA)  
Factory Reset button. Press for 7 seconds to reset factory default settings.  
Power adapter connection  
POWER IN  
CONSOLE  
FXS1 - FXS8  
Console connection for a PC.  
FXS ports to be connected to physical PSTN lines.  
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Once the GXW is turned on and configured, the front display panel indicates the status of the unit.  
FIGURE 2: DIAGRAM OF GXW–400X DISPLAY PANEL  
GXW - 400x  
Display LEDs  
FXS port status  
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL  
Power LED  
Ready LED  
LAN LED  
PC LED  
Indicates Power. Remains ON when Power is connected and turned ON.  
Remains ON after boot-up.  
Indicates LAN (or WAN) port activity  
Indicates PC (or LAN) port activity  
LEDs 1 - 8  
Indicate status of the respective FXS Ports on the back panel  
Busy - ON (Solid Green)  
Available - OFF  
NOTE: All LEDs display green when ON.  
SAFETY COMPLIANCES  
The GXW–400x is compliant with various safety standards including FCC/CE. Its power adaptor is  
compliant with UL standard.  
WARRANTY  
Grandstream has a reseller agreement with our reseller customer. End users should contact the company  
from whom you purchased the product for replacement, repair or refund. If you purchased the product  
directly from Grandstream, contact your Grandstream Sales and Service Representative for a RMA  
(Return Materials Authorization) number. Grandstream reserves the right to remedy warranty policy  
without prior notification.  
Warning: Please do not use a different power adaptor with the GXW-400x as it may cause damage to  
the products and void the manufacturer warranty.  
Grandstream Networks, Inc.  
GXW-400x User Manual  
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APPLICATION DESCRIPTION  
There are three scenarios where the GXW–400x series can be effectively used to enable any business to  
leverage the benefits of VoIP and the Internet.  
SCENARIO ONE: GXW- 400X FXS GATEWAY CONFIGURATION / PBX SCENARIO, VOIP  
Anywhere in the world  
FXS  
PBX  
Trunks  
PSTN  
Analog  
Lines  
Trunk  
Line  
4 or 8 Ports  
INTERNET  
CLOUD  
GXW FXS Series  
SCENARIO TWO: GXW- 400X FXS GATEWAY CONFIGURATION / NO PBX SCENARIO, VOIP  
Anywhere in the world  
PSTN  
Analog  
Lines  
4 or 8 Ports  
INTERNET  
CLOUD  
GXW FXS Series  
FXS Gateway acts as a ‘virtual office’,  
connecting analog lines directly to the  
gateway  
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SCENARIO THREE: GXW–410x & GXW–400x SCENARIO / TOLL-FREE CALLING BETWEEN LOCATIONS  
Branch A - Boston, MA  
4 employees  
PSTN  
Analog  
Lines  
FX0 Lines  
Internet  
Cloud  
PSTN  
Cloud  
GXW 400x  
GXW 410x  
Branch B – Denver, CO  
GXW–400X FEATURES  
The GXW–400x is feature rich and is a next generation IP voice gateway that is interoperable and  
compatible with leading IP-PBXs, SoftSwitches and SIP platforms. The GXW–400x FXS Gateway series  
is auto-configurable, remotely manageable and scalable. There are two FXS models, the GXW-4004 and  
GXW-4008, each offering superb voice quality, traditional telephony functionality, easy deployment, and 4  
or 8 FXS ports respectively. Each model features flexible dialing plans, PSTN failover, integrated call  
routing to support a pure IP network call and an external power supply.  
SOFTWARE FEATURES OVERVIEW  
Two SIP accounts & profiles  
Multi-party conferencing  
Supports Voice Codecs:  
G711(a/µ, Annex I & II), G723.1A, G726 (ADPCM), G728, G729 A/B/E, iLBC and GSM  
T.38 Fax  
G.168 Echo Cancellation  
Voice Activity Detection (VAD) with CNG (comfort noise generation) and PLC (packet loss  
concealment)  
Supports PSTN/PBX analog telephone sets or analog trunks  
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TABLE 3: GXW–400X SOFTWARE FEATURES  
GXW– 400x FXS Analog Gateway Series  
Telephone Interfaces  
GXW-4004: 4 ports, 2 SIP accounts & profiles  
GXW-4008: 8 ports, 2 SIP accounts & profiles  
FXS, RJ-11  
Network Interface  
LED Indicators  
Two (2) 10M/100 Mbps, RJ-45  
Power and Line LEDs  
Voice over Packet  
Capabilities  
Voice Activity Detection (VAD) with CNG (comfort noise generation) and  
PLC (packet loss concealment), Dynamic Jitter Buffer,  
Modem detection & auto-switch to G.711,  
Packetized Voice Protocol Unit (supports RTP/RTCP and AAL2 protocol),  
G.168 compliant Echo Cancellation, LEC (line echo cancellation) with NLP  
PSTN Fail-over  
PSTN failover on power failure  
Voice Compression  
G.711 + Annex I (PLC), Annex II (VAD/CNG format) encoder and decoder,  
G.723.1A, G.726(ADPCM), G.729A/B/E, GSM, iLBC  
G.726 provides proprietary VAD, CNG, and signal power estimation  
Voice Play Out unit (reordering, fixed and adaptive jitter buffer, clock  
synchronization), AGC (automatic gain control),  
Status output, Decoder controlling via voice packet header  
DHCP Server/Client  
Fax over IP  
Yes, NAT Router or Switched Mode  
T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to  
G.711 for Fax Pass-through (pending),  
Fax Datapump V.17, V.19, V.27ter, V.29 for T.38 fax relay  
Diffserve, TOS, 802.1 P/Q VLAN tagging  
RTP/RTCP  
QoS  
IP Transport  
DTMF Method  
flexible DTMF transmission method, User interface of In-audio, RFC2833,  
and/or SIP Info  
IP Signaling  
Provisioning  
Control  
SIP (RFC 3261)  
TFTP, HTTP, HTTPS (pending)  
TLS/SIPS  
Management  
Syslog support, HTTPS and telnet (pending), remote management using  
Web browser  
Power  
Output: 12VDC / Input: 100–240 VAC/50-60 Hz  
Operational: 32o–104oF or 0o–40oC  
Environmental  
Storage: 10o–130o F / Humidity: 10–90% Non-condensing  
Dimensions (HxWxD)  
Mounting  
35 x 135 x 225 mm  
Rack mount, Wall mount, Desktop  
Short and long haul  
Caller ID  
REN3: Up to150 ft on 24 AWG line  
Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID  
Polarity Reversal / Wink Yes  
EMC  
EN55022/EN55024 and FCC part15 Class B  
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HARDWARE FEATURES  
4 or 8 FXS ports  
Two RJ-45 (switched or routed)  
Auto detect Ethernet polarity  
PC Console connection  
PSTN fail-over port  
TABLE 4: HARDWARE SPECIFICATION OF GXW–400X  
LAN interface  
LED  
2 x RJ45 10/100Mbps  
8 LEDs (GREEN)  
Universal Switching  
Power Adaptor  
Input: 100-240V AC, 50/60Hz, 0.5A Max  
Output: 12V DC, 1.25A  
UL certified  
Dimension  
Weight  
225mm (L) x 135mm (W) x 35mm (H)  
0.29 lbs (3.5 oz)  
Temperature  
32~104°F  
0~40°C  
Humidity  
10% - 90% (non-condensing)  
FCC, CE  
Compliance  
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USING THE WEB CONFIGURATION PAGES  
1. Connect the Power to the GXW-400x unit.  
2. Connect an Ethernet cable between the LAN port on GXW–400x to your PC.  
3. Your PC will be assigned an IP address like 192.168.2.100  
4. Launch web browser and type http://192.168.2.1 at Address of web browser. This will let you to  
connect to build-in web server of GXW–400x.  
5. Login to the GUI using the password “admin” for the Advanced Configuration Page or “123” for  
the Basic Configuration Page.  
CONFIGURING THE GXW–400X  
1. There are six different tabs (Status, Basic Settings, Advanced Settings, Profile 1, Profile 2 and FXS  
Ports) on the top of the screen (after login). To open each page, click on the tab.  
2. Click on Profile 1 to enter your SIP Server/ SIP Proxy/Registrar information. Enter the IP Address (or  
FQDN) of the Server under: SIP Server and/or Outbound Proxy.  
3. Click on FXS ports to enter the extensions or account information. You will need to fill in the  
following information for each extension. Once the extensions are configured, you are finished.  
FXS SIP User Auth ID Authentication  
Name  
Profile  
Port  
ID  
Password  
1
2
3
4
5
6
7
8
601  
602  
603  
604  
605  
606  
607  
608  
601  
602  
603  
604  
605  
606  
607  
608  
***  
***  
***  
***  
***  
***  
***  
***  
Ext 1  
Ext 2  
Ext 3  
Ext 4  
Ext 5  
Ext 6  
Ext 7  
Ext 8  
Profile1  
Profile1  
Profile1  
Profile1  
Profile1  
Profile1  
Profile1  
Profile1  
4. Click Update after changing any setting and then Re-boot to confirm changes.  
5. After reboot, check the Status Page to confirm the extensions are successfully registered. You can  
now use your standard phones connected to ports FXS1 to FXS8 to make calls.  
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)  
This is a tricky setting. It really depends on your network settings, so set this parameter to Yes or No,  
whichever works. 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.  
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DTMF METHODS  
DTMF Settings are in Profile pages.  
DTMF in-audio  
DTMF via RTP (RFC2833)  
DTMF via SIP INFO  
Enable one or more DTMF methods based on your PBX system.  
PREFERRED VOCODER (CODEC)  
The GXW–400x supports a broad range of voice codecs. Under Profile web pages, choose your  
preferred order of different codecs:  
PCMU (or G711µ)  
PCMA (or G711a)  
G729 A/B/E  
G723  
G726-32  
iLBC  
G728  
GSM  
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BASIC OPERATIONS  
UNDERSTANDING GXW VOICE PROMPTS  
GXW–400x has a stored voice prompt menu for quick browsing and simple configuration. Enter *** on  
the standard analog phone connected to any FXS port to use the voice prompt menu.  
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 – 06, 47, 86 or 99 Menu option  
01  
“DHCP Mode”, or  
“Static IP Mode”  
Enter ‘9’ to toggle the selection  
Static IP Mode”: configure all the IP address information through  
menu options 02 to 05.  
Dynamic IP Mode”: IP address information is retrieved from the  
DHCP server automatically at reboot.  
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  
“DNS Server “ + IP address  
Preferred Vocoder  
Same as Menu option 02  
Same as Menu option 02  
Same as Menu option 02  
Enter “9” to go to the next selection in the list:  
PCM U / PCM A  
G-723  
G-729  
iLBC  
G-726-32  
G728  
G729E  
10  
12  
13  
“MAC Address”  
Announces the Mac address of the unit.  
WAN Port Web Access  
Firmware Server IP Address  
Enter “9” to toggle between enable and disable  
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  
99  
“Direct IP Calling”  
User will be prompted with a dial tone to dial a 12-digit IP address  
to make a direct IP call.  
(For details, see Make a Direct IP Call)  
“RESET”  
Enter “9” to reboot the device; or  
Enter MAC address to restore factory default setting.  
“Invalid Entry”  
Automatically returns to Main Menu  
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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 PHONE CALLS  
CALLING PHONE OR EXTENSION NUMBERS  
There are currently two methods to make an extension number call:  
1. Dial the numbers directly and wait for 4 (default) seconds.  
2. Dial the numbers directly, and press # (assuming that “use # as dial key is selected in web  
configuration).  
Examples:  
To dial another extension on the same proxy, such as 1008, simply pick up the attached phone,  
dial 1008 and then press the # or wait for 4 seconds.  
To dial a PSTN number such as 6266667890, you may need a prefix number followed by the  
phone number. Please check with your VoIP service provider for correct information. If your  
phone is assigned a PSTN-like number such as 6265556789, it is likely that you dial “1” prior to  
the number - 16266667890.  
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. A direct IP call can be completed if:  
1. Both GXW–400X and other VoIP Device, have public IP addresses, or  
2. Both GXW–400X and other VoIP Device are on the same LAN using private IP addresses, or  
3. Both GXW–400X and other VoIP Device can be connected through a router using public or  
private IP addresses (with necessary port forwarding or DMZ).  
To make a direct IP call  
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 a 12-digit target IP address after the dial tone and voice prompt “Direct IP Calling”  
Destination ports can be specified by using “*4” (encoding for “:”) followed by the port number.  
NOTE: When completing direct IP call, the “Use Random Port” should set to “NO”. You can not make  
direct IP calls between FXS1 to FXS2 since they are using same IP.  
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EXAMPLES OF DIRECT IP CALLS:  
1. If the target IP address is 192.168.0.10, the dialing convention is:  
Voice Prompt with option 47, then 192 168 000 010 followed by the “#” key or wait 4 seconds.  
2. If the target IP address/port is 192.168.1.20:5062, then the dialing convention is:  
Voice Prompt with option 47, then 192168001020*45062 followed by the “#” key or wait 4  
seconds.  
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 (series of quick consecutive dial tones) during conversation indicates an incoming call (if  
the call waiting feature is enabled). Press the flash button to place the current caller on hold and switch  
to the other call. 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:  
1. Caller A presses FLASH on the analog phone to hear the dial tone.  
2. Caller A dials *87 then dials caller C’s number, and then # (or wait for 4 seconds)  
3. Caller 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.  
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 GXW 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.  
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3-WAY CONFERENCING  
Star Code Style 3-way Conference  
Caller A wants to bring third Caller C into conference:  
1. Caller A presses FLASH to receive a dial tone. (Hook Flash for older model phones)  
2. Caller A dials Caller C’s number then # (or wait for 4 seconds).  
3. If Caller C answers call, then Caller A presses “flash” to bring Parties B & C into conference.  
4. If Caller C does not answer the call, Caller A can press “flash” to resume call with Caller B.  
Bellcore Style 3-way Conference  
The GXW–400x also supports Bellcore style 3-way Conference. Ensure to enable “Use Bellcore Style 3-  
way Conference” under Profile Page. Follow directions for Star Code Style.  
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CALL FEATURES  
The GXW–400x supports traditional and advanced telephony features. These features are easy to use  
with any phone.  
TABLE 6: CALL FEATURES TABLE (STAR CODE)  
Key  
*23  
*30  
*31  
*67  
*82  
*50  
*51  
*70  
*71  
*72  
Call Features  
3-way Conferencing  
Block CallerID (for all-config change)  
Send CallerID (for all-config change)  
Block CallerID (per call)  
Send CallerID (per call)  
Disable Call Waiting (for all-config change)  
Enable Call Waiting (for all-config change)  
Disable Call Waiting (Per Call)  
Enable Call Waiting (Per Call)  
Unconditional Call Forward: Dial “*72” for a dial tone. Dial the forwarding number  
followed by “#”. Wait for dial tone and hang up. (dial tone indicates successful forward)  
*73  
*87  
*90  
Cancel Unconditional Call Forward: Dial “*73” and wait for dial tone, then hang up.  
Blind Transfer  
Busy Call Forward: Dial “*90” for a dial tone. Dial 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” for a dial tone. Dial 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.  
PSTN PASS THROUGH/LIFE LINE  
The RJ-11 line jack on the GXW–400x side will function as a pass through jack when GXW–400x is out of  
power. The pass through/life line enables the user to use the analog phone for PSTN calls directly  
without using an access code.  
T.38 FAX SUPPORT  
GXW–400x 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. To send or receive faxes in fax pass  
through mode, users must select all the Preferred Codecs to be PCMU/PCMA (G.711-u/a).  
End User Configuration  
Once this HTTP request is entered and sent from a Web browser, the GXW–400x will respond with a  
login screen. There are two default passwords for the login page:  
User Level:  
Password: Web pages allowed:  
End User Level  
Administrator Level  
123  
admin  
Only Status and Basic Settings  
All pages can be browsed.  
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TABLE 7: BASIC SETTINGS PAGE DEFINITIONS  
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.  
There are two modes to operate the GXW–400x:  
IP Address  
DHCP mode: all the field values for the Static IP mode are not used (even  
though they are still saved in the Flash memory.) The GXW–400x acquires its IP  
address from the first DHCP server it discovers from the LAN it is connected.  
Using the PPPoE feature: set the PPPoE account settings. The GXW–400x will  
establish a PPPoE session if any of the PPPoE fields is set.  
Static IP mode: configure the IP address, Subnet Mask, Default Router IP  
address, DNS Server 1 (primary), DNS Server 2 (secondary) fields. These fields  
are set to zero by default.  
Time Zone  
Controls how the date/time is displayed according to the specified time zone.  
TBD  
Self Defined Time  
Zone  
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.  
WAN side HTTP  
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.  
Reply to ICMP on  
WAN port  
If set to “Yes”, the GXW400x will respond to the PING command from other  
computers, but it also is vulnerable to the DOS attack. Default is No.  
Cloned WAN MAC  
Addr  
Enables a specific MAC address. Set in Hex format.  
LAN Subnet Mask  
LAN DHCP Base IP  
Sets the LAN subnet mask. Default value is 255.255.255.0  
Base IP for the LAN port which functions as a Gateway for the subnet.  
Default value is 192.168.2.1.  
DHCP IP Lease Time 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  
Forward all WAN IP traffic to a specific IP address if no matching port is used by  
GXW400x or defined in port forwarding.  
Port Forwarding  
Forwards a matching (TCP/UDP) port to a specific LAN IP address with a  
specific (TCP/UDP) port.  
In addition to the Basic Settings configuration page, the end user also has access to the Device Status  
page.  
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TABLE 8: STATUS PAGE DEFINITIONS  
MAC Address  
The device ID in HEX format. This is needed for ISP troubleshooting.  
WAN IP Address  
Product Model  
Software Version  
Shows LAN IP address of GXW–400x  
Contains the product model info.  
Program: This is the main software release. Boot and Loader are not changed  
often.  
System Up Time  
PPPoE Link Up  
Shows system up time since the last reboot.  
Shows whether the PPPoE connection is running if connected to DSL modem.  
Detected NAT Type  
Shows type of NAT the GXW–400x is connected to via its LAN port. It is based  
on STUN protocol.  
Port  
Indicates whether the different FXS Ports are registered to the SIP server(s).  
ADVANCED USER SETTINGS  
ADVANCED USER CONFIGURATION  
To access the advanced user configuration page, follow the directions for Using the Web Configuration  
Pages. Advanced User configuration includes the end user configuration and the advanced  
configurations including: SIP configuration, Codec selection, NAT Traversal Setting and other  
miscellaneous configuration.  
FIGURE 3: ADVANCED USER CONFIGURATION SCREENSHOT  
Grandstream Device Configuration  
Password  
Login  
All Rights Reserved Grandstream Networks, Inc. 2005-2006  
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TABLE 9: ADVANCED CONFIGURATION PAGE DEFINITIONS  
Admin  
Administrator password. Only the administrator can configure the “Advanced Settings”  
Password  
page. Password field is purposely left blank for security reasons 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.  
Keep-alive  
interval  
This parameter specifies how often the GXW–400x sends a blank UDP packet to the  
SIP server to keep the “hole” on the NAT open. Default is 20 seconds.  
Firmware  
Upgrade &  
Provisioning  
Enables GXW–400x to download firmware or configuration file through either the TFTP  
or HTTP server.  
Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or  
not blank, the GXW–400x 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: Please do NOT interrupt the TFTP upgrade process (especially the power supply)  
as this will damage the device. Depending on the network environment this process  
can take up to 15 or 20 minutes.  
Via HTTP Server The URL for the HTTP server used for firmware upgrade and configuration via HTTP.  
For example, ttp://provisioning.mycompany.com:6688/Grandstream/1.0.0.36  
:6688” is the specific TCP port where the HTTP server is listening; it can be omitted if  
using default port 80.  
Note: If Auto Upgrade is set to No, GXW–400x will only do HTTP download once at  
boot up.  
Automatic  
Upgrade  
Choose “Yes” to enable automatic upgrade and provisioning. In “Check for new  
firmware every” field, enter the number of days to enable GXW–400x to check the  
server for firmware upgrade or configuration in the defined period of days. When set to  
No, GXW–400x will only do upgrade once at boot up. “Always check for New  
Firmware.” Check New Firmware only when F/W pre/suffix changes”  
Authenticate  
Conf File  
If set to Yes, config file is authenticated before acceptance. This protects the  
configuration from an unauthorized change.  
Firmware Key  
For firmware encryption. It should be 32 digit in Hexadecimal Representation. End  
user should keep it blank.  
Lock Keypad  
Update  
If set to “Yes”, the configuration update via keypad is disabled.  
NTP server  
URI or IP address of the NTP (Network Time Protocol) server. Used by the phone to  
synchronize the date and time.  
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Syslog Server  
Syslog Level  
The IP address or URL of System log server. This feature is especially useful for the  
ITSP (Internet Telephone Service Provider)  
Select the GXW–400x 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  
TABLE 10: FXS PORTS CONFIGURATION DEFINITIONS  
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 GXW–400x to register to (SIP) servers  
of ITSP.  
Name  
Name  
Profile ID  
Select the corresponding Profile ID (1/2)  
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TABLE 11: PROFILE PAGE DEFINITIONS  
Profile Active  
SIP Server  
When set to Yes the SIP Profile is activated.  
SIP Server’s IP address or Domain name provided by VoIP service provider.  
Outbound Proxy IP address or Domain name of Outbound Proxy, or Media Gateway, or Session  
Border Controller. Used by GXW–400x 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.  
This parameter defines whether the GXW–400x NAT traversal mechanism is  
activated or not. If activated (by choosing “Yes”) and a STUN server is also  
specified, then the GXW–400x 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 GXW–400x 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 GXW–  
400x will periodically (every 20 seconds or so) send a blank UDP packet (with no  
payload data) to the SIP server to keep the “hole” on the NAT open.  
Use DNS SRV:  
Default is No. If set to “Yes” the client will use DNS SRV to look up server.  
User ID is Phone If the GXW–400x has an assigned PSTN telephone number, this field should be set  
Number  
to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be  
attached to the “From” header in SIP request.  
SIP Registration This parameter controls whether the GXW–400x 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  
Default is No. If set to “Yes,” user can place outgoing calls even when not  
w/o Registration registered (if allowed by ITSP) but is unable to receive incoming calls.  
Register  
Expiration  
Allows the user to specify the time frequency (in minutes) for the GXW–400x to  
refresh its registration with the specified registrar. The default interval is 60 minutes  
(or 1 hour). The maximum interval is 65535 minutes (about 45 days).  
Local SIP port  
Local RTP Port  
Defines the local SIP port the GXW–400x will listen and transmit. The default value  
for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and Account  
4 respectively.  
Defines the local RTP-RTCP 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 RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP  
and port_value+3 for its RTCP. The default value is 5004.  
Refer to Use  
Used for call transfer, incase server requires this setting.  
Target Contact  
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DTMF Payload  
Type  
Sets the payload type for DTMF using RFC2833.  
DTMF in-audio  
DTMF via RTP  
Send DTMF as inband (in-audio).  
Send DTMF via RTP (According to RFC 2833).  
Send DTMF via SIP INFO message.  
DTMF via SIP  
INFO  
Send Flash  
Event  
Default is NO. If set to yes, flash will be sent as DTMF event.  
Enable Call  
Features  
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are supported locally  
Off-Hook Auto  
Dial  
Allows the user to configure a User ID or extension number to be automatically  
dialed upon off-hook. Only the user part of a SIP address needs to be entered  
here. The phone will automatically append the “@” and the host portion of the  
corresponding SIP address.  
Proxy Require  
Use NAT IP  
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.  
NAT IP address used in SIP/SDP message. Default is blank.  
Distinctive  
Ringtone  
CustomRing 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.  
Disable Call  
Waiting  
Default is No.  
Disable Call  
Default is No.  
Waiting Tone  
Ring Timeout  
Incoming call will stop ringing when not picked up given a specific period of time.  
No Key Entry  
Timeout  
Default is 4 seconds.  
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.  
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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  
“(Re)Dial” key. If set to “No”, this “#” key can be included as part of number.  
Subscribe for  
MWI  
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 in outgoing INVITE message will  
be set to anonymous, blocking Caller ID.  
Special Feature  
Default is Standard. Choose the selection to meet some special requirements from  
Softswitch vendors.  
Preferred  
Vocoder  
The GXW–400x supports up to 5 different Vocoder types including G.711 A-/U-law,  
GSM, G.723.1, G.729A/B. The user can configure Vocoders in a preference list  
that will be included with the same preference order in SDP message. The first  
Vocoder 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.  
iLBC Frame Size Sets the iLBC frame size in 20ms or 30ms  
iLBC Payload  
type  
Defines payload type for iLBC. Default value is 97. The valid range is between 96  
and 127.  
G726-16 Payload Default value is 98. Range is from 96 to 127.  
type  
G726 - 24  
Payload type  
Default value is 99. Range is from 96 to 127.  
Default value is 100. 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.  
Default is No.  
G726 - 32  
Payload type  
G726 - 40  
Payload type  
G729E payload  
type  
VAD  
Fax Mode  
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec  
PCMU/PCMA)  
Jitter Buffer Type Select either Fixed or Adaptive based on network conditions.  
Jitter Buffer  
Length  
Select Low, Medium or High based on network conditions.  
SLIC Setting  
Dependent on standard phone type (and location)  
Caller ID Scheme • Bellcore/Telcordia  
• ETSI-DTMF  
• ETSI-FSK  
• SIN 227 - BT  
• NTT Japan  
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Polarity Reversal Default is No. If set to “Yes”, polarity will be reversed upon call establishment and  
termination.  
Hook Flash  
timing  
Gain  
Handset volume adjustment. RX is for receiving volume, TX is for transmission  
volume. Default values are 0dB for both parameters. Loudest volume: +6dB  
Lowest volume: -6dB.  
Call Progress/  
Ring Tones  
Configure ring or tone frequencies according to preference. By default tones are  
set to North American frequencies. Frequencies should be configured with known  
values to avoid uncomfortable high pitch sounds.  
SAVING THE CONFIGURATION CHANGES  
Once a change is made, press the “Update” button in the Configuration Menu. The GXW–400x will  
display the following screen to confirm that the changes have been saved. Reboot or power cycle the  
GXW–400x after all the changes are made so that those changes can take effect.  
FIGURE4: SAVING CONFIGURATION CHANGES SCREENSHOT  
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FIRMWARE UPGRADE  
Our latest official release can be downloaded from: http://www.grandstream.com/y-firmware.htm. To  
upgrade your unit firmware, follow these steps:  
1. Under Advanced Settings webpage, enter your TFTP or HTTP Server IP address (or FQDN) next  
to the “Firmware Upgrade: Upgrade Server” field.  
2. Select via TFTP or HTTP accordingly.  
3. If you plan to use Automatic Upgrade, set it to “Yes”, otherwise No (this will make it check for  
upgrade every time you reboot).  
UPGRADE THROUGH HTTP  
To upgrade firmware via HTTP, the field “Firmware Upgrade and Provisioning: Upgrade Via” needs to be  
set to HTTP. The “Firmware Server Path” should be set to where the firmware files are located.  
For example, the user can use the following URL in the Firmware Server Path:firmware.mycompany.com:  
6688/Grandstream/1.0.0.29 where firmware.mycompany.com is the FQDN of the HTTP server. It can  
also be in IP address format. “:6688” is the TCP port the HTTP server listening to, default http server  
listens to port 80. “/Grandstream/1.0.0.29” is the RELATIVE directory to the root dir on HTTP web server.  
UPGRADE THROUGH TFTP  
To upgrade firmware via TFTP, set the field “Firmware Upgrade and Provisioning: Upgrade Via” to TFTP.  
The TFTP server can be configured in either IP address format or FQDN.  
To configure the TFTP server via the Web configuration interface, follow these five steps:  
1. Open your browser to input the IP address of the GXW–400x.  
2. Enter the admin password to enter the configuration screen.  
3. Enter the TFTP server address or URL in the “Firmware Server Path” field near the  
bottom of the configuration screen.  
4. Once the “Firmware Server Path” is set, update the change by clicking the “Update”  
button.  
5. Reboot or power cycle the unit.  
If the configured updating server is found and a new code image is available, the GXW–400x will retrieve  
the new image files by downloading them into the GXW–400x’s SRAM. During this stage, the GXW–  
400x’s LED will blink until the checking/downloading process is completed. Upon verification of  
checksum, the new code image will be saved into the Flash. If TFTP fails for any reason (e.g., TFTP  
server is not responding, there are no code image files available for upgrade, or checksum test fails, etc),  
the GXW–400x will stop the TFTP process and simply boot using the existing code image in the flash.  
A firmware upgrade may take as long as 20 minutes over the Internet, or 20+ seconds if performed on a  
LAN. Grandstream recommends conducting firmware upgrades in a controlled LAN environment if  
possible.  
NO LOCAL TFTP SERVER  
For users who do not have a local TFTP server, Grandstream provides a NAT-friendly TFTP server on  
the public Internet for users to download the latest firmware upgrade automatically. Please check the  
Services section of Grandstream’s Web site to obtain this TFTP server IP address. Alternatively, user  
can download and install a free TFTP or HTTP server in his LAN for a firmware upgrade. A free  
Windows version TFTP server can be downloaded from:  
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Directions For Downloading TFTP Server:  
1. Unzip the file and put all of the files under the root directory of the TFTP server.  
2. Put the PC running the TFTP server and the GXW–400x in the same LAN segment.  
3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive  
Only" to "Transmit Only" for the firmware upgrade.  
4. Start the TFTP server, in the phone’s web configuration page.  
5. Configure the Firmware Server Path with the IP address of the PC.  
6. Update the change and reboot the unit.  
You can also download the free HTTP server from http://httpd.apache.org/ or just use Microsoft IIS web  
RESTORE FACTORY DEFAULT SETTING  
WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone.  
Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will  
not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP  
service provider.  
FACTORY RESET  
The only way to reset default factory settings is to follows 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.  
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GLOSSARY OF TERMS  
ADSL Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit  
from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800 kbps upstream,  
depending on line distance.  
AGC Automatic Gain Control is an electronic system found in many types of devices. Its purpose is to  
control the gain of a system in order to maintain some measure of performance over a changing range of  
real world conditions.  
ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically  
IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The protocol  
operates below the network layer as a part of the interface between the OSI network and OSI link layer. It  
ATA Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP, like  
Grandstream HT series products.  
CODEC Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A)  
converter for translating the signals from the outside world to digital, and back again.  
CNG Comfort Noise Generator, generate artificial background noise used in radio and wireless  
communications to fill the silent time in a transmission resulting from voice activity detection.  
DATAGRAM A data packet carrying its own address information so it can be independently routed from  
its source to the destination computer  
DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or  
compressed. Lossy compression algorithms ordinarily decimate while sub-sampling.  
DECT Digital Enhanced Cordless Telecommunications: A standard developed by the European  
Telecommunication Standard Institute from 1988, governing pan-European digital mobile telephony.  
DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public  
switched telephone network, Closed User Groups (CUGs), Local Area Networks, and wireless local loop.  
The DECT Common Interface radio standard is a multi-carrier time division multiple access, time division  
duplex (MC-TDMA-TDD) radio transmission technique using ten radio frequency channels from 1880 to  
1930 MHz, each divided into 24 time slots of 10ms, and twelve full-duplex accesses per carrier, for a total  
of 120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all 12  
possible accesses (time slots) simultaneously by using different frequencies or using only one frequency.  
All signaling information is transmitted from the RFP within a multi-frame (16 frames). Voice signals are  
digitally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation.  
DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain  
names into IP addresses  
DID Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going  
through an attendant or auto-attendant.  
DSP Digital Signal Processor. A specialized CPU used for digital signal processing. Grandstream  
products all have DSP chips built inside.  
DTMF Dual Tone Multi Frequency. The standard tone-pairs used on telephone terminals for dialing  
using in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals  
support only 12 of them (0-9, * and #).  
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FQDN Fully Qualified Domain Name. A FQDN consists of a host and domain name, including top-level  
domain. For example, www.grandstream.com is a fully qualified domain name. www is the host,  
Grandstream is the second-level domain, and and.com is the top level domain.  
FXS Foreign eXchange Office. An FXS device can be an analog phone, answering machine, fax, or  
anything that handles a call from the telephone company like AT&T. They should also operate the same  
way when connected to an FXS interface.  
An FXS interface will accept calls from FXS or PSTN interfaces. All countries and regions have  
their own standards.  
FXS is complimentary to FXS (and the PSTN).  
FXS Foreign eXchange Station. An FXS device has hardware to generate the ring signal to the FXS  
extension (usually an analog phone).  
An FXS device will allow any FXS device to operate as if it were connected to the phone  
company. This makes your PBX the POTS+PSTN for the phone.  
The FXS Interface connects to FXS devices (by an FXS interface, of course).  
DHCP The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the  
configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses, to  
deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to provide  
other configuration information such as the addresses for printer, time and news servers.  
ECHO CANCELLATION Echo Cancellation is used in telephony to describe the process of removing  
echo from a voice communication in order to improve voice quality on a telephone call. In addition to  
improving quality, this process improves bandwidth savings achieved through silence suppression by  
preventing echo from traveling across a network. There are two types of echo of relevance in telephony:  
acoustic echo and hybrid echo. Speech compression techniques and digital processing delay often  
contribute to echo generation in telephone networks.  
H.323 A suite of standards for multimedia conferences on traditional packet-switched networks.  
HTTP Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and retrieve  
functions of a server  
IP Internet Protocol. A packet-based protocol for delivering data across networks.  
IP-PBX IP-based Private Branch Exchange  
IP Telephony (Internet Protocol telephony, also known as Voice over IP Telephony) A general term for  
the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and  
other forms of information that have traditionally been carried over the dedicated circuit-switched  
connections of the public switched telephone network (PSTN). The basic steps involved in originating an  
IP Telephony call are conversion of the analog voice signal to digital format and compression/translation  
of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched  
networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony  
are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is  
only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP  
Telephony software essentially provides free telephone calls anywhere in the world. However, the  
challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border  
controllers resolve this issue by providing quality assurance comparable to legacy telephone systems.  
IVR IVR is a software application that accepts a combination of voice telephone input and touch-tone  
keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and  
perhaps other media.  
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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-  
bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for  
Ethernet is 1500 byte.  
NAT Network Address Translation  
NTP Network Time Protocol, a protocol to exchange and synchronize time over networks The port used  
is UDP 123 Grandstream products using NTP to get time from Internet  
OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.  
OBP/SBCs are put into the signaling and media path between calling and called Caller. The OBP/SBC  
acts as if it was the called VoIP phone and places a second call to the called Caller. The effect of this  
behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the  
OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private  
OBP/SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network.  
Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks  
with internet connections using NAT.  
PPPoE Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in  
Ethernet frames. It is used mainly with cable modem and DSL services.  
PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or  
called POT (Plain Old Telephone), or circuit switched network.  
RTCP Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time  
Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does not  
transport any data itself. It is used periodically to transmit control packets to participants in a streaming  
multimedia session. The primary function of RTCP is to provide feedback on the quality of service being  
provided by RTP.  
RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video  
over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first  
published in 1996 as RFC 1889  
SDP Session Description Protocol is a format for describing streaming media initialization parameters. It  
has been published by the IETF as RFC 2327.  
SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261).  
SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice  
transmission and uses fewer resources and is considerably less complex than H.323. All Grandstream  
products are SIP based  
STUN Simple Traversal of UDP over NATs is a network protocol allowing clients behind NAT (or multiple  
NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by  
the NAT with a particular local port. This information is used to set up UDP communication between two  
hosts that are both behind NAT routers. The protocol is defined in RFC 3489. STUN will usually work well  
with non-symmetric NAT routers.  
TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP,  
applications on networked hosts can create connections to one another, over which they can exchange  
data or packets. The protocol guarantees reliable and in-order delivery of sender to receiver data.  
TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very  
basic form of FTP; It uses UDP (port 69) as its transport protocol.  
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UDP User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using  
UDP, programs on networked computers can send short messages known as datagrams to one another.  
UDP does not provide the reliability and ordering guarantees that TCP does; datagrams may arrive out of  
order or go missing without notice. However, as a result, UDP is faster and more efficient for many  
lightweight or time-sensitive purposes.  
VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing  
wherein, the presence or absence of human speech is detected from the audio samples.  
VLAN A virtual LAN, known as a VLAN, is a logically-independent network. Several VLANs can co-exist  
on a single physical switch. It is usually refer to the IEEE 802.1Q tagging protocol.  
VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of  
signaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc.  
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