Grandstream Networks GXW4104, GXW4108 User Manual

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Innovative IP Voice & Video

Grandstream Analog IP Gateway GXW410x

Quick Installation Guide

SW version 1.0.0.27

WARNING: Please DO NOT power cycle the GXW410x when LED lights are flashing during system boot up or firmware upgrade. You may corrupt firmware images and cause the unit to malfunction.

Overview

The GXW-410x offers an easy to manage, easy to configure IP communications solution for any small business or businesses with virtual and/or branch locations who want to leverage their broadband network and/or add new IP Technology to their current phone system. The Grandstream Enterprise Analog VoIP Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls. There are two models - the GXW4104 and GXW4108, which have either 4 and 8 FXO ports respectively. The installation is the same for either model.

A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-410x series. In this environment, the SIP server handles SIP registration and call control and the GXW410x processes media conversion between IP and PSTN calls. By design, the system supports the North American call progress tones and signaling standards on PSTN sides.

GXW4100 FEATURES

TFTP and HTTP firmware upgrade support

Multiple SIP accounts, associated with physical line ports, each account corresponding to one of the multiple SIP profile

Multiple SIP profiles, max of 3 profiles per system. Each profile hosts 0 to multiple number of SIP accounts, depending on user need

One stage and two stage dialing

Two stage dialing means when after dialing the number to the GXW, be it from VoIP to GXW or from PSTN to GXW, a second dial-tone prompts users to input the final destination number to finish final dialing.

One stage dialing means user only hear dial-tone once and input a final destination number along with a pre-fix. One stage dialing need SIP server to support SIP call forward via a dial-plan.

VoIP to PSTN call setup and teardown

Channel configurable for one stage or two stage dialing, Default is 2 stage dialing.

PSTN to VoIP call setup and teardown

Channel configurable for one stage or two stage dialing, Default is 2 stage dialing. One stage dialing requires user to configure Off-Hook Auto Dial to a SIP Number.

Support: G711, G723, G729, and GSM

Line echo canceller g.168 support

Flexible DTMF transmission method User Interface of In-audio, RFC2833, and SIP Info

Round-robin port scheduling to ensure available lines to access PSTN networks

Configurable channel dialing to improve dial-out reliability

o digit length: default 100ms

odigit volume: gain [-31,0]dB, default -11dB

odial pause between digits: default 100ms

owait for dial-tone: yes/no, default yes (1 for Yes, 2 for No)

oone-stage ( use 1 ) or 2 stage (use 2) dialing: default of 2 stage dialing

oSyntax: ch (or chan or channel) x-y: val; ch …

Configurable PSTN Termination

oEnable current disconnect: default of disabled. Some special PBXs and CO lines use line power drop to indicate PSTN hang-up. When this is the configuration, please consult your PSTN line service provider for the correct PSTN disconnect method.

oAC termination impedance: default North America. This impedance works with parameters of Busy/Re-order tone in Call Progress Table. Users have to set BUSY/REORDER tone values to enable this parameter.

oBusy or re-order tones: following busy or reorder tone of call progress tones is used to teardown regular PSTN call if detected

Configurable call progress/termination tones via pattern matching

oDial-tone: f1/f2(350/440), v1/v2( -11/ -11), on1/off1(0/0), on2/off2(0/0)

o Ring back tone: f1/f2(default 440/480), on/off(default 2s/4s) o Busy tone: f1/f2(480/620), on/off(0.5/0.5s), duration (8s)

o Re-order tone: f1/f2( 480/620 ), on/off(25/25), duration (default 8s)

o Confirmation tone: f1/f2(350/440), on/off(0.1/0.1s), duration (default 8s) o Usage Syntax:

o ch x-y: f1(or freq1 or frequency1)=val1@vol1, f2 (or freq2 or frequency2) = val2@vol2, c (or cad or cadence) = on1/off1–on2/off2–on3/off3; ch3: …… o x,y - 0-9 digit.

o Configure Channel voice settings,

o Voice volume: gain control, [-31, 31], default 1 dB o Audio input gain: [-31, 31], default 0 dB

o Silence Suppression: 1 – enabled, 2 - disabled, default is 1

oLine echo cancellation: 1 – enabled, 2 – disabled; default is 1

Configure other channel settings, PSTN Silence Timeout, default 60 sec. This serves as a last measure to address PSTN run-away calls. It is not supposed to replace above regular PSTN disconnect methods.

DTMF Method via : default value is in-audio

1in-audio

2– RFC2833

3in-audio and RFC2833

4– SIP Info

5in-audio and RFC2833

www.grandstream.com

info@grandstream.com

GXW-410x Quick Install Guide

VoIPon Solutions www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0) 1245 600560

Grandstream Networks GXW4104, GXW4108 User Manual

Innovative IP Voice & Video

6 – SIP Info and RFC2833

7 – in-audio, RFC2833, and SIP Info

www.grandstream.com

info@grandstream.com

GXW-410x Quick Install Guide

VoIPon Solutions www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0) 1245 600560

Innovative IP Voice & Video

QUICK INSTALLATION AND CONFIGURATION

APPLICATION DESCRIPTION

GXW410x series can be deployed in the following enterprise topology:

GXW-410x Configuration

PSTN

 

FXO

 

 

lines

GXW4100

 

 

 

IP-PBX

Or

SIP Registrar

IP/LAN

 

IP/WAN

 

 

 

301

801

802

803

302

 

 

PSTN Analog Phones

Package and Installation

Unpack and check all accessories included:

one 12V universal power adapter,

one Ethernet cable,

one device unit.

WARNING: use only the power adapter included in the GXW-410x package. Using an alternative power adapter may permanently damage the unit.

Installation

1.Connect power and cables

2.Connect Ethernet cable to a hub and power adapter to power wall outlet

3.Assign GXW410x IP Address

4.Use another Ethernet cable to connect your PC to the same hub that GXW4100 is connected.

5.Assign your PC to a static IP that is within the same subnet of default IP address of GXW4100 192.168.0.160. i.e 192.168.0.x. X can be any number between 2 and 254 except 160.

6.Launch web browser and type http://192.168.0.160 at Address of web browser. This will let you to connect to build-in web server of GXW4100.

You may change to use DHCP or PPPoE connection or another static IP address according to your local network environment.

Simple Configuration As a Media Gateway to Access PSTN Networks

GXW410x can be configured to work with any leading SIP server, such as Asterisk, for a pure media gateway to access PSTN networks. In such applications, the user only needs to configure GXW410xgateway Stage Dialing field and Sip Server field.

For a simple set-up, users only need to configure a SIP server field for default SIP Profile 1. This field should be configured to point to the SIP server to be used with the GXW410x.

For advanced applications, the user is required to choose at least one SIP server field from the SIP profiles and one stage dialing under system Channel configuration table. On SIP server sides, the SIP server must be configured to forward user PSTN calls to the GXW410x.

Please be aware that by default, the system uses North American PSTN settings and TWO STAGE dialing to access PSTN networks for VOIP to PSTN calls and PSTN to VOIP calls. Two stage dialing means the end-user will hear dial-tone twice. First dial-tone is used to let users to input destination number in the same network of the calling networks. Second dialtone is used to let users to input final destination number.

The GXW410x also supports ONE STAGE dialing, which means users only need to input one final PSTN number when first dial-tone is heard for calls from VOIP to PSTN. This requires configuring both SIP server and GXW410x to one stage dialing (see last section of quick guide and user manual for one stage dialing). For one stage PSTN to VOIP calls, user needs to configure off-hook auto dial field (see last section on sample configuration)

Extensive Configuration for Multiple Users

www.grandstream.com

info@grandstream.com

GXW-410x Quick Install Guide

VoIPon Solutions www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0) 1245 600560

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