Basic Configuration Grandstream GXW4104 FXO Gateway with Asterisk Pbx

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GrandStreamGxw4104In this post I want to show how to configure the GXW410x to work with Asterisk Pbx.

1) Update the device to the latest firmware. At the time of writing this post latest firmware revision is 1.4.1.5: if you have a firmware before 1.3.4.13 you have to upgrade before to 1.3.4.13 and after that upgrade to the latest firmware because it is not possible to upgrade directly.
2) In Asterisk.

[1001]
type=friend
secret= <password>
host=dynamic
context=from-trunk
insecure=very
call-limit=1
nat=no
canreinvite=no
dtmfmode=rfc2833
qualify=yes

[1002]
type=friend
secret= <password>
host=dynamic
context=from-trunk
insecure=very
call-limit=1
nat=no
canreinvite=no
dtmfmode=rfc2833
qualify=yes

[1003]
type=friend
secret= <password>
host=dynamic
context=from-trunk
insecure=very
call-limit=1
nat=no
canreinvite=no
dtmfmode=rfc2833
qualify=yes

[1004]
type=friend
secret= <password>
host=dynamic
context=from-trunk
insecure=very
call-limit=1
nat=no
canreinvite=no
dtmfmode=rfc2833
qualify=yes

GrandStream4104-WebNwStatus

 

 

 

 

 

3) Device via web interface

Accounts->Account1->General Settings
Account Active: Yes
Account Name: General
SIP Server:
Outbound Proxy:

Accounts->Account1->Network Settings
Use DNS SRV: No
NAT Traversal (STUN): No, but send keep-alive
Proxy-Require:

Accounts->Account1->SIP Settings
SIP Registration: Yes
Unregister On Reboot: Yes
Register Expiration: 60
SIP Reg Failure Retry Wait: 20
SIP Transport: UDP
Session Expiration: 180
Special Feature: Standard
Account Active

Accounts->Account2->General Settings
Account Active: No

Accounts->Account3->General Settings
Account Active: No

Accounts->User Account
Channels: 1 - SIP User ID: 1001 - Authenticate ID: 1001 - Authen Password: <password> - SIP Account: Account1
Channels: 2 - SIP User ID: 1002 - Authenticate ID: 1001 - Authen Password: <password> - SIP Account: Account1
Channels: 3 - SIP User ID: 1003 - Authenticate ID: 1001 - Authen Password: <password> - SIP Account: Account1
Channels: 4 - SIP User ID: 1004 - Authenticate ID: 1001 - Authen Password: <password> - SIP Account: Account1

Settings->Call Settings
Voice Frames per TX: 2

Settings->Channels Settings
DTMF Methods(1-7): 2

Networks->Advanced Settings
Layer 3 QoS: 63

For all lines
The following fields need to be set according to the PSTN Service provider. For a detailed list of the worldwide database of call progress tones in the world please check “Various Tones used in nation networks (according to ITU-T recommendation E.180)” in linkografia. The values here are related to Italy.

FXO Lines-> FXO Settings
Dial Tone: ch1-4:f1=425@-14,f2=425@-14,c=200/200-600/1000;
Ringback Tone: ch1-4:f1=440@-19,f2=480@-19,c=2000/4000;
Busy Tone: ch1-4:f1=425@-11,f2=425@-11,c=200/200;
Reorder Tone: ch1-4:f1=425@-11,f2=425@-11,c=200/200;
Enable Current Disconnect(Y/N): ch1-4:N;
Enable Tone Disconnect: ch1-4:Y;
Enable Call Supervision: ch1-4:N;
Number of Rings Before Pickup: ch1-4:3;
Caller ID Scheme: ch1-4:2;
Caller ID Transport Type: ch1-4:1;

FXO Lines-> Dialing
Wait for Dial-Tone(Y/N): ch1-4:N;
Stage Method(1/2): ch1-4:1;
Min Delay Before Dialing Out: ch1-4:50;

Linkografia
Various Tones used in nation networks (according to ITU-T recommendation E.180)

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