Docking AIU-8 FXO Service Board and Master System Asterisk16.15.1

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Docking AIU-8 FXO Service Board and Master System Asterisk16.15.1

This article introduces the detailed steps to connect the service board to Asterisk.

Follow the steps below to configure a two-way call between the phone and the gateway.

Outgoing calls: from Asterisk SIP extension 3002 to the gateway via trunk 1008.

Incoming call: call from outside line to gateway, call to Asterisk through SIP trunk 1008, then the call is sent to 3002 SIP extension through Asterisk.

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In the following steps, the following parameters are mandatory, other parameters can be configured according to your needs.

Step 1: Create a SIP trunk on the Asterisk server

Please add the following lines to /etc/asterisk/sip.conf to create a SIP trunk (1001) and a SIP extension (3002).

[1001]

username=1001

secret=1001

host=dynamic

port=5060

type=friend

context = from-gsm

[3002]

username=3002

secret=3002

host=dynamic

port=5060

type=friend

context = from-internal

After editing, save and exit, restart the SIP service of Asterisk server

 

 

Step 2: Edit Dialing Rules in Asterisk

Edit the dialing rules in /etc/asterisk/extensions.conf.

[from-internal]

exten => _x.,1,Dial(sip/1001/${EXTEN})

exten => _x.,n,hangup()

[from-gsm]

exten =>_x.,1,Dial(sip/3001)

exten =>_x.,n,Hangup()

 

 

Step 3: Set the network parameters in the network

Log in to the network in your browser and set the network parameters. The following figure is an example for reference only.

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Step 4: Create a SIP endpoint in the network

Set up SIP trunking. The following figure shows detailed information on how to set it up.

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For other parameters in SIP, please set them according to your requirements, as they are not necessary in simple calls.

 

 

Step 5: Set up routing rules in the network

Set the routing rules for outbound and inbound calls as shown below.

Inbound routing rules

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Outbound routing rules规则

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Step 6: Register a SIP extension through the software

Use SIP software such as Xlite, eyeBeam to register a SIP extension (3001). After all the above steps, you can try to make a call.

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Use SIP software such as Xlite, eyeBeam to register a SIP extension (3002).

Test:

Incoming call test: Call the number of Port1 on the gateway through your phone and see if 3002 will ring. If 3002 rings, it means your configuration is successful; otherwise, it means there is something wrong with your configuration, please check the configuration.

Outgoing call test: Call your cell phone number on the 3002 extension registered in the software phone. If your phone rings, it means your configuration is OK; otherwise, please check your configuration.

 

Parent topic: AIU-8

 

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