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Digital Gateways(OIAD)

  • folder icon closed folder iconUser Manual
    • Digital Gateway L20X User Manual
    • Digital Gateway L301 User Manual
  • folder icon closed folder iconFunction Guide
    • PRI Signaling Interface Manual
    • R2 Signaling Interface Manual
    • SS7 Signaling Interface Manual
    • New Version of Digital Gateway Connects To lssabel
    • New Version of Digital Gateway Connects To 3CX
    • New Version of Digital Gateway Connects To Asterisk
    • New Version of Digital Gateway Connects To FreePBX

Digital Gateways(OIAD)

  • folder icon closed folder iconUser Manual
    • Digital Gateway L20X User Manual
    • Digital Gateway L301 User Manual
  • folder icon closed folder iconFunction Guide
    • PRI Signaling Interface Manual
    • R2 Signaling Interface Manual
    • SS7 Signaling Interface Manual
    • New Version of Digital Gateway Connects To lssabel
    • New Version of Digital Gateway Connects To 3CX
    • New Version of Digital Gateway Connects To Asterisk
    • New Version of Digital Gateway Connects To FreePBX

New Version of Digital Gateway Connects To Asterisk

Estimated reading: 2 minutes 13 views

This document provides a detailed guide on connecting the digital network gateway to Asterisk.

Please follow the steps below to configure two-way communication between the phone and the gateway.

Outgoing calls: Make a call from the Asterisk SIP extension to the gateway via the SIP trunk.

Incoming calls: Make a call from the external line to the gateway, then call Asterisk via the SIP trunk 10000, and finally have Asterisk forward the call to the SIP extension.

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Step 1. Create a SIP trunk in the Asterisk server

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

Step 2. Edit the 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. Create a SIP trunk on the digital gateway

Go to Profiles-> Select any profiles to add the SIP main server, and enter the Asterisk address.

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After successfully adding and saving, go to Trunk Settings >> SIP Trunk (as shown in the figure below, the digital gateway acts as the client; if it is a point-to-point connection, disable registration and select “No”

and enter the remote IP address in the IP Direct Routing Table)

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Dialing Test

Use SIP software (such as Xlite, eyeBeam) to register the SIP extension (3002). After completing the above steps, you can attempt to make a call.

Incoming call test: Dial the number of the network gateway port 1 using your phone to see if 3002 rings. If 3002 rings, your configuration is successful; otherwise, there is an issue with your configuration, please check.

Outgoing call test: Use the registered 3002 extension in the software phone to dial your phone number. If your mobile phone rings, it means your configuration is successful; otherwise, please check your configuration.

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