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Digital Gateways

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  • folder icon closed folder iconFunctions Guide
    • SNMP
    • How to configure PRI signalling
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    • How to configure SS7 signalling
    • How to use Auto Provisioning
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  • folder icon closed folder iconApplication Notes
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    • Digital Gateways Connect with Elastix
    • connect E1 gateway with elastix via sip trunk
  • folder icon closed folder iconFAQ
    • SNMP Guide on Digital gateway
    • Digital Gateway connect with 3CX
    • Digital Gateway connect with Issabel
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  5. Digital Gateway connect with Asterisk16.15.1
Updated on November 23, 2022

Digital Gateways

  • folder icon closed folder iconUser Manual
    • DGW-100XR User Manual
    • DGW-L1 User Manual
    • DGW-L20X User Manual
  • folder icon closed folder iconFunctions Guide
    • SNMP
    • How to configure PRI signalling
    • How to configure MFC/R2 signalling
    • How to configure SS7 signalling
    • How to use Auto Provisioning
    • How to use DID function with OpenVox T1/E1 Gateway
  • folder icon closed folder iconApplication Notes
    • OpenVox TR069 Technote
    • Digital Gateways Connect with Elastix
    • connect E1 gateway with elastix via sip trunk
  • folder icon closed folder iconFAQ
    • SNMP Guide on Digital gateway
    • Digital Gateway connect with 3CX
    • Digital Gateway connect with Issabel
    • Digital Gateway connect with Asterisk16.15.1
    • Digital Gateway connect with Openvox UC
    • Digital Gateway connect with Freepbx 15

Digital Gateway connect with Asterisk16.15.1

Estimated reading: 2 minutes 38 views

This document mainly describes the detailed steps of connecting the digital gateway with Asterisk.

Follow the steps below to configure two-way calls between the phone and the gateway:

Outgoing call: from Asterisk SIP extension 2001 to the gateway through relay 10000;

Incoming call: call from an outside line to the gateway, through SIP trunk 10000 to Asterisk, and then send the call to the 2001 SIP extension through Asterisk;

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Step1. Create a SIP Trunk in Asterisk Server

Please add the following lines in /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 and restart SIP service in Asterisk Server

Step 2. Edit Dial Rules In Asterisk

Edit dial 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()

Step3. Create SIP Trunk on Digial Gateway.

Go to VoIP->VoIP Endpoints. Add an SIP Trunk.

After successfully adding and saving, you can see this trunk in the home page of the digital gateway and its status is OK.

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Step4. Create inbound and outbound routes on digital gateway.

Go to Routing->Call Routing Rules. Add inbound and outbound routes.

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Test call:

Taking advantage of SIP software such as Xlite, eyeBeam to register a SIP extension(2001). After all above steps, you can try to make calls.

Incoming call test: Dial the number of port 1 on the gateway via your mobile to see if 2001 will ring. If 2001 rings, it means your configuration is successful; unless, it means there is something wrong with your configuration, please check it.
Outgoing call test: Dial your mobile number on 2001 extension registered in a software phone. If your mobile rings, it means your configuration is ok; unless, please check your configuration.

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