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FreePBX + Didlogic Tutorial
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FreePBX + Didlogic Tutorial

agoldenbergagoldenberg Member, Provider

Does anyone have a link to a good tutorial to set up FreePBX with a DID from Didlogic.

I followed their docs but they are somewhat out of date.

Cannot make or Receive calls at the moment.

Comments

  • We do a lot of PBX hosting. Whats the issue?

    It should be fairly straightforward. In your DIDlogic panel you make sure the did is set to send to your PBX in the format [email protected] where 1234567890 is the DID number with full international and pbx.yourdomain.com is the domain of your Freepbx server. If you havent changed the default SIP port then this will be enough and if you have to 5066 for example you will need to add this [email protected]:5066

    Then in your Freepbx Admin setup in Connectivity->Trunks add a trunk that will allow DID Logic's servers to communicate with your PBX. Give it a trunk name down at the Peer Details section. They will have supplied these details which go in Peer Details.

    It will look something like this

    type=peer
    qualify=yes
    insecure=port,invite
    host=xxxx.didlogic.com
    dtmfmode=rfc2833
    disallow=all
    context=from-trunk
    allow=alaw&ulaw&g729&gsm&g723

    You may need to setup multiple trunks if DIDLogic have more than one traffic carrier server (for redundancy)

    This step allows the DIDLogic routed call on your DID to get to your PBX.

    Check your logs now and you should see call attempts reaching your PBX

    Now that the call gets there you need to handle it in Connectivity->Incoming Routes

    This is simple. Here you just need to give the route a name and specify the DID number in the field of the same name in the format above 1234567890

    At the end of the page then you tell the PBX what to do with calls on this number in Set Destination. You can send it to one Extension which would be the norm or if you want to get fancy you can send to an IVR (which you build separately in Freepbx) an have people choose options on extensions.

    Calls should now connect and you can see this detail in the Asterisk logs also.

    Hope this helps

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