Issue SIP Trunk


I don’t know how to configure or integrate the SIP Trunk with Freepbx. My idea is do a IVR using SIP trunk.

Please, can someone help me?


Everything is pretty well documented in the Wiki. has literally everything, from installation to troubleshooting…

  1. VoIP providers often have sample trunk configurations. Once you get it right, your asterisk sip status should show it’s registered.
  2. Create your IVR, something simple at first.
  3. Then you need a catch all (any did, any cid) inbound route which goes to a destination, such as your ivr.

I always have the inbound route go to a time condition first which determines if we are open or closed. The time condition match (or not matched) goes to one of two ivr destinations.

I agree with @dghundt about how your production system should be set up, but I get from the original posting that you aren’t close to that yet.

  1. You need to set up a “trunk” which is how you connect your SIP provider to your PBX>
  2. Set up a simple “any/any” inbound route (same as above).
  3. Finally, set up a single, simple destination for that inbound route. I like to use an announcement or a voicemail account so that I know that the connection is getting made. Using voicemail also lets you test things like NAT setting (if you can’t hear the announcement or you can’t hear the recording, your NAT settings are probably not quite right.

As you get more and more pieces working, you can move into recording your IVR announcement and setting up the destinations for the various options you want to put on your IVR. At this point, setting up a Time Condition is a terrific addition to the process.

Small steps - get one thing working at a time, and move on to the next thing.

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