Possible Codec problem when calling with callfile

I’ve got a system that uses an ARI script to dial a local phone number and, when the local person picks up, dials the remote phone. This script, originally written by @wardmundy about 15 years ago has been working flawlessly for me until last week.

I had to rebuild the PBX server and now, occasionally, when we make an outbound call, the recipient at the remote end will pick up the call and our end will “hang up” the call immediately. The sketchy part is that it only happens once in every 10+ calls. Now, we did restore the system from an old backup, which restored the Chan-SIP configuration (which was working on another server literally the day before).

I’m thinking this is a codec mismatch problem (the logs imply that as well) but I have both the trunk and the extensions set to try a full array of codecs, including the usual suspects (ulaw and alaw). The tricky part is, though, that when we hand dial the call, it always connects. This leads me to believe that if there’s a codec list on the ARI interface, it’s getting in the way and establishing the call to a codec we don’t use. The catch there is I’m pretty sure there’s no such thing as an ARI Codec list. It should be using the codec list from the trunk, but that should match with the remote phone (since everything is running through the PSTN) and when we hand dial the call, it goes through.

I’ve been through the logs a thousand times and I just can’t get a handle on it. I’m open to suggestions:

  • Would changing the trunks to PJ-SIP make any difference?
  • Does ARI get a vote in the call’s codec?
  • Is there something in Asterisk 15 that sounds familiar?

If it is a codec issue, I would expect it to show in the call trace with an error message. Can you pastebin a call trace for us?

I’ll try to capture one today.

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