IAX2 issues with

I have 3 asterisk servers and I have several SIP trunk providers. Everything was working fine with and other versions of 1.4 and 1.2. But as soon as I upgraded to I started having garbled calls when the call came from a SIP provider then through IAX2 to another server. I thought at first it was because of keyrotate and because one server was and one was Even after upgrading both to and setting keyrotate=no, i’m still getting garbled audio. A switch back to and everything is fine again. I’m using the exact same config files. Calls coming in over SIP directly to the server work fine. Its only when it is relayed through IAX2 that it gets garbled.

Here is some new information about this problem. It works fine when the first server is but the destination server sending the audio is If the destination server is then the problem occures. Here is a current diagram of call flow:

ITSP -> (SIP) -> pbx2 -> (IAX2) -> pbx1

Currently pbx2 is and pbx1 is If pbx1 is I get garbled audio. It does not seem to matter what version of asterisk pbx2 is running.

Does anyone have any ideas how to fix this?


You will find that the resources available through this forum are primarily configuration, troubleshooting, and vendor experts… but few around here are experts on the inner workings of astersk and debugging at the source code/channel driver level.

I would open a bug report at bugs.digium.com. I believe I read another post on here yesterday complaining of a very similar problem when calling Voicemail through an IAX trunk.


Just make sure you read the bug submission guidelines, otherwise you will wait a week + only to have your bug dismissed temporarily because you didn’t provide all the required information.

Good luck!!

here is the other thread… you probably want to get with this guy and open a report together. (assuming he is running the same version of 1.6)