I have 3 asterisk servers and I have several SIP trunk providers. Everything was working fine with 1.6.0.9 and other versions of 1.4 and 1.2. But as soon as I upgraded to 1.6.1.0 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 1.6.0.9 and one was 1.6.1.0. Even after upgrading both to 1.6.1.0 and setting keyrotate=no, i’m still getting garbled audio. A switch back to 1.6.0.9 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 1.6.1.0 but the destination server sending the audio is 1.6.0.9. If the destination server is 1.6.1.0 then the problem occures. Here is a current diagram of call flow:
ITSP -> (SIP) -> pbx2 -> (IAX2) -> pbx1
Currently pbx2 is 1.6.1.0 and pbx1 is 1.6.0.9. If pbx1 is 1.6.1.0 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?
Thanks,
Michael