I have 3 asterisk servers and I have several SIP trunk providers. Everything was working fine with 220.127.116.11 and other versions of 1.4 and 1.2. But as soon as I upgraded to 18.104.22.168 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 22.214.171.124 and one was 126.96.36.199. Even after upgrading both to 188.8.131.52 and setting keyrotate=no, i’m still getting garbled audio. A switch back to 184.108.40.206 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 220.127.116.11 but the destination server sending the audio is 18.104.22.168. If the destination server is 22.214.171.124 then the problem occures. Here is a current diagram of call flow:
ITSP -> (SIP) -> pbx2 -> (IAX2) -> pbx1
Currently pbx2 is 126.96.36.199 and pbx1 is 188.8.131.52. If pbx1 is 184.108.40.206 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?