well i have asterisk 1.4.2 - i have a intermittent problems looks like related to codecs.
My calls sometimes are connected fine, sometimes i get one way audio, sometimes the system just hangs up on me like in the following example:
Call on SIP/Euro-083862c8 left from hold
– SIP/Euro-083862c8 is making progress passing it to SIP/72269588-0837c588
– Call on SIP/Euro-083862c8 left from hold
– SIP/Euro-083862c8 answered SIP/72269588-0837c588
– AGI Script a2billing.php completed, returning 0
I’m using a2billing for billing. I disabled all the codecs except g729.
Whenever i use DIDs for calling cards or sipura adapter to make calls - i cannot get consisted g729 usage - sometimes system is using it , sometimes it is using something else. I have 10 g729 channels purchased. i have sip.conf set to g729 only, my did’s support g729 and my sipura phone set only for g729. Despite that ( show g729 ) , seems like whenever i get to use g729 on both sides - the call is fine. So that let’s me to believe asterisk is having problems with translations or …
Any ideas???
Seems like problems started when i upgraded to 1.4.2 from 1.2.14
I started having sip bridging issues and audio problems. Is anyone else having these problems???