I am experiencing a strange issue.
I’m setting up a new Asterisk server. Using the latest 188.8.131.52 branch, on Ubuntu JeOS (based on 10.04 LTS), and running inside a virtual PC on VMWare ESXi. I have a test extension that plays an announcement and some music to the caller.
When using a SIP phone directly registered with the server, no problems. When calling via an IAX trunk (i.e. registering the SIP phone to my house’s Asterisk server, then IAX trunk to the newly configured server), after a random number of seconds - but usually within 30 - the audio stops and dial plan execution simply stalls. I can hang up the call and try again, same thing. IAX debug shows the ACK messages are still being processed, it’s just the audio that stalls.
It seems this is an interaction between the 1.6.2 branck of Asterisk and the virtual environment. Another user has raised a bug for this issue but the only help he got was that the Asterisk developers don’t consider it to be an Asterisk bug because the issue does not crop up on a non-virtualised server. I would beg to differ, as according to the bug’s creator, the 1.6.1 branch of Asterisk works OK. But that’s probably a losing battle.
Anyway, I was wondering if anyone else has seen this issue, or similar - where the dialplan execution stalls for no readily apparent reason and without error messages?
Similarly, has anyone had success with the 1.6.2 branck of Asterisk running inside a virtual environment?
I’m keen to stick on the 1.6.2 branch as future releases will (if I understand correctly) be built on this code and I don’t want to be limited to an older branch whose development will cease. I’d really like to get Asterisk running inside the virtual environment, too, if I possibly can.