Problems with no audio

Hi everybody

We are telecom provider, using ASTERISK as a SIP TRUNK server for at our customers.
Among many systems, we support SWYX.

We are having some problems, when re-direction (to external destinaton) is activated on SWYX.

If a call get a voiceprompt, or just a sound, and then redirects the call, audio works great.
But if redirection, is done without any sound (ivr or similar) and the call goes direcly out to the external number, there are no audio at all.

We have had a other Asterisk server, where we made it work.
It was our first server - it was hosted/rented by a other telecom operator, but they will not give us the “config” information, as they mean it belong to them; So we are pretty f***** right now.

If we do not get the problem solved, we will loose “tons” of money.

FYI: We can make it work with OpenSips, but SWYX can not work against OpenSips directly, it must run through Asterisk.

Do any one know/are fimilar with a solution, to this problem?


There’s a full write up of the settings (including pictures) that you need to follow to set up External SIP for FreePBX. I have battled with this problem for many months but the solution here is invaluable!. … -asterisk/
IT Support London

Your configuration is not clear to me, but my immediate thought is that you are lacking a time source. Typically you need dahdi dummy loaded (unless you have real dahdi hardware) and internal timing enabled in asterisk.conf.