We have been using asterisk 10.5.0 for quite some time on our dedicated server.
Recently we moved to asterisk 10.6 and then to 10.7 respectively.
We are no longer to route audio in the way it should have been routed using directmedia.
Following is our sip.conf :
context = default ; Default context for incoming calls
We are using a2billing as our VOIP gateway. Directmedia seems to work absolutely fine till 10.5 but not beyond. Are we doing something wrong over here?
Please help if anybody is facing the same issue. The audio is getting disconnected after 1-2 seconds.
(Founder toRing VOIP)