I installed a Trixbox (last release) and A2Billing (1.3.3). I had this problem:
a device configured with freepbx can connect to a sip trunk placing the call and having correct audio
a device created by a2billing can place the call correctly (sip proto is ok) but hears no audio at all.
Sniffing the net, I saw this: when the "freepbx configured" device places the call, the sip provider (betamax) gives a 183 code with a "Connection Address" that's different from the sip address itself. Asterisk correctly sends all RTP packet to this new IP.
When, instead, a a2billing device places the call and gets the same 183 message back with the Connection Ip address, asterisk starts sending rtp packets to the same "sip server" ip address, so missing all audio frames.
Any hints?
Alfonso
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