I have problems to bypass Asterisk with RTP Stream.
In this scenario I need asterisk just as a SIP signaling relay between two different domains without handling SDP protocol.
The asterisk server is also behind an Application Layer Gatreway which performs all NAT/Traversal Operation.
The scnario is
We should use asterisk just to interpret the sent digit in order to redirect to the right ALG IP address.
What happens is that after Asterisk gets the invite from ALG it puts in the SDP part its pown address,so in this way also the RTP is addressed to asterisk.
In this way it performs also media proxy manipulation instead I would like to configure just for SIP without any manipulation.
Is it possible to this on Asterisk? My release is 1.2.24
I was looking a lot before posting but I acould not find any answer.
Thanks in advance