We are trying to setup a SIP trunk between Avaya and Nortel through Cisco ASA. The SIP session seems to be successful and we can dial from Nortel to Avaya. The problem is that after dialing the phone rings but two ends can not hear each other. That is RTP (voice) communication is not working.
As part of SIP, both the PBX send the IP address and port number to connect to to establish voice session.
My assumption is that as we are NATing on ASA the TCP communication works but the original IP address and port numbers in SIP header doesn't change due to which routing breaks.
I tried using SIP inspection on ASA and that didn't work.
Any suggestions are appreciated!!