I have an ARI application that answer a call, then call another SIP server (not Asterisk), and creates a mixing bridge between the calls. The problem is that the RTP ports used between the bridge and the other server are not correct, resulting in one way audio. Using wireshark we see that Asterisk send audio on the same port as it receives it from the other server, instead using a different port. When calling this other server directly from an extension, all works fine.
Any idea what can cause it and how to resolve?