Hi, I have the following setup:
Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints With directmedia=no I can make a call between the two SIP endpoints; the RTP stream being passed through the Asterisk box. Obviously, this is sub-optimal. I attempted to enable bridging of the call between the 2 endpoints directly, given that they are on the same non-routeable private net. With directmedia=nonat, I see Asterisk report the bridging of the calls but both sides of the call are routed to the originating endpoint so effectively, the call becomes an echo-loop. There is no audio on the second end-point although the call remains up. I assume this is some sort of firewall/nat/routing issue. Could someone explain what is possibly going on and perhaps offer a solution? Cheers, Richard.
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