Hi,<div><br></div><div>I have the following setup:</div><div><br></div><div>Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints</div><div><br></div><div>With directmedia=no I can make a call between the two SIP endpoints; the RTP stream being passed through the Asterisk box.</div>
<div><br></div><div>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.</div><div><br></div><div>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.</div>
<div><br></div><div>I assume this is some sort of firewall/nat/routing issue. Could someone explain what is possibly going on and perhaps offer a solution?</div><div><br></div><div>Cheers,</div><div><br></div><div>Richard.</div>