Hi,<div>I can imagine the gravity of the task you're stuck at. Can you take a sip trace, open it up in wireshark and see the SDPs which are exchanged between the two ends from OpenSIPS server. I'm pretty sure that if you didn't change anything then it must be the network/NAT changes which resulted in no-audio at all. <br>
All I can imagine is that the two end points.. i.e Asterisk and Phone both trying to send Audio to Private subnet of other side thereby resulting in no-audio at all.</div><div><br></div><div>If your one end point had a public IP negotiated in SDP then you can expect at least one-way audio.</div>
<div><br></div><div>Take a SIP trace and reply here, maybe some other expert could comment then too.</div><div><br></div><div>Regards,</div><div>Sammy.<br><br><div class="gmail_quote">On Sat, Dec 17, 2011 at 2:43 AM, Nick Khamis <span dir="ltr"><<a href="mailto:symack@gmail.com">symack@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello Everyone,<br>
<br>
I had one way audo (out) for weeks, which was ok for our testing.<br>
Plans were to integrate RTPProxy to manage two way audo, but<br>
for now one way was ok.<br>
<br>
Not sure what the hell I changed, and now I have no audio at all.<br>
This is for an OpenSIPS -> Asterisk integration. I'm really streesed<br>
about this and not sure where to start debugging this thing anymore.<br>
<br>
Please Help,<br>
<br>
Nick.<br>
<br>
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