[OpenSIPS-Users] Totally Stunned about this No Audio Going Out

[Sammy Govind]

Hi,
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.
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.

If your one end point had a public IP negotiated in SDP then you can expect
at least one-way audio.

Take a SIP trace and reply here, maybe some other expert could comment then
too.

Regards,
Sammy.

On Sat, Dec 17, 2011 at 2:43 AM, Nick Khamis <symack at gmail.com> wrote:

> Hello Everyone,
>
> I had one way audo (out) for weeks, which was ok for our testing.
> Plans were to integrate RTPProxy to manage two way audo, but
> for now one way was ok.
>
> Not sure what the hell I changed, and now I have no audio at all.
> This is for an OpenSIPS -> Asterisk integration. I'm really streesed
> about this and not sure where to start debugging this thing anymore.
>
> Please Help,
>
> Nick.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20111219/02fad3f2/attachment.htm>