[OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

Chris Stone axisml at gmail.com
Tue Feb 8 03:14:44 CET 2011

Sorry all for the last message - too quick on the Send button.....

On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink <opensips-users at voipro.nl> wrote:
> Hi Chris,
> That config should't touch the Contact header, and yet that's also been
> modified:
> In:  Contact:<sip:+13038382386 at ...
> Out: Contact:<sip:+13038382386 at ...
> Are you sure nothing else is touching the message?

Yes, absolutely. The packets were captured on the Opensips server - in
from upstream provider and then the next packet relaying the invite to
backend Asterisk server. The upstream provider is, of course, on a
remote network. The Asterisk server is on the same LAN - only a switch
separating the 2 servers. The only application on the server that
touched the packets would be Opensips.

So I'm not nuts - something very weird is going on here.... eh?



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