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

Ovidiu Sas osas at voipembedded.com
Tue Feb 8 04:05:36 CET 2011

On Mon, Feb 7, 2011 at 9:14 PM, Chris Stone <axisml at gmail.com> wrote:
> 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?

Yes.  For sure you are using a different config.
Are you running multiple servers on that box?
Or are you having some virtual machines there?

More information about the Users mailing list