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

[Henk Hesselink]

Chris,

Weird indeed.  Dave Singers suggestions are excellent, maybe also check
that there are no old OpenSIPS processes hanging around (with ps axu).
I occasionally see a process that won't die and then the restart will
fail.

Cheers,

Henk Hesselink


On 08-02-11 03:14, Chris Stone 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 208.94.157.10 ...
>> Out: Contact:<sip:+13038382386 at 67.212.153.178 ...
>>
>> 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?
>
>
> Regards,
>
> Chris
>
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