[Kamailio-Users] Problem with sip uri in contact field

[Geir O. Jensen]

Hm. We do rather a lot of things with the SIP uri, both reformatting the
content and changing hosts depending on the number.

However, I thought this was OK since anything that was changed would be
"record routed" and handled via the loose route portion.

I see now that that only works if the contact field is a full SIP uri (as
our regular Gateway does) because it's never been a problem.

The ACK that is sent from the SNOM gets caught by the looseroute script.
Should it?

I'm a bit confused now, can't really see how I'm going to fix this... Our
script is rather large and ugly :/

I was just under the impression that loose route meant the packets "knew"
where they were going...

- Geir


> -----Original Message-----
> From: Iñaki Baz Castillo [mailto:ibc at aliax.net] 
> Sent: 3. mars 2009 11:41
> To: Geir O. Jensen
> Cc: kamailio
> Subject: Re: [Kamailio-Users] Problem with sip uri in contact field
> 
> 2009/3/3 Geir O. Jensen <geir.o.jensen at uninett.no>:
> > As far as I can see, OpenSER get's the ACK from the phone, fails to 
> > follow the contact field and tries to send the ACK to itself.
> >
> > U 2009/03/03 11:03:58.968311 192.168.10.10:5060 -> 
> 192.168.10.1:5060 
> > ACK sip:192.168.20.1 SIP/2.0
> >
> > U 2009/03/03 11:03:58.972547 192.168.10.1:5060 -> 192.168.10.1:5060 
> > ACK sip:domain.net SIP/2.0
> 
> This is an ACK for a INVITE 200, so it's a new transaction (a 
> in-dialog request) rather than part of the INVITE transaction 
> (as an ACK for a [3456]XX response would be).
> 
> The RURI is modified by Kamailio, this means that your script 
> is doing it.
> What do you do with your in-dialog requests? what do you do 
> with ACK (INVITE-200)?
> Are you using "$ru=" or "sethostport"/"writehostport" for them?
> 
> 
> --
> Iñaki Baz Castillo
> <ibc at aliax.net>
> 
>