[asterisk-users] SIP response 482 "Loop Detected"

[--[ UxBoD ]--]

----- Original Message -----
> On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- <uxbod at splatnix.net>
> wrote:
> >
> > ----- Original Message -----
> >> Hi,
> >>
> >> We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
> >> that
> >> we are unable to URI dial our clients. We run a multi-tenant server
> >> and have set sip.conf to forward calls to a public context based on
> >> incoming domain name. This was all working before but not it is
> >> complaining of a loop back as the source and target server are the
> >> same.
> >>
> >> Any ideas on how to overcome this problem as we dial our clients
> >> based
> >> on their email address.
> >
> > Grabbing a SIP debug I see:
> >
> > <--- Transmitting (no NAT) to 10.172.120.5:5060 --->
> > SIP/2.0 100 Trying
> > Via: SIP/2.0/UDP
> > 10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
> > From: "User A" <sip:usera at 172.30.14.8>;tag=c3zqlidz1u
> > To: <sip:userb at seconddomain.com>
> > Call-ID: 66b3314cc6d1-jxu0nhluv4zt
> > CSeq: 2 INVITE
> > Server: secret
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> > INFO
> > Supported: replaces, timer
> > Require: timer
> > Session-Expires: 1800;refresher=uas
> > Contact: <sip:userb at 172.30.14.8>
> > Content-Length: 0
> >
> > And am guessing that as the source from IP matches the Contact:
> > address Asterisk sees that as a loop ?
> 
> I don't know these things, but you should probably post more of a SIP
> trace. Maybe turn on full sip debug to a file for long enough to see
> what the SIP conversation looks like that asterisk 1.6.2.9 is having
> with itself.
>