On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- <ux...@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:us...@172.30.14.8>;tag=c3zqlidz1u > To: <sip:us...@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:us...@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. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users