----- 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 ? -- Thanks, Phil -- _____________________________________________________________________ -- 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
