----- Original Message ----- > 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. >
>From what I have read "hairpin" calls are not supported by asterisk; so am >guessing something has been fixed in the 1.6.2.X branch that should have not >worked in 1.6.1.X anyway :) While I continue the research have implemented >using a workaround via the AstDB and the following changes to the uri-dial >plan: exten => _[a-zA-Z0-9].,n,GotoIf(${DB_EXISTS(URI/${ext...@${sipdomain})}?inturi:exturi) exten => _[a-zA-Z0-9].,n(inturi),Goto(${DB(URI/${ext...@${sipdomain})}) exten => _[a-zA-Z0-9].,n(exturi),Macro(uridial,${ext...@${sipdomain}) This is a bit of pain as we have to make sure we update the DB when a new inbound URI is added; though it works and means we can stick with the 1.6.2.X branch. Would be interested to hear from a dev though as to whether they think it should work as we originally had it configured ? -- 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