----- Original Message ----- > ----- Original Message ----- > > On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- <[email protected]> > > 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:[email protected]>;tag=c3zqlidz1u > > > To: <sip:[email protected]> > > > 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:[email protected]> > > > 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 ?
Do you think this should be raised as a issue in bugtraq or at least brought up on the asterisk-dev mailing list ? -- 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
