On Tue, Jul 6, 2010 at 11:08 AM, --[ UxBoD ]-- <[email protected]> wrote: > ----- 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 >
Turn on sip debug for everything, posting just one sip packet doesn't tell much. Knowing if asterisk is sending udp packets to itself or not is a fairly important detail. I'd go with the issue tracker -- _____________________________________________________________________ -- 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
