It looks like interop issue with dialog matching between asterisk and freeswitch. Which version of asterisk is it? Which version of FreeSWITCH? You may want to provide a trace of the whole call starting with the invite.
FS is having trouble identifying what call asterisk wants to cancel. 2009/3/19 Steven Ward <[email protected]> > I have phones registered to a FS box, and an * box. There is a sip trunk > between the two boxes. > > A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone > while it's still ringing, this is what I get on the sip trace on FS: > > ... > > 2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 > switch_ivr_originate() Ring Ready sofia/internal/[email protected]! > recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: > ------------------------------------------------------------------------ > CANCEL > sip:[email protected]<sip%[email protected]>SIP/2.0 > Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport > From: "Steve" <sip:[email protected] <sip%[email protected]> > >;tag=as25193d44 > To: <sip:[email protected]<sip%[email protected]> > > > Call-ID: [email protected] > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 > From: "Steve" <sip:[email protected] <sip%[email protected]> > >;tag=as25193d44 > To: <sip:[email protected]<sip%[email protected]> > >;tag=c5Z8Q1e93p7KD > Call-ID: [email protected] > CSeq: 103 CANCEL > Content-Length: 0 > > -------------------------------------------------------- > > > The effect is that the FS keeps on ringing - it doesn't detect the hangup. > > > When I call from a FS phone (1000) to another FS phone (12345), and I hang > up the calling phone > while it's still ringing, this is what I get on the sip trace: > > ... > > send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: > ------------------------------------------------------------------------ > CANCEL sip:[email protected]:63054;rinstance=64e968d7a1317bc3 SIP/2.0 > Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a > Max-Forwards: 69 > From: "Extension 1000" <sip:[email protected] <sip%[email protected]> > >;tag=meK8yUgpgU2Zc > To: <sip:[email protected]:63054;rinstance=64e968d7a1317bc3> > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a > Contact: <sip:[email protected]:63054;rinstance=64e968d7a1317bc3> > To: <sip:[email protected]:63054 > ;rinstance=64e968d7a1317bc3>;tag=db12c87a > From: "Extension 1000"<sip:[email protected] <sip%[email protected]> > >;tag=meK8yUgpgU2Zc > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 CANCEL > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a > To: <sip:[email protected]:63054 > ;rinstance=64e968d7a1317bc3>;tag=db12c87a > From: "Extension 1000"<sip:[email protected] <sip%[email protected]> > >;tag=meK8yUgpgU2Zc > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 INVITE > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ... > > It works just fine. Any ideas? I'm not sure where to go with this. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
