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[steve.d.ward at gmail.com]

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/12345@11.2.22.45 ([email]sofia/internal/12345@11.2.22.45[/email])!
recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950:
   ------------------------------------------------------------------------
   CANCEL sip:12345@b-pbx-sip-3.abc.xyz.net ([email]sip%3A12345@b-pbx-sip-3.abc.xyz.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport
   From: "Steve" <sip:54321@11.2.22.45 ([email]sip%3A54321@11.2.22.45[/email])>;tag=as25193d44
   To: <sip:12345@b-pbx-sip-3.abc.xyz.net ([email]sip%3A12345@b-pbx-sip-3.abc.xyz.net[/email])>
   Call-ID: 0c0614d866a62841546cbf3340224682@11.2.22.45 (0c0614d866a62841546cbf3340224682@11.2.22.45)
   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:54321@11.2.22.45 ([email]sip%3A54321@11.2.22.45[/email])>;tag=as25193d44
   To: <sip:12345@b-pbx-sip-3.abc.xyz.net ([email]sip%3A12345@b-pbx-sip-3.abc.xyz.net[/email])>;tag=c5Z8Q1e93p7KD
   Call-ID: 0c0614d866a62841546cbf3340224682@11.2.22.45 (0c0614d866a62841546cbf3340224682@11.2.22.45)
   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:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0
   Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a
   Max-Forwards: 69
   From: "Extension 1000" <sip:1000@11.2.22.46 ([email]sip%3A1000@11.2.22.46[/email])>;tag=meK8yUgpgU2Zc
   To: <sip:12345@11.2.56.106:63054;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:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>
   To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a
   From: "Extension 1000"<sip:1000@11.2.22.46 ([email]sip%3A1000@11.2.22.46[/email])>;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:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a
   From: "Extension 1000"<sip:1000@11.2.22.46 ([email]sip%3A1000@11.2.22.46[/email])>;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.

[anthony.minessale at g...]

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 <steve.d.ward@gmail.com (steve.d.ward@gmail.com)>