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>
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! 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<sip%3A12345@b-pbx-sip-3.abc.xyz.net>SIP/2.0 Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport From: "Steve" <sip:54321@11.2.22.45 <sip%3A54321@11.2.22.45>
;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net>
Call-ID: 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 <sip%3A54321@11.2.22.45>
;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net> ;tag=c5Z8Q1e93p7KD Call-ID: 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 <sip%3A1000@11.2.22.46>
;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 <sip%3A1000@11.2.22.46>
;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 <sip%3A1000@11.2.22.46>
;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.
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