It looks like interop issue with dialog matching between asterisk and freeswitch.<br>Which version of asterisk is it? Which version of FreeSWITCH?<br>You may want to provide a trace of the whole call starting with the invite.<br>
<br>FS is having trouble identifying what call asterisk wants to cancel.<br><br><br><div class="gmail_quote">2009/3/19 Steven Ward <span dir="ltr"><<a href="mailto:steve.d.ward@gmail.com">steve.d.ward@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><p>I have phones registered to a FS box, and an * box. There is a sip trunk between the two boxes.</p>
<p>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:</p>
<p>...</p>
<p>2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready <a href="mailto:sofia/internal/12345@11.2.22.45" target="_blank">sofia/internal/12345@11.2.22.45</a>!<br>recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950:<br>
------------------------------------------------------------------------<br> CANCEL <a href="mailto:sip%3A12345@b-pbx-sip-3.abc.xyz.net" target="_blank">sip:12345@b-pbx-sip-3.abc.xyz.net</a> SIP/2.0<br> Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport<br>
From: "Steve" <<a href="mailto:sip%3A54321@11.2.22.45" target="_blank">sip:54321@11.2.22.45</a>>;tag=as25193d44<br> To: <<a href="mailto:sip%3A12345@b-pbx-sip-3.abc.xyz.net" target="_blank">sip:12345@b-pbx-sip-3.abc.xyz.net</a>><br>
Call-ID: <a href="mailto:0c0614d866a62841546cbf3340224682@11.2.22.45" target="_blank">0c0614d866a62841546cbf3340224682@11.2.22.45</a><br> CSeq: 103 CANCEL<br> User-Agent: Asterisk PBX<br> Max-Forwards: 70<br> Content-Length: 0</p>
<p> ------------------------------------------------------------------------<br>send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572:<br> ------------------------------------------------------------------------<br>
SIP/2.0 481 Call/Transaction Does Not Exist<br> Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060<br> From: "Steve" <<a href="mailto:sip%3A54321@11.2.22.45" target="_blank">sip:54321@11.2.22.45</a>>;tag=as25193d44<br>
To: <<a href="mailto:sip%3A12345@b-pbx-sip-3.abc.xyz.net" target="_blank">sip:12345@b-pbx-sip-3.abc.xyz.net</a>>;tag=c5Z8Q1e93p7KD<br> Call-ID: <a href="mailto:0c0614d866a62841546cbf3340224682@11.2.22.45" target="_blank">0c0614d866a62841546cbf3340224682@11.2.22.45</a><br>
CSeq: 103 CANCEL<br> Content-Length: 0</p>
<p> --------------------------------------------------------<br> <br> <br>The effect is that the FS keeps on ringing - it doesn't detect the hangup.</p>
<p><br>When I call from a FS phone (1000) to another FS phone (12345), and I hang up the calling phone<br>while it's still ringing, this is what I get on the sip trace:</p>
<p>...</p>
<p>send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163:<br> ------------------------------------------------------------------------<br> CANCEL sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0<br>
Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a<br> Max-Forwards: 69<br> From: "Extension 1000" <<a href="mailto:sip%3A1000@11.2.22.46" target="_blank">sip:1000@11.2.22.46</a>>;tag=meK8yUgpgU2Zc<br>
To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3><br> Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862<br> CSeq: 112626727 CANCEL<br> Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"<br>
Content-Length: 0</p>
<p> ------------------------------------------------------------------------<br>recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863:<br> ------------------------------------------------------------------------<br>
SIP/2.0 200 OK<br> Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a<br> Contact: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3><br> To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a<br>
From: "Extension 1000"<<a href="mailto:sip%3A1000@11.2.22.46" target="_blank">sip:1000@11.2.22.46</a>>;tag=meK8yUgpgU2Zc<br> Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862<br> CSeq: 112626727 CANCEL<br>
User-Agent: X-Lite release 1011s stamp 41150<br> Content-Length: 0</p>
<p> ------------------------------------------------------------------------<br>recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334:<br> ------------------------------------------------------------------------<br>
SIP/2.0 487 Request Terminated<br> Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a<br> To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a<br> From: "Extension 1000"<<a href="mailto:sip%3A1000@11.2.22.46" target="_blank">sip:1000@11.2.22.46</a>>;tag=meK8yUgpgU2Zc<br>
Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862<br> CSeq: 112626727 INVITE<br> User-Agent: X-Lite release 1011s stamp 41150<br> Content-Length: 0</p>
<p> ...<br> <br>It works just fine. Any ideas? I'm not sure where to go with this. Thanks.</p>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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