<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">This appears to be a bug in FreeSWITCH. Can you please test this on current svn trunk and if it is still a problem, please report this as a bug to <a href="http://jira.freeswitch.org">http://jira.freeswitch.org</a>.<div><br></div><div>MIke</div><div><br><div><div>On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I note that its missing the to tag from the 180 sent 5 seconds earlier (I think thats okay) but the via branch tag is also different, which seems wrong. Can anyone else chime in, I can't recall the dialog matching rules of early dialog like this.<div><br></div><div>Mike</div><div><br><div><div>On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>Here it is:</div> <div> </div> <div><a href="mailto:freeswitch@b-pbx-lab-1">freeswitch@b-pbx-lab-1</a>> recv 517 bytes from udp/[10.1.21.44]:5060 at 13:53:07.644865:<br> ------------------------------------------------------------------------<br> OPTIONS sip:<a href="http://b-pbx-lab-1.mynet.net">b-pbx-lab-1.mynet.net</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport<br> From: "Unknown" <<a href="mailto:sip%3AUnknown@10.1.21.44">sip:Unknown@10.1.21.44</a>>;tag=as5adee8f4<br> To: <sip:<a href="http://b-pbx-lab-1.mynet.net">b-pbx-lab-1.mynet.net</a>><br> Contact: <<a href="mailto:sip%3AUnknown@10.1.21.44">sip:Unknown@10.1.21.44</a>><br> Call-ID: <a href="mailto:2e6222b16df27200056f742a070f0b56@10.1.21.44">2e6222b16df27200056f742a070f0b56@10.1.21.44</a><br> CSeq: 102 OPTIONS<br> User-Agent: Asterisk PBX<br> Max-Forwards: 70<br> Date: Tue, 24 Mar 2009 13:53:07 GMT<br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br> Supported: replaces<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:<br> ------------------------------------------------------------------------<br> SIP/2.0 200 OK<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060<br> From: "Unknown" <<a href="mailto:sip%3AUnknown@10.1.21.44">sip:Unknown@10.1.21.44</a>>;tag=as5adee8f4<br> To: <sip:<a href="http://b-pbx-lab-1.mynet.net">b-pbx-lab-1.mynet.net</a>>;tag=DytraHp3K84aD<br> Call-ID: <a href="mailto:2e6222b16df27200056f742a070f0b56@10.1.21.44">2e6222b16df27200056f742a070f0b56@10.1.21.44</a><br> CSeq: 102 OPTIONS<br> Contact: <<a href="sip:10.1.21.45">sip:10.1.21.45</a>><br> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported<br> Accept: application/sdp<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: 100rel, timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:<br> ------------------------------------------------------------------------<br> INVITE <a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>><br> Contact: <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>><br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 102 INVITE<br> User-Agent: Asterisk PBX<br> Max-Forwards: 70<br> Date: Tue, 24 Mar 2009 13:53:11 GMT<br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br> Supported: replaces<br> Content-Type: application/sdp<br> Content-Length: 258</div> <div> v=0<br> o=root 4756 4756 IN IP4 10.1.21.44<br> s=session<br> c=IN IP4 10.1.21.44<br> t=0 0<br> m=audio 17956 RTP/AVP 0 8 101<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=silenceSupp:off - - - -<br> a=ptime:20<br> a=sendrecv<br> ------------------------------------------------------------------------<br>send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:<br> ------------------------------------------------------------------------<br> SIP/2.0 100 Trying<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>><br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 102 INVITE<br> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:<br> ------------------------------------------------------------------------<br> SIP/2.0 407 Proxy Authentication Required<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>>;tag=e7KHcc76gHUXr<br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 102 INVITE<br> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported<br> Accept: application/sdp<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: 100rel, timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br> Proxy-Authenticate: Digest realm="10.1.21.44", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth"<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:<br> ------------------------------------------------------------------------<br> ACK <a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>>;tag=e7KHcc76gHUXr<br> Contact: <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>><br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 102 ACK<br> User-Agent: Asterisk PBX<br> Max-Forwards: 70<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306:<br> ------------------------------------------------------------------------<br> INVITE <a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>><br> Contact: <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>><br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 103 INVITE<br> User-Agent: Asterisk PBX<br> Max-Forwards: 70<br> Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", algorithm=MD5, uri="<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", nc=00000001<br> Date: Tue, 24 Mar 2009 13:53:11 GMT<br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br> Supported: replaces<br> Content-Type: application/sdp<br> Content-Length: 258</div> <div> v=0<br> o=root 4756 4757 IN IP4 10.1.21.44<br> s=session<br> c=IN IP4 10.1.21.44<br> t=0 0<br> m=audio 17956 RTP/AVP 0 8 101<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=silenceSupp:off - - - -<br> a=ptime:20<br> a=sendrecv<br> ------------------------------------------------------------------------<br>send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526:<br> ------------------------------------------------------------------------<br> SIP/2.0 100 Trying<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>><br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 103 INVITE<br> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel <a href="mailto:sofia/internal/70904@10.1.21.44">sofia/internal/70904@10.1.21.44</a> [1d28557e-187b-11de-8c60-ad87768304bc]<br> 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing Steve->70904 in context default<br>2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/<a href="sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes">sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes</a> [1d3a376c-187b-11de-8c60-ad87768304bc]<br> send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291:<br> ------------------------------------------------------------------------<br> INVITE <a href="sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c">sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p<br> Max-Forwards: 69<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.45">sip:70904@10.1.21.45</a>>;tag=gS62F28DB372F<br> To: <<a href="sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c">sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c</a>><br> Call-ID: f4992499-931d-122c-34b1-003018ae1862<br> CSeq: 112833059 INVITE<br> Contact: <<a href="http://sip:mod_sofia@10.1.21.45:5060">sip:mod_sofia@10.1.21.45:5060</a>><br> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: 100rel, timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br> Content-Type: application/sdp<br> Content-Disposition: session<br> Content-Length: 328<br> Remote-Party-ID: "Steve" <<a href="mailto:sip%3A70904@10.1.21.45">sip:70904@10.1.21.45</a>>;screen=yes;privacy=off</div> <div> v=0<br> o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45<br> s=FreeSWITCH<br> c=IN IP4 10.1.21.45<br> t=0 0<br> m=audio 22432 RTP/AVP 0 9 8 3 101 13<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:9 G722/8000<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:3 GSM/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=rtpmap:13 CN/8000<br> a=ptime:20<br> ------------------------------------------------------------------------<br> recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589:<br> ------------------------------------------------------------------------<br> SIP/2.0 180 Ringing<br> Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p<br> Contact: <<a href="sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c">sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c</a>><br> To: <<a href="sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c">sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c</a>>;tag=fa138551<br> From: "Steve"<<a href="mailto:sip%3A70904@10.1.21.45">sip:70904@10.1.21.45</a>>;tag=gS62F28DB372F<br> Call-ID: f4992499-931d-122c-34b1-003018ae1862<br> CSeq: 112833059 INVITE<br> User-Agent: X-Lite release 1011s stamp 41150<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/<a href="sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes">sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes</a>!<br> send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060:<br> ------------------------------------------------------------------------<br> SIP/2.0 180 Ringing<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>>;tag=FgDae7QaetHgm<br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 103 INVITE<br> Contact: <<a href="sip:mod_sofia@10.1.21.45:5060;transport=udp">sip:mod_sofia@10.1.21.45:5060;transport=udp</a>><br> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported<br> Accept: application/sdp<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: 100rel, timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() Ring-Ready <a href="mailto:sofia/internal/70904@10.1.21.44">sofia/internal/70904@10.1.21.44</a>!<br> 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready <a href="mailto:sofia/internal/70904@10.1.21.44">sofia/internal/70904@10.1.21.44</a>!<br>recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013:<br> ------------------------------------------------------------------------<br> CANCEL <a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>><br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 103 CANCEL<br> User-Agent: Asterisk PBX<br> Max-Forwards: 70<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------<br>send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618:<br> ------------------------------------------------------------------------<br> SIP/2.0 481 Call/Transaction Does Not Exist<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44">sip:70904@10.1.21.44</a>>;tag=as4863e49a<br> To: <<a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net">sip:70904@b-pbx-lab-1.mynet.net</a>>;tag=FgDae7QaetHgm<br> Call-ID: <a href="mailto:4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44">4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44</a><br> CSeq: 103 CANCEL<br> Content-Length: 0</div> <div> ------------------------------------------------------------------------</div> <div><br><br> </div> <div class="gmail_quote">2009/3/24 Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span><br> <blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"> <div bgcolor="#FFFFFF"> <div>This means we could not match the cancel to a current call dialog. I would need to see the full sip trace of the call to know why, but typically this is because of not matching call Id or to or from tags</div> <div><br></div> <div>Mike <div> <div></div> <div class="h5"><br><br>On Mar 24, 2009, at 9:43 AM, Steven Ward <<a href="mailto:steve.d.ward@gmail.com" target="_blank">steve.d.ward@gmail.com</a>> wrote:<br><br></div></div></div> <div> <div></div> <div class="h5"> <div></div> <blockquote type="cite"> <div> <div>A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while the call is still ringing does not work.</div> <div> </div> <div>Why is this request resulting in a 481?</div> <div> </div> <div>I appreciate the help - I'm still just starting to learn SIP & FS. The CANCEL request and 481 response appear as follows on my FS console:</div> <div> </div> <div> </div> <div>recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:<br> ------------------------------------------------------------------------<br> CANCEL <a href="mailto:sip%3A70904@b-pbx-lab-1.mynet.net" target="_blank">sip:</a><a href="mailto:70904@b-pbx-lab-1.mynet.net" target="_blank">70904@b-pbx-lab-1.mynet.net</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44" target="_blank">sip:</a><a href="mailto:70904@10.1.21.44" target="_blank">70904@10.1.21.44</a>>;tag=as7f6965ea<br> To: <<a href="mailto:sip%3A70904@b-lab-1.mynet.net" target="_blank">sip:</a><a href="mailto:70904@b-lab-1.mynet.net" target="_blank">70904@b-lab-1.mynet.net</a>><br> Call-ID: <a href="mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44" target="_blank"></a><a href="mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44" target="_blank">237598fd102b739a03b4a4047bf69843@10.1.21.44</a><br> CSeq: 103 CANCEL<br> User-Agent: Asterisk PBX<br> Max-Forwards: 70<br> Content-Length: 0</div><p> ------------------------------------------------------------------------<br>send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:<br> ------------------------------------------------------------------------<br> SIP/2.0 481 Call/Transaction Does Not Exist<br> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060<br> From: "Steve" <<a href="mailto:sip%3A70904@10.1.21.44" target="_blank">sip:</a><a href="mailto:70904@10.1.21.44" target="_blank">70904@10.1.21.44</a>>;tag=as7f6965ea<br> To: <<a href="mailto:sip%3A70904@b-lab-1.mynet.net" target="_blank">sip:</a><a href="mailto:70904@b-lab-1.mynet.net" target="_blank">70904@b-lab-1.mynet.net</a>>;tag=71m745HKHKyjc<br> Call-ID: <a href="mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44" target="_blank"></a><a href="mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44" target="_blank">237598fd102b739a03b4a4047bf69843@10.1.21.44</a><br> CSeq: 103 CANCEL<br> Content-Length: 0</p> <div> --------------------------------------</div> <div> </div> <div> </div> <div> </div> <div>Thanks. - SW</div></div></blockquote></div></div> <blockquote type="cite"> <div><span>_______________________________________________</span><br><span>Freeswitch-users mailing list</span><br><span><a href="mailto:Freeswitch-users@lists.freeswitch.org" target="_blank">Freeswitch-users@lists.freeswitch.org</a></span><br> <span><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></span><br><span>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank"></a><a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a></span><br> <span><a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a></span><br></div></blockquote></div><br>_______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a><br><br></blockquote></div><br> _______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></div></blockquote></div><br></div></body></html>