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[Freeswitch-users] sip cancel request fails


 
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steve.d.ward at gmail.com
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PostPosted: Tue Mar 24, 2009 8:50 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

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.
 
Why is this request resulting in a 481?
 
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:
 
 
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   ------------------------------------------------------------------------
   CANCEL sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as7f6965ea
   To: <sip:70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])>
   Call-ID: 237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as7f6965ea
   To: <sip:70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])>;tag=71m745HKHKyjc
   Call-ID: 237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
   CSeq: 103 CANCEL
   Content-Length: 0    --------------------------------------
 
 
 
Thanks.  - SW
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steve.d.ward at gmail.com
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PostPosted: Tue Mar 24, 2009 8:50 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

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.
 
Why is this request resulting in a 481?
 
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:
 
 
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   ------------------------------------------------------------------------
   CANCEL sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as7f6965ea
   To: <sip:70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])>
   Call-ID: 237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as7f6965ea
   To: <sip:70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])>;tag=71m745HKHKyjc
   Call-ID: 237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
   CSeq: 103 CANCEL
   Content-Length: 0    --------------------------------------
 
 
 
Thanks.  - SW
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mike at jerris.com
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PostPosted: Tue Mar 24, 2009 8:55 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

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


Mike

On Mar 24, 2009, at 9:43 AM, Steven Ward <steve.d.ward@gmail.com (steve.d.ward@gmail.com)> wrote:



Quote:
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.

Why is this request resulting in a 481?

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:


recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
------------------------------------------------------------------------
CANCEL sip:[url=mailto:70904@b-pbx-lab-1.mynet.net]70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])[/url] SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
From: "Steve" <sip:[url=mailto:70904@10.1.21.44]70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])[/url]>;tag=as7f6965ea
To: <sip:[url=mailto:70904@b-lab-1.mynet.net]70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])[/url]>
Call-ID: [url=mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44]237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)[/url]
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
------------------------------------------------------------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
From: "Steve" <sip:[url=mailto:70904@10.1.21.44]70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])[/url]>;tag=as7f6965ea
To: <sip:[url=mailto:70904@b-lab-1.mynet.net]70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])[/url]>;tag=71m745HKHKyjc
Call-ID: [url=mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44]237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)[/url]
CSeq: 103 CANCEL
Content-Length: 0 --------------------------------------



Thanks. - SW

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steve.d.ward at gmail.com
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PostPosted: Tue Mar 24, 2009 9:10 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

Here it is:
 
freeswitch@b-pbx-lab-1 ([email]freeswitch@b-pbx-lab-1[/email])> recv 517 bytes from udp/[10.1.21.44]:5060 at 13:53:07.644865:
   ------------------------------------------------------------------------
   OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
   From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
   To: <sip:b-pbx-lab-1.mynet.net>
   Contact: <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>
   Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
   CSeq: 102 OPTIONS
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:07 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Length: 0
   ------------------------------------------------------------------------
send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
   From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
   To: <sip:b-pbx-lab-1.mynet.net>;tag=DytraHp3K84aD
   Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
   CSeq: 102 OPTIONS
   Contact: <sip:10.1.21.45>
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   ------------------------------------------------------------------------
recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:
   ------------------------------------------------------------------------
   INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4756 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0
   ------------------------------------------------------------------------
send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="10.1.21.44", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth"
   Content-Length: 0
   ------------------------------------------------------------------------
recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:
   ------------------------------------------------------------------------
   ACK sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
   Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 ACK
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0
   ------------------------------------------------------------------------
recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306:
   ------------------------------------------------------------------------
   INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", algorithm=MD5, uri="sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", nc=00000001
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4757 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0
   ------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email]) [1d28557e-187b-11de-8c60-ad87768304bc]
2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing Steve->70904 in context default
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes [1d3a376c-187b-11de-8c60-ad87768304bc]
send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291:
   ------------------------------------------------------------------------
   INVITE sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p
   Max-Forwards: 69
   From: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
   To: <sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>
   Call-ID: f4992499-931d-122c-34b1-003018ae1862
   CSeq: 112833059 INVITE
   Contact: <sip:mod_sofia@10.1.21.45:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 328
   Remote-Party-ID: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;screen=yes;privacy=off
   v=0
   o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45
   s=FreeSWITCH
   c=IN IP4 10.1.21.45
   t=0 0
   m=audio 22432 RTP/AVP 0 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   ------------------------------------------------------------------------
recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p
   Contact: <sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>
   To: <sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>;tag=fa138551
   From: "Steve"<sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
   Call-ID: f4992499-931d-122c-34b1-003018ae1862
   CSeq: 112833059 INVITE
   User-Agent: X-Lite release 1011s stamp 41150
   Content-Length: 0
   ------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes!
send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 INVITE
   Contact: <sip:mod_sofia@10.1.21.45:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   ------------------------------------------------------------------------
2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() Ring-Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013:
   ------------------------------------------------------------------------
   CANCEL sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0
   ------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 CANCEL
   Content-Length: 0
   ------------------------------------------------------------------------


 
2009/3/24 Michael Jerris <mike@jerris.com (mike@jerris.com)>
Quote:
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


Mike


On Mar 24, 2009, at 9:43 AM, Steven Ward <steve.d.ward@gmail.com (steve.d.ward@gmail.com)> wrote:






Quote:
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.
 
Why is this request resulting in a 481?
 
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:
 
 
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   ------------------------------------------------------------------------
   CANCEL sip:[url=mailto:70904@b-pbx-lab-1.mynet.net]70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])[/url] SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: "Steve" <sip:[url=mailto:70904@10.1.21.44]70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])[/url]>;tag=as7f6965ea
   To: <sip:[url=mailto:70904@b-lab-1.mynet.net]70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])[/url]>
   Call-ID: [url=mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44]237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)[/url]
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: "Steve" <sip:[url=mailto:70904@10.1.21.44]70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])[/url]>;tag=as7f6965ea
   To: <sip:[url=mailto:70904@b-lab-1.mynet.net]70904@b-lab-1.mynet.net ([email]sip%3A70904@b-lab-1.mynet.net[/email])[/url]>;tag=71m745HKHKyjc
   Call-ID: [url=mailto:237598fd102b739a03b4a4047bf69843@10.1.21.44]237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)[/url]
   CSeq: 103 CANCEL
   Content-Length: 0    --------------------------------------
 
 
 
Thanks.  - SW

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mike at jerris.com
Guest





PostPosted: Tue Mar 24, 2009 10:09 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

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.

Mike

On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:
Quote:
Here it is:

freeswitch@b-pbx-lab-1 ([email]freeswitch@b-pbx-lab-1[/email])> recv 517 bytes from udp/[10.1.21.44]:5060 at 13:53:07.644865:
------------------------------------------------------------------------
OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
To: <sip:b-pbx-lab-1.mynet.net>
Contact: <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>
Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Mar 2009 13:53:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
------------------------------------------------------------------------
send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
To: <sip:b-pbx-lab-1.mynet.net>;tag=DytraHp3K84aD
Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
CSeq: 102 OPTIONS
Contact: <[url=sip:10.1.21.45]sip:10.1.21.45[/url]>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
------------------------------------------------------------------------
recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:
------------------------------------------------------------------------
INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Mar 2009 13:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 4756 4756 IN IP4 10.1.21.44
s=session
c=IN IP4 10.1.21.44
t=0 0
m=audio 17956 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Content-Length: 0
------------------------------------------------------------------------
send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:
------------------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="10.1.21.44", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:
------------------------------------------------------------------------
ACK sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306:
------------------------------------------------------------------------
INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", algorithm=MD5, uri="sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", nc=00000001
Date: Tue, 24 Mar 2009 13:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 4756 4757 IN IP4 10.1.21.44
s=session
c=IN IP4 10.1.21.44
t=0 0
m=audio 17956 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Content-Length: 0
------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email]) [1d28557e-187b-11de-8c60-ad87768304bc]
2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing Steve->70904 in context default
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes[/url] [1d3a376c-187b-11de-8c60-ad87768304bc]
send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291:
------------------------------------------------------------------------
INVITE [url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url] SIP/2.0
Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p
Max-Forwards: 69
From: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
To: <[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url]>
Call-ID: f4992499-931d-122c-34b1-003018ae1862
CSeq: 112833059 INVITE
Contact: <sip:mod_sofia@10.1.21.45:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 328
Remote-Party-ID: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;screen=yes;privacy=off
v=0
o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45
s=FreeSWITCH
c=IN IP4 10.1.21.45
t=0 0
m=audio 22432 RTP/AVP 0 9 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
------------------------------------------------------------------------
recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589:
------------------------------------------------------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p
Contact: <[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url]>
To: <[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url]>;tag=fa138551
From: "Steve"<sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
Call-ID: f4992499-931d-122c-34b1-003018ae1862
CSeq: 112833059 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes[/url]!
send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060:
------------------------------------------------------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 INVITE
Contact: <[url=sip:mod_sofia@10.1.21.45:5060;transport=udp]sip:mod_sofia@10.1.21.45:5060;transport=udp[/url]>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
------------------------------------------------------------------------
2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() Ring-Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013:
------------------------------------------------------------------------
CANCEL sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618:
------------------------------------------------------------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 CANCEL
Content-Length: 0
------------------------------------------------------------------------



2009/3/24 Michael Jerris <mike@jerris.com (mike@jerris.com)>
Quote:
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


Mike


On Mar 24, 2009, at 9:43 AM, Steven Ward <steve.d.ward@gmail.com (steve.d.ward@gmail.com)> wrote:






Quote:
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.

Why is this request resulting in a 481?

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:


recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
------------------------------------------------------------------------
CANCEL sip: ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])70904@b-pbx-lab-1.mynet.net (70904@b-pbx-lab-1.mynet.net) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
From: "Steve" <sip: ([email]sip%3A70904@10.1.21.44[/email])70904@10.1.21.44 (70904@10.1.21.44)>;tag=as7f6965ea
To: <sip: ([email]sip%3A70904@b-lab-1.mynet.net[/email])70904@b-lab-1.mynet.net (70904@b-lab-1.mynet.net)>
Call-ID: (237598fd102b739a03b4a4047bf69843@10.1.21.44)237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
------------------------------------------------------------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
From: "Steve" <sip: ([email]sip%3A70904@10.1.21.44[/email])70904@10.1.21.44 (70904@10.1.21.44)>;tag=as7f6965ea
To: <sip: ([email]sip%3A70904@b-lab-1.mynet.net[/email])70904@b-lab-1.mynet.net (70904@b-lab-1.mynet.net)>;tag=71m745HKHKyjc
Call-ID: (237598fd102b739a03b4a4047bf69843@10.1.21.44)237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
CSeq: 103 CANCEL
Content-Length: 0 --------------------------------------



Thanks. - SW

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mike at jerris.com
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PostPosted: Tue Mar 24, 2009 10:12 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

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 http://jira.freeswitch.org.

MIke

On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote:
Quote:
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.

Mike

On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:
Quote:
Here it is:

freeswitch@b-pbx-lab-1 ([email]freeswitch@b-pbx-lab-1[/email])> recv 517 bytes from udp/[10.1.21.44]:5060 at 13:53:07.644865:
------------------------------------------------------------------------
OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
To: <sip:b-pbx-lab-1.mynet.net>
Contact: <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>
Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Mar 2009 13:53:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
------------------------------------------------------------------------
send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
To: <sip:b-pbx-lab-1.mynet.net>;tag=DytraHp3K84aD
Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
CSeq: 102 OPTIONS
Contact: <[url=sip:10.1.21.45]sip:10.1.21.45[/url]>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
------------------------------------------------------------------------
recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:
------------------------------------------------------------------------
INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Mar 2009 13:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 4756 4756 IN IP4 10.1.21.44
s=session
c=IN IP4 10.1.21.44
t=0 0
m=audio 17956 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Content-Length: 0
------------------------------------------------------------------------
send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:
------------------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="10.1.21.44", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:
------------------------------------------------------------------------
ACK sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306:
------------------------------------------------------------------------
INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", algorithm=MD5, uri="sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", nc=00000001
Date: Tue, 24 Mar 2009 13:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 4756 4757 IN IP4 10.1.21.44
s=session
c=IN IP4 10.1.21.44
t=0 0
m=audio 17956 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Content-Length: 0
------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email]) [1d28557e-187b-11de-8c60-ad87768304bc]
2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing Steve->70904 in context default
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes[/url] [1d3a376c-187b-11de-8c60-ad87768304bc]
send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291:
------------------------------------------------------------------------
INVITE [url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url] SIP/2.0
Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p
Max-Forwards: 69
From: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
To: <[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url]>
Call-ID: f4992499-931d-122c-34b1-003018ae1862
CSeq: 112833059 INVITE
Contact: <sip:mod_sofia@10.1.21.45:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 328
Remote-Party-ID: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;screen=yes;privacy=off
v=0
o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45
s=FreeSWITCH
c=IN IP4 10.1.21.45
t=0 0
m=audio 22432 RTP/AVP 0 9 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
------------------------------------------------------------------------
recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589:
------------------------------------------------------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p
Contact: <[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url]>
To: <[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c[/url]>;tag=fa138551
From: "Steve"<sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
Call-ID: f4992499-931d-122c-34b1-003018ae1862
CSeq: 112833059 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/[url=sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes]sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes[/url]!
send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060:
------------------------------------------------------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 INVITE
Contact: <[url=sip:mod_sofia@10.1.21.45:5060;transport=udp]sip:mod_sofia@10.1.21.45:5060;transport=udp[/url]>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
------------------------------------------------------------------------
2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() Ring-Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013:
------------------------------------------------------------------------
CANCEL sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618:
------------------------------------------------------------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060
From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
CSeq: 103 CANCEL
Content-Length: 0
------------------------------------------------------------------------



2009/3/24 Michael Jerris <mike@jerris.com (mike@jerris.com)>
Quote:
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


Mike


On Mar 24, 2009, at 9:43 AM, Steven Ward <steve.d.ward@gmail.com (steve.d.ward@gmail.com)> wrote:






Quote:
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.

Why is this request resulting in a 481?

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:


recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
------------------------------------------------------------------------
CANCEL sip: ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])70904@b-pbx-lab-1.mynet.net (70904@b-pbx-lab-1.mynet.net) SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
From: "Steve" <sip: ([email]sip%3A70904@10.1.21.44[/email])70904@10.1.21.44 (70904@10.1.21.44)>;tag=as7f6965ea
To: <sip: ([email]sip%3A70904@b-lab-1.mynet.net[/email])70904@b-lab-1.mynet.net (70904@b-lab-1.mynet.net)>
Call-ID: (237598fd102b739a03b4a4047bf69843@10.1.21.44)237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
------------------------------------------------------------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
From: "Steve" <sip: ([email]sip%3A70904@10.1.21.44[/email])70904@10.1.21.44 (70904@10.1.21.44)>;tag=as7f6965ea
To: <sip: ([email]sip%3A70904@b-lab-1.mynet.net[/email])70904@b-lab-1.mynet.net (70904@b-lab-1.mynet.net)>;tag=71m745HKHKyjc
Call-ID: (237598fd102b739a03b4a4047bf69843@10.1.21.44)237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
CSeq: 103 CANCEL
Content-Length: 0 --------------------------------------



Thanks. - SW

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steve.d.ward at gmail.com
Guest





PostPosted: Tue Mar 24, 2009 10:19 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

Mike,
 
Thanks for taking the time to look at this - I appreciate it.  I'll go ahead and test it out on the current svn trunk.
 
- Steve


2009/3/24 Michael Jerris <mike@jerris.com (mike@jerris.com)>
Quote:
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 http://jira.freeswitch.org.

MIke


On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote:

Quote:
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.

Mike

On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:

Quote:
Here it is:
 
freeswitch@b-pbx-lab-1 ([email]freeswitch@b-pbx-lab-1[/email])> recv 517 bytes from udp/[10.1.21.44]:5060 at 13:53:07.644865:
   ------------------------------------------------------------------------
   OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
   From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
   To: <sip:b-pbx-lab-1.mynet.net>
   Contact: <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>
   Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
   CSeq: 102 OPTIONS
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:07 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Length: 0
   ------------------------------------------------------------------------
send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
   From: "Unknown" <sip:Unknown@10.1.21.44 ([email]sip%3AUnknown@10.1.21.44[/email])>;tag=as5adee8f4
   To: <sip:b-pbx-lab-1.mynet.net>;tag=DytraHp3K84aD
   Call-ID: 2e6222b16df27200056f742a070f0b56@10.1.21.44 (2e6222b16df27200056f742a070f0b56@10.1.21.44)
   CSeq: 102 OPTIONS
   Contact: <sip:10.1.21.45>
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   ------------------------------------------------------------------------
recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:
   ------------------------------------------------------------------------
   INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4756 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0
   ------------------------------------------------------------------------
send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="10.1.21.44", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth"
   Content-Length: 0
   ------------------------------------------------------------------------
recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:
   ------------------------------------------------------------------------
   ACK sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=e7KHcc76gHUXr
   Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 102 ACK
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0
   ------------------------------------------------------------------------
recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306:
   ------------------------------------------------------------------------
   INVITE sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Contact: <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", algorithm=MD5, uri="sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", nc=00000001
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4757 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0
   ------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email]) [1d28557e-187b-11de-8c60-ad87768304bc]
2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing Steve->70904 in context default
2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes [1d3a376c-187b-11de-8c60-ad87768304bc]
send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291:
   ------------------------------------------------------------------------
   INVITE sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p
   Max-Forwards: 69
   From: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
   To: <sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>
   Call-ID: f4992499-931d-122c-34b1-003018ae1862
   CSeq: 112833059 INVITE
   Contact: <sip:mod_sofia@10.1.21.45:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 328
   Remote-Party-ID: "Steve" <sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;screen=yes;privacy=off
   v=0
   o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45
   s=FreeSWITCH
   c=IN IP4 10.1.21.45
   t=0 0
   m=audio 22432 RTP/AVP 0 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   ------------------------------------------------------------------------
recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p
   Contact: <sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>
   To: <sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>;tag=fa138551
   From: "Steve"<sip:70904@10.1.21.45 ([email]sip%3A70904@10.1.21.45[/email])>;tag=gS62F28DB372F
   Call-ID: f4992499-931d-122c-34b1-003018ae1862
   CSeq: 112833059 INVITE
   User-Agent: X-Lite release 1011s stamp 41150
   Content-Length: 0
   ------------------------------------------------------------------------
2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:70904@10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes!
send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 INVITE
   Contact: <sip:mod_sofia@10.1.21.45:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   ------------------------------------------------------------------------
2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() Ring-Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/70904@10.1.21.44 ([email]sofia/internal/70904@10.1.21.44[/email])!
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013:
   ------------------------------------------------------------------------
   CANCEL sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0
   ------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060
   From: "Steve" <sip:70904@10.1.21.44 ([email]sip%3A70904@10.1.21.44[/email])>;tag=as4863e49a
   To: <sip:70904@b-pbx-lab-1.mynet.net ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])>;tag=FgDae7QaetHgm
   Call-ID: 4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44 (4db5b31f3f9d99c436804e4b54277b3e@10.1.21.44)
   CSeq: 103 CANCEL
   Content-Length: 0
   ------------------------------------------------------------------------


 
2009/3/24 Michael Jerris <mike@jerris.com (mike@jerris.com)>
Quote:
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


Mike


On Mar 24, 2009, at 9:43 AM, Steven Ward <steve.d.ward@gmail.com (steve.d.ward@gmail.com)> wrote:






Quote:
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.
 
Why is this request resulting in a 481?
 
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:
 
 
recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   ------------------------------------------------------------------------
   CANCEL sip: ([email]sip%3A70904@b-pbx-lab-1.mynet.net[/email])70904@b-pbx-lab-1.mynet.net (70904@b-pbx-lab-1.mynet.net) SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: "Steve" <sip: ([email]sip%3A70904@10.1.21.44[/email])70904@10.1.21.44 (70904@10.1.21.44)>;tag=as7f6965ea
   To: <sip: ([email]sip%3A70904@b-lab-1.mynet.net[/email])70904@b-lab-1.mynet.net (70904@b-lab-1.mynet.net)>
   Call-ID: (237598fd102b739a03b4a4047bf69843@10.1.21.44)237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: "Steve" <sip: ([email]sip%3A70904@10.1.21.44[/email])70904@10.1.21.44 (70904@10.1.21.44)>;tag=as7f6965ea
   To: <sip: ([email]sip%3A70904@b-lab-1.mynet.net[/email])70904@b-lab-1.mynet.net (70904@b-lab-1.mynet.net)>;tag=71m745HKHKyjc
   Call-ID: (237598fd102b739a03b4a4047bf69843@10.1.21.44)237598fd102b739a03b4a4047bf69843@10.1.21.44 (237598fd102b739a03b4a4047bf69843@10.1.21.44)
   CSeq: 103 CANCEL
   Content-Length: 0    --------------------------------------
 
 
 
Thanks.  - SW

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woof at nortel.com
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PostPosted: Tue Mar 24, 2009 11:47 am    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

Woof!

Appears to be a recently fixed * bug:

0014431: Bad branch parameter value in CANCEL request
http://bugs.digium.com/view.php?id=14431

--Woof!




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steve.d.ward at gmail.com
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PostPosted: Tue Mar 24, 2009 1:57 pm    Post subject: [Freeswitch-users] sip cancel request fails Reply with quote

Ah ha!
 
Thanks for finding that.
 
I updated my * server and I'm all set.
 
Many thanks for all the feedback and help.


2009/3/24 Andy Spitzer <woof@nortel.com (woof@nortel.com)>
Quote:
Woof!

Appears to be a recently fixed * bug:

0014431: Bad branch parameter value in CANCEL request
http://bugs.digium.com/view.php?id=14431

--Woof!





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