[Freeswitch-users] sip cancel request fails
Michael Jerris
mike at jerris.com
Tue Mar 24 07:54:34 PDT 2009
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:
> Here it is:
>
> freeswitch at b-pbx-lab-1> 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 at 10.1.21.44>;tag=as5adee8f4
> To: <sip:b-pbx-lab-1.mynet.net>
> Contact: <sip:Unknown at 10.1.21.44>
> Call-ID: 2e6222b16df27200056f742a070f0b56 at 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 at 10.1.21.44>;tag=as5adee8f4
> To: <sip:b-pbx-lab-1.mynet.net>;tag=DytraHp3K84aD
> Call-ID: 2e6222b16df27200056f742a070f0b56 at 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 at b-pbx-lab-1.mynet.net SIP/2.0
> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
> From: "Steve" <sip:70904 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>
> Contact: <sip:70904 at 10.1.21.44>
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 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 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 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 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>;tag=e7KHcc76gHUXr
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 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 at b-pbx-lab-1.mynet.net SIP/2.0
> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
> From: "Steve" <sip:70904 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>;tag=e7KHcc76gHUXr
> Contact: <sip:70904 at 10.1.21.44>
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 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 at b-pbx-lab-1.mynet.net SIP/2.0
> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport
> From: "Steve" <sip:70904 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>
> Contact: <sip:70904 at 10.1.21.44>
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 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 at b-pbx-lab-1.mynet.net
> ", 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 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 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 at 10.1.21.44 [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 at 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 at 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 at 10.1.21.45>;tag=gS62F28DB372F
> To: <sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>
> Call-ID: f4992499-931d-122c-34b1-003018ae1862
> CSeq: 112833059 INVITE
> Contact: <sip:mod_sofia at 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 at 10.1.21.45>;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 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>
> To: <sip:
> 70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>;tag=fa138551
> From: "Steve"<sip:70904 at 10.1.21.45>;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 at 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 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>;tag=FgDae7QaetHgm
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44
> CSeq: 103 INVITE
> Contact: <sip:mod_sofia at 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 at 10.1.21.44!
> 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692
> switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44!
> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013:
>
> ------------------------------------------------------------------------
> CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0
> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport
> From: "Steve" <sip:70904 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 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 at 10.1.21.44>;tag=as4863e49a
> To: <sip:70904 at b-pbx-lab-1.mynet.net>;tag=FgDae7QaetHgm
> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44
> CSeq: 103 CANCEL
> Content-Length: 0
>
> ------------------------------------------------------------------------
>
>
>
> 2009/3/24 Michael Jerris <mike at jerris.com>
> 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 at gmail.com>
> wrote:
>
>> 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 at 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:70904 at 10.1.21.44>;tag=as7f6965ea
>> To: <sip:70904 at b-lab-1.mynet.net>
>> Call-ID: 237598fd102b739a03b4a4047bf69843 at 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 at 10.1.21.44>;tag=as7f6965ea
>> To: <sip:70904 at b-lab-1.mynet.net>;tag=71m745HKHKyjc
>> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44
>> CSeq: 103 CANCEL
>> Content-Length: 0
>>
>> --------------------------------------
>>
>>
>>
>> Thanks. - SW
>> _______________________________________________
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>
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