[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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