[Freeswitch-users] sip cancel request fails

Tue Mar 24 07:59:45 PDT 2009

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:

> 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
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>> http://www.freeswitch.org
>>
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>

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