[Freeswitch-users] sip cancel request fails

Steven Ward steve.d.ward at gmail.com
Tue Mar 24 06:57:10 PDT 2009


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 <sip%3AUnknown at 10.1.21.44>
>;tag=as5adee8f4
   To: <sip:b-pbx-lab-1.mynet.net>
   Contact: <sip:Unknown at 10.1.21.44 <sip%3AUnknown 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 <sip%3AUnknown 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%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 at b-pbx-lab-1.mynet.net>>
   Contact: <sip:70904 at 10.1.21.44 <sip%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 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%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 at b-pbx-lab-1.mynet.net>
>;tag=e7KHcc76gHUXr
   Contact: <sip:70904 at 10.1.21.44 <sip%3A70904 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%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 at b-pbx-lab-1.mynet.net>>
   Contact: <sip:70904 at 10.1.21.44 <sip%3A70904 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<sip%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 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 <sip%3A70904 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 <sip%3A70904 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 <sip%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 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%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 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 <sip%3A70904 at 10.1.21.44>
>;tag=as4863e49a
   To: <sip:70904 at b-pbx-lab-1.mynet.net <sip%3A70904 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%3A70904 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 <sip%3A70904 at 10.1.21.44>
> >;tag=as7f6965ea
>    To: <sip:70904 at b-lab-1.mynet.net <sip%3A70904 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 <sip%3A70904 at 10.1.21.44>
> >;tag=as7f6965ea
>    To: <sip:70904 at b-lab-1.mynet.net <sip%3A70904 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
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/eca4313e/attachment-0002.html 


More information about the FreeSWITCH-users mailing list