[Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls

Wed Apr 1 23:09:56 PDT 2009

I probably shouldn't be doing this for you, but...

http://bugs.digium.com/view.php?id=14431

;)

On Wed, Apr 1, 2009 at 7:41 PM, Alfonso Pinto <elhodred at gmail.com> wrote:
> I've searched in google about it and only found a message about the
> same, Anthony asked for more information and nobody answer.
>
> I've tried with an IP phone (aastra 57i) and the same happens.
>
> Thank you
>
> 2009/4/2 Brian West <brian at freeswitch.org>:
>> I'm pretty sure this is a bug in Asterisk something to do with dialog
>> matching... I think if you search the archives you'll see about it.
>> /b
>> On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote:
>>
>> Hi guys,
>>
>> I've using asterisk as PSTN gateway. When a call arrives from PSTN, I
>> send the call to freeswitch and this route the call to a SIP gateway.
>>
>> When caller cancels the  call (hangups before callee answers), I get
>> this on asterisk CLI:
>>
>> chan_sip.c:13056 handle_response: Remote host can't match request
>> CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up.
>>
>> I'm using asterisk 1.4.23.1 and freeswitch 1.0.3
>>
>> This is the sip call flow:
>>
>> u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060
>> INVITE sip:666666666 at 1.1.1.1 SIP/2.0.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 102 INVITE.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Date: Wed, 01 Apr 2009 21:03:12 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>> Supported: replaces.
>> Content-Type: application/sdp.
>> Content-Length: 265.
>> .
>> v=0.
>> o=root 29347 29347 IN IP4 2.2.2.2.
>> s=session.
>> c=IN IP4 2.2.2.2.
>> t=0 0.
>> m=audio 13846 RTP/AVP 18 101.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=silenceSupp:off - - - -.
>> a=ptime:20.
>> a=sendrecv.
>>
>>
>> U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 102 INVITE.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060
>> SIP/2.0 407 Proxy Authentication Required.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 102 INVITE.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, presence, dialog, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer.
>> Proxy-Authenticate: Digest realm="1.1.1.1",
>> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5,
>> qop="auth".
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060
>> ACK sip:666666666 at 1.1.1.1 SIP/2.0.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 102 ACK.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060
>> INVITE sip:666666666 at 1.1.1.1 SIP/2.0.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 INVITE.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1",
>> algorithm=MD5, uri="sip:666666666 at 1.1.1.1",
>> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7",
>> response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth,
>> cnonce="47efcad4", nc=00000001.
>> Date: Wed, 01 Apr 2009 21:03:12 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>> Supported: replaces.
>> Content-Type: application/sdp.
>> Content-Length: 265.
>> .
>> v=0.
>> o=root 29347 29348 IN IP4 2.2.2.2.
>> s=session.
>> c=IN IP4 2.2.2.2.
>> t=0 0.
>> m=audio 13846 RTP/AVP 18 101.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=silenceSupp:off - - - -.
>> a=ptime:20.
>> a=sendrecv.
>>
>>
>> U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 INVITE.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060
>> INVITE sip:666666666 at 3.3.3.3 SIP/2.0.
>> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB.
>> Max-Forwards: 69.
>> From: "999999999" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> CSeq: 113193247 INVITE.
>> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>> Supported: 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: 387.
>> Remote-Party-ID: "999999999" <sip:999999999 at 3.3.3.3>;screen=yes;privacy=off.
>> .
>> v=0.
>> o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1.
>> s=FreeSWITCH.
>> c=IN IP4 1.1.1.1.
>> t=0 0.
>> m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13.
>> a=rtpmap:18 G729/8000.
>> a=rtpmap:4 G723/8000.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:9 G722/8000.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=rtpmap:13 CN/8000.
>> a=ptime:20.
>>
>>
>> U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB.
>> From: "999999999" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Date: Fri, 05 Jan 2001 07:46:57 GMT.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> Server: Cisco-SIPGateway/IOS-12.x.
>> CSeq: 113193247 INVITE.
>> Allow-Events: telephone-event.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060
>> SIP/2.0 183 Session Progress.
>> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB.
>> From: "999999999" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Date: Fri, 05 Jan 2001 07:46:57 GMT.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> Server: Cisco-SIPGateway/IOS-12.x.
>> CSeq: 113193247 INVITE.
>> Allow-Events: telephone-event.
>> Contact: <sip:666666666 at 3.3.3.3:5060>.
>> Content-Disposition: session;handling=required.
>> Content-Type: application/sdp.
>> Content-Length: 300.
>> .
>> v=0.
>> o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3.
>> s=SIP Call.
>> c=IN IP4 3.3.3.3.
>> t=0 0.
>> m=audio 19398 RTP/AVP 18 13 101.
>> c=IN IP4 3.3.3.3.
>> a=rtpmap:18 G729/8000.
>> a=rtpmap:13 CN/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:40.
>>
>>
>> U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060
>> SIP/2.0 183 Session Progress.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 INVITE.
>> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>> Supported: 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: 292.
>> .
>> v=0.
>> o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1.
>> s=FreeSWITCH.
>> c=IN IP4 1.1.1.1.
>> t=0 0.
>> m=audio 20620 RTP/AVP 18 101.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=silenceSupp:off - - - -.
>> a=ptime:20.
>>
>>
>> U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060
>> CANCEL sip:666666666 at 1.1.1.1 SIP/2.0.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 CANCEL.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060
>> SIP/2.0 481 Call/Transaction Does Not Exist.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 CANCEL.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB.
>> From: "999999999" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Date: Fri, 05 Jan 2001 07:46:57 GMT.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> Server: Cisco-SIPGateway/IOS-12.x.
>> CSeq: 113193247 INVITE.
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE, NOTIFY, INFO.
>> Allow-Events: telephone-event.
>> Contact: <sip:666666666 at 3.3.3.3:5060>.
>> Content-Type: application/sdp.
>> Content-Length: 300.
>> .
>> v=0.
>> o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3.
>> s=SIP Call.
>> c=IN IP4 3.3.3.3.
>> t=0 0.
>> m=audio 19398 RTP/AVP 18 13 101.
>> c=IN IP4 3.3.3.3.
>> a=rtpmap:18 G729/8000.
>> a=rtpmap:13 CN/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:40.
>>
>>
>> U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060
>> ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj.
>> Max-Forwards: 70.
>> From: "999999999" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> CSeq: 113193247 ACK.
>> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 INVITE.
>> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>> Supported: 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: 292.
>> .
>> v=0.
>> o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1.
>> s=FreeSWITCH.
>> c=IN IP4 1.1.1.1.
>> t=0 0.
>> m=audio 20620 RTP/AVP 18 101.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=silenceSupp:off - - - -.
>> a=ptime:20.
>>
>>
>> U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060
>> ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 ACK.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060
>> BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 104 BYE.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1",
>> algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060",
>> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7",
>> response="21ee4a61f1751494e2e96254dd007a4c", qop=auth,
>> cnonce="6bc43301", nc=00000002.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060.
>> From: "999999999" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 104 BYE.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>> Supported: timer, precondition, path, replaces.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060
>> BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe.
>> Max-Forwards: 70.
>> From: "999999999" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> CSeq: 113193248 BYE.
>> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>> Supported: timer, precondition, path, replaces.
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe.
>> From: "999999999" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Date: Fri, 05 Jan 2001 07:47:32 GMT.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> Server: Cisco-SIPGateway/IOS-12.x.
>> Content-Length: 0.
>> CSeq: 113193248 BYE.
>> .
>>
>> Please, can somebody tell me what is happening?.
>>
>> Thanks in advance.
>>
>> Regards.
>>
>> _______________________________________________
>> 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
>>
>> Brian West
>> brian at freeswitch.org
>> -- Meet us a ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>> _______________________________________________
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>> 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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-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com