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

Wed Apr 1 17:09:42 PDT 2009

One question more, maybe a stupid one: How can I search the archives?
I didn't find nothing in lists.freeswitch.org.

Regards

2009/4/2 Brian West <brian at freeswitch.org>:
> Follow this
> thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html
> /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.
>
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> Brian West
> brian at freeswitch.org
> -- Meet us a ClueCon!  http://www.cluecon.com
>
>
>
>
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