Hi, Updating asterisk to version 1.4.24 solved the problem. Thanks guys. Regards. 2009/4/2 Brian West <brian@freeswitch.org>:
Follow this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.htm... /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@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@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=ceKFmNU84B90c. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@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@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 INVITE. Contact: <sip:gw+primus@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@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@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@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@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@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@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@1.1.1.1. CSeq: 103 INVITE. Contact: <sip:mod_sofia@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@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@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@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@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@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 ACK. Contact: <sip:gw+primus@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@1.1.1.1. CSeq: 103 INVITE. Contact: <sip:mod_sofia@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@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@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@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e@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@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@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193248 BYE. Contact: <sip:gw+primus@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@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@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@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com
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