Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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.
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@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
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@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@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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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@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@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@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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-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
hehe I emailed it to him off list :) /b On Apr 2, 2009, at 1:09 AM, Kristian Kielhofner wrote:
I probably shouldn't be doing this for you, but...
http://bugs.digium.com/view.php?id=14431
;)
Brian West brian@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com
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.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West brian@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com
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@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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If you go to google and input "site:lists.freeswitch.org blah" /b On Apr 1, 2009, at 7:09 PM, Alfonso Pinto wrote:
One question more, maybe a stupid one: How can I search the archives? I didn't find nothing in lists.freeswitch.org.
Regards
Brian West brian@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com
Alfonso Pinto <elhodred@gmail.com> wrote:
One question more, maybe a stupid one: How can I search the archives?
http://www.gmane.org/ The searching tool they use, Xapian, tends to give good relevance ranking, at least in my experience.
Thank you so much, gmane gives me correct results. Instead, trying to search the thread Brian emailed to me with site:lists.freeswitch.org doesn't give the correct response, thread doesn't appears. Regards 2009/4/2 Jason White <jason@jasonjgw.net>:
Alfonso Pinto <elhodred@gmail.com> wrote:
One question more, maybe a stupid one: How can I search the archives?
The searching tool they use, Xapian, tends to give good relevance ranking, at least in my experience.
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participants (4)
-
Alfonso Pinto
-
Brian West
-
Jason White
-
Kristian Kielhofner