[Freeswitch-users] 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 '271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "9" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "9" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: "9" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, cnonce="47efcad4", nc=0001. 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: "9" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. Max-Forwards: 69. From: "9" ;tag=e050QBXFZXN6K. To: . Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 INVITE. Contact: . 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: "9" ;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
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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 '271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "9" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "9" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: "9" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, cnonce="47efcad4", nc=0001. 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: "9" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. Max-Forwards: 69. From: "9" 559066...@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: . Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 INVITE. Contact: . 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: "9" 99...@3.
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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 : > 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 '271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "9" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "9" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "9" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=0001. > 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: "9" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "9" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 I
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
Follow this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html /b On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: Hi guys, I've using asterisk as PSTN gateway. When a call arrives from PSTN, I send the call to freeswitch and this route the call to a SIP gateway. When caller cancels the call (hangups before callee answers), I get this on asterisk CLI: chan_sip.c:13056 handle_response: Remote host can't match request CANCEL to call '271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "9" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "9" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: "9" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, cnonce="47efcad4", nc=0001. 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: "9" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. Max-Forwards: 69. From: "9" 559066...@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: . Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 INVITE. Contact: . 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: "9" 99...@3.3.3.3>;screen=yes;privacy=off. . v=0. o=F
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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 : > Follow this > thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "9" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "9" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "9" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=0001. > 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: "9" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "9" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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 br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ 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
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
Alfonso Pinto 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. ___ 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
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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 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 : >> 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 '271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. >> From: "9" ;tag=as26208773. >> To: . >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. >> To: ;tag=ceKFmNU84B90c. >> Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. >> From: "9" ;tag=as26208773. >> To: ;tag=ceKFmNU84B90c. >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. >> From: "9" ;tag=as26208773. >> To: . >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1", >> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", >> response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, >> cnonce="47efcad4", nc=0001. >> 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: "9" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. >> CSeq: 103 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Content-Length: 0
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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 br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ 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
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
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 : > Alfonso Pinto 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. > > > ___ > 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 > ___ 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
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls [SOLVED]
Hi, Updating asterisk to version 1.4.24 solved the problem. Thanks guys. Regards. 2009/4/2 Brian West : > Follow this > thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "9" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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: "9" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "9" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "9" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=0001. > 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: "9" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80...@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:66...@3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "9" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UP
