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cdeacon

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  1. cdeacon
    Here are my logs from the point that the auto attendant forwards the call to the extension: [7] 2008/09/29 14:12:48: SIP Rx tcp:172.16.116.104:5065: REFER sip:999@172.16.116.105:1181;transport=tcp SIP/2.0 FROM: <sip:999@172.16.116.104;user=phone>;epid=06A5C4BC0B;tag=c19311cfd TO: <sip:999@172.16.116.104;user=phone>;tag=53059 CSEQ: 1 REFER CALL-ID: 88725d22@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.16.116.104:5065;branch=z9hG4bK6d898cb2 CONTACT: <sip:OBSRNDVMMX01.obsrnd.obsglobal.com:5065;transport=Tcp;maddr=172.16.116.104 ;ms-opaque=ede52a3158c87bd0>;automata CONTENT-LENGTH: 0 REFER-TO: <sip:212@172.16.116.105:1181;transport=tcp;user=phone> REFERRED-BY: <sip:999@172.16.116.104;user=phone> USER-AGENT: RTCC/3.0.0.0 [7] 2008/09/29 14:12:48: SIP Tx tcp:172.16.116.104:5065: SIP/2.0 202 Accepted Via: SIP/2.0/TCP 172.16.116.104:5065;branch=z9hG4bK6d898cb2 From: <sip:999@172.16.116.104;user=phone>;tag=c19311cfd;epid=06A5C4BC0B To: <sip:999@172.16.116.104;user=phone>;tag=53059 Call-ID: 88725d22@pbx CSeq: 1 REFER Contact: <sip:999@172.16.116.105:1181;transport=tcp> User-Agent: pbxnsip-PBX/3.0.0.2998 Content-Length: 0 [5] 2008/09/29 14:12:48: Redirecting call to 212 [5] 2008/09/29 14:12:48: Call 88725d22@pbx#53059: Last request not finished [7] 2008/09/29 14:12:48: SIP Tx tcp:172.16.116.104:5065: BYE sip:OBSRNDVMMX01.obsrnd.obsglobal.com:5065;transport=Tcp;maddr=172.16.116.104 SIP/2.0 Via: SIP/2.0/TCP 172.16.116.105:1181;branch=z9hG4bK-d1b5d0ea5696fcc1fcaf23c8260956a0;rport From: "OBS Global" <sip:999@172.16.116.104;user=phone>;tag=53059 To: <sip:999@172.16.116.104;user=phone>;tag=c19311cfd Call-ID: 88725d22@pbx CSeq: 25625 BYE Max-Forwards: 70 Contact: <sip:999@172.16.116.105:1181;transport=tcp> RTP-RxStat: Dur=15,Pkt=528,Oct=90816,Underun=0 RTP-TxStat: Dur=14,Pkt=699,Oct=117264 Content-Length: 0 [5] 2008/09/29 14:12:48: Dialplan ThinkTel: 212 goes to extension [8] 2008/09/29 14:12:48: Play audio_moh/noise.wav [7] 2008/09/29 14:12:48: UDP: Opening socket on port 54846 [7] 2008/09/29 14:12:48: UDP: Opening socket on port 54847 [7] 2008/09/29 14:12:48: SIP Tx tcp:172.16.116.102:5060: INVITE sip:+12049759212@172.16.116.102;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205 To: "OBS Global" <sip:999@obsrnd.obsglobal.com> Call-ID: 9b698194@pbx CSeq: 11279 INVITE Max-Forwards: 70 Contact: <sip:212@172.16.116.105:1182;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Alert-Info: <http://127.0.0.1/Bellcore-dr2> P-Asserted-Identity: "OBS Global" <sip:999@obsrnd.obsglobal.com> Content-Type: application/sdp Content-Length: 294 v=0 o=- 14904 14904 IN IP4 172.16.116.105 s=- c=IN IP4 172.16.116.105 t=0 0 m=audio 54846 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/09/29 14:12:48: UDP: Opening socket on port 51212 [7] 2008/09/29 14:12:48: UDP: Opening socket on port 51213 [7] 2008/09/29 14:12:48: SIP Tx udp:172.16.116.200:2051: INVITE sip:212@172.16.116.200:2051;line=rpwp3qf5 SIP/2.0 Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-6b60059434809ee17999ff5b626bdef1;rport From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559 To: "OBS Global" <sip:999@obsrnd.obsglobal.com> Call-ID: 7ec7d03a@pbx CSeq: 22046 INVITE Max-Forwards: 70 Contact: <sip:212@172.16.116.105:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Alert-Info: <http://127.0.0.1/Bellcore-dr2> P-Asserted-Identity: "OBS Global" <sip:999@obsrnd.obsglobal.com> Content-Type: application/sdp Content-Length: 294 v=0 o=- 54913 54913 IN IP4 172.16.116.105 s=- c=IN IP4 172.16.116.105 t=0 0 m=audio 51212 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/09/29 14:12:48: 1B9E0136@159.18.161.101#b6d60dcb50: Media-aware pass-through mode [7] 2008/09/29 14:12:48: SIP Rx tcp:172.16.116.102:5060: SIP/2.0 100 Trying FROM: "Chris Deacon"<sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205 TO: "OBS Global"<sip:999@obsrnd.obsglobal.com> CSEQ: 11279 INVITE CALL-ID: 9b698194@pbx VIA: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport CONTENT-LENGTH: 0 [7] 2008/09/29 14:12:48: SIP Rx udp:172.16.116.200:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-6b60059434809ee17999ff5b626bdef1;rport=5060 From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559 To: "OBS Global" <sip:999@obsrnd.obsglobal.com>;tag=yv82uxv0iq Call-ID: 7ec7d03a@pbx CSeq: 22046 INVITE Contact: <sip:212@172.16.116.200:2051;line=rpwp3qf5>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 [7] 2008/09/29 14:12:48: SIP Tx udp:172.16.116.200:2051: PRACK sip:212@172.16.116.200:2051;line=rpwp3qf5 SIP/2.0 Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-636dea0941850e26b19f08a829dc50f9;rport From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559 To: "OBS Global" <sip:999@obsrnd.obsglobal.com>;tag=yv82uxv0iq Call-ID: 7ec7d03a@pbx CSeq: 22047 PRACK Max-Forwards: 70 Contact: <sip:212@172.16.116.105:5060;transport=udp> RAck: 1 22046 INVITE P-Asserted-Identity: "OBS Global" <sip:999@obsrnd.obsglobal.com> Content-Length: 0 [8] 2008/09/29 14:12:48: Play audio_en/ringback.wav [8] 2008/09/29 14:12:48: Call 88725d22@pbx#53059: Response does not correspond to open request [7] 2008/09/29 14:12:48: SIP Rx udp:172.16.116.200:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-636dea0941850e26b19f08a829dc50f9;rport=5060 From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559 To: "OBS Global" <sip:999@obsrnd.obsglobal.com>;tag=yv82uxv0iq Call-ID: 7ec7d03a@pbx CSeq: 22047 PRACK Contact: <sip:212@172.16.116.200:2051;line=rpwp3qf5>;flow-id=1 Content-Length: 0 [7] 2008/09/29 14:12:48: Call 7ec7d03a@pbx#11559: Clear last request [7] 2008/09/29 14:12:48: SIP Rx tcp:172.16.116.102:5060: SIP/2.0 183 Session Progress FROM: "Chris Deacon"<sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205 TO: OBS Global<sip:999@obsrnd.obsglobal.com>;epid=4414844359;tag=389ae44573 CSEQ: 11279 INVITE CALL-ID: 9b698194@pbx VIA: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer [7] 2008/09/29 14:12:49: SIP Rx tcp:172.16.116.104:5065: SIP/2.0 200 OK FROM: "OBS Global"<sip:999@172.16.116.104;user=phone>;tag=53059 TO: <sip:999@172.16.116.104;user=phone>;tag=c19311cfd;epid=06A5C4BC0B CSEQ: 25625 BYE CALL-ID: 88725d22@pbx VIA: SIP/2.0/TCP 172.16.116.105:1181;branch=z9hG4bK-d1b5d0ea5696fcc1fcaf23c8260956a0;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [8] 2008/09/29 14:12:49: Call 88725d22@pbx#53059: Response does not correspond to open request [5] 2008/09/29 14:12:49: BYE Response: Terminate 88725d22@pbx [7] 2008/09/29 14:12:49: Other Ports: 3 [7] 2008/09/29 14:12:49: Call Port: 1B9E0136@159.18.161.101#b6d60dcb50 [7] 2008/09/29 14:12:49: Call Port: 7ec7d03a@pbx#11559 [7] 2008/09/29 14:12:49: Call Port: 9b698194@pbx#62205 [7] 2008/09/29 14:12:49: SIP Rx tcp:172.16.116.102:5060: SIP/2.0 180 Ringing FROM: "Chris Deacon"<sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205 TO: OBS Global<sip:999@obsrnd.obsglobal.com>;epid=4414844359;tag=389ae44573 CSEQ: 11279 INVITE CALL-ID: 9b698194@pbx VIA: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer [7] 2008/09/29 14:12:50: SIP Tr udp:159.18.161.67:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK8751.c2ab7894.0 Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-0c26c6f69825fcf9328601343cba48ea-159.18.161.101-1 Record-Route: <sip:2049758698@159.18.161.67;ftag=159.18.161.101+1+1ba2d9+59b98b75;lr=on> From: Chris Deacon <sip:2049820218@159.18.161.101:5060;transport=udp>;tag=159.18.161.101+1+1ba2d9+59b98b75;isup-oli=00 To: <sip:2049758698@205.200.204.5>;tag=b6d60dcb50 Call-ID: 1B9E0136@159.18.161.101 CSeq: 684567237 INVITE Contact: <sip:2049758698@172.16.116.105:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Content-Type: application/sdp Content-Length: 208 v=0 o=- 38924 38924 IN IP4 172.16.116.105 s=- c=IN IP4 172.16.116.105 t=0 0 m=audio 62836 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrec
  2. cdeacon
    Hey there. I've been setting up an OCS2007 with Exchange UM environment with success thus far with routing calls between SNOM320 sets and OC clients. I'm now looking into the UM portion, specifically the Auto Attendant. I can call in, receive the greeting and ask to be routed to an extension. Once it dials the extension the phone & communicator ring, which is all fine. My issue is that if I hang up the phone while the AA is calling the extension, the extension keeps ringing and then forwards to voice mail. How can I get the auto attendant to terminate the call when the outside caller has ended the call?
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