asterisk2*CLI> sip set debug on SIP Debugging enabled <--- SIP read from UDP:137.28.94.174:50437 ---> INVITE sip:999999@172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK1d7f16bb From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:04 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "364774" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 18574 0 IN IP4 137.28.94.174 s=SIP Call t=0 0 m=audio 21670 RTP/AVP 0 8 18 96 c=IN IP4 137.28.94.174 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv <-------------> --- (18 headers 13 lines) --- == Using SIP RTP CoS mark 5 Sending to 137.28.94.174 : 5061 (no NAT) Using INVITE request as basis request - 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 Found peer '364774' for '364774' from 137.28.94.174:50437 asterisk2*CLI> <--- Reliably Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK1d7f16bb;received=137.28.94.174 From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: ;tag=as42dba452 Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 CSeq: 101 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e9e1af4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174' in 32000 ms (Method: INVITE) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50465 ---> ACK sip:999999@172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK1d7f16bb From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: ;tag=as42dba452 Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 Date: Thu, 17 Sep 2009 21:41:04 GMT CSeq: 101 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50437 ---> INVITE sip:999999@172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK69b61ed5 From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:04 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Authorization: Digest username="364774",realm="asterisk",uri="sip:999999@172.28.1.42",response="29fc8c18242d76a2fd03387cc573ab5f",nonce="7e9e1af4",algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "364774" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 18574 0 IN IP4 137.28.94.174 s=SIP Call t=0 0 m=audio 21670 RTP/AVP 0 8 18 96 c=IN IP4 137.28.94.174 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv <-------------> --- (19 headers 13 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) Using INVITE request as basis request - 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 Found peer '364774' for '364774' from 137.28.94.174:50437 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Peer audio RTP is at port 137.28.94.174:21670 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 96 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 137.28.94.174:21670 Looking for 999999 in phones (domain 172.28.1.42) list_route: hop: asterisk2*CLI> <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK69b61ed5;received=137.28.94.174 From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [999999@phones:1] Answer("SIP/364774-09aa5d58", "") in new stack Audio is at 172.28.1.42 port 11790 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk2*CLI> <--- Reliably Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK69b61ed5;received=137.28.94.174 From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: ;tag=as73c3701a Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 286 v=0 o=root 1089043006 1089043006 IN IP4 172.28.1.42 s=Asterisk PBX 1.6.2.0-rc1 c=IN IP4 172.28.1.42 t=0 0 m=audio 11790 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [999999@phones:2] Set("SIP/364774-09aa5d58", "MBEXT=364774") in new stack -- Executing [999999@phones:3] Dial("SIP/364774-09aa5d58", "SIP/exchange2/340000") in new stack == Using SIP RTP CoS mark 5 Audio is at 172.28.1.42 port 18608 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.28.1.68:5067: INVITE sip:340000@exch08.uwec.edu:5067 SIP/2.0 Via: SIP/2.0/TCP 172.28.1.42:5060;branch=z9hG4bK14f1b293;rport Max-Forwards: 70 From: "364774" ;tag=as11ac9eb9 To: Contact: Call-ID: 73b6b3350ea1dc283968224b35d64692@172.28.1.42 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0-rc1 Date: Thu, 17 Sep 2009 16:24:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 884513040 884513040 IN IP4 172.28.1.42 s=Asterisk PBX 1.6.2.0-rc1 c=IN IP4 172.28.1.42 t=0 0 m=audio 18608 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50437 ---> ACK sip:999999@172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK211ce1a0 From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: ;tag=as73c3701a Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:04 GMT CSeq: 102 ACK User-Agent: Cisco-CP7960G/8.0 Authorization: Digest username="364774",realm="asterisk",uri="sip:999999@172.28.1.42",response="29fc8c18242d76a2fd03387cc573ab5f",nonce="7e9e1af4",algorithm=MD5 Remote-Party-ID: "364774" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk2*CLI> <--- SIP read from UDP:172.28.129.99:50561 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK31fb17c2 From: ;tag=0014a9713f5d0142391703ce-148a09ce To: Call-ID: 0014a971-3f5d0003-45f08c58-5b4104c9@172.28.129.99 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:12 GMT CSeq: 307 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 240 <-------------> --- (12 headers 0 lines) --- Sending to 172.28.129.99 : 5061 (no NAT) <--- Transmitting (no NAT) to 172.28.129.99:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK31fb17c2;received=172.28.129.99 From: ;tag=0014a9713f5d0142391703ce-148a09ce To: ;tag=as3c9a7220 Call-ID: 0014a971-3f5d0003-45f08c58-5b4104c9@172.28.129.99 CSeq: 307 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27ec61f8" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0014a971-3f5d0003-45f08c58-5b4104c9@172.28.129.99' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:172.28.129.99:50561 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK7b3cc2cd From: ;tag=0014a9713f5d0142391703ce-148a09ce To: Call-ID: 0014a971-3f5d0003-45f08c58-5b4104c9@172.28.129.99 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:12 GMT CSeq: 308 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="6200",realm="asterisk",uri="sip:172.28.1.42",response="8c15b458032538bb677b088689f2f58b",nonce="27ec61f8",algorithm=MD5 Content-Length: 0 Expires: 240 <-------------> --- (13 headers 0 lines) --- Sending to 172.28.129.99 : 5061 (no NAT) asterisk2*CLI> <--- Transmitting (no NAT) to 172.28.129.99:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK7b3cc2cd;received=172.28.129.99 From: ;tag=0014a9713f5d0142391703ce-148a09ce To: ;tag=as3c9a7220 Call-ID: 0014a971-3f5d0003-45f08c58-5b4104c9@172.28.129.99 CSeq: 308 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 240 Contact: ;expires=240 Date: Thu, 17 Sep 2009 16:24:25 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0014a971-3f5d0003-45f08c58-5b4104c9@172.28.129.99' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:172.28.129.99:50562 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK6300dea6 From: ;tag=0014a9713f5d014356508244-166197a5 To: Call-ID: 0014a971-3f5d0004-64f4b8d7-2cee8957@172.28.129.99 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:12 GMT CSeq: 306 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 240 <-------------> --- (12 headers 0 lines) --- Sending to 172.28.129.99 : 5061 (no NAT) <--- Transmitting (no NAT) to 172.28.129.99:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK6300dea6;received=172.28.129.99 From: ;tag=0014a9713f5d014356508244-166197a5 To: ;tag=as3a3aaa42 Call-ID: 0014a971-3f5d0004-64f4b8d7-2cee8957@172.28.129.99 CSeq: 306 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ab4ce71" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0014a971-3f5d0004-64f4b8d7-2cee8957@172.28.129.99' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:172.28.129.99:50562 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK39af1611 From: ;tag=0014a9713f5d014356508244-166197a5 To: Call-ID: 0014a971-3f5d0004-64f4b8d7-2cee8957@172.28.129.99 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:12 GMT CSeq: 307 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="6201",realm="asterisk",uri="sip:172.28.1.42",response="5e2ca8cfd754c9d7ea3397657f63cdfd",nonce="1ab4ce71",algorithm=MD5 Content-Length: 0 Expires: 240 <-------------> --- (13 headers 0 lines) --- Sending to 172.28.129.99 : 5061 (no NAT) asterisk2*CLI> <--- Transmitting (no NAT) to 172.28.129.99:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK39af1611;received=172.28.129.99 From: ;tag=0014a9713f5d014356508244-166197a5 To: ;tag=as3a3aaa42 Call-ID: 0014a971-3f5d0004-64f4b8d7-2cee8957@172.28.129.99 CSeq: 307 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 240 Contact: ;expires=240 Date: Thu, 17 Sep 2009 16:24:25 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0014a971-3f5d0004-64f4b8d7-2cee8957@172.28.129.99' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from TCP:172.28.1.74:28017 ---> OPTIONS sip:172.28.1.42:5060 SIP/2.0 FROM: ;epid=70576ACED4;tag=119562e256 TO: CSEQ: 10106 OPTIONS CALL-ID: d6b14b692c894dd28a50a358685d9650 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.28.1.74:28017;branch=z9hG4bK453ccdcd ACCEPT: application/sdp CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 <-------------> --- (10 headers 0 lines) --- Looking for s in default (domain 172.28.1.42) asterisk2*CLI> <--- Transmitting (no NAT) to 172.28.1.74:28017 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.28.1.74:28017;branch=z9hG4bK453ccdcd;received=172.28.1.74 From: ;epid=70576ACED4;tag=119562e256 To: ;tag=as352e0ce3 Call-ID: d6b14b692c894dd28a50a358685d9650 CSeq: 10106 OPTIONS Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'd6b14b692c894dd28a50a358685d9650' in 32000 ms (Method: OPTIONS) Reliably Transmitting (no NAT) to 137.28.5.37:5060: OPTIONS sip:137.28.5.37 SIP/2.0 Via: SIP/2.0/UDP 172.28.1.42:5060;branch=z9hG4bK1a1bbeef;rport Max-Forwards: 70 From: "asterisk" ;tag=as4f08ea5f To: Contact: Call-ID: 7473433e22b2660f6e0449db479413dc@172.28.1.42 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-rc1 Date: Thu, 17 Sep 2009 16:24:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk2*CLI> <--- SIP read from UDP:137.28.5.37:5060 ---> SIP/2.0 200 OK Date: Thu, 17 Sep 2009 21:41:27 GMT Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "asterisk" ;tag=as4f08ea5f Content-Length: 0 To: ;tag=1138012929 Call-ID: 7473433e22b2660f6e0449db479413dc@172.28.1.42 Via: SIP/2.0/UDP 172.28.1.42:5060;branch=z9hG4bK1a1bbeef;rport CSeq: 102 OPTIONS <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '7473433e22b2660f6e0449db479413dc@172.28.1.42' Method: OPTIONS Reliably Transmitting (no NAT) to 137.28.5.36:5060: OPTIONS sip:137.28.5.36 SIP/2.0 Via: SIP/2.0/UDP 172.28.1.42:5060;branch=z9hG4bK26c3fef7;rport Max-Forwards: 70 From: "asterisk" ;tag=as06560c60 To: Contact: Call-ID: 60a49147395c204d3bc92c887b265afd@172.28.1.42 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-rc1 Date: Thu, 17 Sep 2009 16:24:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk2*CLI> <--- SIP read from UDP:137.28.5.36:5060 ---> SIP/2.0 200 OK Date: Thu, 17 Sep 2009 21:41:30 GMT Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "asterisk" ;tag=as06560c60 Content-Length: 0 To: ;tag=1212530487 Call-ID: 60a49147395c204d3bc92c887b265afd@172.28.1.42 Via: SIP/2.0/UDP 172.28.1.42:5060;branch=z9hG4bK26c3fef7;rport CSeq: 102 OPTIONS <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '60a49147395c204d3bc92c887b265afd@172.28.1.42' Method: OPTIONS asterisk2*CLI> <--- SIP read from UDP:172.28.129.99:50561 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK4e8cd473 From: ;tag=0014a9713f5d0144690403e1-2b3d5faf To: Call-ID: 0014a971-3f5d0002-3a5dd9db-130d5e70@172.28.129.99 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:40 GMT CSeq: 300 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 240 <-------------> --- (12 headers 0 lines) --- Sending to 172.28.129.99 : 5061 (no NAT) <--- Transmitting (no NAT) to 172.28.129.99:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK4e8cd473;received=172.28.129.99 From: ;tag=0014a9713f5d0144690403e1-2b3d5faf To: ;tag=as2b46f09a Call-ID: 0014a971-3f5d0002-3a5dd9db-130d5e70@172.28.129.99 CSeq: 300 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06ec6b0e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0014a971-3f5d0002-3a5dd9db-130d5e70@172.28.129.99' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:172.28.129.99:50561 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK0cfb1ad3 From: ;tag=0014a9713f5d0144690403e1-2b3d5faf To: Call-ID: 0014a971-3f5d0002-3a5dd9db-130d5e70@172.28.129.99 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:40 GMT CSeq: 301 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="6200",realm="asterisk",uri="sip:172.28.1.42",response="7e7958c7bcb4a35ca6dc5d288de1cb4b",nonce="06ec6b0e",algorithm=MD5 Content-Length: 0 Expires: 240 <-------------> --- (13 headers 0 lines) --- Sending to 172.28.129.99 : 5061 (no NAT) asterisk2*CLI> <--- Transmitting (no NAT) to 172.28.129.99:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.129.99:5061;branch=z9hG4bK0cfb1ad3;received=172.28.129.99 From: ;tag=0014a9713f5d0144690403e1-2b3d5faf To: ;tag=as2b46f09a Call-ID: 0014a971-3f5d0002-3a5dd9db-130d5e70@172.28.129.99 CSeq: 301 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 240 Contact: ;expires=240 Date: Thu, 17 Sep 2009 16:24:54 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0014a971-3f5d0002-3a5dd9db-130d5e70@172.28.129.99' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0014a971-3f5d0003-45f08c58-5b4104c9@172.28.129.99' Method: REGISTER Really destroying SIP dialog '0014a971-3f5d0004-64f4b8d7-2cee8957@172.28.129.99' Method: REGISTER asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50437 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK4c5ec395 From: ;tag=000f8fe922b6006d627601b2-528c9d1a To: Call-ID: 000f8fe9-22b60002-3a4df9e7-03ae763a@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:53 GMT CSeq: 145 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 240 <-------------> --- (12 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK4c5ec395;received=137.28.94.174 From: ;tag=000f8fe922b6006d627601b2-528c9d1a To: ;tag=as278ea696 Call-ID: 000f8fe9-22b60002-3a4df9e7-03ae763a@137.28.94.174 CSeq: 145 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0d7e30c0" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60002-3a4df9e7-03ae763a@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50437 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK577ddc38 From: ;tag=000f8fe922b6006d627601b2-528c9d1a To: Call-ID: 000f8fe9-22b60002-3a4df9e7-03ae763a@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:53 GMT CSeq: 146 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="364774",realm="asterisk",uri="sip:172.28.1.42",response="82ed005c72459eee8842366f5cc3ecdb",nonce="0d7e30c0",algorithm=MD5 Content-Length: 0 Expires: 240 <-------------> --- (13 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) asterisk2*CLI> <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK577ddc38;received=137.28.94.174 From: ;tag=000f8fe922b6006d627601b2-528c9d1a To: ;tag=as278ea696 Call-ID: 000f8fe9-22b60002-3a4df9e7-03ae763a@137.28.94.174 CSeq: 146 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 240 Contact: ;expires=240 Date: Thu, 17 Sep 2009 16:25:09 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60002-3a4df9e7-03ae763a@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50437 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK30a86267 From: ;tag=000f8fe922b6006e6e3b59bc-2c115b08 To: Call-ID: 000f8fe9-22b60003-0d823bd0-6db0d586@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:53 GMT CSeq: 145 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 240 <-------------> --- (12 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK30a86267;received=137.28.94.174 From: ;tag=000f8fe922b6006e6e3b59bc-2c115b08 To: ;tag=as2ea0f87d Call-ID: 000f8fe9-22b60003-0d823bd0-6db0d586@137.28.94.174 CSeq: 145 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2b5f5d43" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60003-0d823bd0-6db0d586@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50437 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK33e91b31 From: ;tag=000f8fe922b6006e6e3b59bc-2c115b08 To: Call-ID: 000f8fe9-22b60003-0d823bd0-6db0d586@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:53 GMT CSeq: 146 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="364774",realm="asterisk",uri="sip:172.28.1.42",response="bd90ed3379c78440baf0942664b71e6c",nonce="2b5f5d43",algorithm=MD5 Content-Length: 0 Expires: 240 <-------------> --- (13 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) asterisk2*CLI> <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK33e91b31;received=137.28.94.174 From: ;tag=000f8fe922b6006e6e3b59bc-2c115b08 To: ;tag=as2ea0f87d Call-ID: 000f8fe9-22b60003-0d823bd0-6db0d586@137.28.94.174 CSeq: 146 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 240 Contact: ;expires=240 Date: Thu, 17 Sep 2009 16:25:09 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60003-0d823bd0-6db0d586@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50438 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK741d7ef7 From: ;tag=000f8fe922b6006f230c2e45-05dbab58 To: Call-ID: 000f8fe9-22b60004-58235296-52b57b83@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:54 GMT CSeq: 145 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 240 <-------------> --- (12 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK741d7ef7;received=137.28.94.174 From: ;tag=000f8fe922b6006f230c2e45-05dbab58 To: ;tag=as364037fe Call-ID: 000f8fe9-22b60004-58235296-52b57b83@137.28.94.174 CSeq: 145 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6787207b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60004-58235296-52b57b83@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50438 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK3a0fa9de From: ;tag=000f8fe922b6006f230c2e45-05dbab58 To: Call-ID: 000f8fe9-22b60004-58235296-52b57b83@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:54 GMT CSeq: 146 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="365234",realm="asterisk",uri="sip:172.28.1.42",response="bfdcea90d169e6780ff3e155e904fbe8",nonce="6787207b",algorithm=MD5 Content-Length: 0 Expires: 240 <-------------> --- (13 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) asterisk2*CLI> <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK3a0fa9de;received=137.28.94.174 From: ;tag=000f8fe922b6006f230c2e45-05dbab58 To: ;tag=as364037fe Call-ID: 000f8fe9-22b60004-58235296-52b57b83@137.28.94.174 CSeq: 146 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 240 Contact: ;expires=240 Date: Thu, 17 Sep 2009 16:25:09 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60004-58235296-52b57b83@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50439 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK75ec01e5 From: ;tag=000f8fe922b600705d538299-79dda94c To: Call-ID: 000f8fe9-22b60005-2dff728b-6d038092@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:54 GMT CSeq: 145 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 240 <-------------> --- (12 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK75ec01e5;received=137.28.94.174 From: ;tag=000f8fe922b600705d538299-79dda94c To: ;tag=as6defd800 Call-ID: 000f8fe9-22b60005-2dff728b-6d038092@137.28.94.174 CSeq: 145 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="793e71cd" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60005-2dff728b-6d038092@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50439 ---> REGISTER sip:172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK4dc6b54f From: ;tag=000f8fe922b600705d538299-79dda94c To: Call-ID: 000f8fe9-22b60005-2dff728b-6d038092@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:41:54 GMT CSeq: 146 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="362994",realm="asterisk",uri="sip:172.28.1.42",response="4f5f0d165d8e2b67924aa9c74c471c78",nonce="793e71cd",algorithm=MD5 Content-Length: 0 Expires: 240 <-------------> --- (13 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) asterisk2*CLI> <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK4dc6b54f;received=137.28.94.174 From: ;tag=000f8fe922b600705d538299-79dda94c To: ;tag=as6defd800 Call-ID: 000f8fe9-22b60005-2dff728b-6d038092@137.28.94.174 CSeq: 146 REGISTER Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 240 Contact: ;expires=240 Date: Thu, 17 Sep 2009 16:25:09 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000f8fe9-22b60005-2dff728b-6d038092@137.28.94.174' in 32000 ms (Method: REGISTER) asterisk2*CLI> <--- SIP read from TCP:172.28.1.68:19763 ---> OPTIONS sip:172.28.1.42:5060 SIP/2.0 FROM: ;epid=770E3F755A;tag=db16c85045 TO: CSEQ: 18 OPTIONS CALL-ID: e89dc1b2af604a25b0956e5dbd6f9264 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.28.1.68:19763;branch=z9hG4bK94a5a844 ACCEPT: application/sdp CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 <-------------> --- (10 headers 0 lines) --- Looking for s in default (domain 172.28.1.42) asterisk2*CLI> <--- Transmitting (no NAT) to 172.28.1.68:19763 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.28.1.68:19763;branch=z9hG4bK94a5a844;received=172.28.1.68 From: ;epid=770E3F755A;tag=db16c85045 To: ;tag=as43cd849e Call-ID: e89dc1b2af604a25b0956e5dbd6f9264 CSeq: 18 OPTIONS Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e89dc1b2af604a25b0956e5dbd6f9264' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '0014a971-3f5d0002-3a5dd9db-130d5e70@172.28.129.99' Method: REGISTER asterisk2*CLI> <--- SIP read from UDP:137.28.94.174:50437 ---> BYE sip:999999@172.28.1.42 SIP/2.0 Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK03bbfdec From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: ;tag=as73c3701a Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 Max-Forwards: 70 Date: Thu, 17 Sep 2009 21:42:11 GMT CSeq: 103 BYE User-Agent: Cisco-CP7960G/8.0 Content-Length: 0 Authorization: Digest username="364774",realm="asterisk",uri="sip:999999@172.28.1.42",response="1948c85051d1a6527cb9650dce876a8f",nonce="7e9e1af4",algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Sending to 137.28.94.174 : 5061 (no NAT) <--- Transmitting (no NAT) to 137.28.94.174:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 137.28.94.174:5061;branch=z9hG4bK03bbfdec;received=137.28.94.174 From: "364774" ;tag=000f8fe922b6006c4729212d-7f7e86b1 To: ;tag=as73c3701a Call-ID: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174 CSeq: 103 BYE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> asterisk2*CLI> sip set debug off SIP Debugging Disabled