[Sep 28 16:20:24] Connected to Asterisk 1.6.2.0-beta4 currently running on asterisk (pid = 20533) asterisk*CLI> core set verbose 50 Verbosity was 0 and is now 50 asterisk*CLI> core set debug 50 Core debug was 0 and is now 50 asterisk*CLI> sip set history on SIP History Recording Enabled (use 'sip show history') asterisk*CLI> sip set debug on SIP Debugging enabled [2009-09-28 16:20:56] <--- SIP read from UDP:192.168.0.30:5060 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-ae940c3 From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 101 INVITE Max-Forwards: 70 Contact: CrossComm, Inc. Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 1226527 1226527 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16460 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2009-09-28 16:20:56] --- (14 headers 18 lines) --- [2009-09-28 16:20:56] == Using SIP RTP TOS bits 16 [2009-09-28 16:20:56] == Using SIP RTP CoS mark 5 [2009-09-28 16:20:56] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 16:20:56] Using INVITE request as basis request - af00e7f7-1143ca13@192.168.0.30 [2009-09-28 16:20:56] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-28 16:20:56] <--- Reliably Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-ae940c3;received=192.168.0.30 From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: ;tag=as266dcab4 Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 101 INVITE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.crosscomm.net", nonce="2c035955" Content-Length: 0 <------------> [2009-09-28 16:20:56] Scheduling destruction of SIP dialog 'af00e7f7-1143ca13@192.168.0.30' in 6400 ms (Method: INVITE) [2009-09-28 16:20:56] <--- SIP read from UDP:192.168.0.30:5060 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-ae940c3 From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: ;tag=as266dcab4 Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 101 ACK Max-Forwards: 70 Contact: CrossComm, Inc. User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:20:56] --- (10 headers 0 lines) --- [2009-09-28 16:20:56] <--- SIP read from UDP:192.168.0.30:5060 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-df83152f From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="2c035955",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="4a501d5b223f291d1b1112737a607a73" Contact: CrossComm, Inc. Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 1226527 1226527 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16460 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2009-09-28 16:20:56] --- (15 headers 18 lines) --- [2009-09-28 16:20:56] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 16:20:56] Using INVITE request as basis request - af00e7f7-1143ca13@192.168.0.30 [2009-09-28 16:20:56] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-28 16:20:56] Found RTP audio format 0 [2009-09-28 16:20:56] Found RTP audio format 2 [2009-09-28 16:20:56] Found RTP audio format 4 [2009-09-28 16:20:56] Found RTP audio format 8 [2009-09-28 16:20:56] Found RTP audio format 18 [2009-09-28 16:20:56] Found RTP audio format 96 [2009-09-28 16:20:56] Found RTP audio format 97 [2009-09-28 16:20:56] Found RTP audio format 98 [2009-09-28 16:20:56] Found RTP audio format 101 [2009-09-28 16:20:56] Peer audio RTP is at port 192.168.0.30:16460 [2009-09-28 16:20:56] Found audio description format PCMU for ID 0 [2009-09-28 16:20:56] Found audio description format G726-32 for ID 2 [2009-09-28 16:20:56] Found audio description format G723 for ID 4 [2009-09-28 16:20:56] Found audio description format PCMA for ID 8 [2009-09-28 16:20:56] Found audio description format G729a for ID 18 [2009-09-28 16:20:56] Found audio description format G726-40 for ID 96 [2009-09-28 16:20:56] Found audio description format G726-24 for ID 97 [2009-09-28 16:20:56] Found audio description format G726-16 for ID 98 [2009-09-28 16:20:56] Found audio description format telephone-event for ID 101 [2009-09-28 16:20:56] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [2009-09-28 16:20:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-09-28 16:20:56] Peer audio RTP is at port 192.168.0.30:16460 [2009-09-28 16:20:56] Looking for 8000 in softphones (domain asterisk.crosscomm.net) [2009-09-28 16:20:56] list_route: hop: [2009-09-28 16:20:56] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-df83152f;received=192.168.0.30 From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-09-28 16:20:56] -- Executing [8000@softphones:1] ConfBridge("SIP/line3-482c8fd0", "8000") in new stack [2009-09-28 16:20:56] Audio is at 64.105.202.244 port 20016 [2009-09-28 16:20:56] Adding codec 0x4 (ulaw) to SDP [2009-09-28 16:20:56] Adding codec 0x8 (alaw) to SDP [2009-09-28 16:20:56] Adding non-codec 0x1 (telephone-event) to SDP [2009-09-28 16:20:56] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-df83152f;received=192.168.0.30 From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: ;tag=as5fb80a11 Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 297 v=0 o=root 1376066602 1376066602 IN IP4 64.105.202.244 s=Asterisk PBX 1.6.2.0-beta4 c=IN IP4 64.105.202.244 t=0 0 m=audio 20016 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-28 16:20:56] -- Playing 'conf-onlyperson.slin' (language 'en') [2009-09-28 16:21:01] <--- SIP read from UDP:64.61.93.190:5060 ---> INVITE sip:19192460171@64.105.202.244:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKc0b1.a1daaf32.0 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKc0b1.4350e62.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK022e706f;rport=5060 From: "Martin Brendan " ;tag=as0c0039c0 To: Contact: Call-ID: 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 68 Remote-Party-ID: "Martin Brendan " ;privacy=off;screen=no Date: Mon, 28 Sep 2009 20:21:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 410 v=0 o=root 29667 29667 IN IP4 64.61.93.170 s=session c=IN IP4 64.61.93.170 t=0 0 m=audio 19496 RTP/AVP 0 8 3 97 111 5 7 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2009-09-28 16:21:01] --- (18 headers 19 lines) --- [2009-09-28 16:21:01] == Using SIP RTP TOS bits 16 [2009-09-28 16:21:01] == Using SIP RTP CoS mark 5 [2009-09-28 16:21:01] Sending to 64.61.93.190 : 5060 (no NAT) [2009-09-28 16:21:01] Using INVITE request as basis request - 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 [2009-09-28 16:21:01] Found peer 'VoicePulse-Primary-Brendan' for '8479227343' from 64.61.93.190:5060 [2009-09-28 16:21:01] Found RTP audio format 0 [2009-09-28 16:21:01] Found RTP audio format 8 [2009-09-28 16:21:01] Found RTP audio format 3 [2009-09-28 16:21:01] Found RTP audio format 97 [2009-09-28 16:21:01] Found RTP audio format 111 [2009-09-28 16:21:01] Found RTP audio format 5 [2009-09-28 16:21:01] Found RTP audio format 7 [2009-09-28 16:21:01] Found RTP audio format 101 [2009-09-28 16:21:01] Peer audio RTP is at port 64.61.93.170:19496 [2009-09-28 16:21:01] Found audio description format PCMU for ID 0 [2009-09-28 16:21:01] Found audio description format PCMA for ID 8 [2009-09-28 16:21:01] Found audio description format GSM for ID 3 [2009-09-28 16:21:01] Found audio description format iLBC for ID 97 [2009-09-28 16:21:01] Found audio description format G726-32 for ID 111 [2009-09-28 16:21:01] Found audio description format DVI4 for ID 5 [2009-09-28 16:21:01] Found audio description format LPC for ID 7 [2009-09-28 16:21:01] Found audio description format telephone-event for ID 101 [2009-09-28 16:21:01] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xcae (gsm|ulaw|alaw|g726|adpcm|lpc10|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [2009-09-28 16:21:01] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-09-28 16:21:01] Peer audio RTP is at port 64.61.93.170:19496 [2009-09-28 16:21:01] Looking for 19192460171 in inbound (domain 64.105.202.244) [2009-09-28 16:21:01] list_route: hop: [2009-09-28 16:21:01] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKc0b1.a1daaf32.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKc0b1.4350e62.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK022e706f;rport=5060 Record-Route: From: "Martin Brendan " ;tag=as0c0039c0 o: Call-ID: 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-09-28 16:21:01] -- Executing [19192460171@inbound:1] ConfBridge("SIP/VoicePulse-Primary-Brendan-10367280", "8000,mMsc") in new stack [2009-09-28 16:21:01] Audio is at 64.105.202.244 port 20008 [2009-09-28 16:21:01] Adding codec 0x2 (gsm) to SDP [2009-09-28 16:21:01] Adding codec 0x4 (ulaw) to SDP [2009-09-28 16:21:01] Adding codec 0x8 (alaw) to SDP [2009-09-28 16:21:01] Adding non-codec 0x1 (telephone-event) to SDP [2009-09-28 16:21:01] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKc0b1.a1daaf32.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKc0b1.4350e62.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK022e706f;rport=5060 Record-Route: From: "Martin Brendan " ;tag=as0c0039c0 To: ;tag=as08f5ee1b Call-ID: 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 318 v=0 o=root 712800917 712800917 IN IP4 64.105.202.244 s=Asterisk PBX 1.6.2.0-beta4 c=IN IP4 64.105.202.244 t=0 0 m=audio 20008 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-28 16:21:01] -- Playing 'conf-onlyone.slin' (language 'en') [2009-09-28 16:21:05] <--- SIP read from UDP:192.168.0.30:5060 ---> SUBSCRIBE sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-246fcfba From: CrossComm, Inc. ;tag=f0fff61673677df3 To: CrossComm, Inc. ;tag=as35894806 Call-ID: f353b712-ec105127@192.168.0.30 CSeq: 49583 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="50424a67",uri="sip:asterisk.crosscomm.net",algorithm=MD5,response="609aeab1fda1388c708673839efbe578" Contact: CrossComm, Inc. Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:21:05] --- (14 headers 0 lines) --- [2009-09-28 16:21:05] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again. [2009-09-28 16:21:05] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-246fcfba;received=192.168.0.30 From: CrossComm, Inc. ;tag=f0fff61673677df3 To: CrossComm, Inc. ;tag=as35894806 Call-ID: f353b712-ec105127@192.168.0.30 CSeq: 49583 SUBSCRIBE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:21:05] Really destroying SIP dialog 'f353b712-ec105127@192.168.0.30' Method: SUBSCRIBE [2009-09-28 16:21:09] Reliably Transmitting (no NAT) to 192.168.0.30:5060: OPTIONS sip:line3@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK72a786ca;rport Max-Forwards: 70 From: "asterisk" ;tag=as5c007557 To: Contact: Call-ID: 23f98dac71a094896ecefff473765bcd@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-beta4 Date: Mon, 28 Sep 2009 20:21:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:21:09] <--- SIP read from UDP:192.168.0.30:5060 ---> SIP/2.0 486 Busy Here To: ;tag=f76be8be46253182i0 From: "asterisk" ;tag=as5c007557 Call-ID: 23f98dac71a094896ecefff473765bcd@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK72a786ca Server: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2009-09-28 16:21:09] --- (10 headers 0 lines) --- [2009-09-28 16:21:09] Really destroying SIP dialog '23f98dac71a094896ecefff473765bcd@64.105.202.244' Method: OPTIONS [2009-09-28 16:21:09] Reliably Transmitting (no NAT) to 67.108.9.165:5060: OPTIONS sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK572acc27;rport Max-Forwards: 70 From: "asterisk" ;tag=as521dfabd To: Contact: Call-ID: 04876e5b5c5ae52e58d5039775ca2912@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-beta4 Date: Mon, 28 Sep 2009 20:21:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:21:09] Reliably Transmitting (no NAT) to 64.61.93.190:5060: OPTIONS sip:jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK5f5c2b08;rport Max-Forwards: 70 From: "asterisk" ;tag=as1d1097dc To: Contact: Call-ID: 4e1686f61afa356f455363bc74b6c283@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-beta4 Date: Mon, 28 Sep 2009 20:21:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:21:09] Reliably Transmitting (no NAT) to 192.168.0.30:5061: OPTIONS sip:line4@192.168.0.30:5061 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK78bbb74c;rport Max-Forwards: 70 From: "asterisk" ;tag=as5cbd6d8b To: Contact: Call-ID: 7670ab426b7bbfb436bb9f8c79a010c8@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-beta4 Date: Mon, 28 Sep 2009 20:21:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:21:09] <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 200 OK to keepalive Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK5f5c2b08;rport=5060 From: "asterisk" ;tag=as1d1097dc To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.b36d Call-ID: 4e1686f61afa356f455363bc74b6c283@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:21:09] --- (8 headers 0 lines) --- [2009-09-28 16:21:09] Really destroying SIP dialog '4e1686f61afa356f455363bc74b6c283@64.105.202.244' Method: OPTIONS [2009-09-28 16:21:09] <--- SIP read from UDP:192.168.0.30:5061 ---> SIP/2.0 200 OK To: ;tag=3de4cede920481e2i1 From: "asterisk" ;tag=as5cbd6d8b Call-ID: 7670ab426b7bbfb436bb9f8c79a010c8@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK78bbb74c Server: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2009-09-28 16:21:09] --- (10 headers 0 lines) --- [2009-09-28 16:21:09] Really destroying SIP dialog '7670ab426b7bbfb436bb9f8c79a010c8@64.105.202.244' Method: OPTIONS [2009-09-28 16:21:09] <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 200 OK to keepalive Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK572acc27;rport=5060 From: "asterisk" ;tag=as521dfabd To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.e496 Call-ID: 04876e5b5c5ae52e58d5039775ca2912@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:21:09] --- (8 headers 0 lines) --- [2009-09-28 16:21:09] Really destroying SIP dialog '04876e5b5c5ae52e58d5039775ca2912@64.105.202.244' Method: OPTIONS <--- SIP read from UDP:192.168.0.30:5060 ---> SUBSCRIBE sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-45fe0cf0 From: CrossComm, Inc. ;tag=40074c287f28dac2 To: CrossComm, Inc. ;tag=as35894806 Call-ID: 29673ca0-a438e4aa@192.168.0.30 CSeq: 56409 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="50424a67",uri="sip:asterisk.crosscomm.net",algorithm=MD5,response="609aeab1fda1388c708673839efbe578" Contact: CrossComm, Inc. Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:21:35] --- (14 headers 0 lines) --- [2009-09-28 16:21:35] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again. [2009-09-28 16:21:35] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-45fe0cf0;received=192.168.0.30 From: CrossComm, Inc. ;tag=40074c287f28dac2 To: CrossComm, Inc. ;tag=as35894806 Call-ID: 29673ca0-a438e4aa@192.168.0.30 CSeq: 56409 SUBSCRIBE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:21:35] Really destroying SIP dialog '29673ca0-a438e4aa@192.168.0.30' Method: SUBSCRIBE <--- SIP read from UDP:64.61.93.190:5060 ---> CANCEL sip:19192460171@64.105.202.244:5060 SIP/2.0 Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKc0b1.a1daaf32.0 From: "Martin Brendan " ;tag=as0c0039c0 Call-ID: 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 To: CSeq: 102 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:21:44] --- (9 headers 0 lines) --- [2009-09-28 16:21:44] Sending to 64.61.93.190 : 5060 (no NAT) [2009-09-28 16:21:44] <--- Reliably Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKc0b1.a1daaf32.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKc0b1.4350e62.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK022e706f;rport=5060 From: "Martin Brendan " ;tag=as0c0039c0 To: ;tag=as08f5ee1b Call-ID: 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:21:44] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKc0b1.a1daaf32.0;received=64.61.93.190 From: "Martin Brendan " ;tag=as0c0039c0 To: ;tag=as08f5ee1b Call-ID: 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 CSeq: 102 CANCEL Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:21:44] <--- SIP read from UDP:64.61.93.190:5060 ---> ACK sip:19192460171@64.105.202.244:5060 SIP/2.0 Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKc0b1.a1daaf32.0 From: "Martin Brendan " ;tag=as0c0039c0 Call-ID: 2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170 To: ;tag=as08f5ee1b CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:21:44] --- (9 headers 0 lines) --- [2009-09-28 16:21:44] Really destroying SIP dialog '2ff281b92105b33d1363e3e6737f7e2c@64.61.93.170' Method: ACK [2009-09-28 16:21:44] <--- SIP read from UDP:192.168.0.30:5060 ---> CANCEL sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-df83152f From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 102 CANCEL Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="2c035955",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="5c509570c4ed61d805cf6a134623e29b" User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:21:44] --- (10 headers 0 lines) --- [2009-09-28 16:21:44] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 16:21:44] <--- Reliably Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-df83152f;received=192.168.0.30 From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: ;tag=as5fb80a11 Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:21:44] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-df83152f;received=192.168.0.30 From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: ;tag=as5fb80a11 Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 102 CANCEL Server: Asterisk PBX 1.6.2.0-beta4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:21:44] <--- SIP read from UDP:192.168.0.30:5060 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-df83152f From: CrossComm, Inc. ;tag=26d2eea73721f323o0 To: ;tag=as5fb80a11 Call-ID: af00e7f7-1143ca13@192.168.0.30 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="2c035955",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="4a501d5b223f291d1b1112737a607a73" Contact: CrossComm, Inc. User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:21:44] --- (11 headers 0 lines) --- [2009-09-28 16:21:45] Really destroying SIP dialog 'af00e7f7-1143ca13@192.168.0.30' Method: ACK