Debian GNU/Linux comes with ABSOLUTELY NO WARRANTY, to the extent permitted by applicable law. Last login: Mon Sep 28 14:35:57 2009 from 192.168.0.125 root@asterisk:~# asterisk -R Asterisk SVN-trunk-r220496, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= [Sep 28 14:39:39] Connected to Asterisk SVN-trunk-r220496 currently running on asterisk (pid = 23238) [2009-09-28 14:39:41] 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=z9hG4bK33e3e6a0 Max-Forwards: 70 From: "asterisk" ;tag=as61b8c83f To: Contact: Call-ID: 22fdb2581342d1cb07d4dcd417459500@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:39:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> sip set debug on SIP Debugging re-enabled [2009-09-28 14:39:41] <--- SIP read from UDP:192.168.0.30:5060 ---> SIP/2.0 200 OK To: ;tag=f76be8be46253182i0 From: "asterisk" ;tag=as61b8c83f Call-ID: 22fdb2581342d1cb07d4dcd417459500@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK33e3e6a0 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 14:39:41] --- (10 headers 0 lines) --- [2009-09-28 14:39:41] Really destroying SIP dialog '22fdb2581342d1cb07d4dcd417459500@64.105.202.244' Method: OPTIONS [2009-09-28 14:39:41] 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=z9hG4bK6ef261ca Max-Forwards: 70 From: "asterisk" ;tag=as3337f960 To: Contact: Call-ID: 5cae9b431759cc510e36488e13319b02@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:39:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 14:39:41] <--- SIP read from UDP:192.168.0.30:5061 ---> SIP/2.0 200 OK To: ;tag=3de4cede920481e2i1 From: "asterisk" ;tag=as3337f960 Call-ID: 5cae9b431759cc510e36488e13319b02@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK6ef261ca 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 14:39:41] --- (10 headers 0 lines) --- [2009-09-28 14:39:41] Really destroying SIP dialog '5cae9b431759cc510e36488e13319b02@64.105.202.244' Method: OPTIONS [2009-09-28 14:39:42] 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=z9hG4bK50f05702 Max-Forwards: 70 From: "asterisk" ;tag=as455ccda3 To: Contact: Call-ID: 52cf56f551d5f24f41d7443311fc6aaa@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:39:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 14:39:42] <--- 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=z9hG4bK50f05702;rport=5060 From: "asterisk" ;tag=as455ccda3 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.eaa9 Call-ID: 52cf56f551d5f24f41d7443311fc6aaa@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:39:42] --- (8 headers 0 lines) --- [2009-09-28 14:39:42] Really destroying SIP dialog '52cf56f551d5f24f41d7443311fc6aaa@64.105.202.244' Method: OPTIONS [2009-09-28 14:39:42] 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=z9hG4bK4b0d2f24 Max-Forwards: 70 From: "asterisk" ;tag=as4cf3ad06 To: Contact: Call-ID: 1384f72670cde52d3313ee983723d979@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:39:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 14:39:42] <--- 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=z9hG4bK4b0d2f24;rport=5060 From: "asterisk" ;tag=as4cf3ad06 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.4168 Call-ID: 1384f72670cde52d3313ee983723d979@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:39:42] --- (8 headers 0 lines) --- [2009-09-28 14:39:42] Really destroying SIP dialog '1384f72670cde52d3313ee983723d979@64.105.202.244' Method: OPTIONS asterisk*CLI> sip set history on SIP History Recording Enabled (use 'sip show history') asterisk*CLI> [2009-09-28 14:40:05] <--- 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-2123c607 From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: Call-ID: 50dbd1c6-ee34967c@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: 395 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 621323 621323 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16444 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 14:40:05] --- (14 headers 18 lines) --- [2009-09-28 14:40:05] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 14:40:05] Using INVITE request as basis request - 50dbd1c6-ee34967c@192.168.0.30 [2009-09-28 14:40:05] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-28 14:40:05] <--- 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-2123c607;received=192.168.0.30 From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: ;tag=as0ad5ce2e Call-ID: 50dbd1c6-ee34967c@192.168.0.30 CSeq: 101 INVITE Server: Asterisk PBX SVN-trunk-r220496 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.crosscomm.net", nonce="77cec199" Content-Length: 0 <------------> [2009-09-28 14:40:05] Scheduling destruction of SIP dialog '50dbd1c6-ee34967c@192.168.0.30' in 6400 ms (Method: INVITE) [2009-09-28 14:40:05] <--- 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-2123c607 From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: ;tag=as0ad5ce2e Call-ID: 50dbd1c6-ee34967c@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 14:40:05] --- (10 headers 0 lines) --- [2009-09-28 14:40:05] <--- 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-da5aef2b From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: Call-ID: 50dbd1c6-ee34967c@192.168.0.30 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="77cec199",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="8049880518aa3faaabd2347cbec71941" Contact: CrossComm, Inc. Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 395 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 621323 621323 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16444 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 14:40:05] --- (15 headers 18 lines) --- [2009-09-28 14:40:05] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 14:40:05] Using INVITE request as basis request - 50dbd1c6-ee34967c@192.168.0.30 [2009-09-28 14:40:05] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-28 14:40:05] Found RTP audio format 0 [2009-09-28 14:40:05] Found RTP audio format 2 [2009-09-28 14:40:05] Found RTP audio format 4 [2009-09-28 14:40:05] Found RTP audio format 8 [2009-09-28 14:40:05] Found RTP audio format 18 [2009-09-28 14:40:05] Found RTP audio format 96 [2009-09-28 14:40:05] Found RTP audio format 97 [2009-09-28 14:40:05] Found RTP audio format 98 [2009-09-28 14:40:05] Found RTP audio format 101 [2009-09-28 14:40:05] Peer audio RTP is at port 192.168.0.30:16444 [2009-09-28 14:40:05] Found audio description format PCMU for ID 0 [2009-09-28 14:40:05] Found audio description format G726-32 for ID 2 [2009-09-28 14:40:05] Found audio description format G723 for ID 4 [2009-09-28 14:40:05] Found audio description format PCMA for ID 8 [2009-09-28 14:40:05] Found audio description format G729a for ID 18 [2009-09-28 14:40:05] Found audio description format G726-40 for ID 96 [2009-09-28 14:40:05] Found audio description format G726-24 for ID 97 [2009-09-28 14:40:05] Found audio description format G726-16 for ID 98 [2009-09-28 14:40:05] Found audio description format telephone-event for ID 101 [2009-09-28 14:40:05] 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 14:40:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2009-09-28 14:40:05] Peer audio RTP is at port 192.168.0.30:16444 [2009-09-28 14:40:05] Looking for 8000 in softphones (domain asterisk.crosscomm.net) [2009-09-28 14:40:05] list_route: hop: [2009-09-28 14:40:05] <--- 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-da5aef2b;received=192.168.0.30 From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: Call-ID: 50dbd1c6-ee34967c@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r220496 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> asterisk*CLI> asterisk*CLI> [2009-09-28 14:40:11] <--- 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=z9hG4bKa7c5.45d97b84.0 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKa7c5.0848c6e4.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK6709cb41;rport=5060 From: "Martin Brendan " ;tag=as422736c8 To: Contact: Call-ID: 4dc853f11983eae7656af93305777fad@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 18:40:11 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 14814 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 14:40:11] --- (18 headers 19 lines) --- [2009-09-28 14:40:11] Sending to 64.61.93.190 : 5060 (no NAT) [2009-09-28 14:40:11] Using INVITE request as basis request - 4dc853f11983eae7656af93305777fad@64.61.93.170 [2009-09-28 14:40:11] Found peer 'VoicePulse-Primary-Brendan' for '8479227343' from 64.61.93.190:5060 [2009-09-28 14:40:11] Found RTP audio format 0 [2009-09-28 14:40:11] Found RTP audio format 8 [2009-09-28 14:40:11] Found RTP audio format 3 [2009-09-28 14:40:11] Found RTP audio format 97 [2009-09-28 14:40:11] Found RTP audio format 111 [2009-09-28 14:40:11] Found RTP audio format 5 [2009-09-28 14:40:11] Found RTP audio format 7 [2009-09-28 14:40:11] Found RTP audio format 101 [2009-09-28 14:40:11] Peer audio RTP is at port 64.61.93.170:14814 [2009-09-28 14:40:11] Found audio description format PCMU for ID 0 [2009-09-28 14:40:11] Found audio description format PCMA for ID 8 [2009-09-28 14:40:11] Found audio description format GSM for ID 3 [2009-09-28 14:40:11] Found audio description format iLBC for ID 97 [2009-09-28 14:40:11] Found audio description format G726-32 for ID 111 [2009-09-28 14:40:11] Found audio description format DVI4 for ID 5 [2009-09-28 14:40:11] Found audio description format LPC for ID 7 [2009-09-28 14:40:11] Found audio description format telephone-event for ID 101 [2009-09-28 14:40:11] 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 14:40:11] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2009-09-28 14:40:11] Peer audio RTP is at port 64.61.93.170:14814 [2009-09-28 14:40:11] Looking for 19192460171 in inbound (domain 64.105.202.244) [2009-09-28 14:40:11] list_route: hop: [2009-09-28 14:40:11] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKa7c5.45d97b84.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKa7c5.0848c6e4.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK6709cb41;rport=5060 Record-Route: From: "Martin Brendan " ;tag=as422736c8 To: Call-ID: 4dc853f11983eae7656af93305777fad@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r220496 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> asterisk*CLI> asterisk*CLI> [2009-09-28 14:40:12] NOTICE[23266]: chan_sip.c:11668 sip_reregister: -- Re-registration for ocY97PJM89@jfk-primary.voicepulse.com [2009-09-28 14:40:12] REGISTER 12 headers, 0 lines [2009-09-28 14:40:12] Reliably Transmitting (no NAT) to 64.61.93.190:5060: REGISTER sip:jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK496671e5 Max-Forwards: 70 From: ;tag=as1e715866 To: Call-ID: 19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244 CSeq: 110 REGISTER User-Agent: Asterisk PBX SVN-trunk-r220496 Authorization: Digest username="ocY97PJM89", realm="jfk-primary.voicepulse.com", algorithm=MD5, uri="sip:jfk-primary.voicepulse.com", nonce="4ac102fd3be03bb8cb8f47f3ae98ab5239df93d1", response="8abc8e5bd6cc529f66c32dfa8ebf6efd", qop=auth, cnonce="53d59075", nc=00000004 Expires: 120 Contact: Content-Length: 0 --- [2009-09-28 14:40:12] <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK496671e5;rport=5060 From: ;tag=as1e715866 To: Call-ID: 19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244 CSeq: 110 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:12] --- (8 headers 0 lines) --- [2009-09-28 14:40:12] <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 64.105.202.244:5060;rport=5060;branch=z9hG4bK496671e5 From: ;tag=as1e715866 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.ec98 Call-ID: 19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244 CSeq: 110 REGISTER WWW-Authenticate: Digest realm="jfk-primary.voicepulse.com", nonce="4ac10438fd29abf3c263b6aa9708c55d48be57d4", qop="auth", stale=true Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:12] --- (9 headers 0 lines) --- [2009-09-28 14:40:12] Responding to challenge, registration to domain/host name jfk-primary.voicepulse.com [2009-09-28 14:40:12] REGISTER 12 headers, 0 lines [2009-09-28 14:40:12] Reliably Transmitting (no NAT) to 64.61.93.190:5060: REGISTER sip:jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK2e1d0678 Max-Forwards: 70 From: ;tag=as1ba1e851 To: Call-ID: 19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244 CSeq: 111 REGISTER User-Agent: Asterisk PBX SVN-trunk-r220496 Authorization: Digest username="ocY97PJM89", realm="jfk-primary.voicepulse.com", algorithm=MD5, uri="sip:jfk-primary.voicepulse.com", nonce="4ac10438fd29abf3c263b6aa9708c55d48be57d4", response="86ddd2d8ebf479a40f079e9146dd7bf8", qop=auth, cnonce="79f250b8", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- [2009-09-28 14:40:12] <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK2e1d0678;rport=5060 From: ;tag=as1ba1e851 To: Call-ID: 19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244 CSeq: 111 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:12] --- (8 headers 0 lines) --- [2009-09-28 14:40:12] <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.105.202.244:5060;rport=5060;branch=z9hG4bK2e1d0678 From: ;tag=as1ba1e851 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.9877 Call-ID: 19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244 CSeq: 111 REGISTER Contact: ;expires=120;received="sip:64.105.202.244:5060" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:12] --- (9 headers 0 lines) --- [2009-09-28 14:40:12] Scheduling destruction of SIP dialog '19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244' in 32000 ms (Method: REGISTER) [2009-09-28 14:40:12] NOTICE[23266]: chan_sip.c:18621 handle_response_register: Outbound Registration: Expiry for jfk-primary.voicepulse.com is 120 sec (Scheduling reregistration in 105 s) [2009-09-28 14:40:13] NOTICE[23266]: chan_sip.c:11668 sip_reregister: -- Re-registration for ocY97PJM89@jfk-backup.voicepulse.com [2009-09-28 14:40:13] REGISTER 12 headers, 0 lines [2009-09-28 14:40:13] Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK75084452 Max-Forwards: 70 From: ;tag=as54a4dd42 To: Call-ID: 101babe656d9d0c34966414f082d3dad@64.105.202.244 CSeq: 114 REGISTER User-Agent: Asterisk PBX SVN-trunk-r220496 Authorization: Digest username="ocY97PJM89", realm="jfk-backup.voicepulse.com", algorithm=MD5, uri="sip:jfk-backup.voicepulse.com", nonce="4ac102b800001c12aa864bbcc27ac1e8149a3abe1d153962", response="43b046ccd445aca57b34004c5e072d8d", qop=auth, cnonce="1dc223f4", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- [2009-09-28 14:40:13] channe <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 64.105.202.244:5060;rport=5060;branch=z9hG4bK75084452 From: ;tag=as54a4dd42 To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.0788 Call-ID: 101babe656d9d0c34966414f082d3dad@64.105.202.244 CSeq: 114 REGISTER WWW-Authenticate: Digest realm="jfk-backup.voicepulse.com", nonce="4ac103210000295a8aa8b1be8e9eecee9ea0482111265a97", qop="auth", stale=true Server: OpenSIPS (1.4.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:13] --- (9 headers 0 lines) --- [2009-09-28 14:40:13] Responding to challenge, registration to domain/host name jfk-backup.voicepulse.com [2009-09-28 14:40:13] REGISTER 12 headers, 0 lines [2009-09-28 14:40:13] Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK2255460a Max-Forwards: 70 From: ;tag=as0ca7caa2 To: Call-ID: 101babe656d9d0c34966414f082d3dad@64.105.202.244 CSeq: 115 REGISTER User-Agent: Asterisk PBX SVN-trunk-r220496 Authorization: Digest username="ocY97PJM89", realm="jfk-backup.voicepulse.com", algorithm=MD5, uri="sip:jfk-backup.voicepulse.com", nonce="4ac103210000295a8aa8b1be8e9eecee9ea0482111265a97", response="7863cc027488251d7314aebd2703b3aa", qop=auth, cnonce="6832abfc", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- [2009-09-28 14:40:13] channe <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK2255460a;rport=5060 From: ;tag=as0ca7caa2 To: Call-ID: 101babe656d9d0c34966414f082d3dad@64.105.202.244 CSeq: 115 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:13] --- (8 headers 0 lines) --- [2009-09-28 14:40:13] channe <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.105.202.244:5060;rport=5060;branch=z9hG4bK2255460a From: ;tag=as0ca7caa2 To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.0d4b Call-ID: 101babe656d9d0c34966414f082d3dad@64.105.202.244 CSeq: 115 REGISTER Contact: ;expires=120;received="sip:64.105.202.244:5060" Server: OpenSIPS (1.4.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:13] --- (9 headers 0 lines) --- [2009-09-28 14:40:13] Scheduling destruction of SIP dialog '101babe656d9d0c34966414f082d3dad@64.105.202.244' in 32000 ms (Method: REGISTER) [2009-09-28 14:40:13] NOTICE[23266]: chan_sip.c:18621 handle_response_register: Outbound Registration: Expiry for jfk-backup.voicepulse.com is 120 sec (Scheduling reregistration in 105 s) asterisk*CLI> [2009-09-28 14:40:41] 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=z9hG4bK31332b3b Max-Forwards: 70 From: "asterisk" ;tag=as4a8646ee To: Contact: Call-ID: 0f9b39d254a2826830f7a3b53a86b3bb@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:40:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 14:40:41] <--- SIP read from UDP:192.168.0.30:5060 ---> SIP/2.0 486 Busy Here To: ;tag=f76be8be46253182i0 From: "asterisk" ;tag=as4a8646ee Call-ID: 0f9b39d254a2826830f7a3b53a86b3bb@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK31332b3b 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 14:40:41] --- (10 headers 0 lines) --- [2009-09-28 14:40:41] Really destroying SIP dialog '0f9b39d254a2826830f7a3b53a86b3bb@64.105.202.244' Method: OPTIONS [2009-09-28 14:40:41] 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=z9hG4bK35b525e6 Max-Forwards: 70 From: "asterisk" ;tag=as3c4bb4fb To: Contact: Call-ID: 02f0093d1e367b293c9957105bc88000@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:40:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 14:40:41] <--- SIP read from UDP:192.168.0.30:5061 ---> SIP/2.0 200 OK To: ;tag=3de4cede920481e2i1 From: "asterisk" ;tag=as3c4bb4fb Call-ID: 02f0093d1e367b293c9957105bc88000@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK35b525e6 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 14:40:41] --- (10 headers 0 lines) --- [2009-09-28 14:40:41] Really destroying SIP dialog '02f0093d1e367b293c9957105bc88000@64.105.202.244' Method: OPTIONS [2009-09-28 14:40:42] 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=z9hG4bK3845384a Max-Forwards: 70 From: "asterisk" ;tag=as3dc9d040 To: Contact: Call-ID: 16ae692c705298a362a4e16f572241ad@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 14:40:42] <--- 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=z9hG4bK3845384a;rport=5060 From: "asterisk" ;tag=as3dc9d040 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.3b13 Call-ID: 16ae692c705298a362a4e16f572241ad@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:42] --- (8 headers 0 lines) --- [2009-09-28 14:40:42] Really destroying SIP dialog '16ae692c705298a362a4e16f572241ad@64.105.202.244' Method: OPTIONS [2009-09-28 14:40:42] 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=z9hG4bK23ab5c35 Max-Forwards: 70 From: "asterisk" ;tag=as7ee90a12 To: Contact: Call-ID: 1f590ade4f1c57346ba9acbc37afc2d3@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r220496 Date: Mon, 28 Sep 2009 18:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 14:40:42] <--- 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=z9hG4bK23ab5c35;rport=5060 From: "asterisk" ;tag=as7ee90a12 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.0af6 Call-ID: 1f590ade4f1c57346ba9acbc37afc2d3@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:42] --- (8 headers 0 lines) --- [2009-09-28 14:40:42] Really destroying SIP dialog '1f590ade4f1c57346ba9acbc37afc2d3@64.105.202.244' Method: OPTIONS [2009-09-28 14:40:43] <--- 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-da5aef2b From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: Call-ID: 50dbd1c6-ee34967c@192.168.0.30 CSeq: 102 CANCEL Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="77cec199",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="340600290760bee2de666d8e4c8175dd" User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 14:40:43] --- (10 headers 0 lines) --- [2009-09-28 14:40:43] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 14:40:43] <--- 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-da5aef2b;received=192.168.0.30 From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: ;tag=as4d1334e9 Call-ID: 50dbd1c6-ee34967c@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r220496 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 14:40:43] <--- 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-da5aef2b;received=192.168.0.30 From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: ;tag=as4d1334e9 Call-ID: 50dbd1c6-ee34967c@192.168.0.30 CSeq: 102 CANCEL Server: Asterisk PBX SVN-trunk-r220496 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 14:40:43] <--- 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-da5aef2b From: CrossComm, Inc. ;tag=2d10a93e749ce564o0 To: ;tag=as4d1334e9 Call-ID: 50dbd1c6-ee34967c@192.168.0.30 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="77cec199",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="8049880518aa3faaabd2347cbec71941" Contact: CrossComm, Inc. User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 14:40:43] --- (11 headers 0 lines) --- [2009-09-28 14:40:43] Really destroying SIP dialog '50dbd1c6-ee34967c@192.168.0.30' Method: ACK [2009-09-28 14:40:44] Really destroying SIP dialog '19d8af12450ab72d1cedcbbf095a3cff@64.105.202.244' Method: REGISTER [2009-09-28 14:40:45] <--- 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=z9hG4bKa7c5.45d97b84.0 From: "Martin Brendan " ;tag=as422736c8 Call-ID: 4dc853f11983eae7656af93305777fad@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 14:40:45] --- (9 headers 0 lines) --- [2009-09-28 14:40:45] Sending to 64.61.93.190 : 5060 (no NAT) [2009-09-28 14:40:45] <--- 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=z9hG4bKa7c5.45d97b84.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKa7c5.0848c6e4.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK6709cb41;rport=5060 From: "Martin Brendan " ;tag=as422736c8 To: ;tag=as760ecfe0 Call-ID: 4dc853f11983eae7656af93305777fad@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r220496 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 14:40:45] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKa7c5.45d97b84.0;received=64.61.93.190 From: "Martin Brendan " ;tag=as422736c8 To: ;tag=as760ecfe0 Call-ID: 4dc853f11983eae7656af93305777fad@64.61.93.170 CSeq: 102 CANCEL Server: Asterisk PBX SVN-trunk-r220496 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 14:40:45] <--- 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=z9hG4bKa7c5.45d97b84.0 From: "Martin Brendan " ;tag=as422736c8 Call-ID: 4dc853f11983eae7656af93305777fad@64.61.93.170 To: ;tag=as760ecfe0 CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 14:40:45] --- (9 headers 0 lines) --- [2009-09-28 14:40:45] Really destroying SIP dialog '4dc853f11983eae7656af93305777fad@64.61.93.170' Method: ACK [2009-09-28 14:40:45] Really destroying SIP dialog '101babe656d9d0c34966414f082d3dad@64.105.202.244' Method: REGISTER asterisk*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry 67.108.9.165 ocY97PJM89 101babe656d9d0c 0x0 (nothing) No 64.61.93.190 ocY97PJM89 19d8af12450ab72 0x0 (nothing) No 192.168.0.30 line3 50dbd1c6-ee3496 0x4 (ulaw) No Rx: INVITE 64.61.93.190 ocY97PJM89 4dc853f11983eae 0x4 (ulaw) No Rx: INVITE 4 active SIP dialogs asterisk*CLI> sip show history 50dbd1c6-ee3496 asterisk*CLI> * SIP Call 1. Rx INVITE / 101 INVITE / sip:8000@asterisk.crosscomm.net 2. AuthChal Auth challenge sent for - nc 0 3. TxRespRel SIP/2.0 / 101 INVITE - 401 Unauthorized 4. SchedDestroy 6400 ms 5. Rx ACK / 101 ACK / sip:8000@asterisk.crosscomm.net 6. Rx INVITE / 102 INVITE / sip:8000@asterisk.crosscomm.net 7. CancelDestroy 8. Invite New call: 50dbd1c6-ee34967c@192.168.0.30 9. AuthOK Auth challenge succesful for line3 10. NewChan Channel SIP/line3-1071fd60 - from 50dbd1c6-ee34967c@192.168.0.3 11. TxResp SIP/2.0 / 102 INVITE - 100 Trying asterisk*CLI> asterisk*CLI> sip show history 4dc853f11983eae asterisk*CLI> * SIP Call 1. Rx INVITE / 102 INVITE / sip:19192460171@64.105.202.244:5060 2. NewChan Channel SIP/VoicePulse-Primary-Brendan-1072a538 - from 4dc853f1 3. TxResp SIP/2.0 / 102 INVITE - 100 Trying