558600051@default-d532,2", "8600051|K") in new stack [Oct 23 23:53:36] WARNING[6927]: app_meetme.c:3134 admin_exec: Conference number '8600051' not found! [Oct 23 23:53:36] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-d532,2", "") in new stack [Oct 23 23:53:36] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-d532,2' [Oct 23 23:53:36] -- Executing [h@default:1] DeadAGI("Local/55558600051@default-d532,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Oct 23 23:53:36] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Oct 23 23:53:38] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 23 23:53:38] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 23 23:53:42] Really destroying SIP dialog '315c0163013a70e57a908e6274ea052d@127.0.0.2' Method: REGISTER [Oct 23 23:54:01] == Parsing '/etc/asterisk/manager.conf': [Oct 23 23:54:01] Found [Oct 23 23:54:01] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 23 23:54:01] == Parsing '/etc/asterisk/manager.conf': [Oct 23 23:54:01] Found [Oct 23 23:54:01] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 23 23:54:01] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 23 23:54:01] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 23 23:54:02] -- Executing [12503802844@default:1] AGI("SIP/gs102-0000009b", "agi://127.0.0.1:4577/call_log") in new stack [Oct 23 23:54:02] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Oct 23 23:54:02] -- Executing [12503802844@default:2] Dial("SIP/gs102-0000009b", "SIP/switch2voip/12503802844|10000|To") in new stack [Oct 23 23:54:02] Audio is at 192.168.1.135 port 10368 [Oct 23 23:54:02] Adding codec 0x4 (ulaw) to SDP [Oct 23 23:54:02] Adding non-codec 0x1 (telephone-event) to SDP [Oct 23 23:54:02] Reliably Transmitting (NAT) to 66.33.147.150:5060: INVITE sip:12503802844@66.33.147.150;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1785cdda;rport From: "Test Admin Phone" ;tag=as4f7f71f2 To: Contact: Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Test Admin Phone" ;privacy=off;screen=no Date: Mon, 24 Oct 2011 06:54:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 213 v=0 o=root 3322 3322 IN IP4 192.168.1.135 s=session c=IN IP4 192.168.1.135 t=0 0 m=audio 10368 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Oct 23 23:54:02] -- Called switch2voip/12503802844 [Oct 23 23:54:02] <--- SIP read from 66.33.147.150:5060 ---> SIP/2.0 407 Unauthorized Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1785cdda;received=24.69.86.249;rport=5060 From: "Test Admin Phone" ;tag=as4f7f71f2 To: Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 102 INVITE Contact: Server: Net2Phone Carrier Proxy-Authenticate: Digest realm="net2phone",nonce="B559E9EFA3C530BAB4FFC886A27A1556" Content-Length: 0 <-------------> [Oct 23 23:54:02] --- (10 headers 0 lines) --- [Oct 23 23:54:02] Transmitting (NAT) to 66.33.147.150:5060: ACK sip:12503802844@66.33.147.150;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1785cdda;rport From: "Test Admin Phone" ;tag=as4f7f71f2 To: Contact: Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Test Admin Phone" ;privacy=off;screen=no Content-Length: 0 --- [Oct 23 23:54:02] Audio is at 192.168.1.135 port 10368 [Oct 23 23:54:02] Adding codec 0x4 (ulaw) to SDP [Oct 23 23:54:02] Adding non-codec 0x1 (telephone-event) to SDP [Oct 23 23:54:02] Reliably Transmitting (NAT) to 66.33.147.150:5060: INVITE sip:12503802844@66.33.147.150;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;rport From: "Test Admin Phone" ;tag=as4f7f71f2 To: Contact: Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Test Admin Phone" ;privacy=off;screen=no Proxy-Authorization: Digest username="5208913412", realm="net2phone", algorithm=MD5, uri="sip:12503802844@66.33.147.150;cpd=on", nonce="B559E9EFA3C530BAB4FFC886A27A1556", response="2041179d24a32fa1e36943232c172e77" Date: Mon, 24 Oct 2011 06:54:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 213 v=0 o=root 3322 3323 IN IP4 192.168.1.135 s=session c=IN IP4 192.168.1.135 t=0 0 m=audio 10368 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Oct 23 23:54:02] <--- SIP read from 66.33.147.150:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;received=24.69.86.249;rport=5060 From: "Test Admin Phone" ;tag=as4f7f71f2 To: ;tag=ccid-188300447-1-1851 Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 103 INVITE Contact: Server: Net2Phone Carrier Content-Length: 0 <-------------> [Oct 23 23:54:02] --- (9 headers 0 lines) --- [Oct 23 23:54:04] <--- SIP read from 66.33.147.150:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;received=24.69.86.249;rport=5060 From: "Test Admin Phone" ;tag=as4f7f71f2 To: ;tag=ccid-188300447-1-1851 Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 103 INVITE Contact: Server: Net2Phone Carrier Content-Length: 214 Content-Type: application/sdp v=0 o=5208913412 188300447 188300447 IN IP4 216.53.10.3 s=SIP Call c=IN IP4 216.53.10.3 t=0 0 m=audio 20798 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Oct 23 23:54:04] --- (10 headers 10 lines) --- [Oct 23 23:54:04] Found RTP audio format 0 [Oct 23 23:54:04] Found RTP audio format 101 [Oct 23 23:54:04] Found audio description format PCMU for ID 0 [Oct 23 23:54:04] Found audio description format telephone-event for ID 101 [Oct 23 23:54:04] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Oct 23 23:54:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Oct 23 23:54:04] Peer audio RTP is at port 216.53.10.3:20798 [Oct 23 23:54:04] -- SIP/switch2voip-0000009c is making progress passing it to SIP/gs102-0000009b [Oct 23 23:54:05] <--- SIP read from 66.33.147.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;received=24.69.86.249;rport=5060 From: "Test Admin Phone" ;tag=as4f7f71f2 To: ;tag=ccid-188300447-1-1851 Allow: ACK,BYE,CANCEL,INVITE,OPTIONS Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 103 INVITE Contact: Server: Net2Phone Carrier Content-Length: 214 ontent-Type: application/sdp v=0 o=5208913412 188300447 188300447 IN IP4 216.53.10.3 s=SIP Call c=IN IP4 216.53.10.3 t=0 0 m=audio 20798 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Oct 23 23:54:05] --- (11 headers 10 lines) --- [Oct 23 23:54:05] Found RTP audio format 0 [Oct 23 23:54:05] Found RTP audio format 101 [Oct 23 23:54:05] Found audio description format PCMU for ID 0 [Oct 23 23:54:05] Found audio description format telephone-event for ID 101 [Oct 23 23:54:05] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Oct 23 23:54:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Oct 23 23:54:05] Peer audio RTP is at port 216.53.10.3:20798 [Oct 23 23:54:05] list_route: hop: [Oct 23 23:54:05] set_destination: Parsing for address/port to send to [Oct 23 23:54:05] set_destination: set destination to 66.33.147.150, port 5060 [Oct 23 23:54:05] Transmitting (NAT) to 66.33.147.150:5060: ACK sip:66.33.147.150:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK2400bf41;rport From: "Test Admin Phone" ;tag=as4f7f71f2 To: ;tag=ccid-188300447-1-1851 Contact: Call-ID: 2d52e4021ef494dc39b270ce1917da19@192.168.1.135 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Test Admin Phone" ;privacy=off;screen=no Content-Length: 0 --- [Oct 23 23:54:05] -- SIP/switch2voip-0000009c answered SIP/gs102-0000009b [Oct 23 23:54:06] == Parsing '/etc/asterisk/manager.conf': [Oct 23 23:54:06] Found [Oct 23 23:54:06] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 23 23:54:06] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 23 23:54:10] Reliably Transmitting (NAT) to 66.33.147.150:5060: OPTIONS sip:66.33.147.150;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1e9cab9b;rport From: "asterisk" ;tag=as50188c95 To: Contact: Call-ID: 392b03f10af210f453ddff55729be94f@192.168.1.135 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 24 Oct 2011 06:54:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Oct 23 23:54:10] <--- SIP read from 66.33.147.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1e9cab9b;received=24.69.86.249;rport=5060 From: "asterisk" ;tag=as50188c95 To: Call-ID: 392b03f10af210f453ddff55729be94f@192.168.1.135 CSeq: 102 OPTIONS Contact: Server: Net2Phone Carrier Content-Length: 0 <-------------> [Oct 23 23:54:10] --- (9 headers 0 lines) --- [Oct 23 23:54:10] Really destroying SIP dialog '392b03f10af210f453ddff55729be94f@192.168.1.135' Method: OPTIONS vicibox*CLI>