[Jun 12 09:44:11] DEBUG[18144] chan_sip.c: Auto destroying SIP dialog '583cea7946a25ab80f497cdf4ad7a932@130.10.160.5:5060' [Jun 12 09:44:11] DEBUG[18144] chan_sip.c: Destroying SIP dialog 583cea7946a25ab80f497cdf4ad7a932@130.10.160.5:5060 [Jun 12 09:44:12] VERBOSE[18133] asterisk.c: -- Remote UNIX connection [Jun 12 09:44:12] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 5f4a7b2c539a7ead1f1609813eefbbf2@130.10.20.220:5060 - OPTIONS (No RTP) [Jun 12 09:44:12] DEBUG[18144] acl.c: For destination '10.0.0.1', our source address is '130.10.20.220'. [Jun 12 09:44:12] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:44:12] DEBUG[18144] chan_sip.c: Initializing initreq for method OPTIONS - callid 7ade89476ba030361356f0b11e723f60@130.10.20.220:5060 [Jun 12 09:44:12] DEBUG[18144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.1:5060 [Jun 12 09:44:12] DEBUG[18144] chan_sip.c: Stopping retransmission on '7ade89476ba030361356f0b11e723f60@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:44:12] DEBUG[18144] chan_sip.c: Destroying SIP dialog 7ade89476ba030361356f0b11e723f60@130.10.20.220:5060 [Jun 12 09:44:13] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:44:15] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:44:16] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:44:17] DEBUG[18144] chan_sip.c: Auto destroying SIP dialog '9ff10e9a48678ae4' [Jun 12 09:44:17] DEBUG[18144] chan_sip.c: Destroying SIP dialog 9ff10e9a48678ae4 [Jun 12 09:44:17] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0xb6f65bf8' [Jun 12 09:44:17] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0xb6f45628' [Jun 12 09:44:26] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 051893d73690626c5cfd08d30f11084d@130.10.20.220:5060 - OPTIONS (No RTP) [Jun 12 09:44:26] DEBUG[18144] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:44:26] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:44:26] DEBUG[18144] chan_sip.c: Initializing initreq for method OPTIONS - callid 5a662ade3ca511cd06bfe355616a7da9@130.10.20.220:5060 [Jun 12 09:44:26] DEBUG[18144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:26] DEBUG[18144] chan_sip.c: = Looking for Call ID: 5a662ade3ca511cd06bfe355616a7da9@130.10.20.220:5060 (Checking To) --From tag as22cd5a7e --To-tag as0cc661bb [Jun 12 09:44:26] DEBUG[18144] chan_sip.c: Stopping retransmission on '5a662ade3ca511cd06bfe355616a7da9@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:44:26] DEBUG[18144] chan_sip.c: Destroying SIP dialog 5a662ade3ca511cd06bfe355616a7da9@130.10.20.220:5060 [Jun 12 09:44:28] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:44:30] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:44:31] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:10.0.0.155:5060 ---> INVITE sip:2204@130.10.20.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKd38b205e6fc6e4060 Max-Forwards: 70 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: Call-ID: 791e658937df8782 CSeq: 1151097176 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE Allow-Events: talk, hold Contact: "2202" Supported: replaces, 100rel User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 341 v=0 o=OpenStage-Line_0 2046032545 1044122655 IN IP4 10.0.0.155 s=SIP Call c=IN IP4 10.0.0.155 t=0 0 m=audio 5010 RTP/AVP 8 0 9 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv <-------------> [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 0 [ 56]: INVITE sip:2204@130.10.20.220:5060;transport=udp SIP/2.0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKd38b205e6fc6e4060 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 3 [ 71]: From: "2202" ;tag=8d46f5140b;epid=SC309517 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 4 [ 33]: To: [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 5 [ 25]: Call-ID: 791e658937df8782 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 6 [ 23]: CSeq: 1151097176 INVITE [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 7 [ 54]: Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 8 [ 24]: Allow-Events: talk, hold [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 9 [ 56]: Contact: "2202" [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 10 [ 27]: Supported: replaces, 100rel [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 11 [ 46]: User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 12 [ 32]: X-Siemens-Call-Type: ST-insecure [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 14 [ 19]: Content-Length: 341 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 15 [ 0]: [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 0 [ 3]: v=0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 1 [ 58]: o=OpenStage-Line_0 2046032545 1044122655 IN IP4 10.0.0.155 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.155 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 5 [ 33]: m=audio 5010 RTP/AVP 8 0 9 18 101 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 8 [ 20]: a=rtpmap:9 G722/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 12 [ 19]: a=fmtp:18 annexb=no [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-15 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 14 [ 10]: a=sendrecv [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: --- (15 headers 15 lines) --- [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: = Looking for Call ID: 791e658937df8782 (Checking From) --From tag 8d46f5140b --To-tag [Jun 12 09:44:37] DEBUG[18144] acl.c: For destination '10.0.0.155', our source address is '130.10.20.220'. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 791e658937df8782 - INVITE (No RTP) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 12 09:44:37] DEBUG[18144] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel" [Jun 12 09:44:37] DEBUG[18144] sip/reqresp_parser.c: Found SIP option: -replaces- [Jun 12 09:44:37] DEBUG[18144] sip/reqresp_parser.c: Matched SIP option: replaces [Jun 12 09:44:37] DEBUG[18144] sip/reqresp_parser.c: Found SIP option: -100rel- [Jun 12 09:44:37] DEBUG[18144] sip/reqresp_parser.c: Matched SIP option: 100rel [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '10.0.0.155' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '10.0.0.155' and port ''. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Sending to 10.0.0.155:5060 (NAT) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Initializing initreq for method INVITE - callid 791e658937df8782 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Using INVITE request as basis request - 791e658937df8782 [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '130.10.20.220:5060' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found peer '2202' for '2202' from 10.0.0.155:5060 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: <--- Reliably Transmitting (NAT) to 10.0.0.155:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKd38b205e6fc6e4060;received=10.0.0.155;rport=5060 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: ;tag=as6e6a5277 Call-ID: 791e658937df8782 CSeq: 1151097176 INVITE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="396f50c0" Content-Length: 0 <------------> [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8604 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.0.155:5060 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Scheduling destruction of SIP dialog '791e658937df8782' in 32000 ms (Method: INVITE) [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:10.0.0.155:5060 ---> ACK sip:2204@130.10.20.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKd38b205e6fc6e4060 Max-Forwards: 70 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: ;tag=as6e6a5277 Call-ID: 791e658937df8782 CSeq: 1151097176 ACK User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 Content-Length: 0 <-------------> [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 0 [ 53]: ACK sip:2204@130.10.20.220:5060;transport=udp SIP/2.0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKd38b205e6fc6e4060 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 3 [ 71]: From: "2202" ;tag=8d46f5140b;epid=SC309517 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 4 [ 48]: To: ;tag=as6e6a5277 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 5 [ 25]: Call-ID: 791e658937df8782 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 6 [ 20]: CSeq: 1151097176 ACK [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 7 [ 46]: User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: --- (9 headers 0 lines) --- [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: = Looking for Call ID: 791e658937df8782 (Checking From) --From tag 8d46f5140b --To-tag as6e6a5277 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8604 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Stopping retransmission on '791e658937df8782' of Response 1151097176: Match Found [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:10.0.0.155:5060 ---> INVITE sip:2204@130.10.20.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKb116b96afeb771c3b Max-Forwards: 70 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: Call-ID: 791e658937df8782 CSeq: 1151097177 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE Allow-Events: talk, hold Authorization: Digest username="2202",realm="asterisk",nonce="396f50c0",uri="sip:2204@130.10.20.220:5060;transport=udp",response="e9b6992df752a348d0ad45f0693aec4e",algorithm=MD5 Contact: "2202" Supported: replaces, 100rel User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 341 v=0 o=OpenStage-Line_0 2046032545 1044122655 IN IP4 10.0.0.155 s=SIP Call c=IN IP4 10.0.0.155 t=0 0 m=audio 5010 RTP/AVP 8 0 9 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv <-------------> [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 0 [ 56]: INVITE sip:2204@130.10.20.220:5060;transport=udp SIP/2.0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKb116b96afeb771c3b [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 3 [ 71]: From: "2202" ;tag=8d46f5140b;epid=SC309517 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 4 [ 33]: To: [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 5 [ 25]: Call-ID: 791e658937df8782 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 6 [ 23]: CSeq: 1151097177 INVITE [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 7 [ 54]: Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 8 [ 24]: Allow-Events: talk, hold [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 9 [177]: Authorization: Digest username="2202",realm="asterisk",nonce="396f50c0",uri="sip:2204@130.10.20.220:5060;transport=udp",response="e9b6992df752a348d0ad45f0693aec4e",algorithm=MD5 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 10 [ 56]: Contact: "2202" [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 11 [ 27]: Supported: replaces, 100rel [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 12 [ 46]: User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 13 [ 32]: X-Siemens-Call-Type: ST-insecure [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 15 [ 19]: Content-Length: 341 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 16 [ 0]: [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 0 [ 3]: v=0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 1 [ 58]: o=OpenStage-Line_0 2046032545 1044122655 IN IP4 10.0.0.155 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.155 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 5 [ 33]: m=audio 5010 RTP/AVP 8 0 9 18 101 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 8 [ 20]: a=rtpmap:9 G722/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 12 [ 19]: a=fmtp:18 annexb=no [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-15 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Body 14 [ 10]: a=sendrecv [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: --- (16 headers 15 lines) --- [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: = Looking for Call ID: 791e658937df8782 (Checking From) --From tag 8d46f5140b --To-tag [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '130.10.20.220:5060' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port '5060'. [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '130.10.20.220:5060' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port '5060'. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '10.0.0.155' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '10.0.0.155' and port ''. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Sending to 10.0.0.155:5060 (NAT) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Initializing initreq for method INVITE - callid 791e658937df8782 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Using INVITE request as basis request - 791e658937df8782 [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '130.10.20.220:5060' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found peer '2202' for '2202' from 10.0.0.155:5060 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6f33588' [Jun 12 09:44:37] DEBUG[18144] res_rtp_asterisk.c: Allocated port 16786 for RTP instance '0xb6f33588' [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: RTP instance '0xb6f33588' is setup and ready to go [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6f76c00' [Jun 12 09:44:37] DEBUG[18144] res_rtp_asterisk.c: Allocated port 15700 for RTP instance '0xb6f76c00' [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: RTP instance '0xb6f76c00' is setup and ready to go [Jun 12 09:44:37] DEBUG[18144] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6f76c00' [Jun 12 09:44:37] VERBOSE[18144] netsock2.c: == Using SIP VIDEO TOS bits 136 [Jun 12 09:44:37] VERBOSE[18144] netsock2.c: == Using SIP VIDEO CoS mark 6 [Jun 12 09:44:37] DEBUG[18144] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6f33588' [Jun 12 09:44:37] VERBOSE[18144] netsock2.c: == Using SIP RTP TOS bits 184 [Jun 12 09:44:37] VERBOSE[18144] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Setting NAT on RTP to Off [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Setting NAT on VRTP to Off [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing session-level SDP o=OpenStage-Line_0 2046032545 1044122655 IN IP4 10.0.0.155... OK. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '10.0.0.155' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '10.0.0.155' and port ''. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.155... OK. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found RTP audio format 8 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Setting payload 8 based on m type on 0xb74daf64 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found RTP audio format 0 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Setting payload 0 based on m type on 0xb74daf64 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found RTP audio format 9 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Setting payload 9 based on m type on 0xb74daf64 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found RTP audio format 18 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Setting payload 18 based on m type on 0xb74daf64 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found RTP audio format 101 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Setting payload 101 based on m type on 0xb74daf64 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found audio description format G722 for ID 9 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found audio description format G729 for ID 18 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED OR FAILED. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Incorporating payload 0 on 0xb74daf64 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Incorporating payload 8 on 0xb74daf64 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Incorporating payload 9 on 0xb74daf64 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Incorporating payload 18 on 0xb74daf64 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Incorporating payload 101 on 0xb74daf64 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Capabilities: us - 0x3c1008 (alaw|g722|h261|h263|h263p|h264), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1008 (alaw|g722) [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 12 09:44:37] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f33588' [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Peer audio RTP is at port 10.0.0.155:5010 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Copying payload 0 from 0xb74daf64 to 0xb6f33734 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Copying payload 8 from 0xb74daf64 to 0xb6f33734 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Copying payload 9 from 0xb74daf64 to 0xb6f33734 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Copying payload 18 from 0xb74daf64 to 0xb6f33734 [Jun 12 09:44:37] DEBUG[18144] rtp_engine.c: Copying payload 101 from 0xb74daf64 to 0xb6f33734 [Jun 12 09:44:37] DEBUG[18144] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb6f33588' [Jun 12 09:44:37] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f76c00' [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Peer doesn't provide video [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: We're settling with these formats: 0x1008 (alaw|g722) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Checking SIP call limits for device 2202 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Updating call counter for incoming call [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Call from peer '2202' is 1 out of 4 [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '130.10.20.220:5060' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:37] DEBUG[18144] netsock2.c: Splitting '130.10.20.220:5060' into... [Jun 12 09:44:37] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: Looking for 2204 in from-internal-4803977-acl (domain 130.10.20.220) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: *** Our native formats are 0x1000 (g722) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: *** Joint capabilities are 0x1008 (alaw|g722) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: *** Our capabilities are 0x3c1008 (alaw|g722|h261|h263|h263p|h264) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: build_route: Contact hop: "2202" [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: list_route: hop: [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: SIP/2202-00000015: New call is still down.... Trying... [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.155:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKb116b96afeb771c3b;received=10.0.0.155;rport=5060 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: Call-ID: 791e658937df8782 CSeq: 1151097177 INVITE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.0.155:5060 [Jun 12 09:44:37] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - 2202 [Jun 12 09:44:37] DEBUG[18136] chan_sip.c: Checking device state for peer 2202 [Jun 12 09:44:37] DEBUG[18136] devicestate.c: Changing state for SIP/2202 - state 3 (Busy) [Jun 12 09:44:37] DEBUG[18136] devicestate.c: device 'SIP/2202' state '3' [Jun 12 09:44:37] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - 2202 [Jun 12 09:44:37] DEBUG[18136] chan_sip.c: Checking device state for peer 2202 [Jun 12 09:44:37] DEBUG[18136] devicestate.c: Changing state for SIP/2202 - state 3 (Busy) [Jun 12 09:44:37] DEBUG[18136] devicestate.c: device 'SIP/2202' state '3' [Jun 12 09:44:37] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:37] DEBUG[18161] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2202 CallerIDName: Sebastian Kirchner AccountCode: Exten: 2204 Context: from-internal-4803977-acl Uniqueid: 1371023077.22 [Jun 12 09:44:37] DEBUG[18161] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2202 CallerIDName: Sebastian Kirchner ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023077.22 [Jun 12 09:44:37] DEBUG[19616] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2202 CallerIDName: Sebastian Kirchner AccountCode: Exten: 2204 Context: from-internal-4803977-acl Uniqueid: 1371023077.22 [Jun 12 09:44:37] DEBUG[19616] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2202 CallerIDName: Sebastian Kirchner ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023077.22 [Jun 12 09:44:37] DEBUG[20059] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2202 CallerIDName: Sebastian Kirchner AccountCode: Exten: 2204 Context: from-internal-4803977-acl Uniqueid: 1371023077.22 [Jun 12 09:44:37] DEBUG[20059] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2202 CallerIDName: Sebastian Kirchner ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023077.22 [Jun 12 09:44:37] DEBUG[20571] pbx.c: Function result is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'EXTEN' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:1] NoOp("SIP/2202-00000015", "ACL check for 2202 - destination 2204") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Function result is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:2] Set("SIP/2202-00000015", "_CALLER=2202") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'EXTEN' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:3] Set("SIP/2202-00000015", "CALLEDNUMBER=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Macro' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:4] Macro("SIP/2202-00000015", "executeagi,set-callerid.agi") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:1] NoOp("SIP/2202-00000015", "Execute AGI") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:2] GotoIf("SIP/2202-00000015", "0?agihost0failed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is 'set-callerid.agi' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'AGI' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:3] AGI("SIP/2202-00000015", "agi://130.10.14.249/set-callerid.agi") in new stack [Jun 12 09:44:37] DEBUG[20571] res_agi.c: Wow, connected! [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLER' is '2202' [Jun 12 09:44:37] DEBUG[18161] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/2202-00000015 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) Uniqueid: 1371023077.22 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:37] DEBUG[19616] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/2202-00000015 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) Uniqueid: 1371023077.22 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:37] DEBUG[20059] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/2202-00000015 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) Uniqueid: 1371023077.22 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:37] VERBOSE[20571] res_agi.c: -- AGI Script agi://130.10.14.249/set-callerid.agi completed, returning 0 [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: AGI [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:4] GotoIf("SIP/2202-00000015", "1?agisuccess") in new stack [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Goto (macro-executeagi,s,7) [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:7] NoOp("SIP/2202-00000015", "AGI SUCCESS") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLER' is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'SIPAddHeader' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:5] SIPAddHeader("SIP/2202-00000015", "Call-Info: ;purpose=icon") in new stack [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: SIP Header added "Call-Info: ;purpose=icon" as __SIPADDHEADER01 [Jun 12 09:44:37] DEBUG[20571] pbx.c: Function result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:6] NoOp("SIP/2202-00000015", "REDIRECTINGCOUNT = 0") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'FORWARDARRAY' is NULL [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:7] NoOp("SIP/2202-00000015", "FORWARDARRAY = ") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Function result is '' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Function result is '' [Jun 12 09:44:37] DEBUG[20571] func_strings.c: c1=48, c2=57 [Jun 12 09:44:37] DEBUG[20571] func_strings.c: Allowed: 0123456789 [Jun 12 09:44:37] DEBUG[20571] pbx.c: Function result is '' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:8] GotoIf("SIP/2202-00000015", "1?acl") in new stack [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Goto (from-internal-4803977-acl,2204,14) [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLER' is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDNUMBER' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Macro' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:14] Macro("SIP/2202-00000015", "executeagi,acl.agi?user=2202&callednum=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:1] NoOp("SIP/2202-00000015", "Execute AGI") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:2] GotoIf("SIP/2202-00000015", "0?agihost0failed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is 'acl.agi?user=2202&callednum=2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'AGI' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:3] AGI("SIP/2202-00000015", "agi://130.10.14.249/acl.agi?user=2202&callednum=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] res_agi.c: Wow, connected! [Jun 12 09:44:37] VERBOSE[20571] res_agi.c: -- AGI Script agi://130.10.14.249/acl.agi?user=2202&callednum=2204 completed, returning 0 [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: AGI [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:4] GotoIf("SIP/2202-00000015", "1?agisuccess") in new stack [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Goto (macro-executeagi,s,7) [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:7] NoOp("SIP/2202-00000015", "AGI SUCCESS") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLER' is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDNUMBER' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Macro' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:15] Macro("SIP/2202-00000015", "executeagi,diallog.agi?user=2202&type=0&number=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:1] NoOp("SIP/2202-00000015", "Execute AGI") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:2] GotoIf("SIP/2202-00000015", "0?agihost0failed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is 'diallog.agi?user=2202&type=0&number=2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'AGI' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:3] AGI("SIP/2202-00000015", "agi://130.10.14.249/diallog.agi?user=2202&type=0&number=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] res_agi.c: Wow, connected! [Jun 12 09:44:37] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:37] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - soft2202 [Jun 12 09:44:37] DEBUG[18137] chan_sip.c: Checking device state for peer soft2202 [Jun 12 09:44:37] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - web2202 [Jun 12 09:44:37] DEBUG[18137] chan_sip.c: Checking device state for peer web2202 [Jun 12 09:44:37] DEBUG[18137] app_queue.c: Extension '2202@hints' changed to state '3' (Busy) but we don't care because they're not a member of any queue. [Jun 12 09:44:37] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - soft2202 [Jun 12 09:44:37] DEBUG[18137] chan_sip.c: Checking device state for peer soft2202 [Jun 12 09:44:37] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - web2202 [Jun 12 09:44:37] DEBUG[18137] chan_sip.c: Checking device state for peer web2202 [Jun 12 09:44:37] DEBUG[18171] app_queue.c: Device 'SIP/2202' changed to state '3' (Busy) [Jun 12 09:44:37] DEBUG[18171] app_queue.c: Device 'SIP/2202' changed to state '3' (Busy) [Jun 12 09:44:37] DEBUG[18161] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2202 Context: hints Hint: SIP/2202&SIP/soft2202&SIP/web2202 Status: 2 [Jun 12 09:44:37] DEBUG[19616] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2202 Context: hints Hint: SIP/2202&SIP/soft2202&SIP/web2202 Status: 2 [Jun 12 09:44:37] DEBUG[20059] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2202 Context: hints Hint: SIP/2202&SIP/soft2202&SIP/web2202 Status: 2 [Jun 12 09:44:37] VERBOSE[20571] res_agi.c: -- AGI Script agi://130.10.14.249/diallog.agi?user=2202&type=0&number=2204 completed, returning 0 [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: AGI [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:4] GotoIf("SIP/2202-00000015", "1?agisuccess") in new stack [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Goto (macro-executeagi,s,7) [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:7] NoOp("SIP/2202-00000015", "AGI SUCCESS") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDNUMBER' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Goto' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977-acl:16] Goto("SIP/2202-00000015", "from-internal-4803977,2204,1") in new stack [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Goto (from-internal-4803977,2204,1) [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLER' is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'EXTEN' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Gosub' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [2204@from-internal-4803977:1] Gosub("SIP/2202-00000015", "lookup-called-user,s,1(2202,2204,6,130.10.20.220,asterisk-GuruPlug)") in new stack [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Channel SIP/2202-00000015 has no datastore, so we're allocating one. [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG1' to '2202' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG2' to '2204' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG3' to '6' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG4' to '130.10.20.220' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG5' to 'asterisk-GuruPlug' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG2' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:1] Set("SIP/2202-00000015", "CALLEDNUMBER=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG2' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG2' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG2' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GosubIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:2] GosubIf("SIP/2202-00000015", "1?lookup-calling-party,s,1(2202):serverfailed") in new stack [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG1' to '2202' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG2' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG3' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG4' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG5' to '' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is '2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Macro' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-calling-party:1] Macro("SIP/2202-00000015", "executeagi,lookup.agi?user=2202") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:1] NoOp("SIP/2202-00000015", "Execute AGI") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:2] GotoIf("SIP/2202-00000015", "0?agihost0failed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is 'lookup.agi?user=2202' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'AGI' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:3] AGI("SIP/2202-00000015", "agi://130.10.14.249/lookup.agi?user=2202") in new stack [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[20571] res_agi.c: Wow, connected! [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] VERBOSE[20571] res_agi.c: -- AGI Script agi://130.10.14.249/lookup.agi?user=2202 completed, returning 0 [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: AGI [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:4] GotoIf("SIP/2202-00000015", "1?agisuccess") in new stack [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Goto (macro-executeagi,s,7) [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:7] NoOp("SIP/2202-00000015", "AGI SUCCESS") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'OUTGOINGSERVERID' is '6' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-calling-party:2] Set("SIP/2202-00000015", "CALLINGPARTYOUTGOINGSERVERID=6") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'OUTGOINGSERVERIP' is '130.10.20.220' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-calling-party:3] Set("SIP/2202-00000015", "CALLINGPARTYOUTGOINGSERVERIP=130.10.20.220") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'OUTGOINGSERVERNAME' is 'asterisk-asteriskdev2' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-calling-party:4] Set("SIP/2202-00000015", "CALLINGPARTYOUTGOINGSERVERNAME=asterisk-asteriskdev2") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLINGPARTYOUTGOINGSERVERID' is '6' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLINGPARTYOUTGOINGSERVERIP' is '130.10.20.220' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLINGPARTYOUTGOINGSERVERNAME' is 'asterisk-asteriskdev2' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-calling-party:5] GotoIf("SIP/2202-00000015", "0?lookupfailed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Return' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-calling-party:6] Return("SIP/2202-00000015", "1") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'GOSUB_RETVAL' is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:3] Set("SIP/2202-00000015", "CALLINGPARTYEXISTS=1") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDNUMBER' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Gosub' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:4] Gosub("SIP/2202-00000015", "lookup-called-party,s,1(2204)") in new stack [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG1' to '2204' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG2' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG3' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG4' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG5' to '' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:1] Set("SIP/2202-00000015", "LOGGEDINSERVERID=") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:2] Set("SIP/2202-00000015", "LOGGEDINSERVERIP=") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:3] Set("SIP/2202-00000015", "LOGGEDINSERVERNAME=") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Macro' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:4] Macro("SIP/2202-00000015", "executeagi,lookup.agi?user=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:1] NoOp("SIP/2202-00000015", "Execute AGI") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:2] GotoIf("SIP/2202-00000015", "0?agihost0failed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'ARG1' is 'lookup.agi?user=2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'AGI' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:3] AGI("SIP/2202-00000015", "agi://130.10.14.249/lookup.agi?user=2204") in new stack [Jun 12 09:44:37] DEBUG[20571] res_agi.c: Wow, connected! [Jun 12 09:44:37] VERBOSE[20571] res_agi.c: -- AGI Script agi://130.10.14.249/lookup.agi?user=2204 completed, returning 0 [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: AGI [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:4] GotoIf("SIP/2202-00000015", "1?agisuccess") in new stack [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Goto (macro-executeagi,s,7) [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: GotoIf [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@macro-executeagi:7] NoOp("SIP/2202-00000015", "AGI SUCCESS") in new stack [Jun 12 09:44:37] DEBUG[20571] app_macro.c: Executed application: NoOp [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'LOGGEDINSERVERID' is '9' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:5] Set("SIP/2202-00000015", "CALLEDPARTYLOGGEDINSERVERID=9") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'LOGGEDINSERVERIP' is '130.10.160.5' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:6] Set("SIP/2202-00000015", "CALLEDPARTYLOGGEDINSERVERIP=130.10.160.5") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'LOGGEDINSERVERNAME' is 'asterisk-GuruPlug' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:7] Set("SIP/2202-00000015", "CALLEDPARTYLOGGEDINSERVERNAME=asterisk-GuruPlug") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDPARTYLOGGEDINSERVERID' is '9' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDPARTYLOGGEDINSERVERIP' is '130.10.160.5' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDPARTYLOGGEDINSERVERNAME' is 'asterisk-GuruPlug' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:8] GotoIf("SIP/2202-00000015", "0?lookupfailed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Return' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-party:9] Return("SIP/2202-00000015", "1") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'GOSUB_RETVAL' is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:5] Set("SIP/2202-00000015", "CALLEDPARTYEXISTS=1") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDPARTYEXISTS' is '1' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:6] GotoIf("SIP/2202-00000015", "0?failed") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDPARTYLOGGEDINSERVERID' is '9' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'SERVERID' is '6' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Expression result is '0' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'GotoIf' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:7] GotoIf("SIP/2202-00000015", "0?diallocal") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Not taking any branch [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDPARTYLOGGEDINSERVERNAME' is 'asterisk-GuruPlug' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Set' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:8] Set("SIP/2202-00000015", "CALLEDSERVER=asterisk-GuruPlug") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Gosub' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@lookup-called-user:9] Gosub("SIP/2202-00000015", "to-node,s,1") in new stack [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG1' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG2' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG3' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG4' to '' [Jun 12 09:44:37] DEBUG[20571] app_stack.c: Setting 'ARG5' to '' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'NoOp' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@to-node:1] NoOp("SIP/2202-00000015", "HangUp") in new stack [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDSERVER' is 'asterisk-GuruPlug' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Result of 'CALLEDNUMBER' is '2204' [Jun 12 09:44:37] DEBUG[20571] pbx.c: Launching 'Dial' [Jun 12 09:44:37] VERBOSE[20571] pbx.c: -- Executing [s@to-node:2] Dial("SIP/2202-00000015", "SIP/asterisk-GuruPlug/2204,,o") in new stack [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Asked to create a SIP channel with formats: 0x1000 (g722) [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Allocating new SIP dialog for 661765f005b48b0d61c0f3990e835ab3@130.10.20.220:5060 - INVITE (No RTP) [Jun 12 09:44:37] DEBUG[20571] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9b980d8' [Jun 12 09:44:37] DEBUG[20571] res_rtp_asterisk.c: Allocated port 13774 for RTP instance '0x9b980d8' [Jun 12 09:44:37] DEBUG[20571] rtp_engine.c: RTP instance '0x9b980d8' is setup and ready to go [Jun 12 09:44:37] DEBUG[20571] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x94e23c8' [Jun 12 09:44:37] DEBUG[20571] res_rtp_asterisk.c: Allocated port 11698 for RTP instance '0x94e23c8' [Jun 12 09:44:37] DEBUG[20571] rtp_engine.c: RTP instance '0x94e23c8' is setup and ready to go [Jun 12 09:44:37] DEBUG[20571] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x94e23c8' [Jun 12 09:44:37] VERBOSE[20571] netsock2.c: == Using SIP VIDEO TOS bits 136 [Jun 12 09:44:37] VERBOSE[20571] netsock2.c: == Using SIP VIDEO CoS mark 6 [Jun 12 09:44:37] DEBUG[20571] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9b980d8' [Jun 12 09:44:37] VERBOSE[20571] netsock2.c: == Using SIP RTP TOS bits 184 [Jun 12 09:44:37] VERBOSE[20571] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Setting NAT on RTP to Off [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Setting NAT on VRTP to Off [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 12 09:44:37] DEBUG[20571] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: *** Our native formats are 0x2003c0000 (speex16|h261|h263|h263p|h264) [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: *** Our capabilities are 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x200000000 (speex16) [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: *** Our preferred formats from the incoming channel are 0x1000 (g722) [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jun 12 09:44:37] DEBUG[20571] rtp_engine.c: Seeded SDP of 'SIP/asterisk-GuruPlug-00000016' with that of 'SIP/2202-00000015' [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable DIALEDTIME. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ANSWEREDTIME. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable DIALEDPEERNAME. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable DIALEDPEERNUMBER. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable DIALSTATUS. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARGC. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG5. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG4. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG3. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG2. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG1. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLEDSERVER. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLEDPARTYEXISTS. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable GOSUB_RETVAL. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLEDPARTYLOGGEDINSERVERNAME. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLEDPARTYLOGGEDINSERVERIP. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLEDPARTYLOGGEDINSERVERID. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable MACRO_DEPTH. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable AGISTATUS. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable AJ_AGISTATUS. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable LOGGEDINSERVERIP. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable LOGGEDINSERVERNAME. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable LOGGEDINSERVERID. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable OUTGOINGSERVERIP. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable OUTGOINGSERVERNAME. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable OUTGOINGSERVERID. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLINGPARTYEXISTS. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLINGPARTYOUTGOINGSERVERNAME. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLINGPARTYOUTGOINGSERVERIP. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLINGPARTYOUTGOINGSERVERID. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable CALLEDNUMBER. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARGC. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG5. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG4. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG3. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG2. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable ARG1. [Jun 12 09:44:37] DEBUG[20571] channel.c: Copying hard-transferable variable SIPADDHEADER01. [Jun 12 09:44:37] DEBUG[20571] channel.c: Copying soft-transferable variable CALLER. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable SIPCALLID. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable SIPDOMAIN. [Jun 12 09:44:37] DEBUG[20571] channel.c: Not copying variable SIPURI. [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Outgoing Call for 2204 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Updating call counter for outgoing call [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Call to peer 'asterisk-GuruPlug' is 1 out of 2147483647 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Adding SIP Header "Call-Info" with content :;purpose=icon: [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: This call needs video offers! [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: ** Our capability: 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) Video flag: False Text flag: False [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: ** Our prefcodec: 0x1000 (g722) [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Audio is at 13774 [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Video is at 130.10.20.220:11698 [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Adding codec 0x200000000 (speex16) to SDP [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Adding codec 0x200 (speex) to SDP [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Adding video codec 0x40000 (h261) to SDP [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Adding video codec 0x80000 (h263) to SDP [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Adding video codec 0x200000 (h264) to SDP [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: -- Done with adding codecs to SDP [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Done building SDP. Settling with this capability: 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Initializing initreq for method INVITE - callid 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 0 [ 36]: INVITE sip:2204@130.10.160.5 SIP/2.0 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;rport [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 3 [ 73]: From: "Sebastian Kirchner (2202)" ;tag=as1935e828 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 4 [ 27]: To: [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 5 [ 38]: Contact: [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 6 [ 60]: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 8 [ 24]: User-Agent: asteriskdev2 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 9 [ 35]: Date: Wed, 12 Jun 2013 07:44:37 GMT [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 12 [ 80]: Call-Info: ;purpose=icon [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 13 [ 73]: P-Asserted-Identity: "Sebastian Kirchner (2202)" [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jun 12 09:44:37] VERBOSE[20571] chan_sip.c: Reliably Transmitting (NAT) to 130.10.160.5:5060: INVITE sip:2204@130.10.160.5 SIP/2.0 Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;rport Max-Forwards: 70 From: "Sebastian Kirchner (2202)" ;tag=as1935e828 To: Contact: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 102 INVITE User-Agent: asteriskdev2 Date: Wed, 12 Jun 2013 07:44:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Call-Info: ;purpose=icon P-Asserted-Identity: "Sebastian Kirchner (2202)" Content-Type: application/sdp Content-Length: 444 v=0 o=root 725587763 725587763 IN IP4 130.10.20.220 s=Asterisk PBX 1.8.20.1~dfsg-1ubuntu1 c=IN IP4 130.10.20.220 b=CT:128 t=0 0 m=audio 13774 RTP/AVP 117 110 101 a=rtpmap:117 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 11698 RTP/AVP 31 34 98 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv --- [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8607 [Jun 12 09:44:37] DEBUG[20571] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:37] VERBOSE[20571] app_dial.c: -- Called SIP/asterisk-GuruPlug/2204 [Jun 12 09:44:37] DEBUG[20571] channel.c: Set channel SIP/asterisk-GuruPlug-00000016 to read format slin [Jun 12 09:44:37] DEBUG[20571] channel.c: Set channel SIP/2202-00000015 to write format slin [Jun 12 09:44:37] DEBUG[20571] channel.c: Set channel SIP/2202-00000015 to read format slin [Jun 12 09:44:37] DEBUG[18161] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: from-node Uniqueid: 1371023077.23 [Jun 12 09:44:37] DEBUG[18161] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:37] DEBUG[18161] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/2202-00000015 Destination: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: ConnectedLineName: UniqueID: 1371023077.22 DestUniqueID: 1371023077.23 Dialstring: asterisk-GuruPlug/2204 [Jun 12 09:44:37] DEBUG[19616] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: from-node Uniqueid: 1371023077.23 [Jun 12 09:44:37] DEBUG[19616] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:37] DEBUG[19616] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/2202-00000015 Destination: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: ConnectedLineName: UniqueID: 1371023077.22 DestUniqueID: 1371023077.23 Dialstring: asterisk-GuruPlug/2204 [Jun 12 09:44:37] DEBUG[20059] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: from-node Uniqueid: 1371023077.23 [Jun 12 09:44:37] DEBUG[20059] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:37] DEBUG[20059] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/2202-00000015 Destination: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: ConnectedLineName: UniqueID: 1371023077.22 DestUniqueID: 1371023077.23 Dialstring: asterisk-GuruPlug/2204 [Jun 12 09:44:37] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Jun 12 09:44:37] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;received=130.10.20.220;rport=5060 From: "Sebastian Kirchner (2202)" ;tag=as1935e828 To: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 102 INVITE Server: Guru Plug Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;received=130.10.20.220;rport=5060 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 2 [ 73]: From: "Sebastian Kirchner (2202)" ;tag=as1935e828 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 3 [ 27]: To: [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 4 [ 60]: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 6 [ 17]: Server: Guru Plug [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 9 [ 37]: Contact: [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 12 09:44:37] VERBOSE[18144] chan_sip.c: --- (11 headers 0 lines) --- [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: = Looking for Call ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 (Checking To) --From tag as1935e828 --To-tag [Jun 12 09:44:37] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:44:37] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:44:37] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 6 (Ringing) [Jun 12 09:44:37] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '6' [Jun 12 09:44:37] DEBUG[20571] channel.c: Set channel SIP/asterisk-GuruPlug-00000016 to write format slin [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: *** SIP TIMER: Cancelling retransmission #8607 - INVITE (got response) [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060' Request 102: Found [Jun 12 09:44:37] DEBUG[18144] chan_sip.c: SIP response 100 to standard invite [Jun 12 09:44:37] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:37] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:37] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:37] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:37] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:37] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:38] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;received=130.10.20.220;rport=5060 From: "Sebastian Kirchner (2202)" ;tag=as1935e828 To: ;tag=as0cf77563 Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 102 INVITE Server: Guru Plug Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Frank Castle" Content-Length: 0 <-------------> [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;received=130.10.20.220;rport=5060 [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 2 [ 73]: From: "Sebastian Kirchner (2202)" ;tag=as1935e828 [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 3 [ 42]: To: ;tag=as0cf77563 [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 4 [ 60]: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 6 [ 17]: Server: Guru Plug [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 9 [ 37]: Contact: [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 10 [ 60]: P-Asserted-Identity: "Frank Castle" [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 12 09:44:38] VERBOSE[18144] chan_sip.c: --- (12 headers 0 lines) --- [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: = Looking for Call ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 (Checking To) --From tag as1935e828 --To-tag as0cf77563 [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060' Request 102: Found [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: SIP response 180 to standard invite [Jun 12 09:44:38] DEBUG[18144] chan_sip.c: build_route: Contact hop: [Jun 12 09:44:38] VERBOSE[18144] chan_sip.c: list_route: hop: [Jun 12 09:44:38] DEBUG[18161] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2204 CallerIDName: Frank Castle Uniqueid: 1371023077.23 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:38] DEBUG[18161] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 [Jun 12 09:44:38] DEBUG[19616] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2204 CallerIDName: Frank Castle Uniqueid: 1371023077.23 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:38] DEBUG[19616] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 [Jun 12 09:44:38] DEBUG[20059] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 CallerIDNum: 2204 CallerIDName: Frank Castle Uniqueid: 1371023077.23 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 12 09:44:38] DEBUG[20059] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 [Jun 12 09:44:38] VERBOSE[20571] app_dial.c: -- SIP/asterisk-GuruPlug-00000016 is ringing [Jun 12 09:44:38] DEBUG[20571] rtp_engine.c: Setting early bridge SDP of 'SIP/2202-00000015' with that of 'SIP/asterisk-GuruPlug-00000016' [Jun 12 09:44:38] VERBOSE[20571] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.155:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKb116b96afeb771c3b;received=10.0.0.155;rport=5060 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: ;tag=as2a991fae Call-ID: 791e658937df8782 CSeq: 1151097177 INVITE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Frank Castle" Content-Length: 0 <------------> [Jun 12 09:44:38] DEBUG[20571] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.0.0.155:5060 [Jun 12 09:44:38] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:44:38] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:44:38] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 6 (Ringing) [Jun 12 09:44:38] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '6' [Jun 12 09:44:38] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 12 09:44:39] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> OPTIONS sip:s@130.10.20.220:5060 SIP/2.0 Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK7d94cc62;rport Max-Forwards: 70 From: "asterisk" ;tag=as280908b6 To: Contact: Call-ID: 30504a1d3853977017124fb66f5c1875@130.10.160.5:5060 CSeq: 102 OPTIONS User-Agent: Guru Plug Date: Wed, 12 Jun 2013 07:45:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 0 [ 40]: OPTIONS sip:s@130.10.20.220:5060 SIP/2.0 [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK7d94cc62;rport [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as280908b6 [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 4 [ 30]: To: [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 5 [ 41]: Contact: [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 6 [ 59]: Call-ID: 30504a1d3853977017124fb66f5c1875@130.10.160.5:5060 [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 8 [ 21]: User-Agent: Guru Plug [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 9 [ 35]: Date: Wed, 12 Jun 2013 07:45:11 GMT [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 12 09:44:39] VERBOSE[18144] chan_sip.c: --- (13 headers 0 lines) --- [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: = Looking for Call ID: 30504a1d3853977017124fb66f5c1875@130.10.160.5:5060 (Checking From) --From tag as280908b6 --To-tag [Jun 12 09:44:39] DEBUG[18144] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 30504a1d3853977017124fb66f5c1875@130.10.160.5:5060 - OPTIONS (No RTP) [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 12 09:44:39] DEBUG[18144] netsock2.c: Splitting '130.10.20.220:5060' into... [Jun 12 09:44:39] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:39] DEBUG[18144] netsock2.c: Splitting '130.10.160.5' into... [Jun 12 09:44:39] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port ''. [Jun 12 09:44:39] VERBOSE[18144] chan_sip.c: Looking for s in default (domain 130.10.20.220) [Jun 12 09:44:39] VERBOSE[18144] chan_sip.c: <--- Transmitting (NAT) to 130.10.160.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK7d94cc62;received=130.10.160.5;rport=5060 From: "asterisk" ;tag=as280908b6 To: ;tag=as29cbc25b Call-ID: 30504a1d3853977017124fb66f5c1875@130.10.160.5:5060 CSeq: 102 OPTIONS Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Jun 12 09:44:39] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:39] VERBOSE[18144] chan_sip.c: Scheduling destruction of SIP dialog '30504a1d3853977017124fb66f5c1875@130.10.160.5:5060' in 32000 ms (Method: OPTIONS) [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;received=130.10.20.220;rport=5060 From: "Sebastian Kirchner (2202)" ;tag=as1935e828 To: ;tag=as0cf77563 Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 102 INVITE Server: Guru Plug Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Frank Castle" Content-Type: application/sdp Content-Length: 456 v=0 o=root 1306371035 1306371035 IN IP4 130.10.160.5 s=Asterisk PBX 10.12.2 c=IN IP4 130.10.160.5 b=CT:128 t=0 0 m=audio 10770 RTP/AVP 117 110 101 a=rtpmap:117 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 10550 RTP/AVP 31 34 98 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv <-------------> [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK60f812dc;received=130.10.20.220;rport=5060 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 2 [ 73]: From: "Sebastian Kirchner (2202)" ;tag=as1935e828 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 3 [ 42]: To: ;tag=as0cf77563 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 4 [ 60]: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 6 [ 17]: Server: Guru Plug [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 9 [ 37]: Contact: [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 10 [ 60]: P-Asserted-Identity: "Frank Castle" [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 12 [ 19]: Content-Length: 456 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 13 [ 0]: [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 0 [ 3]: v=0 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 1 [ 48]: o=root 1306371035 1306371035 IN IP4 130.10.160.5 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 10.12.2 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 3 [ 21]: c=IN IP4 130.10.160.5 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 4 [ 8]: b=CT:128 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 5 [ 5]: t=0 0 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 6 [ 33]: m=audio 10770 RTP/AVP 117 110 101 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 7 [ 24]: a=rtpmap:117 speex/16000 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 8 [ 23]: a=rtpmap:110 speex/8000 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 13 [ 10]: a=sendrecv [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 14 [ 33]: m=video 10550 RTP/AVP 31 34 98 99 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 15 [ 22]: a=rtpmap:31 H261/90000 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 16 [ 22]: a=rtpmap:34 H263/90000 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 17 [ 27]: a=rtpmap:98 h263-1998/90000 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 18 [ 22]: a=rtpmap:99 H264/90000 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Body 19 [ 10]: a=sendrecv [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: --- (13 headers 20 lines) --- [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: = Looking for Call ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 (Checking To) --From tag as1935e828 --To-tag as0cf77563 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Acked pending invite 102 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Stopping retransmission on '40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: SIP response 200 to standard invite [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing session-level SDP o=root 1306371035 1306371035 IN IP4 130.10.160.5... OK. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing session-level SDP s=Asterisk PBX 10.12.2... UNSUPPORTED OR FAILED. [Jun 12 09:44:43] DEBUG[18144] netsock2.c: Splitting '130.10.160.5' into... [Jun 12 09:44:43] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port ''. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing session-level SDP c=IN IP4 130.10.160.5... OK. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing session-level SDP b=CT:128... UNSUPPORTED OR FAILED. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found RTP audio format 117 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Setting payload 117 based on m type on 0xb74db4d4 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found RTP audio format 110 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Setting payload 110 based on m type on 0xb74db4d4 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found RTP audio format 101 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Setting payload 101 based on m type on 0xb74db4d4 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found audio description format speex for ID 117 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:117 speex/16000... OK. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found audio description format speex for ID 110 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000... OK. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found RTP video format 31 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Setting payload 31 based on m type on 0xb74dc214 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found RTP video format 34 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Setting payload 34 based on m type on 0xb74dc214 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found RTP video format 98 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Setting payload 98 based on m type on 0xb74dc214 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found RTP video format 99 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Setting payload 99 based on m type on 0xb74dc214 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found video description format H261 for ID 31 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:31 H261/90000... OK. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found video description format H263 for ID 34 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:34 H263/90000... OK. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found video description format h263-1998 for ID 98 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:98 h263-1998/90000... OK. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Found video description format H264 for ID 99 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:99 H264/90000... OK. [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Incorporating payload 101 on 0xb74db4d4 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Incorporating payload 110 on 0xb74db4d4 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Incorporating payload 117 on 0xb74db4d4 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Incorporating payload 31 on 0xb74dc214 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Incorporating payload 34 on 0xb74dc214 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Incorporating payload 98 on 0xb74dc214 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Incorporating payload 99 on 0xb74dc214 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Capabilities: us - 0x2003c0200 (speex|speex16|h261|h263|h263p|h264), peer - audio=0x200000200 (speex|speex16)/video=0x3c0000 (h261|h263|h263p|h264)/text=0x0 (nothing), combined - 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 12 09:44:43] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9b980d8' [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Peer audio RTP is at port 130.10.160.5:10770 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Copying payload 101 from 0xb74db4d4 to 0x9b98284 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Copying payload 110 from 0xb74db4d4 to 0x9b98284 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Copying payload 117 from 0xb74db4d4 to 0x9b98284 [Jun 12 09:44:43] DEBUG[18144] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9b980d8' [Jun 12 09:44:43] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x94e23c8' [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Peer video RTP is at port 130.10.160.5:10550 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Copying payload 31 from 0xb74dc214 to 0x94e2574 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Copying payload 34 from 0xb74dc214 to 0x94e2574 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Copying payload 98 from 0xb74dc214 to 0x94e2574 [Jun 12 09:44:43] DEBUG[18144] rtp_engine.c: Copying payload 99 from 0xb74dc214 to 0x94e2574 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: We're settling with these formats: 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: We have an owner, now see if we need to change this call [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Updating call counter for outgoing call [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: build_route: Contact hop: [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: list_route: hop: [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Strict routing enforced for session 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 12 09:44:43] DEBUG[18144] netsock2.c: Splitting '130.10.160.5:5060' into... [Jun 12 09:44:43] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port '5060'. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: set_destination: set destination to 130.10.160.5:5060 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: Transmitting (NAT) to 130.10.160.5:5060: ACK sip:2204@130.10.160.5:5060 SIP/2.0 Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK1cc9085c;rport Max-Forwards: 70 From: "Sebastian Kirchner (2202)" ;tag=as1935e828 To: ;tag=as0cf77563 Contact: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 102 ACK User-Agent: asteriskdev2 Content-Length: 0 --- [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Trying to put 'ACK sip:220' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:43] VERBOSE[20571] app_dial.c: -- SIP/asterisk-GuruPlug-00000016 answered SIP/2202-00000015 [Jun 12 09:44:43] DEBUG[20571] rtp_engine.c: Setting early bridge SDP of 'SIP/2202-00000015' with that of 'SIP/asterisk-GuruPlug-00000016' [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: SIP answering channel: SIP/2202-00000015 [Jun 12 09:44:43] DEBUG[20571] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: Setting framing from config on incoming call [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: ** Our capability: 0x1008 (alaw|g722) Video flag: True Text flag: True [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 12 09:44:43] VERBOSE[20571] chan_sip.c: Audio is at 16786 [Jun 12 09:44:43] VERBOSE[20571] chan_sip.c: Adding codec 0x1000 (g722) to SDP [Jun 12 09:44:43] VERBOSE[20571] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jun 12 09:44:43] VERBOSE[20571] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: -- Done with adding codecs to SDP [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: Done building SDP. Settling with this capability: 0x1008 (alaw|g722) [Jun 12 09:44:43] VERBOSE[20571] chan_sip.c: <--- Reliably Transmitting (NAT) to 10.0.0.155:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKb116b96afeb771c3b;received=10.0.0.155;rport=5060 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: ;tag=as2a991fae Call-ID: 791e658937df8782 CSeq: 1151097177 INVITE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Frank Castle" Content-Type: application/sdp Content-Length: 277 v=0 o=root 2144383301 2144383301 IN IP4 130.10.20.220 s=Asterisk PBX 1.8.20.1~dfsg-1ubuntu1 c=IN IP4 130.10.20.220 t=0 0 m=audio 16786 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8611 [Jun 12 09:44:43] DEBUG[20571] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.155:5060 [Jun 12 09:44:43] DEBUG[20571] features.c: bridge answer set, chan answer set [Jun 12 09:44:43] DEBUG[20571] features.c: Removing dialed interfaces datastore on SIP/asterisk-GuruPlug-00000016 since we're bridging [Jun 12 09:44:43] DEBUG[20571] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 12 09:44:43] DEBUG[20571] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 12 09:44:43] DEBUG[20571] rtp_engine.c: Channel codec0 = unknown is not codec1 = unknown, cannot native bridge in RTP. [Jun 12 09:44:43] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:44:43] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:44:43] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 2 (In use) [Jun 12 09:44:43] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '2' [Jun 12 09:44:43] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:44:43] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:44:43] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 2 (In use) [Jun 12 09:44:43] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '2' [Jun 12 09:44:43] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - 2202 [Jun 12 09:44:43] DEBUG[18136] chan_sip.c: Checking device state for peer 2202 [Jun 12 09:44:43] DEBUG[18136] devicestate.c: Changing state for SIP/2202 - state 3 (Busy) [Jun 12 09:44:43] DEBUG[18136] devicestate.c: device 'SIP/2202' state '3' [Jun 12 09:44:43] DEBUG[18161] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 [Jun 12 09:44:43] DEBUG[18161] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: 2204 ConnectedLineName: Frank Castle Uniqueid: 1371023077.22 [Jun 12 09:44:43] DEBUG[18161] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 Uniqueid: 1371023077.23 AccountCode: OldAccountCode: [Jun 12 09:44:43] DEBUG[18161] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2202-00000015 Channel2: SIP/asterisk-GuruPlug-00000016 Uniqueid1: 1371023077.22 Uniqueid2: 1371023077.23 CallerID1: 2202 CallerID2: 2204 [Jun 12 09:44:43] DEBUG[19616] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 [Jun 12 09:44:43] DEBUG[19616] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: 2204 ConnectedLineName: Frank Castle Uniqueid: 1371023077.22 [Jun 12 09:44:43] DEBUG[19616] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 Uniqueid: 1371023077.23 AccountCode: OldAccountCode: [Jun 12 09:44:43] DEBUG[19616] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2202-00000015 Channel2: SIP/asterisk-GuruPlug-00000016 Uniqueid1: 1371023077.22 Uniqueid2: 1371023077.23 CallerID1: 2202 CallerID2: 2204 [Jun 12 09:44:43] DEBUG[20059] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Uniqueid: 1371023077.23 [Jun 12 09:44:43] DEBUG[20059] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/2202-00000015 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: 2204 ConnectedLineName: Frank Castle Uniqueid: 1371023077.22 [Jun 12 09:44:43] DEBUG[20059] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 Uniqueid: 1371023077.23 AccountCode: OldAccountCode: [Jun 12 09:44:43] DEBUG[20059] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/2202-00000015 Channel2: SIP/asterisk-GuruPlug-00000016 Uniqueid1: 1371023077.22 Uniqueid2: 1371023077.23 CallerID1: 2202 CallerID2: 2204 [Jun 12 09:44:43] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:43] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:43] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:10.0.0.155:5060 ---> ACK sip:2204@130.10.20.220:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKd02b7beba091f6cca Max-Forwards: 70 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: ;tag=as2a991fae Call-ID: 791e658937df8782 CSeq: 1151097177 ACK Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE Allow-Events: talk, hold Authorization: Digest username="2202",realm="asterisk",nonce="396f50c0",uri="sip:2204@130.10.20.220:5060;transport=udp",response="e9b6992df752a348d0ad45f0693aec4e",algorithm=MD5 User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 Content-Length: 0 <-------------> [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 0 [ 39]: ACK sip:2204@130.10.20.220:5060 SIP/2.0 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKd02b7beba091f6cca [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 3 [ 71]: From: "2202" ;tag=8d46f5140b;epid=SC309517 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 4 [ 48]: To: ;tag=as2a991fae [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 5 [ 25]: Call-ID: 791e658937df8782 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 6 [ 20]: CSeq: 1151097177 ACK [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 7 [ 54]: Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 8 [ 24]: Allow-Events: talk, hold [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 9 [177]: Authorization: Digest username="2202",realm="asterisk",nonce="396f50c0",uri="sip:2204@130.10.20.220:5060;transport=udp",response="e9b6992df752a348d0ad45f0693aec4e",algorithm=MD5 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 10 [ 46]: User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 12 09:44:43] VERBOSE[18144] chan_sip.c: --- (12 headers 0 lines) --- [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: = Looking for Call ID: 791e658937df8782 (Checking From) --From tag 8d46f5140b --To-tag as2a991fae [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8611 [Jun 12 09:44:43] DEBUG[18144] chan_sip.c: Stopping retransmission on '791e658937df8782' of Response 1151097177: Match Found [Jun 12 09:44:43] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:43] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:43] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:43] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - soft2202 [Jun 12 09:44:43] DEBUG[18137] chan_sip.c: Checking device state for peer soft2202 [Jun 12 09:44:43] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - web2202 [Jun 12 09:44:43] DEBUG[18137] chan_sip.c: Checking device state for peer web2202 [Jun 12 09:44:43] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 12 09:44:43] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 12 09:44:43] DEBUG[18171] app_queue.c: Device 'SIP/2202' changed to state '3' (Busy) [Jun 12 09:44:43] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:44:43] DEBUG[20571] res_rtp_asterisk.c: Ooh, format changed from unknown to speex16 [Jun 12 09:44:43] DEBUG[20571] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x9b980d8' [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> INVITE sip:*962204@130.10.20.220 SIP/2.0 Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK00c438b3;rport Max-Forwards: 70 From: "User 2202" ;tag=as3837f858 To: Contact: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 102 INVITE User-Agent: Guru Plug Date: Wed, 12 Jun 2013 07:45:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Alert-Info: P-Asserted-Identity: "User 2202" Content-Type: application/sdp Content-Length: 454 v=0 o=root 164719518 164719518 IN IP4 130.10.160.5 s=Asterisk PBX 10.12.2 c=IN IP4 130.10.160.5 b=CT:128 t=0 0 m=audio 11872 RTP/AVP 117 110 101 a=rtpmap:117 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 11550 RTP/AVP 31 34 98 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv <-------------> [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 0 [ 40]: INVITE sip:*962204@130.10.20.220 SIP/2.0 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK00c438b3;rport [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 3 [ 56]: From: "User 2202" ;tag=as3837f858 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 4 [ 31]: To: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 5 [ 37]: Contact: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 6 [ 59]: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 8 [ 21]: User-Agent: Guru Plug [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 9 [ 35]: Date: Wed, 12 Jun 2013 07:45:16 GMT [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 12 [ 84]: Alert-Info: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 13 [ 56]: P-Asserted-Identity: "User 2202" [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 15 [ 19]: Content-Length: 454 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 16 [ 0]: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 0 [ 3]: v=0 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 1 [ 46]: o=root 164719518 164719518 IN IP4 130.10.160.5 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 10.12.2 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 3 [ 21]: c=IN IP4 130.10.160.5 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 4 [ 8]: b=CT:128 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 5 [ 5]: t=0 0 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 6 [ 33]: m=audio 11872 RTP/AVP 117 110 101 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 7 [ 24]: a=rtpmap:117 speex/16000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 8 [ 23]: a=rtpmap:110 speex/8000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 13 [ 10]: a=sendrecv [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 14 [ 33]: m=video 11550 RTP/AVP 31 34 98 99 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 15 [ 22]: a=rtpmap:31 H261/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 16 [ 22]: a=rtpmap:34 H263/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 17 [ 27]: a=rtpmap:98 h263-1998/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 18 [ 22]: a=rtpmap:99 H264/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 19 [ 10]: a=sendrecv [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: --- (16 headers 20 lines) --- [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: = Looking for Call ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 (Checking From) --From tag as3837f858 --To-tag [Jun 12 09:44:44] DEBUG[18144] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 - INVITE (No RTP) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 12 09:44:44] DEBUG[18144] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer" [Jun 12 09:44:44] DEBUG[18144] sip/reqresp_parser.c: Found SIP option: -replaces- [Jun 12 09:44:44] DEBUG[18144] sip/reqresp_parser.c: Matched SIP option: replaces [Jun 12 09:44:44] DEBUG[18144] sip/reqresp_parser.c: Found SIP option: -timer- [Jun 12 09:44:44] DEBUG[18144] sip/reqresp_parser.c: Matched SIP option: timer [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.160.5:5060' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port '5060'. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Sending to 130.10.160.5:5060 (NAT) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Initializing initreq for method INVITE - callid 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Using INVITE request as basis request - 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.160.5' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port ''. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found peer '2202' for '2202' from 130.10.160.5:5060 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: <--- Reliably Transmitting (NAT) to 130.10.160.5:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK00c438b3;received=130.10.160.5;rport=5060 From: "User 2202" ;tag=as3837f858 To: ;tag=as4ad74684 Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 102 INVITE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="675679ed" Content-Length: 0 <------------> [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8614 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Scheduling destruction of SIP dialog '68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060' in 32000 ms (Method: INVITE) [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> ACK sip:*962204@130.10.20.220 SIP/2.0 Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK00c438b3;rport Max-Forwards: 70 From: "User 2202" ;tag=as3837f858 To: ;tag=as4ad74684 Contact: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 102 ACK User-Agent: Guru Plug Content-Length: 0 <-------------> [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 0 [ 37]: ACK sip:*962204@130.10.20.220 SIP/2.0 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK00c438b3;rport [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 3 [ 56]: From: "User 2202" ;tag=as3837f858 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 4 [ 46]: To: ;tag=as4ad74684 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 5 [ 37]: Contact: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 6 [ 59]: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 8 [ 21]: User-Agent: Guru Plug [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: --- (10 headers 0 lines) --- [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: = Looking for Call ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 (Checking From) --From tag as3837f858 --To-tag as4ad74684 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8614 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Stopping retransmission on '68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060' of Response 102: Match Found [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> INVITE sip:*962204@130.10.20.220 SIP/2.0 Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK28113614;rport Max-Forwards: 70 From: "User 2202" ;tag=as3837f858 To: Contact: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 103 INVITE User-Agent: Guru Plug Authorization: Digest username="asterisk-asteriskdev2", realm="asterisk", algorithm=MD5, uri="sip:*962204@130.10.20.220", nonce="675679ed", response="cbf411c49ad7e282c10e491706033199" Date: Wed, 12 Jun 2013 07:45:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Alert-Info: P-Asserted-Identity: "User 2202" Content-Type: application/sdp Content-Length: 454 v=0 o=root 164719518 164719519 IN IP4 130.10.160.5 s=Asterisk PBX 10.12.2 c=IN IP4 130.10.160.5 b=CT:128 t=0 0 m=audio 11872 RTP/AVP 117 110 101 a=rtpmap:117 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 11550 RTP/AVP 31 34 98 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv <-------------> [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 0 [ 40]: INVITE sip:*962204@130.10.20.220 SIP/2.0 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK28113614;rport [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 3 [ 56]: From: "User 2202" ;tag=as3837f858 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 4 [ 31]: To: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 5 [ 37]: Contact: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 6 [ 59]: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 8 [ 21]: User-Agent: Guru Plug [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 9 [183]: Authorization: Digest username="asterisk-asteriskdev2", realm="asterisk", algorithm=MD5, uri="sip:*962204@130.10.20.220", nonce="675679ed", response="cbf411c49ad7e282c10e491706033199" [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 10 [ 35]: Date: Wed, 12 Jun 2013 07:45:16 GMT [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 13 [ 84]: Alert-Info: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 14 [ 56]: P-Asserted-Identity: "User 2202" [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 16 [ 19]: Content-Length: 454 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Header 17 [ 0]: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 0 [ 3]: v=0 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 1 [ 46]: o=root 164719518 164719519 IN IP4 130.10.160.5 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 10.12.2 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 3 [ 21]: c=IN IP4 130.10.160.5 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 4 [ 8]: b=CT:128 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 5 [ 5]: t=0 0 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 6 [ 33]: m=audio 11872 RTP/AVP 117 110 101 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 7 [ 24]: a=rtpmap:117 speex/16000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 8 [ 23]: a=rtpmap:110 speex/8000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 13 [ 10]: a=sendrecv [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 14 [ 33]: m=video 11550 RTP/AVP 31 34 98 99 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 15 [ 22]: a=rtpmap:31 H261/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 16 [ 22]: a=rtpmap:34 H263/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 17 [ 27]: a=rtpmap:98 h263-1998/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 18 [ 22]: a=rtpmap:99 H264/90000 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Body 19 [ 10]: a=sendrecv [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: --- (17 headers 20 lines) --- [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: = Looking for Call ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 (Checking From) --From tag as3837f858 --To-tag [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.20.220' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.20.220' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.160.5:5060' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port '5060'. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Sending to 130.10.160.5:5060 (NAT) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Initializing initreq for method INVITE - callid 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Using INVITE request as basis request - 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.160.5' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port ''. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found peer 'asterisk-GuruPlug' for 'asterisk-asteriskdev2' from 130.10.160.5:5060 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6f77c58' [Jun 12 09:44:44] DEBUG[18144] res_rtp_asterisk.c: Allocated port 10832 for RTP instance '0xb6f77c58' [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: RTP instance '0xb6f77c58' is setup and ready to go [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6f36ce8' [Jun 12 09:44:44] DEBUG[18144] res_rtp_asterisk.c: Allocated port 13532 for RTP instance '0xb6f36ce8' [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: RTP instance '0xb6f36ce8' is setup and ready to go [Jun 12 09:44:44] DEBUG[18144] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6f36ce8' [Jun 12 09:44:44] VERBOSE[18144] netsock2.c: == Using SIP VIDEO TOS bits 136 [Jun 12 09:44:44] VERBOSE[18144] netsock2.c: == Using SIP VIDEO CoS mark 6 [Jun 12 09:44:44] DEBUG[18144] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6f77c58' [Jun 12 09:44:44] VERBOSE[18144] netsock2.c: == Using SIP RTP TOS bits 184 [Jun 12 09:44:44] VERBOSE[18144] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Setting NAT on RTP to Off [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Setting NAT on VRTP to Off [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing session-level SDP o=root 164719518 164719519 IN IP4 130.10.160.5... OK. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing session-level SDP s=Asterisk PBX 10.12.2... UNSUPPORTED OR FAILED. [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.160.5' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port ''. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing session-level SDP c=IN IP4 130.10.160.5... OK. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing session-level SDP b=CT:128... UNSUPPORTED OR FAILED. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found RTP audio format 117 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Setting payload 117 based on m type on 0xb74daf64 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found RTP audio format 110 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Setting payload 110 based on m type on 0xb74daf64 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found RTP audio format 101 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Setting payload 101 based on m type on 0xb74daf64 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found audio description format speex for ID 117 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:117 speex/16000... OK. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found audio description format speex for ID 110 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000... OK. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found RTP video format 31 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Setting payload 31 based on m type on 0xb74dbca4 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found RTP video format 34 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Setting payload 34 based on m type on 0xb74dbca4 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found RTP video format 98 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Setting payload 98 based on m type on 0xb74dbca4 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found RTP video format 99 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Setting payload 99 based on m type on 0xb74dbca4 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found video description format H261 for ID 31 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:31 H261/90000... OK. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found video description format H263 for ID 34 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:34 H263/90000... OK. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found video description format h263-1998 for ID 98 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:98 h263-1998/90000... OK. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Found video description format H264 for ID 99 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:99 H264/90000... OK. [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Incorporating payload 101 on 0xb74daf64 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Incorporating payload 110 on 0xb74daf64 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Incorporating payload 117 on 0xb74daf64 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Incorporating payload 31 on 0xb74dbca4 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Incorporating payload 34 on 0xb74dbca4 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Incorporating payload 98 on 0xb74dbca4 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Incorporating payload 99 on 0xb74dbca4 [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Capabilities: us - 0x2003c0200 (speex|speex16|h261|h263|h263p|h264), peer - audio=0x200000200 (speex|speex16)/video=0x3c0000 (h261|h263|h263p|h264)/text=0x0 (nothing), combined - 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 12 09:44:44] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f77c58' [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Peer audio RTP is at port 130.10.160.5:11872 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Copying payload 101 from 0xb74daf64 to 0xb6f77e04 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Copying payload 110 from 0xb74daf64 to 0xb6f77e04 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Copying payload 117 from 0xb74daf64 to 0xb6f77e04 [Jun 12 09:44:44] DEBUG[18144] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb6f77c58' [Jun 12 09:44:44] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f36ce8' [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Peer video RTP is at port 130.10.160.5:11550 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Copying payload 31 from 0xb74dbca4 to 0xb6f36e94 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Copying payload 34 from 0xb74dbca4 to 0xb6f36e94 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Copying payload 98 from 0xb74dbca4 to 0xb6f36e94 [Jun 12 09:44:44] DEBUG[18144] rtp_engine.c: Copying payload 99 from 0xb74dbca4 to 0xb6f36e94 [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: We're settling with these formats: 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Checking SIP call limits for device asterisk-GuruPlug [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Updating call counter for incoming call [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Call from peer 'asterisk-GuruPlug' is 2 out of 2147483647 [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.20.220' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.20.220' and port ''. [Jun 12 09:44:44] DEBUG[18144] netsock2.c: Splitting '130.10.160.5' into... [Jun 12 09:44:44] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port ''. [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: Looking for *962204 in from-node (domain 130.10.20.220) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: *** Our native formats are 0x2003c0000 (speex16|h261|h263|h263p|h264) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: *** Joint capabilities are 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: *** Our capabilities are 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x200000000 (speex16) [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: build_route: Contact hop: [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: list_route: hop: [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: SIP/asterisk-GuruPlug-00000017: New call is still down.... Trying... [Jun 12 09:44:44] VERBOSE[18144] chan_sip.c: <--- Transmitting (NAT) to 130.10.160.5:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK28113614;received=130.10.160.5;rport=5060 From: "User 2202" ;tag=as3837f858 To: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 103 INVITE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 12 09:44:44] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:44] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:44:44] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:44:44] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 2 (In use) [Jun 12 09:44:44] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '2' [Jun 12 09:44:44] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:44:44] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:44:44] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 2 (In use) [Jun 12 09:44:44] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '2' [Jun 12 09:44:44] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:44] DEBUG[18161] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2202 CallerIDName: User 2202 AccountCode: Exten: *962204 Context: from-node Uniqueid: 1371023084.24 [Jun 12 09:44:44] DEBUG[18161] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023084.24 [Jun 12 09:44:44] DEBUG[19616] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2202 CallerIDName: User 2202 AccountCode: Exten: *962204 Context: from-node Uniqueid: 1371023084.24 [Jun 12 09:44:44] DEBUG[19616] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023084.24 [Jun 12 09:44:44] DEBUG[20059] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 2202 CallerIDName: User 2202 AccountCode: Exten: *962204 Context: from-node Uniqueid: 1371023084.24 [Jun 12 09:44:44] DEBUG[20059] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023084.24 [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [*962204@from-node:1] NoOp("SIP/asterisk-GuruPlug-00000017", " - From Node - ") in new stack [Jun 12 09:44:44] DEBUG[20572] pbx.c: Result of 'EXTEN' is '*962204' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'Goto' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [*962204@from-node:2] Goto("SIP/asterisk-GuruPlug-00000017", "local-serviceexts,*962204,1") in new stack [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Goto (local-serviceexts,*962204,1) [Jun 12 09:44:44] DEBUG[20572] pbx.c: Result of 'EXTEN' is '*962204' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'Goto' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [*962204@local-serviceexts:1] Goto("SIP/asterisk-GuruPlug-00000017", "local-vmhandler,vm2204,1") in new stack [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Goto (local-vmhandler,vm2204,1) [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:1] NoOp("SIP/asterisk-GuruPlug-00000017", "Voicemail") in new stack [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'Macro' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:2] Macro("SIP/asterisk-GuruPlug-00000017", "setLanguage") in new stack [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [s@macro-setLanguage:1] NoOp("SIP/asterisk-GuruPlug-00000017", "Set Language") in new stack [Jun 12 09:44:44] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:44:44] DEBUG[20572] pbx.c: Function result is '2202' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Expression result is '0' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [s@macro-setLanguage:2] GotoIf("SIP/asterisk-GuruPlug-00000017", "0?:endSetLanguage") in new stack [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Goto (macro-setLanguage,s,21) [Jun 12 09:44:44] DEBUG[20572] app_macro.c: Executed application: GotoIf [Jun 12 09:44:44] DEBUG[20572] pbx.c: Function result is 'de' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [s@macro-setLanguage:21] NoOp("SIP/asterisk-GuruPlug-00000017", "Language Set to de") in new stack [Jun 12 09:44:44] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'NoCDR' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:3] NoCDR("SIP/asterisk-GuruPlug-00000017", "") in new stack [Jun 12 09:44:44] DEBUG[20572] pbx.c: Result of 'EXTEN' is 'vm2204' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'Set' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:4] Set("SIP/asterisk-GuruPlug-00000017", "USERID=2204") in new stack [Jun 12 09:44:44] DEBUG[20572] pbx.c: Result of 'USERID' is '2204' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'Macro' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:5] Macro("SIP/asterisk-GuruPlug-00000017", "executeagi,vm-announcement.agi?user=2204") in new stack [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [s@macro-executeagi:1] NoOp("SIP/asterisk-GuruPlug-00000017", "Execute AGI") in new stack [Jun 12 09:44:44] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:44:44] DEBUG[20572] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:44:44] DEBUG[20572] pbx.c: Expression result is '0' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [s@macro-executeagi:2] GotoIf("SIP/asterisk-GuruPlug-00000017", "0?agihost0failed") in new stack [Jun 12 09:44:44] DEBUG[20572] pbx.c: Not taking any branch [Jun 12 09:44:44] DEBUG[20572] app_macro.c: Executed application: GotoIf [Jun 12 09:44:44] DEBUG[20572] pbx.c: Result of 'ARG1' is 'vm-announcement.agi?user=2204' [Jun 12 09:44:44] DEBUG[20572] pbx.c: Launching 'AGI' [Jun 12 09:44:44] VERBOSE[20572] pbx.c: -- Executing [s@macro-executeagi:3] AGI("SIP/asterisk-GuruPlug-00000017", "agi://130.10.14.249/vm-announcement.agi?user=2204") in new stack [Jun 12 09:44:44] DEBUG[20572] res_agi.c: Wow, connected! [Jun 12 09:44:44] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:44] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 12 09:44:44] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 12 09:44:44] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:44] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:44] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:44] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:44] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:44] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:44] DEBUG[18161] manager.c: Running action 'Getvar' [Jun 12 09:44:44] DEBUG[18161] pbx.c: Result of 'AJ_TRACE_ID' is NULL [Jun 12 09:44:45] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:44:46] VERBOSE[20572] res_agi.c: -- AGI Script agi://130.10.14.249/vm-announcement.agi?user=2204 completed, returning 0 [Jun 12 09:44:46] DEBUG[20572] app_macro.c: Executed application: AGI [Jun 12 09:44:46] DEBUG[20572] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:46] DEBUG[20572] pbx.c: Expression result is '1' [Jun 12 09:44:46] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:44:46] VERBOSE[20572] pbx.c: -- Executing [s@macro-executeagi:4] GotoIf("SIP/asterisk-GuruPlug-00000017", "1?agisuccess") in new stack [Jun 12 09:44:46] VERBOSE[20572] pbx.c: -- Goto (macro-executeagi,s,7) [Jun 12 09:44:46] DEBUG[20572] app_macro.c: Executed application: GotoIf [Jun 12 09:44:46] DEBUG[20572] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:44:46] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:46] VERBOSE[20572] pbx.c: -- Executing [s@macro-executeagi:7] NoOp("SIP/asterisk-GuruPlug-00000017", "AGI SUCCESS") in new stack [Jun 12 09:44:46] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:44:46] DEBUG[20572] pbx.c: Launching 'Answer' [Jun 12 09:44:46] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:6] Answer("SIP/asterisk-GuruPlug-00000017", "") in new stack [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: SIP answering channel: SIP/asterisk-GuruPlug-00000017 [Jun 12 09:44:46] DEBUG[20572] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: Setting framing from config on incoming call [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: This call needs video offers! [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: ** Our capability: 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) Video flag: False Text flag: True [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Audio is at 10832 [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Video is at 130.10.20.220:13532 [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Adding codec 0x200000000 (speex16) to SDP [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Adding codec 0x200 (speex) to SDP [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Adding video codec 0x40000 (h261) to SDP [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Adding video codec 0x80000 (h263) to SDP [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Adding video codec 0x200000 (h264) to SDP [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: -- Done with adding codecs to SDP [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: Done building SDP. Settling with this capability: 0x2003c0200 (speex|speex16|h261|h263|h263p|h264) [Jun 12 09:44:46] VERBOSE[20572] chan_sip.c: <--- Reliably Transmitting (NAT) to 130.10.160.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK28113614;received=130.10.160.5;rport=5060 From: "User 2202" ;tag=as3837f858 To: ;tag=as4e8878ef Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 103 INVITE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 446 v=0 o=root 1815434556 1815434556 IN IP4 130.10.20.220 s=Asterisk PBX 1.8.20.1~dfsg-1ubuntu1 c=IN IP4 130.10.20.220 b=CT:128 t=0 0 m=audio 10832 RTP/AVP 117 110 101 a=rtpmap:117 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 13532 RTP/AVP 31 34 98 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv <------------> [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8617 [Jun 12 09:44:46] DEBUG[20572] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:46] DEBUG[18161] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023084.24 [Jun 12 09:44:46] DEBUG[19616] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023084.24 [Jun 12 09:44:46] DEBUG[20059] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Uniqueid: 1371023084.24 [Jun 12 09:44:46] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:44:46] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:44:46] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 2 (In use) [Jun 12 09:44:46] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '2' [Jun 12 09:44:46] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:44:46] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> ACK sip:*962204@130.10.20.220:5060 SIP/2.0 Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK7e0fa066;rport Max-Forwards: 70 From: "User 2202" ;tag=as3837f858 To: ;tag=as4e8878ef Contact: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 103 ACK User-Agent: Guru Plug Content-Length: 0 <-------------> [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 0 [ 42]: ACK sip:*962204@130.10.20.220:5060 SIP/2.0 [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK7e0fa066;rport [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 3 [ 56]: From: "User 2202" ;tag=as3837f858 [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 4 [ 46]: To: ;tag=as4e8878ef [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 5 [ 37]: Contact: [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 6 [ 59]: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 8 [ 21]: User-Agent: Guru Plug [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 12 09:44:46] VERBOSE[18144] chan_sip.c: --- (10 headers 0 lines) --- [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: = Looking for Call ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 (Checking From) --From tag as3837f858 --To-tag as4e8878ef [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8617 [Jun 12 09:44:46] DEBUG[18144] chan_sip.c: Stopping retransmission on '68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060' of Response 103: Match Found [Jun 12 09:44:46] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:44:46] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 12 09:44:46] DEBUG[20572] pbx.c: Result of 'FILE' is '/tmp/1371023084.24' [Jun 12 09:44:46] DEBUG[20572] pbx.c: Launching 'Read' [Jun 12 09:44:46] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:7] Read("SIP/asterisk-GuruPlug-00000017", "VMMENUINPUT,/tmp/1371023084.24,1,,1,1") in new stack [Jun 12 09:44:46] VERBOSE[20572] app_read.c: -- Accepting a maximum of 1 digits. [Jun 12 09:44:46] DEBUG[20572] channel.c: Set channel SIP/asterisk-GuruPlug-00000017 to write format slin [Jun 12 09:44:46] DEBUG[20572] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jun 12 09:44:46] VERBOSE[20572] file.c: -- Playing '/tmp/1371023084.24.slin' (language 'de') [Jun 12 09:44:46] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Jun 12 09:44:46] DEBUG[20572] res_rtp_asterisk.c: Ooh, format changed from unknown to speex16 [Jun 12 09:44:46] DEBUG[20571] res_rtp_asterisk.c: Ooh, format changed from unknown to g722 [Jun 12 09:44:46] DEBUG[20571] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160 [Jun 12 09:44:46] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:44:48] DEBUG[20571] res_rtp_asterisk.c: Got RTCP report of 52 bytes [Jun 12 09:44:48] DEBUG[18161] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62439 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Jun 12 09:44:48] DEBUG[19616] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62439 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Jun 12 09:44:48] DEBUG[20059] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62439 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Jun 12 09:44:48] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023088.3180691456 SentRTP: 20480 SentPackets: 129 SentOctets: 20640 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 1892801996 DLSR: 0.0060 (sec) [Jun 12 09:44:48] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023088.3180691456 SentRTP: 20480 SentPackets: 129 SentOctets: 20640 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 1892801996 DLSR: 0.0060 (sec) [Jun 12 09:44:48] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023088.3180691456 SentRTP: 20480 SentPackets: 129 SentOctets: 20640 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 1892801996 DLSR: 0.0060 (sec) [Jun 12 09:44:48] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023088.3264049152 SentRTP: 3360350608 SentPackets: 251 SentOctets: 2885 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 7.3240 (sec) [Jun 12 09:44:48] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023088.3264049152 SentRTP: 3360350608 SentPackets: 251 SentOctets: 2885 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 7.3240 (sec) [Jun 12 09:44:48] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023088.3264049152 SentRTP: 3360350608 SentPackets: 251 SentOctets: 2885 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 7.3240 (sec) [Jun 12 09:44:48] DEBUG[20571] res_rtp_asterisk.c: Got RTCP report of 44 bytes [Jun 12 09:44:48] DEBUG[18161] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42352.0000000000 DLSR: 0.0600(sec) RTT: 1(sec) [Jun 12 09:44:48] DEBUG[19616] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42352.0000000000 DLSR: 0.0600(sec) RTT: 1(sec) [Jun 12 09:44:48] DEBUG[20059] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42352.0000000000 DLSR: 0.0600(sec) RTT: 1(sec) [Jun 12 09:44:51] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:11873 OurSSRC: 2103453475 SentNTP: 1371023091.0242221056 SentRTP: 79040 SentPackets: 247 SentOctets: 2835 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0002 TheirLastSR: 0 DLSR: 9.5870 (sec) [Jun 12 09:44:51] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:11873 OurSSRC: 2103453475 SentNTP: 1371023091.0242221056 SentRTP: 79040 SentPackets: 247 SentOctets: 2835 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0002 TheirLastSR: 0 DLSR: 9.5870 (sec) [Jun 12 09:44:51] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:11873 OurSSRC: 2103453475 SentNTP: 1371023091.0242221056 SentRTP: 79040 SentPackets: 247 SentOctets: 2835 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0002 TheirLastSR: 0 DLSR: 9.5870 (sec) [Jun 12 09:44:52] DEBUG[20572] channel.c: Scheduling timer at (138 requested / 138 actual) timer ticks per second [Jun 12 09:44:52] DEBUG[20572] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 12 09:44:52] DEBUG[20572] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 12 09:44:52] DEBUG[20572] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 12 09:44:52] DEBUG[20572] channel.c: Set channel SIP/asterisk-GuruPlug-00000017 to write format speex16 [Jun 12 09:44:53] VERBOSE[20572] app_read.c: -- User entered nothing. [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'VMMENUINPUT' is '' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Expression result is '0' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:44:53] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:8] GotoIf("SIP/asterisk-GuruPlug-00000017", "0?vm-menu") in new stack [Jun 12 09:44:53] DEBUG[20572] pbx.c: Not taking any branch [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'RECORD' is 'YES' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:53] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:9] NoOp("SIP/asterisk-GuruPlug-00000017", "Execute voicemail YES") in new stack [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'RECORD' is 'YES' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Expression result is '0' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:44:53] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:10] GotoIf("SIP/asterisk-GuruPlug-00000017", "0?vm-end") in new stack [Jun 12 09:44:53] DEBUG[20572] pbx.c: Not taking any branch [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'USERID' is '2204' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'EPOCH' is '1371023093' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Launching 'Set' [Jun 12 09:44:53] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:11] Set("SIP/asterisk-GuruPlug-00000017", "RECFILENAME=2204-1371023093") in new stack [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'EPOCH' is '1371023093' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:44:53] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:12] NoOp("SIP/asterisk-GuruPlug-00000017", "Record Start 1371023093") in new stack [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'EPOCH' is '1371023093' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Launching 'Set' [Jun 12 09:44:53] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:13] Set("SIP/asterisk-GuruPlug-00000017", "START=1371023093") in new stack [Jun 12 09:44:53] DEBUG[20572] pbx.c: Result of 'RECFILENAME' is '2204-1371023093' [Jun 12 09:44:53] DEBUG[20572] pbx.c: Launching 'Record' [Jun 12 09:44:53] VERBOSE[20572] pbx.c: -- Executing [vm2204@local-vmhandler:14] Record("SIP/asterisk-GuruPlug-00000017", "/tmp/2204-1371023093:wav,0,600,k") in new stack [Jun 12 09:44:53] DEBUG[20572] channel.c: Set channel SIP/asterisk-GuruPlug-00000017 to write format gsm [Jun 12 09:44:53] DEBUG[20572] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jun 12 09:44:53] VERBOSE[20572] file.c: -- Playing 'beep.gsm' (language 'de') [Jun 12 09:44:53] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Jun 12 09:44:53] DEBUG[20572] res_rtp_asterisk.c: Difference is 16800, ms is 1070 [Jun 12 09:44:53] DEBUG[20571] res_rtp_asterisk.c: Got RTCP report of 52 bytes [Jun 12 09:44:53] DEBUG[18161] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62636 SequenceNumberCycles: 0 IAJitter: 227 LastSR: 42352.2952790016 DLSR: 4.9916(sec) RTT: 1(sec) [Jun 12 09:44:53] DEBUG[19616] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62636 SequenceNumberCycles: 0 IAJitter: 227 LastSR: 42352.2952790016 DLSR: 4.9916(sec) RTT: 1(sec) [Jun 12 09:44:53] DEBUG[20059] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62636 SequenceNumberCycles: 0 IAJitter: 227 LastSR: 42352.2952790016 DLSR: 4.9916(sec) RTT: 1(sec) [Jun 12 09:44:53] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023093.3181309952 SentRTP: 52160 SentPackets: 327 SentOctets: 52320 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0017 TheirLastSR: 1893129676 DLSR: 0.0070 (sec) [Jun 12 09:44:53] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023093.3181309952 SentRTP: 52160 SentPackets: 327 SentOctets: 52320 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0017 TheirLastSR: 1893129676 DLSR: 0.0070 (sec) [Jun 12 09:44:53] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023093.3181309952 SentRTP: 52160 SentPackets: 327 SentOctets: 52320 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0017 TheirLastSR: 1893129676 DLSR: 0.0070 (sec) [Jun 12 09:44:53] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023093.3264016384 SentRTP: 3360430608 SentPackets: 501 SentOctets: 5792 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 0 DLSR: 12.3240 (sec) [Jun 12 09:44:53] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023093.3264016384 SentRTP: 3360430608 SentPackets: 501 SentOctets: 5792 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 0 DLSR: 12.3240 (sec) [Jun 12 09:44:53] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023093.3264016384 SentRTP: 3360430608 SentPackets: 501 SentOctets: 5792 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 0 DLSR: 12.3240 (sec) [Jun 12 09:44:53] DEBUG[20571] res_rtp_asterisk.c: Got RTCP report of 44 bytes [Jun 12 09:44:53] DEBUG[18161] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42357.0000000000 DLSR: 0.0600(sec) RTT: 1(sec) [Jun 12 09:44:53] DEBUG[19616] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42357.0000000000 DLSR: 0.0600(sec) RTT: 1(sec) [Jun 12 09:44:53] DEBUG[20059] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42357.0000000000 DLSR: 0.0600(sec) RTT: 1(sec) [Jun 12 09:44:53] DEBUG[20572] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 12 09:44:53] DEBUG[20572] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 12 09:44:53] DEBUG[20572] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 12 09:44:53] DEBUG[20572] channel.c: Set channel SIP/asterisk-GuruPlug-00000017 to write format speex16 [Jun 12 09:44:53] DEBUG[20572] chan_sip.c: Strict routing enforced for session 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:53] VERBOSE[20572] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 12 09:44:53] DEBUG[20572] netsock2.c: Splitting '130.10.160.5:5060' into... [Jun 12 09:44:53] DEBUG[20572] netsock2.c: ...host '130.10.160.5' and port '5060'. [Jun 12 09:44:53] VERBOSE[20572] chan_sip.c: set_destination: set destination to 130.10.160.5:5060 [Jun 12 09:44:53] VERBOSE[20572] chan_sip.c: Reliably Transmitting (NAT) to 130.10.160.5:5060: INFO sip:2202@130.10.160.5:5060 SIP/2.0 Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK4589e17c;rport Max-Forwards: 70 From: ;tag=as4e8878ef To: "User 2202" ;tag=as3837f858 Contact: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 102 INFO User-Agent: asteriskdev2 Content-Type: application/media_control+xml Content-Length: 205 --- [Jun 12 09:44:53] DEBUG[20572] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8619 [Jun 12 09:44:53] DEBUG[20572] chan_sip.c: Trying to put 'INFO sip:22' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:53] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK4589e17c;received=130.10.20.220;rport=5060 From: ;tag=as4e8878ef To: "User 2202" ;tag=as3837f858 Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 102 INFO Server: Guru Plug Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK4589e17c;received=130.10.20.220;rport=5060 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 2 [ 48]: From: ;tag=as4e8878ef [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 3 [ 54]: To: "User 2202" ;tag=as3837f858 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 4 [ 59]: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 5 [ 14]: CSeq: 102 INFO [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 6 [ 17]: Server: Guru Plug [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 12 09:44:53] VERBOSE[18144] chan_sip.c: --- (10 headers 0 lines) --- [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: = Looking for Call ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 (Checking To) --From tag as4e8878ef --To-tag as3837f858 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8619 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Stopping retransmission on '68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060' of Request 102: Match Found [Jun 12 09:44:53] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> INFO sip:2202@130.10.20.220:5060 SIP/2.0 Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK05e8fb19;rport Max-Forwards: 70 From: ;tag=as0cf77563 To: "Sebastian Kirchner (2202)" ;tag=as1935e828 Contact: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 102 INFO User-Agent: Guru Plug Content-Type: application/media_control+xml Content-Length: 205 <-------------> [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 0 [ 40]: INFO sip:2202@130.10.20.220:5060 SIP/2.0 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK05e8fb19;rport [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 3 [ 44]: From: ;tag=as0cf77563 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 4 [ 71]: To: "Sebastian Kirchner (2202)" ;tag=as1935e828 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 5 [ 37]: Contact: [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 6 [ 60]: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 7 [ 14]: CSeq: 102 INFO [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 8 [ 21]: User-Agent: Guru Plug [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 9 [ 43]: Content-Type: application/media_control+xml [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 10 [ 19]: Content-Length: 205 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Header 11 [ 0]: [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Body 0 [175]: [Jun 12 09:44:53] VERBOSE[18144] chan_sip.c: --- (11 headers 1 lines) --- [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: = Looking for Call ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 (Checking From) --From tag as0cf77563 --To-tag as1935e828 [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: **** Received INFO (13) - Command in SIP INFO [Jun 12 09:44:53] VERBOSE[18144] chan_sip.c: Receiving INFO! [Jun 12 09:44:53] VERBOSE[18144] chan_sip.c: <--- Transmitting (NAT) to 130.10.160.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK05e8fb19;received=130.10.160.5;rport=5060 From: ;tag=as0cf77563 To: "Sebastian Kirchner (2202)" ;tag=as1935e828 Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 102 INFO Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jun 12 09:44:53] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:44:53] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:54] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 8 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:55] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:11873 OurSSRC: 2103453475 SentNTP: 1371023096.0408997888 SentRTP: 124640 SentPackets: 337 SentOctets: 4153 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0202 TheirLastSR: 0 DLSR: 14.6270 (sec) [Jun 12 09:44:56] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:11873 OurSSRC: 2103453475 SentNTP: 1371023096.0408997888 SentRTP: 124640 SentPackets: 337 SentOctets: 4153 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0202 TheirLastSR: 0 DLSR: 14.6270 (sec) [Jun 12 09:44:56] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:11873 OurSSRC: 2103453475 SentNTP: 1371023096.0408997888 SentRTP: 124640 SentPackets: 337 SentOctets: 4153 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0202 TheirLastSR: 0 DLSR: 14.6270 (sec) [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:56] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 8 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:57] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[20571] res_rtp_asterisk.c: Got RTCP report of 52 bytes [Jun 12 09:44:58] DEBUG[18161] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62643 SequenceNumberCycles: 0 IAJitter: 145 LastSR: 42357.1073741824 DLSR: 4.9913(sec) RTT: 77(sec) [Jun 12 09:44:58] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023098.3470667776 SentRTP: 53120 SentPackets: 333 SentOctets: 53280 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0300 TheirLastSR: 1893457356 DLSR: 0.0010 (sec) [Jun 12 09:44:58] DEBUG[18161] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023098.3470917632 SentRTP: 3360511248 SentPackets: 753 SentOctets: 12127 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 0 DLSR: 17.3750 (sec) [Jun 12 09:44:58] DEBUG[19616] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62643 SequenceNumberCycles: 0 IAJitter: 145 LastSR: 42357.1073741824 DLSR: 4.9913(sec) RTT: 77(sec) [Jun 12 09:44:58] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023098.3470667776 SentRTP: 53120 SentPackets: 333 SentOctets: 53280 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0300 TheirLastSR: 1893457356 DLSR: 0.0010 (sec) [Jun 12 09:44:58] DEBUG[19616] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023098.3470917632 SentRTP: 3360511248 SentPackets: 753 SentOctets: 12127 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 0 DLSR: 17.3750 (sec) [Jun 12 09:44:58] DEBUG[20059] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 10.0.0.155:5011 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 62643 SequenceNumberCycles: 0 IAJitter: 145 LastSR: 42357.1073741824 DLSR: 4.9913(sec) RTT: 77(sec) [Jun 12 09:44:58] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 10.0.0.155:5011 OurSSRC: 1026437510 SentNTP: 1371023098.3470667776 SentRTP: 53120 SentPackets: 333 SentOctets: 53280 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0300 TheirLastSR: 1893457356 DLSR: 0.0010 (sec) [Jun 12 09:44:58] DEBUG[20059] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 130.10.160.5:10771 OurSSRC: 1747013801 SentNTP: 1371023098.3470917632 SentRTP: 3360511248 SentPackets: 753 SentOctets: 12127 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 0 DLSR: 17.3750 (sec) [Jun 12 09:44:58] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:58] DEBUG[20571] res_rtp_asterisk.c: Got RTCP report of 44 bytes [Jun 12 09:44:59] DEBUG[18161] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42362.3758096384 DLSR: 0.0080(sec) RTT: 90(sec) [Jun 12 09:44:59] DEBUG[19616] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42362.3758096384 DLSR: 0.0080(sec) RTT: 90(sec) [Jun 12 09:44:59] DEBUG[20059] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 130.10.160.5:10771 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 51274 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 42362.3758096384 DLSR: 0.0080(sec) RTT: 90(sec) [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 8 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:44:59] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:00] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 8 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:01] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:02] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Challenge' [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Login' [Jun 12 09:45:03] VERBOSE[20680] manager.c: == Manager 'obelisk' logged on from 130.10.237.168 [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Command' [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Command' [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 8 instead [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Status' [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Agents' [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:10.0.0.155:5060 ---> BYE sip:2204@130.10.20.220:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKe4fe7379e034e18e1 Max-Forwards: 70 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: ;tag=as2a991fae Call-ID: 791e658937df8782 CSeq: 1151097178 BYE Authorization: Digest username="2202",realm="asterisk",nonce="396f50c0",uri="sip:2204@130.10.20.220:5060",response="1b7d4a6aaf21d26d4ecea46526133f50",algorithm=MD5 Supported: 100rel, replaces User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 X-Siemens-RTP-Stats: PS=991,OS=158560,PR=333,OR=53280,PL=0,JI=1051,LA=548,SS=1,EN=9,DE=9,IE=13,TCLW=56 Content-Length: 0 <-------------> [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 0 [ 39]: BYE sip:2204@130.10.20.220:5060 SIP/2.0 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKe4fe7379e034e18e1 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 3 [ 71]: From: "2202" ;tag=8d46f5140b;epid=SC309517 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 4 [ 48]: To: ;tag=as2a991fae [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 5 [ 25]: Call-ID: 791e658937df8782 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 6 [ 20]: CSeq: 1151097178 BYE [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 7 [163]: Authorization: Digest username="2202",realm="asterisk",nonce="396f50c0",uri="sip:2204@130.10.20.220:5060",response="1b7d4a6aaf21d26d4ecea46526133f50",algorithm=MD5 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 8 [ 27]: Supported: 100rel, replaces [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 9 [ 46]: User-Agent: OpenStage_60_V3 R1.41.0 SIP 130205 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 10 [102]: X-Siemens-RTP-Stats: PS=991,OS=158560,PR=333,OR=53280,PL=0,JI=1051,LA=548,SS=1,EN=9,DE=9,IE=13,TCLW=56 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: --- (12 headers 0 lines) --- [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: = Looking for Call ID: 791e658937df8782 (Checking From) --From tag 8d46f5140b --To-tag as2a991fae [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Initializing initreq for method BYE - callid 791e658937df8782 [Jun 12 09:45:03] DEBUG[18144] netsock2.c: Splitting '10.0.0.155' into... [Jun 12 09:45:03] DEBUG[18144] netsock2.c: ...host '10.0.0.155' and port ''. [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: Sending to 10.0.0.155:5060 (NAT) [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Setting SIP_ALREADYGONE on dialog 791e658937df8782 [Jun 12 09:45:03] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f33588' [Jun 12 09:45:03] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f76c00' [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: Scheduling destruction of SIP dialog '791e658937df8782' in 32000 ms (Method: BYE) [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Received bye, issuing owner hangup [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.155:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.155;branch=z9hG4bKe4fe7379e034e18e1;received=10.0.0.155;rport=5060 From: "2202" ;tag=8d46f5140b;epid=SC309517 To: ;tag=as2a991fae Call-ID: 791e658937df8782 CSeq: 1151097178 BYE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.155:5060 [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'QueueStatus' [Jun 12 09:45:03] DEBUG[20571] channel.c: Didn't get a frame from channel: SIP/2202-00000015 [Jun 12 09:45:03] DEBUG[20571] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 12 09:45:03] DEBUG[20571] channel.c: Bridge stops bridging channels SIP/2202-00000015 and SIP/asterisk-GuruPlug-00000016 [Jun 12 09:45:03] DEBUG[20571] cdr.c: Dropping CDR ! [Jun 12 09:45:03] DEBUG[20571] channel.c: Hanging up channel 'SIP/asterisk-GuruPlug-00000016' [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Hangup call SIP/asterisk-GuruPlug-00000016, SIP callid 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: update_call_counter(2204) - decrement call limit counter on hangup [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Updating call counter for outgoing call [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Call to peer 'asterisk-GuruPlug' removed from call limit 2147483647 [Jun 12 09:45:03] DEBUG[20571] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9b980d8' [Jun 12 09:45:03] DEBUG[20571] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x94e23c8' [Jun 12 09:45:03] VERBOSE[20571] chan_sip.c: Scheduling destruction of SIP dialog '40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060' in 6400 ms (Method: INFO) [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Strict routing enforced for session 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:45:03] VERBOSE[20571] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 12 09:45:03] DEBUG[20571] netsock2.c: Splitting '130.10.160.5:5060' into... [Jun 12 09:45:03] DEBUG[20571] netsock2.c: ...host '130.10.160.5' and port '5060'. [Jun 12 09:45:03] VERBOSE[20571] chan_sip.c: set_destination: set destination to 130.10.160.5:5060 [Jun 12 09:45:03] VERBOSE[20571] chan_sip.c: Reliably Transmitting (NAT) to 130.10.160.5:5060: BYE sip:2204@130.10.160.5:5060 SIP/2.0 Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK109daa97;rport Max-Forwards: 70 From: "Sebastian Kirchner (2202)" ;tag=as1935e828 To: ;tag=as0cf77563 Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 103 BYE User-Agent: asteriskdev2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8622 [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Trying to put 'BYE sip:220' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:45:03] DEBUG[20571] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jun 12 09:45:03] DEBUG[20571] pbx.c: Spawn extension (to-node,s,2) exited non-zero on 'SIP/2202-00000015' [Jun 12 09:45:03] VERBOSE[20571] pbx.c: == Spawn extension (to-node, s, 2) exited non-zero on 'SIP/2202-00000015' [Jun 12 09:45:03] DEBUG[20571] channel.c: Soft-Hanging up channel 'SIP/2202-00000015' [Jun 12 09:45:03] DEBUG[20571] channel.c: Hanging up channel 'SIP/2202-00000015' [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Hangup call SIP/2202-00000015, SIP callid 791e658937df8782 [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: update_call_counter(2202) - decrement call limit counter on hangup [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Updating call counter for incoming call [Jun 12 09:45:03] DEBUG[20571] chan_sip.c: Call from peer '2202' removed from call limit 4 [Jun 12 09:45:03] DEBUG[20571] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f33588' [Jun 12 09:45:03] DEBUG[20571] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f76c00' [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[18161] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2202-00000015 Channel2: SIP/asterisk-GuruPlug-00000016 Uniqueid1: 1371023077.22 Uniqueid2: 1371023077.23 CallerID1: 2202 CallerID2: 2204 [Jun 12 09:45:03] DEBUG[18161] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 Uniqueid: 1371023077.23 CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[18161] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2202-00000015 UniqueID: 1371023077.22 DialStatus: ANSWER [Jun 12 09:45:03] DEBUG[18161] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2202-00000015 Uniqueid: 1371023077.22 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: 2204 ConnectedLineName: Frank Castle Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[19616] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2202-00000015 Channel2: SIP/asterisk-GuruPlug-00000016 Uniqueid1: 1371023077.22 Uniqueid2: 1371023077.23 CallerID1: 2202 CallerID2: 2204 [Jun 12 09:45:03] DEBUG[19616] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 Uniqueid: 1371023077.23 CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[19616] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2202-00000015 UniqueID: 1371023077.22 DialStatus: ANSWER [Jun 12 09:45:03] DEBUG[19616] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2202-00000015 Uniqueid: 1371023077.22 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: 2204 ConnectedLineName: Frank Castle Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[20059] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2202-00000015 Channel2: SIP/asterisk-GuruPlug-00000016 Uniqueid1: 1371023077.22 Uniqueid2: 1371023077.23 CallerID1: 2202 CallerID2: 2204 [Jun 12 09:45:03] DEBUG[20059] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 Uniqueid: 1371023077.23 CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[20059] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2202-00000015 UniqueID: 1371023077.22 DialStatus: ANSWER [Jun 12 09:45:03] DEBUG[20059] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2202-00000015 Uniqueid: 1371023077.22 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: 2204 ConnectedLineName: Frank Castle Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[20680] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/2202-00000015 Channel2: SIP/asterisk-GuruPlug-00000016 Uniqueid1: 1371023077.22 Uniqueid2: 1371023077.23 CallerID1: 2202 CallerID2: 2204 [Jun 12 09:45:03] DEBUG[20680] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000016 Uniqueid: 1371023077.23 CallerIDNum: 2204 CallerIDName: Frank Castle ConnectedLineNum: 2202 ConnectedLineName: Sebastian Kirchner (2202) Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[20680] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/2202-00000015 UniqueID: 1371023077.22 DialStatus: ANSWER [Jun 12 09:45:03] DEBUG[20680] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/2202-00000015 Uniqueid: 1371023077.22 CallerIDNum: 2202 CallerIDName: Sebastian Kirchner (2202) ConnectedLineNum: 2204 ConnectedLineName: Frank Castle Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:03] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:45:03] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:45:03] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:45:03] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 2 (In use) [Jun 12 09:45:03] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '2' [Jun 12 09:45:03] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:45:03] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:45:03] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 2 (In use) [Jun 12 09:45:03] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '2' [Jun 12 09:45:03] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - 2202 [Jun 12 09:45:03] DEBUG[18136] chan_sip.c: Checking device state for peer 2202 [Jun 12 09:45:03] DEBUG[18136] devicestate.c: Changing state for SIP/2202 - state 1 (Not in use) [Jun 12 09:45:03] DEBUG[18136] devicestate.c: device 'SIP/2202' state '1' [Jun 12 09:45:03] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - 2202 [Jun 12 09:45:03] DEBUG[18136] chan_sip.c: Checking device state for peer 2202 [Jun 12 09:45:03] DEBUG[18136] devicestate.c: Changing state for SIP/2202 - state 1 (Not in use) [Jun 12 09:45:03] DEBUG[18136] devicestate.c: device 'SIP/2202' state '1' [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK109daa97;received=130.10.20.220;rport=5060 From: "Sebastian Kirchner (2202)" ;tag=as1935e828 To: ;tag=as0cf77563 Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 CSeq: 103 BYE Server: Guru Plug Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK109daa97;received=130.10.20.220;rport=5060 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 2 [ 73]: From: "Sebastian Kirchner (2202)" ;tag=as1935e828 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 3 [ 42]: To: ;tag=as0cf77563 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 4 [ 60]: Call-ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 6 [ 17]: Server: Guru Plug [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: --- (10 headers 0 lines) --- [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: = Looking for Call ID: 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 (Checking To) --From tag as1935e828 --To-tag as0cf77563 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8622 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Stopping retransmission on '40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060' of Request 103: Match Found [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Destroying SIP dialog 40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060 [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: Really destroying SIP dialog '40c976456ebaa4075379c09440b3d43a@130.10.20.220:5060' Method: INFO [Jun 12 09:45:03] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0x9b980d8' [Jun 12 09:45:03] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0x94e23c8' [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: <--- SIP read from UDP:130.10.160.5:5060 ---> BYE sip:*962204@130.10.20.220:5060 SIP/2.0 Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK4cdac7ec;rport Max-Forwards: 70 From: "User 2202" ;tag=as3837f858 To: ;tag=as4e8878ef Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 104 BYE User-Agent: Guru Plug Authorization: Digest username="asterisk-asteriskdev2", realm="asterisk", algorithm=MD5, uri="sip:*962204@130.10.20.220:5060", nonce="675679ed", response="67a37404fc020a0772d226d5f088e5ba" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 0 [ 42]: BYE sip:*962204@130.10.20.220:5060 SIP/2.0 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK4cdac7ec;rport [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 3 [ 56]: From: "User 2202" ;tag=as3837f858 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 4 [ 46]: To: ;tag=as4e8878ef [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 5 [ 59]: Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 6 [ 13]: CSeq: 104 BYE [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 7 [ 21]: User-Agent: Guru Plug [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 8 [188]: Authorization: Digest username="asterisk-asteriskdev2", realm="asterisk", algorithm=MD5, uri="sip:*962204@130.10.20.220:5060", nonce="675679ed", response="67a37404fc020a0772d226d5f088e5ba" [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: --- (12 headers 0 lines) --- [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: = Looking for Call ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 (Checking From) --From tag as3837f858 --To-tag as4e8878ef [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Initializing initreq for method BYE - callid 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:45:03] DEBUG[18144] netsock2.c: Splitting '130.10.160.5:5060' into... [Jun 12 09:45:03] DEBUG[18144] netsock2.c: ...host '130.10.160.5' and port '5060'. [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: Sending to 130.10.160.5:5060 (NAT) [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Setting SIP_ALREADYGONE on dialog 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:45:03] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f77c58' [Jun 12 09:45:03] DEBUG[18144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f36ce8' [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: Scheduling destruction of SIP dialog '68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060' in 6400 ms (Method: BYE) [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Received bye, issuing owner hangup [Jun 12 09:45:03] VERBOSE[18144] chan_sip.c: <--- Transmitting (NAT) to 130.10.160.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 130.10.160.5:5060;branch=z9hG4bK4cdac7ec;received=130.10.160.5;rport=5060 From: "User 2202" ;tag=as3837f858 To: ;tag=as4e8878ef Call-ID: 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 CSeq: 104 BYE Server: asteriskdev2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jun 12 09:45:03] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:45:03] DEBUG[18159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Jun 12 09:45:03] DEBUG[20572] app_record.c: Got hangup [Jun 12 09:45:03] DEBUG[20572] pbx.c: Spawn extension (local-vmhandler,vm2204,14) exited non-zero on 'SIP/asterisk-GuruPlug-00000017' [Jun 12 09:45:03] VERBOSE[20572] pbx.c: == Spawn extension (local-vmhandler, vm2204, 14) exited non-zero on 'SIP/asterisk-GuruPlug-00000017' [Jun 12 09:45:03] DEBUG[20572] channel.c: Soft-Hanging up channel 'SIP/asterisk-GuruPlug-00000017' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'EPOCH' is '1371023103' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Launching 'Set' [Jun 12 09:45:03] VERBOSE[20572] pbx.c: -- Executing [h@local-vmhandler:1] Set("SIP/asterisk-GuruPlug-00000017", "END=1371023103") in new stack [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'USERID' is '2204' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Launching 'Macro' [Jun 12 09:45:03] VERBOSE[20572] pbx.c: -- Executing [h@local-vmhandler:2] Macro("SIP/asterisk-GuruPlug-00000017", "executedeadagi,vm-record.agi?user=2204") in new stack [Jun 12 09:45:03] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:45:03] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:1] NoOp("SIP/asterisk-GuruPlug-00000017", "Execute AGI") in new stack [Jun 12 09:45:03] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:45:03] DEBUG[20572] pbx.c: Expression result is '0' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:45:03] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:2] GotoIf("SIP/asterisk-GuruPlug-00000017", "0?agihost0failed") in new stack [Jun 12 09:45:03] DEBUG[20572] pbx.c: Not taking any branch [Jun 12 09:45:03] DEBUG[20572] app_macro.c: Executed application: GotoIf [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'ARG1' is 'vm-record.agi?user=2204' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Launching 'AGI' [Jun 12 09:45:03] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:3] AGI("SIP/asterisk-GuruPlug-00000017", "agi://130.10.14.249/vm-record.agi?user=2204") in new stack [Jun 12 09:45:03] DEBUG[20572] res_agi.c: Hungup channel detected, running agi in dead mode. [Jun 12 09:45:03] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:45:03] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:45:03] DEBUG[20572] res_agi.c: Wow, connected! [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'START' is '1371023093' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'RECFILENAME' is '2204-1371023093' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'END' is '1371023103' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'RECFILENAME' is '2204-1371023093' [Jun 12 09:45:03] DEBUG[20572] pbx.c: Result of 'END' is '1371023103' [Jun 12 09:45:03] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:45:03] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Getvar' [Jun 12 09:45:03] DEBUG[20680] manager.c: Running action 'Logoff' [Jun 12 09:45:03] VERBOSE[20680] manager.c: == Manager 'obelisk' logged off from 130.10.237.168 [Jun 12 09:45:03] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:45:03] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:45:03] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:45:03] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:45:03] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:45:03] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - soft2202 [Jun 12 09:45:03] DEBUG[18137] chan_sip.c: Checking device state for peer soft2202 [Jun 12 09:45:03] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - web2202 [Jun 12 09:45:03] DEBUG[18137] chan_sip.c: Checking device state for peer web2202 [Jun 12 09:45:03] DEBUG[18137] app_queue.c: Extension '2202@hints' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 12 09:45:03] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - soft2202 [Jun 12 09:45:03] DEBUG[18137] chan_sip.c: Checking device state for peer soft2202 [Jun 12 09:45:03] DEBUG[18137] devicestate.c: No provider found, checking channel drivers for SIP - web2202 [Jun 12 09:45:03] DEBUG[18137] chan_sip.c: Checking device state for peer web2202 [Jun 12 09:45:03] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 12 09:45:03] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 12 09:45:03] DEBUG[18171] app_queue.c: Device 'SIP/2202' changed to state '1' (Not in use) [Jun 12 09:45:03] DEBUG[18171] app_queue.c: Device 'SIP/2202' changed to state '1' (Not in use) [Jun 12 09:45:03] DEBUG[18161] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2202 Context: hints Hint: SIP/2202&SIP/soft2202&SIP/web2202 Status: 0 [Jun 12 09:45:03] DEBUG[19616] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2202 Context: hints Hint: SIP/2202&SIP/soft2202&SIP/web2202 Status: 0 [Jun 12 09:45:03] DEBUG[20059] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 2202 Context: hints Hint: SIP/2202&SIP/soft2202&SIP/web2202 Status: 0 [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Challenge' [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Login' [Jun 12 09:45:04] VERBOSE[20783] manager.c: == Manager 'obelisk' logged on from 130.10.237.168 [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Command' [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Command' [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Status' [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Agents' [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'QueueStatus' [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Getvar' [Jun 12 09:45:04] DEBUG[20783] pbx.c: Result of 'BRIDGEPEER' is NULL [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Getvar' [Jun 12 09:45:04] DEBUG[20783] pbx.c: Result of 'DIALEDPEERNUMBER' is NULL [Jun 12 09:45:04] DEBUG[20783] manager.c: Running action 'Logoff' [Jun 12 09:45:04] VERBOSE[20783] manager.c: == Manager 'obelisk' logged off from 130.10.237.168 [Jun 12 09:45:04] VERBOSE[20572] res_agi.c: -- AGI Script agi://130.10.14.249/vm-record.agi?user=2204 completed, returning 0 [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: AGI [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Expression result is '1' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:4] GotoIf("SIP/asterisk-GuruPlug-00000017", "1?agisuccess") in new stack [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Goto (macro-executedeadagi,s,7) [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: GotoIf [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:7] NoOp("SIP/asterisk-GuruPlug-00000017", "AGI SUCCESS") in new stack [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'USERID' is '2204' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'DIALSTATUS' is NULL [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'Macro' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [h@local-vmhandler:3] Macro("SIP/asterisk-GuruPlug-00000017", "executedeadagi,diallog.agi?user=2204&typename=") in new stack [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:1] NoOp("SIP/asterisk-GuruPlug-00000017", "Execute AGI") in new stack [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'AGIHOST0' is NULL [Jun 12 09:45:04] DEBUG[20572] pbx.c: Expression result is '0' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:2] GotoIf("SIP/asterisk-GuruPlug-00000017", "0?agihost0failed") in new stack [Jun 12 09:45:04] DEBUG[20572] pbx.c: Not taking any branch [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: GotoIf [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'ARG1' is 'diallog.agi?user=2204&typename=' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'AGI' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:3] AGI("SIP/asterisk-GuruPlug-00000017", "agi://130.10.14.249/diallog.agi?user=2204&typename=") in new stack [Jun 12 09:45:04] DEBUG[20572] res_agi.c: Hungup channel detected, running agi in dead mode. [Jun 12 09:45:04] DEBUG[20572] res_agi.c: Wow, connected! [Jun 12 09:45:04] VERBOSE[20572] res_agi.c: -- AGI Script agi://130.10.14.249/diallog.agi?user=2204&typename= completed, returning 0 [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: AGI [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Expression result is '1' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'GotoIf' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:4] GotoIf("SIP/asterisk-GuruPlug-00000017", "1?agisuccess") in new stack [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Goto (macro-executedeadagi,s,7) [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: GotoIf [Jun 12 09:45:04] DEBUG[20572] pbx.c: Result of 'AGISTATUS' is 'SUCCESS' [Jun 12 09:45:04] DEBUG[20572] pbx.c: Launching 'NoOp' [Jun 12 09:45:04] VERBOSE[20572] pbx.c: -- Executing [s@macro-executedeadagi:7] NoOp("SIP/asterisk-GuruPlug-00000017", "AGI SUCCESS") in new stack [Jun 12 09:45:04] DEBUG[20572] app_macro.c: Executed application: NoOp [Jun 12 09:45:04] DEBUG[20572] channel.c: Hanging up channel 'SIP/asterisk-GuruPlug-00000017' [Jun 12 09:45:04] DEBUG[20572] chan_sip.c: Hangup call SIP/asterisk-GuruPlug-00000017, SIP callid 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:45:04] DEBUG[20572] chan_sip.c: update_call_counter(asterisk-GuruPlug) - decrement call limit counter on hangup [Jun 12 09:45:04] DEBUG[20572] chan_sip.c: Updating call counter for incoming call [Jun 12 09:45:04] DEBUG[20572] chan_sip.c: Call from peer 'asterisk-GuruPlug' removed from call limit 2147483647 [Jun 12 09:45:04] DEBUG[20572] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f77c58' [Jun 12 09:45:04] DEBUG[20572] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6f36ce8' [Jun 12 09:45:04] DEBUG[18161] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 Uniqueid: 1371023084.24 CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:04] DEBUG[19616] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 Uniqueid: 1371023084.24 CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:04] DEBUG[20059] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/asterisk-GuruPlug-00000017 Uniqueid: 1371023084.24 CallerIDNum: 2202 CallerIDName: User 2202 ConnectedLineNum: ConnectedLineName: Cause: 16 Cause-txt: Normal Clearing [Jun 12 09:45:04] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:45:04] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:45:04] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:45:04] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 1 (Not in use) [Jun 12 09:45:04] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '1' [Jun 12 09:45:04] DEBUG[18136] devicestate.c: No provider found, checking channel drivers for SIP - asterisk-GuruPlug [Jun 12 09:45:04] DEBUG[18136] chan_sip.c: Checking device state for peer asterisk-GuruPlug [Jun 12 09:45:04] DEBUG[18136] devicestate.c: Changing state for SIP/asterisk-GuruPlug - state 1 (Not in use) [Jun 12 09:45:04] DEBUG[18136] devicestate.c: device 'SIP/asterisk-GuruPlug' state '1' [Jun 12 09:45:04] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:45:04] DEBUG[18132] cel_pgsql.c: inserting a CEL record. [Jun 12 09:45:04] DEBUG[18132] cel_radius.c: Unable to create RADIUS record. CEL not recorded! [Jun 12 09:45:04] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 12 09:45:04] DEBUG[18171] app_queue.c: Device 'SIP/asterisk-GuruPlug' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 12 09:45:10] DEBUG[18144] chan_sip.c: Auto destroying SIP dialog '68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060' [Jun 12 09:45:10] DEBUG[18144] chan_sip.c: Destroying SIP dialog 68b6c33a2e225fcf4ca91a195c57b558@130.10.160.5:5060 [Jun 12 09:45:10] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0xb6f77c58' [Jun 12 09:45:10] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0xb6f36ce8' [Jun 12 09:45:11] DEBUG[18144] chan_sip.c: Auto destroying SIP dialog '30504a1d3853977017124fb66f5c1875@130.10.160.5:5060' [Jun 12 09:45:11] DEBUG[18144] chan_sip.c: Destroying SIP dialog 30504a1d3853977017124fb66f5c1875@130.10.160.5:5060 [Jun 12 09:45:12] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 30286a9d13e211381e3cd1750dd5f325@130.10.20.220:5060 - OPTIONS (No RTP) [Jun 12 09:45:12] DEBUG[18144] acl.c: For destination '10.0.0.1', our source address is '130.10.20.220'. [Jun 12 09:45:12] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:45:12] DEBUG[18144] chan_sip.c: Initializing initreq for method OPTIONS - callid 2803493e32adb79b78040d067a1d119b@130.10.20.220:5060 [Jun 12 09:45:12] DEBUG[18144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.1:5060 [Jun 12 09:45:12] DEBUG[18144] chan_sip.c: = Looking for Call ID: 2803493e32adb79b78040d067a1d119b@130.10.20.220:5060 (Checking To) --From tag as054964dc --To-tag 1085747899 [Jun 12 09:45:12] DEBUG[18144] chan_sip.c: Stopping retransmission on '2803493e32adb79b78040d067a1d119b@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:45:12] DEBUG[18144] chan_sip.c: Destroying SIP dialog 2803493e32adb79b78040d067a1d119b@130.10.20.220:5060 [Jun 12 09:45:13] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:45:15] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:45:16] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:45:26] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 238dc58b1360a21b5fc645385f5117a0@130.10.20.220:5060 - OPTIONS (No RTP) [Jun 12 09:45:26] DEBUG[18144] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:45:26] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:45:26] DEBUG[18144] chan_sip.c: Initializing initreq for method OPTIONS - callid 24526dfe51fe7b6b53fb806e20191cb9@130.10.20.220:5060 [Jun 12 09:45:26] DEBUG[18144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:45:26] DEBUG[18144] chan_sip.c: Stopping retransmission on '24526dfe51fe7b6b53fb806e20191cb9@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:45:26] DEBUG[18144] chan_sip.c: Destroying SIP dialog 24526dfe51fe7b6b53fb806e20191cb9@130.10.20.220:5060 [Jun 12 09:45:28] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:45:30] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:45:31] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:45:35] DEBUG[18144] chan_sip.c: Auto destroying SIP dialog '791e658937df8782' [Jun 12 09:45:35] DEBUG[18144] chan_sip.c: Destroying SIP dialog 791e658937df8782 [Jun 12 09:45:35] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0xb6f33588' [Jun 12 09:45:35] DEBUG[18144] rtp_engine.c: Destroyed RTP instance '0xb6f76c00' [Jun 12 09:45:39] DEBUG[18144] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:45:39] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:45:39] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 59174066349d179455f409976dbb7008@130.10.160.5:5060 - OPTIONS (No RTP) [Jun 12 09:45:39] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:45:43] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:45:45] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:45:46] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:45:58] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:46:00] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:46:01] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:46:11] DEBUG[18144] chan_sip.c: Auto destroying SIP dialog '59174066349d179455f409976dbb7008@130.10.160.5:5060' [Jun 12 09:46:11] DEBUG[18144] chan_sip.c: Destroying SIP dialog 59174066349d179455f409976dbb7008@130.10.160.5:5060 [Jun 12 09:46:12] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 2f0d6c9564820bc758cd901f4e5368f1@130.10.20.220:5060 - OPTIONS (No RTP) [Jun 12 09:46:12] DEBUG[18144] acl.c: For destination '10.0.0.1', our source address is '130.10.20.220'. [Jun 12 09:46:12] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:46:12] DEBUG[18144] chan_sip.c: Initializing initreq for method OPTIONS - callid 650dcd9d45d57ce369fdb83037749ab9@130.10.20.220:5060 [Jun 12 09:46:12] DEBUG[18144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.1:5060 [Jun 12 09:46:12] DEBUG[18144] chan_sip.c: Stopping retransmission on '650dcd9d45d57ce369fdb83037749ab9@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:46:12] DEBUG[18144] chan_sip.c: Destroying SIP dialog 650dcd9d45d57ce369fdb83037749ab9@130.10.20.220:5060 [Jun 12 09:46:13] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:46:15] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:46:16] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:46:26] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 37a43b557cff16a62aab418d570c1858@130.10.20.220:5060 - OPTIONS (No RTP) [Jun 12 09:46:26] DEBUG[18144] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:46:26] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:46:26] DEBUG[18144] chan_sip.c: Initializing initreq for method OPTIONS - callid 4cba0b18556e02a2148b45b47b8372a7@130.10.20.220:5060 [Jun 12 09:46:26] DEBUG[18144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:46:26] DEBUG[18144] chan_sip.c: Stopping retransmission on '4cba0b18556e02a2148b45b47b8372a7@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:46:26] DEBUG[18144] chan_sip.c: Destroying SIP dialog 4cba0b18556e02a2148b45b47b8372a7@130.10.20.220:5060 [Jun 12 09:46:28] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:46:30] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:46:31] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:46:39] DEBUG[18144] acl.c: For destination '130.10.160.5', our source address is '130.10.20.220'. [Jun 12 09:46:39] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:46:39] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 574cfaad32ab879754ac6f3905f6d3ba@130.10.160.5:5060 - OPTIONS (No RTP) [Jun 12 09:46:39] DEBUG[18144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 130.10.160.5:5060 [Jun 12 09:46:43] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:46:45] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:46:46] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:46:58] DEBUG[20059] manager.c: Running action 'Ping' [Jun 12 09:47:00] DEBUG[18161] manager.c: Running action 'Ping' [Jun 12 09:47:01] DEBUG[19616] manager.c: Running action 'Ping' [Jun 12 09:47:11] DEBUG[18144] chan_sip.c: Auto destroying SIP dialog '574cfaad32ab879754ac6f3905f6d3ba@130.10.160.5:5060' [Jun 12 09:47:11] DEBUG[18144] chan_sip.c: Destroying SIP dialog 574cfaad32ab879754ac6f3905f6d3ba@130.10.160.5:5060 [Jun 12 09:47:12] DEBUG[18144] chan_sip.c: Allocating new SIP dialog for 36ba344b2a3c7ad308f4cc345919e65b@130.10.20.220:5060 - OPTIONS (No RTP) [Jun 12 09:47:12] DEBUG[18144] acl.c: For destination '10.0.0.1', our source address is '130.10.20.220'. [Jun 12 09:47:12] DEBUG[18144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 130.10.20.220:5060 [Jun 12 09:47:12] DEBUG[18144] chan_sip.c: Initializing initreq for method OPTIONS - callid 71f9f8c21c6d435756d16a7f56d057a5@130.10.20.220:5060 [Jun 12 09:47:12] DEBUG[18144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.1:5060 [Jun 12 09:47:12] DEBUG[18144] chan_sip.c: Stopping retransmission on '71f9f8c21c6d435756d16a7f56d057a5@130.10.20.220:5060' of Request 102: Match Found [Jun 12 09:47:12] DEBUG[18144] chan_sip.c: Destroying SIP dialog 71f9f8c21c6d435756d16a7f56d057a5@130.10.20.220:5060 [Jun 12 09:47:13] DEBUG[20059] manager.c: Running action 'Ping'