<-------------> [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: <--- SIP read from TLS:172.23.10.95:5061 ---> INVITE sip:49501@192.168.233.250:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e Route: Max-Forwards: 70 From: ;tag=db0c2b65a4 To: Call-ID: 386ec67fd1f94f0f CSeq: 378731545 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE Allow-Events: talk, hold Contact: Supported: replaces, 100rel User-Agent: OpenStage_60_V3 R0.40.0 SIP DEV X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 319 v=0 o=OpenStage-Line_0 261517019 893362067 IN IP4 172.23.10.95 s=SIP Call c=IN IP4 172.23.10.95 t=0 0 m=audio 5010 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/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 <-------------> [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: --- (16 headers 14 lines) --- [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Sending to 172.23.10.95:5061 (no NAT) [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Using INVITE request as basis request - 386ec67fd1f94f0f [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found peer '49502' for '49502' from 172.23.10.95:5061 [Aug 18 10:46:49] VERBOSE[18491] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found RTP audio format 8 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found RTP audio format 0 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found RTP audio format 18 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found RTP audio format 101 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found audio description format G729 for ID 18 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Peer audio RTP is at port 172.23.10.95:5010 [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: Looking for 49501 in sen (domain 192.168.233.250:5061) [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: list_route: hop: [Aug 18 10:46:49] VERBOSE[18491] chan_sip.c: <--- Transmitting (no NAT) to 172.23.10.95:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e;received=172.23.10.95 From: ;tag=db0c2b65a4 To: Call-ID: 386ec67fd1f94f0f CSeq: 378731545 INVITE Server: Asterisk PBX 1.8.4-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 18 10:46:49] VERBOSE[21636] pbx.c: -- Executing [49501@sen:1] Set("SIP/49502-00000014", "CONNECTEDLINE(all)=49501 <49501>") in new stack [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: <--- Transmitting (no NAT) to 172.23.10.95:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e;received=172.23.10.95 From: ;tag=db0c2b65a4 To: ;tag=as60f9c8ef Call-ID: 386ec67fd1f94f0f CSeq: 378731545 INVITE Server: Asterisk PBX 1.8.4-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "49501" Content-Length: 0 <------------> [Aug 18 10:46:49] VERBOSE[21636] pbx.c: -- Executing [49501@sen:2] Dial("SIP/49502-00000014", "SIP/49501") in new stack [Aug 18 10:46:49] VERBOSE[21636] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: Audio is at 5060 [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: Reliably Transmitting (no NAT) to 172.23.10.96:5060: INVITE sip:49501@172.23.10.96:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.233.250:5060;branch=z9hG4bK52b58f81 Max-Forwards: 70 From: "49502" ;tag=as393ff3c6 To: Contact: Call-ID: 21548279214a162f5247c0f669b78fab@192.168.233.250:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.4-rc2 Date: Thu, 18 Aug 2011 09:46:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "49502" Content-Type: application/sdp Content-Length: 316 v=0 o=root 524542948 524542948 IN IP4 192.168.233.250 s=Asterisk PBX 1.8.4-rc2 c=IN IP4 192.168.233.250 t=0 0 m=audio 17106 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Aug 18 10:46:49] VERBOSE[21636] app_dial.c: -- Called 49501 [Aug 18 10:46:49] VERBOSE[9660] chan_sip.c: <--- SIP read from UDP:172.23.10.96:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.233.250:5060;branch=z9hG4bK52b58f81 From: "49502" ;tag=as393ff3c6 To: ;tag=2486063310 Call-ID: 21548279214a162f5247c0f669b78fab@192.168.233.250:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE Allow-Events: hold Contact: Server: OpenStage_80_V3 R0.36.0 SIP DEV Content-Length: 0 <-------------> [Aug 18 10:46:49] VERBOSE[9660] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:46:49] VERBOSE[21636] app_dial.c: -- SIP/49501-00000015 is ringing [Aug 18 10:46:49] VERBOSE[21636] chan_sip.c: <--- Transmitting (no NAT) to 172.23.10.95:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e;received=172.23.10.95 From: ;tag=db0c2b65a4 To: ;tag=as60f9c8ef Call-ID: 386ec67fd1f94f0f CSeq: 378731545 INVITE Server: Asterisk PBX 1.8.4-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Call-Info: ;purpose=call-completion;m=NR Content-Length: 0 <------------> [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: <--- SIP read from TLS:172.23.10.95:5061 ---> CANCEL sip:49501@192.168.233.250:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e Route: Max-Forwards: 70 From: ;tag=db0c2b65a4 To: Call-ID: 386ec67fd1f94f0f CSeq: 378731545 CANCEL User-Agent: OpenStage_60_V3 R0.40.0 SIP DEV Warning: 399 172.23.10.95 "Call is terminated" Content-Length: 0 <-------------> [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: Sending to 172.23.10.95:5061 (no NAT) [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.23.10.95:5061 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e;received=172.23.10.95 From: ;tag=db0c2b65a4 To: ;tag=as60f9c8ef Call-ID: 386ec67fd1f94f0f CSeq: 378731545 INVITE Server: Asterisk PBX 1.8.4-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Call-Info: ;purpose=call-completion;m=NR Content-Length: 0 <------------> [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: <--- Transmitting (no NAT) to 172.23.10.95:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e;received=172.23.10.95 From: ;tag=db0c2b65a4 To: ;tag=as60f9c8ef Call-ID: 386ec67fd1f94f0f CSeq: 378731545 CANCEL Server: Asterisk PBX 1.8.4-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Call-Info: ;purpose=call-completion;m=NR Content-Length: 0 <------------> [Aug 18 10:46:52] VERBOSE[21636] chan_sip.c: Scheduling destruction of SIP dialog '21548279214a162f5247c0f669b78fab@192.168.233.250:5060' in 32000 ms (Method: INVITE) [Aug 18 10:46:52] VERBOSE[21636] chan_sip.c: Reliably Transmitting (no NAT) to 172.23.10.96:5060: CANCEL sip:49501@172.23.10.96:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.233.250:5060;branch=z9hG4bK52b58f81 Max-Forwards: 70 From: "49502" ;tag=as393ff3c6 To: Call-ID: 21548279214a162f5247c0f669b78fab@192.168.233.250:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.8.4-rc2 Content-Length: 0 --- [Aug 18 10:46:52] VERBOSE[21636] chan_sip.c: Scheduling destruction of SIP dialog '21548279214a162f5247c0f669b78fab@192.168.233.250:5060' in 32000 ms (Method: INVITE) [Aug 18 10:46:52] VERBOSE[21636] pbx.c: == Spawn extension (sen, 49501, 2) exited non-zero on 'SIP/49502-00000014' [Aug 18 10:46:52] VERBOSE[9660] chan_sip.c: Really destroying SIP dialog '386ec67fd1f94f0f' Method: CANCEL [Aug 18 10:46:52] VERBOSE[9660] chan_sip.c: <--- SIP read from UDP:172.23.10.96:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.233.250:5060;branch=z9hG4bK52b58f81 From: "49502" ;tag=as393ff3c6 To: ;tag=2486063310 Call-ID: 21548279214a162f5247c0f669b78fab@192.168.233.250:5060 CSeq: 102 CANCEL Server: OpenStage_80_V3 R0.36.0 SIP DEV Content-Length: 0 <-------------> [Aug 18 10:46:52] VERBOSE[9660] chan_sip.c: --- (8 headers 0 lines) --- [Aug 18 10:46:52] VERBOSE[9660] chan_sip.c: <--- SIP read from UDP:172.23.10.96:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.233.250:5060;branch=z9hG4bK52b58f81 From: "49502" ;tag=as393ff3c6 To: ;tag=2486063310 Call-ID: 21548279214a162f5247c0f669b78fab@192.168.233.250:5060 CSeq: 102 INVITE Server: OpenStage_80_V3 R0.36.0 SIP DEV Content-Length: 0 <-------------> [Aug 18 10:46:52] VERBOSE[9660] chan_sip.c: --- (8 headers 0 lines) --- [Aug 18 10:46:52] VERBOSE[9660] chan_sip.c: Transmitting (no NAT) to 172.23.10.96:5060: ACK sip:49501@172.23.10.96:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.233.250:5060;branch=z9hG4bK52b58f81 Max-Forwards: 70 From: "49502" ;tag=as393ff3c6 To: ;tag=2486063310 Contact: Call-ID: 21548279214a162f5247c0f669b78fab@192.168.233.250:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.4-rc2 Content-Length: 0 --- [Aug 18 10:46:52] VERBOSE[9660] chan_sip.c: Really destroying SIP dialog '21548279214a162f5247c0f669b78fab@192.168.233.250:5060' Method: INVITE [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: <--- SIP read from TLS:172.23.10.95:5061 ---> ACK sip:49501@192.168.233.250:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK110ae4cc8711bd17e Route: Max-Forwards: 70 From: ;tag=db0c2b65a4 To: ;tag=as60f9c8ef Call-ID: 386ec67fd1f94f0f CSeq: 378731545 ACK User-Agent: OpenStage_60_V3 R0.40.0 SIP DEV Content-Length: 0 <-------------> [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: <--- SIP read from TLS:172.23.10.95:5061 ---> SUBSCRIBE sip:07ff955f1a84858d257ed9745dc0b281@192.168.233.250:5061;transport=tls SIP/2.0 Accept: application/call-completion Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK17c485e121fc32486 Route: Max-Forwards: 70 From: ;tag=271b80fe1b;epid=SC25e334 To: Call-ID: 73f7d3e74e87ca8e CSeq: 406322410 SUBSCRIBE Allow: NOTIFY Contact: Event: call-completion Expires: 3600 User-Agent: OpenStage_60_V3 R0.40.0 SIP DEV Content-Length: 0 <-------------> [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: --- (15 headers 0 lines) --- [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: Creating new subscription [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: Sending to 172.23.10.95:5061 (no NAT) [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: list_route: hop: [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: Found peer '49502' for '49502' from 172.23.10.95:5061 [Aug 18 10:46:52] WARNING[18491] chan_sip.c: Invalid URI 'sip:07ff955f1a84858d257ed9745dc0b281@192.168.233.250:5061' in CC subscribe [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: <--- Transmitting (no NAT) to 172.23.10.95:5061 ---> SIP/2.0 404 Not Found Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK17c485e121fc32486;received=172.23.10.95 From: ;tag=271b80fe1b;epid=SC25e334 To: ;tag=as0612ec2e Call-ID: 73f7d3e74e87ca8e CSeq: 406322410 SUBSCRIBE Server: Asterisk PBX 1.8.4-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: Scheduling destruction of SIP dialog '73f7d3e74e87ca8e' in 3610000 ms (Method: SUBSCRIBE) [Aug 18 10:46:52] NOTICE[18491] chan_sip.c: Got SUBSCRIBE for extension 07ff955f1a84858d257ed9745dc0b281@sen from 172.23.10.95:5061, but there is no hint for that extension. [Aug 18 10:46:52] VERBOSE[18491] chan_sip.c: <--- Transmitting (no NAT) to 172.23.10.95:5061 ---> SIP/2.0 404 Not found Via: SIP/2.0/TLS 172.23.10.95:5061;branch=z9hG4bK17c485e121fc32486;received=172.23.10.95 From: ;tag=271b80fe1b;epid=SC25e334 To: ;tag=as0612ec2e Call-ID: 73f7d3e74e87ca8e CSeq: 406322410 SUBSCRIBE Server: Asterisk PBX 1.8.4-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:47:03] VERBOSE[9660] chan_sip.c: <--- SIP read from UDP:172.23.4.45:5060 ---> <-------------> [Aug 18 10:47:23] VERBOSE[9660] chan_sip.c: <--- SIP read from UDP:172.23.4.45:5060 ---> <-------------> [Aug 18 10:47:43] VERBOSE[9660] chan_sip.c: <--- SIP read from UDP:172.23.4.45:5060 --->