-- Accepting call from '7327041000' to '7327049020' on channel 0/1, span 1 -- Executing [7327049020@default:1] Macro("DAHDI/1-1", "writeincomingsipcallid") in new stack -- Executing [s@macro-writeincomingsipcallid:1] Set("DAHDI/1-1", "CDR(SIPCALLID1)=") in new stack [Mar 10 11:20:18] DEBUG[16786]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [7327049020@default:2] Goto("DAHDI/1-1", "incoming|7327049020|1") in new stack -- Goto (incoming,7327049020,1) -- Executing [7327049020@incoming:1] Dial("DAHDI/1-1", "SIP/7327049020@fsa-fsdev") in new stack Audio is at 64.19.145.13 port 15576 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 64.19.145.4:5060: INVITE sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK52d25ffe;rport From: "asterisk" ;tag=as61584e50 To: Contact: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Mar 2009 15:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 184 v=0 o=root 16694 16694 IN IP4 64.19.145.13 s=session c=IN IP4 64.19.145.13 t=0 0 m=audio 15576 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 7327049020@fsa-fsdev fsdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK52d25ffe;received=64.19.145.13;rport=5060 From: "asterisk" ;tag=as61584e50 To: ;tag=as090eb864 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7eb7bcaf" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 64.19.145.4:5060: ACK sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK52d25ffe;rport From: "asterisk" ;tag=as61584e50 To: ;tag=as090eb864 Contact: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 64.19.145.13 port 15576 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 64.19.145.4:5060: INVITE sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4769e2bc;rport From: "asterisk" ;tag=as61584e50 To: Contact: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="fsdev", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.4", nonce="7eb7bcaf", response="7c223f1741161aa43859549e026f4eb9" Date: Tue, 10 Mar 2009 15:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 184 v=0 o=root 16694 16695 IN IP4 64.19.145.13 s=session c=IN IP4 64.19.145.13 t=0 0 m=audio 15576 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- fsdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4769e2bc;received=64.19.145.13;rport=5060 From: "asterisk" ;tag=as61584e50 To: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- fsdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4769e2bc;received=64.19.145.13;rport=5060 From: "asterisk" ;tag=as61584e50 To: ;tag=as4ad462f6 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- SIP/fsa-fsdev-09c4df40 is ringing fsdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4769e2bc;received=64.19.145.13;rport=5060 From: "asterisk" ;tag=as61584e50 To: ;tag=as4ad462f6 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 12304 12304 IN IP4 64.19.145.4 s=session c=IN IP4 64.19.145.4 t=0 0 m=audio 17268 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (12 headers 10 lines) --- Found RTP audio format 0 Peer audio RTP is at port 64.19.145.4:17268 Found audio description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 64.19.145.4:17268 list_route: hop: [Mar 10 11:20:22] DEBUG[16769]: chan_sip.c:6153 reqprep: Strict routing enforced for session 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 set_destination: Parsing for address/port to send to set_destination: set destination to 64.19.145.4, port 5060 Transmitting (NAT) to 64.19.145.4:5060: ACK sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK29345ccf;rport From: "asterisk" ;tag=as61584e50 To: ;tag=as4ad462f6 Contact: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fsdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> INVITE sip:asterisk@64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;rport From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Contact: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 186 v=0 o=root 12304 12305 IN IP4 64.19.186.173 s=session c=IN IP4 64.19.186.173 t=0 0 m=audio 10000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> -- SIP/fsa-fsdev-09c4df40 answered DAHDI/1-1 --- (13 headers 10 lines) --- Scheduling destruction of SIP dialog '1b0c8d42467afe7a592a8f501910acde@64.19.145.13' in 6400 ms (Method: INVITE) [Mar 10 11:20:22] DEBUG[16786]: chan_sip.c:6153 reqprep: Strict routing enforced for session 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 set_destination: Parsing for address/port to send to set_destination: set destination to 64.19.145.4, port 5060 Reliably Transmitting (NAT) to 64.19.145.4:5060: BYE sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK57096a05;rport From: "asterisk" ;tag=as61584e50 To: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="fsdev", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.4", nonce="7eb7bcaf", response="f62046127af6a34e9ae6c41e152d659f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (incoming, 7327049020, 1) exited non-zero on 'DAHDI/1-1' [Mar 10 11:20:22] DEBUG[16786]: chan_dahdi.c:3366 dahdi_setoption: Set option AUDIO MODE, value: ON(1) on DAHDI/1-1 [Mar 10 11:20:22] DEBUG[16786]: chan_dahdi.c:2998 dahdi_hangup: Not yet hungup... Calling hangup once with icause, and clearing call [Mar 10 11:20:22] DEBUG[16786]: chan_dahdi.c:3362 dahdi_setoption: Set option AUDIO MODE, value: OFF(0) on DAHDI/1-1 -- Hungup 'DAHDI/1-1' <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK57096a05;received=64.19.145.13;rport=5060 From: "asterisk" ;tag=as61584e50 To: ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '1b0c8d42467afe7a592a8f501910acde@64.19.145.13' Method: INVITE fsdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> INVITE sip:asterisk@64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;rport From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Contact: Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 186 v=0 o=root 12304 12305 IN IP4 64.19.186.173 s=session c=IN IP4 64.19.186.173 t=0 0 m=audio 10000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 10 lines) --- Sending to 64.19.145.4 : 5060 (NAT) Using INVITE request as basis request - 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 Found peer 'fsa-fsdev' <--- Reliably Transmitting (NAT) to 64.19.145.4:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;received=64.19.145.4;rport=5060 From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3957dc4f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1b0c8d42467afe7a592a8f501910acde@64.19.145.13' in 6400 ms (Method: INVITE) Retransmitting #1 (NAT) to 64.19.145.4:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;received=64.19.145.4;rport=5060 From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3957dc4f" Content-Length: 0 --- Retransmitting #2 (NAT) to 64.19.145.4:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;received=64.19.145.4;rport=5060 From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3957dc4f" Content-Length: 0 --- Retransmitting #3 (NAT) to 64.19.145.4:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;received=64.19.145.4;rport=5060 From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3957dc4f" Content-Length: 0 --- Retransmitting #4 (NAT) to 64.19.145.4:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;received=64.19.145.4;rport=5060 From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3957dc4f" Content-Length: 0 --- Retransmitting #5 (NAT) to 64.19.145.4:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;received=64.19.145.4;rport=5060 From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3957dc4f" Content-Length: 0 --- Really destroying SIP dialog '5a7a6ff23c82c0be08d839f77467cb58@64.19.145.4' Method: OPTIONS Retransmitting #6 (NAT) to 64.19.145.4:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK7d74ef9b;received=64.19.145.4;rport=5060 From: ;tag=as4ad462f6 To: "asterisk" ;tag=as61584e50 Call-ID: 1b0c8d42467afe7a592a8f501910acde@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3957dc4f" Content-Length: 0 ---