--- (9 headers 0 lines)--- Destroying call '17b71f0428967f603fdecc9242766b6f@65.27.185.178' mfrisch*CLI> <-- SIP read from 67.55.159.160:5060: --- (0 headers 0 lines) Nat keepalive --- mfrisch*CLI> <-- SIP read from 67.55.159.160:5060: INVITE sip:3199859638@65.27.185.178:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKd6e.e8ed189.0 Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK35a7b029;rport=5060 From: "FreeDigits" ;tag=as62ba7f40 To: Contact: Call-ID: 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 16 Date: Wed, 23 Aug 2006 01:11:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 441 v=0 o=root 27784 27784 IN IP4 67.55.159.156 s=session c=IN IP4 67.55.159.156 t=0 0 m=audio 18184 RTP/AVP 0 8 3 18 111 5 10 7 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 19 lines)--- Using INVITE request as basis request - 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156 Sending to 67.55.159.160 : 5060 (non-NAT) Found peer 'freedigits' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 111 Found RTP audio format 5 Found RTP audio format 10 Found RTP audio format 7 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 67.55.159.156:18184 Found description format PCMU Found description format PCMA Found description format GSM Found description format G729 Found description format G726-32 Found description format DVI4 Found description format L16 Found description format LPC Found description format iLBC Found description format telephone-event Capabilities: us - 0x2 (gsm), peer - audio=0x5fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|ilbc)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 3199859638 in from-trunk (domain 65.27.185.178) list_route: hop: Transmitting (NAT) to 67.55.159.160:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKd6e.e8ed189.0;received=67.55.159.160 Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK35a7b029;rport=5060 From: "FreeDigits" ;tag=as62ba7f40 To: Call-ID: 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/3199859638-081ecdd8", "FROM_DID=3199859638") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "FAX_RX=420") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "FAX_RX_EMAIL=mfrisch@mfrisch.hopto.org") in new stack -- Executing Goto("SIP/3199859638-081ecdd8", "ext-local|420|1") in new stack -- Goto (ext-local,420,1) -- Executing Macro("SIP/3199859638-081ecdd8", "exten-vm|420|420") in new stack -- Executing Macro("SIP/3199859638-081ecdd8", "user-callerid") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "0?report") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "0?start") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "REALCALLERIDNUM=FreeDigits") in new stack -- Executing NoOp("SIP/3199859638-081ecdd8", "REALCALLERIDNUM is FreeDigits") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "AMPUSER=") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "AMPUSERCIDNAME=") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "1?report") in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp("SIP/3199859638-081ecdd8", "Using CallerID "FreeDigits" ") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "VMBOX=420") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "EXTTOCALL=420") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "CFUEXT=") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "RT=15") in new stack -- Executing Macro("SIP/3199859638-081ecdd8", "record-enable|420|IN") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/3199859638-081ecdd8", "recordingcheck|20060822-211201|1156295521.90") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/3199859638-081ecdd8", "No recording needed") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "0?dolocaldial|1") in new stack -- Executing Macro("SIP/3199859638-081ecdd8", "dial|15|tr|420") in new stack -- Executing AGI("SIP/3199859638-081ecdd8", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp("SIP/3199859638-081ecdd8", "Returned from dialparties with no extensions to call") in new stack -- Executing NoOp("SIP/3199859638-081ecdd8", "DIALSTATUS is ") in new stack -- Executing GosubIf("SIP/3199859638-081ecdd8", "0?docfu|1") in new stack -- Executing NoOp("SIP/3199859638-081ecdd8", "Voicemail is 420") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "0?s-|1") in new stack -- Executing NoOp("SIP/3199859638-081ecdd8", "Sending to Voicemail box 420") in new stack -- Executing Macro("SIP/3199859638-081ecdd8", "vm|420|") in new stack -- Executing Macro("SIP/3199859638-081ecdd8", "user-callerid") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "0?report") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "1?start") in new stack -- Goto (macro-user-callerid,s,4) -- Executing NoOp("SIP/3199859638-081ecdd8", "REALCALLERIDNUM is FreeDigits") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "AMPUSER=") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "AMPUSERCIDNAME=") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "1?report") in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp("SIP/3199859638-081ecdd8", "Using CallerID "FreeDigits" ") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "VMGAIN=") in new stack -- Executing Goto("SIP/3199859638-081ecdd8", "s-|1") in new stack -- Goto (macro-vm,s-,1) -- Executing Macro("SIP/3199859638-081ecdd8", "get-vmcontext|420") in new stack -- Executing Set("SIP/3199859638-081ecdd8", "VMCONTEXT=default") in new stack -- Executing GotoIf("SIP/3199859638-081ecdd8", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing NoOp("SIP/3199859638-081ecdd8", "") in new stack -- Executing VoiceMail("SIP/3199859638-081ecdd8", "420@default|u") in new stack We're at 65.27.185.178 port 11430 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 67.55.159.160:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKd6e.e8ed189.0;received=67.55.159.160 Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK35a7b029;rport=5060 Record-Route: From: "FreeDigits" ;tag=as62ba7f40 To: ;tag=as1f163b67 Call-ID: 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=root 5282 5282 IN IP4 65.27.185.178 s=session c=IN IP4 65.27.185.178 t=0 0 m=audio 11430 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'vm-theperson' (language 'en') mfrisch*CLI> <-- SIP read from 67.55.159.160:5060: ACK sip:3199859638@65.27.185.178:5060 SIP/2.0 Via: SIP/2.0/UDP 67.55.159.160;branch=0 Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK5241ca31;rport=5060 From: "FreeDigits" ;tag=as62ba7f40 To: ;tag=as1f163b67 Contact: Call-ID: 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 16 Content-Length: 0 --- (11 headers 0 lines)--- -- Playing 'digits/4' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') mfrisch*CLI> <-- SIP read from 67.55.159.160:5060: --- (0 headers 0 lines) Nat keepalive --- Destroying call '7ca5fb4138f30e8071bd20561d565088@127.0.0.1' -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/420/tmp/b18WVg format: wav49, 0x81b5960 -- x=1, open writing: /var/spool/asterisk/voicemail/default/420/tmp/b18WVg format: gsm, 0x818fb98 -- x=2, open writing: /var/spool/asterisk/voicemail/default/420/tmp/b18WVg format: wav, 0x8187ed0 mfrisch*CLI> <-- SIP read from 67.55.159.160:5060: BYE sip:3199859638@65.27.185.178:5060 SIP/2.0 Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKe6e.dd5167f6.0 Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK29e05a13;rport=5060 From: "FreeDigits" ;tag=as62ba7f40 To: ;tag=as1f163b67 Contact: Call-ID: 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 16 Content-Length: 0 --- (11 headers 0 lines)--- Sending to 67.55.159.160 : 5060 (NAT) Transmitting (NAT) to 67.55.159.160:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKe6e.dd5167f6.0;received=67.55.159.160 Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK29e05a13;rport=5060 From: "FreeDigits" ;tag=as62ba7f40 To: ;tag=as1f163b67 Call-ID: 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0