rendered paste body--- (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: <sip:67.55.159.160;ftag=as62ba7f40;lr=on>
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" <sip:FreeDigits@67.55.159.156>;tag=as62ba7f40
To: <sip:3199859638@freedigits.net>
Contact: <sip:FreeDigits@67.55.159.156:5060>
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: <sip:67.55.159.160;ftag=as62ba7f40;lr=on>
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" <sip:FreeDigits@67.55.159.156>;tag=as62ba7f40
To: <sip:3199859638@freedigits.net>
Call-ID: 41a7ff391ab346054850a0bd4e1b3ffd@67.55.159.156
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3199859638@65.27.185.178>
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" <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" <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: <sip:67.55.159.160;ftag=as62ba7f40;lr=on>
From: "FreeDigits" <sip:FreeDigits@67.55.159.156>;tag=as62ba7f40
To: <sip:3199859638@freedigits.net>;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: <sip:3199859638@65.27.185.178>
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" <sip:FreeDigits@67.55.159.156>;tag=as62ba7f40
To: <sip:3199859638@freedigits.net>;tag=as1f163b67
Contact: <sip:FreeDigits@67.55.159.156:5060>
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" <sip:FreeDigits@67.55.159.156>;tag=as62ba7f40
To: <sip:3199859638@freedigits.net>;tag=as1f163b67
Contact: <sip:FreeDigits@67.55.159.156:5060>
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" <sip:FreeDigits@67.55.159.156>;tag=as62ba7f40
To: <sip:3199859638@freedigits.net>;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: <sip:3199859638@65.27.185.178>
Content-Length: 0