rendered paste body -- Executing Set("SIP/644-0114", "EXTTOCALL=800") in new stack
-- Executing Set("SIP/644-0114", "CFUEXT=") in new stack
-- Executing Set("SIP/644-0114", "RT=15") in new stack
-- Executing Macro("SIP/644-0114", "record-enable|800|IN") in new stack
-- Executing GotoIf("SIP/644-0114", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/644-0114", "recordingcheck|20061012-163740|1160685460.858") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20061012-163740|1160685460.858: Inbound recording enabled.
recordingcheck|20061012-163740|1160685460.858: CALLFILENAME=20061012-163740-1160685460.858
-- AGI Script recordingcheck completed, returning 0
-- Executing Monitor("SIP/644-0114", "wav49|20061012-163740-1160685460.858| mb") in new stack
-- Executing GotoIf("SIP/644-0114", "0?dolocaldial|1") in new stack
-- Executing Macro("SIP/644-0114", "dial|15|tr|800") in new stack
-- Executing AGI("SIP/644-0114", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1
dialparties.agi: Caller ID name is 'Mark Chaney' number is '644'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 800 to extension map
-- dialparties.agi: Extension 800 cf is disabled
-- dialparties.agi: Extension 800 do not disturb is disabled
> dialparties.agi: extnum: 800
> dialparties.agi: exthascw: 0
> dialparties.agi: exthascfb: 0
> dialparties.agi: extcfb:
> dialparties.agi: exthascfu: 0
> dialparties.agi: extcfu:
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
> dialparties.agi: ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 800
-- dialparties.agi: DbSet CALLTRACE/800 to 644
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/644-0114", "SIP/800|15|tr") in new stack
We're at 66.199.224.216 port 12888
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 74.139.32.6:1437:
INVITE sip:800@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK1b0c6cf1;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as6726efac
To: <sip:800@192.168.1.150:5060>
Contact: <sip:644@66.199.224.216>
Call-ID: 2c59dceb18d56fa865c4dea606a7e325@66.199.224.216
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 20:37:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 392 392 IN IP4 66.199.224.216
s=session
c=IN IP4 66.199.224.216
t=0 0
m=audio 12888 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 800
Retransmitting #1 (NAT) to 74.139.32.6:1437:
INVITE sip:800@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK1b0c6cf1;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as6726efac
To: <sip:800@192.168.1.150:5060>
Contact: <sip:644@66.199.224.216>
Call-ID: 2c59dceb18d56fa865c4dea606a7e325@66.199.224.216
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 20:37:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontent-Type: application/sdp
Content-Length: 240
v=0
o=root 392 392 IN IP4 66.199.224.216
s=session
c=IN IP4 66.199.224.216
t=0 0
m=audio 12888 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #2 (NAT) to 74.139.32.6:1437:
INVITE sip:800@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK1b0c6cf1;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as6726efac
To: <sip:800@192.168.1.150:5060>
Contact: <sip:644@66.199.224.216>
Call-ID: 2c59dceb18d56fa865c4dea606a7e325@66.199.224.216
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 20:37:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontent-Type: application/sdp
Content-Length: 240
v=0
o=root 392 392 IN IP4 66.199.224.216
s=session
c=IN IP4 66.199.224.216
t=0 0
m=audio 12888 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #3 (NAT) to 74.139.32.6:1437:
INVITE sip:800@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK1b0c6cf1;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as6726efac
To: <sip:800@192.168.1.150:5060>
Contact: <sip:644@66.199.224.216>
Call-ID: 2c59dceb18d56fa865c4dea606a7e325@66.199.224.216
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 20:37:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontent-Type: application/sdp
Content-Length: 240
v=0
o=root 392 392 IN IP4 66.199.224.216
s=session
c=IN IP4 66.199.224.216
t=0 0
m=audio 12888 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #4 (NAT) to 74.139.32.6:1437:
INVITE sip:800@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK1b0c6cf1;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as6726efac
To: <sip:800@192.168.1.150:5060>
Contact: <sip:644@66.199.224.216>
Call-ID: 2c59dceb18d56fa865c4dea606a7e325@66.199.224.216
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 20:37:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontent-Type: application/sdp
Content-Length: 240
v=0
o=root 392 392 IN IP4 66.199.224.216
s=session
c=IN IP4 66.199.224.216
t=0 0
m=audio 12888 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Nobody picked up in 15000 ms
-- Executing GosubIf("SIP/644-0114", "0?docfu|1") in new stack
-- Executing NoOp("SIP/644-0114", "Voicemail is 800") in new stack
-- Executing GotoIf("SIP/644-0114", "0?s-NOANSWER|1") in new stack
-- Executing NoOp("SIP/644-0114", "Sending to Voicemail box 800") in new stack
-- Executing Macro("SIP/644-0114", "vm|800|NOANSWER") in new stack
-- Executing Macro("SIP/644-0114", "user-callerid") in new stack
-- Executing GotoIf("SIP/644-0114", "0?report") in new stack
-- Executing GotoIf("SIP/644-0114", "1?start") in new stack
-- Goto (macro-user-callerid,s,4)
-- Executing NoOp("SIP/644-0114", "REALCALLERIDNUM is 644") in new stack
-- Executing Set("SIP/644-0114", "AMPUSER=644") in new stack
-- Executing Set("SIP/644-0114", "AMPUSERCIDNAME=Mark Chaney") in new stack
-- Executing GotoIf("SIP/644-0114", "0?report") in new stack
-- Executing Set("SIP/644-0114", "CALLERID(all)=Mark Chaney <644>") in new stack
-- Executing NoOp("SIP/644-0114", "Using CallerID "Mark Chaney" <644>") in new stack
-- Executing Set("SIP/644-0114", "VMGAIN=") in new stack
-- Executing Goto("SIP/644-0114", "s-NOANSWER|1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing Macro("SIP/644-0114", "get-vmcontext|800") in new stack
-- Executing Set("SIP/644-0114", "VMCONTEXT=default") in new stack
-- Executing GotoIf("SIP/644-0114", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing NoOp("SIP/644-0114", "") in new stack
-- Executing VoiceMail("SIP/644-0114", "800@default|u") in new stack
-- Playing 'vm-theperson' (language 'en')
Destroying call '2c59dceb18d56fa865c4dea606a7e325@66.199.224.216'
-- Playing 'digits/8' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/644-0114' in macro 'vm'
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/644-0114' in macro 'exten-vm'
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/644-0114'
-- Registered SIP '800' at 74.139.32.6 port 1443 expires 3600
maxim*CLI>