rendered paste bodymaxim*CLI> sip debug peer 701
SIP Debugging Enabled for IP: 75.23.69.69:5062
-- Executing Macro("SIP/644-1e76", "exten-vm|novm|701") in new stack
-- Executing Macro("SIP/644-1e76", "user-callerid") in new stack
-- Executing GotoIf("SIP/644-1e76", "0?report") in new stack
-- Executing GotoIf("SIP/644-1e76", "0?start") in new stack
-- Executing Set("SIP/644-1e76", "REALCALLERIDNUM=644") in new stack
-- Executing NoOp("SIP/644-1e76", "REALCALLERIDNUM is 644") in new stack
-- Executing Set("SIP/644-1e76", "AMPUSER=644") in new stack
-- Executing Set("SIP/644-1e76", "AMPUSERCIDNAME=Mark Chaney") in new stack
-- Executing GotoIf("SIP/644-1e76", "0?report") in new stack
-- Executing Set("SIP/644-1e76", "CALLERID(all)=Mark Chaney <644>") in new stack
-- Executing NoOp("SIP/644-1e76", "Using CallerID "Mark Chaney" <644>") in new stack
-- Executing Set("SIP/644-1e76", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("SIP/644-1e76", "VMBOX=novm") in new stack
-- Executing Set("SIP/644-1e76", "EXTTOCALL=701") in new stack
-- Executing Set("SIP/644-1e76", "CFUEXT=") in new stack
-- Executing Set("SIP/644-1e76", "RT=") in new stack
-- Executing Macro("SIP/644-1e76", "record-enable|701|IN") in new stack
-- Executing GotoIf("SIP/644-1e76", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/644-1e76", "recordingcheck|20061012-170357|1160687037.866") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20061012-170357|1160687037.866: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/644-1e76", "No recording needed") in new stack
-- Executing GotoIf("SIP/644-1e76", "0?dolocaldial|1") in new stack
-- Executing Macro("SIP/644-1e76", "dial||tr|701") in new stack
-- Executing AGI("SIP/644-1e76", "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 701 to extension map
-- dialparties.agi: Extension 701 cf is disabled
-- dialparties.agi: Extension 701 do not disturb is disabled
> dialparties.agi: extnum: 701
> 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 701
-- dialparties.agi: DbSet CALLTRACE/701 to 644
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/644-1e76", "SIP/701||tr") in new stack
We're at 66.199.224.216 port 14666
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 75.23.69.69:5062:
INVITE sip:701@192.168.1.151:5062 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as706cc5ae
To: <sip:701@192.168.1.151:5062>
Contact: <sip:644@66.199.224.216>
Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 21:03:59 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 14666 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 701
maxim*CLI>
<-- SIP read from 75.23.69.69:5062:
SIP/2.0 100 Trying
To: <sip:701@192.168.1.151:5062>
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as706cc5ae
Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
--- (8 headers 0 lines)---
maxim*CLI>
<-- SIP read from 75.23.69.69:5062:
SIP/2.0 180 Ringing
To: <sip:701@192.168.1.151:5062>;tag=612b8910bf0630d7i0
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as706cc5ae
Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
--- (8 headers 0 lines)---
-- SIP/701-9628 is ringing
12 headers, 0 lines
Reliably Transmitting (NAT) to 75.23.69.69:5062:
OPTIONS sip:701@192.168.1.151:5062 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK02e580e1;rport
From: "Unknown" <sip:Unknown@66.199.224.216>;tag=as1f7402e5
To: <sip:701@192.168.1.151:5062>
Contact: <sip:Unknown@66.199.224.216>
Call-ID: 7060e3c902f21f42179febe9594eecde@66.199.224.216
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 21:04:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
maxim*CLI>
<-- SIP read from 75.23.69.69:5062:
SIP/2.0 200 OK
To: <sip:701@192.168.1.151:5062>;tag=7a86b35ab04987cdi0
From: "Unknown" <sip:Unknown@66.199.224.216>;tag=as1f7402e5
Call-ID: 7060e3c902f21f42179febe9594eecde@66.199.224.216
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK02e580e1
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
--- (10 headers 0 lines)---
Destroying call '7060e3c902f21f42179febe9594eecde@66.199.224.216'
Reliably Transmitting (NAT) to 75.23.69.69:5062:
CANCEL sip:701@192.168.1.151:5062 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as706cc5ae
To: <sip:701@192.168.1.151:5062>
Contact: <sip:644@66.199.224.216>
Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call '33c3314969e66c014b6522f23e3ffd3a@66.199.224.216' in 32000 ms
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/644-1e76' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/644-1e76' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/644-1e76'
maxim*CLI>
<-- SIP read from 75.23.69.69:5062:
SIP/2.0 487 Request Terminated
To: <sip:701@192.168.1.151:5062>;tag=612b8910bf0630d7i0
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as706cc5ae
Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
--- (8 headers 0 lines)---
Transmitting (NAT) to 75.23.69.69:5062:
ACK sip:701@192.168.1.151:5062 SIP/2.0
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4;rport
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as706cc5ae
To: <sip:701@192.168.1.151:5062>;tag=612b8910bf0630d7i0
Contact: <sip:644@66.199.224.216>
Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
maxim*CLI>
<-- SIP read from 75.23.69.69:5062:
SIP/2.0 200 OK
To: <sip:701@192.168.1.151:5062>;tag=612b8910bf0630d7i0
From: "Mark Chaney" <sip:644@66.199.224.216>;tag=as706cc5ae
Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '33c3314969e66c014b6522f23e3ffd3a@66.199.224.216'
maxim*CLI>