<-- SIP read from 204.147.183.18:5060:
INVITE sip:s@204.147.183.18:5060 SIP/2.0
Record-Route: <sip:204.147.183.18;ftag=09e88f42c0411fbd3c36b08559d7b315;lr=on>
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKca18.d35f4442.0
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK78b502d79fc30624e613d354f6c5a7ec;rport=1024
Max-Forwards: 16
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315
To: <sip:08918917@sip-ny1.stanaphone.com>
Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
Contact: Anonymous <sip:204.147.183.55:1024>
Expires: 300
User-Agent: Sippy
cisco-GUID: 4050610959-437850580-2926939359-138976334
h323-conf-id: 4050610959-437850580-2926939359-138976334
Content-Length: 357
Content-Type: application/sdp

v=0
o=CiscoSystemsSIP-GW-UserAgent 6436 6997 IN IP4 204.147.183.18
s=SIP Call
i=oldc:c=IN IP4 66.114.240.12
t=0 0
m=audio 60744 RTP/AVP 0 18 3 101
c=IN IP4 204.147.183.18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
a=nortpproxy:yes

--- (16 headers 15 lines) ---
Using INVITE request as basis request - F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12
Sending to 204.147.183.18 : 5060 (NAT)
Found peer 'stanap'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 204.147.183.18:60744
Found description format PCMU
Found description format G729
Found description format GSM
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in from-trunk (domain 204.147.183.18)
list_route: hop: <sip:204.147.183.18;ftag=09e88f42c0411fbd3c36b08559d7b315;lr=on>
Transmitting (NAT) to 204.147.183.18:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKca18.d35f4442.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK78b502d79fc30624e613d354f6c5a7ec;rport=1024
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315
To: <sip:08918917@sip-ny1.stanaphone.com>
Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.103.144.10>
Content-Length: 0


---
    -- Executing Set("SIP/08918917-b6306f50", "FROM_DID=s") in new stack
    -- Executing Goto("SIP/08918917-b6306f50", "ext-group|600|1") in new stack
    -- Goto (ext-group,600,1)
    -- Executing Macro("SIP/08918917-b6306f50", "user-callerid|") in new stack
    -- Executing NoOp("SIP/08918917-b6306f50", "user-callerid:  9166067933") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "0?report") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "0?start") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "REALCALLERIDNUM=9166067933") in new stack
    -- Executing NoOp("SIP/08918917-b6306f50", "REALCALLERIDNUM is 9166067933") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "AMPUSER=") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "AMPUSERCIDNAME=") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "1?report") in new stack
    -- Goto (macro-user-callerid,s,11)
    -- Executing NoOp("SIP/08918917-b6306f50", "TTL:  ARG1: ") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "0?continue") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "_TTL=64") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/08918917-b6306f50", "Using CallerID "" <9166067933>") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "1?skipdb") in new stack
    -- Goto (ext-group,600,4)
    -- Executing Set("SIP/08918917-b6306f50", "__NODEST=") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "__BLKVM_OVERRIDE=BLKVM/600/SIP/08918917-b6306f50") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "__BLKVM_BASE=600") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "DB(BLKVM/600/SIP/08918917-b6306f50)=TRUE") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "RRNODEST=") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "__NODEST=600") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "1?REPCID") in new stack
    -- Goto (ext-group,600,14)
    -- Executing NoOp("SIP/08918917-b6306f50", "CALLERID(name) is ") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "RecordMethod=Group") in new stack
    -- Executing Macro("SIP/08918917-b6306f50", "record-enable|1234-2134-3122|Group") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/08918917-b6306f50", "recordingcheck|20070206-135502|1170788102.54") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/08918917-b6306f50", "No recording needed") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "RingGroupMethod=ringall") in new stack
    -- Executing Macro("SIP/08918917-b6306f50", "dial|20|trw|1234-2134-3122") in new stack
    -- Executing DeadAGI("SIP/08918917-b6306f50", "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 'unknown' number is '9166067933'
  dialparties.agi: Methodology of ring is  'ringall'
    --  dialparties.agi: Added extension 1234 to extension map
    --  dialparties.agi: Added extension 2134 to extension map
    --  dialparties.agi: Added extension 3122 to extension map
    --  dialparties.agi: Extension 1234 cf is disabled
    --  dialparties.agi: Extension 2134 cf is disabled
    --  dialparties.agi: Extension 3122 cf is disabled
    --  dialparties.agi: Extension 1234 do not disturb is disabled
    --  dialparties.agi: Extension 2134 do not disturb is disabled
    --  dialparties.agi: Extension 3122 do not disturb is disabled
    --  dialparties.agi: dbset CALLTRACE/1234 to 9166067933
    --  dialparties.agi: dbset CALLTRACE/2134 to 9166067933
    --  dialparties.agi: dbset CALLTRACE/3122 to 9166067933
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Dial("SIP/08918917-b6306f50", "SIP/19166067933&SIP/2134&IAX2/3122|20|trwM(auto-blkvm)") in new stack
Destroying call '3bd8ccd750fa8d8d0f27817438d06071@127.0.0.1'
Destroying call '4a12e65132e89def702da9482e483d83@127.0.0.1'
  == Everyone is busy/congested at this time (3:0/0/3)
    -- Executing Set("SIP/08918917-b6306f50", "RingGroupMethod=") in new stack
    -- Executing GotoIf("SIP/08918917-b6306f50", "0?nodest") in new stack
    -- Executing Set("SIP/08918917-b6306f50", "__NODEST=") in new stack
    -- Executing DBdel("SIP/08918917-b6306f50", "BLKVM/600/SIP/08918917-b6306f50") in new stack
    -- DBdel: family=BLKVM, key=600/SIP/08918917-b6306f50
    -- Executing Goto("SIP/08918917-b6306f50", "app-blackhole|busy|1") in new stack
    -- Goto (app-blackhole,busy,1)
    -- Executing NoOp("SIP/08918917-b6306f50", "Blackhole Dest: Busy") in new stack
    -- Executing Answer("SIP/08918917-b6306f50", "") in new stack
We're at 38.103.144.10 port 19286
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKca18.d35f4442.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK78b502d79fc30624e613d354f6c5a7ec;rport=1024
Record-Route: <sip:204.147.183.18;ftag=09e88f42c0411fbd3c36b08559d7b315;lr=on>
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315
To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as79a9e528
Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.103.144.10>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 12008 12008 IN IP4 38.103.144.10
s=session
c=IN IP4 38.103.144.10
t=0 0
m=audio 19286 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing PlayTones("SIP/08918917-b6306f50", "busy") in new stack
    -- Executing Busy("SIP/08918917-b6306f50", "20") in new stack
PP1*CLI>
<-- SIP read from 204.147.183.18:5060:
ACK sip:s@204.147.183.18:5060 SIP/2.0
Via: SIP/2.0/UDP 204.147.183.18;branch=0
Via: SIP/2.0/UDP 204.147.183.55:1024;rport=1024;branch=z9hG4bKe2a7227874947282377905a055c235ed
Max-Forwards: 16
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315
To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as79a9e528
Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12
CSeq: 200 ACK
Expires: 300
User-Agent: Sippy