<-- SIP read from 204.147.183.18:5060:INVITE sip:s@204.147.183.18:5060 SIP/2.0Record-Route: <sip:204.147.183.18;ftag=09e88f42c0411fbd3c36b08559d7b315;lr=on>Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKca18.d35f4442.0Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK78b502d79fc30624e613d354f6c5a7ec;rport=1024Max-Forwards: 16From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315To: <sip:08918917@sip-ny1.stanaphone.com>Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12CSeq: 200 INVITEContact: Anonymous <sip:204.147.183.55:1024>Expires: 300User-Agent: Sippycisco-GUID: 4050610959-437850580-2926939359-138976334h323-conf-id: 4050610959-437850580-2926939359-138976334Content-Length: 357Content-Type: application/sdpv=0o=CiscoSystemsSIP-GW-UserAgent 6436 6997 IN IP4 204.147.183.18s=SIP Calli=oldc:c=IN IP4 66.114.240.12t=0 0m=audio 60744 RTP/AVP 0 18 3 101c=IN IP4 204.147.183.18a=rtpmap:0 PCMU/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=yesa=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=direction:passivea=nortpproxy:yes--- (16 headers 15 lines) ---Using INVITE request as basis request - F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12Sending to 204.147.183.18 : 5060 (NAT)Found peer 'stanap'Found RTP audio format 0Found RTP audio format 18Found RTP audio format 3Found RTP audio format 101Peer audio RTP is at port 204.147.183.18:60744Found description format PCMUFound description format G729Found description format GSMFound description format telephone-eventCapabilities: 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 TryingVia: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKca18.d35f4442.0;received=204.147.183.18Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK78b502d79fc30624e613d354f6c5a7ec;rport=1024From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315To: <sip:08918917@sip-ny1.stanaphone.com>Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12CSeq: 200 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <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 stackDestroying 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 stackWe're at 38.103.144.10 port 19286Adding codec 0x4 (ulaw) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (NAT) to 204.147.183.18:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKca18.d35f4442.0;received=204.147.183.18Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK78b502d79fc30624e613d354f6c5a7ec;rport=1024Record-Route: <sip:204.147.183.18;ftag=09e88f42c0411fbd3c36b08559d7b315;lr=on>From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as79a9e528Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12CSeq: 200 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Type: application/sdpContent-Length: 218v=0o=root 12008 12008 IN IP4 38.103.144.10s=sessionc=IN IP4 38.103.144.10t=0 0m=audio 19286 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - ---- -- Executing PlayTones("SIP/08918917-b6306f50", "busy") in new stack -- Executing Busy("SIP/08918917-b6306f50", "20") in new stackPP1*CLI><-- SIP read from 204.147.183.18:5060:ACK sip:s@204.147.183.18:5060 SIP/2.0Via: SIP/2.0/UDP 204.147.183.18;branch=0Via: SIP/2.0/UDP 204.147.183.55:1024;rport=1024;branch=z9hG4bKe2a7227874947282377905a055c235edMax-Forwards: 16From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=09e88f42c0411fbd3c36b08559d7b315To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as79a9e528Call-ID: F1731497-1A1911D4-AE788CDF-8489C4E@66.114.240.12CSeq: 200 ACKExpires: 300User-Agent: Sippy