rendered paste body -- Started music on hold, class 'default', on SIP/pbx.kicks-ass.net-08cc8e60
-- Executing Macro("SIP/100-08c98958", "dialout-trunk|2|8614900||") in new stack
-- Executing Set("SIP/100-08c98958", "DIAL_TRUNK=2") in new stack
-- Executing Set("SIP/100-08c98958", "_NODEST=") in new stack
-- Executing Set("SIP/100-08c98958", "DIAL_NUMBER=8614900") in new stack
-- Executing Set("SIP/100-08c98958", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/100-08c98958", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/100-08c98958", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,8)
-- Executing Set("SIP/100-08c98958", "GROUP()=OUT_2") in new stack
-- Executing Macro("SIP/100-08c98958", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/100-08c98958", "user-callerid: device 100") in new stack
-- Executing GotoIf("SIP/100-08c98958", "0?report") in new stack
-- Executing GotoIf("SIP/100-08c98958", "0?start") in new stack
-- Executing Set("SIP/100-08c98958", "REALCALLERIDNUM=100") in new stack
-- Executing NoOp("SIP/100-08c98958", "REALCALLERIDNUM is 100") in new stack
-- Executing Set("SIP/100-08c98958", "AMPUSER=100") in new stack
-- Executing Set("SIP/100-08c98958", "AMPUSERCIDNAME=test") in new stack
-- Executing GotoIf("SIP/100-08c98958", "0?report") in new stack
-- Executing Set("SIP/100-08c98958", "CALLERID(all)=test <100>") in new stack
-- Executing Set("SIP/100-08c98958", "REALCALLERIDNUM=100") in new stack
-- Executing NoOp("SIP/100-08c98958", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/100-08c98958", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/100-08c98958", "Using CallerID "test" <100>") in new stack
-- Executing Macro("SIP/100-08c98958", "record-enable|100|OUT") in new stack
-- Executing GotoIf("SIP/100-08c98958", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/100-08c98958", "recordingcheck|20070529-081939|1180441179.16") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070529-081939|1180441179.16: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/100-08c98958", "No recording needed") in new stack
-- Executing GotoIf("SIP/100-08c98958", "0?skipoutcid") in new stack
-- Executing Set("SIP/100-08c98958", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/100-08c98958", "outbound-callerid|2") in new stack
-- Executing GotoIf("SIP/100-08c98958", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/100-08c98958", "REALCALLERIDNUM is 100") in new stack
-- Executing GotoIf("SIP/100-08c98958", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/100-08c98958", "USEROUTCID=") in new stack
-- Executing Set("SIP/100-08c98958", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/100-08c98958", "TRUNKOUTCID=3525471396") in new stack
-- Executing GotoIf("SIP/100-08c98958", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/100-08c98958", "0?usercid") in new stack
-- Executing Set("SIP/100-08c98958", "CALLERID(all)=3525471396") in new stack
-- Executing GotoIf("SIP/100-08c98958", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/100-08c98958", "CallerID set to "" <3525471396>") in new stack
-- Executing GotoIf("SIP/100-08c98958", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,16)
-- Executing DeadAGI("SIP/100-08c98958", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 011.
> fixlocalprefix: Using pattern 1800NXXXXXX
> fixlocalprefix: Using pattern 1866NXXXXXX
> fixlocalprefix: Using pattern 1877NXXXXXX
> fixlocalprefix: Using pattern 1888NXXXXXX
> fixlocalprefix: Using pattern 352|NXXXXXX
> fixlocalprefix: Using pattern NXXNXXXXXX
> fixlocalprefix: Using pattern NXXXXXX
== fixlocalprefix: Dialpattern NXXXXXX matched. 8614900 -> 8614900
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/100-08c98958", "OUTNUM=8614900") in new stack
-- Executing Set("SIP/100-08c98958", "custom=SIP/vtwhite") in new stack
-- Executing GotoIf("SIP/100-08c98958", "0?customtrunk") in new stack
-- Executing Dial("SIP/100-08c98958", "SIP/vtwhite/8614900|300|") in new stack
We're at 76.4.108.63 port 14606
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (NAT) to 216.120.251.172:5060:
INVITE sip:8614900@sip.vtwhite.com SIP/2.0
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK26b76728;rport
From: "3525471396" <sip:3525471396@sip.vtwhite.com>;tag=as4ae6c158
To: <sip:8614900@sip.vtwhite.com>
Contact: <sip:3525471396@76.4.108.63>
Call-ID: 0c2eee0330c03e597611578338d6fb2d@sip.vtwhite.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 29 May 2007 12:19:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 212
v=0
o=root 3118 3118 IN IP4 76.4.108.63
s=session
c=IN IP4 76.4.108.63
t=0 0
m=audio 14606 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called vtwhite/8614900
asterisk1*CLI>
<-- SIP read from 216.120.251.172:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK26b76728;rport=10110
From: "3525471396" <sip:3525471396@sip.vtwhite.com>;tag=as4ae6c158
To: <sip:8614900@sip.vtwhite.com>
Call-ID: 0c2eee0330c03e597611578338d6fb2d@sip.vtwhite.com
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 216.120.251.172:5060 "Noisy feedback tells: pid=8515 req_src_ip=76.4.108.63 req_src_port=10110 in_uri=sip:8614900@sip.vtwhite.com out_uri=sip:8614900@sip.vtwhite.com via_cnt==1"
--- (9 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 216.120.251.172:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK26b76728;rport=10110
From: "3525471396" <sip:3525471396@sip.vtwhite.com>;tag=as4ae6c158
To: <sip:8614900@sip.vtwhite.com>
Call-ID: 0c2eee0330c03e597611578338d6fb2d@sip.vtwhite.com
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 216.120.251.172:5060 "Noisy feedback tells: pid=8515 req_src_ip=76.4.108.63 req_src_port=10110 in_uri=sip:8614900@sip.vtwhite.com out_uri=sip:+8614900@ser1.vtnoc.net via_cnt==1"
--- (9 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 216.120.251.172:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK26b76728;rport=10110
Record-Route: <sip:64.118.82.166;lr=on;ftag=as4ae6c158>,<sip:216.120.251.172;lr=on;ftag=as4ae6c158>
From: "3525471396" <sip:3525471396@sip.vtwhite.com>;tag=as4ae6c158
To: <sip:8614900@sip.vtwhite.com>;tag=VPST506071629579
Call-ID: 0c2eee0330c03e597611578338d6fb2d@sip.vtwhite.com
CSeq: 102 INVITE
Contact: <sip:+8614900@4.68.251.11:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 184
v=0
o=- 1180441180 1180441181 IN IP4 209.244.42.249
s=-
c=IN IP4 209.244.42.249
t=0 0
m=audio 60352 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.244.42.249:60352
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/vtwhite-08c94950 is making progress passing it to SIP/100-08c98958
12 headers, 0 lines
Reliably Transmitting (NAT) to 216.120.251.172:5060:
OPTIONS sip:sip.vtwhite.com SIP/2.0
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK6abf598e;rport
From: "Unknown" <sip:Unknown@76.4.108.63>;tag=as5c19d615
To: <sip:sip.vtwhite.com>
Contact: <sip:Unknown@76.4.108.63>
Call-ID: 680028e217182d12640f32632889acb6@76.4.108.63
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 29 May 2007 12:19:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 216.120.251.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK6abf598e;rport=10110
From: "Unknown" <sip:Unknown@76.4.108.63>;tag=as5c19d615
To: <sip:sip.vtwhite.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.7541
Call-ID: 680028e217182d12640f32632889acb6@76.4.108.63
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Support:
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 216.120.251.172:5060 "Noisy feedback tells: pid=8511 req_src_ip=76.4.108.63 req_src_port=10110 in_uri=sip:sip.vtwhite.com out_uri=sip:sip.vtwhite.com via_cnt==1"
--- (13 headers 0 lines) ---
Destroying call '680028e217182d12640f32632889acb6@76.4.108.63'
asterisk1*CLI>