-- 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" ;tag=as4ae6c158 To: Contact: 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" ;tag=as4ae6c158 To: 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" ;tag=as4ae6c158 To: 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: , From: "3525471396" ;tag=as4ae6c158 To: ;tag=VPST506071629579 Call-ID: 0c2eee0330c03e597611578338d6fb2d@sip.vtwhite.com CSeq: 102 INVITE Contact: 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" ;tag=as5c19d615 To: Contact: 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" ;tag=as5c19d615 To: ;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>