asterisk1*CLI> <-- SIP read from 192.168.1.14:21680: --- (0 headers 1 lines) --- Destroying call 'Mjc4OWE5NjVhZjQ2NGQ0ZTY5NmNmZDY2NzZhMGRiYzY.' Destroying call '4cf6aa554837e3b60a8105bd6f54a055@127.0.0.1' asterisk1*CLI> <-- SIP read from 212.98.65.15:32782: INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0 Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222- From:;tag=1949364555-1195663028222- To:"36938115 36938115" Call-ID:BW173708222211107-277357524@212.98.65.15 CSeq:831059968 INVITE Contact: Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:multipart/mixed,application/media_control+xml,application/sdp Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:284 v=0 o=BroadWorks 117961895 1 IN IP4 212.98.65.50 s=- c=IN IP4 212.98.65.50 t=0 0 m=audio 24900 RTP/AVP 8 18 0 101 c=IN IP4 212.98.65.50 a=fmtp:18 annexb=yes a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 13 lines) --- Using INVITE request as basis request - BW173708222211107-277357524@212.98.65.15 Sending to 212.98.65.15 : 5060 (NAT) Found peer 'v2tel15' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 212.98.65.50:24900 Found description format PCMA Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for v2telin36938115 in from-trunk (domain 130.226.238.29) list_route: hop: Transmitting (NAT) to 212.98.65.15:32782: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15 From: ;tag=1949364555-1195663028222- To: "36938115 36938115" Call-ID: BW173708222211107-277357524@212.98.65.15 CSeq: 831059968 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/36938115-09e71428", "__FROM_DID=v2telin36938115") in new stack -- Executing Goto("SIP/36938115-09e71428", "ext-local|115|1") in new stack -- Goto (ext-local,115,1) -- Executing Macro("SIP/36938115-09e71428", "exten-vm|novm|115") in new stack -- Executing Macro("SIP/36938115-09e71428", "user-callerid") in new stack -- Executing NoOp("SIP/36938115-09e71428", "user-callerid: 26849929") in new stack -- Executing GotoIf("SIP/36938115-09e71428", "0?report") in new stack -- Executing GotoIf("SIP/36938115-09e71428", "0?start") in new stack -- Executing Set("SIP/36938115-09e71428", "REALCALLERIDNUM=26849929") in new stack -- Executing NoOp("SIP/36938115-09e71428", "REALCALLERIDNUM is 26849929") in new stack -- Executing Set("SIP/36938115-09e71428", "AMPUSER=") in new stack -- Executing Set("SIP/36938115-09e71428", "AMPUSERCIDNAME=") in new stack -- Executing GotoIf("SIP/36938115-09e71428", "1?report") in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp("SIP/36938115-09e71428", "TTL: ARG1: novm") in new stack -- Executing GotoIf("SIP/36938115-09e71428", "0?continue") in new stack -- Executing Set("SIP/36938115-09e71428", "__TTL=64") in new stack -- Executing GotoIf("SIP/36938115-09e71428", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/36938115-09e71428", "Using CallerID "" <26849929>") in new stack -- Executing Set("SIP/36938115-09e71428", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/36938115-09e71428", "VMBOX=novm") in new stack -- Executing Set("SIP/36938115-09e71428", "EXTTOCALL=115") in new stack -- Executing Set("SIP/36938115-09e71428", "CFUEXT=") in new stack -- Executing Set("SIP/36938115-09e71428", "CFBEXT=") in new stack -- Executing Set("SIP/36938115-09e71428", "RT=") in new stack -- Executing Macro("SIP/36938115-09e71428", "record-enable|115|IN") in new stack -- Executing GotoIf("SIP/36938115-09e71428", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/36938115-09e71428", "recordingcheck|20071121-173708|1195663028.84") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20071121-173708|1195663028.84: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/36938115-09e71428", "No recording needed") in new stack -- Executing Macro("SIP/36938115-09e71428", "dial||tr|115") in new stack -- Executing DeadAGI("SIP/36938115-09e71428", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi asterisk1*CLI> <-- SIP read from 212.98.65.15:32782: INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0 Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222- From:;tag=1949364555-1195663028222- To:"36938115 36938115" Call-ID:BW173708222211107-277357524@212.98.65.15 CSeq:831059968 INVITE Contact: Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:multipart/mixed,application/media_control+xml,application/sdp Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:284 v=0 o=BroadWorks 117961895 1 IN IP4 212.98.65.50 s=- c=IN IP4 212.98.65.50 t=0 0 m=audio 24900 RTP/AVP 8 18 0 101 c=IN IP4 212.98.65.50 a=fmtp:18 annexb=yes a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 13 lines) --- Ignoring this INVITE request Transmitting (NAT) to 212.98.65.15:32782: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15 From: ;tag=1949364555-1195663028222- To: "36938115 36938115" Call-ID: BW173708222211107-277357524@212.98.65.15 CSeq: 831059968 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is '26849929' dialparties.agi: Methodology of ring is 'none' > dialparties.agi: USE_CONFIRMATION: 'FALSE' > dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 115 to extension map -- dialparties.agi: Extension 115 cf is disabled -- dialparties.agi: Extension 115 do not disturb is disabled > dialparties.agi: extnum: 115 > dialparties.agi: exthascw: 1 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: -- dialparties.agi: dbset CALLTRACE/115 to 26849929 == Manager 'admin' logged off from 127.0.0.1 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/36938115-09e71428", "SIP/115||tr") in new stack We're at 192.168.1.111 port 10514 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (NAT) to 192.168.1.13:17454: INVITE sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport From: "26849929" ;tag=as159d06dc To: Contact: Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Nov 2007 16:37:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 2523 2523 IN IP4 192.168.1.111 s=session c=IN IP4 192.168.1.111 t=0 0 m=audio 10514 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 115 Transmitting (NAT) to 212.98.65.15:32782: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15 From: ;tag=1949364555-1195663028222- To: "36938115 36938115";tag=as55b58f2d Call-ID: BW173708222211107-277357524@212.98.65.15 CSeq: 831059968 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.1.13:17454: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport=5060 Contact: To: ;tag=cf7ab53d From: "26849929";tag=as159d06dc Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111 CSeq: 102 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 --- (9 headers 0 lines) --- -- SIP/115-09e769f0 is ringing asterisk1*CLI> <-- SIP read from 212.98.65.15:32782: INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0 Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222- From:;tag=1949364555-1195663028222- To:"36938115 36938115" Call-ID:BW173708222211107-277357524@212.98.65.15 CSeq:831059968 INVITE Contact: Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:multipart/mixed,application/media_control+xml,application/sdp Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:284 v=0 o=BroadWorks 117961895 1 IN IP4 212.98.65.50 s=- c=IN IP4 212.98.65.50 t=0 0 m=audio 24900 RTP/AVP 8 18 0 101 c=IN IP4 212.98.65.50 a=fmtp:18 annexb=yes a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 13 lines) --- Ignoring this INVITE request Transmitting (NAT) to 212.98.65.15:32782: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15 From: ;tag=1949364555-1195663028222- To: "36938115 36938115" Call-ID: BW173708222211107-277357524@212.98.65.15 CSeq: 831059968 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.1.5:52136: --- (0 headers 1 lines) --- asterisk1*CLI> <-- SIP read from 192.168.1.3:15436: --- (0 headers 1 lines) --- asterisk1*CLI> <-- SIP read from 212.98.65.15:32782: INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0 Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222- From:;tag=1949364555-1195663028222- To:"36938115 36938115" Call-ID:BW173708222211107-277357524@212.98.65.15 CSeq:831059968 INVITE Contact: Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:multipart/mixed,application/media_control+xml,application/sdp Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:284 v=0 o=BroadWorks 117961895 1 IN IP4 212.98.65.50 s=- c=IN IP4 212.98.65.50 t=0 0 m=audio 24900 RTP/AVP 8 18 0 101 c=IN IP4 212.98.65.50 a=fmtp:18 annexb=yes a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 13 lines) --- Ignoring this INVITE request Transmitting (NAT) to 212.98.65.15:32782: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15 From: ;tag=1949364555-1195663028222- To: "36938115 36938115" Call-ID: BW173708222211107-277357524@212.98.65.15 CSeq: 831059968 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.1.13:17454: SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport=5060 To: ;tag=cf7ab53d From: "26849929";tag=as159d06dc Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111 CSeq: 102 INVITE User-Agent: X-Lite release 1011s stamp 41150 Warning: 399 devnull "User reject" Content-Length: 0 --- (9 headers 0 lines) --- -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.1.13 Transmitting (NAT) to 192.168.1.13:17454: ACK sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport From: "26849929" ;tag=as159d06dc To: ;tag=cf7ab53d Contact: Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/115-09e769f0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Set("SIP/36938115-09e71428", "SV_DIALSTATUS=CONGESTION") in new stack -- Executing GosubIf("SIP/36938115-09e71428", "0?docfu|1") in new stack -- Executing GosubIf("SIP/36938115-09e71428", "0?docfb|1") in new stack -- Executing Set("SIP/36938115-09e71428", "DIALSTATUS=CONGESTION") in new stack -- Executing NoOp("SIP/36938115-09e71428", "Voicemail is novm") in new stack -- Executing GotoIf("SIP/36938115-09e71428", "1?s-CONGESTION|1") in new stack -- Goto (macro-exten-vm,s-CONGESTION,1) -- Executing PlayTones("SIP/36938115-09e71428", "congestion") in new stack We're at 130.226.238.29 port 14872 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 212.98.65.15:32782: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15 From: ;tag=1949364555-1195663028222- To: "36938115 36938115";tag=as55b58f2d Call-ID: BW173708222211107-277357524@212.98.65.15 CSeq: 831059968 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2523 2523 IN IP4 130.226.238.29 s=session c=IN IP4 130.226.238.29 t=0 0 m=audio 14872 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Congestion("SIP/36938115-09e71428", "10") in new stack Transmitting (NAT) to 212.98.65.15:32782: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15 From: ;tag=1949364555-1195663028222- To: "36938115 36938115";tag=as55b58f2d Call-ID: BW173708222211107-277357524@212.98.65.15 CSeq: 831059968 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: User alerting, no answer --- == Spawn extension (macro-exten-vm, s-CONGESTION, 2) exited non-zero on 'SIP/36938115-09e71428' in macro 'exten-vm' == Spawn extension (macro-exten-vm, s-CONGESTION, 2) exited non-zero on 'SIP/36938115-09e71428' Destroying call '44f7628f094a42de64b89e5714b000f6@192.168.1.111' Destroying call 'BW173708222211107-277357524@212.98.65.15' asterisk1*CLI> <-- SIP read from 192.168.1.2:36142: --- (0 headers 1 lines) --- asterisk1*CLI> <-- SIP read from 192.168.1.13:17454: --- (0 headers 1 lines) --- asterisk1*CLI>