SIP Debugging enabled <--- SIP read from UDP:77.72.169.129:5060 ---> INVITE sip:77XXXXX1@92.37.7.105:5060 SIP/2.0 Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK7443ab1277294938aad26d2dd4a6b65f From: "+04XXXXX11" ;tag=120313ac506904dd1c56779 To: Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 CSeq: 0 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 317 v=0 o=axxxxx6 1354991299 1354991299 IN IP4 80.239.235.99 s=SIP Call c=IN IP4 80.239.235.99 t=0 0 m=audio 11698 RTP/AVP 0 8 3 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 <-------------> --- (11 headers 14 lines) --- Sending to 77.72.169.129:5060 (NAT) Using INVITE request as basis request - c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 Found peer 'in12voip' for '+04XXXXX11' from 77.72.169.129:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 80.239.235.99:11698 Looking for 77XXXXX1 in from-12v (domain 92.37.7.105) list_route: hop: <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK7443ab1277294938aad26d2dd4a6b65f;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c56779 To: Call-ID: c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 CSeq: 0 INVITE Server: FPBX-2.8.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [77XXXXX1@from-12v:1] GotoIf("SIP/in12voip-00000000", "0?noplusatstart") in new stack -- Executing [77XXXXX1@from-12v:2] NoOp("SIP/in12voip-00000000", "Changing Caller ID number from +04XXXXX11 to 04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:3] NoOp("SIP/in12voip-00000000", "Changing Caller ID name from +04XXXXX11 to 04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:4] Set("SIP/in12voip-00000000", "CALLERID(num)=04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:5] Set("SIP/in12voip-00000000", "CALLERID(name)=04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:6] NoOp("SIP/in12voip-00000000", "Changing Caller ID name from "04XXXXX11" <04XXXXX11> to 04XXXXX11" <04XXXXX11>") in new stack -- Executing [77XXXXX1@from-12v:7] Goto("SIP/in12voip-00000000", "from-trunk,77XXXXX1,1") in new stack -- Goto (from-trunk,77XXXXX1,1) -- Executing [77XXXXX1@from-trunk:1] Set("SIP/in12voip-00000000", "__FROM_DID=77XXXXX1") in new stack -- Executing [77XXXXX1@from-trunk:2] Gosub("SIP/in12voip-00000000", "app-blacklist-check,s,1") in new stack -- Executing [s@app-blacklist-check:1] GotoIf("SIP/in12voip-00000000", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:2] Set("SIP/in12voip-00000000", "CALLED_BLACKLIST=1") in new stack -- Executing [s@app-blacklist-check:3] Return("SIP/in12voip-00000000", "") in new stack -- Executing [77XXXXX1@from-trunk:3] ExecIf("SIP/in12voip-00000000", "0 ?Set(CALLERID(name)=04XXXXX11)") in new stack -- Executing [77XXXXX1@from-trunk:4] Set("SIP/in12voip-00000000", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [77XXXXX1@from-trunk:5] Set("SIP/in12voip-00000000", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [77XXXXX1@from-trunk:6] Goto("SIP/in12voip-00000000", "from-did-direct,106,1") in new stack -- Goto (from-did-direct,106,1) -- Executing [106@from-did-direct:1] Macro("SIP/in12voip-00000000", "exten-vm,novm,106") in new stack -- Executing [s@macro-exten-vm:1] Macro("SIP/in12voip-00000000", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/in12voip-00000000", "AMPUSER=04XXXXX11") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/in12voip-00000000", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/in12voip-00000000", "1?Set(REALCALLERIDNUM=04XXXXX11)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/in12voip-00000000", "AMPUSER=") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/in12voip-00000000", "AMPUSERCIDNAME=") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/in12voip-00000000", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("SIP/in12voip-00000000", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/in12voip-00000000", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("SIP/in12voip-00000000", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/in12voip-00000000", "CALLERID(number)=04XXXXX11") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/in12voip-00000000", "CALLERID(name)=04XXXXX11") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/in12voip-00000000", "Using CallerID "04XXXXX11" <04XXXXX11>") in new stack -- Executing [s@macro-exten-vm:2] Set("SIP/in12voip-00000000", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:3] Set("SIP/in12voip-00000000", "VMBOX=novm") in new stack -- Executing [s@macro-exten-vm:4] Set("SIP/in12voip-00000000", "__EXTTOCALL=106") in new stack -- Executing [s@macro-exten-vm:5] Set("SIP/in12voip-00000000", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:6] Set("SIP/in12voip-00000000", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:7] Set("SIP/in12voip-00000000", "RT=""") in new stack -- Executing [s@macro-exten-vm:8] Macro("SIP/in12voip-00000000", "record-enable,106,IN") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/in12voip-00000000", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/in12voip-00000000", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/in12voip-00000000", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/in12voip-00000000", "1?IN") in new stack -- Goto (macro-record-enable,s,20) -- Executing [s@macro-record-enable:20] ExecIf("SIP/in12voip-00000000", "1?MacroExit()") in new stack -- Executing [s@macro-exten-vm:9] Macro("SIP/in12voip-00000000", "dial-one,"",tr,106") in new stack -- Executing [s@macro-dial-one:1] Set("SIP/in12voip-00000000", "DEXTEN=106") in new stack -- Executing [s@macro-dial-one:2] Set("SIP/in12voip-00000000", "DIALSTATUS_CW=") in new stack -- Executing [s@macro-dial-one:3] GosubIf("SIP/in12voip-00000000", "0?screen,1") in new stack -- Executing [s@macro-dial-one:4] GosubIf("SIP/in12voip-00000000", "0?cf,1") in new stack -- Executing [s@macro-dial-one:5] GotoIf("SIP/in12voip-00000000", "1?skip1") in new stack -- Goto (macro-dial-one,s,8) -- Executing [s@macro-dial-one:8] GotoIf("SIP/in12voip-00000000", "0?nodial") in new stack -- Executing [s@macro-dial-one:9] GotoIf("SIP/in12voip-00000000", "0?continue") in new stack -- Executing [s@macro-dial-one:10] Set("SIP/in12voip-00000000", "EXTHASCW=") in new stack -- Executing [s@macro-dial-one:11] GotoIf("SIP/in12voip-00000000", "1?next1:cwinusebusy") in new stack -- Goto (macro-dial-one,s,12) -- Executing [s@macro-dial-one:12] GotoIf("SIP/in12voip-00000000", "0?docfu:skip3") in new stack -- Goto (macro-dial-one,s,16) -- Executing [s@macro-dial-one:16] GotoIf("SIP/in12voip-00000000", "1?next2:continue") in new stack -- Goto (macro-dial-one,s,17) -- Executing [s@macro-dial-one:17] GotoIf("SIP/in12voip-00000000", "1?continue") in new stack -- Goto (macro-dial-one,s,25) -- Executing [s@macro-dial-one:25] GotoIf("SIP/in12voip-00000000", "0?nodial") in new stack -- Executing [s@macro-dial-one:26] GosubIf("SIP/in12voip-00000000", "1?dstring,1:dlocal,1") in new stack -- Executing [dstring@macro-dial-one:1] Set("SIP/in12voip-00000000", "DSTRING=") in new stack -- Executing [dstring@macro-dial-one:2] Set("SIP/in12voip-00000000", "DEVICES=106") in new stack -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/in12voip-00000000", "0?Return()") in new stack -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/in12voip-00000000", "0?Set(DEVICES=06)") in new stack -- Executing [dstring@macro-dial-one:5] Set("SIP/in12voip-00000000", "LOOPCNT=1") in new stack -- Executing [dstring@macro-dial-one:6] Set("SIP/in12voip-00000000", "ITER=1") in new stack -- Executing [dstring@macro-dial-one:7] Set("SIP/in12voip-00000000", "THISDIAL=SIP/106") in new stack -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/in12voip-00000000", "1?zap2dahdi,1") in new stack -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/in12voip-00000000", "0?Return()") in new stack -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/in12voip-00000000", "NEWDIAL=") in new stack -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/in12voip-00000000", "LOOPCNT2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/in12voip-00000000", "ITER2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/in12voip-00000000", "THISPART2=SIP/106") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/in12voip-00000000", "0?Set(THISPART2=DAHDI/106)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/in12voip-00000000", "NEWDIAL=SIP/106&") in new stack -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/in12voip-00000000", "ITER2=2") in new stack -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/in12voip-00000000", "0?begin2") in new stack -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/in12voip-00000000", "THISDIAL=SIP/106") in new stack -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/in12voip-00000000", "") in new stack -- Executing [dstring@macro-dial-one:9] Set("SIP/in12voip-00000000", "DSTRING=SIP/106&") in new stack -- Executing [dstring@macro-dial-one:10] Set("SIP/in12voip-00000000", "ITER=2") in new stack -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/in12voip-00000000", "0?begin") in new stack -- Executing [dstring@macro-dial-one:12] Set("SIP/in12voip-00000000", "DSTRING=SIP/106") in new stack -- Executing [dstring@macro-dial-one:13] Return("SIP/in12voip-00000000", "") in new stack -- Executing [s@macro-dial-one:27] GotoIf("SIP/in12voip-00000000", "0?nodial") in new stack -- Executing [s@macro-dial-one:28] GotoIf("SIP/in12voip-00000000", "0?skiptrace") in new stack -- Executing [s@macro-dial-one:29] GosubIf("SIP/in12voip-00000000", "1?ctset,1:ctclear,1") in new stack -- Executing [ctset@macro-dial-one:1] Set("SIP/in12voip-00000000", "DB(CALLTRACE/106)=04XXXXX11") in new stack -- Executing [ctset@macro-dial-one:2] Return("SIP/in12voip-00000000", "") in new stack -- Executing [s@macro-dial-one:30] Set("SIP/in12voip-00000000", "D_OPTIONS=tr") in new stack -- Executing [s@macro-dial-one:31] ExecIf("SIP/in12voip-00000000", "0?SIPAddHeader(Alert-Info: )") in new stack -- Executing [s@macro-dial-one:32] ExecIf("SIP/in12voip-00000000", "0?SIPAddHeader()") in new stack -- Executing [s@macro-dial-one:33] ExecIf("SIP/in12voip-00000000", "0?Set(CHANNEL(musicclass)=)") in new stack -- Executing [s@macro-dial-one:34] GosubIf("SIP/in12voip-00000000", "0?qwait,1") in new stack -- Executing [s@macro-dial-one:35] Set("SIP/in12voip-00000000", "__CWIGNORE=") in new stack -- Executing [s@macro-dial-one:36] Set("SIP/in12voip-00000000", "__KEEPCID=TRUE") in new stack -- Executing [s@macro-dial-one:37] Dial("SIP/in12voip-00000000", "SIP/106,"",tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 13030 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.150:5062: INVITE sip:106@192.168.0.150:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK11b150f5;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as0fb73b90 To: Contact: Call-ID: 29b523d34cd7e193670f086873a4fc7d@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.15.0) Date: Sat, 08 Dec 2012 18:28:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 1126686595 1126686595 IN IP4 192.168.0.11 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.0.11 t=0 0 m=audio 13030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called SIP/106 <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK7443ab1277294938aad26d2dd4a6b65f;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c56779 To: ;tag=as16cb5fb8 Call-ID: c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 CSeq: 0 INVITE Server: FPBX-2.8.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:192.168.0.150:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK11b150f5;rport From: "04XXXXX11" ;tag=as0fb73b90 To: Call-ID: 29b523d34cd7e193670f086873a4fc7d@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: Yealink SIP-T22P 7.60.79.7 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.0.150:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK11b150f5;rport From: "04XXXXX11" ;tag=as0fb73b90 To: ;tag=1376295483 Call-ID: 29b523d34cd7e193670f086873a4fc7d@192.168.0.11:5060 CSeq: 102 INVITE Contact: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T22P 7.60.79.7 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 <-------------> --- (11 headers 0 lines) --- list_route: hop: -- SIP/106-00000001 is ringing <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK7443ab1277294938aad26d2dd4a6b65f;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c56779 To: ;tag=as16cb5fb8 Call-ID: c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 CSeq: 0 INVITE Server: FPBX-2.8.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Really destroying SIP dialog '3077384e773c083e52e86177583b75b7@38.126.208.166' Method: REGISTER <--- SIP read from UDP:77.72.169.129:5060 ---> CANCEL sip:77XXXXX1@92.37.7.105:5060 SIP/2.0 Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK7443ab1277294938aad26d2dd4a6b65f From: "+04XXXXX11" ;tag=120313ac506904dd1c56779 To: ;tag=as16cb5fb8 Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 CSeq: 0 CANCEL Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK7443ab1277294938aad26d2dd4a6b65f;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c56779 To: ;tag=as16cb5fb8 Call-ID: c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 CSeq: 0 CANCEL Server: FPBX-2.8.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:77.72.169.129:5060 ---> ACK sip:77XXXXX1@92.37.7.105:5060 SIP/2.0 Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bKfb6fe384f1b34529899c76e9ce9aa5dc From: "+04XXXXX11" ;tag=120313ac506904dd1c56779 To: ;tag=as16cb5fb8 Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: c96d9eaa0d3c4e49a0deef29b7fe75ef@77.72.169.129 CSeq: 0 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.0.150:5062 ---> <-------------> elastix*CLI> exit