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=z9hG4bK0097d9d566b742259fccce591af73635 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@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 1354991911 1354991911 IN IP4 77.72.168.110 s=SIP Call c=IN IP4 77.72.168.110 t=0 0 m=audio 11342 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 - 80f0a25e8fc24bfb923fd4233210e12e@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 77.72.168.110:11342 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=z9hG4bK0097d9d566b742259fccce591af73635;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@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,105,1") in new stack -- Goto (from-did-direct,105,1) -- Executing [105@from-did-direct:1] Macro("SIP/in12voip-00000000", "exten-vm,novm,105") 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=105") 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,105,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,105") in new stack -- Executing [s@macro-dial-one:1] Set("SIP/in12voip-00000000", "DEXTEN=105") 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=105") 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=05)") 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/105") 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/105") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/in12voip-00000000", "0?Set(THISPART2=DAHDI/105)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/in12voip-00000000", "NEWDIAL=SIP/105&") 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/105") 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/105&") 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/105") 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/105)=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/105,"",tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 17274 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 192.168.0.2:16410: INVITE sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK185ba256;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as4e4830ec To: Contact: Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.15.0) Date: Sat, 08 Dec 2012 18:38:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 144566831 144566831 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 17274 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/105 <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK0097d9d566b742259fccce591af73635;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: ;tag=as76833e47 Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@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 <------------> Retransmitting #1 (NAT) to 192.168.0.2:16410: INVITE sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK185ba256;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as4e4830ec To: Contact: Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.15.0) Date: Sat, 08 Dec 2012 18:38:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 144566831 144566831 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 17274 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 --- <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK185ba256;rport=5060 Contact: To: ;tag=3758686a From: "04XXXXX11";tag=as4e4830ec Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- list_route: hop: <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK185ba256;rport=5060 Contact: To: ;tag=3758686a From: "04XXXXX11";tag=as4e4830ec Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- list_route: hop: -- SIP/105-00000001 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=z9hG4bK0097d9d566b742259fccce591af73635;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: ;tag=as76833e47 Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@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/105-00000001 is ringing <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK185ba256;rport=5060 Contact: To: ;tag=3758686a From: "04XXXXX11";tag=as4e4830ec Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 183 v=0 o=- 2 2 IN IP4 192.168.0.2 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.2 t=0 0 m=audio 60422 RTP/AVP 8 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.0.2:60422 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.2:16410 Transmitting (NAT) to 192.168.0.2:16410: ACK sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK524c6716;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as4e4830ec To: ;tag=3758686a Contact: Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.8.15.0) Content-Length: 0 --- -- SIP/105-00000001 answered SIP/in12voip-00000000 Audio is at 16874 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK0097d9d566b742259fccce591af73635;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: ;tag=as76833e47 Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@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-Type: application/sdp Content-Length: 233 v=0 o=root 399717030 399717030 IN IP4 92.37.7.105 s=Asterisk PBX 1.8.15.0 c=IN IP4 92.37.7.105 t=0 0 m=audio 16874 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- 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=z9hG4bK0097d9d566b742259fccce591af73635 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: ;tag=as76833e47 Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@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:77.72.169.129:5060 ---> BYE sip:77XXXXX1@92.37.7.105:5060 SIP/2.0 Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK0b3a1ad312b94bd3bd2e0692ebbb60a1 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: ;tag=as76833e47 Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@77.72.169.129 CSeq: 1 BYE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 77.72.169.129:5060 (NAT) Scheduling destruction of SIP dialog '80f0a25e8fc24bfb923fd4233210e12e@77.72.169.129' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK0b3a1ad312b94bd3bd2e0692ebbb60a1;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c5736e To: ;tag=as76833e47 Call-ID: 80f0a25e8fc24bfb923fd4233210e12e@77.72.169.129 CSeq: 1 BYE 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 <------------> -- Executing [h@macro-dial-one:1] Macro("SIP/in12voip-00000000", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/in12voip-00000000", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/in12voip-00000000", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/in12voip-00000000", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("SIP/in12voip-00000000", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("SIP/in12voip-00000000", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("SIP/in12voip-00000000", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("SIP/in12voip-00000000", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("SIP/in12voip-00000000", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("SIP/in12voip-00000000", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("SIP/in12voip-00000000", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("SIP/in12voip-00000000", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("SIP/in12voip-00000000", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("SIP/in12voip-00000000", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/in12voip-00000000' in macro 'hangupcall' == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/in12voip-00000000' Scheduling destruction of SIP dialog '7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.2:16410 Reliably Transmitting (NAT) to 192.168.0.2:16410: BYE sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK2f01cb2c;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as4e4830ec To: ;tag=3758686a Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 103 BYE User-Agent: FPBX-2.8.1(1.8.15.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/in12voip-00000000' in macro 'dial-one' == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/in12voip-00000000' in macro 'exten-vm' == Spawn extension (from-did-direct, 105, 1) exited non-zero on 'SIP/in12voip-00000000' Really destroying SIP dialog '80f0a25e8fc24bfb923fd4233210e12e@77.72.169.129' Method: BYE Retransmitting #1 (NAT) to 192.168.0.2:16410: BYE sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK2f01cb2c;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as4e4830ec To: ;tag=3758686a Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 103 BYE User-Agent: FPBX-2.8.1(1.8.15.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK2f01cb2c;rport=5060 Contact: To: ;tag=3758686a From: "04XXXXX11";tag=as4e4830ec Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 103 BYE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060' Method: INVITE <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK2f01cb2c;rport=5060 Contact: To: ;tag=3758686a From: "04XXXXX11";tag=as4e4830ec Call-ID: 7c751b5a42e6a4446fc00f202233379e@192.168.0.11:5060 CSeq: 103 BYE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.0.150:5062 ---> <-------------> Really destroying SIP dialog '7bab575c3f8ef01737849c1d5f086f70@38.126.208.166' Method: REGISTER <--- SIP read from UDP:192.168.0.2:16410 ---> <-------------> <--- 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=z9hG4bK74076a2c75334f0fa21094c0985b246a From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 315 v=0 o=axxxxx6 1354991930 1354991930 IN IP4 62.41.83.113 s=SIP Call c=IN IP4 62.41.83.113 t=0 0 m=audio 11758 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 - 0c0e1733cf874cd29955876d97333ab6@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 62.41.83.113:11758 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=z9hG4bK74076a2c75334f0fa21094c0985b246a;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 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-00000002", "0?noplusatstart") in new stack -- Executing [77XXXXX1@from-12v:2] NoOp("SIP/in12voip-00000002", "Changing Caller ID number from +04XXXXX11 to 04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:3] NoOp("SIP/in12voip-00000002", "Changing Caller ID name from +04XXXXX11 to 04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:4] Set("SIP/in12voip-00000002", "CALLERID(num)=04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:5] Set("SIP/in12voip-00000002", "CALLERID(name)=04XXXXX11") in new stack -- Executing [77XXXXX1@from-12v:6] NoOp("SIP/in12voip-00000002", "Changing Caller ID name from "04XXXXX11" <04XXXXX11> to 04XXXXX11" <04XXXXX11>") in new stack -- Executing [77XXXXX1@from-12v:7] Goto("SIP/in12voip-00000002", "from-trunk,77XXXXX1,1") in new stack -- Goto (from-trunk,77XXXXX1,1) -- Executing [77XXXXX1@from-trunk:1] Set("SIP/in12voip-00000002", "__FROM_DID=77XXXXX1") in new stack -- Executing [77XXXXX1@from-trunk:2] Gosub("SIP/in12voip-00000002", "app-blacklist-check,s,1") in new stack -- Executing [s@app-blacklist-check:1] GotoIf("SIP/in12voip-00000002", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:2] Set("SIP/in12voip-00000002", "CALLED_BLACKLIST=1") in new stack -- Executing [s@app-blacklist-check:3] Return("SIP/in12voip-00000002", "") in new stack -- Executing [77XXXXX1@from-trunk:3] ExecIf("SIP/in12voip-00000002", "0 ?Set(CALLERID(name)=04XXXXX11)") in new stack -- Executing [77XXXXX1@from-trunk:4] Set("SIP/in12voip-00000002", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [77XXXXX1@from-trunk:5] Set("SIP/in12voip-00000002", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [77XXXXX1@from-trunk:6] Goto("SIP/in12voip-00000002", "from-did-direct,105,1") in new stack -- Goto (from-did-direct,105,1) -- Executing [105@from-did-direct:1] Macro("SIP/in12voip-00000002", "exten-vm,novm,105") in new stack -- Executing [s@macro-exten-vm:1] Macro("SIP/in12voip-00000002", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/in12voip-00000002", "AMPUSER=04XXXXX11") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/in12voip-00000002", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/in12voip-00000002", "1?Set(REALCALLERIDNUM=04XXXXX11)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/in12voip-00000002", "AMPUSER=") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/in12voip-00000002", "AMPUSERCIDNAME=") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/in12voip-00000002", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("SIP/in12voip-00000002", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/in12voip-00000002", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("SIP/in12voip-00000002", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/in12voip-00000002", "CALLERID(number)=04XXXXX11") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/in12voip-00000002", "CALLERID(name)=04XXXXX11") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/in12voip-00000002", "Using CallerID "04XXXXX11" <04XXXXX11>") in new stack -- Executing [s@macro-exten-vm:2] Set("SIP/in12voip-00000002", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:3] Set("SIP/in12voip-00000002", "VMBOX=novm") in new stack -- Executing [s@macro-exten-vm:4] Set("SIP/in12voip-00000002", "__EXTTOCALL=105") in new stack -- Executing [s@macro-exten-vm:5] Set("SIP/in12voip-00000002", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:6] Set("SIP/in12voip-00000002", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:7] Set("SIP/in12voip-00000002", "RT=""") in new stack -- Executing [s@macro-exten-vm:8] Macro("SIP/in12voip-00000002", "record-enable,105,IN") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/in12voip-00000002", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/in12voip-00000002", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/in12voip-00000002", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/in12voip-00000002", "1?IN") in new stack -- Goto (macro-record-enable,s,20) -- Executing [s@macro-record-enable:20] ExecIf("SIP/in12voip-00000002", "1?MacroExit()") in new stack -- Executing [s@macro-exten-vm:9] Macro("SIP/in12voip-00000002", "dial-one,"",tr,105") in new stack -- Executing [s@macro-dial-one:1] Set("SIP/in12voip-00000002", "DEXTEN=105") in new stack -- Executing [s@macro-dial-one:2] Set("SIP/in12voip-00000002", "DIALSTATUS_CW=") in new stack -- Executing [s@macro-dial-one:3] GosubIf("SIP/in12voip-00000002", "0?screen,1") in new stack -- Executing [s@macro-dial-one:4] GosubIf("SIP/in12voip-00000002", "0?cf,1") in new stack -- Executing [s@macro-dial-one:5] GotoIf("SIP/in12voip-00000002", "1?skip1") in new stack -- Goto (macro-dial-one,s,8) -- Executing [s@macro-dial-one:8] GotoIf("SIP/in12voip-00000002", "0?nodial") in new stack -- Executing [s@macro-dial-one:9] GotoIf("SIP/in12voip-00000002", "0?continue") in new stack -- Executing [s@macro-dial-one:10] Set("SIP/in12voip-00000002", "EXTHASCW=") in new stack -- Executing [s@macro-dial-one:11] GotoIf("SIP/in12voip-00000002", "1?next1:cwinusebusy") in new stack -- Goto (macro-dial-one,s,12) -- Executing [s@macro-dial-one:12] GotoIf("SIP/in12voip-00000002", "0?docfu:skip3") in new stack -- Goto (macro-dial-one,s,16) -- Executing [s@macro-dial-one:16] GotoIf("SIP/in12voip-00000002", "1?next2:continue") in new stack -- Goto (macro-dial-one,s,17) -- Executing [s@macro-dial-one:17] GotoIf("SIP/in12voip-00000002", "1?continue") in new stack -- Goto (macro-dial-one,s,25) -- Executing [s@macro-dial-one:25] GotoIf("SIP/in12voip-00000002", "0?nodial") in new stack -- Executing [s@macro-dial-one:26] GosubIf("SIP/in12voip-00000002", "1?dstring,1:dlocal,1") in new stack -- Executing [dstring@macro-dial-one:1] Set("SIP/in12voip-00000002", "DSTRING=") in new stack -- Executing [dstring@macro-dial-one:2] Set("SIP/in12voip-00000002", "DEVICES=105") in new stack -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/in12voip-00000002", "0?Return()") in new stack -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/in12voip-00000002", "0?Set(DEVICES=05)") in new stack -- Executing [dstring@macro-dial-one:5] Set("SIP/in12voip-00000002", "LOOPCNT=1") in new stack -- Executing [dstring@macro-dial-one:6] Set("SIP/in12voip-00000002", "ITER=1") in new stack -- Executing [dstring@macro-dial-one:7] Set("SIP/in12voip-00000002", "THISDIAL=SIP/105") in new stack -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/in12voip-00000002", "1?zap2dahdi,1") in new stack -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/in12voip-00000002", "0?Return()") in new stack -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/in12voip-00000002", "NEWDIAL=") in new stack -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/in12voip-00000002", "LOOPCNT2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/in12voip-00000002", "ITER2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/in12voip-00000002", "THISPART2=SIP/105") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/in12voip-00000002", "0?Set(THISPART2=DAHDI/105)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/in12voip-00000002", "NEWDIAL=SIP/105&") in new stack -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/in12voip-00000002", "ITER2=2") in new stack -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/in12voip-00000002", "0?begin2") in new stack -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/in12voip-00000002", "THISDIAL=SIP/105") in new stack -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/in12voip-00000002", "") in new stack -- Executing [dstring@macro-dial-one:9] Set("SIP/in12voip-00000002", "DSTRING=SIP/105&") in new stack -- Executing [dstring@macro-dial-one:10] Set("SIP/in12voip-00000002", "ITER=2") in new stack -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/in12voip-00000002", "0?begin") in new stack -- Executing [dstring@macro-dial-one:12] Set("SIP/in12voip-00000002", "DSTRING=SIP/105") in new stack -- Executing [dstring@macro-dial-one:13] Return("SIP/in12voip-00000002", "") in new stack -- Executing [s@macro-dial-one:27] GotoIf("SIP/in12voip-00000002", "0?nodial") in new stack -- Executing [s@macro-dial-one:28] GotoIf("SIP/in12voip-00000002", "0?skiptrace") in new stack -- Executing [s@macro-dial-one:29] GosubIf("SIP/in12voip-00000002", "1?ctset,1:ctclear,1") in new stack -- Executing [ctset@macro-dial-one:1] Set("SIP/in12voip-00000002", "DB(CALLTRACE/105)=04XXXXX11") in new stack -- Executing [ctset@macro-dial-one:2] Return("SIP/in12voip-00000002", "") in new stack -- Executing [s@macro-dial-one:30] Set("SIP/in12voip-00000002", "D_OPTIONS=tr") in new stack -- Executing [s@macro-dial-one:31] ExecIf("SIP/in12voip-00000002", "0?SIPAddHeader(Alert-Info: )") in new stack -- Executing [s@macro-dial-one:32] ExecIf("SIP/in12voip-00000002", "0?SIPAddHeader()") in new stack -- Executing [s@macro-dial-one:33] ExecIf("SIP/in12voip-00000002", "0?Set(CHANNEL(musicclass)=)") in new stack -- Executing [s@macro-dial-one:34] GosubIf("SIP/in12voip-00000002", "0?qwait,1") in new stack -- Executing [s@macro-dial-one:35] Set("SIP/in12voip-00000002", "__CWIGNORE=") in new stack -- Executing [s@macro-dial-one:36] Set("SIP/in12voip-00000002", "__KEEPCID=TRUE") in new stack -- Executing [s@macro-dial-one:37] Dial("SIP/in12voip-00000002", "SIP/105,"",tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 16404 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 192.168.0.2:16410: INVITE sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as7baec6d7 To: Contact: Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.15.0) Date: Sat, 08 Dec 2012 18:38:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 296923846 296923846 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 16404 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/105 <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK74076a2c75334f0fa21094c0985b246a;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: ;tag=as5c3952f4 Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 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 <------------> Retransmitting #1 (NAT) to 192.168.0.2:16410: INVITE sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as7baec6d7 To: Contact: Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.15.0) Date: Sat, 08 Dec 2012 18:38:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 296923846 296923846 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 16404 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 --- <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport=5060 Contact: To: ;tag=87106c3d From: "04XXXXX11";tag=as7baec6d7 Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- list_route: hop: <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport=5060 Contact: To: ;tag=87106c3d From: "04XXXXX11";tag=as7baec6d7 Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- list_route: hop: -- SIP/105-00000003 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=z9hG4bK74076a2c75334f0fa21094c0985b246a;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: ;tag=as5c3952f4 Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 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/105-00000003 is ringing <--- 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=z9hG4bK74076a2c75334f0fa21094c0985b246a From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: ;tag=as5c3952f4 Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 CANCEL Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 77.72.169.129:5060 (NAT) <--- Reliably Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK74076a2c75334f0fa21094c0985b246a;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: ;tag=as5c3952f4 Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 INVITE 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 <------------> <--- Transmitting (NAT) to 77.72.169.129:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK74076a2c75334f0fa21094c0985b246a;received=77.72.169.129;rport=5060 From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: ;tag=as5c3952f4 Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 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 <------------> Scheduling destruction of SIP dialog '676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 192.168.0.2:16410: CANCEL sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as7baec6d7 To: Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 CANCEL User-Agent: FPBX-2.8.1(1.8.15.0) Content-Length: 0 --- Scheduling destruction of SIP dialog '676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060' in 6400 ms (Method: INVITE) == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/in12voip-00000002' in macro 'dial-one' == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/in12voip-00000002' in macro 'exten-vm' == Spawn extension (from-did-direct, 105, 1) exited non-zero on 'SIP/in12voip-00000002' -- Executing [h@from-did-direct:1] Macro("SIP/in12voip-00000002", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/in12voip-00000002", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/in12voip-00000002", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/in12voip-00000002", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("SIP/in12voip-00000002", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("SIP/in12voip-00000002", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("SIP/in12voip-00000002", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("SIP/in12voip-00000002", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("SIP/in12voip-00000002", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("SIP/in12voip-00000002", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("SIP/in12voip-00000002", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("SIP/in12voip-00000002", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("SIP/in12voip-00000002", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi <--- 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=z9hG4bKafd3794f0d0a471ca34c988330ba147d From: "+04XXXXX11" ;tag=120313ac506904dd1c573d5 To: ;tag=as5c3952f4 Contact: sip:+04XXXXX11@77.72.169.129:5060 Call-ID: 0c0e1733cf874cd29955876d97333ab6@77.72.169.129 CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("SIP/in12voip-00000002", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/in12voip-00000002' in macro 'hangupcall' == Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/in12voip-00000002' Retransmitting #1 (NAT) to 192.168.0.2:16410: CANCEL sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as7baec6d7 To: Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 CANCEL User-Agent: FPBX-2.8.1(1.8.15.0) Content-Length: 0 --- Really destroying SIP dialog '0c0e1733cf874cd29955876d97333ab6@77.72.169.129' Method: ACK <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport=5060 Contact: To: ;tag=87106c3d From: "04XXXXX11";tag=as7baec6d7 Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 CANCEL User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport=5060 To: ;tag=87106c3d From: "04XXXXX11";tag=as7baec6d7 Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 192.168.0.2:16410: ACK sip:105@192.168.0.2:16410;rinstance=7065c19b8e16ded1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport Max-Forwards: 70 From: "04XXXXX11" ;tag=as7baec6d7 To: ;tag=87106c3d Contact: Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.8.15.0) Content-Length: 0 --- Scheduling destruction of SIP dialog '676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.0.2:16410 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK479e8478;rport=5060 To: ;tag=b2000514 From: "04XXXXX11";tag=as7baec6d7 Call-ID: 676c02b873f45be00a1be196678e7f2b@192.168.0.11:5060 CSeq: 102 CANCEL Content-Length: 0 <-------------> --- (7 headers 0 lines) --- elastix*CLI> exit