trixbox1*CLI> sip set debug on SIP Debugging enabled trixbox1*CLI> <--- SIP read from UDP://192.168.254.5:5061 ---> INVITE sip:02163847880@192.168.254.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.5:5061;branch=z9hG4bK-a8583058 From: PSTN ;tag=ea44ff2fa77dff40o1 To: Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 101 INVITE Max-Forwards: 70 Contact: PSTN Expires: 240 User-Agent: Linksys/SPA3000-3.1.10(GWd) Content-Length: 446 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 9493028 9493028 IN IP4 192.168.254.5 s=- c=IN IP4 192.168.254.5 t=0 0 m=audio 16430 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (14 headers 20 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 Sending to 192.168.254.5 : 5061 (no NAT) Using INVITE request as basis request - 8d91451b-1ec69244@192.168.254.5 No user '192.168.254.254' in SIP users list Found peer 'pstn sip' for '192.168.254.254' from 192.168.254.5:5061 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.5:16430 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Found audio description format G729a for ID 18 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/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 192.168.254.5:16430 Peer video RTP is at port 192.168.254.5:12853 Looking for 02163847880 in from-pstn (domain 192.168.254.254) list_route: hop: trixbox1*CLI> <--- Transmitting (no NAT) to 192.168.254.5:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.5:5061;branch=z9hG4bK-a8583058;received=192.168.254.5 From: PSTN ;tag=ea44ff2fa77dff40o1 To: Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [02163847880@from-pstn:1] Set("SIP/PSTN-b5913a00", "__FROM_DID=02163847880") in new stack -- Executing [02163847880@from-pstn:2] ExecIf("SIP/PSTN-b5913a00", "0 ?Set(CALLERID(name)=192168254254)") in new stack -- Executing [02163847880@from-pstn:3] Set("SIP/PSTN-b5913a00", "FAX_RX_EMAIL=sorcerer@sorcerer.dyndns.org") in new stack -- Executing [02163847880@from-pstn:4] Set("SIP/PSTN-b5913a00", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [02163847880@from-pstn:5] Set("SIP/PSTN-b5913a00", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [02163847880@from-pstn:6] Goto("SIP/PSTN-b5913a00", "ext-group,600,1") in new stack -- Goto (ext-group,600,1) -- Executing [600@ext-group:1] Macro("SIP/PSTN-b5913a00", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/PSTN-b5913a00", "AMPUSER=192168254254") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/PSTN-b5913a00", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/PSTN-b5913a00", "1?Set(REALCALLERIDNUM=192168254254)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/PSTN-b5913a00", "AMPUSER=") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/PSTN-b5913a00", "AMPUSERCIDNAME=") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/PSTN-b5913a00", "1?report") in new stack -- Goto (macro-user-callerid,s,11) -- Executing [s@macro-user-callerid:11] GotoIf("SIP/PSTN-b5913a00", "0?continue") in new stack -- Executing [s@macro-user-callerid:12] Set("SIP/PSTN-b5913a00", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:13] GotoIf("SIP/PSTN-b5913a00", "1?continue") in new stack -- Goto (macro-user-callerid,s,20) -- Executing [s@macro-user-callerid:20] NoOp("SIP/PSTN-b5913a00", "Using CallerID "PSTN" <192168254254>") in new stack -- Executing [600@ext-group:2] GotoIf("SIP/PSTN-b5913a00", "1?skipdb") in new stack -- Goto (ext-group,600,4) -- Executing [600@ext-group:4] Set("SIP/PSTN-b5913a00", "__NODEST=") in new stack -- Executing [600@ext-group:5] Set("SIP/PSTN-b5913a00", "__BLKVM_OVERRIDE=BLKVM/600/SIP/PSTN-b5913a00") in new stack -- Executing [600@ext-group:6] Set("SIP/PSTN-b5913a00", "__BLKVM_BASE=600") in new stack -- Executing [600@ext-group:7] Set("SIP/PSTN-b5913a00", "DB(BLKVM/600/SIP/PSTN-b5913a00)=TRUE") in new stack -- Executing [600@ext-group:8] Set("SIP/PSTN-b5913a00", "RRNODEST=") in new stack -- Executing [600@ext-group:9] Set("SIP/PSTN-b5913a00", "__NODEST=600") in new stack -- Executing [600@ext-group:10] Set("SIP/PSTN-b5913a00", "RecordMethod=Group") in new stack -- Executing [600@ext-group:11] Macro("SIP/PSTN-b5913a00", "record-enable,999-993,Group") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/PSTN-b5913a00", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI("SIP/PSTN-b5913a00", "recordingcheck,20091015-221532,1255634131.87") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck,20091015-221532,1255634131.87: Recording enable for 999 recordingcheck,20091015-221532,1255634131.87: CALLFILENAME=g999-20091015-221532-1255634131.87 -- AGI Script recordingcheck completed, returning 0 -- Executing [s@macro-record-enable:999] MixMonitor("SIP/PSTN-b5913a00", "g999-20091015-221532-1255634131.87.wav,,") in new stack == Begin MixMonitor Recording SIP/PSTN-b5913a00 -- Executing [600@ext-group:12] Set("SIP/PSTN-b5913a00", "RingGroupMethod=ringall") in new stack -- Executing [600@ext-group:13] Macro("SIP/PSTN-b5913a00", "dial,20,tr,999-993") in new stack -- Executing [s@macro-dial:1] GotoIf("SIP/PSTN-b5913a00", "1?dial") in new stack -- Goto (macro-dial,s,3) -- Executing [s@macro-dial:3] AGI("SIP/PSTN-b5913a00", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: Caller ID name is 'PSTN' number is '192168254254' > dialparties.agi: USE_CONFIRMATION: 'FALSE' > dialparties.agi: RINGGROUP_INDEX: '' dialparties.agi: Methodology of ring is 'ringall' -- dialparties.agi: Added extension 999 to extension map -- dialparties.agi: Added extension 993 to extension map -- dialparties.agi: Extension 999 cf is disabled -- dialparties.agi: Extension 993 cf is disabled -- dialparties.agi: Extension 999 do not disturb is disabled -- dialparties.agi: Extension 993 do not disturb is disabled > dialparties.agi: extnum 999 has: cw: 1; hascfb: 0 [] hascfu: 0 [] -- dialparties.agi: dbset CALLTRACE/999 to 192168254254 > dialparties.agi: extnum 993 has: cw: 1; hascfb: 0 [] hascfu: 0 [] -- dialparties.agi: dbset CALLTRACE/993 to 192168254254 -- dialparties.agi: Filtered ARG3: 999-993 > dialparties.agi: NODEST: 600 adding M(auto-blkvm) to dialopts: trM(auto-blkvm) > dialparties.agi: NODEST: 600 blkvm enabled macro already in dialopts: trM(auto-blkvm) == Manager 'admin' logged off from 127.0.0.1 -- AGI Script dialparties.agi completed, returning 0 -- Executing [s@macro-dial:7] Dial("SIP/PSTN-b5913a00", "SIP/999&SIP/993,20,trM(auto-blkvm)") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 Audio is at 192.168.254.254 port 32378 Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.254.11:5060: INVITE sip:999@192.168.254.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK5f80c936;rport Max-Forwards: 70 From: "PSTN" ;tag=as0fb399fc To: Contact: Call-ID: 2b09262a1b6db8d775b0944108043b82@192.168.254.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Date: Thu, 15 Oct 2009 19:15:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 224 v=0 o=root 205746368 205746368 IN IP4 192.168.254.254 s=Asterisk PBX 1.6.0.10-FONCORE-r40 c=IN IP4 192.168.254.254 t=0 0 m=audio 32378 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 999 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 trixbox1*CLI> <--- SIP read from UDP://192.168.254.11:5060 ---> SIP/2.0 100 Trying To: From: "PSTN" ;tag=as0fb399fc Call-ID: 2b09262a1b6db8d775b0944108043b82@192.168.254.254 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK5f80c936 Server: Linksys/PAP2-3.1.22(LS) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '0d3d613b0d3fc27e7eff80375e4be427@127.0.0.1' Method: INVITE trixbox1*CLI> <--- Transmitting (no NAT) to 192.168.254.5:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.5:5061;branch=z9hG4bK-a8583058;received=192.168.254.5 From: PSTN ;tag=ea44ff2fa77dff40o1 To: ;tag=as72b53622 Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> trixbox1*CLI> <--- SIP read from UDP://192.168.254.11:5060 ---> SIP/2.0 180 Ringing To: ;tag=78c081425c24a026i0 From: "PSTN" ;tag=as0fb399fc Call-ID: 2b09262a1b6db8d775b0944108043b82@192.168.254.254 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK5f80c936 Server: Linksys/PAP2-3.1.22(LS) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/999-084ba070 is ringing trixbox1*CLI> <--- Transmitting (no NAT) to 192.168.254.5:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.5:5061;branch=z9hG4bK-a8583058;received=192.168.254.5 From: PSTN ;tag=ea44ff2fa77dff40o1 To: ;tag=as72b53622 Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.254.11:5060: OPTIONS sip:999@192.168.254.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK49b66fd7;rport Max-Forwards: 70 From: "Unknown" ;tag=as079d971e To: Contact: Call-ID: 46bfa6b92c04596c6e9bb3be74bb2861@192.168.254.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Date: Thu, 15 Oct 2009 19:15:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.254.5:5060: OPTIONS sip:995@192.168.254.5:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK7b29e231;rport Max-Forwards: 70 From: "Unknown" ;tag=as444fcb29 To: Contact: Call-ID: 1a4d9ddb6845600e608132682b41cbda@192.168.254.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Date: Thu, 15 Oct 2009 19:15:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- trixbox1*CLI> <--- SIP read from UDP://192.168.254.11:5060 ---> SIP/2.0 486 Busy Here To: ;tag=9c61311b75c6bcaai0 From: "Unknown" ;tag=as079d971e Call-ID: 46bfa6b92c04596c6e9bb3be74bb2861@192.168.254.254 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK49b66fd7 Server: Linksys/PAP2-3.1.22(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '46bfa6b92c04596c6e9bb3be74bb2861@192.168.254.254' Method: OPTIONS trixbox1*CLI> <--- SIP read from UDP://192.168.254.5:5060 ---> SIP/2.0 200 OK To: ;tag=72bb98ea8517600ai0 From: "Unknown" ;tag=as444fcb29 Call-ID: 1a4d9ddb6845600e608132682b41cbda@192.168.254.254 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK7b29e231;rport=5060 Server: Linksys/SPA3000-3.1.10(GWd) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1a4d9ddb6845600e608132682b41cbda@192.168.254.254' Method: OPTIONS trixbox1*CLI> <--- SIP read from UDP://192.168.254.11:5060 ---> SIP/2.0 200 OK To: ;tag=78c081425c24a026i0 From: "PSTN" ;tag=as0fb399fc Call-ID: 2b09262a1b6db8d775b0944108043b82@192.168.254.254 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK5f80c936 Contact: ilker Server: Linksys/PAP2-3.1.22(LS) Content-Length: 259 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 9489921 9489921 IN IP4 192.168.254.11 s=- c=IN IP4 192.168.254.11 t=0 0 m=audio 16396 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (12 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.11:16396 Found audio description format PCMA for ID 8 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.254.11:16396 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.11, port 5060 Transmitting (no NAT) to 192.168.254.11:5060: ACK sip:999@192.168.254.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK1ffdc2db;rport Max-Forwards: 70 From: "PSTN" ;tag=as0fb399fc To: ;tag=78c081425c24a026i0 Contact: Call-ID: 2b09262a1b6db8d775b0944108043b82@192.168.254.254 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Content-Length: 0 --- -- SIP/999-084ba070 answered SIP/PSTN-b5913a00 -- Executing [s@macro-auto-blkvm:1] Set("SIP/999-084ba070", "__MACRO_RESULT=") in new stack -- Executing [s@macro-auto-blkvm:2] DBdel("SIP/999-084ba070", "BLKVM/600/SIP/PSTN-b5913a00") in new stack -- DBdel: family=BLKVM, key=600/SIP/PSTN-b5913a00 Audio is at 192.168.254.254 port 33374 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.254.5:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.5:5061;branch=z9hG4bK-a8583058;received=192.168.254.5 From: PSTN ;tag=ea44ff2fa77dff40o1 To: ;tag=as72b53622 Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 306 v=0 o=root 1293438424 1293438424 IN IP4 192.168.254.254 s=Asterisk PBX 1.6.0.10-FONCORE-r40 c=IN IP4 192.168.254.254 t=0 0 m=audio 33374 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> trixbox1*CLI> <--- SIP read from UDP://192.168.254.5:5061 ---> ACK sip:02163847880@192.168.254.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.5:5061;branch=z9hG4bK-3717dc53 From: PSTN ;tag=ea44ff2fa77dff40o1 To: ;tag=as72b53622 Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 101 ACK Max-Forwards: 70 Contact: PSTN User-Agent: Linksys/SPA3000-3.1.10(GWd) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- trixbox1*CLI> <--- SIP read from UDP://192.168.254.11:5060 ---> BYE sip:192168254254@192.168.254.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-4343f894 From: ;tag=78c081425c24a026i0 To: "PSTN" ;tag=as0fb399fc Call-ID: 2b09262a1b6db8d775b0944108043b82@192.168.254.254 CSeq: 101 BYE Max-Forwards: 70 User-Agent: Linksys/PAP2-3.1.22(LS) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.254.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.254.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.11:5060;branch=z9hG4bK-4343f894;received=192.168.254.11 From: ;tag=78c081425c24a026i0 To: "PSTN" ;tag=as0fb399fc Call-ID: 2b09262a1b6db8d775b0944108043b82@192.168.254.254 CSeq: 101 BYE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@macro-dial:1] Macro("SIP/PSTN-b5913a00", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/PSTN-b5913a00", "w") in new stack -- Executing [s@macro-hangupcall:2] NoCDR("SIP/PSTN-b5913a00", "") in new stack -- Executing [s@macro-hangupcall:3] GotoIf("SIP/PSTN-b5913a00", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [s@macro-hangupcall:6] GotoIf("SIP/PSTN-b5913a00", "0?skipblkvm") in new stack -- Executing [s@macro-hangupcall:7] NoOp("SIP/PSTN-b5913a00", "Cleaning Up Block VM Flag: BLKVM/600/SIP/PSTN-b5913a00") in new stack -- Executing [s@macro-hangupcall:8] DBdel("SIP/PSTN-b5913a00", "BLKVM/600/SIP/PSTN-b5913a00") in new stack -- DBdel: family=BLKVM, key=600/SIP/PSTN-b5913a00 -- DBdel: Error deleting key from database. -- Executing [s@macro-hangupcall:9] GotoIf("SIP/PSTN-b5913a00", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing [s@macro-hangupcall:11] Hangup("SIP/PSTN-b5913a00", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/PSTN-b5913a00' in macro 'hangupcall' == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/PSTN-b5913a00' == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/PSTN-b5913a00' in macro 'dial' == Spawn extension (ext-group, 600, 13) exited non-zero on 'SIP/PSTN-b5913a00' == End MixMonitor Recording SIP/PSTN-b5913a00 Scheduling destruction of SIP dialog '8d91451b-1ec69244@192.168.254.5' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.5, port 5061 Reliably Transmitting (no NAT) to 192.168.254.5:5061: BYE sip:192.168.254.5:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK4f788983;rport Max-Forwards: 70 From: ;tag=as72b53622 To: PSTN ;tag=ea44ff2fa77dff40o1 Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- trixbox1*CLI> <--- SIP read from UDP://192.168.254.5:5061 ---> SIP/2.0 200 OK To: PSTN ;tag=ea44ff2fa77dff40o1 From: ;tag=as72b53622 Call-ID: 8d91451b-1ec69244@192.168.254.5 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.254.254:5060;branch=z9hG4bK4f788983;rport=5060 Server: Linksys/SPA3000-3.1.10(GWd) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '2b09262a1b6db8d775b0944108043b82@192.168.254.254' Method: BYE Really destroying SIP dialog '8d91451b-1ec69244@192.168.254.5' Method: ACK trixbox1*CLI>