asterisk1*CLI> sip debug SIP Debugging enabled asterisk1*CLI> <-- SIP read from 192.168.0.104:47681: INVITE sip:7000@192.168.0.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-9b318f2b053afa50-1--d87543-;rport Max-Forwards: 70 Contact: To: "7000" From: "Doogies Extension";tag=36272437 Call-ID: cf6b000fc22b4c24@dm95YWdlcg.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeamLicensed release 9999a stamp 25770 Content-Length: 436 v=0 o=- 211292324 2 IN IP4 192.168.0.104 s= c=IN IP4 192.168.0.104 t=0 0 m=audio 47682 RTP/AVP 107 100 106 6 0 105 8 3 4 5 101 a=alt:1 1 : sojasovc syysekfa 192.168.0.104 47682 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:04DEEEC8A2D7463BB9BADE665BF795BB --- (13 headers 15 lines) --- Using INVITE request as basis request - cf6b000fc22b4c24@dm95YWdlcg.. Sending to 192.168.0.104 : 47681 (NAT) Reliably Transmitting (NAT) to 192.168.0.104:47681: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-9b318f2b053afa50-1--d87543-;received=192.168.0.104;rport=47681 From: "Doogies Extension";tag=36272437 To: "7000";tag=as6ee7be5a Call-ID: cf6b000fc22b4c24@dm95YWdlcg.. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78e03a35" Content-Length: 0 --- Scheduling destruction of call 'cf6b000fc22b4c24@dm95YWdlcg..' in 15000 ms Found user '2200' <-- SIP read from 192.168.0.104:47681: ACK sip:7000@192.168.0.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-9b318f2b053afa50-1--d87543-;rport To: "7000";tag=as6ee7be5a From: "Doogies Extension";tag=36272437 Call-ID: cf6b000fc22b4c24@dm95YWdlcg.. CSeq: 1 ACK Content-Length: 0 --- (7 headers 0 lines) --- asterisk1*CLI> <-- SIP read from 192.168.0.104:47681: INVITE sip:7000@192.168.0.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-b16a0724d1177555-1--d87543-;rport Max-Forwards: 70 Contact: To: "7000" From: "Doogies Extension";tag=36272437 Call-ID: cf6b000fc22b4c24@dm95YWdlcg.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="2200",realm="asterisk",nonce="78e03a35",uri="sip:7000@192.168.0.103",response="d127683acae9ddee834512ed118d7b15",algorithm=MD5 Supported: eventlist User-Agent: eyeBeamLicensed release 9999a stamp 25770 Content-Length: 436 v=0 o=- 211292324 2 IN IP4 192.168.0.104 s= c=IN IP4 192.168.0.104 t=0 0 m=audio 47682 RTP/AVP 107 100 106 6 0 105 8 3 4 5 101 a=alt:1 1 : sojasovc syysekfa 192.168.0.104 47682 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:04DEEEC8A2D7463BB9BADE665BF795BB --- (14 headers 15 lines) --- Using INVITE request as basis request - cf6b000fc22b4c24@dm95YWdlcg.. Sending to 192.168.0.104 : 47681 (NAT) Found user '2200' Found RTP audio format 107 Found RTP audio format 100 Found RTP audio format 106 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 105 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 4 Found RTP audio format 5 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.104:47682 Found description format BV32 Found description format SPEEX Found description format SPEEX-FEC Found description format SPEEX-FEC Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x22f (g723|gsm|ulaw|alaw|adpcm|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 7000 in from-internal (domain 192.168.0.103) list_route: hop: Transmitting (NAT) to 192.168.0.104:47681: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-b16a0724d1177555-1--d87543-;received=192.168.0.104;rport=47681 From: "Doogies Extension";tag=36272437 To: "7000" Call-ID: cf6b000fc22b4c24@dm95YWdlcg.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/2200-08757218", "exten-vm|7000|7000") in new stack -- Executing Macro("SIP/2200-08757218", "user-callerid") in new stack -- Executing NoOp("SIP/2200-08757218", "user-callerid: device 2200") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?start") in new stack -- Executing Set("SIP/2200-08757218", "REALCALLERIDNUM=2200") in new stack -- Executing NoOp("SIP/2200-08757218", "REALCALLERIDNUM is 2200") in new stack -- Executing Set("SIP/2200-08757218", "AMPUSER=2200") in new stack -- Executing Set("SIP/2200-08757218", "AMPUSERCIDNAME=Doogie PC") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack -- Executing Set("SIP/2200-08757218", "CALLERID(all)=Doogie PC <2200>") in new stack -- Executing Set("SIP/2200-08757218", "REALCALLERIDNUM=2200") in new stack -- Executing NoOp("SIP/2200-08757218", "TTL: ARG1: 7000") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?continue") in new stack -- Executing Set("SIP/2200-08757218", "_TTL=64") in new stack -- Executing GotoIf("SIP/2200-08757218", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/2200-08757218", "Using CallerID "Doogie PC" <2200>") in new stack -- Executing Set("SIP/2200-08757218", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/2200-08757218", "VMBOX=7000") in new stack -- Executing Set("SIP/2200-08757218", "EXTTOCALL=7000") in new stack -- Executing Set("SIP/2200-08757218", "CFUEXT=") in new stack -- Executing Set("SIP/2200-08757218", "RT=17") in new stack -- Executing Macro("SIP/2200-08757218", "record-enable|7000|IN") in new stack -- Executing GotoIf("SIP/2200-08757218", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/2200-08757218", "recordingcheck|20070406-092448|1175865888.732") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070406-092448|1175865888.732: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/2200-08757218", "No recording needed") in new stack -- Executing Macro("SIP/2200-08757218", "dial|17|tr|7000") in new stack -- Executing DeadAGI("SIP/2200-08757218", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'Doogie PC' number is '2200' dialparties.agi: Methodology of ring is 'none' > dialparties.agi: USE_CONFIRMATION: 'FALSE' > dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 7000 to extension map -- dialparties.agi: Extension 7000 cf is disabled -- dialparties.agi: Extension 7000 do not disturb is disabled > dialparties.agi: extnum: 7000 > dialparties.agi: exthascw: 1 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: -- dialparties.agi: dbset CALLTRACE/7000 to 2200 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/2200-08757218", "SIP/7000|17|tr") in new stack Destroying call '3e9d0e9a1944918352c8fffc15fab152@127.0.0.1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing GosubIf("SIP/2200-08757218", "0?docfu|1") in new stack -- Executing NoOp("SIP/2200-08757218", "Voicemail is 7000") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?s-CHANUNAVAIL|1") in new stack -- Executing NoOp("SIP/2200-08757218", "Sending to Voicemail box 7000") in new stack -- Executing Macro("SIP/2200-08757218", "vm|7000|CHANUNAVAIL") in new stack -- Executing Macro("SIP/2200-08757218", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/2200-08757218", "user-callerid: Doogie PC 2200") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack -- Executing GotoIf("SIP/2200-08757218", "1?start") in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp("SIP/2200-08757218", "REALCALLERIDNUM is 2200") in new stack -- Executing Set("SIP/2200-08757218", "AMPUSER=2200") in new stack -- Executing Set("SIP/2200-08757218", "AMPUSERCIDNAME=Doogie PC") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack -- Executing Set("SIP/2200-08757218", "CALLERID(all)=Doogie PC <2200>") in new stack -- Executing Set("SIP/2200-08757218", "REALCALLERIDNUM=2200") in new stack -- Executing NoOp("SIP/2200-08757218", "TTL: 64 ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/2200-08757218", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/2200-08757218", "Using CallerID "Doogie PC" <2200>") in new stack -- Executing Set("SIP/2200-08757218", "VMGAIN=") in new stack -- Executing GotoIf("SIP/2200-08757218", "1?s-CHANUNAVAIL|1") in new stack -- Goto (macro-vm,s-CHANUNAVAIL,1) -- Executing Macro("SIP/2200-08757218", "get-vmcontext|7000") in new stack -- Executing Set("SIP/2200-08757218", "VMCONTEXT=default") in new stack -- Executing GotoIf("SIP/2200-08757218", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing NoOp("SIP/2200-08757218", "") in new stack -- Executing VoiceMail("SIP/2200-08757218", "7000@default|u") in new stack We're at 192.168.0.103 port 10286 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.104:47681: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-b16a0724d1177555-1--d87543-;received=192.168.0.104;rport=47681 From: "Doogies Extension";tag=36272437 To: "7000";tag=as5171fcf2 Call-ID: cf6b000fc22b4c24@dm95YWdlcg.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 3113 3113 IN IP4 192.168.0.103 s=session c=IN IP4 192.168.0.103 t=0 0 m=audio 10286 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'vm-theperson' (language 'en') asterisk1*CLI> <-- SIP read from 192.168.0.104:47681: ACK sip:7000@192.168.0.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-c55d581b561ce05c-1--d87543-;rport Max-Forwards: 70 Contact: To: "7000";tag=as5171fcf2 From: "Doogies Extension";tag=36272437 Call-ID: cf6b000fc22b4c24@dm95YWdlcg.. CSeq: 2 ACK Proxy-Authorization: Digest username="2200",realm="asterisk",nonce="78e03a35",uri="sip:7000@192.168.0.103",response="d127683acae9ddee834512ed118d7b15",algorithm=MD5 User-Agent: eyeBeamLicensed release 9999a stamp 25770 Content-Length: 0 --- (11 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (NAT) to 86.108.53.127:60987: OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0 Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport From: "Unknown" ;tag=as784a1432 To: Contact: Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') Retransmitting #1 (NAT) to 86.108.53.127:60987: OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0 Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport From: "Unknown" ;tag=as784a1432 To: Contact: Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- Playing 'digits/0' (language 'en') Retransmitting #2 (NAT) to 86.108.53.127:60987: OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0 Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport From: "Unknown" ;tag=as784a1432 To: Contact: Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.112:5060: OPTIONS sip:2000@192.168.0.112:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK6c2cb148;rport From: "Unknown" ;tag=as5de0f7a0 To: Contact: Call-ID: 41ceba507a4736af32c1f31351f5f7c9@192.168.0.103 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.112:5061: OPTIONS sip:2100@192.168.0.112:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1063e71a;rport From: "Unknown" ;tag=as24dca3d4 To: Contact: Call-ID: 5d8317305dfe9de12837ba260240206d@192.168.0.103 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.0.112:5060: SIP/2.0 200 OK To: ;tag=e2d5239726b10c7di0 From: "Unknown" ;tag=as5de0f7a0 Call-ID: 41ceba507a4736af32c1f31351f5f7c9@192.168.0.103 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK6c2cb148 Server: Sipura/SPA2000-3.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Accept-Language: en_us --- (11 headers 0 lines) --- Destroying call '41ceba507a4736af32c1f31351f5f7c9@192.168.0.103' asterisk1*CLI> <-- SIP read from 192.168.0.112:5061: SIP/2.0 200 OK To: ;tag=75105ac74f83bf0di1 From: "Unknown" ;tag=as24dca3d4 Call-ID: 5d8317305dfe9de12837ba260240206d@192.168.0.103 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1063e71a Server: Sipura/SPA2000-3.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Accept-Language: en_us --- (11 headers 0 lines) --- Destroying call '5d8317305dfe9de12837ba260240206d@192.168.0.103' 12 headers, 0 lines Reliably Transmitting (no NAT) to 69.158.200.30:5060: OPTIONS sip:mcstoronto.ca SIP/2.0 Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK44637480;rport From: "Unknown" ;tag=as12c84306 To: Contact: Call-ID: 4fb960cb34344d322244d05b2306eb9c@64.231.189.60 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 69.158.200.30:5060: SIP/2.0 404 Not Found From: "Unknown";tag=as12c84306 To: ;tag=485685578 Call-ID: 4fb960cb34344d322244d05b2306eb9c@64.231.189.60 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.231.189.60:5060;rport=5060;branch=z9hG4bK44637480 contact: supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,com.nortelnetworks.im.encryption Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '4fb960cb34344d322244d05b2306eb9c@64.231.189.60' -- Playing 'digits/0' (language 'en') 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.104:47681: OPTIONS sip:2200@192.168.0.104:47681;rinstance=799a10aa2b7e2adc SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK3e24c4b8;rport From: "Unknown" ;tag=as1af7228e To: Contact: Call-ID: 35f83474080a7e6400eab899525cb58b@192.168.0.103 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.0.104:47681: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK3e24c4b8;rport=5060 Contact: To: ;tag=8602e969 From: "Unknown";tag=as1af7228e Call-ID: 35f83474080a7e6400eab899525cb58b@192.168.0.103 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeamLicensed release 9999a stamp 25770 Content-Length: 0 --- (13 headers 0 lines) --- Destroying call '35f83474080a7e6400eab899525cb58b@192.168.0.103' Retransmitting #3 (NAT) to 86.108.53.127:60987: OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0 Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport From: "Unknown" ;tag=as784a1432 To: Contact: Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- Playing 'vm-isunavail' (language 'en') Retransmitting #4 (NAT) to 86.108.53.127:60987: OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0 Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport From: "Unknown" ;tag=as784a1432 To: Contact: Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Apr 2007 13:24:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '0f03d8a7506ac0e5721e56725265888f@64.231.189.60' asterisk1*CLI> <-- SIP read from 192.168.0.104:47681: --- (0 headers 1 lines) --- -- Playing 'vm-intro' (language 'en') REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 69.158.200.30:5060: REGISTER sip:mcstoronto.ca SIP/2.0 Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK70a65362;rport From: ;tag=as17802292 To: Call-ID: 7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1 CSeq: 109 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="dv008446", realm="Realm", algorithm=MD5, uri="sip:mcstoronto.ca", nonce="MTE3NTg2NTY4MTk1NjJjNGZlNmNhZjJkN2QxZjdmY2FlZTY2YTQ3ZmExNGU2", response="70d0611882545e7009f46bb06ad1663a", opaque="", qop=auth, cnonce="1bad84ac", nc=00000003 Expires: 120 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> sip debug <-- SIP read from 69.158.200.30:5060: SIP/2.0 100 Trying From: ;tag=as17802292 To: Call-ID: 7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1 CSeq: 109 REGISTER Via: SIP/2.0/UDP 64.231.189.60:5060;rport=5060;branch=z9hG4bK70a65362 Content-Length: 0 --- (7 headers 0 lines) --- asterisk1*CLI> sip debug <-- SIP read from 69.158.200.30:5060: SIP/2.0 200 Registration Successful From: "DOUGHAN TURK";tag=as17802292 To: ;tag=1231123177 Call-ID: 7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1 CSeq: 109 REGISTER Via: SIP/2.0/UDP 64.231.189.60:5060;rport=5060;branch=z9hG4bK70a65362 contact: ;expires=109 supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,com.nortelnetworks.im.encryption Content-Length: 0 --- (9 headers 0 lines) --- Scheduling destruction of call '7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1' in 32000 ms asterisk1*CLI> sip no debug SIP Debugging Disabled -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/7000/tmp/z5RN8c format: wav49, 0x86ca178 -- x=1, open writing: /var/spool/asterisk/voicemail/default/7000/tmp/z5RN8c format: wav, 0x86d30d8 -- User hung up == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/2200-08757218' in macro 'vm' == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/2200-08757218' in macro 'exten-vm' == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/2200-08757218'