Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Terry Cole Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="b03af597df8fef523facbb3608d8b8d2",username="200",realm="asterisk",nonce="03a824d1",algorithm=MD5,uri="sip:201@24.176.126.46" User-Agent: Aastra 9112i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 181996649 IN IP4 192.168.200.11 s=SIP Call c=IN IP4 192.168.200.11 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - --- (16 headers 13 lines) --- Using INVITE request as basis request - a85738a9afd6fd630c362e1dbbe004e1@192.168.200.11 Sending to 192.168.200.11 : 5060 (NAT) Found user '200' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.200.11:3000 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 201 in from-internal (domain 24.176.126.46) list_route: hop: Transmitting (NAT) to 24.176.126.45:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.200.11;branch=z9hG4bK95933da61;received=24.176.126.45 From: Terry Cole ;tag=cfcff26be31280e To: 201 Call-ID: a85738a9afd6fd630c362e1dbbe004e1@192.168.200.11 CSeq: 821688029 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/200-089a42b8", "exten-vm|201|201") in new stack -- Executing Macro("SIP/200-089a42b8", "user-callerid") in new stack -- Executing NoOp("SIP/200-089a42b8", "user-callerid: device 200") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?report") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?start") in new stack -- Executing Set("SIP/200-089a42b8", "REALCALLERIDNUM=200") in new stack -- Executing NoOp("SIP/200-089a42b8", "REALCALLERIDNUM is 200") in new stack -- Executing Set("SIP/200-089a42b8", "AMPUSER=200") in new stack -- Executing Set("SIP/200-089a42b8", "AMPUSERCIDNAME=Terrys Desk") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?report") in new stack -- Executing Set("SIP/200-089a42b8", "CALLERID(all)=Terrys Desk <200>") in new stack -- Executing Set("SIP/200-089a42b8", "REALCALLERIDNUM=200") in new stack -- Executing NoOp("SIP/200-089a42b8", "TTL: ARG1: 201") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?continue") in new stack -- Executing Set("SIP/200-089a42b8", "_TTL=64") in new stack -- Executing GotoIf("SIP/200-089a42b8", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/200-089a42b8", "Using CallerID "Terrys Desk" <200>") in new stack -- Executing Set("SIP/200-089a42b8", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/200-089a42b8", "VMBOX=201") in new stack -- Executing Set("SIP/200-089a42b8", "EXTTOCALL=201") in new stack -- Executing Set("SIP/200-089a42b8", "CFUEXT=") in new stack -- Executing Set("SIP/200-089a42b8", "RT=15") in new stack -- Executing Macro("SIP/200-089a42b8", "record-enable|201|IN") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/200-089a42b8", "recordingcheck|20070506-151643|1178482603.129") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070506-151643|1178482603.129: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/200-089a42b8", "No recording needed") in new stack -- Executing Macro("SIP/200-089a42b8", "dial|15|tr|201") in new stack -- Executing DeadAGI("SIP/200-089a42b8", "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 1epbx modes... dialparties.agi: Caller ID name is 'Terrys Desk' number is '200' dialparties.agi: Methodology of ring is 'none' > dialparties.agi: USE_CONFIRMATION: 'FALSE' > dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 201 to extension map -- dialparties.agi: Extension 201 cf is disabled -- dialparties.agi: Extension 201 do not disturb is disabled > dialparties.agi: extnum: 201odes... > dialparties.agi: exthascw: 1odes... > dialparties.agi: exthascfb: 0des... > dialparties.agi: extcfb: x modes... > dialparties.agi: exthascfu: 0des... > dialparties.agi: extcfu: x modes... -- dialparties.agi: dbset CALLTRACE/201 to 200 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/200-089a42b8", "SIP/201|15|tr") in new stack Destroying call '646cd299107b584e22fb36cf3fcd7740@127.0.0.1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing GosubIf("SIP/200-089a42b8", "0?docfu|1") in new stack -- Executing NoOp("SIP/200-089a42b8", "Voicemail is 201") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?s-CHANUNAVAIL|1") in new stack -- Executing NoOp("SIP/200-089a42b8", "Sending to Voicemail box 201") in new stack -- Executing Macro("SIP/200-089a42b8", "vm|201|CHANUNAVAIL") in new stack -- Executing Macro("SIP/200-089a42b8", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/200-089a42b8", "user-callerid: Terrys Desk 200") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?report") in new stack -- Executing GotoIf("SIP/200-089a42b8", "1?start") in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp("SIP/200-089a42b8", "REALCALLERIDNUM is 200") in new stack -- Executing Set("SIP/200-089a42b8", "AMPUSER=200") in new stack -- Executing Set("SIP/200-089a42b8", "AMPUSERCIDNAME=Terrys Desk") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?report") in new stack -- Executing Set("SIP/200-089a42b8", "CALLERID(all)=Terrys Desk <200>") in new stack -- Executing Set("SIP/200-089a42b8", "REALCALLERIDNUM=200") in new stack -- Executing NoOp("SIP/200-089a42b8", "TTL: 64 ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/200-089a42b8", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/200-089a42b8", "Using CallerID "Terrys Desk" <200>") in new stack -- Executing Set("SIP/200-089a42b8", "VMGAIN=") in new stack -- Executing GotoIf("SIP/200-089a42b8", "1?s-CHANUNAVAIL|1") in new stack -- Goto (macro-vm,s-CHANUNAVAIL,1) -- Executing Macro("SIP/200-089a42b8", "get-vmcontext|201") in new stack -- Executing Set("SIP/200-089a42b8", "VMCONTEXT=default") in new stack -- Executing GotoIf("SIP/200-089a42b8", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing NoOp("SIP/200-089a42b8", "") in new stack -- Executing VoiceMail("SIP/200-089a42b8", "201@default|u") in new stack We're at 192.168.200.22 port 12304 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 24.176.126.45:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.11;branch=z9hG4bK95933da61;received=24.176.126.45 From: Terry Cole ;tag=cfcff26be31280e To: 201 ;tag=as77620810 Call-ID: a85738a9afd6fd630c362e1dbbe004e1@192.168.200.11 CSeq: 821688029 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2920 2920 IN IP4 192.168.200.22 s=session c=IN IP4 192.168.200.22 t=0 0 m=audio 12304 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 - - - - --- -- Playing 'vm-theperson' (language 'en') asterisk1*CLI> * Retrieving #freepbx modes... <-- SIP read from 192.168.200.11:5060: ACK sip:201@192.168.200.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.11;branch=z9hG4bKe8ea96c26 Max-Forwards: 70 Content-Length: 0 To: 201 ;tag=as77620810 From: Terry Cole ;tag=cfcff26be31280e Call-ID: a85738a9afd6fd630c362e1dbbe004e1@192.168.200.11 CSeq: 821688029 ACK Contact: Terry Cole Proxy-Authorization:Digest response="b58ea62e903725f5ec4bdc2a5c62776f",username="200",realm="asterisk",nonce="03a824d1",algorithm=MD5,uri="sip:201@192.168.200.22" User-Agent: Aastra 9112i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 --- (11 headers 0 lines) --- asterisk1*CLI> * Retrieving #freepbx modes... <-- SIP read from 192.168.200.11:5060: BYE sip:201@192.168.200.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.11;branch=z9hG4bK611edbb0f Max-Forwards: 70 Content-Length: 0 To: 201 ;tag=as77620810 From: Terry Cole ;tag=cfcff26be31280e Call-ID: a85738a9afd6fd630c362e1dbbe004e1@192.168.200.11 CSeq: 821688030 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="b4de2a8a308b07fa626c60cc04dd5596",username="200",realm="asterisk",nonce="03a824d1",algorithm=MD5,uri="sip:201@192.168.200.22" User-Agent: Aastra 9112i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 --- (12 headers 0 lines) --- Sending to 192.168.200.11 : 5060 (NAT) Transmitting (NAT) to 192.168.200.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.11;branch=z9hG4bK611edbb0f;received=192.168.200.11 From: Terry Cole ;tag=cfcff26be31280e To: 201 ;tag=as77620810 Call-ID: a85738a9afd6fd630c362e1dbbe004e1@192.168.200.11 Seq: 821688030 BYEetrieving #freepbx modes... User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: No route to destination --- == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/200-089a42b8' in macro 'vm' == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/200-089a42b8' in macro 'exten-vm' == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/200-089a42b8' Destroying call 'a85738a9afd6fd630c362e1dbbe004e1@192.168.200.11' asterisk1*CLI> * Retrieving #freepbx modes...