sip debug peer 301 SIP Debugging Enabled for IP: 192.168.202.60:5060 Destroying call 'FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60' Reliably Transmitting (no NAT) to 192.168.202.60:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000003d04471be76000009a4000007c4;received=192.168.202.60 From: "Administrator";tag=2094089012093 To: ;tag=as3620d665 Call-ID: C712650C-4544-4662-B306-9F1CD493DF01@192.168.202.60 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="6402b862" Content-Length: 0 --- Scheduling destruction of call 'C712650C-4544-4662-B306-9F1CD493DF01@192.168.202.60' in 15000 ms Using INVITE request as basis request - C712650C-4544-4662-B306-9F1CD493DF01@192.168.202.60 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.202.60:49166 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 299 in from-internal (domain 172.31.2.83) list_route: hop: Transmitting (no NAT) to 192.168.202.60:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000003d04471be7600000c21000007c5;received=192.168.202.60 From: "Administrator";tag=2094089012093 To: Call-ID: C712650C-4544-4662-B306-9F1CD493DF01@192.168.202.60 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/301-5c2e", "exten-vm|299|299") in new stack -- Executing Macro("SIP/301-5c2e", "user-callerid") in new stack -- Executing GotoIf("SIP/301-5c2e", "0?report") in new stack -- Executing GotoIf("SIP/301-5c2e", "0?start") in new stack -- Executing Set("SIP/301-5c2e", "REALCALLERIDNUM=301") in new stack -- Executing NoOp("SIP/301-5c2e", "REALCALLERIDNUM is 301") in new stack -- Executing Set("SIP/301-5c2e", "AMPUSER=301") in new stack -- Executing Set("SIP/301-5c2e", "AMPUSERCIDNAME=test1") in new stack -- Executing GotoIf("SIP/301-5c2e", "0?report") in new stack -- Executing Set("SIP/301-5c2e", "CALLERID(all)=test1 <301>") in new stack -- Executing NoOp("SIP/301-5c2e", "Using CallerID "test1" <301>") in new stack -- Executing Set("SIP/301-5c2e", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/301-5c2e", "VMBOX=299") in new stack -- Executing Set("SIP/301-5c2e", "EXTTOCALL=299") in new stack -- Executing Set("SIP/301-5c2e", "CFUEXT=") in new stack -- Executing Set("SIP/301-5c2e", "RT=15") in new stack -- Executing Macro("SIP/301-5c2e", "record-enable|299|IN") in new stack -- Executing GotoIf("SIP/301-5c2e", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/301-5c2e", "recordingcheck|20060522-153510|1148304910.2") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060522-153510|1148304910.2: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/301-5c2e", "No recording needed") in new stack -- Executing GotoIf("SIP/301-5c2e", "0?dolocaldial|1") in new stack -- Executing Macro("SIP/301-5c2e", "dial|15|tr|299") in new stack -- Executing AGI("SIP/301-5c2e", "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 'test1' number is '301' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 299 to extension map -- dialparties.agi: Extension 299 cf is disabled -- dialparties.agi: Extension 299 do not disturb is disabled > dialparties.agi: extnum: 299 > dialparties.agi: exthascw: 0 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 > dialparties.agi: ExtensionState: 0 -- dialparties.agi: Checking CW and CFB status for extension 299 -- dialparties.agi: DbSet CALLTRACE/299 to 301 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/301-5c2e", "SIP/299|15|tr") in new stack -- Called 299 Transmitting (no NAT) to 192.168.202.60:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000003d04471be7600000c21000007c5;received=192.168.202.60 From: "Administrator";tag=2094089012093 To: ;tag=as3bcc6349 Call-ID: C712650C-4544-4662-B306-9F1CD493DF01@192.168.202.60 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- SIP/299-2718 is ringing -- SIP/299-2718 answered SIP/301-5c2e We're at 172.31.2.83 port 16282 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 192.168.202.60:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000003d04471be7600000c21000007c5;received=192.168.202.60 From: "Administrator";tag=2094089012093 To: ;tag=as3bcc6349 Call-ID: C712650C-4544-4662-B306-9F1CD493DF01@192.168.202.60 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 24148 24148 IN IP4 172.31.2.83 s=session c=IN IP4 172.31.2.83 t=0 0 m=audio 16282 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 - - - - --- -- Attempting native bridge of SIP/301-5c2e and SIP/299-2718 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.202.60:5060: OPTIONS sip:301@192.168.202.60 SIP/2.0 Via: SIP/2.0/UDP 172.31.2.83:5060;branch=z9hG4bK5c9d317e From: "Unknown" ;tag=as22082c13 To: Contact: Call-ID: 27edc7fd342e693a6a4f3f9f506a4085@172.31.2.83 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 22 May 2006 13:35:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '27edc7fd342e693a6a4f3f9f506a4085@172.31.2.83' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/301-5c2e' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/301-5c2e' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/301-5c2e' 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.202.60:5060: OPTIONS sip:301@192.168.202.60 SIP/2.0 Via: SIP/2.0/UDP 172.31.2.83:5060;branch=z9hG4bK589a4676 From: "Unknown" ;tag=as649b5513 To: Contact: Call-ID: 059c579c03160fc64eedc9c64d97791f@172.31.2.83 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 22 May 2006 13:36:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '059c579c03160fc64eedc9c64d97791f@172.31.2.83' Transmitting (no NAT) to 192.168.202.60:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000000104471bed20000048c000007cd;received=192.168.202.60 From: ;tag=210337184397 To: Call-ID: FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60 CSeq: 349 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.202.60:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000000104471bed20000048c000007cd;received=192.168.202.60 From: ;tag=210337184397 To: ;tag=as7d1de5b0 Call-ID: FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60 CSeq: 349 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="453166cc" Content-Length: 0 --- Scheduling destruction of call 'FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60' in 15000 ms Using latest REGISTER request as basis request Transmitting (no NAT) to 192.168.202.60:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000000104471bed300006911000007d0;received=192.168.202.60 From: ;tag=2103385930570 To: Call-ID: FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60 CSeq: 350 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.202.60:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.202.60;rport;branch=z9hG4bKc0a8ca3c000000104471bed300006911000007d0;received=192.168.202.60 From: ;tag=2103385930570 To: ;tag=as7d1de5b0 Call-ID: FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60 CSeq: 350 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Mon, 22 May 2006 13:36:43 GMT Content-Length: 0 --- Scheduling destruction of call 'FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60' in 15000 ms 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.202.60:5060: NOTIFY sip:301@192.168.202.60 SIP/2.0 Via: SIP/2.0/UDP 172.31.2.83:5060;branch=z9hG4bK579387eb From: "Unknown" ;tag=as05270db1 To: Contact: Call-ID: 34658ead5e92b5cc6d0bb59419cb6c0c@172.31.2.83 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 91 essages-Waiting: no Message-Account: sip:asterisk@172.31.2.83 Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '34658ead5e92b5cc6d0bb59419cb6c0c@172.31.2.83' in 15000 ms Destroying call '34658ead5e92b5cc6d0bb59419cb6c0c@172.31.2.83' Destroying call 'FC2F2E53-8183-4F2D-B86B-2E1D4F03871A@192.168.202.60' 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.202.60:5060: OPTIONS sip:301@192.168.202.60 SIP/2.0 Via: SIP/2.0/UDP 172.31.2.83:5060;branch=z9hG4bK6342122e From: "Unknown" ;tag=as0557612d To: Contact: Call-ID: 01015d637edab5a32973e47a79a0a2bb@172.31.2.83 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 22 May 2006 13:37:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '01015d637edab5a32973e47a79a0a2bb@172.31.2.83'