=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.05.23 00:00:41 =~=~=~=~=~=~=~=~=~=~=~= Reliably Transmitting (NAT) to 194.120.0.198:5060: OPTIONS sip:sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK0bbdd262;rport From: "Unknown" ;tag=as55bfa670 To: Contact: Call-ID: 56b6b95e2b1a3b300b3863f16bf34d43@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:01:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK0bbdd262;rport From: "Unknown" ;tag=as55bfa670 To: Contact: sip:194.120.0.198:5060 Call-ID: 56b6b95e2b1a3b300b3863f16bf34d43@192.168.2.31 CSeq: 102 OPTIONS Supported: foo User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Accept: application/sdp Accept-Encoding: Accept-Language: <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '56b6b95e2b1a3b300b3863f16bf34d43@192.168.2.31' Method: OPTIONS s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> <-------------> s07*CLI> Reliably Transmitting (no NAT) to 192.168.2.112:5060: OPTIONS sip:1@192.168.2.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK14e617e3 From: "Unknown" ;tag=as3e7b0424 To: Contact: Call-ID: 0b8193192069279759d633865ea086be@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:02:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK14e617e3;received=192.168.2.31 From: "Unknown" ;tag=as3e7b0424 To: "unknown" ;tag=293cc9c803 Call-ID: 0b8193192069279759d633865ea086be@192.168.2.31 CSeq: 102 OPTIONS Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '0b8193192069279759d633865ea086be@192.168.2.31' Method: OPTIONS s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> REGISTER sip:S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a802700000037f4a17206700007f8600000121;rport From: "unknown" ;tag=2d53c9f1a2 To: Contact: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 45 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.112 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a802700000037f4a17206700007f8600000121;rport;received=192.168.2.112 From: "unknown" ;tag=2d53c9f1a2 To: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 45 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> s07*CLI> <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a802700000037f4a17206700007f8600000121;rport;received=192.168.2.112 From: "unknown" ;tag=2d53c9f1a2 To: ;tag=as7ca8a332 Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 45 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66adc8ca" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7357702EAD61475099AA7F9970BE11A10xc0a80270' in 32000 ms (Method: REGISTER) s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> REGISTER sip:S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003804a1720670000097a00000124;rport From: "unknown" ;tag=2d53c9f1a2 To: Contact: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 46 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Authorization: Digest username="1",realm="asterisk",nonce="66adc8ca",uri="sip:S07",response="2ca09b0f67b00a23b4c0480ca1ac3416",algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.112 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003804a1720670000097a00000124;rport;received=192.168.2.112 From: "unknown" ;tag=2d53c9f1a2 To: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 46 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> s07*CLI> -- Registered SIP '1' at 192.168.2.112 port 5060 s07*CLI> Reliably Transmitting (no NAT) to 192.168.2.112:5060: OPTIONS sip:1@192.168.2.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK19bd495d From: "Unknown" ;tag=as0662edc0 To: Contact: Call-ID: 359aa7cc17ecac2b7894452311c42034@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:02:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003804a1720670000097a00000124;rport;received=192.168.2.112 From: "unknown" ;tag=2d53c9f1a2 To: ;tag=as7ca8a332 Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 46 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Fri, 22 May 2009 22:02:21 GMT Content-Length: 0 <------------> s07*CLI> Scheduling destruction of SIP dialog '7357702EAD61475099AA7F9970BE11A10xc0a80270' in 32000 ms (Method: REGISTER) s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK19bd495d;received=192.168.2.31 From: "Unknown" ;tag=as0662edc0 To: "unknown" ;tag=5627c9f1d1 Call-ID: 359aa7cc17ecac2b7894452311c42034@192.168.2.31 CSeq: 102 OPTIONS Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '359aa7cc17ecac2b7894452311c42034@192.168.2.31' Method: OPTIONS s07*CLI> Scheduling destruction of SIP dialog '2b929ee14e6f7fe9453967252448096e@192.168.2.31' in 6400 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 192.168.2.112:5060: NOTIFY sip:1@192.168.2.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK4d87e4a3 From: "Unknown" ;tag=as2381bf6f To: Contact: Call-ID: 2b929ee14e6f7fe9453967252448096e@192.168.2.31 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 87 Messages-Waiting: no Message-Account: sip:*97@192.168.2.31 Voice-Message: 0/0 (0/0) --- s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK4d87e4a3;received=192.168.2.31 From: "Unknown" ;tag=as2381bf6f To: "unknown" ;tag=2bf5c9f902 Call-ID: 2b929ee14e6f7fe9453967252448096e@192.168.2.31 CSeq: 102 NOTIFY Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2b929ee14e6f7fe9453967252448096e@192.168.2.31' Method: NOTIFY s07*CLI> REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 194.120.0.198:5060: REGISTER sip:sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK74cc85b0;rport From: ;tag=as72215d6a To: Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 128 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="damousys", realm="sip1.voipbuster.com", algorithm=MD5, uri="sip:sip1.voipbuster.com", nonce="3077547578", response="ab190d6e0a8fccee95c3a6ef31f52b91" Expires: 120 Contact: Event: registration Content-Length: 0 --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK74cc85b0;rport From: ;tag=as72215d6a To: Contact: sip:194.120.0.198:5060 Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 128 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sip1.voipbuster.com",nonce="3077643218",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Responding to challenge, registration to domain/host name sip1.voipbuster.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 194.120.0.198:5060: REGISTER sip:sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK50df03eb;rport From: ;tag=as7d375f5e To: Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 129 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="damousys", realm="sip1.voipbuster.com", algorithm=MD5, uri="sip:sip1.voipbuster.com", nonce="3077643218", response="371fcf8b8c249e65b49fa2f415174d27" Expires: 120 Contact: Event: registration Content-Length: 0 --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK50df03eb;rport From: ;tag=as7d375f5e To: Contact: ;expires=120 Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 129 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31' in 32000 ms (Method: REGISTER) s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> <-------------> s07*CLI> Really destroying SIP dialog '7357702EAD61475099AA7F9970BE11A10xc0a80270' Method: REGISTER s07*CLI> Reliably Transmitting (NAT) to 194.120.0.198:5060: OPTIONS sip:sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK35bba6ea;rport From: "Unknown" ;tag=as2e76ed96 To: Contact: Call-ID: 17ae64343c0be359059e2b003145d586@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:02:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK35bba6ea;rport From: "Unknown" ;tag=as2e76ed96 To: Contact: sip:194.120.0.198:5060 Call-ID: 17ae64343c0be359059e2b003145d586@192.168.2.31 CSeq: 102 OPTIONS Supported: foo User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Accept: application/sdp Accept-Encoding: Accept-Language: <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '17ae64343c0be359059e2b003145d586@192.168.2.31' Method: OPTIONS s07*CLI> q <--- SIP read from 192.168.2.112:5060 ---> <-------------> s07*CLI> quit Executing last minute cleanups [root@s07 ~]# asterisk -rvvvv Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found Connected to Asterisk 1.4.24.1 currently running on s07 (pid = 2382) s07*CLI> Verbosity is at least 4 s07*CLI> Really destroying SIP dialog '7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31' Method: REGISTER s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> INVITE sip:0736560974@S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003984a17209700001cb000000129;rport From: "unknown" ;tag=5da0caada0 To: Contact: Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 1 INVITE Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 368 Content-Type: application/sdp Supported: replaces,norefersub,timer v=0 o=- 3452018455 3452018455 IN IP4 192.168.2.112 s=SJphone c=IN IP4 192.168.2.112 t=0 0 m=audio 49174 RTP/AVP 3 97 98 8 0 101 c=IN IP4 192.168.2.112 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=setup:active a=sendrecv <-------------> --- (12 headers 17 lines) --- Sending to 192.168.2.112 : 5060 (NAT) Using INVITE request as basis request - BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 <--- Reliably Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003984a17209700001cb000000129;rport;received=192.168.2.112 From: "unknown" ;tag=5da0caada0 To: ;tag=as00416c31 Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="601cff03" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270' in 32000 ms (Method: INVITE) Found user '1' <--- SIP read from 192.168.2.112:5060 ---> ACK sip:0736560974@S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003984a17209700001cb000000129;rport From: "unknown" ;tag=5da0caada0 To: ;tag=as00416c31 Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 1 ACK Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 192.168.2.112:5060 ---> INVITE sip:0736560974@S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003994a172097000014480000012b;rport From: "unknown" ;tag=5da0caada0 To: Contact: Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 2 INVITE Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 368 Content-Type: application/sdp Supported: replaces,norefersub,timer Proxy-Authorization: Digest username="1",realm="asterisk",nonce="601cff03",uri="sip:0736560974@S07",response="662c86a7032ff256c8b3d095535d6638",algorithm=MD5 v=0 o=- 3452018455 3452018455 IN IP4 192.168.2.112 s=SJphone c=IN IP4 192.168.2.112 t=0 0 m=audio 49174 RTP/AVP 3 97 98 8 0 101 c=IN IP4 192.168.2.112 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=setup:active a=sendrecv <-------------> --- (13 headers 17 lines) --- Sending to 192.168.2.112 : 5060 (NAT) Using INVITE request as basis request - BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 Found user '1' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.112:49174 Found audio description format GSM for ID 3 Found audio description format iLBC for ID 97 Found audio description format iLBC for ID 98 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=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 192.168.2.112:49174 Looking for 0736560974 in from-internal (domain S07) list_route: hop: <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003994a172097000014480000012b;rport;received=192.168.2.112 From: "unknown" ;tag=5da0caada0 To: Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [0736560974@from-internal:1] Macro("SIP/1-1de84aa0", "user-callerid|SKIPTTL|") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1-1de84aa0", "AMPUSER=1") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1-1de84aa0", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1-1de84aa0", "1|Set|REALCALLERIDNUM=1") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1-1de84aa0", "AMPUSER=1") in new stack s07*CLI> -- Executing [s@macro-user-callerid:5] Set("SIP/1-1de84aa0", "AMPUSERCIDNAME=damousys") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1-1de84aa0", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1-1de84aa0", "AMPUSERCID=1") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1-1de84aa0", "CALLERID(all)="damousys" <1>") in new stack -- Executing [s@macro-user-callerid:9] Set("SIP/1-1de84aa0", "REALCALLERIDNUM=1") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1-1de84aa0", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("SIP/1-1de84aa0", "Using CallerID "damousys" <1>") in new stack -- Executing [0736560974@from-internal:2] Set("SIP/1-1de84aa0", "_NODEST=") in new stack -- Executing [0736560974@from-internal:3] Macro("SIP/1-1de84aa0", "record-enable|1|OUT|") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/1-1de84aa0", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI("SIP/1-1de84aa0", "recordingcheck|20090523-000314|1243029794.9") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck s07*CLI> recordingcheck|20090523-000314|1243029794.9: Outbound recording not enabled s07*CLI> -- AGI Script recordingcheck completed, returning 0 s07*CLI> -- Executing [s@macro-record-enable:5] MacroExit("SIP/1-1de84aa0", "") in new stack s07*CLI> -- Executing [0736560974@from-internal:4] Macro("SIP/1-1de84aa0", "dialout-trunk|2|0736560974||") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:1] Set("SIP/1-1de84aa0", "DIAL_TRUNK=2") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1-1de84aa0", "0?sub-pincheck|s|1") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1-1de84aa0", "0?disabletrunk|1") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:4] Set("SIP/1-1de84aa0", "DIAL_NUMBER=0736560974") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:5] Set("SIP/1-1de84aa0", "DIAL_TRUNK_OPTIONS=tr") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:6] Set("SIP/1-1de84aa0", "OUTBOUND_GROUP=OUT_2") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1-1de84aa0", "1?nomax") in new stack s07*CLI> -- Goto (macro-dialout-trunk,s,9) s07*CLI> -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1-1de84aa0", "0?skipoutcid") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:10] Set("SIP/1-1de84aa0", "DIAL_TRUNK_OPTIONS=") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1-1de84aa0", "outbound-callerid|2") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1-1de84aa0", "0|SetCallerPres|") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1-1de84aa0", "1|Set|REALCALLERIDNUM=1") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1-1de84aa0", "1?normcid") in new stack s07*CLI> -- Goto (macro-outbound-callerid,s,6) s07*CLI> -- Executing [s@macro-outbound-callerid:6] Set("SIP/1-1de84aa0", "USEROUTCID=damousys") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:7] Set("SIP/1-1de84aa0", "EMERGENCYCID=") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:8] Set("SIP/1-1de84aa0", "TRUNKOUTCID=0031857852105") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1-1de84aa0", "1?trunkcid") in new stack s07*CLI> -- Goto (macro-outbound-callerid,s,12) s07*CLI> -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1-1de84aa0", "1|Set|CALLERID(all)=0031857852105") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1-1de84aa0", "1|Set|CALLERID(all)=damousys") in new stack s07*CLI> -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1-1de84aa0", "0|SetCallerPres|prohib_passed_screen") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1-1de84aa0", "0|AGI|fixlocalprefix") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:13] Set("SIP/1-1de84aa0", "OUTNUM=0736560974") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:14] Set("SIP/1-1de84aa0", "custom=SIP/VoipBuster") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1-1de84aa0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1-1de84aa0", "dialout-trunk-predial-hook|") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1-1de84aa0", "") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1-1de84aa0", "0?bypass|1") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1-1de84aa0", "0?customtrunk") in new stack s07*CLI> -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1-1de84aa0", "SIP/VoipBuster/0736560974|300|") in new stack s07*CLI> Audio is at 192.168.2.31 port 10898 s07*CLI> Adding codec 0x4 (ulaw) to SDP s07*CLI> Adding codec 0x8 (alaw) to SDP s07*CLI> Adding codec 0x2 (gsm) to SDP s07*CLI> Reliably Transmitting (NAT) to 194.120.0.198:5060: INVITE sip:0736560974@sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK4dc26a88;rport From: "damousys" ;tag=as6674a06d To: Contact: Call-ID: 2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:03:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 229 v=0 o=root 2382 2382 IN IP4 192.168.2.31 s=session c=IN IP4 192.168.2.31 t=0 0 m=audio 10898 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- s07*CLI> -- Called VoipBuster/0736560974 s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK4dc26a88;rport From: "damousys" ;tag=as6674a06d To: Contact: sip:0736560974@194.120.0.198:5060 Call-ID: 2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sip1.voipbuster.com",nonce="3225484515",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 194.120.0.198:5060: ACK sip:0736560974@sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK4dc26a88;rport From: "damousys" ;tag=as6674a06d To: Contact: Call-ID: 2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 192.168.2.31 port 10898 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (NAT) to 194.120.0.198:5060: INVITE sip:0736560974@sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK3bfdb5c6;rport From: "damousys" ;tag=as6674a06d To: Contact: C s07*CLI> all-ID: 2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="damousys", realm="sip1.voipbuster.com", algorithm=MD5, uri="sip:0736560974@sip1.voipbuster.com", nonce="3225484515", response="c149c7e7c2fb593486d9f6ca78e12485" Date: Fri, 22 May 2009 22:03:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 229 v=0 o=root 2382 2383 IN IP4 192.168.2.31 s=session c=IN IP4 192.168.2.31 t=0 0 m=audio 10898 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK3bfdb5c6;rport From: "damousys" ;tag=as6674a06d To: Contact: sip:0736560974@194.120.0.198:5060 Call-ID: 2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> s07*CLI> --- (10 headers 0 lines) --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK3bfdb5c6;rport From: "damousys" ;tag=as6674a06d To: Contact: sip:0736560974@194.120.0.198:5060 Call-ID: 2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 194.120.0.198:5060: ACK sip:0736560974@sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK3bfdb5c6;rport From: "damousys" ;tag=as6674a06d To: Contact: Call-ID: 2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- s07*CLI> -- SIP/VoipBuster-1de88e80 is circuit-busy s07*CLI> == Everyone is busy/congested at this time (1:0/1/0) s07*CLI> -- Executing [s@macro-dialout-trunk:20] Goto("SIP/1-1de84aa0", "s-CONGESTION|1") in new stack s07*CLI> -- Goto (macro-dialout-trunk,s-CONGESTION,1) s07*CLI> -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/1-1de84aa0", "1?noreport") in new stack s07*CLI> -- Goto (macro-dialout-trunk,s-CONGESTION,3) s07*CLI> -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/1-1de84aa0", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack s07*CLI> -- Executing [0736560974@from-internal:5] Macro("SIP/1-1de84aa0", "outisbusy|") in new stack s07*CLI> -- Executing [s@macro-outisbusy:1] Playback("SIP/1-1de84aa0", "all-circuits-busy-now|noanswer") in new stack s07*CLI> Audio is at 192.168.2.31 port 17444 s07*CLI> Adding codec 0x4 (ulaw) to SDP s07*CLI> Adding codec 0x8 (alaw) to SDP s07*CLI> Adding non-codec 0x1 (telephone-event) to SDP s07*CLI> <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003994a172097000014480000012b;rport;received=192.168.2.112 From: "unknown" ;tag=5da0caada0 To: ;tag=as62f557a5 Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: s07*CLI> Content-Type: application/sdp Content-Length: 262 v=0 o=root 2382 2382 IN IP4 192.168.2.31 s=session c=IN IP4 192.168.2.31 t=0 0 m=audio 17444 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 <------------> s07*CLI> -- Playing 'all-circuits-busy-now' (language 'en') s07*CLI> Really destroying SIP dialog '2d186bf93310bcd12fb96003166728cd@sip1.voipbuster.com' Method: INVITE s07*CLI> -- Executing [s@macro-outisbusy:2] Playback("SIP/1-1de84aa0", "pls-try-call-later|noanswer") in new stack -- Playing 'pls-try-call-later' (language 'en') s07*CLI> -- Executing [s@macro-outisbusy:3] Macro("SIP/1-1de84aa0", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1-1de84aa0", "vw") in new stack -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1-1de84aa0", "") in new stack -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1-1de84aa0", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1-1de84aa0", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1-1de84aa0", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing [s@macro-hangupcall:11] Hangup("SIP/1-1de84aa0", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1-1de84aa0' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/1-1de84aa0' in macro 'outisbusy' == Spawn extension (from-internal, 0736560974, 5) exited non-zero on 'SIP/1-1de84aa0' -- Executing [h@from-internal:1] Macro("SIP/1-1de84aa0", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1-1de84aa0", "vw") in new stack -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1-1de84aa0", "") in new stack -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1-1de84aa0", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1-1de84aa0", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1-1de84aa0", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing [s@macro-hangupcall:11] Hangup("SIP/1-1de84aa0", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1-1de84aa0' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1-1de84aa0' Scheduling destruction of SIP dialog 'BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003994a172097000014480000012b;rport;received=192.168.2.112 From: "unknown" ;tag=5da0caada0 To: ;tag=as62f557a5 Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> ACK sip:0736560974@S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003994a172097000014480000012b;rport From: "unknown" ;tag=5da0caada0 To: ;tag=as62f557a5 Call-ID: BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270 CSeq: 2 ACK Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Proxy-Authorization: Digest username="1",realm="asterisk",nonce="601cff03",uri="sip:0736560974@S07",response="662c86a7032ff256c8b3d095535d6638",algorithm=MD5 <-------------> --- (10 headers 0 lines) --- s07*CLI> Reliably Transmitting (no NAT) to 192.168.2.112:5060: OPTIONS sip:1@192.168.2.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK2a134cc3 From: "Unknown" ;tag=as06b1e383 To: Contact: Call-ID: 6972cafb39690c2406f0f0c82a72b037@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:03:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK2a134cc3;received=192.168.2.31 From: "Unknown" ;tag=as06b1e383 To: "unknown" ;tag=4955cac748 Call-ID: 6972cafb39690c2406f0f0c82a72b037@192.168.2.31 CSeq: 102 OPTIONS Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '6972cafb39690c2406f0f0c82a72b037@192.168.2.31' Method: OPTIONS s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> <-------------> s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> <-------------> s07*CLI> Really destroying SIP dialog 'BE5A67AF148347B1B4BFCAA3794802CD0xc0a80270' Method: ACK s07*CLI> Reliably Transmitting (NAT) to 194.120.0.198:5060: OPTIONS sip:sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK79f2143b;rport From: "Unknown" ;tag=as7cdbad4d To: Contact: Call-ID: 1e4438295e6182f25082e05c0554afd5@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:03:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK79f2143b;rport From: "Unknown" ;tag=as7cdbad4d To: Contact: sip:194.120.0.198:5060 Call-ID: 1e4438295e6182f25082e05c0554afd5@192.168.2.31 CSeq: 102 OPTIONS Supported: foo User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Accept: application/sdp Accept-Encoding: Accept-Language: <-------------> --- (13 headers 0 lines) --- s07*CLI> Really destroying SIP dialog '1e4438295e6182f25082e05c0554afd5@192.168.2.31' Method: OPTIONS s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> <-------------> s07*CLI> Reliably Transmitting (no NAT) to 192.168.2.112:5060: OPTIONS sip:1@192.168.2.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK7ed84b5e From: "Unknown" ;tag=as111a3a1c To: Contact: Call-ID: 4f9019ae370aef4219804a55312eae89@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:04:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK7ed84b5e;received=192.168.2.31 From: "Unknown" ;tag=as111a3a1c To: "unknown" ;tag=681fcb9ccf Call-ID: 4f9019ae370aef4219804a55312eae89@192.168.2.31 CSeq: 102 OPTIONS Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '4f9019ae370aef4219804a55312eae89@192.168.2.31' Method: OPTIONS s07*CLI> REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 194.120.0.198:5060: REGISTER sip:sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK5b83d7ce;rport From: ;tag=as241e9269 To: Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 130 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="damousys", realm="sip1.voipbuster.com", algorithm=MD5, uri="sip:sip1.voipbuster.com", nonce="3077643218", response="371fcf8b8c249e65b49fa2f415174d27" Expires: 120 Contact: Event: registration Content-Length: 0 --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK5b83d7ce;rport From: ;tag=as241e9269 To: Contact: sip:194.120.0.198:5060 Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 130 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sip1.voipbuster.com",nonce="3077738781",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Responding to challenge, registration to domain/host name sip1.voipbuster.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 194.120.0.198:5060: REGISTER sip:sip1.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK6e972b44;rport From: ;tag=as36c02ee5 To: Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 131 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="damousys", realm="sip1.voipbuster.com", algorithm=MD5, uri="sip:sip1.voipbuster.com", nonce="3077738781", response="ce93eebdaa2587dac30024e04a9b0511" Expires: 120 Contact: Event: registration Content-Length: 0 --- s07*CLI> <--- SIP read from 194.120.0.198:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK6e972b44;rport From: ;tag=as36c02ee5 To: Contact: ;expires=120 Call-ID: 7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31 CSeq: 131 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '7d4883eb2e5c416d6acd4ebc3883a437@192.168.2.31' in 32000 ms (Method: REGISTER) s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> REGISTER sip:S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003ae4a1720df0000245d00000131;rport From: "unknown" ;tag=5da4cbc6ac To: Contact: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 47 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.112 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003ae4a1720df0000245d00000131;rport;received=192.168.2.112 From: "unknown" ;tag=5da4cbc6ac To: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 47 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> s07*CLI> <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003ae4a1720df0000245d00000131;rport;received=192.168.2.112 From: "unknown" ;tag=5da4cbc6ac To: ;tag=as1e984132 Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 47 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29cf7b65" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7357702EAD61475099AA7F9970BE11A10xc0a80270' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.2.112:5060 ---> REGISTER sip:S07 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003af4a1720df0000438900000134;rport From: "unknown" ;tag=5da4cbc6ac To: Contact: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 48 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Authorization: Digest username="1",realm="asterisk",nonce="29cf7b65",uri="sip:S07",response="0ca2d916e846846a0de10c0a86e84333",algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.112 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003af4a1720df0000438900000134;rport;received=192.168.2.112 From: "unknown" ;tag=5da4cbc6ac To: Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 48 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> s07*CLI> -- Registered SIP '1' at 192.168.2.112 port 5060 s07*CLI> Reliably Transmitting (no NAT) to 192.168.2.112:5060: OPTIONS sip:1@192.168.2.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK3aa89d44 From: "Unknown" ;tag=as42a204f0 To: Contact: Call-ID: 512179793eeaf8fe6b0063002297601b@192.168.2.31 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 May 2009 22:04:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- s07*CLI> <--- Transmitting (no NAT) to 192.168.2.112:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.112;branch=z9hG4bKc0a80270000003af4a1720df0000438900000134;rport;received=192.168.2.112 From: "unknown" ;tag=5da4cbc6ac To: ;tag=as1e984132 Call-ID: 7357702EAD61475099AA7F9970BE11A10xc0a80270 CSeq: 48 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Fri, 22 May 2009 22:04:33 GMT Content-Length: 0 <------------> s07*CLI> Scheduling destruction of SIP dialog '7357702EAD61475099AA7F9970BE11A10xc0a80270' in 32000 ms (Method: REGISTER) s07*CLI> <--- SIP read from 192.168.2.112:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK3aa89d44;received=192.168.2.31 From: "Unknown" ;tag=as42a204f0 To: "unknown" ;tag=1072cbc6fa Call-ID: 512179793eeaf8fe6b0063002297601b@192.168.2.31 CSeq: 102 OPTIONS Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) <-------------> s07*CLI> --- (8 headers 0 lines) --- s07*CLI> Really destroying SIP dialog '512179793eeaf8fe6b0063002297601b@192.168.2.31' Method: OPTIONS s07*CLI> Scheduling destruction of SIP dialog '4fe09ed942f5bdda39359d2972dbc8ae@192.168.2.31' in 6400 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 192.168.2.112:5060: NOTIFY sip:1@192.168.2.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK2c6ddd05 From: "Unknown" ;tag=as3ce4bbfc To: Contact: Call-ID: 4fe09ed942f5bdda39359d2972dbc8ae@192.168.2.31 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 87 Messages-Waiting: no Message-Account: sip:*97@192.168.2.31 Voice-Message: 0/0 (0/0) --- <--- SIP read from 192.168.2.112:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.31:5060;branch=z9hG4bK2c6ddd05;received=192.168.2.31 From: "Unknown" ;tag=as3ce4bbfc To: "unknown" ;tag=5d20cbcdec Call-ID: 4fe09ed942f5bdda39359d2972dbc8ae@192.168.2.31 CSeq: 102 NOTIFY Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '4fe09ed942f5bdda39359d2972dbc8ae@192.168.2.31' Method: NOTIFY s07*CLI>