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- <------------>
- <--- SIP read from UDP:192.168.1.201:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7321dd16;received=192.168.1.202 ;rport=5060
- From: "3000" <sip:3000@192.168.1.202>;tag=as146d1bbf
- To: <sip:2001@192.168.1.201>;tag=as320d1474
- Call-ID: 063d0bb833d9a848487cbc944ddc02ae@192.168.1.202:5060
- CSeq: 102 INVITE
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d877a7b"
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Transmitting (NAT) to 192.168.1.201:5060:
- ACK sip:2001@192.168.1.201 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7321dd16;rport
- Max-Forwards: 70
- From: "3000" <sip:3000@192.168.1.202>;tag=as146d1bbf
- To: <sip:2001@192.168.1.201>;tag=as320d1474
- Contact: <sip:3000@192.168.1.202:5060>
- Call-ID: 063d0bb833d9a848487cbc944ddc02ae@192.168.1.202:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.8.21.0
- Content-Length: 0
- ---
- [Apr 28 22:21:03] NOTICE[1179]: chan_sip.c:20959 handle_response_invite: Failed t o authenticate on INVITE to '"3000" <sip:3000@192.168.1.202>;tag=as146d1bbf'
- -- SIP/central1-00000040 is circuit-busy
- == Everyone is busy/congested at this time (1:0/1/0)
- -- Stopped music on hold on SIP/3000-0000003f
- -- Auto fallthrough, channel 'SIP/3000-0000003f' status is 'CONGESTION'
- <--- Reliably Transmitting (NAT) to 192.168.1.4:5060 --->
- SIP/2.0 503 Service Unavailable
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-72b7a679eb9f0ba4-1---d 8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4e38ce48
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as2b7cadfd
- Call-ID: M2ZmYzIyNjhlMTNkZWNjZDlmZGI0ODRkYmJhZGM2NDc.
- CSeq: 2 INVITE
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- X-Asterisk-HangupCause: Call Rejected
- X-Asterisk-HangupCauseCode: 21
- Content-Length: 0
- <------------>
- Really destroying SIP dialog '063d0bb833d9a848487cbc944ddc02ae@192.168.1.202:5060 ' Method: INVITE
- <--- SIP read from UDP:192.168.1.4:5060 --->
- ACK sip:2001@192.168.1.202;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-72b7a679eb9f0ba4-1---d 8754z-
- Max-Forwards: 70
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as2b7cadfd
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4e38ce48
- Call-ID: M2ZmYzIyNjhlMTNkZWNjZDlmZGI0ODRkYmJhZGM2NDc.
- CSeq: 2 ACK
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- Really destroying SIP dialog 'M2ZmYzIyNjhlMTNkZWNjZDlmZGI0ODRkYmJhZGM2NDc.' Metho d: ACK
- Really destroying SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' Method: REGISTER
- <--- SIP read from UDP:192.168.1.9:5066 --->
- REGISTER sip:192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-a60582e5
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 223 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="74d24849",uri="sip: 192.168.1.202",algorithm=MD5,response="a7e2d0010ab03644f673a8bc1ec93e6a"
- Contact: "3001" <sip:3001@192.168.1.9:5066>;expires=36
- User-Agent: Sipura/SPA941-4.1.8
- Content-Length: 0
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- <------------->
- --- (12 headers 0 lines) ---
- Sending to 192.168.1.9:5066 (NAT)
- <--- Transmitting (NAT) to 192.168.1.9:5066 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-a60582e5;received=192.168.1.9;rp ort=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as6baecdec
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 223 REGISTER
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37d48d5f"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' in 32000 ms (Method: REGISTER)
- <--- SIP read from UDP:192.168.1.9:5066 --->
- REGISTER sip:192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-fc72ba1e
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 224 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="37d48d5f",uri="sip: 192.168.1.202",algorithm=MD5,response="1f547f45b641b191ee0f5b0dad7f77a6"
- Contact: "3001" <sip:3001@192.168.1.9:5066>;expires=36
- User-Agent: Sipura/SPA941-4.1.8
- Content-Length: 0
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- <------------->
- --- (12 headers 0 lines) ---
- Sending to 192.168.1.9:5066 (NAT)
- Reliably Transmitting (NAT) to 192.168.1.9:5066:
- OPTIONS sip:3001@192.168.1.9:5066 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK4b5373e8;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as59900734
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 68a8d102175667dd392c99122f9dd21a@192.168.1.202:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 01:21:12 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- <--- Transmitting (NAT) to 192.168.1.9:5066 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-fc72ba1e;received=192.168.1.9;rp ort=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as6baecdec
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 224 REGISTER
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Expires: 60
- Contact: <sip:3001@192.168.1.9:5066>;expires=60
- Date: Mon, 29 Apr 2013 01:21:12 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '24b8d401527e33eb5cd5b1ec4668a247@192.168.1. 202:5060' in 6400 ms (Method: NOTIFY)
- Reliably Transmitting (NAT) to 192.168.1.9:5066:
- NOTIFY sip:3001@192.168.1.9:5066 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK47f698f6;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as6657280f
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 24b8d401527e33eb5cd5b1ec4668a247@192.168.1.202:5060
- CSeq: 102 NOTIFY
- User-Agent: Asterisk PBX 1.8.21.0
- Event: message-summary
- Content-Type: application/simple-message-summary
- Content-Length: 93
- Messages-Waiting: no
- Message-Account: sip:asterisk@192.168.1.202
- Voice-Message: 0/0 (0/0)
- ---
- Scheduling destruction of SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' in 32000 ms (Method: REGISTER)
- <--- SIP read from UDP:192.168.1.9:5066 --->
- SIP/2.0 200 OK
- To: <sip:3001@192.168.1.9:5066>;tag=3730351cb34f217fi0
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as59900734
- Call-ID: 68a8d102175667dd392c99122f9dd21a@192.168.1.202:5060
- CSeq: 102 OPTIONS
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK4b5373e8
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- <------------->
- --- (9 headers 0 lines) ---
- Really destroying SIP dialog '68a8d102175667dd392c99122f9dd21a@192.168.1.202:5060 ' Method: OPTIONS
- <--- SIP read from UDP:192.168.1.9:5066 --->
- SIP/2.0 200 OK
- To: <sip:3001@192.168.1.9:5066>;tag=3730351cb34f217fi0
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as6657280f
- Call-ID: 24b8d401527e33eb5cd5b1ec4668a247@192.168.1.202:5060
- CSeq: 102 NOTIFY
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK47f698f6
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- Really destroying SIP dialog '24b8d401527e33eb5cd5b1ec4668a247@192.168.1.202:5060 ' Method: NOTIFY
- asterisk*CLI> sip set debug off
- SIP Debugging Disabled
- asterisk*CLI> exit
- Executing last minute cleanups
- Asterisk ending (0).
- root@asterisk:/etc/asterisk# nano debugc1.txt
- GNU nano 2.2.4 Fichero: debugc1.txt
- Parsing /etc/asterisk/asterisk.conf
- Seeding global EID '00:17:31:19:92:83' from 'eth0' using 'siocgifhwaddr'
- Asterisk 1.8.21.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
- Created by Mark Spencer <markster@digium.com>
- 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.
- =========================================================================
- Connected to Asterisk 1.8.21.0 currently running on asterisk (pid = 1162)
- asterisk*CLI> ^M^[[0KVerbosity is at least 38
- Core debug is at least 2
- ^M^[[Kasterisk*CLI> exit^H^H^H^Hupdatedb^[[15Gsip set debug off^H^Hn^[[K^[[15Gexit^[[K^H^H^H^Hsip set debug on
- asterisk*CLI> ^M^[[0KSIP Debugging enabled
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.9:5066 --->
- NOTIFY sip:192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-ee6a353b
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: <sip:192.168.1.202>
- Call-ID: 554ddcd8-5f93cf3@192.168.1.9
- CSeq: 250 NOTIFY
- Max-Forwards: 70
- Event: keep-alive
- User-Agent: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- <--- Transmitting (NAT) to 192.168.1.9:5066 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-ee6a353b;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: <sip:192.168.1.202>;tag=as6d44df28
- Call-ID: 554ddcd8-5f93cf3@192.168.1.9
- CSeq: 250 NOTIFY
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
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