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- 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 = 1568)
- asterisk*CLI> ^M^[[0KVerbosity is at least 37
- Core debug was 0 and is now 2
- ^M^[[Kasterisk*CLI> ^M^[[0K == Using SIP RTP CoS mark 5
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Executing [2001@default:1] ^[[1;36mDial^[[0m("^[[1;35mSIP/3000-00000000^[[0m", "^[[1;35mSIP/central1/2001,30,Ttm^[[0m") in new stack
- ^M^[[Kasterisk*CLI> ^M^[[0K == Using SIP RTP CoS mark 5
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Called SIP/central1/2001
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Started music on hold, class 'default', on SIP/3000-00000000
- ^M^[[Kasterisk*CLI> ^M^[[0K[Apr 29 16:51:20] ^[[1;33mNOTICE^[[0m[1585]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m20959^[[0m ^[[1;37mhandle_response_invite^[[0m: Failed to authe$
- -- SIP/central1-00000001 is circuit-busy
- ^M^[[Kasterisk*CLI> ^M^[[0K == Everyone is busy/congested at this time (1:0/1/0)
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Stopped music on hold on SIP/3000-00000000
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Auto fallthrough, channel 'SIP/3000-00000000' status is 'CONGESTION'
- ^M^[[Kasterisk*CLI> sip set debug on
- asterisk*CLI> ^M^[[0KSIP Debugging enabled
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- INVITE sip:3000@192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0b161d10;rport
- Max-Forwards: 70
- From: "2001" <sip:2001@192.168.1.201>;tag=as52f3269a
- To: <sip:3000@192.168.1.202>
- Contact: <sip:2001@192.168.1.201:5060>
- Call-ID: 4331ba2852d8ae6562692e7b23d2b627@192.168.1.201:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 16:06:50 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 262
- v=0
- o=root 1123005417 1123005417 IN IP4 192.168.1.201
- s=Asterisk PBX 1.8.21.0
- c=IN IP4 192.168.1.201
- t=0 0
- m=audio 19342 RTP/AVP 0 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------->
- --- (14 headers 12 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.201:5060 (NAT)
- Using INVITE request as basis request - 4331ba2852d8ae6562692e7b23d2b627@192.168.1.201:5060
- Found peer 'central1' for '2001' from 192.168.1.201:5060
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Reliably Transmitting (NAT) to 192.168.1.201:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0b161d10;received=192.168.1.201;rport=5060
- From: "2001" <sip:2001@192.168.1.201>;tag=as52f3269a
- To: <sip:3000@192.168.1.202>;tag=as30a5c3e0
- Call-ID: 4331ba2852d8ae6562692e7b23d2b627@192.168.1.201:5060
- 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="240d42bf"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '4331ba2852d8ae6562692e7b23d2b627@192.168.1.201:5060' in 6400 ms (Method: INVITE)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- ACK sip:3000@192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0b161d10;rport
- Max-Forwards: 70
- From: "2001" <sip:2001@192.168.1.201>;tag=as52f3269a
- To: <sip:3000@192.168.1.202>;tag=as30a5c3e0
- Contact: <sip:2001@192.168.1.201:5060>
- Call-ID: 4331ba2852d8ae6562692e7b23d2b627@192.168.1.201:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.8.21.0
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (10 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' Method: REGISTER
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-b777e59f
- 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: 2034 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="52fee18d",uri="sip:192.168.1.202",algorithm=MD5,response="fbe384097e45cc6c9ff7af8b2501bcbd"
- 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) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.9:5066 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-b777e59f;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as5da69f5c
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 2034 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="47dc249a"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' in 32000 ms (Method: REGISTER)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-aadbf618
- 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: 2035 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="47dc249a",uri="sip:192.168.1.202",algorithm=MD5,response="ac3cffd67af1243adbe8b1f3498cec38"
- 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) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.9:5066 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably 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=z9hG4bK1ca3ddd0;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as7ecc34ed
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 5d0ec6611a341ac6527741873bdb219e@192.168.1.202:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 19:52:03 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-aadbf618;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as5da69f5c
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 2035 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 19:52:03 GMT
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog '6a8de94131f8ec6c06ebb07f48b7fbc0@192.168.1.202:5060' in 6400 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably 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=z9hG4bK3ac65042;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as51db82b2
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 6a8de94131f8ec6c06ebb07f48b7fbc0@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)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=as7ecc34ed
- Call-ID: 5d0ec6611a341ac6527741873bdb219e@192.168.1.202:5060
- CSeq: 102 OPTIONS
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK1ca3ddd0
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (9 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '5d0ec6611a341ac6527741873bdb219e@192.168.1.202:5060' Method: OPTIONS
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=as51db82b2
- Call-ID: 6a8de94131f8ec6c06ebb07f48b7fbc0@192.168.1.202:5060
- CSeq: 102 NOTIFY
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK3ac65042
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (8 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '6a8de94131f8ec6c06ebb07f48b7fbc0@192.168.1.202:5060' Method: NOTIFY
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '4331ba2852d8ae6562692e7b23d2b627@192.168.1.201:5060' Method: ACK
- ^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-aafbe87a
- 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: 2869 NOTIFY
- Max-Forwards: 70
- Event: keep-alive
- User-Agent: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (10 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-aafbe87a;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: <sip:192.168.1.202>;tag=as0de98357
- Call-ID: 554ddcd8-5f93cf3@192.168.1.9
- CSeq: 2869 NOTIFY
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog '554ddcd8-5f93cf3@192.168.1.9' in 32000 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- INVITE sip:3000@192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK78b4b886;rport
- Max-Forwards: 70
- From: "2001" <sip:2001@192.168.1.201>;tag=as4d1e93e3
- To: <sip:3000@192.168.1.202>
- Contact: <sip:2001@192.168.1.201:5060>
- Call-ID: 6d5a1349711340ff0cf725641c6b61b3@192.168.1.201:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 16:06:57 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 262
- v=0
- o=root 1619176956 1619176956 IN IP4 192.168.1.201
- s=Asterisk PBX 1.8.21.0
- c=IN IP4 192.168.1.201
- t=0 0
- m=audio 12906 RTP/AVP 0 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------->
- --- (14 headers 12 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.201:5060 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KUsing INVITE request as basis request - 6d5a1349711340ff0cf725641c6b61b3@192.168.1.201:5060
- ^M^[[Kasterisk*CLI> ^M^[[0KFound peer 'central1' for '2001' from 192.168.1.201:5060
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Reliably Transmitting (NAT) to 192.168.1.201:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK78b4b886;received=192.168.1.201;rport=5060
- From: "2001" <sip:2001@192.168.1.201>;tag=as4d1e93e3
- To: <sip:3000@192.168.1.202>;tag=as79ad0df9
- Call-ID: 6d5a1349711340ff0cf725641c6b61b3@192.168.1.201:5060
- 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="6a5cadaf"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '6d5a1349711340ff0cf725641c6b61b3@192.168.1.201:5060' in 6400 ms (Method: INVITE)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- ACK sip:3000@192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK78b4b886;rport
- Max-Forwards: 70
- From: "2001" <sip:2001@192.168.1.201>;tag=as4d1e93e3
- To: <sip:3000@192.168.1.202>;tag=as79ad0df9
- Contact: <sip:2001@192.168.1.201:5060>
- Call-ID: 6d5a1349711340ff0cf725641c6b61b3@192.168.1.201:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.8.21.0
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (10 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.4:5060 --->
- INVITE sip:2001@192.168.1.202;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-8d332c180e126ab4-1---d8754z-
- Max-Forwards: 70
- Contact: <sip:3000@190.122.72.253:5060;transport=UDP>
- To: <sip:2001@192.168.1.202;transport=UDP>
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 1 INVITE
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Content-Type: application/sdp
- Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
- User-Agent: Zoiper rev.11137
- Allow-Events: presence, kpml
- Content-Length: 331
- v=0
- o=Zoiper_user 0 0 IN IP4 190.122.72.253
- s=Zoiper_session
- c=IN IP4 190.122.72.253
- t=0 0
- m=audio 8000 RTP/AVP 3 0 8 110 98 101
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:110 speex/8000
- a=rtpmap:98 iLBC/8000
- a=fmtp:98 mode=30
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=sendrecv
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (14 headers 15 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.4:5060 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KUsing INVITE request as basis request - MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- ^M^[[Kasterisk*CLI> ^M^[[0KFound peer '3000' for '3000' from 192.168.1.4:5060
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Reliably Transmitting (NAT) to 192.168.1.4:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-8d332c180e126ab4-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as7220bfcd
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 1 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="5f6962a4"
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog 'MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.' in 6400 ms (Method: INVITE)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-8d332c180e126ab4-1---d8754z-
- Max-Forwards: 70
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as7220bfcd
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 1 ACK
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (8 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.4:5060 --->
- INVITE sip:2001@192.168.1.202;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-e593059da3fc135b-1---d8754z-
- Max-Forwards: 70
- Contact: <sip:3000@190.122.72.253:5060;transport=UDP>
- To: <sip:2001@192.168.1.202;transport=UDP>
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 2 INVITE
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Content-Type: application/sdp
- Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
- User-Agent: Zoiper rev.11137
- Authorization: Digest username="3000",realm="asterisk",nonce="5f6962a4",uri="sip:2001@192.168.1.202;transport=UDP",response="a95748ad7c6f51cd2861cc5e57471507",algorith$
- Allow-Events: presence, kpml
- Content-Length: 331
- v=0
- o=Zoiper_user 0 0 IN IP4 190.122.72.253
- s=Zoiper_session
- c=IN IP4 190.122.72.253
- t=0 0
- m=audio 8000 RTP/AVP 3 0 8 110 98 101
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:110 speex/8000
- a=rtpmap:98 iLBC/8000
- a=fmtp:98 mode=30
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=sendrecv
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (15 headers 15 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.4:5060 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KUsing INVITE request as basis request - MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- ^M^[[Kasterisk*CLI> ^M^[[0KFound peer '3000' for '3000' from 192.168.1.4:5060
- ^M^[[Kasterisk*CLI> ^M^[[0K == Using SIP RTP CoS mark 5
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 3
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 0
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 8
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 110
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 98
- Found RTP audio format 101
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format GSM for ID 3
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format PCMU for ID 0
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format PCMA for ID 8
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format speex for ID 110
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format iLBC for ID 98
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format telephone-event for ID 101
- ^M^[[Kasterisk*CLI> ^M^[[0KCapabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 ($
- ^M^[[Kasterisk*CLI> ^M^[[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- ^M^[[Kasterisk*CLI> ^M^[[0KPeer audio RTP is at port 190.122.72.253:8000
- ^M^[[Kasterisk*CLI> ^M^[[0KLooking for 2001 in default (domain 192.168.1.202)
- ^M^[[Kasterisk*CLI> ^M^[[0Klist_route: hop: <sip:3000@190.122.72.253:5060;transport=UDP>
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Transmitting (NAT) to 192.168.1.4:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-e593059da3fc135b-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- To: <sip:2001@192.168.1.202;transport=UDP>
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 2 INVITE
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:2001@192.168.1.202:5060>
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Executing [2001@default:1] ^[[1;36mDial^[[0m("^[[1;35mSIP/3000-00000002^[[0m", "^[[1;35mSIP/central1/2001,30,Ttm^[[0m") in new stack
- ^M^[[Kasterisk*CLI> ^M^[[0K == Using SIP RTP CoS mark 5
- ^M^[[Kasterisk*CLI> ^M^[[0KAudio is at 19040
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x4 (ulaw) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x2 (gsm) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding non-codec 0x1 (telephone-event) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.201:5060:
- INVITE sip:2001@192.168.1.201 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK54e6d1ed;rport
- Max-Forwards: 70
- From: "3000" <sip:3000@192.168.1.202>;tag=as159b7043
- To: <sip:2001@192.168.1.201>
- Contact: <sip:3000@192.168.1.202:5060>
- Call-ID: 567bedc93a0015c720341f4b34953bcf@192.168.1.202:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 19:52:11 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 262
- v=0
- o=root 1934654958 1934654958 IN IP4 192.168.1.202
- s=Asterisk PBX 1.8.21.0
- c=IN IP4 192.168.1.202
- t=0 0
- m=audio 19040 RTP/AVP 0 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Called SIP/central1/2001
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Started music on hold, class 'default', on SIP/3000-00000002
- ^M^[[Kasterisk*CLI> ^M^[[0KAudio is at 12834
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x2 (gsm) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x4 (ulaw) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x8 (alaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Transmitting (NAT) to 192.168.1.4:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-e593059da3fc135b-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as7918e276
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 2 INVITE
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- ^M^[[Kasterisk*CLI> ^M^[[0KSupported: replaces, timer
- Contact: <sip:2001@192.168.1.202:5060>
- Content-Type: application/sdp
- Content-Length: 280
- v=0
- o=root 8187567 8187567 IN IP4 192.168.1.202
- s=Asterisk PBX 1.8.21.0
- c=IN IP4 192.168.1.202
- t=0 0
- m=audio 12834 RTP/AVP 3 0 8 101
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=z9hG4bK54e6d1ed;received=192.168.1.202;rport=5060
- From: "3000" <sip:3000@192.168.1.202>;tag=as159b7043
- To: <sip:2001@192.168.1.201>;tag=as17b3a9e2
- Call-ID: 567bedc93a0015c720341f4b34953bcf@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="180145f6"
- 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=z9hG4bK54e6d1ed;rport
- Max-Forwards: 70
- From: "3000" <sip:3000@192.168.1.202>;tag=as159b7043
- To: <sip:2001@192.168.1.201>;tag=as17b3a9e2
- Contact: <sip:3000@192.168.1.202:5060>
- Call-ID: 567bedc93a0015c720341f4b34953bcf@192.168.1.202:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.8.21.0
- Content-Length: 0
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K[Apr 29 16:52:11] ^[[1;33mNOTICE^[[0m[1585]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m20959^[[0m ^[[1;37mhandle_response_invite^[[0m: Failed to authe$
- -- SIP/central1-00000003 is circuit-busy
- ^M^[[Kasterisk*CLI> ^M^[[0K == Everyone is busy/congested at this time (1:0/1/0)
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Stopped music on hold on SIP/3000-00000002
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Auto fallthrough, channel 'SIP/3000-00000002' status is 'CONGESTION'
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-e593059da3fc135b-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as7918e276
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- 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
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '567bedc93a0015c720341f4b34953bcf@192.168.1.202:5060' Method: INVITE
- ^M^[[Kasterisk*CLI> ^M^[[0KRetransmitting #1 (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-e593059da3fc135b-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as7918e276
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- 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
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-e593059da3fc135b-1---d8754z-
- Max-Forwards: 70
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as7918e276
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 2 ACK
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- Really destroying SIP dialog 'MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.' Method: ACK
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-e593059da3fc135b-1---d8754z-
- Max-Forwards: 70
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as7918e276
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=3c1a1232
- Call-ID: MWI2MWI2ZmZjMjk3MjFiZDRmNmJjZWYzYmM3MzI0YTA.
- CSeq: 2 ACK
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '6d5a1349711340ff0cf725641c6b61b3@192.168.1.201:5060' Method: ACK
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.4:5060 --->
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.4:5060:
- OPTIONS sip:3000@190.122.72.253:5060;rinstance=13c51bb503937355;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK59714a59;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as5e59626c
- To: <sip:3000@190.122.72.253:5060;rinstance=13c51bb503937355;transport=UDP>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 3f75bf8346ab2f3202b140375fe290d6@192.168.1.202:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 19:52:17 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.4:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK59714a59;rport=5060
- Contact: <sip:192.168.1.4:5060>
- To: <sip:3000@190.122.72.253:5060;rinstance=13c51bb503937355;transport=UDP>;tag=e87d7932
- From: "asterisk"<sip:asterisk@192.168.1.202>;tag=as5e59626c
- Call-ID: 3f75bf8346ab2f3202b140375fe290d6@192.168.1.202:5060
- CSeq: 102 OPTIONS
- Accept: application/sdp, application/sdp
- Accept-Language: en
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
- User-Agent: Zoiper rev.11137
- Allow-Events: presence, kpml
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- Really destroying SIP dialog '3f75bf8346ab2f3202b140375fe290d6@192.168.1.202:5060' Method: OPTIONS
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- OPTIONS sip:192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK01896e29;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as5fe491d1
- To: <sip:192.168.1.202>
- Contact: <sip:asterisk@192.168.1.201:5060>
- Call-ID: 7417cc4626d792fa71ab679d2533c278@192.168.1.201:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 16:07:10 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- Looking for s in default (domain 192.168.1.202)
- <--- Transmitting (NAT) to 192.168.1.201:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK01896e29;received=192.168.1.201;rport=5060
- From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as5fe491d1
- To: <sip:192.168.1.202>;tag=as2e846901
- Call-ID: 7417cc4626d792fa71ab679d2533c278@192.168.1.201:5060
- CSeq: 102 OPTIONS
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:192.168.1.202:5060>
- Accept: application/sdp
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '7417cc4626d792fa71ab679d2533c278@192.168.1.201:5060' in 32000 ms (Method: OPTIONS)
- ^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-aebccf5d
- 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: 2870 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-aebccf5d;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: <sip:192.168.1.202>;tag=as0de98357
- Call-ID: 554ddcd8-5f93cf3@192.168.1.9
- CSeq: 2870 NOTIFY
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '554ddcd8-5f93cf3@192.168.1.9' in 32000 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' Method: REGISTER
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.201:5060:
- OPTIONS sip:192.168.1.201 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7216ad02;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as095c061a
- To: <sip:192.168.1.201>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 3f82b6fd0afd388d0d70725c69220bce@192.168.1.202:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 19:52:36 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7216ad02;received=192.168.1.202;rport=5060
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as095c061a
- To: <sip:192.168.1.201>;tag=as50deca58
- Call-ID: 3f82b6fd0afd388d0d70725c69220bce@192.168.1.202:5060
- CSeq: 102 OPTIONS
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:192.168.1.201:5060>
- Accept: application/sdp
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Really destroying SIP dialog '3f82b6fd0afd388d0d70725c69220bce@192.168.1.202:5060' Method: OPTIONS
- ^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-69a8e3b3
- 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: 2871 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-69a8e3b3;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: <sip:192.168.1.202>;tag=as0de98357
- Call-ID: 554ddcd8-5f93cf3@192.168.1.9
- CSeq: 2871 NOTIFY
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '554ddcd8-5f93cf3@192.168.1.9' in 32000 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-af30ebdd
- 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: 2036 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="47dc249a",uri="sip:192.168.1.202",algorithm=MD5,response="ac3cffd67af1243adbe8b1f3498cec38"
- 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-af30ebdd;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as7844c86c
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 2036 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="443639c1"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' in 32000 ms (Method: REGISTER)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-23b40d6a
- 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: 2037 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="443639c1",uri="sip:192.168.1.202",algorithm=MD5,response="a8bc29c58c21cd976201d49132702c2c"
- 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)
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably 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=z9hG4bK2b6da75e;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as70f3054d
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 15d4c0cf19a100d570a622ec3ccb6ab6@192.168.1.202:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 19:52:37 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-23b40d6a;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as7844c86c
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 2037 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 19:52:37 GMT
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog '2c2aacb2792f3444254e99d07e476960@192.168.1.202:5060' in 6400 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably 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=z9hG4bK5b915271;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as3ba17a97
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 2c2aacb2792f3444254e99d07e476960@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)
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' in 32000 ms (Method: REGISTER)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=as70f3054d
- Call-ID: 15d4c0cf19a100d570a622ec3ccb6ab6@192.168.1.202:5060
- CSeq: 102 OPTIONS
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK2b6da75e
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (9 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '15d4c0cf19a100d570a622ec3ccb6ab6@192.168.1.202:5060' Method: OPTIONS
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=as3ba17a97
- Call-ID: 2c2aacb2792f3444254e99d07e476960@192.168.1.202:5060
- CSeq: 102 NOTIFY
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK5b915271
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (8 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '2c2aacb2792f3444254e99d07e476960@192.168.1.202:5060' Method: NOTIFY
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.4:5060 --->
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '7417cc4626d792fa71ab679d2533c278@192.168.1.201:5060' Method: OPTIONS
- ^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-5fa17ae7
- 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: 2872 NOTIFY
- Max-Forwards: 70
- Event: keep-alive
- User-Agent: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (10 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-5fa17ae7;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: <sip:192.168.1.202>;tag=as0de98357
- Call-ID: 554ddcd8-5f93cf3@192.168.1.9
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog '554ddcd8-5f93cf3@192.168.1.9' in 32000 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.4:5060 --->
- INVITE sip:2001@192.168.1.202;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-f82eb1fb18a6ef1e-1---d8754z-
- Max-Forwards: 70
- Contact: <sip:3000@190.122.72.253:5060;transport=UDP>
- To: <sip:2001@192.168.1.202;transport=UDP>
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 1 INVITE
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Content-Type: application/sdp
- Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
- User-Agent: Zoiper rev.11137
- Allow-Events: presence, kpml
- Content-Length: 331
- v=0
- o=Zoiper_user 0 0 IN IP4 190.122.72.253
- s=Zoiper_session
- c=IN IP4 190.122.72.253
- t=0 0
- m=audio 8000 RTP/AVP 3 0 8 110 98 101
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:110 speex/8000
- a=rtpmap:98 iLBC/8000
- a=fmtp:98 mode=30
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=sendrecv
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (14 headers 15 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.4:5060 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KUsing INVITE request as basis request - OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- ^M^[[Kasterisk*CLI> ^M^[[0KFound peer '3000' for '3000' from 192.168.1.4:5060
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Reliably Transmitting (NAT) to 192.168.1.4:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-f82eb1fb18a6ef1e-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as063eae8c
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 1 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="3a178e56"
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog 'OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.' in 6400 ms (Method: INVITE)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-f82eb1fb18a6ef1e-1---d8754z-
- Max-Forwards: 70
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as063eae8c
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 1 ACK
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (8 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.4:5060 --->
- INVITE sip:2001@192.168.1.202;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-563fc751a1cb6e80-1---d8754z-
- Max-Forwards: 70
- Contact: <sip:3000@190.122.72.253:5060;transport=UDP>
- To: <sip:2001@192.168.1.202;transport=UDP>
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 2 INVITE
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Content-Type: application/sdp
- Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
- User-Agent: Zoiper rev.11137
- Authorization: Digest username="3000",realm="asterisk",nonce="3a178e56",uri="sip:2001@192.168.1.202;transport=UDP",response="6cc6bb98d3c82b2c179bc3bae4c09bd3",algorith$
- Allow-Events: presence, kpml
- Content-Length: 331
- v=0
- o=Zoiper_user 0 0 IN IP4 190.122.72.253
- s=Zoiper_session
- c=IN IP4 190.122.72.253
- t=0 0
- m=audio 8000 RTP/AVP 3 0 8 110 98 101
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:110 speex/8000
- a=rtpmap:98 iLBC/8000
- a=fmtp:98 mode=30
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=sendrecv
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (15 headers 15 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.4:5060 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KUsing INVITE request as basis request - OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- ^M^[[Kasterisk*CLI> ^M^[[0KFound peer '3000' for '3000' from 192.168.1.4:5060
- ^M^[[Kasterisk*CLI> ^M^[[0K == Using SIP RTP CoS mark 5
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 3
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 0
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 8
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 110
- ^M^[[Kasterisk*CLI> ^M^[[0KFound RTP audio format 98
- Found RTP audio format 101
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format GSM for ID 3
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format PCMU for ID 0
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format PCMA for ID 8
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format speex for ID 110
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format iLBC for ID 98
- ^M^[[Kasterisk*CLI> ^M^[[0KFound audio description format telephone-event for ID 101
- ^M^[[Kasterisk*CLI> ^M^[[0KCapabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 ($
- ^M^[[Kasterisk*CLI> ^M^[[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- ^M^[[Kasterisk*CLI> ^M^[[0KPeer audio RTP is at port 190.122.72.253:8000
- Looking for 2001 in default (domain 192.168.1.202)
- ^M^[[Kasterisk*CLI> ^M^[[0Klist_route: hop: <sip:3000@190.122.72.253:5060;transport=UDP>
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Transmitting (NAT) to 192.168.1.4:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-563fc751a1cb6e80-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- To: <sip:2001@192.168.1.202;transport=UDP>
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 2 INVITE
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:2001@192.168.1.202:5060>
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Executing [2001@default:1] ^[[1;36mDial^[[0m("^[[1;35mSIP/3000-00000004^[[0m", "^[[1;35mSIP/central1/2001,30,Ttm^[[0m") in new stack
- ^M^[[Kasterisk*CLI> ^M^[[0K == Using SIP RTP CoS mark 5
- ^M^[[Kasterisk*CLI> ^M^[[0KAudio is at 19026
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x4 (ulaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding non-codec 0x1 (telephone-event) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably Transmitting (NAT) to 192.168.1.201:5060:
- INVITE sip:2001@192.168.1.201 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK00c124db;rport
- Max-Forwards: 70
- From: "3000" <sip:3000@192.168.1.202>;tag=as221104e9
- To: <sip:2001@192.168.1.201>
- Contact: <sip:3000@192.168.1.202:5060>
- Call-ID: 692ee0184971534473676a4d1ed1569a@192.168.1.202:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 19:52:56 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 262
- v=0
- o=root 1848754928 1848754928 IN IP4 192.168.1.202
- s=Asterisk PBX 1.8.21.0
- c=IN IP4 192.168.1.202
- t=0 0
- m=audio 19026 RTP/AVP 0 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Called SIP/central1/2001
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Started music on hold, class 'default', on SIP/3000-00000004
- ^M^[[Kasterisk*CLI> ^M^[[0KAudio is at 16188
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x2 (gsm) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0KAdding non-codec 0x1 (telephone-event) to SDP
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Transmitting (NAT) to 192.168.1.4:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 190.122.72.253:5060;branch=z9hG4bK-d8754z-563fc751a1cb6e80-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as1208b427
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 2 INVITE
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- ^M^[[Kasterisk*CLI> ^M^[[0KSupported: replaces, timer
- Contact: <sip:2001@192.168.1.202:5060>
- Content-Type: application/sdp
- Content-Length: 286
- v=0
- o=root 1314216853 1314216853 IN IP4 192.168.1.202
- s=Asterisk PBX 1.8.21.0
- c=IN IP4 192.168.1.202
- t=0 0
- m=audio 16188 RTP/AVP 3 0 8 101
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=z9hG4bK00c124db;received=192.168.1.202;rport=5060
- From: "3000" <sip:3000@192.168.1.202>;tag=as221104e9
- To: <sip:2001@192.168.1.201>;tag=as5caa7d00
- Call-ID: 692ee0184971534473676a4d1ed1569a@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="55bd9d54"
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KTransmitting (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=z9hG4bK00c124db;rport
- Max-Forwards: 70
- From: "3000" <sip:3000@192.168.1.202>;tag=as221104e9
- To: <sip:2001@192.168.1.201>;tag=as5caa7d00
- Contact: <sip:3000@192.168.1.202:5060>
- Call-ID: 692ee0184971534473676a4d1ed1569a@192.168.1.202:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.8.21.0
- Content-Length: 0
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K[Apr 29 16:52:56] ^[[1;33mNOTICE^[[0m[1585]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m20959^[[0m ^[[1;37mhandle_response_invite^[[0m: Failed to authe$
- ^M^[[Kasterisk*CLI> ^M^[[0K -- SIP/central1-00000005 is circuit-busy
- ^M^[[Kasterisk*CLI> ^M^[[0K == Everyone is busy/congested at this time (1:0/1/0)
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Stopped music on hold on SIP/3000-00000004
- ^M^[[Kasterisk*CLI> ^M^[[0K -- Auto fallthrough, channel 'SIP/3000-00000004' status is 'CONGESTION'
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-563fc751a1cb6e80-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as1208b427
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- 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
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '692ee0184971534473676a4d1ed1569a@192.168.1.202:5060' Method: INVITE
- ^M^[[Kasterisk*CLI> ^M^[[0KRetransmitting #1 (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-563fc751a1cb6e80-1---d8754z-;received=192.168.1.4;rport=5060
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as1208b427
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- 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
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-563fc751a1cb6e80-1---d8754z-
- Max-Forwards: 70
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as1208b427
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 2 ACK
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (8 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog 'OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.' Method: ACK
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-563fc751a1cb6e80-1---d8754z-
- Max-Forwards: 70
- To: <sip:2001@192.168.1.202;transport=UDP>;tag=as1208b427
- From: "3000"<sip:3000@192.168.1.202;transport=UDP>;tag=4a3b2c2c
- Call-ID: OWUzMzQ1ODUxOWEzNTIwZWIzNGRiZWNlMDZhYTAxMjU.
- CSeq: 2 ACK
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (8 headers 0 lines) ---
- ^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-6291816f
- 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: 2873 NOTIFY
- Max-Forwards: 70
- Event: keep-alive
- User-Agent: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (10 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-6291816f;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: <sip:192.168.1.202>;tag=as0de98357
- Call-ID: 554ddcd8-5f93cf3@192.168.1.9
- CSeq: 2873 NOTIFY
- Server: Asterisk PBX 1.8.21.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog '554ddcd8-5f93cf3@192.168.1.9' in 32000 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- INVITE sip:3000@192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1dc38b11;rport
- Max-Forwards: 70
- From: "2001" <sip:2001@192.168.1.201>;tag=as39c3d829
- To: <sip:3000@192.168.1.202>
- Contact: <sip:2001@192.168.1.201:5060>
- Call-ID: 60133cf340722c0e3a120c4b500719dc@192.168.1.201:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 16:07:58 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 260
- v=0
- o=root 700161715 700161715 IN IP4 192.168.1.201
- s=Asterisk PBX 1.8.21.0
- c=IN IP4 192.168.1.201
- t=0 0
- m=audio 13830 RTP/AVP 0 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (14 headers 12 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.201:5060 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KUsing INVITE request as basis request - 60133cf340722c0e3a120c4b500719dc@192.168.1.201:5060
- ^M^[[Kasterisk*CLI> ^M^[[0KFound peer 'central1' for '2001' from 192.168.1.201:5060
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- Reliably Transmitting (NAT) to 192.168.1.201:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1dc38b11;received=192.168.1.201;rport=5060
- From: "2001" <sip:2001@192.168.1.201>;tag=as39c3d829
- To: <sip:3000@192.168.1.202>;tag=as4412c3a0
- Call-ID: 60133cf340722c0e3a120c4b500719dc@192.168.1.201:5060
- 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="58b2613d"
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog '60133cf340722c0e3a120c4b500719dc@192.168.1.201:5060' in 6400 ms (Method: INVITE)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- SIP read from UDP:192.168.1.201:5060 --->
- ACK sip:3000@192.168.1.202 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1dc38b11;rport
- Max-Forwards: 70
- From: "2001" <sip:2001@192.168.1.201>;tag=as39c3d829
- To: <sip:3000@192.168.1.202>;tag=as4412c3a0
- Contact: <sip:2001@192.168.1.201:5060>
- Call-ID: 60133cf340722c0e3a120c4b500719dc@192.168.1.201:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.8.21.0
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (10 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' Method: REGISTER
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-b99c969d
- 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: 2038 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="443639c1",uri="sip:192.168.1.202",algorithm=MD5,response="a8bc29c58c21cd976201d49132702c2c"
- 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
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (12 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.9:5066 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-b99c969d;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as7cb2b865
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 2038 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="5d021d94"
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' in 32000 ms (Method: REGISTER)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-f3fc469a
- 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: 2039 REGISTER
- Max-Forwards: 70
- Authorization: Digest username="3001",realm="asterisk",nonce="5d021d94",uri="sip:192.168.1.202",algorithm=MD5,response="2071f32c648f473ecfb6759d354ee0c4"
- 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
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (12 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KSending to 192.168.1.9:5066 (NAT)
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably 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=z9hG4bK1a3fe658;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as286bfa96
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 31376ca52343ac7757cc3da7677726a4@192.168.1.202:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 1.8.21.0
- Date: Mon, 29 Apr 2013 19:53:11 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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-f3fc469a;received=192.168.1.9;rport=5066
- From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
- To: "3001" <sip:3001@192.168.1.202>;tag=as7cb2b865
- Call-ID: ccf9627c-fa28d39f@192.168.1.9
- CSeq: 2039 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 19:53:11 GMT
- Content-Length: 0
- <------------>
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog '13dc6f2c37afc4fa1069ee847a71b6b0@192.168.1.202:5060' in 6400 ms (Method: NOTIFY)
- ^M^[[Kasterisk*CLI> ^M^[[0KReliably 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=z9hG4bK2a01de7b;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as43d41fc7
- To: <sip:3001@192.168.1.9:5066>
- Contact: <sip:asterisk@192.168.1.202:5060>
- Call-ID: 13dc6f2c37afc4fa1069ee847a71b6b0@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)
- ---
- ^M^[[Kasterisk*CLI> ^M^[[0KScheduling destruction of SIP dialog 'ccf9627c-fa28d39f@192.168.1.9' in 32000 ms (Method: REGISTER)
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=as286bfa96
- Call-ID: 31376ca52343ac7757cc3da7677726a4@192.168.1.202:5060
- CSeq: 102 OPTIONS
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK1a3fe658
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (9 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '31376ca52343ac7757cc3da7677726a4@192.168.1.202:5060' Method: OPTIONS
- ^M^[[Kasterisk*CLI> ^M^[[0K
- <--- 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=as43d41fc7
- Call-ID: 13dc6f2c37afc4fa1069ee847a71b6b0@192.168.1.202:5060
- CSeq: 102 NOTIFY
- Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK2a01de7b
- Server: Sipura/SPA941-4.1.8
- Content-Length: 0
- <------------->
- ^M^[[Kasterisk*CLI> ^M^[[0K--- (8 headers 0 lines) ---
- ^M^[[Kasterisk*CLI> ^M^[[0KReally destroying SIP dialog '13dc6f2c37afc4fa1069ee847a71b6b0@192.168.1.202:5060' Method: NOTIFY
- ^M^[[Kasterisk*CLI> sip set debug on^M^[[0KReally destroying SIP dialog '60133cf340722c0e3a120c4b500719dc@192.168.1.201:5060' Method: ACK
- ^M^[[Kasterisk*CLI> sip set debug on^H^[[K^H^[[Koff
- asterisk*CLI> ^M^[[0KSIP Debugging Disabled
- ^M^[[Kasterisk*CLI> exit
- Executing last minute cleanups
- Asterisk ending (0).
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