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mark mirasol
Dec 26, 2015, 6:16:21 AM12/26/15
to discuss-doubango
Hello guys,
I'm trying to setup sipml5 and webrtc2sip with vicibox. I'm able to connect to asterisk but for some reason after sipml5 connects to the conference there is no audio.
Here's what my console looks like. Please advise what needs to be done to get this working. Thank you.
linux-xosz:~ # webrtc2sip --config=/opt/webrtc2sip/sbin/config.xml
*******************************************************************
Copyright (C) 2012-2013 Doubango Telecom <http://www.doubango.org>
PRODUCT: webrtc2sip
HOME PAGE: http://webrtc2sip.org
LICENCE: GPLv3 or proprietary
VERSION: 2.6.0
'quit' to quit the application.
*******************************************************************
SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
*INFO: transport = udp://*:10061
*INFO: transport = ws://*:10061
*INFO: transport = wss://*:10062
*INFO: enable-rtp-symetric = yes
*INFO: enable-100rel = no
*INFO: enable-media-coder = yes
*INFO: enable-videojb = yes
*INFO: video-size-pref = vga
*INFO: rtp-buffsize = 65535
*INFO: avpf-tail-length = [100-400]
*INFO: srtp-mode = optional
*INFO: srtp-type = dtls
*INFO: dtmf-type = rfc4733
*INFO: codecs = pcmu
*INFO: UnRegister codec: PCMU, G.711u codec (native)
*INFO: codec-opus-maxrates = 48000;48000
*INFO: stun-server = stun.l.google.com;19302;-;-
*INFO: enable-icestun = yes
*INFO: max-fds = -1
*INFO: nameserver = 8.8.8.8
*INFO: ssl-certificates =
/home/cg/mycert/private/key.csr.server1.pem;
/home/cg/myca/certs/crt.server1.pem;
*;
no
*INFO: transport = c2c://*:10070
*INFO: transport = c2cs://*:10072
*INFO: database = sqlite;*
*INFO: sqlite3_threadsafe = 1
*INFO: Database opened = TRUE
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=8, pipeW=9
*INFO: Socket added[TCP/IPv4 transport]: fd=8, tail.count=1
*INFO: master fd=3
*INFO: Socket added[TCP/IPv4 transport]: fd=3, tail.count=2
*INFO: Transport::run() - enter
*INFO: Starting [TCP/IPv4 transport] server with IP {0.0.0.0} on port {10070} using fd {3} with type {9}...
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=10, pipeW=11
*INFO: Socket added[TLS/IPv4 transport]: fd=10, tail.count=1
*INFO: master fd=4
*INFO: Socket added[TLS/IPv4 transport]: fd=4, tail.count=2
*INFO: Stack running in SERVER mode
*INFO: tsk_timer_manager_start
*INFO: Transport::run() - enter
*INFO: Starting [TLS/IPv4 transport] server with IP {0.0.0.0} on port {10072} using fd {4} with type {17}...
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: SIP STACK::run -- START
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=15, pipeW=16
*INFO: Socket added[SIP transport]: fd=15, tail.count=1
*INFO: master fd=12
*INFO: Socket added[SIP transport]: fd=12, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=17, pipeW=18
*INFO: Transport::run() - enter
*INFO: Socket added[SIP transport]: fd=17, tail.count=1
*INFO: master fd=13
*INFO: Socket added[SIP transport]: fd=13, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: Transport::run() - enter
*INFO: pipeR fd=19, pipeW=20
*INFO: Socket added[SIP transport]: fd=19, tail.count=1
*INFO: master fd=14
*INFO: Socket added[SIP transport]: fd=14, tail.count=2
*INFO: Starting [SIP transport] server with IP {209.133.200.178} on port {10061} using fd {13} with type {64}...
*INFO: Transport::run() - enter
*INFO: Starting [SIP transport] server with IP {209.133.200.178} on port {10061} using fd {12} with type {2}...
*INFO: SIP STACK -- START
*INFO: Starting [SIP transport] server with IP {209.133.200.178} on port {10062} using fd {14} with type {128}...
*INFO: ioctlt(13), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=21)
*INFO: Socket added[SIP transport]: fd=21, tail.count=3
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 21
*INFO: #1 peers in the 'SIP transport' transport
*INFO: WebSocket Peer accepted/connected with fd = 21
*INFO: *** Stream Peer destroyed ***
*INFO: #0 peers in the 'SIP transport' transport
*INFO: #1 peers in the 'SIP transport' transport
*INFO: WebSocket handshake message: GET / HTTP/1.1
Host: 209.133.200.178:10061
Connection: Upgrade
Pragma: no-cache
Cache-Control: no-cache
Upgrade: websocket
Origin: http://www.doubango.org
Sec-WebSocket-Version: 13
User-Agent: Mozilla/5.0 (Windows NT 6.1) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/47.0.2526.106 Safari/537.36
Accept-Encoding: gzip, deflate, sdch
Accept-Language: en-US,en;q=0.8
Cookie: 25b97abf00fb1f974605272b1828266ewebphone_backup=39304ff562d00cf62f9775cceb8a54ddfb2d20699a9669af22318228022c174feb0fdaae89405d0a658a654908af9be16b1803f2af87e9d8d3663f67d2eeb3f5e8a34a00309a9d4748f3e8e41395c5be70963e290818fd0f881e313a22b3dcf17a182e7b6d569467706a6a14c3fe436af05c9280247f2a88e358869f4311a0997e91997d2e3e4bdb7844ba0f7ab9c0ea378f8cd7b289c344026c410c; __utma=149296821.162806626.1450923597.1451038963.1451089099.4; __utmc=149296821; __utmz=149296821.1450923597.1.1.utmcsr=(direct)|utmccn=(direct)|utmcmd=(none)
Sec-WebSocket-Key: SXcHOntmayPa4RTsE01+vw==
Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits
Sec-WebSocket-Protocol: sip
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;rport
From: "7002"<sip:70...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8
To: "7002"<sip:70...@127.0.0.1>
Contact: "7002"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214
CSeq: 21735 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO: Sending DNS query to "96.31.73.169"
*INFO: CloseSocket(22)
*INFO: CloseSocket(23)
*INFO: DNS NAPTR (127.0.0.1) query returned zero result
*INFO:
SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;rport
From: "7002"<sip:70...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8
To: "7002"<sip:70...@127.0.0.1>
Contact: "7002"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214
CSeq: 21735 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;ws-hacked=WS
*INFO:
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;ws-hacked=WS
From: "7002"<sip:70...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8
To: "7002"<sip:70...@127.0.0.1>;tag=as3e564bab
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214
CSeq: 21735 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d6d6f83"
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;rport
From: "7002"<sip:70...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8
To: "7002"<sip:70...@127.0.0.1>
Contact: "7002"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214
CSeq: 21736 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="6d6d6f83",uri="sip:asterisk",response="593b063652a10782f90c82f5496623e9",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO: Sending DNS query to "96.31.73.169"
*INFO: CloseSocket(22)
*INFO: CloseSocket(23)
*INFO: DNS NAPTR (127.0.0.1) query returned zero result
*INFO:
SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;rport
From: "7002"<sip:70...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8
To: "7002"<sip:70...@127.0.0.1>
Contact: "7002"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214
CSeq: 21736 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="6d6d6f83",uri="sip:asterisk",response="593b063652a10782f90c82f5496623e9",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;ws-hacked=WS
*INFO:
RECV:OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK305a6a22;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as4ddb6e06
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Sat, 26 Dec 2015 07:24:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;ws-hacked=WS
From: "7002"<sip:70...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8
To: "7002"<sip:70...@127.0.0.1>;tag=as3e564bab
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214
CSeq: 21736 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;expires=200
Date: Sat, 26 Dec 2015 07:24:26 GMT
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:
RECV:NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK11f013bd;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: no
Message-Account: sip:aste...@209.133.200.178
Voice-Message: 0/0 (0/0)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK305a6a22
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as4ddb6e06
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:
SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK305a6a22
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as4ddb6e06
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO:
RECV:NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK11f013bd;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: no
Message-Account: sip:aste...@209.133.200.178
Voice-Message: 0/0 (0/0)
*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK11f013bd
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 NOTIFY
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK11f013bd
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 NOTIFY
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO:
RECV:INVITE sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK78fc7efa;rport
Max-Forwards: 70
From: "ACagcW14511140941001100110011001" <sip:80080...@209.133.200.178>;tag=as31f6df12
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
Contact: <sip:80080...@209.133.200.178:5060>
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Sat, 26 Dec 2015 07:25:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "ACagcW14511140941001100110011001" <sip:80080...@209.133.200.178>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 214458807 214458807 IN IP4 209.133.200.178
s=Asterisk PBX 1.8.32.3-vici
c=IN IP4 209.133.200.178
t=0 0
m=audio 16076 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
*INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE
*INFO:
SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK78fc7efa
From: "ACagcW14511140941001100110011001"<sip:80080...@209.133.200.178>;tag=as31f6df12
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 INVITE
Content-Length: 0
*INFO: is_ice_active=0,
is_ro_hold_resume_changed=0,
is_ro_provisional_final_matching=0,
is_ro_media_lines_changed=0,
is_ro_network_info_changed=0,
is_ro_loopback_address=0,
is_media_type_changed=0,
is_ro_codecs_changed=0
*INFO: tdav_consumer_audio_init()
**WARN: function: "tdav_session_audio_ctor()"
file: "src/audio/tdav_session_audio.c"
line: "794"
MSG: No Audio denoiser found
*INFO: Create speekup jitter buffer
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: dtls.remote.setup=passive
*INFO: State machine: s0000_Started_2_Ringing_X_iINVITE
*INFO: State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
*INFO:
SEND: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK78fc7efa
From: "ACagcW14511140941001100110011001"<sip:80080...@209.133.200.178>;tag=as31f6df12
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;tag=1311098130
Contact: <sip:70...@209.133.200.178:10061;transport=udp>
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
***ERROR: function: "tsk_params_get_param_value()"
file: "src/tsk_params.c"
line: "219"
MSG: Invalid parameter
***ERROR: function: "tsk_params_get_param_value()"
file: "src/tsk_params.c"
line: "219"
MSG: Invalid parameter
*INFO: State machine: x0500_Current_2_Current_X_oINVITE
*INFO: tsk_timer_manager_start
*INFO: ICE CTX::run -- START
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: State machine: ICE_Started_2_GatheringHostCandidates_X_GatherHostCandidates
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports [209.133.200.178:59096]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = 209.133.200.178
*INFO: State machine: ICE_GatheringHostCandidates_2_GatheringHostCandidatesDone_X_Success
*INFO: ICE using STUN server: stun.l.google.com:19302
*INFO: ICE callback: Gathering host candidates succeed
*INFO: State machine: ICE_GatheringHostCandidatesDone_2_GatheringReflexiveCandidates_X_GatherReflexiveCandidates
*INFO: ICE reflexive candidates gathering ...0,500000
*INFO: Skipping redundant candidate address=209.133.200.178 and port=59097, fd=24, already_skipped(0)=no
*INFO: ICE reflexive candidates gathering ...1,0
*INFO: Skipping redundant candidate address=209.133.200.178 and port=59096, fd=25, already_skipped(1)=no
*INFO: srflx_addr_count_added=0, srflx_addr_count_skipped=2
*INFO: Candidate: K1ivuVRSOTdWoNf 1 udp 2130706431 209.133.200.178 59096 typ host
*INFO: Candidate: K1ivuVRSOTdWoNf 2 udp 2130706430 209.133.200.178 59097 typ host
*INFO: State machine: ICE_fsm_GatheringReflexiveCandidates_2_GatheringReflexiveCandidatesDone_X_Success
*INFO: ICE callback: Gathering reflexive candidates succeed
*INFO: State machine: ICE_Any_2_GatheringCompleted_X_GatheringComplet
*INFO: ICE callback: Gathering candidates completed
*INFO: State machine: c0000_Started_2_Outgoing_X_oINVITE
*INFO: tdav_consumer_audio_init()
**WARN: function: "tdav_session_audio_ctor()"
file: "src/audio/tdav_session_audio.c"
line: "794"
MSG: No Audio denoiser found
*INFO: Create speekup jitter buffer
*INFO: ICE enabled on RTP manager
*INFO: dtls.remote.setup=active
*INFO: Add call-id = 'b9fce8a6-9e99-4665-7041-4020ab1d6555' to peer with local fd = 21
*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 209.133.200.178:10061;rport=10061;branch=z9hG4bK-1962783856
From: <sip:80080...@209.133.200.178>;tag=969623956
Call-ID: b9fce8a6-9e99-4665-7041-4020ab1d6555
CSeq: 1682046286 INVITE
Content-Length: 0
*INFO: State machine: x0000_Any_2_Any_X_i1xx
*INFO:
RECV:OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK764533d1;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as47d3bbc4
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Sat, 26 Dec 2015 07:25:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK764533d1
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as47d3bbc4
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:
SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK764533d1
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as47d3bbc4
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
navaismo
Dec 28, 2015, 10:58:45 AM12/28/15
to discuss-doubango
Your ViciBox has an internal or external IP address? Also check the rtp debug of your asterisk to see the IP address to deliver the RTP(audio). And take a look over the STUN null workaround.
From: "7002"<sip:...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8To: "7002"<sip:...@127.0.0.1>Contact: "7002"<sip:7002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214CSeq: 21735 REGISTERContent-Length: 0Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>Max-Forwards: 70User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18Organization: Doubango TelecomSupported: path*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send*INFO: Sending DNS query to "96.31.73.169"*INFO: CloseSocket(22)*INFO: CloseSocket(23)*INFO: DNS NAPTR (127.0.0.1) query returned zero result*INFO:SEND: REGISTER sip:asterisk SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;rport
From: "7002"<sip:...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8To: "7002"<sip:...@127.0.0.1>
Contact: "7002"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"Call-ID: 2a606762-525a-4328-f4f5-60089afb5214CSeq: 21735 REGISTERContent-Length: 0Route: <sip:>Max-Forwards: 70User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18Organization: Doubango TelecomSupported: pathVia: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;ws-hacked=WS*INFO:RECV:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;received=209.133.200.178;rport=10061Via: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKPEI7OSg1i8SQm6Z1ZOvgJu9pe2i0FIWK;ws-hacked=WS
From: "7002"<sip:...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8To: "7002"<sip:...@127.0.0.1>;tag=as3e564bab
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214CSeq: 21735 REGISTERServer: Asterisk PBX 1.8.32.3-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d6d6f83"Content-Length: 0*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:asterisk SIP/2.0Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;rport
From: "7002"<sip:...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8To: "7002"<sip:...@127.0.0.1>Contact: "7002"<sip:7002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214CSeq: 21736 REGISTERContent-Length: 0Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>Max-Forwards: 70Authorization: Digest username="7002",realm="asterisk",nonce="6d6d6f83",uri="sip:asterisk",response="593b063652a10782f90c82f5496623e9",algorithm=MD5User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18Organization: Doubango TelecomSupported: path*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send*INFO: Sending DNS query to "96.31.73.169"*INFO: CloseSocket(22)*INFO: CloseSocket(23)*INFO: DNS NAPTR (127.0.0.1) query returned zero result*INFO:SEND: REGISTER sip:asterisk SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;rport
From: "7002"<sip:...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8To: "7002"<sip:...@127.0.0.1>
Contact: "7002"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"Call-ID: 2a606762-525a-4328-f4f5-60089afb5214CSeq: 21736 REGISTERContent-Length: 0Route: <sip:>Max-Forwards: 70Authorization: Digest username="7002",realm="asterisk",nonce="6d6d6f83",uri="sip:asterisk",response="593b063652a10782f90c82f5496623e9",algorithm=MD5User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18Organization: Doubango TelecomSupported: pathVia: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;ws-hacked=WS*INFO:RECV:OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK305a6a22;rportMax-Forwards: 70
From: "asterisk" <sip:as...@209.133.200.178>;tag=as4ddb6e06
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Sat, 26 Dec 2015 07:24:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send*INFO:RECV:SIP/2.0 200 OKVia: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;received=209.133.200.178;rport=10061Via: SIP/2.0/TCP 180.191.109.46:51047;rport;branch=z9hG4bKWIKyoU6Cg8qNsvBL4AJZfMvsIzC9IpxZ;ws-hacked=WS
From: "7002"<sip:...@127.0.0.1>;tag=SDmxp2Bh1UUWas9TAnf8To: "7002"<sip:...@127.0.0.1>;tag=as3e564bab
Call-ID: 2a606762-525a-4328-f4f5-60089afb5214CSeq: 21736 REGISTERServer: Asterisk PBX 1.8.32.3-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerExpires: 200Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;expires=200Date: Sat, 26 Dec 2015 07:24:26 GMTContent-Length: 0*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699*INFO:RECV:NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK11f013bd;rportMax-Forwards: 70
From: "asterisk" <sip:as...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFYUser-Agent: Asterisk PBX 1.8.32.3-viciEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 95Messages-Waiting: no
Message-Account: sip:as...@209.133.200.178
Voice-Message: 0/0 (0/0)*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 405 Method Not AllowedVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK305a6a22
From: "asterisk"<sip:as...@209.133.200.178>;tag=as4ddb6e06
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 OPTIONSContent-Length: 0*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699*INFO:SEND: SIP/2.0 405 Method Not AllowedVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK305a6a22
From: "asterisk"<sip:as...@209.133.200.178>;tag=as4ddb6e06
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 OPTIONSContent-Length: 0*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK*INFO: === NICT terminated ===*INFO: *** NICT destroyed ****INFO:RECV:NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK11f013bd;rportMax-Forwards: 70
From: "asterisk" <sip:as...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFYUser-Agent: Asterisk PBX 1.8.32.3-viciEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 95Messages-Waiting: no
Message-Account: sip:as...@209.133.200.178
Voice-Message: 0/0 (0/0)*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 481 Dialog/Transaction Does Not ExistVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK11f013bd
From: "asterisk"<sip:as...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 NOTIFYContent-Length: 0*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699*INFO:SEND: SIP/2.0 481 Dialog/Transaction Does Not ExistVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK11f013bd
From: "asterisk"<sip:as...@209.133.200.178>;tag=as1f06335e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 NOTIFYContent-Length: 0*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK*INFO: === NICT terminated ===*INFO: *** NICT destroyed ****INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK*INFO: === NICT terminated ===*INFO: *** NICT destroyed ****INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK*INFO: === NICT terminated ===*INFO: *** NICT destroyed ****INFO:RECV:INVITE sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK78fc7efa;rportMax-Forwards: 70
From: "ACagcW14511140941001100110011001" <sip:800...@209.133.200.178>;tag=as31f6df12
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>Contact: <sip:80080...@209.133.200.178:5060>
CSeq: 102 INVITEUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Sat, 26 Dec 2015 07:25:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timer
Remote-Party-ID: "ACagcW14511140941001100110011001" <sip:800...@209.133.200.178>;party=calling;privacy=off;screen=no
Content-Type: application/sdpContent-Length: 269v=0o=root 214458807 214458807 IN IP4 209.133.200.178s=Asterisk PBX 1.8.32.3-vicic=IN IP4 209.133.200.178t=0 0m=audio 16076 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv*INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE*INFO:SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK78fc7efa
From: "ACagcW14511140941001100110011001"<sip:800...@209.133.200.178>;tag=as31f6df12
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 INVITEContent-Length: 0*INFO: is_ice_active=0,is_ro_hold_resume_changed=0,is_ro_provisional_final_matching=0,is_ro_media_lines_changed=0,is_ro_network_info_changed=0,is_ro_loopback_address=0,is_media_type_changed=0,is_ro_codecs_changed=0*INFO: tdav_consumer_audio_init()**WARN: function: "tdav_session_audio_ctor()"file: "src/audio/tdav_session_audio.c"line: "794"MSG: No Audio denoiser found*INFO: Create speekup jitter buffer*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports*INFO: RTP/RTCP manager[End]: Trying to bind to random ports*INFO: dtls.remote.setup=passive*INFO: State machine: s0000_Started_2_Ringing_X_iINVITE*INFO: State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx*INFO:SEND: SIP/2.0 180 RingingVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK78fc7efa
From: "ACagcW14511140941001100110011001"<sip:800...@209.133.200.178>;tag=as31f6df12
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>;tag=1311098130Contact: <sip:70...@209.133.200.178:10061;transport=udp>
From: <sip:800...@209.133.200.178>;tag=969623956
Call-ID: b9fce8a6-9e99-4665-7041-4020ab1d6555CSeq: 1682046286 INVITEContent-Length: 0*INFO: State machine: x0000_Any_2_Any_X_i1xx*INFO:RECV:OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK764533d1;rportMax-Forwards: 70
From: "asterisk" <sip:as...@209.133.200.178>;tag=as47d3bbc4
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Sat, 26 Dec 2015 07:25:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 405 Method Not AllowedVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK764533d1
From: "asterisk"<sip:as...@209.133.200.178>;tag=as47d3bbc4
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
CSeq: 102 OPTIONSContent-Length: 0*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699*INFO:SEND: SIP/2.0 405 Method Not AllowedVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK764533d1
From: "asterisk"<sip:as...@209.133.200.178>;tag=as47d3bbc4
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51047;ws-src-proto=ws>
mark mirasol
Dec 29, 2015, 11:03:44 PM12/29/15
to discuss-doubango
Hello navaismo. Thank you for replying.
Our vicibox uses an external ip, 209.133.200.178.
Please see sipml5 screenshots.
For some reason, rtp debug shows the error "no such command". Please see attached screenshot.
I followed the setups in the link below to run the rtp debug.
Please advise.
Our instance is ViciBox Redux v.6.0.4 running on Asterisk 1.8.32.3-vici.
Thank you in advance.
...
mark mirasol
Dec 30, 2015, 12:19:14 AM12/30/15
to discuss-doubango
I set rtp debug on. This is what the asterisk log looks like.
linux-xosz:~ # asterisk -vvvvvvvvvR
Asterisk 1.8.32.3-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <mark...@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 1.8.32.3-vici currently running on linux-xosz (pid = 2051)
Verbosity is at least 21
[Dec 30 00:13:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:13:06] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:13:06] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:13:08]
<--- SIP read from UDP:209.133.200.178:10061 --->
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKbESA1hpiTBNN444tazkyir2ZN3vSqfig;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61874 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKbESA1hpiTBNN444tazkyir2ZN3vSqfig;ws-hacked=WS
<------------->
[Dec 30 00:13:08] --- (14 headers 0 lines) ---
[Dec 30 00:13:08] Sending to 209.133.200.178:10061 (NAT)
[Dec 30 00:13:08]
<--- Transmitting (NAT) to 209.133.200.178:10061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKbESA1hpiTBNN444tazkyir2ZN3vSqfig;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKbESA1hpiTBNN444tazkyir2ZN3vSqfig;ws-hacked=WS
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4da072fe
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61874 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53788be1"
Content-Length: 0
<------------>
[Dec 30 00:13:08] Scheduling destruction of SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' in 32000 ms (Method: REGISTER)
[Dec 30 00:13:09]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK41f64774;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as41a0ec20
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 15a3572104cf042d...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:13:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:13:09] --- (13 headers 0 lines) ---
[Dec 30 00:13:09] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:13:09]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK41f64774;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as41a0ec20
To: <sip:209.133.200.178>;tag=as10c2e485
Call-ID: 15a3572104cf042d...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:13:09] Scheduling destruction of SIP dialog '15a3572104cf042d...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:13:09]
<--- SIP read from UDP:209.133.200.178:10061 --->
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61875 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="53788be1",uri="sip:asterisk",response="6b46764aaf18d55af746111c413c87d6",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;ws-hacked=WS
<------------->
[Dec 30 00:13:09] --- (15 headers 0 lines) ---
[Dec 30 00:13:09] Sending to 209.133.200.178:10061 (NAT)
[Dec 30 00:13:09] -- Registered SIP '7002' at 209.133.200.178:10061
[Dec 30 00:13:09] Reliably Transmitting (NAT) to 209.133.200.178:10061:
OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK4b262a89;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as7db7041d
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:13:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:13:09]
<--- Transmitting (NAT) to 209.133.200.178:10061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;ws-hacked=WS
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4da072fe
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61875 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200
Date: Wed, 30 Dec 2015 05:13:09 GMT
Content-Length: 0
<------------>
[Dec 30 00:13:09] Scheduling destruction of SIP dialog '6f7fd43f2948d036...@209.133.200.178:5060' in 32000 ms (Method: NOTIFY)
[Dec 30 00:13:09] Reliably Transmitting (NAT) to 209.133.200.178:10061:
NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK601becb8;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as2e1c170c
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: no
Message-Account: sip:aste...@209.133.200.178
Voice-Message: 0/0 (0/0)
---
[Dec 30 00:13:09] Scheduling destruction of SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' in 32000 ms (Method: REGISTER)
[Dec 30 00:13:09]
<--- SIP read from UDP:209.133.200.178:10061 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4b262a89
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as7db7041d
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[Dec 30 00:13:09] --- (7 headers 0 lines) ---
[Dec 30 00:13:09] NOTICE[2545]: chan_sip.c:21672 handle_response_peerpoke: Peer '7002' is now Reachable. (434ms / 2000ms)
[Dec 30 00:13:09] Really destroying SIP dialog '539f965042ba384e...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:13:09] Retransmitting #1 (NAT) to 209.133.200.178:10061:
NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK601becb8;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as2e1c170c
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: no
Message-Account: sip:aste...@209.133.200.178
Voice-Message: 0/0 (0/0)
---
[Dec 30 00:13:10]
<--- SIP read from UDP:209.133.200.178:10061 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK601becb8
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as2e1c170c
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
[Dec 30 00:13:10] --- (7 headers 0 lines) ---
[Dec 30 00:13:10] Really destroying SIP dialog '6f7fd43f2948d036...@209.133.200.178:5060' Method: NOTIFY
[Dec 30 00:13:31] Really destroying SIP dialog '1197d94255e27395...@199.21.115.215' Method: OPTIONS
[Dec 30 00:13:35] Reliably Transmitting (NAT) to 208.74.75.250:5060:
OPTIONS sip:208.74.75.250 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK03babc65;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as64b1e129
To: <sip:208.74.75.250>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:13:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:13:35]
<--- SIP read from UDP:208.74.75.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK03babc65;rport=5060
To: <sip:208.74.75.250>
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as64b1e129
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[Dec 30 00:13:35] --- (7 headers 0 lines) ---
[Dec 30 00:13:35] Really destroying SIP dialog '67eda84c3d740a16...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:13:41] Really destroying SIP dialog '15a3572104cf042d...@199.21.115.215' Method: OPTIONS
[Dec 30 00:13:41] Really destroying SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' Method: REGISTER
[Dec 30 00:13:50] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:13:50] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:13:50] -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-00007917;2", "8600052,K") in new stack
[Dec 30 00:13:50] WARNING[9462]: app_meetme.c:4840 admin_exec: Conference number '8600052' not found!
[Dec 30 00:13:50] -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-00007917;2", "") in new stack
[Dec 30 00:13:50] == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-00007917;2'
[Dec 30 00:13:50] -- Executing [h@default:1] AGI("Local/55558600052@default-00007917;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Dec 30 00:13:50] -- <Local/55558600052@default-00007917;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Dec 30 00:13:51] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:13:51] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:13:56] Reliably Transmitting (NAT) to 66.148.120.167:5060:
OPTIONS sip:66.148.120.167 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK36afb48e;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as69c74b1c
To: <sip:66.148.120.167>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:13:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:13:56]
<--- SIP read from UDP:66.148.120.167:5060 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK36afb48e;rport
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as69c74b1c
To: <sip:66.148.120.167>;tag=301214151303
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0
<------------->
[Dec 30 00:13:56] --- (8 headers 0 lines) ---
[Dec 30 00:13:56] Really destroying SIP dialog '577719eb3264f74d...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:13:59]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK0c5d1747;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as6d3b1432
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 78c6786f2fc65d70...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:13:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:13:59] --- (13 headers 0 lines) ---
[Dec 30 00:13:59] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:13:59]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK0c5d1747;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as6d3b1432
To: <sip:209.133.200.178>;tag=as39cddcbe
Call-ID: 78c6786f2fc65d70...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:13:59] Scheduling destruction of SIP dialog '78c6786f2fc65d70...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:14:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:14:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:14:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:14:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:14:03] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:14:03] == Using SIP RTP CoS mark 5
[Dec 30 00:14:03] Audio is at 16912
[Dec 30 00:14:03] Adding codec 0x4 (ulaw) to SDP
[Dec 30 00:14:03] Adding codec 0x2 (gsm) to SDP
[Dec 30 00:14:03] Adding non-codec 0x1 (telephone-event) to SDP
[Dec 30 00:14:03] Reliably Transmitting (NAT) to 209.133.200.178:10061:
INVITE sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK4be00157;rport
Max-Forwards: 70
From: "ACagcW14514524391001100110011001" <sip:80080...@209.133.200.178>;tag=as40e46c0e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:80080...@209.133.200.178:5060>
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:14:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "ACagcW14514524391001100110011001" <sip:80080...@209.133.200.178>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 267
v=0
o=root 12243625 12243625 IN IP4 209.133.200.178
s=Asterisk PBX 1.8.32.3-vici
c=IN IP4 209.133.200.178
t=0 0
m=audio 16912 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
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4be00157
From: "ACagcW14514524391001100110011001"<sip:80080...@209.133.200.178>;tag=as40e46c0e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 INVITE
Content-Length: 0
<------------->
[Dec 30 00:14:03] --- (7 headers 0 lines) ---
[Dec 30 00:14:03]
<--- SIP read from UDP:209.133.200.178:10061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4be00157
From: "ACagcW14514524391001100110011001"<sip:80080...@209.133.200.178>;tag=as40e46c0e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345
Contact: <sip:70...@209.133.200.178:10061;transport=udp>
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
[Dec 30 00:14:03] --- (9 headers 0 lines) ---
[Dec 30 00:14:03] list_route: hop: <sip:70...@209.133.200.178:10061;transport=udp>
[Dec 30 00:14:06] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:14:06] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:14:09]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK752c4f5c;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as38ac5c8c
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 3abb6d644a73cc41...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:14:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:14:09] --- (13 headers 0 lines) ---
[Dec 30 00:14:09] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:14:09]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK752c4f5c;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as38ac5c8c
To: <sip:209.133.200.178>;tag=as773073d0
Call-ID: 3abb6d644a73cc41...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:14:09] Scheduling destruction of SIP dialog '3abb6d644a73cc41...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:14:09] Reliably Transmitting (NAT) to 209.133.200.178:10061:
OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK6261efd9;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as56f3d206
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:14:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK6261efd9
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as56f3d206
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[Dec 30 00:14:10] --- (7 headers 0 lines) ---
[Dec 30 00:14:10] Really destroying SIP dialog '692e9dda45864384...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:14:31] Really destroying SIP dialog '78c6786f2fc65d70...@199.21.115.215' Method: OPTIONS
[Dec 30 00:14:33] Scheduling destruction of SIP dialog '1b9c795737d74d56...@209.133.200.178:5060' in 27776 ms (Method: INVITE)
[Dec 30 00:14:33] Reliably Transmitting (NAT) to 209.133.200.178:10061:
CANCEL sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK4be00157;rport
Max-Forwards: 70
From: "ACagcW14514524391001100110011001" <sip:80080...@209.133.200.178>;tag=as40e46c0e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.32.3-vici
Content-Length: 0
---
[Dec 30 00:14:33] Scheduling destruction of SIP dialog '1b9c795737d74d56...@209.133.200.178:5060' in 27776 ms (Method: INVITE)
[Dec 30 00:14:33] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:14:33]
<--- SIP read from UDP:209.133.200.178:10061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4be00157
From: "ACagcW14514524391001100110011001"<sip:80080...@209.133.200.178>;tag=as40e46c0e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345
Contact: <sip:70...@209.133.200.178:10061;transport=udp>
CSeq: 102 CANCEL
Content-Length: 0
<------------->
[Dec 30 00:14:33] --- (8 headers 0 lines) ---
[Dec 30 00:14:33]
<--- SIP read from UDP:209.133.200.178:10061 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4be00157
From: "ACagcW14514524391001100110011001"<sip:80080...@209.133.200.178>;tag=as40e46c0e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=487; text="Request Cancelled"
<------------->
[Dec 30 00:14:33] --- (8 headers 0 lines) ---
[Dec 30 00:14:33] Transmitting (NAT) to 209.133.200.178:10061:
ACK sip:70...@209.133.200.178:10061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK4be00157;rport
Max-Forwards: 70
From: "ACagcW14514524391001100110011001" <sip:80080...@209.133.200.178>;tag=as40e46c0e
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345
Contact: <sip:80080...@209.133.200.178:5060>
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.3-vici
Content-Length: 0
---
[Dec 30 00:14:33] Scheduling destruction of SIP dialog '1b9c795737d74d56...@209.133.200.178:5060' in 27776 ms (Method: INVITE)
[Dec 30 00:14:35] Reliably Transmitting (NAT) to 208.74.75.250:5060:
OPTIONS sip:208.74.75.250 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK168cfe39;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as550b737a
To: <sip:208.74.75.250>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:14:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:14:35]
<--- SIP read from UDP:208.74.75.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK168cfe39;rport=5060
To: <sip:208.74.75.250>
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as550b737a
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[Dec 30 00:14:35] --- (7 headers 0 lines) ---
[Dec 30 00:14:35] Really destroying SIP dialog '6a8142882fd2e597...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:14:41] Really destroying SIP dialog '3abb6d644a73cc41...@199.21.115.215' Method: OPTIONS
[Dec 30 00:14:50]
<--- SIP read from UDP:209.133.200.178:10061 --->
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKg7HixXHaRpVwG99uhbtQHxER1un3PaO8;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61876 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="53788be1",uri="sip:asterisk",response="6b46764aaf18d55af746111c413c87d6",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKg7HixXHaRpVwG99uhbtQHxER1un3PaO8;ws-hacked=WS
<------------->
[Dec 30 00:14:50] --- (14 headers 0 lines) ---
[Dec 30 00:14:50] Sending to 209.133.200.178:10061 (NAT)
[Dec 30 00:14:50]
<--- Transmitting (NAT) to 209.133.200.178:10061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKg7HixXHaRpVwG99uhbtQHxER1un3PaO8;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKg7HixXHaRpVwG99uhbtQHxER1un3PaO8;ws-hacked=WS
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61876 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b735138"
Content-Length: 0
<------------>
[Dec 30 00:14:50] Scheduling destruction of SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' in 32000 ms (Method: REGISTER)
[Dec 30 00:14:50]
<--- SIP read from UDP:209.133.200.178:10061 --->
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61877 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="5b735138",uri="sip:asterisk",response="2fd81998744e2f2f9ad2b0af5bdc3062",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;ws-hacked=WS
<------------->
[Dec 30 00:14:50] --- (14 headers 0 lines) ---
[Dec 30 00:14:50] Sending to 209.133.200.178:10061 (NAT)
[Dec 30 00:14:50] Reliably Transmitting (NAT) to 209.133.200.178:10061:
OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK6bad4f26;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as6516a2a7
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:14:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:14:50]
<--- Transmitting (NAT) to 209.133.200.178:10061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;ws-hacked=WS
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61877 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200
Date: Wed, 30 Dec 2015 05:14:50 GMT
Content-Length: 0
<------------>
[Dec 30 00:14:50] Scheduling destruction of SIP dialog '331ba133294209f7...@209.133.200.178:5060' in 35136 ms (Method: NOTIFY)
[Dec 30 00:14:50] Reliably Transmitting (NAT) to 209.133.200.178:10061:
NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK660f8e63;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as1d032a79
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: no
Message-Account: sip:aste...@209.133.200.178
Voice-Message: 0/0 (0/0)
---
[Dec 30 00:14:50] Scheduling destruction of SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' in 32000 ms (Method: REGISTER)
[Dec 30 00:14:50]
<--- SIP read from UDP:209.133.200.178:10061 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK6bad4f26
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as6516a2a7
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[Dec 30 00:14:50] --- (7 headers 0 lines) ---
[Dec 30 00:14:50] Really destroying SIP dialog '3521e224118b364a...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:14:50] Retransmitting #1 (NAT) to 209.133.200.178:10061:
NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK660f8e63;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as1d032a79
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: no
Message-Account: sip:aste...@209.133.200.178
Voice-Message: 0/0 (0/0)
---
[Dec 30 00:14:50]
<--- SIP read from UDP:209.133.200.178:10061 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK660f8e63
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as1d032a79
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
[Dec 30 00:14:50] --- (7 headers 0 lines) ---
[Dec 30 00:14:50] Really destroying SIP dialog '331ba133294209f7...@209.133.200.178:5060' Method: NOTIFY
[Dec 30 00:14:55]
<--- SIP read from UDP:209.133.200.178:10061 --->
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bK593udChhO9hbO5y7SwDlOEHLTUd0ik91;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=0;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61878 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="5b735138",uri="sip:asterisk",response="2fd81998744e2f2f9ad2b0af5bdc3062",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bK593udChhO9hbO5y7SwDlOEHLTUd0ik91;ws-hacked=WS
<------------->
[Dec 30 00:14:55] --- (14 headers 0 lines) ---
[Dec 30 00:14:55] Sending to 209.133.200.178:10061 (NAT)
[Dec 30 00:14:55] NOTICE[2545]: chan_sip.c:15062 check_auth: Correct auth, but based on stale nonce received from '"mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL'
[Dec 30 00:14:55]
<--- Transmitting (NAT) to 209.133.200.178:10061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bK593udChhO9hbO5y7SwDlOEHLTUd0ik91;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bK593udChhO9hbO5y7SwDlOEHLTUd0ik91;ws-hacked=WS
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61878 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1de80666", stale=true
Content-Length: 0
<------------>
[Dec 30 00:14:55] Scheduling destruction of SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' in 32000 ms (Method: REGISTER)
[Dec 30 00:14:56]
<--- SIP read from UDP:209.133.200.178:10061 --->
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKlnxpRMcnS3WPWTTm7E2UX1EvBqn2bWmM;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=0;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61879 REGISTER
Content-Length: 0
Route: <sip:>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="1de80666",uri="sip:asterisk",response="a201c070a2f4cc50b52e19d93a6632b1",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKlnxpRMcnS3WPWTTm7E2UX1EvBqn2bWmM;ws-hacked=WS
<------------->
[Dec 30 00:14:56] --- (14 headers 0 lines) ---
[Dec 30 00:14:56] Sending to 209.133.200.178:10061 (NAT)
[Dec 30 00:14:56] -- Unregistered SIP '7002'
[Dec 30 00:14:56]
<--- Transmitting (NAT) to 209.133.200.178:10061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKlnxpRMcnS3WPWTTm7E2UX1EvBqn2bWmM;received=209.133.200.178;rport=10061
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKlnxpRMcnS3WPWTTm7E2UX1EvBqn2bWmM;ws-hacked=WS
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61879 REGISTER
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 0
Date: Wed, 30 Dec 2015 05:14:56 GMT
Content-Length: 0
<------------>
[Dec 30 00:14:56] Scheduling destruction of SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' in 32000 ms (Method: REGISTER)
[Dec 30 00:14:56] Reliably Transmitting (NAT) to 66.148.120.167:5060:
OPTIONS sip:66.148.120.167 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK6a0f4991;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as3b7230b1
To: <sip:66.148.120.167>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:14:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:14:56]
<--- SIP read from UDP:66.148.120.167:5060 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK6a0f4991;rport
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as3b7230b1
To: <sip:66.148.120.167>;tag=301215151303
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0
<------------->
[Dec 30 00:14:56] --- (8 headers 0 lines) ---
[Dec 30 00:14:56] Really destroying SIP dialog '11d647f016879652...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:14:58] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:14:58] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:14:58] -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-00007918;2", "8600052,K") in new stack
[Dec 30 00:14:58] WARNING[9595]: app_meetme.c:4840 admin_exec: Conference number '8600052' not found!
[Dec 30 00:14:58] -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-00007918;2", "") in new stack
[Dec 30 00:14:58] == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-00007918;2'
[Dec 30 00:14:58] -- Executing [h@default:1] AGI("Local/55558600052@default-00007918;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Dec 30 00:14:58] -- <Local/55558600052@default-00007918;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Dec 30 00:14:59] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:14:59] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:14:59]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK50805ab7;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as0d72573b
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 17eee3d364f31211...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:14:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:14:59] --- (13 headers 0 lines) ---
[Dec 30 00:14:59] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:14:59]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK50805ab7;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as0d72573b
To: <sip:209.133.200.178>;tag=as1dfeefa2
Call-ID: 17eee3d364f31211...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:14:59] Scheduling destruction of SIP dialog '17eee3d364f31211...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:15:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:15:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:15:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:15:01] Really destroying SIP dialog '1b9c795737d74d56...@209.133.200.178:5060' Method: INVITE
[Dec 30 00:15:02] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:15:06] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:15:06] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:15:09]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK51e8bac3;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as4cd7071e
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 7194b0a20e5c01e2...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:15:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:15:09] --- (13 headers 0 lines) ---
[Dec 30 00:15:09] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:15:09]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK51e8bac3;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as4cd7071e
To: <sip:209.133.200.178>;tag=as0c0513c4
Call-ID: 7194b0a20e5c01e2...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:15:09] Scheduling destruction of SIP dialog '7194b0a20e5c01e2...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:15:13]
<--- SIP read from UDP:209.126.122.39:5074 --->
INVITE sip:9009725...@209.133.200.178 SIP/2.0
To: 900972599715009<sip:9009725...@209.133.200.178>
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;rport
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Contact: <sip:1...@209.126.122.39:5074>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE
User-Agent: sipcli/v1.8
Content-Type: application/sdp
Content-Length: 284
v=0
o=sipcli-Session 117923307 1062610022 IN IP4 209.126.122.39
s=sipcli
c=IN IP4 209.126.122.39
t=0 0
m=audio 5075 RTP/AVP 18 0 8 101
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
[Dec 30 00:15:13] --- (12 headers 13 lines) ---
[Dec 30 00:15:13] Sending to 209.126.122.39:5074 (NAT)
[Dec 30 00:15:13] Using INVITE request as basis request - d8686922fe4165fef523d8d188793619
[Dec 30 00:15:13] No matching peer for '100' from '209.126.122.39:5074'
[Dec 30 00:15:13]
<--- Reliably Transmitting (NAT) to 209.126.122.39:5074 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
<------------>
[Dec 30 00:15:13] Scheduling destruction of SIP dialog 'd8686922fe4165fef523d8d188793619' in 32000 ms (Method: INVITE)
[Dec 30 00:15:13] Retransmitting #1 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:14] Retransmitting #2 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:16] Retransmitting #3 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:20] Retransmitting #4 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:24] Retransmitting #5 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:28] Really destroying SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' Method: REGISTER
[Dec 30 00:15:28] Retransmitting #6 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:29]
<--- SIP read from UDP:185.40.4.191:5071 --->
INVITE sip:0054417...@209.133.200.178 SIP/2.0
To: 005441729810030<sip:0054417...@209.133.200.178>
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;rport
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Contact: <sip:21...@185.40.4.191:5071>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE
User-Agent: sipcli/v1.8
Content-Type: application/sdp
Content-Length: 279
v=0
o=sipcli-Session 628521815 862000385 IN IP4 185.40.4.191
s=sipcli
c=IN IP4 185.40.4.191
t=0 0
m=audio 5073 RTP/AVP 18 0 8 101
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
[Dec 30 00:15:29] --- (12 headers 13 lines) ---
[Dec 30 00:15:29] Sending to 185.40.4.191:5071 (NAT)
[Dec 30 00:15:29] Using INVITE request as basis request - 808de618525583c867f4f4c0a2e7e13c
[Dec 30 00:15:29] No matching peer for '2101' from '185.40.4.191:5071'
[Dec 30 00:15:29]
<--- Reliably Transmitting (NAT) to 185.40.4.191:5071 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
<------------>
[Dec 30 00:15:29] Scheduling destruction of SIP dialog '808de618525583c867f4f4c0a2e7e13c' in 32000 ms (Method: INVITE)
[Dec 30 00:15:30] Retransmitting #1 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:31] Retransmitting #2 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:31] Really destroying SIP dialog '17eee3d364f31211...@199.21.115.215' Method: OPTIONS
[Dec 30 00:15:32] Retransmitting #7 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:33] Retransmitting #3 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:35] Reliably Transmitting (NAT) to 208.74.75.250:5060:
OPTIONS sip:208.74.75.250 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK328f1331;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as1c6f22c2
To: <sip:208.74.75.250>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:15:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:15:35]
<--- SIP read from UDP:208.74.75.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK328f1331;rport=5060
To: <sip:208.74.75.250>
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as1c6f22c2
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[Dec 30 00:15:35] --- (7 headers 0 lines) ---
[Dec 30 00:15:35] Really destroying SIP dialog '3b805f2a61e4d81c...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:15:36] > Refreshing DNS lookups.
[Dec 30 00:15:36] Retransmitting #8 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:37] Retransmitting #4 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:40] Retransmitting #9 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:41] Retransmitting #5 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:41] Really destroying SIP dialog '7194b0a20e5c01e2...@199.21.115.215' Method: OPTIONS
[Dec 30 00:15:44] Retransmitting #10 (NAT) to 209.126.122.39:5074:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.126.122.39:5074;branch=z9hG4bK-d8686922fe4165fef523d8d188793619;received=209.126.122.39;rport=5074
From: 100<sip:1...@209.133.200.178>;tag=fba51cea
To: 900972599715009<sip:9009725...@209.133.200.178>;tag=as35f88e1d
Call-ID: d8686922fe4165fef523d8d188793619
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58b3ae2f"
Content-Length: 0
---
[Dec 30 00:15:45] WARNING[2545]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission d8686922fe4165fef523d8d188793619 for seqno 1 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Dec 30 00:15:45] WARNING[2545]: chan_sip.c:3884 retrans_pkt: Timeout on d8686922fe4165fef523d8d188793619 on non-critical invite transaction.
[Dec 30 00:15:45] Really destroying SIP dialog 'd8686922fe4165fef523d8d188793619' Method: INVITE
[Dec 30 00:15:45] Retransmitting #6 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:49] Retransmitting #7 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:53] Retransmitting #8 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:15:56] Reliably Transmitting (NAT) to 66.148.120.167:5060:
OPTIONS sip:66.148.120.167 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK2ce80190;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as768dfd94
To: <sip:66.148.120.167>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:15:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:15:56]
<--- SIP read from UDP:66.148.120.167:5060 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK2ce80190;rport
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as768dfd94
To: <sip:66.148.120.167>;tag=301216151303
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0
<------------->
[Dec 30 00:15:56] --- (8 headers 0 lines) ---
[Dec 30 00:15:56] Really destroying SIP dialog '66f21fa93f2a2cb2...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:15:57] Retransmitting #9 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:16:00]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK28ca0c85;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as010d7ae5
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 78dcaf743ad03040...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:16:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:16:00] --- (13 headers 0 lines) ---
[Dec 30 00:16:00] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:16:00]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK28ca0c85;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as010d7ae5
To: <sip:209.133.200.178>;tag=as44eb8574
Call-ID: 78dcaf743ad03040...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:16:00] Scheduling destruction of SIP dialog '78dcaf743ad03040...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:16:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:16:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:16:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:16:01] Retransmitting #10 (NAT) to 185.40.4.191:5071:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.40.4.191:5071;branch=z9hG4bK-808de618525583c867f4f4c0a2e7e13c;received=185.40.4.191;rport=5071
From: 2101<sip:21...@209.133.200.178>;tag=1fc6bfc6
To: 005441729810030<sip:0054417...@209.133.200.178>;tag=as6dd53fec
Call-ID: 808de618525583c867f4f4c0a2e7e13c
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="704360f0"
Content-Length: 0
---
[Dec 30 00:16:01] WARNING[2545]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission 808de618525583c867f4f4c0a2e7e13c for seqno 1 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Dec 30 00:16:01] WARNING[2545]: chan_sip.c:3884 retrans_pkt: Timeout on 808de618525583c867f4f4c0a2e7e13c on non-critical invite transaction.
[Dec 30 00:16:02] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:16:06] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:16:06] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:16:09]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK650bd48c;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as08d8235a
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 64c93f656403fa58...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:16:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:16:09] --- (13 headers 0 lines) ---
[Dec 30 00:16:09] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:16:09]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK650bd48c;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as08d8235a
To: <sip:209.133.200.178>;tag=as5f6d6e27
Call-ID: 64c93f656403fa58...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:16:09] Scheduling destruction of SIP dialog '64c93f656403fa58...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:16:32] Really destroying SIP dialog '78dcaf743ad03040...@199.21.115.215' Method: OPTIONS
[Dec 30 00:16:33] Really destroying SIP dialog '808de618525583c867f4f4c0a2e7e13c' Method: INVITE
[Dec 30 00:16:35] Reliably Transmitting (NAT) to 208.74.75.250:5060:
OPTIONS sip:208.74.75.250 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK56310e77;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as75838008
To: <sip:208.74.75.250>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:16:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:16:35]
<--- SIP read from UDP:208.74.75.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK56310e77;rport=5060
To: <sip:208.74.75.250>
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as75838008
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[Dec 30 00:16:35] --- (7 headers 0 lines) ---
[Dec 30 00:16:35] Really destroying SIP dialog '0b5426197c7b81e4...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:16:41] Really destroying SIP dialog '64c93f656403fa58...@199.21.115.215' Method: OPTIONS
[Dec 30 00:16:56] Reliably Transmitting (NAT) to 66.148.120.167:5060:
OPTIONS sip:66.148.120.167 SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK199381c5;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as10ff819e
To: <sip:66.148.120.167>
Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: Wed, 30 Dec 2015 05:16:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Dec 30 00:16:56]
<--- SIP read from UDP:66.148.120.167:5060 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK199381c5;rport
From: "asterisk" <sip:aste...@209.133.200.178>;tag=as10ff819e
To: <sip:66.148.120.167>;tag=301217151303
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0
<------------->
[Dec 30 00:16:56] --- (8 headers 0 lines) ---
[Dec 30 00:16:56] Really destroying SIP dialog '03ae19bd5c6beef9...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:17:00]
<--- SIP read from UDP:199.21.115.215:5060 --->
OPTIONS sip:209.133.200.178 SIP/2.0
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK006d7840;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as4b84b602
To: <sip:209.133.200.178>
Contact: <sip:aste...@199.21.115.215>
Call-ID: 01a9d6615468dd19...@199.21.115.215
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Wed, 30 Dec 2015 05:17:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
[Dec 30 00:17:00] --- (13 headers 0 lines) ---
[Dec 30 00:17:00] Looking for s in trunkinbound (domain 209.133.200.178)
[Dec 30 00:17:00]
<--- Transmitting (NAT) to 199.21.115.215:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK006d7840;received=199.21.115.215;rport=5060
From: "asterisk" <sip:aste...@199.21.115.215>;tag=as4b84b602
To: <sip:209.133.200.178>;tag=as5dd62f96
Call-ID: 01a9d6615468dd19...@199.21.115.215
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
[Dec 30 00:17:00] Scheduling destruction of SIP dialog '01a9d6615468dd19...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:17:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:17:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 30 00:17:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 30 00:17:02] == Manager 'sendcron' logged off from 127.0.0.1
linux-xosz*CLI>
mark mirasol
Dec 30, 2015, 12:30:13 AM12/30/15
to discuss-doubango
And on chrome, this is what the js console looks like.
This appears to be Chrome
SIPml-api.js:1 SIPML5 API version = 2.0.2
call.htm:161 location=http://209.133.200.178/agc-dev/webphone/sipml5/call.htm
SIPml-api.js:1 'webkitURL' is deprecated. Please use 'URL' instead.
SIPml-api.js:1 User-Agent=Mozilla/5.0 (Windows NT 6.1) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/47.0.2526.106 Safari/537.36
SIPml-api.js:1 WebSocket supported = yes
SIPml-api.js:1 Navigator friendly name = chrome
SIPml-api.js:1 OS friendly name = windows
SIPml-api.js:1 Have WebRTC = yes
SIPml-api.js:1 Have GUM = yes
SIPml-api.js:1 Engine initialized
SIPml-api.js:1 s_websocket_server_url=ws://209.133.200.178:10061
SIPml-api.js:1 s_sip_outboundproxy_url=udp://209.133.200.178
SIPml-api.js:1 b_rtcweb_breaker_enabled=yes
SIPml-api.js:1 b_click2call_enabled=no
SIPml-api.js:1 b_early_ims=yes
SIPml-api.js:1 b_enable_media_stream_cache=no
SIPml-api.js:1 o_bandwidth={}
SIPml-api.js:1 o_video_size={}
SIPml-api.js:1 SIP stack start: proxy='ns313841.ovh.net:12062', realm='<sip:asterisk>', impi='7002', impu='"mark"<sip:70...@209.133.200.178>'
SIPml-api.js:1 Connecting to 'ws://209.133.200.178:10061'
SIPml-api.js:1 ==stack event = starting
SIPml-api.js:1 __tsip_transport_ws_onopen
SIPml-api.js:1 ==stack event = started
SIPml-api.js:1 State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js:1 SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKbESA1hpiTBNN444tazkyir2ZN3vSqfig;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61874 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
SIPml-api.js:1 ==session event = connecting
SIPml-api.js:1 ==session event = sent_request
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.133.200.178:10061;rport=10061;received=209.133.200.178;branch=z9hG4bKbESA1hpiTBNN444tazkyir2ZN3vSqfig
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4da072fe
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61874 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKbESA1hpiTBNN444tazkyir2ZN3vSqfig;ws-hacked=WS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="53788be1",stale=FALSE,algorithm=MD5
SIPml-api.js:1 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js:1 SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61875 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="53788be1",uri="sip:asterisk",response="6b46764aaf18d55af746111c413c87d6",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
Supported: path
SIPml-api.js:1 ==session event = sent_request
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=ws;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4b262a89
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as7db7041d
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:10061;transport=ws>
Max-Forwards: 70
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: 30 Dec 2015 5:13:9 GMT;30
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
SIPml-api.js:1 Not implementedtsk_utils_log_error @ SIPml-api.js:1
SIPml-api.js:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4b262a89
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as7db7041d
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:10061;rport=10061;received=209.133.200.178;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4da072fe
Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61875 REGISTER
Expires: 200
Content-Length: 0
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;ws-hacked=WS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 30 Dec 2015 5:13:9 GMT;30
SIPml-api.js:1 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=ws;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK601becb8
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as2e1c170c
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:10061;transport=ws>
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 95
Max-Forwards: 70
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
SIPml-api.js:1 SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK601becb8
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as2e1c170c
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 NOTIFY
Content-Length: 0
SIPml-api.js:1 ==session event = connected
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=INVITE sip:70...@209.133.200.178:10061 SIP/2.0
Via: SIP/2.0/WS 209.133.200.178:10061;rport;branch=z9hG4bK-1184506698
From: <sip:80080...@209.133.200.178>;tag=754980512
Contact: <sip:8008093395@209.133.200.178:10061;transport=ws>
Call-ID: 5bc51625-b8bc-6240-fcae-bfaeec8fe773
CSeq: 523907086 INVITE
Content-Type: application/sdp
Content-Length: 1074
Max-Forwards: 70
Route: <sip:180.191.109.46:51164;transport=ws;lr>
User-Agent: webrtc2sip Media Server 2.6.0
v=0
o=doubango 1983 678901 IN IP4 209.133.200.178
s=-
c=IN IP4 209.133.200.178
t=0 0
a=acap:1 setup:actpass
a=acap:2 connection:new
a=acap:3 fingerprint:sha-256 39:DD:7A:AA:64:6B:C1:68:C1:C9:67:FE:94:D8:49:D3:A7:3B:3C:B7:04:FE:9A:73:A8:AD:13:E6:56:15:26:C3
a=acap:4 fingerprint:sha-1 90:86:5F:BB:99:0F:37:2F:62:1E:65:18:00:47:3D:AF:5C:F2:43:F7
a=tcap:1 UDP/TLS/RTP/SAVPF UDP/TLS/RTP/SAVP RTP/AVPF
m=audio 10652 RTP/AVP 0 101
c=IN IP4 209.133.200.178
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=pcfg:1 t=1 a=1,2,4|3
a=pcfg:2 t=2 a=1,2,4|3
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:3477872752 cname:c117547c0edf0b0d796c1a6212440c15
a=ssrc:3477872752 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3477872752 label:doubango@audio
a=ice-ufrag:F1EINniGs44Y9ow
a=ice-pwd:e3Gz08NPhIRLvkGC6NYWPE
a=candidate:FfrHF4ZoP2Z1pTl 1 udp 2130706431 209.133.200.178 10652 typ host
a=candidate:FfrHF4ZoP2Z1pTl 2 udp 2130706430 209.133.200.178 10653 typ host
SIPml-api.js:1 State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE
SIPml-api.js:1 SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 209.133.200.178:10061;rport=10061;branch=z9hG4bK-1184506698
From: <sip:80080...@209.133.200.178>;tag=754980512
Call-ID: 5bc51625-b8bc-6240-fcae-bfaeec8fe773
CSeq: 523907086 INVITE
Content-Length: 0
SIPml-api.js:1 ICE servers:[{"url":"stun:null"}]
SIPml-api.js:1 setRemoteDescription(offer)
v=0
o=doubango 1983 678901 IN IP4 209.133.200.178
s=-
c=IN IP4 209.133.200.178
t=0 0
a=setup:actpass
a=connection:new
a=fingerprint:sha-256 39:DD:7A:AA:64:6B:C1:68:C1:C9:67:FE:94:D8:49:D3:A7:3B:3C:B7:04:FE:9A:73:A8:AD:13:E6:56:15:26:C3
a=acap:4 fingerprint:sha-1 90:86:5F:BB:99:0F:37:2F:62:1E:65:18:00:47:3D:AF:5C:F2:43:F7
a=tcap:1 UDP/TLS/RTP/SAVPF UDP/TLS/RTP/SAVP RTP/AVPF
m=audio 10652 RTP/SAVPF 0 101
c=IN IP4 209.133.200.178
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=pcfg:1 t=1 a=1,2,4|3
a=pcfg:2 t=2 a=1,2,4|3
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:3477872752 cname:c117547c0edf0b0d796c1a6212440c15
a=ssrc:3477872752 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3477872752 label:doubango@audio
a=ice-ufrag:F1EINniGs44Y9ow
a=ice-pwd:e3Gz08NPhIRLvkGC6NYWPE
a=candidate:FfrHF4ZoP2Z1pTl 1 udp 2130706431 209.133.200.178 10652 typ host
a=candidate:FfrHF4ZoP2Z1pTl 2 udp 2130706430 209.133.200.178 10653 typ host
SIPml-api.js:3 getUserMedia() no longer works on insecure origins. To use this feature, you should consider switching your application to a secure origin, such as HTTPS. See https://goo.gl/rStTGz for more details.
SIPml-api.js:1 onGetUserMediaError
SIPml-api.js:1 NavigatorUserMediaErrortsk_utils_log_error @ SIPml-api.js:1
SIPml-api.js:1 State machine: tsip_dialog_invite_Started_2_Started_X_any
SIPml-api.js:1 State machine: s0000_Started_2_Ringing_X_iINVITE
SIPml-api.js:1 onSetRemoteDescriptionSuccess
SIPml-api.js:1 onSignalingstateChange:have-remote-offer
SIPml-api.js:1 __on_add_stream
SIPml-api.js:1 ==stack event = m_permission_requested
SIPml-api.js:1 ==stack event = m_permission_refused
SIPml-api.js:1 State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
SIPml-api.js:1 SEND: SIP/2.0 180 Ringing
Via: SIP/2.0/WS 209.133.200.178:10061;rport=10061;branch=z9hG4bK-1184506698
From: <sip:80080...@209.133.200.178>;tag=754980512
To: <sip:70...@209.133.200.178:10061>;tag=0o1OgsqDSo4Np77wQ20A
Contact: <sip:70...@df7jal23ls0d.invalid;transport=ws>
Call-ID: 5bc51625-b8bc-6240-fcae-bfaeec8fe773
CSeq: 523907086 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
SIPml-api.js:1 ==stack event = i_new_call
2SIPml-api.js:1 ==session event = m_stream_audio_remote_added
http://209.133.200.178/agc-dev/webphone/sipml5/null Failed to load resource: the server responded with a status of 404 (Not Found)
SIPml-api.js:1 ==session event = m_stream_audio_remote_added
SIPml-api.js:1 State machine: s0000_Ringing_2_Connected_X_Accept
SIPml-api.js:1 ==session event = connected
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=ws;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK6261efd9
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as56f3d206
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:10061;transport=ws>
Max-Forwards: 70
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: 30 Dec 2015 5:14:9 GMT;30
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
SIPml-api.js:1 Not implementedtsk_utils_log_error @ SIPml-api.js:1
SIPml-api.js:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK6261efd9
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as56f3d206
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
http://209.133.200.178/agc-dev/webphone/sipml5/null Failed to load resource: the server responded with a status of 404 (Not Found)
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=CANCEL sip:70...@209.133.200.178:10061 SIP/2.0
Via: SIP/2.0/WS 209.133.200.178:10061;rport;branch=z9hG4bK-1184506698
From: <sip:80080...@209.133.200.178>;tag=754980512
Call-ID: 5bc51625-b8bc-6240-fcae-bfaeec8fe773
CSeq: 523907086 CANCEL
Content-Length: 0
Max-Forwards: 70
Route: <sip:180.191.109.46:51164;transport=ws;lr>
Route: <sip:180.191.109.46:51164;transport=ws;lr>
SIPml-api.js:1 State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister
SIPml-api.js:1 SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKg7HixXHaRpVwG99uhbtQHxER1un3PaO8;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61876 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="53788be1",uri="sip:asterisk",response="6b46764aaf18d55af746111c413c87d6",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
SIPml-api.js:1 ==session event = sent_request
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.133.200.178:10061;rport=10061;received=209.133.200.178;branch=z9hG4bKg7HixXHaRpVwG99uhbtQHxER1un3PaO8
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61876 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKg7HixXHaRpVwG99uhbtQHxER1un3PaO8;ws-hacked=WS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="5b735138",stale=FALSE,algorithm=MD5
SIPml-api.js:1 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js:1 SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61877 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="5b735138",uri="sip:asterisk",response="2fd81998744e2f2f9ad2b0af5bdc3062",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
SIPml-api.js:1 ==session event = sent_request
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=ws;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK6bad4f26
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as6516a2a7
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:10061;transport=ws>
Max-Forwards: 70
User-Agent: Asterisk PBX 1.8.32.3-vici
Date: 30 Dec 2015 5:14:50 GMT;30
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
SIPml-api.js:1 Not implementedtsk_utils_log_error @ SIPml-api.js:1
SIPml-api.js:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK6bad4f26
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as6516a2a7
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONS
Content-Length: 0
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:10061;rport=10061;received=209.133.200.178;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61877 REGISTER
Expires: 200
Content-Length: 0
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;ws-hacked=WS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 30 Dec 2015 5:14:50 GMT;30
SIPml-api.js:1 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=ws;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK660f8e63
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as1d032a79
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Contact: <sip:aste...@209.133.200.178:10061;transport=ws>
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 95
Max-Forwards: 70
User-Agent: Asterisk PBX 1.8.32.3-vici
Event: message-summary
SIPml-api.js:1 SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK660f8e63
From: "asterisk"<sip:aste...@209.133.200.178>;tag=as1d032a79
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 NOTIFY
Content-Length: 0
SIPml-api.js:1 State machine: x0000_Any_2_Trying_X_shutdown
SIPml-api.js:1 SEND: BYE sip:8008093395@209.133.200.178:10061;transport=ws SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKIHpCgpPdnN4oZKCaqKZT1zwhUuTycwGS;rport
From: <sip:70...@209.133.200.178:10061>;tag=0o1OgsqDSo4Np77wQ20A
To: <sip:80080...@209.133.200.178>;tag=754980512
Call-ID: 5bc51625-b8bc-6240-fcae-bfaeec8fe773
CSeq: 19755 BYE
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
SIPml-api.js:1 ==session event = terminating
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;branch=z9hG4bKIHpCgpPdnN4oZKCaqKZT1zwhUuTycwGS
From: <sip:70...@209.133.200.178:10061>;tag=0o1OgsqDSo4Np77wQ20A
To: <sip:80080...@209.133.200.178>;tag=754980512
Contact: <sip:8008093395@209.133.200.178:10061;transport=ws;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
Call-ID: 5bc51625-b8bc-6240-fcae-bfaeec8fe773
CSeq: 19755 BYE
Content-Length: 0
SIPml-api.js:1 State machine: x0000_Any_2_Terminated_X_i2xxBYE
SIPml-api.js:1 === INVITE Dialog terminated ===
SIPml-api.js:1 PeerConnection::stop()
SIPml-api.js:1 ==session event = terminated
SIPml-api.js:1 The FSM is in the final state
SIPml-api.js:1 State machine: tsip_dialog_register_Any_2_InProgress_X_shutdown
SIPml-api.js:1 SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK593udChhO9hbO5y7SwDlOEHLTUd0ik91;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=0;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61878 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="5b735138",uri="sip:asterisk",response="2fd81998744e2f2f9ad2b0af5bdc3062",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
SIPml-api.js:1 ==session event = terminating
SIPml-api.js:1 ==session event = sent_request
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.133.200.178:10061;rport=10061;received=209.133.200.178;branch=z9hG4bK593udChhO9hbO5y7SwDlOEHLTUd0ik91
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61878 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bK593udChhO9hbO5y7SwDlOEHLTUd0ik91;ws-hacked=WS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="1de80666",stale=TRUE,algorithm=MD5
SIPml-api.js:1 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js:1 SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKlnxpRMcnS3WPWTTm7E2UX1EvBqn2bWmM;rport
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>
Contact: "mark"<sip:70...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=0;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61879 REGISTER
Content-Length: 0
Route: <sip:://:NaN;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="7002",realm="asterisk",nonce="1de80666",uri="sip:asterisk",response="a201c070a2f4cc50b52e19d93a6632b1",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
SIPml-api.js:1 ==session event = sent_request
2http://209.133.200.178/agc-dev/webphone/sipml5/null Failed to load resource: the server responded with a status of 404 (Not Found)
SIPml-api.js:1 __tsip_transport_ws_onmessage
SIPml-api.js:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.133.200.178:10061;rport=10061;received=209.133.200.178;branch=z9hG4bKlnxpRMcnS3WPWTTm7E2UX1EvBqn2bWmM
From: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQL
To: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dc
Call-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29
CSeq: 61879 REGISTER
Expires: 0
Content-Length: 0
Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKlnxpRMcnS3WPWTTm7E2UX1EvBqn2bWmM;ws-hacked=WS
Server: Asterisk PBX 1.8.32.3-vici
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 30 Dec 2015 5:14:56 GMT;30
SIPml-api.js:1 State machine: tsip_dialog_register_InProgress_2_Terminated_X_2xx
SIPml-api.js:1 === REGISTER Dialog terminated ===
http://209.133.200.178/agc-dev/webphone/sipml5/null Failed to load resource: the server responded with a status of 404 (Not Found)
SIPml-api.js:1 ==session event = terminated
SIPml-api.js:1 ==stack event = stopped
SIPml-api.js:1 __tsip_transport_ws_onclose
SIPml-api.js:1 ==stack event = stopped
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.6.2.10Date: Wed, 30 Dec 2015 05:13:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0<------------->[Dec 30 00:13:09] --- (13 headers 0 lines) ---[Dec 30 00:13:09] Looking for s in trunkinbound (domain 209.133.200.178)[Dec 30 00:13:09]<--- Transmitting (NAT) to 199.21.115.215:5060 --->SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK41f64774;received=199.21.115.215;rport=5060From: "asterisk" <sip:aste...@199.21.115.215>;tag=as41a0ec20To: <sip:209.133.200.178>;tag=as10c2e485
CSeq: 102 OPTIONSServer: Asterisk PBX 1.8.32.3-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerAccept: application/sdpContent-Length: 0<------------>
[Dec 30 00:13:09] Scheduling destruction of SIP dialog '15a3572104cf042d1db8cf123412ae...@199.21.115.215' in 32000 ms (Method: OPTIONS)
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Wed, 30 Dec 2015 05:13:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0---[Dec 30 00:13:09]<--- Transmitting (NAT) to 209.133.200.178:10061 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;received=209.133.200.178;rport=10061Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bKbeLNUMx9gnaQWVdlOMniaaLTn6AqPUAk;ws-hacked=WSFrom: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQLTo: "mark"<sip:70...@209.133.200.178>;tag=as4da072feCall-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29CSeq: 61875 REGISTERServer: Asterisk PBX 1.8.32.3-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerExpires: 200Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200Date: Wed, 30 Dec 2015 05:13:09 GMTContent-Length: 0<------------>
[Dec 30 00:13:09] Scheduling destruction of SIP dialog '6f7fd43f2948d0363831f95703f846...@209.133.200.178:5060' in 32000 ms (Method: NOTIFY)
[Dec 30 00:13:09] Reliably Transmitting (NAT) to 209.133.200.178:10061:NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK601becb8;rportMax-Forwards: 70From: "asterisk" <sip:aste...@209.133.200.178>;tag=as2e1c170cTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFYUser-Agent: Asterisk PBX 1.8.32.3-viciEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 95Messages-Waiting: noMessage-Account: sip:aste...@209.133.200.178Voice-Message: 0/0 (0/0)---[Dec 30 00:13:09] Scheduling destruction of SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' in 32000 ms (Method: REGISTER)[Dec 30 00:13:09]<--- SIP read from UDP:209.133.200.178:10061 --->SIP/2.0 405 Method Not AllowedVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4b262a89
From: "asterisk"<sip:asterisk@209.133.200.178>;tag=as7db7041d
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONSContent-Length: 0<------------->[Dec 30 00:13:09] --- (7 headers 0 lines) ---[Dec 30 00:13:09] NOTICE[2545]: chan_sip.c:21672 handle_response_peerpoke: Peer '7002' is now Reachable. (434ms / 2000ms)
[Dec 30 00:13:09] Really destroying SIP dialog '539f965042ba384e015c824a4e45da...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:13:09] Retransmitting #1 (NAT) to 209.133.200.178:10061:NOTIFY sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK601becb8;rportMax-Forwards: 70From: "asterisk" <sip:aste...@209.133.200.178>;tag=as2e1c170cTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 NOTIFYUser-Agent: Asterisk PBX 1.8.32.3-viciEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 95Messages-Waiting: noMessage-Account: sip:aste...@209.133.200.178Voice-Message: 0/0 (0/0)---[Dec 30 00:13:10]<--- SIP read from UDP:209.133.200.178:10061 --->SIP/2.0 481 Dialog/Transaction Does Not ExistVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK601becb8
From: "asterisk"<sip:asterisk@209.133.200.178>;tag=as2e1c170c
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 NOTIFYContent-Length: 0<------------->[Dec 30 00:13:10] --- (7 headers 0 lines) ---
[Dec 30 00:13:10] Really destroying SIP dialog '6f7fd43f2948d0363831f95703f846...@209.133.200.178:5060' Method: NOTIFY[Dec 30 00:13:31] Really destroying SIP dialog '1197d94255e27395266197fe04f44a...@199.21.115.215' Method: OPTIONS
[Dec 30 00:13:35] Reliably Transmitting (NAT) to 208.74.75.250:5060:OPTIONS sip:208.74.75.250 SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK03babc65;rportMax-Forwards: 70From: "asterisk" <sip:aste...@209.133.200.178>;tag=as64b1e129To: <sip:208.74.75.250>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Wed, 30 Dec 2015 05:13:35 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0---[Dec 30 00:13:35]<--- SIP read from UDP:208.74.75.250:5060 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK03babc65;rport=5060To: <sip:208.74.75.250>From: "asterisk" <sip:aste...@209.133.200.178>;tag=as64b1e129
CSeq: 102 OPTIONSContent-Length: 0<------------->[Dec 30 00:13:35] --- (7 headers 0 lines) ---
[Dec 30 00:13:35] Really destroying SIP dialog '67eda84c3d740a16022b957e5090e4...@209.133.200.178:5060' Method: OPTIONS[Dec 30 00:13:41] Really destroying SIP dialog '15a3572104cf042d1db8cf123412ae...@199.21.115.215' Method: OPTIONS
[Dec 30 00:13:41] Really destroying SIP dialog '7f434a93-b8e2-a99b-fb1e-0185884e4d29' Method: REGISTER[Dec 30 00:13:50] == Manager 'sendcron' logged on from 127.0.0.1[Dec 30 00:13:50] == Manager 'sendcron' logged on from 127.0.0.1[Dec 30 00:13:50] -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-00007917;2", "8600052,K") in new stack[Dec 30 00:13:50] WARNING[9462]: app_meetme.c:4840 admin_exec: Conference number '8600052' not found![Dec 30 00:13:50] -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-00007917;2", "") in new stack[Dec 30 00:13:50] == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-00007917;2'[Dec 30 00:13:50] -- Executing [h@default:1] AGI("Local/55558600052@default-00007917;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack[Dec 30 00:13:50] -- <Local/55558600052@default-00007917;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0[Dec 30 00:13:51] == Manager 'sendcron' logged off from 127.0.0.1[Dec 30 00:13:51] == Manager 'sendcron' logged off from 127.0.0.1[Dec 30 00:13:56] Reliably Transmitting (NAT) to 66.148.120.167:5060:OPTIONS sip:66.148.120.167 SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK36afb48e;rportMax-Forwards: 70From: "asterisk" <sip:aste...@209.133.200.178>;tag=as69c74b1cTo: <sip:66.148.120.167>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Wed, 30 Dec 2015 05:13:56 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0---[Dec 30 00:13:56]<--- SIP read from UDP:66.148.120.167:5060 --->SIP/2.0 200 OKCSeq: 102 OPTIONSVia: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK36afb48e;rportFrom: "asterisk" <sip:aste...@209.133.200.178>;tag=as69c74b1c
To: <sip:66.148.120.167>;tag=301214151303Allow: INVITE, ACK, CANCEL, OPTIONS, BYEContent-Length: 0<------------->[Dec 30 00:13:56] --- (8 headers 0 lines) ---
[Dec 30 00:13:56] Really destroying SIP dialog '577719eb3264f74d7fa90ad65a329d...@209.133.200.178:5060' Method: OPTIONS
[Dec 30 00:13:59]<--- SIP read from UDP:199.21.115.215:5060 --->OPTIONS sip:209.133.200.178 SIP/2.0Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK0c5d1747;rportMax-Forwards: 70From: "asterisk" <sip:aste...@199.21.115.215>;tag=as6d3b1432To: <sip:209.133.200.178>Contact: <sip:aste...@199.21.115.215>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.6.2.10Date: Wed, 30 Dec 2015 05:13:59 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0<------------->[Dec 30 00:13:59] --- (13 headers 0 lines) ---[Dec 30 00:13:59] Looking for s in trunkinbound (domain 209.133.200.178)[Dec 30 00:13:59]<--- Transmitting (NAT) to 199.21.115.215:5060 --->SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK0c5d1747;received=199.21.115.215;rport=5060From: "asterisk" <sip:aste...@199.21.115.215>;tag=as6d3b1432To: <sip:209.133.200.178>;tag=as39cddcbe
CSeq: 102 OPTIONSServer: Asterisk PBX 1.8.32.3-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerAccept: application/sdpContent-Length: 0<------------>
[Dec 30 00:13:59] Scheduling destruction of SIP dialog '78c6786f2fc65d706e94105a7ce2cf...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:14:01] == Manager 'sendcron' logged on from 127.0.0.1[Dec 30 00:14:01] == Manager 'sendcron' logged off from 127.0.0.1[Dec 30 00:14:01] == Manager 'sendcron' logged on from 127.0.0.1[Dec 30 00:14:01] == Manager 'sendcron' logged off from 127.0.0.1[Dec 30 00:14:03] == Manager 'sendcron' logged on from 127.0.0.1[Dec 30 00:14:03] == Using SIP RTP CoS mark 5[Dec 30 00:14:03] Audio is at 16912[Dec 30 00:14:03] Adding codec 0x4 (ulaw) to SDP[Dec 30 00:14:03] Adding codec 0x2 (gsm) to SDP[Dec 30 00:14:03] Adding non-codec 0x1 (telephone-event) to SDP[Dec 30 00:14:03] Reliably Transmitting (NAT) to 209.133.200.178:10061:INVITE sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK4be00157;rportMax-Forwards: 70From: "ACagcW14514524391001100110011001" <sip:80080...@209.133.200.178>;tag=as40e46c0eTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>Contact: <sip:80080...@209.133.200.178:5060>
CSeq: 102 INVITEContent-Length: 0<------------->[Dec 30 00:14:03] --- (7 headers 0 lines) ---[Dec 30 00:14:03]<--- SIP read from UDP:209.133.200.178:10061 --->SIP/2.0 180 RingingVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4be00157From: "ACagcW14514524391001100110011001"<sip:80080...@209.133.200.178>;tag=as40e46c0eTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345Contact: <sip:70...@209.133.200.178:10061;transport=udp>
CSeq: 102 INVITEContent-Length: 0Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE<------------->[Dec 30 00:14:03] --- (9 headers 0 lines) ---[Dec 30 00:14:03] list_route: hop: <sip:70...@209.133.200.178:10061;transport=udp>[Dec 30 00:14:06] == Manager 'sendcron' logged on from 127.0.0.1[Dec 30 00:14:06] == Manager 'sendcron' logged off from 127.0.0.1[Dec 30 00:14:09]<--- SIP read from UDP:199.21.115.215:5060 --->OPTIONS sip:209.133.200.178 SIP/2.0Via: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK752c4f5c;rportMax-Forwards: 70From: "asterisk" <sip:aste...@199.21.115.215>;tag=as38ac5c8cTo: <sip:209.133.200.178>Contact: <sip:aste...@199.21.115.215>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.6.2.10Date: Wed, 30 Dec 2015 05:14:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0<------------->[Dec 30 00:14:09] --- (13 headers 0 lines) ---[Dec 30 00:14:09] Looking for s in trunkinbound (domain 209.133.200.178)[Dec 30 00:14:09]<--- Transmitting (NAT) to 199.21.115.215:5060 --->SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 199.21.115.215:5060;branch=z9hG4bK752c4f5c;received=199.21.115.215;rport=5060From: "asterisk" <sip:aste...@199.21.115.215>;tag=as38ac5c8cTo: <sip:209.133.200.178>;tag=as773073d0
CSeq: 102 OPTIONSServer: Asterisk PBX 1.8.32.3-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerAccept: application/sdpContent-Length: 0<------------>
[Dec 30 00:14:09] Scheduling destruction of SIP dialog '3abb6d644a73cc4144eb360c418b61...@199.21.115.215' in 32000 ms (Method: OPTIONS)
[Dec 30 00:14:09] Reliably Transmitting (NAT) to 209.133.200.178:10061:OPTIONS sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK6261efd9;rportMax-Forwards: 70From: "asterisk" <sip:aste...@209.133.200.178>;tag=as56f3d206To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Wed, 30 Dec 2015 05:14:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0---[Dec 30 00:14:10]<--- SIP read from UDP:209.133.200.178:10061 --->SIP/2.0 405 Method Not AllowedVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK6261efd9
From: "asterisk"<sip:asterisk@209.133.200.178>;tag=as56f3d206
To: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 OPTIONSContent-Length: 0<------------->[Dec 30 00:14:10] --- (7 headers 0 lines) ---
[Dec 30 00:14:10] Really destroying SIP dialog '692e9dda4586438451ff4ae177262e...@209.133.200.178:5060' Method: OPTIONS[Dec 30 00:14:31] Really destroying SIP dialog '78c6786f2fc65d706e94105a7ce2cf...@199.21.115.215' Method: OPTIONS[Dec 30 00:14:33] Scheduling destruction of SIP dialog '1b9c795737d74d56121c89d0016cad...@209.133.200.178:5060' in 27776 ms (Method: INVITE)
[Dec 30 00:14:33] Reliably Transmitting (NAT) to 209.133.200.178:10061:CANCEL sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK4be00157;rportMax-Forwards: 70From: "ACagcW14514524391001100110011001" <sip:80080...@209.133.200.178>;tag=as40e46c0eTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>
CSeq: 102 CANCELUser-Agent: Asterisk PBX 1.8.32.3-viciContent-Length: 0---
[Dec 30 00:14:33] Scheduling destruction of SIP dialog '1b9c795737d74d56121c89d0016cad...@209.133.200.178:5060' in 27776 ms (Method: INVITE)
[Dec 30 00:14:33] == Manager 'sendcron' logged off from 127.0.0.1[Dec 30 00:14:33]<--- SIP read from UDP:209.133.200.178:10061 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4be00157From: "ACagcW14514524391001100110011001"<sip:80080...@209.133.200.178>;tag=as40e46c0eTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345Contact: <sip:70...@209.133.200.178:10061;transport=udp>
CSeq: 102 CANCELContent-Length: 0<------------->[Dec 30 00:14:33] --- (8 headers 0 lines) ---[Dec 30 00:14:33]<--- SIP read from UDP:209.133.200.178:10061 --->SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 209.133.200.178:5060;rport=5060;received=209.133.200.178;branch=z9hG4bK4be00157From: "ACagcW14514524391001100110011001"<sip:80080...@209.133.200.178>;tag=as40e46c0eTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345
CSeq: 102 INVITEContent-Length: 0Reason: SIP; cause=487; text="Request Cancelled"<------------->[Dec 30 00:14:33] --- (8 headers 0 lines) ---[Dec 30 00:14:33] Transmitting (NAT) to 209.133.200.178:10061:ACK sip:70...@209.133.200.178:10061;transport=udp SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK4be00157;rportMax-Forwards: 70From: "ACagcW14514524391001100110011001" <sip:80080...@209.133.200.178>;tag=as40e46c0eTo: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;tag=465692345Contact: <sip:80080...@209.133.200.178:5060>
CSeq: 102 ACKUser-Agent: Asterisk PBX 1.8.32.3-viciContent-Length: 0---
[Dec 30 00:14:33] Scheduling destruction of SIP dialog '1b9c795737d74d56121c89d0016cad...@209.133.200.178:5060' in 27776 ms (Method: INVITE)
[Dec 30 00:14:35] Reliably Transmitting (NAT) to 208.74.75.250:5060:OPTIONS sip:208.74.75.250 SIP/2.0Via: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK168cfe39;rportMax-Forwards: 70From: "asterisk" <sip:aste...@209.133.200.178>;tag=as550b737aTo: <sip:208.74.75.250>Contact: <sip:aste...@209.133.200.178:5060>
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Wed, 30 Dec 2015 05:14:35 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0---[Dec 30 00:14:35]<--- SIP read from UDP:208.74.75.250:5060 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 209.133.200.178:5060;branch=z9hG4bK168cfe39;rport=5060To: <sip:208.74.75.250>From: "asterisk" <sip:aste...@209.133.200.178>;tag=as550b737a
CSeq: 102 OPTIONSContent-Length: 0<------------->[Dec 30 00:14:35] --- (7 headers 0 lines) ---
[Dec 30 00:14:35] Really destroying SIP dialog '6a8142882fd2e59742c0da7059eb8e...@209.133.200.178:5060' Method: OPTIONS[Dec 30 00:14:41] Really destroying SIP dialog '3abb6d644a73cc4144eb360c418b61...@199.21.115.215' Method: OPTIONS
CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.8.32.3-viciDate: Wed, 30 Dec 2015 05:14:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Length: 0---[Dec 30 00:14:50]<--- Transmitting (NAT) to 209.133.200.178:10061 --->SIP/2.0 200 OKVia: SIP/2.0/UDP 209.133.200.178:10061;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;received=209.133.200.178;rport=10061Via: SIP/2.0/TCP 180.191.109.46:51164;rport;branch=z9hG4bK1FdI26V7o2FFHV5UucIKF7V08kSbzBmf;ws-hacked=WSFrom: "mark"<sip:70...@209.133.200.178>;tag=THxlLWp8Tt64u2HlLZQLTo: "mark"<sip:70...@209.133.200.178>;tag=as4201d5dcCall-ID: 7f434a93-b8e2-a99b-fb1e-0185884e4d29CSeq: 61877 REGISTERServer: Asterisk PBX 1.8.32.3-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerExpires: 200Contact: <sip:70...@209.133.200.178:10061;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.191.109.46;ws-src-port=51164;ws-src-proto=ws>;expires=200Date: Wed, 30 Dec 2015 05:14:50 GMTContent-Length: 0<------------>
[Dec 30 00:14:50] Scheduling destruction of SIP dialog '331ba133294209f77c5c8506706451...@209.133.200.178:5060' in 35136 ms (Method: NOTIFY)
[Dec 30 00:14:50] Reliably Transmitting (NAT) to 209.133.200.178:10061:NOTIFY sip:70...@209.133.200.178:10061;rtcweb-br
...
navaismo
Dec 30, 2015, 10:17:25 AM12/30/15
to discuss-doubango
Use the TAB key to autocomplete the ftp debug command.
Recently you can only use WSS and no more WS so switch to secure layer. In the past I helped a guy to setup his vicidial with sipml5 to work like mine and he wrote about it https://github.com/noahseis/webrtc2sip/blob/master/readme_install_walkthrough.txt
It is important to check the rtp debug flow in asterisk.
mark mirasol
Dec 30, 2015, 7:31:57 PM12/30/15
to discuss-doubango
Thank you, navaismo. Yes, I followed noah's guide to setup mine. I also asked him for help and he told me that we need to use ssl certificates now.
I'm also following another thread
and it seems all the errors in the log are related to the certificates. I posted the asterisk log with rtp debug on above.
Can you confirm if indeed it's a certificate issue?
mark mirasol
Dec 30, 2015, 10:23:57 PM12/30/15
to discuss-doubango
Navaismo, thank you very much.
I was able to get our instance working. It was indeed a certificate issue and we were able to fix it by recreate the certificates. https://groups.google.com/forum/#!topic/doubango/-6XKVB_Y1kY.
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