Hi,
I am running Asterisk on VM on a windows server (also having public IP). I have following details for peers and trying calling from one to other using SIPML5 application on chrome browser. However I am getting the error “Rejecting secure audio stream without encryption details” everytime.
I went through all related articles but couldn’t solve the issue.
please help what setting should I change??
one thing to notice -
client IP is different in << SIP read from WS:[client-ip]:[client port] >> line and << c=IN IP4 [client-ip] >> line as described in this article
forums.asterisk.org/viewtopic.php?f=1&t=90167&p=199275&hilit=Rejecting+secure+audio+stream+without+encryption+detail#p199275
which might be causing the problem.
IPs
user one - 192.168.32.181
user two - 192.168.49.170
windows server - and 192.168.1.40
VM - 192.168.131.145
details
sip.conf
[code]
[1113]
type=friend
host=dynamic
secret=0000
context=hipath
transport=ws,wss
avpf=yes
;encryption=yes
videosupport=no
icesupport=yes
nat=force_rpot,comedia ;;nat=yes
[1114]
type=friend
host=dynamic
secret=0000
context=hipath
transport=ws,wss,udp
avpf=yes
;encryption=yes
videosupport=no
icesupport=yes
nat=force_rpot,comedia ;;nat=yes //tried with both settings[/code]
JS console output
[code] State machine: c0000_Started_2_Outgoing_X_oINVITE SIPml-api.js?svn=224:1
ICE servers:[{“url”:“stun:stun.l.google.com:19302”},{“url”:“stun:stun.counterpath.net:3478”},{“url”:“stun:numb.viagenie.ca:3478”}] SIPml-api.js?svn=224:1
==stack event = m_permission_requested SIPml-api.js?svn=224:1
==session event = connecting SIPml-api.js?svn=224:1
onGetUserMediaSuccess SIPml-api.js?svn=224:1
createOffer SIPml-api.js?svn=224:1
onCreateSdpSuccess SIPml-api.js?svn=224:1
==stack event = m_permission_accepted SIPml-api.js?svn=224:1
==session event = m_stream_audio_local_added SIPml-api.js?svn=224:1
onSetLocalDescriptionSuccess SIPml-api.js?svn=224:1
7onIceCandidate = undefined SIPml-api.js?svn=224:1
ICE GATHERING COMPLETED! SIPml-api.js?svn=224:1
onIceGatheringCompleted SIPml-api.js?svn=224:1
SEND: INVITE sip:1114@192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFqm1iThoCt3Josc9pgvZCnAH2eBlq6a7;rport From: "1113"sip:1113@192.168.1.40;tag=P1HFRB7d6FlnWT4cQ8Tu To: sip:1114@192.168.1.40 Contact: "1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1113;ha1=bd869170f0156572561fba91dbc4acac;+g.oma.sip-im;+sip.ice;language=“en,fr” Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073 CSeq: 32986 INVITE Content-Type: application/sdp Content-Length: 1767 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 v=0 o=- 7573769006897112000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f m=audio 35209 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 a=rtcp:35209 IN IP4 a=candidate:2942323686 1 udp 2122260223 192.168.32.181 35209 typ host generation 0 a=candidate:2942323686 2 udp 2122260223 192.168.32.181 35209 typ host generation 0 a=candidate:3789797142 1 tcp 1518280447 192.168.32.181 0 typ host generation 0 a=candidate:3789797142 2 tcp 1518280447 192.168.32.181 0 typ host generation 0 a=candidate:1676506229 1 udp 1686052607 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0 a=candidate:1676506229 2 udp 1686052607 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0 a=ice-ufrag:cmS2Mdkr3dIJVTVc a=ice-pwd:1JZIAYodAALxo5r8YA8D18et a=ice-options:google-ice a=fingerprint:sha-256 37:63:96:1E:22:17:D1:45:68:02:F8:9F:28:37:6A:1C:3F:36:51:B2:1E:A8:B4:A3:D6:BE:E9:5F:1F:B1:28:6C a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2622176125 cname:RjotqUOp9+pcE8Ei a=ssrc:2622176125 msid:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f 2d3dc4e5-a26d-46e7-a282-0af274c98081 a=ssrc:2622176125 mslabel:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f a=ssrc:2622176125 label:2d3dc4e5-a26d-46e7-a282-0af274c98081 SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bKFqm1iThoCt3Josc9pgvZCnAH2eBlq6a7 From: "1113"sip:1113@192.168.1.40;tag=P1HFRB7d6FlnWT4cQ8Tu To: sip:1114@192.168.1.40;tag=as050b0fdb Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073 CSeq: 32986 INVITE Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer WWW-Authenticate: Digest realm=“asterisk”,nonce=“1d058aba”,stale=FALSE,algorithm=MD5 SIPml-api.js?svn=224:1
SEND: ACK sip:1114@192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFqm1iThoCt3Josc9pgvZCnAH2eBlq6a7;rport From: "1113"sip:1113@192.168.1.40;tag=P1HFRB7d6FlnWT4cQ8Tu To: sip:1114@192.168.1.40;tag=as050b0fdb Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073 CSeq: 32986 ACK Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 SIPml-api.js?svn=224:1
State machine: x0000_Any_2_Any_X_i401_407_INVITE SIPml-api.js?svn=224:1
SEND: INVITE sip:1114@192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLLmfsw1TX71X126dJCaJrNvFPNoXA1rI;rport From: "1113"sip:1113@192.168.1.40;tag=P1HFRB7d6FlnWT4cQ8Tu To: sip:1114@192.168.1.40 Contact: “1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1113;ha1=bd869170f0156572561fba91dbc4acac;+g.oma.sip-im;+sip.ice;language=“en,fr” Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073 CSeq: 32987 INVITE Content-Type: application/sdp Content-Length: 1767 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Authorization: Digest username=“1113”,realm=“asterisk”,nonce=“1d058aba”,uri="sip:1114@192.168.1.40”,response=“c4e16e428c3f5266f989376016bb04a1”,algorithm=MD5 v=0 o=- 7573769006897112000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f m=audio 35209 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 a=rtcp:35209 IN IP4 a=candidate:2942323686 1 udp 2122260223 192.168.32.181 35209 typ host generation 0 a=candidate:2942323686 2 udp 2122260223 192.168.32.181 35209 typ host generation 0 a=candidate:3789797142 1 tcp 1518280447 192.168.32.181 0 typ host generation 0 a=candidate:3789797142 2 tcp 1518280447 192.168.32.181 0 typ host generation 0 a=candidate:1676506229 1 udp 1686052607 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0 a=candidate:1676506229 2 udp 1686052607 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0 a=ice-ufrag:cmS2Mdkr3dIJVTVc a=ice-pwd:1JZIAYodAALxo5r8YA8D18et a=ice-options:google-ice a=fingerprint:sha-256 37:63:96:1E:22:17:D1:45:68:02:F8:9F:28:37:6A:1C:3F:36:51:B2:1E:A8:B4:A3:D6:BE:E9:5F:1F:B1:28:6C a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2622176125 cname:RjotqUOp9+pcE8Ei a=ssrc:2622176125 msid:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f 2d3dc4e5-a26d-46e7-a282-0af274c98081 a=ssrc:2622176125 mslabel:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f a=ssrc:2622176125 label:2d3dc4e5-a26d-46e7-a282-0af274c98081 SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 488 Not acceptable here Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bKLLmfsw1TX71X126dJCaJrNvFPNoXA1rI From: "1113"sip:1113@192.168.1.40;tag=P1HFRB7d6FlnWT4cQ8Tu To: sip:1114@192.168.1.40;tag=as050b0fdb Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073 CSeq: 32987 INVITE Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer SIPml-api.js?svn=224:1
SEND: ACK sip:1114@192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLLmfsw1TX71X126dJCaJrNvFPNoXA1rI;rport From: "1113"sip:1113@192.168.1.40;tag=P1HFRB7d6FlnWT4cQ8Tu To: sip:1114@192.168.1.40;tag=as050b0fdb Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073 CSeq: 32987 ACK Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 SIPml-api.js?svn=224:1
State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE SIPml-api.js?svn=224:1
=== INVITE Dialog terminated === SIPml-api.js?svn=224:1
PeerConnection::stop() SIPml-api.js?svn=224:1
==session event = i_ao_request SIPml-api.js?svn=224:1
==session event = terminated SIPml-api.js?svn=224:1
The FSM is in the final state SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister SIPml-api.js?svn=224:1
SEND: REGISTER sip:192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKbsns9wccafSW0rd6Nq3wfVbXs71m9NCN;rport From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40 Contact: "1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45050 REGISTER Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Authorization: Digest username=“1113”,realm=“asterisk”,nonce=“5540eea5”,uri=“sip:192.168.1.40”,response=“1a1fd92101d6d73b3dd2d747639481b9”,algorithm=MD5 SIPml-api.js?svn=224:1
==session event = sent_request SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bKbsns9wccafSW0rd6Nq3wfVbXs71m9NCN From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40;tag=as28900697 Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45050 REGISTER Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer WWW-Authenticate: Digest realm=“asterisk”,nonce=“2f05e66e”,stale=FALSE,algorithm=MD5 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 SIPml-api.js?svn=224:1
SEND: REGISTER sip:192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKnHBUoSQAFP7h9nSk8l0nAJMpx9xfKha;rport From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40 Contact: "1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45051 REGISTER Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Authorization: Digest username=“1113”,realm=“asterisk”,nonce=“2f05e66e”,uri=“sip:192.168.1.40”,response=“d580c873292ee99098c1620978ea7162”,algorithm=MD5 SIPml-api.js?svn=224:1
==session event = sent_request SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK7bc122a0 From: "asterisk"sip:asterisk@192.168.1.40;tag=as58b7571b To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.40;transport=WS Call-ID: 4096ae65042e1c727a8047870a5739d7@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Asterisk PBX 11.10.2 Date: 24 Jun 2014 19:14:19 GMT;24 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer SIPml-api.js?svn=224:1
Not implemented SIPml-api.js?svn=224:1
SEND: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK7bc122a0 From: "asterisk"sip:asterisk@192.168.1.40;tag=as58b7571b To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Call-ID: 4096ae65042e1c727a8047870a5739d7@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bKKnHBUoSQAFP7h9nSk8l0nAJMpx9xfKha From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40;tag=as28900697 Contact: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200 Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45051 REGISTER Expires: 200 Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer Date: 24 Jun 2014 19:14:19 GMT;24 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK3550e327 From: "asterisk"sip:asterisk@192.168.1.40;tag=as50b6d26c To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.40;transport=WS Call-ID: 08e27ec1409f1b396c3cf73c601a8d9f@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Asterisk PBX 11.10.2 Date: 24 Jun 2014 19:15:19 GMT;24 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer SIPml-api.js?svn=224:1
Not implemented SIPml-api.js?svn=224:1
SEND: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK3550e327 From: "asterisk"sip:asterisk@192.168.1.40;tag=as50b6d26c To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Call-ID: 08e27ec1409f1b396c3cf73c601a8d9f@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister SIPml-api.js?svn=224:1
SEND: REGISTER sip:192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeDJEEOHisdWHBGSMR5XeFZZgJoetUyri;rport From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40 Contact: "1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45052 REGISTER Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Authorization: Digest username=“1113”,realm=“asterisk”,nonce=“2f05e66e”,uri=“sip:192.168.1.40”,response=“d580c873292ee99098c1620978ea7162”,algorithm=MD5 SIPml-api.js?svn=224:1
==session event = sent_request SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bKeDJEEOHisdWHBGSMR5XeFZZgJoetUyri From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40;tag=as1e174d52 Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45052 REGISTER Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer WWW-Authenticate: Digest realm=“asterisk”,nonce=“0db29802”,stale=FALSE,algorithm=MD5 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 SIPml-api.js?svn=224:1
SEND: REGISTER sip:192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK01WCJiFb6NDIazH7TJ8wir4siTnSMisW;rport From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40 Contact: "1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45053 REGISTER Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Authorization: Digest username=“1113”,realm=“asterisk”,nonce=“0db29802”,uri=“sip:192.168.1.40”,response=“6da1b9866f1e4bb020e5781a3ae81da0”,algorithm=MD5 SIPml-api.js?svn=224:1
==session event = sent_request SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK318931f3 From: "asterisk"sip:asterisk@192.168.1.40;tag=as1d3505ca To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.40;transport=WS Call-ID: 16bd979d3182d8a45185375c265ad02a@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Asterisk PBX 11.10.2 Date: 24 Jun 2014 19:15:59 GMT;24 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer SIPml-api.js?svn=224:1
Not implemented SIPml-api.js?svn=224:1
SEND: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK318931f3 From: "asterisk"sip:asterisk@192.168.1.40;tag=as1d3505ca To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Call-ID: 16bd979d3182d8a45185375c265ad02a@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bK01WCJiFb6NDIazH7TJ8wir4siTnSMisW From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40;tag=as1e174d52 Contact: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200 Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45053 REGISTER Expires: 200 Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer Date: 24 Jun 2014 19:15:59 GMT;24 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK3e6ff783 From: "asterisk"sip:asterisk@192.168.1.40;tag=as5385734d To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.40;transport=WS Call-ID: 4baa20cb55ae597352d473fd40486537@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Asterisk PBX 11.10.2 Date: 24 Jun 2014 19:16:59 GMT;24 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer SIPml-api.js?svn=224:1
Not implemented SIPml-api.js?svn=224:1
SEND: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK3e6ff783 From: "asterisk"sip:asterisk@192.168.1.40;tag=as5385734d To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Call-ID: 4baa20cb55ae597352d473fd40486537@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister SIPml-api.js?svn=224:1
SEND: REGISTER sip:192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB46zxHdHnCmar6RtMwi5SASHBBnaEqyx;rport From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40 Contact: "1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45054 REGISTER Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Authorization: Digest username=“1113”,realm=“asterisk”,nonce=“0db29802”,uri=“sip:192.168.1.40”,response=“6da1b9866f1e4bb020e5781a3ae81da0”,algorithm=MD5 SIPml-api.js?svn=224:1
==session event = sent_request SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bKB46zxHdHnCmar6RtMwi5SASHBBnaEqyx From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40;tag=as253bb2c2 Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45054 REGISTER Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer WWW-Authenticate: Digest realm=“asterisk”,nonce=“3455072f”,stale=FALSE,algorithm=MD5 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 SIPml-api.js?svn=224:1
SEND: REGISTER sip:192.168.1.40 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKRhOJwG5VETwwTD4Oag0bzTI1BDUsMXGf;rport From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40 Contact: "1113"sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45055 REGISTER Content-Length: 0 Route: sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Authorization: Digest username=“1113”,realm=“asterisk”,nonce=“3455072f”,uri=“sip:192.168.1.40”,response=“d2d95b1fd4d8b334c54cd0749c5dffb2”,algorithm=MD5 SIPml-api.js?svn=224:1
==session event = sent_request SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK3da87068 From: "asterisk"sip:asterisk@192.168.1.40;tag=as5407b28e To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.40;transport=WS Call-ID: 0ef4403858088d20304bf9280e77df92@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Asterisk PBX 11.10.2 Date: 24 Jun 2014 19:17:39 GMT;24 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer SIPml-api.js?svn=224:1
Not implemented SIPml-api.js?svn=224:1
SEND: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK3da87068 From: "asterisk"sip:asterisk@192.168.1.40;tag=as5407b28e To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Call-ID: 0ef4403858088d20304bf9280e77df92@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.32.181;branch=z9hG4bKRhOJwG5VETwwTD4Oag0bzTI1BDUsMXGf From: "1113"sip:1113@192.168.1.40;tag=pC5Duy9NZDhiNnn6yGSa To: "1113"sip:1113@192.168.1.40;tag=as253bb2c2 Contact: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200 Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a CSeq: 45055 REGISTER Expires: 200 Content-Length: 0 Server: Asterisk PBX 11.10.2 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer Date: 24 Jun 2014 19:17:39 GMT;24 SIPml-api.js?svn=224:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=224:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=224:1
recv=OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK0fe93456 From: "asterisk"sip:asterisk@192.168.1.40;tag=as11bceb0b To: sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.40;transport=WS Call-ID: 0965e7be6f57c865492ad924579c8e5e@192.168.1.40:0 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Asterisk PBX 11.10.2 Date: 24 Jun 2014 19:18:39 GMT;24 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer SIPml-api.js?svn=224:1
[/code]
Sip set debug on
<--- SIP read from UDP:192.168.146.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK070f8e3f;rport=5060;received=192.168.1.40
From: "asterisk" <sip:asterisk@192.168.1.40>;tag=as1951dd18
To: <sip:192.168.146.1>;tag=246966900
Call-ID: 3022a6e538a55b1d61734231542aee2c@192.168.1.40:5060
CSeq: 102 OPTIONS
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,REFER,INFO,PRACK,UPDATE
Server: HiPath 4000 V6 Common Gateway M5T SIP Stack/4.1.8.14
Content-Type: application/sdp
Content-Length: 855
v=0
o=MxSIP 0 0 IN IP4 0.0.0.0
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 5555 RTP/AVP 8 18 4 96 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 CLEARMODE/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
m=audio 5555 RTP/SAVP 8 18 4 96 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 CLEARMODE/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
m=image 5555 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:375
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (10 headers 37 lines) ---
Reliably Transmitting (NAT) to 192.168.146.1:5060:
OPTIONS sip:192.168.146.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK733a9728;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.40>;tag=as556b2afe
To: <sip:192.168.146.1>
Contact: <sip:asterisk@192.168.1.40:5060>
Call-ID: 76ab957b35b3f34e324eb7807ff8f670@192.168.1.40:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.10.2
Date: Tue, 24 Jun 2014 19:13:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Reliably Transmitting (no NAT) to 192.168.49.170:50857:
OPTIONS sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.40:0;branch=z9hG4bK5c1669e0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.40:0>;tag=as6c6e2728
To: <sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:asterisk@192.168.1.40:0;transport=WS>
Call-ID: 191d7d067b23d889365e470a12143686@192.168.1.40:0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.10.2
Date: Tue, 24 Jun 2014 19:13:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '3022a6e538a55b1d61734231542aee2c@192.168.1.40:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.146.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK733a9728;rport=5060;received=192.168.1.40
From: "asterisk" <sip:asterisk@192.168.1.40>;tag=as556b2afe
To: <sip:192.168.146.1>;tag=1688110548
Call-ID: 76ab957b35b3f34e324eb7807ff8f670@192.168.1.40:5060
CSeq: 102 OPTIONS
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,REFER,INFO,PRACK,UPDATE
Server: HiPath 4000 V6 Common Gateway M5T SIP Stack/4.1.8.14
Content-Type: application/sdp
Content-Length: 855
v=0
o=MxSIP 0 0 IN IP4 0.0.0.0
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 5555 RTP/AVP 8 18 4 96 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 CLEARMODE/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
m=audio 5555 RTP/SAVP 8 18 4 96 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 CLEARMODE/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
m=image 5555 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:375
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (10 headers 37 lines) ---
Really destroying SIP dialog '76ab957b35b3f34e324eb7807ff8f670@192.168.1.40:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.49.170:29340:
OPTIONS sip:1111@192.168.49.170:29340;rinstance=58d56df541c956c9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK247a38af;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.40>;tag=as2833aa91
To: <sip:1111@192.168.49.170:29340;rinstance=58d56df541c956c9>
Contact: <sip:asterisk@192.168.1.40:5060>
Call-ID: 74fcd1ac21ae9f8f596ff7c94fd926b5@192.168.1.40:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.10.2
Date: Tue, 24 Jun 2014 19:13:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Jun 24 12:13:48] NOTICE[17149]: chan_sip.c:15104 sip_reregister: -- Re-registration for 102@192.168.146.1
REGISTER 10 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.146.1:5060:
REGISTER sip:192.168.146.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK38d12340;rport
Max-Forwards: 70
From: <sip:102@192.168.146.1>;tag=as0ec02069
To: <sip:102@192.168.146.1>
Call-ID: 1ba3a05b785379ab33d6f9ee3766fd4b@127.0.1.1
CSeq: 112 REGISTER
User-Agent: Asterisk PBX 11.10.2
Expires: 120
Contact: <sip:102@192.168.1.40:5060>
Content-Length: 0
---
<--- SIP read from UDP:192.168.146.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK38d12340;rport=5060;received=192.168.1.40
From: <sip:102@192.168.146.1>;tag=as0ec02069
To: <sip:102@192.168.146.1>
Call-ID: 1ba3a05b785379ab33d6f9ee3766fd4b@127.0.1.1
CSeq: 112 REGISTER
Server: HiPath 4000 V6 Common Gateway M5T SIP Stack/4.1.8.14
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.49.170:29340 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK247a38af;rport=5060
Contact: <sip:192.168.49.170:29340>
To: <sip:1111@192.168.49.170:29340;rinstance=58d56df541c956c9>;tag=17d7eb1b
From: "asterisk"<sip:asterisk@192.168.1.40>;tag=as2833aa91
Call-ID: 74fcd1ac21ae9f8f596ff7c94fd926b5@192.168.1.40:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces, eventlist
User-Agent: X-Lite release 4.6.1 stamp 73073 b19c773f-W6.1
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '74fcd1ac21ae9f8f596ff7c94fd926b5@192.168.1.40:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.146.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK38d12340;rport=5060;received=192.168.1.40
From: <sip:102@192.168.146.1>;tag=as0ec02069
To: <sip:102@192.168.146.1>;tag=537373495
Call-ID: 1ba3a05b785379ab33d6f9ee3766fd4b@127.0.1.1
CSeq: 112 REGISTER
Contact: <sip:102@192.168.1.40:5060;transport=udp>;expires=120
Expires: 120
Server: HiPath 4000 V6 Common Gateway M5T SIP Stack/4.1.8.14
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
[Jun 24 12:13:48] NOTICE[17149]: chan_sip.c:23615 handle_response_register: Outbound Registration: Expiry for 192.168.146.1 is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '1ba3a05b785379ab33d6f9ee3766fd4b@127.0.1.1' Method: REGISTER
Reliably Transmitting (no NAT) to 192.168.32.181:50632:
OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.40:0;branch=z9hG4bK5013bc04
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.40:0>;tag=as7ca80c09
To: <sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:asterisk@192.168.1.40:0;transport=WS>
Call-ID: 1c89087f5848b8bf4eaff831524705fb@192.168.1.40:0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.10.2
Date: Tue, 24 Jun 2014 19:13:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[Jun 24 12:13:48] NOTICE[17149]: chan_sip.c:15104 sip_reregister: -- Re-registration for 101@192.168.146.1
REGISTER 10 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.146.1:5060:
REGISTER sip:192.168.146.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK06394f8b;rport
Max-Forwards: 70
From: <sip:101@192.168.146.1>;tag=as5efc9f3d
To: <sip:101@192.168.146.1>
Call-ID: 6255ddb35a91645d2cac87de01ca4382@127.0.1.1
CSeq: 112 REGISTER
User-Agent: Asterisk PBX 11.10.2
Expires: 120
Contact: <sip:101@192.168.1.40:5060>
Content-Length: 0
---
<--- SIP read from WS:192.168.49.170:50857 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK5c1669e0
From: "asterisk"<sip:asterisk@192.168.1.40>;tag=as6c6e2728
To: <sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 191d7d067b23d889365e470a12143686@192.168.1.40:0
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from WS:192.168.32.181:50632 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK5013bc04
From: "asterisk"<sip:asterisk@192.168.1.40>;tag=as7ca80c09
To: <sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 1c89087f5848b8bf4eaff831524705fb@192.168.1.40:0
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.146.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK06394f8b;rport=5060;received=192.168.1.40
From: <sip:101@192.168.146.1>;tag=as5efc9f3d
To: <sip:101@192.168.146.1>
Call-ID: 6255ddb35a91645d2cac87de01ca4382@127.0.1.1
CSeq: 112 REGISTER
Server: HiPath 4000 V6 Common Gateway M5T SIP Stack/4.1.8.14
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '191d7d067b23d889365e470a12143686@192.168.1.40:0' Method: OPTIONS
Really destroying SIP dialog '1c89087f5848b8bf4eaff831524705fb@192.168.1.40:0' Method: OPTIONS
<--- SIP read from UDP:192.168.146.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK06394f8b;rport=5060;received=192.168.1.40
From: <sip:101@192.168.146.1>;tag=as5efc9f3d
To: <sip:101@192.168.146.1>;tag=9131883
Call-ID: 6255ddb35a91645d2cac87de01ca4382@127.0.1.1
CSeq: 112 REGISTER
Contact: <sip:101@192.168.1.40:5060;transport=udp>;expires=120
Expires: 120
Server: HiPath 4000 V6 Common Gateway M5T SIP Stack/4.1.8.14
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
[Jun 24 12:13:49] NOTICE[17149]: chan_sip.c:23615 handle_response_register: Outbound Registration: Expiry for 192.168.146.1 is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '6255ddb35a91645d2cac87de01ca4382@127.0.1.1' Method: REGISTER
<--- SIP read from WS:192.168.32.181:50632 --->
INVITE sip:1114@192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFqm1iThoCt3Josc9pgvZCnAH2eBlq6a7;rport
From: "1113"<sip:1113@192.168.1.40>;tag=P1HFRB7d6FlnWT4cQ8Tu
To: <sip:1114@192.168.1.40>
Contact: "1113"<sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1113;ha1=bd869170f0156572561fba91dbc4acac;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073
CSeq: 32986 INVITE
Content-Type: application/sdp
Content-Length: 1767
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
v=0
o=- 7573769006897112000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f
m=audio 35209 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 <public ip>
a=rtcp:35209 IN IP4 <public ip>
a=candidate:2942323686 1 udp 2122260223 192.168.32.181 35209 typ host generation 0
a=candidate:2942323686 2 udp 2122260223 192.168.32.181 35209 typ host generation 0
a=candidate:3789797142 1 tcp 1518280447 192.168.32.181 0 typ host generation 0
a=candidate:3789797142 2 tcp 1518280447 192.168.32.181 0 typ host generation 0
a=candidate:1676506229 1 udp 1686052607 <public ip> 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0
a=candidate:1676506229 2 udp 1686052607 <public ip> 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0
a=ice-ufrag:cmS2Mdkr3dIJVTVc
a=ice-pwd:1JZIAYodAALxo5r8YA8D18et
a=ice-options:google-ice
a=fingerprint:sha-256 37:63:96:1E:22:17:D1:45:68:02:F8:9F:28:37:6A:1C:3F:36:51:B2:1E:A8:B4:A3:D6:BE:E9:5F:1F:B1:28:6C
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2622176125 cname:RjotqUOp9+pcE8Ei
a=ssrc:2622176125 msid:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f 2d3dc4e5-a26d-46e7-a282-0af274c98081
a=ssrc:2622176125 mslabel:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f
a=ssrc:2622176125 label:2d3dc4e5-a26d-46e7-a282-0af274c98081
<------------->
--- (11 headers 40 lines) ---
Using INVITE request as basis request - 458d1196-f111-71e2-98f7-d30d4e4ad073
Found peer '1113' for '1113' from 192.168.32.181:50632
<--- Reliably Transmitting (no NAT) to 192.168.32.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFqm1iThoCt3Josc9pgvZCnAH2eBlq6a7;rport;received=192.168.32.181
From: "1113"<sip:1113@192.168.1.40>;tag=P1HFRB7d6FlnWT4cQ8Tu
To: <sip:1114@192.168.1.40>;tag=as050b0fdb
Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073
CSeq: 32986 INVITE
Server: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d058aba"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '458d1196-f111-71e2-98f7-d30d4e4ad073' in 6400 ms (Method: INVITE)
<--- SIP read from WS:192.168.32.181:50632 --->
ACK sip:1114@192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFqm1iThoCt3Josc9pgvZCnAH2eBlq6a7;rport
From: "1113"<sip:1113@192.168.1.40>;tag=P1HFRB7d6FlnWT4cQ8Tu
To: <sip:1114@192.168.1.40>;tag=as050b0fdb
Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073
CSeq: 32986 ACK
Content-Length: 0
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from WS:192.168.32.181:50632 --->
INVITE sip:1114@192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLLmfsw1TX71X126dJCaJrNvFPNoXA1rI;rport
From: "1113"<sip:1113@192.168.1.40>;tag=P1HFRB7d6FlnWT4cQ8Tu
To: <sip:1114@192.168.1.40>
Contact: "1113"<sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1113;ha1=bd869170f0156572561fba91dbc4acac;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073
CSeq: 32987 INVITE
Content-Type: application/sdp
Content-Length: 1767
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1113",realm="asterisk",nonce="1d058aba",uri="sip:1114@192.168.1.40",response="c4e16e428c3f5266f989376016bb04a1",algorithm=MD5
v=0
o=- 7573769006897112000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f
m=audio 35209 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 <public ip>
a=rtcp:35209 IN IP4 <public ip>
a=candidate:2942323686 1 udp 2122260223 192.168.32.181 35209 typ host generation 0
a=candidate:2942323686 2 udp 2122260223 192.168.32.181 35209 typ host generation 0
a=candidate:3789797142 1 tcp 1518280447 192.168.32.181 0 typ host generation 0
a=candidate:3789797142 2 tcp 1518280447 192.168.32.181 0 typ host generation 0
a=candidate:1676506229 1 udp 1686052607 <public ip> 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0
a=candidate:1676506229 2 udp 1686052607 <public ip> 35209 typ srflx raddr 192.168.32.181 rport 35209 generation 0
a=ice-ufrag:cmS2Mdkr3dIJVTVc
a=ice-pwd:1JZIAYodAALxo5r8YA8D18et
a=ice-options:google-ice
a=fingerprint:sha-256 37:63:96:1E:22:17:D1:45:68:02:F8:9F:28:37:6A:1C:3F:36:51:B2:1E:A8:B4:A3:D6:BE:E9:5F:1F:B1:28:6C
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2622176125 cname:RjotqUOp9+pcE8Ei
a=ssrc:2622176125 msid:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f 2d3dc4e5-a26d-46e7-a282-0af274c98081
a=ssrc:2622176125 mslabel:kzkC0YG0wuA5QmaaCMoXAg7r9EOeS4R6RR4f
a=ssrc:2622176125 label:2d3dc4e5-a26d-46e7-a282-0af274c98081
<------------->
--- (12 headers 40 lines) ---
Using INVITE request as basis request - 458d1196-f111-71e2-98f7-d30d4e4ad073
Found peer '1113' for '1113' from 192.168.32.181:50632
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
[Jun 24 12:14:08] WARNING[17246][C-00000008]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 35209 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
<--- Reliably Transmitting (no NAT) to 192.168.32.181:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLLmfsw1TX71X126dJCaJrNvFPNoXA1rI;rport;received=192.168.32.181
From: "1113"<sip:1113@192.168.1.40>;tag=P1HFRB7d6FlnWT4cQ8Tu
To: <sip:1114@192.168.1.40>;tag=as050b0fdb
Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073
CSeq: 32987 INVITE
Server: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '458d1196-f111-71e2-98f7-d30d4e4ad073' in 6400 ms (Method: INVITE)
<--- SIP read from WS:192.168.32.181:50632 --->
ACK sip:1114@192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLLmfsw1TX71X126dJCaJrNvFPNoXA1rI;rport
From: "1113"<sip:1113@192.168.1.40>;tag=P1HFRB7d6FlnWT4cQ8Tu
To: <sip:1114@192.168.1.40>;tag=as050b0fdb
Call-ID: 458d1196-f111-71e2-98f7-d30d4e4ad073
CSeq: 32987 ACK
Content-Length: 0
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.49.170:29340 --->
<------------->
Really destroying SIP dialog '458d1196-f111-71e2-98f7-d30d4e4ad073' Method: INVITE
<--- SIP read from WS:192.168.32.181:50632 --->
REGISTER sip:192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKbsns9wccafSW0rd6Nq3wfVbXs71m9NCN;rport
From: "1113"<sip:1113@192.168.1.40>;tag=pC5Duy9NZDhiNnn6yGSa
To: "1113"<sip:1113@192.168.1.40>
Contact: "1113"<sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a
CSeq: 45050 REGISTER
Content-Length: 0
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1113",realm="asterisk",nonce="5540eea5",uri="sip:192.168.1.40",response="1a1fd92101d6d73b3dd2d747639481b9",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
<--- Transmitting (no NAT) to 192.168.32.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKbsns9wccafSW0rd6Nq3wfVbXs71m9NCN;rport;received=192.168.32.181
From: "1113"<sip:1113@192.168.1.40>;tag=pC5Duy9NZDhiNnn6yGSa
To: "1113"<sip:1113@192.168.1.40>;tag=as28900697
Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a
CSeq: 45050 REGISTER
Server: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f05e66e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0c390373-c147-1938-d3e0-ff6cf318ad1a' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:192.168.32.181:50632 --->
REGISTER sip:192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKnHBUoSQAFP7h9nSk8l0nAJMpx9xfKha;rport
From: "1113"<sip:1113@192.168.1.40>;tag=pC5Duy9NZDhiNnn6yGSa
To: "1113"<sip:1113@192.168.1.40>
Contact: "1113"<sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a
CSeq: 45051 REGISTER
Content-Length: 0
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1113",realm="asterisk",nonce="2f05e66e",uri="sip:192.168.1.40",response="d580c873292ee99098c1620978ea7162",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.32.181:50632:
OPTIONS sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.40:0;branch=z9hG4bK7bc122a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.40:0>;tag=as58b7571b
To: <sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:asterisk@192.168.1.40:0;transport=WS>
Call-ID: 4096ae65042e1c727a8047870a5739d7@192.168.1.40:0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.10.2
Date: Tue, 24 Jun 2014 19:14:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.32.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKnHBUoSQAFP7h9nSk8l0nAJMpx9xfKha;rport;received=192.168.32.181
From: "1113"<sip:1113@192.168.1.40>;tag=pC5Duy9NZDhiNnn6yGSa
To: "1113"<sip:1113@192.168.1.40>;tag=as28900697
Call-ID: 0c390373-c147-1938-d3e0-ff6cf318ad1a
CSeq: 45051 REGISTER
Server: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Date: Tue, 24 Jun 2014 19:14:19 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0c390373-c147-1938-d3e0-ff6cf318ad1a' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:192.168.32.181:50632 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK7bc122a0
From: "asterisk"<sip:asterisk@192.168.1.40>;tag=as58b7571b
To: <sip:1113@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 4096ae65042e1c727a8047870a5739d7@192.168.1.40:0
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '4096ae65042e1c727a8047870a5739d7@192.168.1.40:0' Method: OPTIONS
<--- SIP read from WS:192.168.49.170:50857 --->
REGISTER sip:192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB9BJNx0W5ug7cD6g5FoRQUPi7OcRMpyD;rport
From: "1114"<sip:1114@192.168.1.40>;tag=1SshOviPeDc9IlP1923w
To: "1114"<sip:1114@192.168.1.40>
Contact: "1114"<sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: fabcf772-700b-08a6-d6d9-c1704e819602
CSeq: 17580 REGISTER
Content-Length: 0
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1114",realm="asterisk",nonce="261810ac",uri="sip:192.168.1.40",response="54885312403fe536c5c90c95e40673a4",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
<--- Transmitting (no NAT) to 192.168.49.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB9BJNx0W5ug7cD6g5FoRQUPi7OcRMpyD;rport;received=192.168.49.170
From: "1114"<sip:1114@192.168.1.40>;tag=1SshOviPeDc9IlP1923w
To: "1114"<sip:1114@192.168.1.40>;tag=as74149598
Call-ID: fabcf772-700b-08a6-d6d9-c1704e819602
CSeq: 17580 REGISTER
Server: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63ebdf4e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'fabcf772-700b-08a6-d6d9-c1704e819602' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:192.168.49.170:50857 --->
REGISTER sip:192.168.1.40 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKS2laBiElXBdcWBuRS0uT8NXzGxDQ1o2J;rport
From: "1114"<sip:1114@192.168.1.40>;tag=1SshOviPeDc9IlP1923w
To: "1114"<sip:1114@192.168.1.40>
Contact: "1114"<sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: fabcf772-700b-08a6-d6d9-c1704e819602
CSeq: 17581 REGISTER
Content-Length: 0
Route: <sip:192.168.1.40:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1114",realm="asterisk",nonce="63ebdf4e",uri="sip:192.168.1.40",response="69ce1525c64b795d653d96ebabec70f7",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.49.170:50857:
OPTIONS sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.40:0;branch=z9hG4bK1c0e1d2e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.40:0>;tag=as3a656297
To: <sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:asterisk@192.168.1.40:0;transport=WS>
Call-ID: 21860ae1563011e62678668c2df4dad2@192.168.1.40:0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.10.2
Date: Tue, 24 Jun 2014 19:14:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.49.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKS2laBiElXBdcWBuRS0uT8NXzGxDQ1o2J;rport;received=192.168.49.170
From: "1114"<sip:1114@192.168.1.40>;tag=1SshOviPeDc9IlP1923w
To: "1114"<sip:1114@192.168.1.40>;tag=as74149598
Call-ID: fabcf772-700b-08a6-d6d9-c1704e819602
CSeq: 17581 REGISTER
Server: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Date: Tue, 24 Jun 2014 19:14:33 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'fabcf772-700b-08a6-d6d9-c1704e819602' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:192.168.49.170:50857 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 192.168.1.40;branch=z9hG4bK1c0e1d2e
From: "asterisk"<sip:asterisk@192.168.1.40>;tag=as3a656297
To: <sip:1114@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 21860ae1563011e62678668c2df4dad2@192.168.1.40:0
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '21860ae1563011e62678668c2df4dad2@192.168.1.40:0' Method: OPTIONS
<--- SIP read from UDP:192.168.49.170:29340 --->