Hello everyone, I have a problem with asterisk v 11.4.0, basically I use SIPml5.api to call a cellphone the call connects no problem, but the only person who can hear the audio is the caller on the PC (calling from the api), when we try the call from within our private networks the audio works both ways
– SIP CONF –
[general]
context=default
allowguest=no=no
realm=doubango.org
udpbindaddr=0.0.0.0:5060
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp,ws,wss
srvlookup=no – RTP CONF –
rtpstart=10000
rtpend=65525
stunaddr=stun.l.google.com:19302– USERS CONF –
[1060]
type=peer
host=dynamic
username=1060
secret=1060
context=default
hasiax=no
hassip=yes
encryption=yes
avpf=yes
icesupport=yes
videosupport=no
directmedia=no
[1061]
type=peer
host=dynamic
username=1061
secret=1061
context=default
hasiax=no
hassip=yes
encryption=yes
avpf=yes
icesupport=yes
videosupport=no
directmedia=no
[100]
nat=force_rport, comedia
qualify=yes
type=friend
host=dynamic
context=default
username=100
secret=100
fromdomain=10.168.1.3 // OUR GSM GATEWAY
icesupport=yes
videosupport=no– SIPml5 JS Console –
SIPML5 API version = 1.2.185 SIPml-api.js:826
User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/34.0.1847.137 Safari/537.36 SIPml-api.js:826
WebSocket supported = yes SIPml-api.js:826
Navigator friendly name = chrome SIPml-api.js:826
OS friendly name = windows SIPml-api.js:826
Have WebRTC = yes SIPml-api.js:826
Have GUM = yes SIPml-api.js:826
Engine initialized SIPml-api.js:826
s_websocket_server_url=ws://xx.xx.xx.xx:8088/ws SIPml-api.js:826
s_sip_outboundproxy_url=udp://xx.xx.xx.xx:5060 SIPml-api.js:826
b_rtcweb_breaker_enabled=no SIPml-api.js:826
b_click2call_enabled=no SIPml-api.js:826
SIP stack start: proxy='ns313841.ovh.net:12060', realm='<sip:xx.xx.xx.xx>', impi='1060', impu='"1060"<sip:1060@xx.xx.xx.xx>' SIPml-api.js:826
Connecting to 'ws://xx.xx.xx.xx:8088/ws' SIPml-api.js:826
==stack event = starting SIPml-api.js:826
__tsip_transport_ws_onopen SIPml-api.js:826
==stack event = started SIPml-api.js:826
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister SIPml-api.js:826
SEND: REGISTER sip:xx.xx.xx.xxSIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKpN16OmCm5ThlX8LvROWTgT2GM9aME8zI;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=DTKz94TI0edrcC5PvD7l
To: "1060"<sip:1060@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: ffb2762b-3659-4f7f-70b2-72430dee3169
CSeq: 15834 REGISTER
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
Supported: path
SIPml-api.js:826
==session event = connecting SIPml-api.js:826
==session event = sent_request SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=yy.yy.yy.yy;branch=z9hG4bKpN16OmCm5ThlX8LvROWTgT2GM9aME8zI
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=DTKz94TI0edrcC5PvD7l
To: "1060"<sip:1060@xx.xx.xx.xx>;tag=as7d477007
Call-ID: ffb2762b-3659-4f7f-70b2-72430dee3169
CSeq: 15834 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="doubango.org",nonce="199a8dfc",stale=FALSE,algorithm=MD5
SIPml-api.js:826
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 SIPml-api.js:826
SEND: REGISTER sip:xx.xx.xx.xxSIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKsOMNqs3ByFzAZizetg8cOi6WubBFNhkK;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=DTKz94TI0edrcC5PvD7l
To: "1060"<sip:1060@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: ffb2762b-3659-4f7f-70b2-72430dee3169
CSeq: 15835 REGISTER
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="doubango.org",nonce="199a8dfc",uri="sip:xx.xx.xx.xx",response="e257ef0ae56f23d5fd02290806ddbb0c",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
Supported: path
SIPml-api.js:826
==session event = sent_request SIPml-api.js:826
State machine: c0000_Started_2_Outgoing_X_oINVITE SIPml-api.js:826
PeerConnectionClass = function RTCPeerConnection() { [native code] } SessionDescriptionClass = function RTCSessionDescription() { [native code] } IceCandidateClass = function RTCIceCandidate() { [native code] } SIPml-api.js:826
ICE servers:[{"url":"stun:stun.l.google.com:19302"}] SIPml-api.js:826
==stack event = m_permission_requested SIPml-api.js:826
==session event = connecting SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=46.135.90.7;branch=z9hG4bKsOMNqs3ByFzAZizetg8cOi6WubBFNhkK
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=DTKz94TI0edrcC5PvD7l
To: "1060"<sip:1060@xx.xx.xx.xx>;tag=as7d477007
Contact: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Call-ID: ffb2762b-3659-4f7f-70b2-72430dee3169
CSeq: 15835 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 20 May 2014 11:53:15 GMT;20
SIPml-api.js:826
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=NOTIFY sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.112:5060;branch=z9hG4bK3644d9ab
From: "asterisk"<sip:asterisk@10.168.1.112>;tag=as6064d973
To: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:asterisk@10.168.1.112:5060;transport=WS>
Call-ID: 4bc9350375e556bf0221338953e0fb2a@10.168.1.112:5060
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 105
Max-Forwards: 70
User-Agent: Asterisk PBX 11.4.0
Event: message-summary
Messages-Waiting: no
Message-Account: sip:asterisk@10.168.1.112;transport=WS
Voice-Message: 0/0 (0/0)
SIPml-api.js:826
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 10.168.1.112:5060;branch=z9hG4bK3644d9ab
From: "asterisk"<sip:asterisk@10.168.1.112>;tag=as6064d973
To: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Call-ID: 4bc9350375e556bf0221338953e0fb2a@10.168.1.112:5060
CSeq: 102 NOTIFY
Content-Length: 0
SIPml-api.js:826
==session event = connected SIPml-api.js:826
onGetUserMediaSuccess SIPml-api.js:826
createOffer SIPml-api.js:826
==stack event = m_permission_accepted SIPml-api.js:826
==session event = m_stream_audio_local_added SIPml-api.js:826
onCreateSdpSuccess SIPml-api.js:826
onSetLocalDescriptionSuccess SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
onIceCandidate = undefined SIPml-api.js:826
ICE GATHERING COMPLETED! SIPml-api.js:826
onIceGatheringCompleted SIPml-api.js:826
SEND: INVITE sip:123456789@xx.xx.xx.xxSIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKotMiesGoZK1Pu8H3eZps243HvfiUSJDk;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:123456789@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29844 INVITE
Content-Type: application/sdp
Content-Length: 2160
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
v=0
o=- 5129446939154120000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Y2oIDXk8xtv2DEe0tT40R9R3e3CbzNEB0WXn
m=audio 56388 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 yy.yy.yy.yy
a=rtcp:56388 IN IP4 yy.yy.yy.yy
a=candidate:1963743260 1 udp 2122260223 192.168.43.20 54398 typ host generation 0
a=candidate:1963743260 2 udp 2122260223 192.168.43.20 54398 typ host generation 0
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54399 typ host generation 0
a=candidate:2999745851 2 udp 2122194687 192.168.56.1 54399 typ host generation 0
a=candidate:999269612 1 tcp 1518280447 192.168.43.20 0 typ host generation 0
a=candidate:999269612 2 tcp 1518280447 192.168.43.20 0 typ host generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:4098121384 1 udp 1686052607 y.yy.yy.yy 56388 typ srflx raddr 192.168.43.20 rport 54398 generation 0
a=candidate:4098121384 2 udp 1686052607 yy.yy.yy.yy 56388 typ srflx raddr 192.168.43.20 rport 54398 generation 0
a=ice-ufrag:/0i2Zo/awOIb1eXD
a=ice-pwd:AlOvtDgJxPiryQzjNKXsU3XX
a=ice-options:google-ice
a=fingerprint:sha-256 EA:4D:9C:6E:9C:81:EF:E9:5A:DA:94:06:16:00:80:FD:CA:B1:DE:74:6D:59:D9:C2:A7:ED:25:B9:C5:C5:4F:1B
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:DkbBfprOz0Uipzpq5+/AfAHJ8Ee2xpN/wi2Yi3fJ
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:D81FnIDSh+Dbo/RMbqoaPmdeP5gX5bUrbd3aKWSL
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:346501379 cname:f7KCkXIFOdSksWDq
a=ssrc:346501379 msid:Y2oIDXk8xtv2DEe0tT40R9R3e3CbzNEB0WXn 4ba9cb7c-6c00-4b0f-b4dd-ded3fc7ef283
a=ssrc:346501379 mslabel:Y2oIDXk8xtv2DEe0tT40R9R3e3CbzNEB0WXn
a=ssrc:346501379 label:4ba9cb7c-6c00-4b0f-b4dd-ded3fc7ef283
SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=yy.yy.yy.yy;branch=z9hG4bKotMiesGoZK1Pu8H3eZps243HvfiUSJDk
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:123456789@xx.xx.xx.xx>;tag=as240f9827
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29844 INVITE
Content-Length: 0
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="doubango.org",nonce="46d7688a",stale=FALSE,algorithm=MD5
SIPml-api.js:826
SEND: ACK sip:732604400@xx.xx.xx.xxSIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKotMiesGoZK1Pu8H3eZps243HvfiUSJDk;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:123456789@xx.xx.xx.xx>;tag=as240f9827
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29844 ACK
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
SIPml-api.js:826
State machine: x0000_Any_2_Any_X_i401_407_INVITE SIPml-api.js:826
SEND: INVITE sip:732604400@178.72.192.195 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKuaCSKlhNmvCJBi8NIoCkFBZiTBtLE4Vm;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:1234596789@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29845 INVITE
Content-Type: application/sdp
Content-Length: 2160
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="doubango.org",nonce="46d7688a",uri="sip:123456789@xx.xx.xx.xx",response="369f51814e329a84cfb24c7602a21983",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
v=0
o=- 5129446939154120000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Y2oIDXk8xtv2DEe0tT40R9R3e3CbzNEB0WXn
m=audio 56388 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 yy.yy.yy.yy
a=rtcp:56388 IN IP4 yy.yy.yy.yy
a=candidate:1963743260 1 udp 2122260223 192.168.43.20 54398 typ host generation 0
a=candidate:1963743260 2 udp 2122260223 192.168.43.20 54398 typ host generation 0
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54399 typ host generation 0
a=candidate:2999745851 2 udp 2122194687 192.168.56.1 54399 typ host generation 0
a=candidate:999269612 1 tcp 1518280447 192.168.43.20 0 typ host generation 0
a=candidate:999269612 2 tcp 1518280447 192.168.43.20 0 typ host generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:4098121384 1 udp 1686052607 yy.yy.yy.yy 56388 typ srflx raddr 192.168.43.20 rport 54398 generation 0
a=candidate:4098121384 2 udp 1686052607 yy.yy.yy.yy 56388 typ srflx raddr 192.168.43.20 rport 54398 generation 0
a=ice-ufrag:/0i2Zo/awOIb1eXD
a=ice-pwd:AlOvtDgJxPiryQzjNKXsU3XX
a=ice-options:google-ice
a=fingerprint:sha-256 EA:4D:9C:6E:9C:81:EF:E9:5A:DA:94:06:16:00:80:FD:CA:B1:DE:74:6D:59:D9:C2:A7:ED:25:B9:C5:C5:4F:1B
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:DkbBfprOz0Uipzpq5+/AfAHJ8Ee2xpN/wi2Yi3fJ
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:D81FnIDSh+Dbo/RMbqoaPmdeP5gX5bUrbd3aKWSL
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:346501379 cname:f7KCkXIFOdSksWDq
a=ssrc:346501379 msid:Y2oIDXk8xtv2DEe0tT40R9R3e3CbzNEB0WXn 4ba9cb7c-6c00-4b0f-b4dd-ded3fc7ef283
a=ssrc:346501379 mslabel:Y2oIDXk8xtv2DEe0tT40R9R3e3CbzNEB0WXn
a=ssrc:346501379 label:4ba9cb7c-6c00-4b0f-b4dd-ded3fc7ef283
SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=yy.yy.yy.yy;branch=z9hG4bKuaCSKlhNmvCJBi8NIoCkFBZiTBtLE4Vm
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:123456789@xx.xx.xx.xx>
Contact: <sip:123456789@10.168.1.112:5060;transport=WS>
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29845 INVITE
Content-Length: 0
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:826
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=yy.yy.yy.yy;branch=z9hG4bKuaCSKlhNmvCJBi8NIoCkFBZiTBtLE4Vm
From: "1060"<sip:1060@178.72.192.195>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:123456789@xx.xx.xx.xx>;tag=as4de1de5d
Contact: <sip:123456789@10.168.1.112:5060;transport=WS>
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29845 INVITE
Content-Length: 0
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:826
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:826
==session event = i_ao_request SIPml-api.js:826
==session event = i_ao_request SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=yy.yy.yy.yy;branch=z9hG4bKuaCSKlhNmvCJBi8NIoCkFBZiTBtLE4Vm
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:123456789@x.xx.xx.xx>;tag=as4de1de5d
Contact: <sip:123456789@10.168.1.112:5060;transport=WS>
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29845 INVITE
Content-Length: 0
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:826
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:826
==session event = i_ao_request SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=yy.yy.yy.yy;branch=z9hG4bKuaCSKlhNmvCJBi8NIoCkFBZiTBtLE4Vm
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:123456789@xx.xx.xx.xx>;tag=as4de1de5d
Contact: <sip:123456789@10.168.1.112:5060;transport=WS>
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29845 INVITE
Content-Type: application/sdp
Content-Length: 685
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
v=0
o=root 751573950 751573950 IN IP4 10.168.1.112
s=Asterisk PBX 11.4.0
c=IN IP4 10.168.1.112
t=0 0
m=audio 14270 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:4e9f68436a28083154c40d7f0c33b239
a=ice-pwd:0f4ee4c61cb8fef5486aad166af6aaf7
a=candidate:Haa80170 1 UDP 2130706431 10.168.1.112 14270 typ host
a=candidate:Sb248c0c3 1 UDP 1694498815 xx.xx.xx.xx 14270 typ srflx
a=candidate:Haa80170 2 UDP 2130706430 10.168.1.112 14271 typ host
a=candidate:Sb248c0c3 2 UDP 1694498814 xx.xx.xx.xx 14270 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:9pdtMT1x749Dl+2mgB7Cbc0B6UDDX6POxOY6SInA
SIPml-api.js:826
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE SIPml-api.js:826
setRemoteDescription(answer)
v=0
o=root 751573950 751573950 IN IP4 10.168.1.112
s=Asterisk PBX 11.4.0
c=IN IP4 10.168.1.112
t=0 0
m=audio 14270 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:4e9f68436a28083154c40d7f0c33b239
a=ice-pwd:0f4ee4c61cb8fef5486aad166af6aaf7
a=candidate:Haa80170 1 UDP 2130706431 10.168.1.112 14270 typ host
a=candidate:Sb248c0c3 1 UDP 1694498815 xx.xx.xx.xx 14270 typ srflx
a=candidate:Haa80170 2 UDP 2130706430 10.168.1.112 14271 typ host
a=candidate:Sb248c0c3 2 UDP 1694498814 xx.xx.xx.xx 14270 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:9pdtMT1x749Dl+2mgB7Cbc0B6UDDX6POxOY6SInA
SIPml-api.js:826
SEND: ACK sip:732604400@10.168.1.112:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK5M5sFeAJpFrQhqvlhlQX;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
To: <sip:732604400@xx.xx.xx.xx>;tag=as4de1de5d
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 29845 ACK
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="doubango.org",nonce="46d7688a",uri="sip:123456789@10.168.1.112:5060;transport=WS",response="6939379cc569c716bac4f98f1b37bd79",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
SIPml-api.js:826
__on_add_stream SIPml-api.js:826
==session event = m_early_media SIPml-api.js:826
onSetRemoteDescriptionSuccess SIPml-api.js:826
==session event = connected SIPml-api.js:826
==session event = m_stream_audio_remote_added SIPml-api.js:826
__tsip_transport_ws_onmessage SIPml-api.js:826
recv=BYE sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.112:5060;branch=z9hG4bK031febbc
From: <sip:123456789@xx.xx.xx.xx>;tag=as4de1de5d
To: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 102 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.4.0
Proxy-Authorization: Digest username="1060",realm="doubango.org",nonce="46d7688a",uri="sip:xx.xx.xx.xx",response="022c47464d80d9d1c2d855561f99243b",algorithm=MD5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
SIPml-api.js:826
State machine: x0000_Any_2_Terminated_X_iBYE SIPml-api.js:826
=== INVITE Dialog terminated === SIPml-api.js:826
PeerConnection::stop() SIPml-api.js:826
SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.168.1.112:5060;branch=z9hG4bK031febbc
From: <sip:123456789@xx.xx.xx.xx>;tag=as4de1de5d
To: "1060"<sip:1060@xx.xx.xx.xx>;tag=VE1c90Vm4KKo7oNZVbR4
Contact: <sip:1060@df7jal23ls0d.invalid;transport=ws>
Call-ID: 4dea1b67-373c-df90-a6c6-daf76a853ce8
CSeq: 102 BYE
Content-Length: 0
SIPml-api.js:826
==session event = terminated SIPml-api.js:826– Asterisk Console –
<--- SIP read from UDP:10.168.1.3:5060 --->
REGISTER sip:10.168.1.101:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6825353319442
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4863-150f3a12
To: <sip:100@10.168.1.101:5060>
Call-ID: 7c1eb3fe34bb-4839-150f3a12@10.168.1.101
CSeq: 6807 REGISTER
Contact: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Expires: 60
User-Agent: 2N VBN 1.18.1.21.6
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 10.168.1.3:5060 (no NAT)
Sending to 10.168.1.3:5060 (no NAT)
<--- Transmitting (NAT) to 10.168.1.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6825353319442;received=10.168.1.3;rport=5060
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4863-150f3a12
To: <sip:100@10.168.1.101:5060>;tag=as1e420547
Call-ID: 7c1eb3fe34bb-4839-150f3a12@10.168.1.101
CSeq: 6807 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="doubango.org", nonce="1ab5d2eb"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7c1eb3fe34bb-4839-150f3a12@10.168.1.101' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.168.1.3:5060 --->
REGISTER sip:10.168.1.101:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6826353319442
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4863-150f3a12
To: <sip:100@10.168.1.101:5060>
Call-ID: 7c1eb3fe34bb-4839-150f3a12@10.168.1.101
CSeq: 6808 REGISTER
Contact: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Authorization: Digest username="100",realm="doubango.org",nonce="1ab5d2eb",uri="sip:10.168.1.101",response="ded6353e5fa75286d8f94d18fd70795c",algorithm=MD5
Expires: 60
User-Agent: 2N VBN 1.18.1.21.6
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 10.168.1.3:5060 (NAT)
Reliably Transmitting (NAT) to 10.168.1.3:5060:
OPTIONS sip:100@10.168.1.3:5060;user=phone;phone-context=100 SIP/2.0
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK3dbde3ce;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.168.1.101>;tag=as540ca4f2
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Contact: <sip:asterisk@10.168.1.101:5060>
Call-ID: 28cc54f84927619730a3540f62ad4a8b@10.168.1.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.4.0
Date: Wed, 21 May 2014 13:53:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (NAT) to 10.168.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6826353319442;received=10.168.1.3;rport=5060
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4863-150f3a12
To: <sip:100@10.168.1.101:5060>;tag=as1e420547
Call-ID: 7c1eb3fe34bb-4839-150f3a12@10.168.1.101
CSeq: 6808 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>;expires=60
Date: Wed, 21 May 2014 13:53:42 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '35a867177c698c3a1e13a77d62759807@10.168.1.3' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 10.168.1.3:5060:
NOTIFY sip:100@10.168.1.3:5060;user=phone;phone-context=100 SIP/2.0
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK5e79a544;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.168.1.3>;tag=as55c22e73
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Contact: <sip:asterisk@10.168.1.101:5060>
Call-ID: 35a867177c698c3a1e13a77d62759807@10.168.1.3
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.4.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90
Messages-Waiting: no
Message-Account: sip:asterisk@10.168.1.3
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '7c1eb3fe34bb-4839-150f3a12@10.168.1.101' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.168.1.3:5060 --->
SIP/2.0 200 OK
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK3dbde3ce;rport
From: "asterisk" <sip:asterisk@10.168.1.101>;tag=as540ca4f2
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Call-ID: 28cc54f84927619730a3540f62ad4a8b@10.168.1.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.4.0
Date: Wed, 21 May 2014 13:53:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '28cc54f84927619730a3540f62ad4a8b@10.168.1.101:5060' Method: OPTIONS
<--- SIP read from UDP:10.168.1.3:5060 --->
SIP/2.0 481 Call Does Not Exist
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK5e79a544;rport
From: "asterisk" <sip:asterisk@10.168.1.3>;tag=as55c22e73
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Call-ID: 35a867177c698c3a1e13a77d62759807@10.168.1.3
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.4.0
Event: message-summary
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '35a867177c698c3a1e13a77d62759807@10.168.1.3' Method: NOTIFY
== Manager 'manager' logged on from xx.xx.xx.xx
-- Added extension '123456789' priority 1 to default
== Manager 'manager' logged off from xx.xx.xx.xx
== Manager 'manager' logged on from xx.xx.xx.xx
-- Added extension '123456789' priority 2 to default
== Manager 'manager' logged off from xx.xx.xx.xx
== Manager 'manager' logged on from xx.xx.xx.xx
== Manager 'manager' logged off from xx.xx.xx.xx
== Manager 'manager' logged on from xx.xx.xx.xx
-- Added extension '88123456789' priority 1 to default
== Manager 'manager' logged off from xx.xx.xx.xx
== WebSocket connection from 'yy.yy.yy.yy:20192' for protocol 'sip' accepted using version '13'
<--- SIP read from WS:yy.yy.yy.yy:20192 --->
REGISTER sip:xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKWwxWATaWTtLaKZJmbpDTsikTPVnJzk6k;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=sVEiIb6Xx3DOhTG0qZai
To: "1060"<sip:1060@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: f8184a08-07cf-558b-9d46-fba19e8e9aef
CSeq: 58461 REGISTER
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
Supported: path
<------------->
--- (13 headers 0 lines) ---
<--- Transmitting (no NAT) to yy.yy.yy.yy:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKWwxWATaWTtLaKZJmbpDTsikTPVnJzk6k;rport;received=yy.yy.yy.yy
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=sVEiIb6Xx3DOhTG0qZai
To: "1060"<sip:1060@xx.xx.xx.xx>;tag=as3f7fe36e
Call-ID: f8184a08-07cf-558b-9d46-fba19e8e9aef
CSeq: 58461 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="doubango.org", nonce="3fa9441a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'f8184a08-07cf-558b-9d46-fba19e8e9aef' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:yy.yy.yy.yy:20192 --->
REGISTER sip:178.72.192.195 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcZNgIGqigboeWQ71qqd9Q1Syfq2CUdvK;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=sVEiIb6Xx3DOhTG0qZai
To: "1060"<sip:1060@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: f8184a08-07cf-558b-9d46-fba19e8e9aef
CSeq: 58462 REGISTER
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="doubango.org",nonce="3fa9441a",uri="sip:xx.xx.xx.xx",response="d2d56e67458af4dfa5e508ca571895c1",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
Supported: path
<------------->
--- (14 headers 0 lines) ---
-- Registered SIP '1060' at yy.yy.yy.yy:20192
<--- Transmitting (no NAT) to yy.yy.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcZNgIGqigboeWQ71qqd9Q1Syfq2CUdvK;rport;received=yy.yy.yy.yy
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=sVEiIb6Xx3DOhTG0qZai
To: "1060"<sip:1060@xx.xx.xx.xx>;tag=as3f7fe36e
Call-ID: f8184a08-07cf-558b-9d46-fba19e8e9aef
CSeq: 58462 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Date: Wed, 21 May 2014 13:53:58 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7869ab652cb5563c6b5ec4f276b2f53d@10.168.1.101:5060' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to yy.yy.yy.yy:20192:
NOTIFY sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.101:5060;branch=z9hG4bK7a0f8f8a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.168.1.101>;tag=as1af55445
To: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:asterisk@10.168.1.101:5060;transport=WS>
Call-ID: 7869ab652cb5563c6b5ec4f276b2f53d@10.168.1.101:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.4.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 105
Messages-Waiting: no
Message-Account: sip:asterisk@10.168.1.101;transport=WS
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog 'f8184a08-07cf-558b-9d46-fba19e8e9aef' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:yy.yy.yy.yy:20192 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 10.168.1.101:5060;branch=z9hG4bK7a0f8f8a
From: "asterisk"<sip:asterisk@10.168.1.101>;tag=as1af55445
To: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Call-ID: 7869ab652cb5563c6b5ec4f276b2f53d@10.168.1.101:5060
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '7869ab652cb5563c6b5ec4f276b2f53d@10.168.1.101:5060' Method: NOTIFY
<--- SIP read from WS:yy.yy.yy.yy:20192 --->
INVITE sip:123456789@xx.xx.xx.xxSIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK5WkV84CH7th8WrFiKs4dersvITs2PES4;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:123456789@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9151 INVITE
Content-Type: application/sdp
Content-Length: 2170
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
v=0
o=- 6839724421067748000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 6jwyj9b0DqRU5BWU4PsHwJjvV6QuRZ8MSwiL
m=audio 19973 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 yy.yy.yy.yy
a=rtcp:19973 IN IP4 yy.yy.yy.yy
a=candidate:683538085 1 udp 2122260223 192.168.137.56 50293 typ host generation 0
a=candidate:683538085 2 udp 2122260223 192.168.137.56 50293 typ host generation 0
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 50294 typ host generation 0
a=candidate:2999745851 2 udp 2122194687 192.168.56.1 50294 typ host generation 0
a=candidate:1715341909 1 tcp 1518280447 192.168.137.56 0 typ host generation 0
a=candidate:1715341909 2 tcp 1518280447 192.168.137.56 0 typ host generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:3828402486 1 udp 1686052607 yy.yy.yy.yy 19973 typ srflx raddr 192.168.137.56 rport 50293 generation 0
a=candidate:3828402486 2 udp 1686052607 yy.yy.yy.yy 19973 typ srflx raddr 192.168.137.56 rport 50293 generation 0
a=ice-ufrag:6WbWLPzneHjg/KQO
a=ice-pwd:iEIFPDAzYsZnxuKLaIMPPdGn
a=ice-options:google-ice
a=fingerprint:sha-256 EA:4D:9C:6E:9C:81:EF:E9:5A:DA:94:06:16:00:80:FD:CA:B1:DE:74:6D:59:D9:C2:A7:ED:25:B9:C5:C5:4F:1B
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:BZYkTtcdk+iM/T8/yrGeSVqLHJHaiHTy1StbdQBX
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VK6+thw1XNAHFbR6hI4qVzraEVuOcsdUuMeOftqu
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:3408717055 cname:D/i0NbOJ6mF8ajqm
a=ssrc:3408717055 msid:6jwyj9b0DqRU5BWU4PsHwJjvV6QuRZ8MSwiL 5041dd72-b3a2-4593-bd6c-6bb157512186
a=ssrc:3408717055 mslabel:6jwyj9b0DqRU5BWU4PsHwJjvV6QuRZ8MSwiL
a=ssrc:3408717055 label:5041dd72-b3a2-4593-bd6c-6bb157512186
<------------->
--- (13 headers 45 lines) ---
Using INVITE request as basis request - 90e448c7-60d0-07eb-a5f7-3dd39149677e
Found peer '1060' for '1060' from yy.yy.yy.yy:20192
<--- Reliably Transmitting (no NAT) to yy.yy.yy.yy:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK5WkV84CH7th8WrFiKs4dersvITs2PES4;rport;received=yy.yy.yy.yy
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:123456789@xx.xx.xx.xx>;tag=as31598004
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9151 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="doubango.org", nonce="77f6bd4c"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '90e448c7-60d0-07eb-a5f7-3dd39149677e' in 32000 ms (Method: INVITE)
<--- SIP read from WS:yy.yy.yy.yy:20192 --->
ACK sip:123456789@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK5WkV84CH7th8WrFiKs4dersvITs2PES4;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:123456789@xx.xx.xx.xx>;tag=as31598004
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9151 ACK
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from WS:yy.yy.yy.yy:20192 --->
INVITE sip:123456789@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB1Dx3HFLCk6TemCrIpBGI8mnBnGNWtD4;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:123456789@xx.xx.xx.xx>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9152 INVITE
Content-Type: application/sdp
Content-Length: 2170
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="doubango.org",nonce="77f6bd4c",uri="sip:123456789@xx.xx.xx.xx",response="23345f77ea81641f029a04b8cb973d6b",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
v=0
o=- 6839724421067748000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 6jwyj9b0DqRU5BWU4PsHwJjvV6QuRZ8MSwiL
m=audio 19973 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 yy.yy.yy.yy
a=rtcp:19973 IN IP4 yy.yy.yy.yy
a=candidate:683538085 1 udp 2122260223 192.168.137.56 50293 typ host generation 0
a=candidate:683538085 2 udp 2122260223 192.168.137.56 50293 typ host generation 0
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 50294 typ host generation 0
a=candidate:2999745851 2 udp 2122194687 192.168.56.1 50294 typ host generation 0
a=candidate:1715341909 1 tcp 1518280447 192.168.137.56 0 typ host generation 0
a=candidate:1715341909 2 tcp 1518280447 192.168.137.56 0 typ host generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host generation 0
a=candidate:3828402486 1 udp 1686052607 yy.yy.yy.yy 19973 typ srflx raddr 192.168.137.56 rport 50293 generation 0
a=candidate:3828402486 2 udp 1686052607 yy.yy.yy.yy 19973 typ srflx raddr 192.168.137.56 rport 50293 generation 0
a=ice-ufrag:6WbWLPzneHjg/KQO
a=ice-pwd:iEIFPDAzYsZnxuKLaIMPPdGn
a=ice-options:google-ice
a=fingerprint:sha-256 EA:4D:9C:6E:9C:81:EF:E9:5A:DA:94:06:16:00:80:FD:CA:B1:DE:74:6D:59:D9:C2:A7:ED:25:B9:C5:C5:4F:1B
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:BZYkTtcdk+iM/T8/yrGeSVqLHJHaiHTy1StbdQBX
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VK6+thw1XNAHFbR6hI4qVzraEVuOcsdUuMeOftqu
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:3408717055 cname:D/i0NbOJ6mF8ajqm
a=ssrc:3408717055 msid:6jwyj9b0DqRU5BWU4PsHwJjvV6QuRZ8MSwiL 5041dd72-b3a2-4593-bd6c-6bb157512186
a=ssrc:3408717055 mslabel:6jwyj9b0DqRU5BWU4PsHwJjvV6QuRZ8MSwiL
a=ssrc:3408717055 label:5041dd72-b3a2-4593-bd6c-6bb157512186
<------------->
--- (14 headers 45 lines) ---
Using INVITE request as basis request - 90e448c7-60d0-07eb-a5f7-3dd39149677e
Found peer '1060' for '1060' from yy.yy.yy.yy
== 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
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port yy.yy.yy.yy:19973
Looking for 123456789in default (domain xx.xx.xx.xx)
list_route: hop: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
<--- Transmitting (no NAT) to yy.yy.yy.yy --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB1Dx3HFLCk6TemCrIpBGI8mnBnGNWtD4;rport;received=37.48.39.42
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:123456789@xx.xx.xx.xx>
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9152 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:123456789@10.168.1.101:5060;transport=WS>
Content-Length: 0
<------------>
-- Executing [123456789@default:1] MixMonitor("SIP/1060-00000000", "SIP/13_123456789.wav,b") in new stack
-- Executing [123456789@default:2] Dial("SIP/1060-00000000", "SIP/123465789@10.168.1.3,,r") in new stack
== Begin MixMonitor Recording SIP/1060-00000000
== Using SIP RTP CoS mark 5
Audio is at 64964
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.168.1.3:5060:
INVITE sip:123456789@10.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK51c35148
Max-Forwards: 70
From: "New User" <sip:1060@10.168.1.101>;tag=as2c1b44e6
To: <sip:123456789@10.168.1.3>
Contact: <sip:1060@10.168.1.101:5060>
Call-ID: 069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Wed, 21 May 2014 13:54:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 1190801425 1190801425 IN IP4 10.168.1.101
s=Asterisk PBX 11.4.0
c=IN IP4 10.168.1.101
t=0 0
m=audio 64964 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/123456789@10.168.1.3
<--- Transmitting (no NAT) to yy.yy.yy.yy:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB1Dx3HFLCk6TemCrIpBGI8mnBnGNWtD4;rport;received=37.48.39.42
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:732604400@xx.xx.xx.xx>;tag=as2784047b
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9152 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:123456789@10.168.1.101:5060;transport=WS>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.168.1.3:5060 --->
SIP/2.0 100 Trying
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK51c35148
From: "New User" <sip:1060@10.168.1.101>;tag=as2c1b44e6
To: <sip:123456789@10.168.1.3>;tag=7c1eb3fe34bb-4864-150f3a2e
Call-ID: 069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060
CSeq: 102 INVITE
Contact: <sip:123456789@10.168.1.3:5060>
User-Agent: 2N VBN 1.18.1.21.6
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:10.168.1.3:5060 --->
SIP/2.0 180 Ringing
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK51c35148
From: "New User" <sip:1060@10.168.1.101>;tag=as2c1b44e6
To: <sip:123456789@10.168.1.3>;tag=7c1eb3fe34bb-4864-150f3a2e
Call-ID: 069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060
CSeq: 102 INVITE
Contact: <sip:123456789@10.168.1.3:5060>
User-Agent: 2N VBN 1.18.1.21.6
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:123456789@10.168.1.3:5060>
-- SIP/10.168.1.3-00000001 is ringing
<--- Transmitting (no NAT) to yy.yy.yy.yy --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB1Dx3HFLCk6TemCrIpBGI8mnBnGNWtD4;rport;received=yy.yy.yy.yy
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:123456789@xx.xx.xx.xx>;tag=as2784047b
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9152 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:123456789@10.168.1.101:5060;transport=WS>
Content-Length: 0
<------------>
Really destroying SIP dialog '7c1eb3fe34bb-4839-150f3a12@10.168.1.101' Method: REGISTER
<--- SIP read from UDP:10.168.1.3:5060 --->
SIP/2.0 200 OK
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK51c35148
From: "New User" <sip:1060@10.168.1.101>;tag=as2c1b44e6
To: <sip:123456789@10.168.1.3>;tag=7c1eb3fe34bb-4864-150f3a2e
Call-ID: 069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060
CSeq: 102 INVITE
Contact: <sip:123456789@10.168.1.3:5060>
User-Agent: 2N VBN 1.18.1.21.6
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Type: application/sdp
Content-Length: 209
v=0
o=VBN 19846 28707 IN IP4 10.168.1.3
s=GSM Call
c=IN IP4 10.168.1.3
t=0 0
a=sendrecv
m=audio 10048 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.168.1.3:10048
list_route: hop: <sip:123456789@10.168.1.3:5060>
set_destination: Parsing <sip:123456789@10.168.1.3:5060> for address/port to send to
set_destination: set destination to 10.168.1.3:5060
Transmitting (no NAT) to 10.168.1.3:5060:
ACK sip:123456789@10.168.1.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK6c9b0d20
Max-Forwards: 70
From: "New User" <sip:1060@10.168.1.101>;tag=as2c1b44e6
To: <sip:123456789@10.168.1.3>;tag=7c1eb3fe34bb-4864-150f3a2e
Contact: <sip:1060@10.168.1.101:5060>
Call-ID: 069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
---
-- SIP/10.168.1.3-00000001 answered SIP/1060-00000000
Audio is at 13764
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to yy.yy.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB1Dx3HFLCk6TemCrIpBGI8mnBnGNWtD4;rport;received=37.48.39.42
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:732604400@xx.xx.xx.xx>;tag=as2784047b
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9152 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:123456789@10.168.1.101:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 687
v=0
o=root 1326460708 1326460708 IN IP4 10.168.1.101
s=Asterisk PBX 11.4.0
c=IN IP4 10.168.1.101
t=0 0
m=audio 13764 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:003936584b282e2c0cd845b170f27e6e
a=ice-pwd:7a696ade76aea0a11157f1e8103e8bf9
a=candidate:Haa80165 1 UDP 2130706431 10.168.1.101 13764 typ host
a=candidate:Sb248c0c3 1 UDP 1694498815 xx.xx.xx.xx 13764 typ srflx
a=candidate:Haa80165 2 UDP 2130706430 10.168.1.101 13765 typ host
a=candidate:Sb248c0c3 2 UDP 1694498814 xx.xx.xx.xx 13764 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:z9ugxF2dnvs+rkwxUWDZLN1H+WarudAvBYawN1bL
<------------>
<--- SIP read from WS:yy.yy.yy.yy --->
ACK sip:732604400@10.168.1.101:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOl4oVbiEz1mRLfCMrcaU;rport
From: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
To: <sip:123456789@xx.xx.xx.xx>;tag=as2784047b
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 9152 ACK
Content-Length: 0
Route: <sip:xx.xx.xx.xx:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="doubango.org",nonce="77f6bd4c",uri="sip:123456789@10.168.1.101:5060;transport=WS",response="4b0d9c82397dea3ba0eba06c04cc77f9",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
<------------->
--- (13 headers 0 lines) ---
<--- SIP read from UDP:10.168.1.3:5060 --->
REGISTER sip:10.168.1.101:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6827353319482
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4865-150f3a3a
To: <sip:100@10.168.1.101:5060>
Call-ID: 7c1eb3fe34bb-4840-150f3a3a@10.168.1.101
CSeq: 6809 REGISTER
Contact: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Expires: 60
User-Agent: 2N VBN 1.18.1.21.6
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 10.168.1.3:5060 (no NAT)
Sending to 10.168.1.3:5060 (no NAT)
<--- Transmitting (NAT) to 10.168.1.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6827353319482;received=10.168.1.3;rport=5060
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4865-150f3a3a
To: <sip:100@10.168.1.101:5060>;tag=as29899ef8
Call-ID: 7c1eb3fe34bb-4840-150f3a3a@10.168.1.101
CSeq: 6809 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="doubango.org", nonce="09104aeb"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7c1eb3fe34bb-4840-150f3a3a@10.168.1.101' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.168.1.3:5060 --->
REGISTER sip:10.168.1.101:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6828353319482
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4865-150f3a3a
To: <sip:100@10.168.1.101:5060>
Call-ID: 7c1eb3fe34bb-4840-150f3a3a@10.168.1.101
CSeq: 6810 REGISTER
Contact: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Authorization: Digest username="100",realm="doubango.org",nonce="09104aeb",uri="sip:10.168.1.101",response="4614264296a784cd56097394633dc00b",algorithm=MD5
Expires: 60
User-Agent: 2N VBN 1.18.1.21.6
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 10.168.1.3:5060 (NAT)
Reliably Transmitting (NAT) to 10.168.1.3:5060:
OPTIONS sip:100@10.168.1.3:5060;user=phone;phone-context=100 SIP/2.0
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK2ece430d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.168.1.101>;tag=as1509567d
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Contact: <sip:asterisk@10.168.1.101:5060>
Call-ID: 63c2a448686d1ef5613f85d74fe59598@10.168.1.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.4.0
Date: Wed, 21 May 2014 13:54:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (NAT) to 10.168.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6828353319482;received=10.168.1.3;rport=5060
From: <sip:100@10.168.1.101:5060>;tag=7c1eb3fe34bb-4865-150f3a3a
To: <sip:100@10.168.1.101:5060>;tag=as29899ef8
Call-ID: 7c1eb3fe34bb-4840-150f3a3a@10.168.1.101
CSeq: 6810 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>;expires=60
Date: Wed, 21 May 2014 13:54:22 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3e679e21386e97373c13860f1a040fa3@10.168.1.3' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 10.168.1.3:5060:
NOTIFY sip:100@10.168.1.3:5060;user=phone;phone-context=100 SIP/2.0
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK4d6f33af;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.168.1.3>;tag=as2c833a21
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Contact: <sip:asterisk@10.168.1.101:5060>
Call-ID: 3e679e21386e97373c13860f1a040fa3@10.168.1.3
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.4.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90
Messages-Waiting: no
Message-Account: sip:asterisk@10.168.1.3
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '7c1eb3fe34bb-4840-150f3a3a@10.168.1.101' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.168.1.3:5060 --->
SIP/2.0 200 OK
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK2ece430d;rport
From: "asterisk" <sip:asterisk@10.168.1.101>;tag=as1509567d
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Call-ID: 63c2a448686d1ef5613f85d74fe59598@10.168.1.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.4.0
Date: Wed, 21 May 2014 13:54:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '63c2a448686d1ef5613f85d74fe59598@10.168.1.101:5060' Method: OPTIONS
<--- SIP read from UDP:10.168.1.3:5060 --->
SIP/2.0 481 Call Does Not Exist
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.101:5060;branch=z9hG4bK4d6f33af;rport
From: "asterisk" <sip:asterisk@10.168.1.3>;tag=as2c833a21
To: <sip:100@10.168.1.3:5060;user=phone;phone-context=100>
Call-ID: 3e679e21386e97373c13860f1a040fa3@10.168.1.3
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.4.0
Event: message-summary
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3e679e21386e97373c13860f1a040fa3@10.168.1.3' Method: NOTIFY
Really destroying SIP dialog 'f8184a08-07cf-558b-9d46-fba19e8e9aef' Method: REGISTER
<--- SIP read from UDP:10.168.1.3:5060 --->
BYE sip:1060@10.168.1.101:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6829353319697
From: <sip:123456789@10.168.1.3>;tag=7c1eb3fe34bb-4864-150f3a2e
To: "New User" <sip:1060@10.168.1.101>;tag=as2c1b44e6
Call-ID: 069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060
CSeq: 19 BYE
Contact: <sip:123456789@10.168.1.3:5060>
User-Agent: 2N VBN 1.18.1.21.6
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 10.168.1.3:5060 (no NAT)
Scheduling destruction of SIP dialog '069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060' in 32000 ms (Method: BYE)
<--- Transmitting (no NAT) to 10.168.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.1.3:5060;branch=z9hG4bK-6829353319697;received=10.168.1.3
From: <sip:123456789@10.168.1.3>;tag=7c1eb3fe34bb-4864-150f3a2e
To: "New User" <sip:1060@10.168.1.101>;tag=as2c1b44e6
Call-ID: 069d221f3490449e6cf7458b693dbd1f@10.168.1.101:5060
CSeq: 19 BYE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (default, 123456789, 2) exited non-zero on 'SIP/1060-00000000'
Scheduling destruction of SIP dialog '90e448c7-60d0-07eb-a5f7-3dd39149677e' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (no NAT) to yy.yy.yy.yy:
BYE sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.101:5060;branch=z9hG4bK3c6dbbbc
Max-Forwards: 70
From: <sip:123456789@xx.xx.xx.xx>;tag=as2784047b
To: "1060"<sip:1060@xx.xx.xx.xx>;tag=VHpt6Bm8IG96oh6TkCOv
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.4.0
Proxy-Authorization: Digest username="1060", realm="doubango.org", algorithm=MD5, uri="sip:xx.xx.xx.xx", nonce="77f6bd4c", response="152c3f531a5c825a6a38664be8a4eb3a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/1060-00000000
<--- SIP read from WS:YY.YY.YY.YY --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 10.168.1.101:5060;branch=z9hG4bK3c6dbbbc
From: <sip:123456789@XX.XX.XX.XX>;tag=as2784047b
To: "1060"<sip:1060@XX.XX.XX.XX>;tag=VHpt6Bm8IG96oh6TkCOv
Contact: <sip:1060@df7jal23ls0d.invalid;transport=ws>
Call-ID: 90e448c7-60d0-07eb-a5f7-3dd39149677e
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '90e448c7-60d0-07eb-a5f7-3dd39149677e' Method: INVITE
Dataobjects*CLI> exit
Asterisk cleanly ending (0).xx.xx.xx.xx - our public IP
yy.yy.yy.yy - T-Mobile public IP
123456789 - my phone number
We have tried multiple versions of asterisk, the only version that works at least within the private network is 11.4, if you would need any other resource to help you solve this problem don´t hesitate to ask