Softphone, webphone and VoIP server Forum

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3/23/2018 10:13 AM
 
Hello, our user could not make calls yesterday from 12:40 until 14:00 and today from 13:30 until 13:45. This is killing for out business. I tried to call to 0040745 956613 and 00 40 720 435 054 but know answer. Can you let me know what is going on? Do you monitor? When will it be solved? Again.. it is killing our business. Please fix it once and for all. 

EVENT, webphone_api.setparameter: loglevel: 5
lib_softphone.js:35 webphone: [13:34:26.063] EVENT, SetParameter user, loglevel: 5
lib_softphone.js:35 webphone: [13:34:26.064] EVENT, SetParameter loglevel: *****
undefined
lib_softphone.js:35 webphone: [13:34:26.124] EVENT, SaveSettingsFile from timer Successfull
lib_softphone.js:35 webphone: [13:34:31.737] EVENT, plhandler initiate Call with engine: webrtc, to number: 0648390xxx
lib_softphone.js:35 webphone: [13:34:31.739] EVENT, USER, API_Call (webrtc), to: 0648390xxx, on line: 1, type: audio
lib_softphone.js:35 webphone: [13:34:31.848] About:
CallMonkey 2.4.1207 (16); Build date: ; OS: Windows 10 64-bit; voipen: webrtc; date: 2018/3/23
Browser: Chrome 65.0.3325.181; Bengine: Blink; GetBrowser: Chrome; GetOs: Windows; Used as SDK: true
lib_softphone.js:35 webphone: [13:34:31.855] Location: https://xx.xx.com/Agent/#AgentAddress
 
lib_softphone.js:35 webphone: [13:34:31.857] EVENT, webrtcapi: audio bandwidth: 0
lib_softphone.js:35 webphone: [13:34:31.857] wpapi_param: loglevel=5
lib_softphone.js:35 webphone: [13:34:31.859] wpapi_param: issdk=true
lib_softphone.js:35 webphone: [13:34:31.860] STATUS,-1,Starting call to 0648390xxx
lib_softphone.js:35 webphone: [13:34:31.863] EVENT, uri = https://usrtcx.webvoipphone.com/mvapireq/?apientry=wsuser&authkey=15523ccdf0117&authid=test&authpwd=*****&brandid=-1&brand
lib_softphone.js:35     webphone: [13:34:31.863] name=CallMonkey&companyname=Blue&upperserver=mv.fieb.nl&needregister=true&callerid=test&displayname=&voicerecupload=&engine=webrt
lib_softphone.js:35         webphone: [13:34:31.863] c&dtmfmode=2
lib_softphone.js:35 webphone: [13:34:31.864] EVENT,httpRequestUri = https://usrtcx.webvoipphone.com/mvapireq/?apientry=wsuser&authkey=15523ccdf0117&authid=test&authpwd=*****&brandid=-
lib_softphone.js:35     webphone: [13:34:31.864] 1&brandname=CallMonkey&companyname=Blue&upperserver=mv.fieb.nl&needregister=true&callerid=test&displayname=&voicerecupload=&engin
lib_softphone.js:35         webphone: [13:34:31.864] e=webrtc&dtmfmode=2;   action = webrtccreateuser
lib_softphone.js:35 webphone: [13:34:31.873] EVENT, HttpRequest method: ws
lib_softphone.js:35 webphone: [13:34:31.881] EVENT, _common: WSRequest url: https://usrtcx.webvoipphone.com/mvstwebsock/mvapireq/?apientry=wsuser&authkey=15523ccdf0117&authid=test&aut
lib_softphone.js:35     webphone: [13:34:31.881] hpwd=*****&brandid=-1&brandname=CallMonkey&companyname=Blue&upperserver=mv.fieb.nl&needregister=true&callerid=test&displayname
lib_softphone.js:35         webphone: [13:34:31.881] =&voicerecupload=&engine=webrtc&dtmfmode=2
lib_softphone.js:35 webphone: [13:34:31.892] EVENT, wsuser already succeeded
lib_softphone.js:35 webphone: [13:34:31.896] EVENT, IsInCall: false
lib_softphone.js:35 webphone: [13:34:31.902] WRTC, EVENT, State machine: c0000_Started_2_Outgoing_X_oINVITE
lib_softphone.js:35 webphone: [13:34:31.919] WRTC, EVENT, __get_lo GetLocalOffer exec
lib_softphone.js:35 webphone: [13:34:31.922] WRTC, EVENT, ICE servers:[{"url":"stun:usrtcx.webvoipphone.com:8090"},{"url":"turn:usrtcx.webvoipphone.com:80?transport=tcp","username":"mzturnusr","credential":"rnd87719318572"}]
VM913:334 [Deprecation] The rtcpMuxPolicy option is being considered for removal and may be removed no earlier than M62, around October 2017. If you depend on it, please see https://www.chromestatus.com/features/5654810086866944 for more details.
WrappedRTCPeerConnection @ VM913:334
RTCPeerConnection @ lib_softphone.js:6
tmedia_session_jsep01.__get_lo @ lib_softphone.js:7
tmedia_session.get_lo @ lib_softphone.js:6
tmedia_session_mgr.get_lo @ lib_softphone.js:6
tsip_dialog_invite.send_offer @ lib_softphone.js:11
tsip_dialog_invite.send_invite @ lib_softphone.js:11
c0000_Started_2_Outgoing_X_oINVITE @ lib_softphone.js:3
tsk_fsm.act @ lib_softphone.js:6
tsip_dialog.fsm_act @ lib_softphone.js:11
tsip_session_invite.call @ lib_softphone.js:9
SIPml.Session.Call.call @ lib_softphone.js:30
SipCall @ lib_softphone.js:56
(anonymous) @ lib_softphone.js:59
setTimeout (async)
P @ lib_softphone.js:59
k @ lib_softphone.js:59
R @ lib_softphone.js:59
call @ webphone_api.js?26.07.2017:183
call @ app.js?v=20180217064218:1
success @ app.js?v=20180217064218:246
callback @ app.js?v=20180217064218:1
onComplete @ app.js?v=20180217064218:1
onStateChange @ app.js?v=20180217064218:1
(anonymous) @ app.js?v=20180217064218:1
XMLHttpRequest.send (async)
request @ app.js?v=20180217064218:1
createCallAttemptUuid @ app.js?v=20180217064218:246
webphoneCallEvent @ app.js?v=20180217064218:246
fire @ app.js?v=20180217064218:1
doFireEvent @ app.js?v=20180217064218:1
g.doFireEvent @ app.js?v=20180217064218:1
fireEventArgs @ app.js?v=20180217064218:1
fireEvent @ app.js?v=20180217064218:1
onWebphoneCallBtnClick @ app.js?v=20180217064218:246
callback @ app.js?v=20180217064218:1
fireHandler @ app.js?v=20180217064218:1
onClick @ app.js?v=20180217064218:1
fire @ app.js?v=20180217064218:1
fire @ app.js?v=20180217064218:1
publish @ app.js?v=20180217064218:1
onRecognized @ app.js?v=20180217064218:1
fire @ app.js?v=20180217064218:1
onTouchEnd @ app.js?v=20180217064218:1
invokeRecognizers @ app.js?v=20180217064218:1
onTouchEnd @ app.js?v=20180217064218:1
(anonymous) @ app.js?v=20180217064218:1
w @ app.js?v=20180217064218:1
requestAnimationFrame (async)
requestAnimationFrame @ app.js?v=20180217064218:1
(anonymous) @ app.js?v=20180217064218:1
doDelegatedEvent @ app.js?v=20180217064218:1
onDelegatedEvent @ app.js?v=20180217064218:1
(anonymous) @ app.js?v=20180217064218:1
lib_softphone.js:35 webphone: [13:34:31.929] WRTC, EVENT, constraintsL: {"audio":{"echoCancellation":true},"video":false,"data":false}
lib_softphone.js:35 webphone: [13:34:31.930] STATUS,1,Call
lib_softphone.js:35 webphone: [13:34:31.938] EVENT, common: SaveCallSession new endpoint added, line: 1; destnr: 0648390xxx; isincoming: 1; callid: 3; setuptime: 1521808471932; connecttime: -1; disctime: -1; setupsent: ; finishedsent: ; mutestate: false; holdstate: false; sipinvite:  
lib_softphone.js:35 webphone: [13:34:31.947] WRTC, EVENT, ==stack event = m_permission_requested
lib_softphone.js:35 webphone: [13:34:31.950] EVENT, webrtcapi: permission requested
lib_softphone.js:35 webphone: [13:34:31.953] STATUS, Waiting for permission
lib_softphone.js:35 webphone: [13:34:31.955] WRTC, EVENT, ==session event = connecting
lib_softphone.js:35 webphone: [13:34:31.956] STATUS,1,Call in progress...
lib_softphone.js:35 webphone: [13:34:31.956] EVENT, common: SaveCallSession endpoint updated, line: 1; destnr: 0648390xxx; isincoming: 1; callid: 3; setuptime: 1521808471956; connecttime: -1; disctime: -1; setupsent: ; finishedsent: ; mutestate: ; holdstate: ; sipinvite:  
lib_softphone.js:35 webphone: [13:34:31.997] WRTC, EVENT, onGetUserMediaSuccess
lib_softphone.js:35 webphone: [13:34:31.998] WRTC, EVENT, createOffer
lib_softphone.js:35 webphone: [13:34:32.002] WRTC, EVENT, onCreateSdpSuccess
lib_softphone.js:35 webphone: [13:34:32.003] WRTC, EVENT, ==stack event = m_permission_accepted
lib_softphone.js:35 webphone: [13:34:32.004] EVENT, webrtcapi: MediaStream permission granted
lib_softphone.js:35 webphone: [13:34:32.007] EVENT,1,Call
lib_softphone.js:35 webphone: [13:34:32.012] WRTC, EVENT, onNegotiationNeeded
lib_softphone.js:35 webphone: [13:34:32.012] WRTC, EVENT, onSetLocalDescriptionSuccess
lib_softphone.js:35 webphone: [13:34:32.013] WRTC, EVENT, onSignalingstateChange:have-local-offer
lib_softphone.js:35 webphone: [13:34:32.015] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.016] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.017] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.019] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.019] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.020] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.020] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.020] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.114] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.123] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.128] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.128] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.129] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.129] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.129] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.129] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.142] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.143] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.146] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.146] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.326] EVENT, common wsuser response: OK: wsuser succeed test
lib_softphone.js:35 webphone: [13:34:32.513] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.514] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.712] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.713] WRTC, EVENT, onIceCandidate = gathering
lib_softphone.js:35 webphone: [13:34:32.713] WRTC, EVENT, ICE gathering manual timeout set to: 3000
lib_softphone.js:35 webphone: [13:34:32.715] WRTC, EVENT, onIceCandidate = complete
lib_softphone.js:35 webphone: [13:34:32.715] WRTC, EVENT, ICE GATHERING COMPLETED!
lib_softphone.js:35 webphone: [13:34:32.716] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:32.756] WRTC, EVENT, __get_lo GetLocalOffer exec
lib_softphone.js:35 webphone: [13:34:32.793] WRTC, EVENT, SEND:
 INVITE sip:0648390xxx@mv.fieb.nl SIP/2.0
    Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKtc0bNsvpATOusyeOqWSvhhqZ3hiZebLV;rport
    From: <sip:test@mv.fieb.nl>;tag=cqsTD7MBRuOveQSiFOX1
    To: <sip:0648390xxx@mv.fieb.nl>
    Contact: <sips:test@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im
    Call-ID: 0a048d86-fb36-a76c-7219-a248bda3f23e
    CSeq: 9451 INVITE
    Content-Type: application/sdp
    Content-Length: 2871
    Max-Forwards: 70
    X-IMStatus: Online
    User-Agent: MWWRTC 1.0
    Organization: Blue
    X-Webdomain: https://backoffice.callmonkey.com
    X-Sy.Uppersrv: mv.fieb.nl
    X-Sy.Upperproxy: mv.fieb.nl
    X-IMStatus: Online
   
    v=0
    o=- 5207836387942722000 2 IN IP4 127.0.0.1
    s=Mizutech - chrome
    t=0 0
    a=group:BUNDLE audio
    a=msid-semantic: WMS 6vSDIMO57DoHCMtCOpS3pdXyHWp2NV93MTnn
    m=audio 80 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
    c=IN IP4 204.12.197.100
    a=rtcp:80 IN IP4 204.12.197.100
    a=candidate:35
lib_softphone.js:35     webphone: [13:34:32.793] 82884834 1 udp 2122255103 2001::9d38:90d7:18c0:3df2:6e7e:aa68 60866 typ host generation 0 network-id 1 network-cost 50
    a=candidate:90380972 1 udp 2122194687 192.168.2.1 60867 typ host generation 0 network-id 2 network-cost 10
    a=candidate:3582884834 2 udp 2122255102 2001::9d38:90d7:18c0:3df2:6e7e:aa68 60868 typ host generation 0 network-id 1 network-cost 50
    a=candidate:90380972 2 udp 2122194686 192.168.2.1 60869 typ host generation 0 network-id 2 network-cost 10
    a=candidate:2601355026 1 tcp 1518275327 2001::9d38:90d7:18c0:3df2:6e7e:aa68 9 typ host tcptype active generation 0 network-id 1 network-cost 50
    a=candidate:1273030236 1 tcp 1518214911 192.168.2.1 9 typ host tcptype active generation 0 network-id 2 network-cost 10
    a=candidate:2601355026 2 tcp 1518275326 2001::9d38:90d7:18c0:3df2:6e7e:aa68 9 typ host tcptype active generation 0 network-id 1 network-cost 50
    a=candidate:1273030236 2 tcp 1518214910 192.168.2.1 9 typ host tcptype active generation 0 network-id 2 network-cost 10
lib_softphone.js:35         webphone: [13:34:32.793]
    a=candidate:2889651577 1 udp 1685987071 145.129.85.151 60867 typ srflx raddr 192.168.2.1 rport 60867 generation 0 network-id 2 network-cost 10
    a=candidate:2889651577 2 udp 1685987070 145.129.85.151 60869 typ srflx raddr 192.168.2.1 rport 60869 generation 0 network-id 2 network-cost 10
    a=candidate:4078253188 1 udp 25042687 204.12.197.100 80 typ relay raddr 145.129.85.151 rport 60321 generation 0 network-id 2 network-cost 10
    a=candidate:4078253188 2 udp 25042686 204.12.197.100 80 typ relay raddr 145.129.85.151 rport 60322 generation 0 network-id 2 network-cost 10
    a=ice-ufrag:JCZ7
    a=ice-pwd:JoDQx5EM77DlvoGqpSKrafHb
    a=ice-options:trickle
    a=fingerprint:sha-256 E8:A3:28:CB:0A:C0:5D:B4:93:9F:5D:91:24:89:73:DF:ED:AC:F8:1D:06:0E:08:E1:DD:88:AA:3A:DA:48:19:58
    a=setup:actpass
    a=mid:audio
    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
    a=sendrecv
    a=rtcp-mux
    a=rtpmap:111 opus/48000/2
    a=rtcp-fb:111 transport-cc
    a=fmtp:111 minptime=10;useinbandfec=1
    a=rtpmap:103 ISAC/16000
    a
lib_softphone.js:35             webphone: [13:34:32.793] =rtpmap:104 ISAC/32000
    a=rtpmap:9 G722/8000
    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:110 telephone-event/48000
    a=rtpmap:112 telephone-event/32000
    a=rtpmap:113 telephone-event/16000
    a=rtpmap:126 telephone-event/8000
    a=ssrc:1226437623 cname:hNjhKJmKS0stpUOD
    a=ssrc:1226437623 msid:6vSDIMO57DoHCMtCOpS3pdXyHWp2NV93MTnn 30bc21b6-891d-497a-8b12-6ddf66f847f2
    a=ssrc:1226437623 mslabel:6vSDIMO57DoHCMtCOpS3pdXyHWp2NV93MTnn
    a=ssrc:1226437623 label:30bc21b6-891d-497a-8b12-6ddf66f847f2
     
lib_softphone.js:35 webphone: [13:34:33.161] WRTC, EVENT, __tsip_transport_ws_onmessage
lib_softphone.js:35 webphone: [13:34:33.209] WRTC, EVENT, recv=
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=60251;received=145.129.85.151;branch=z9hG4bKtc0bNsvpATOusyeOqWSvhhqZ3hiZebLV
From: <sip:test@mv.fieb.nl>;tag=cqsTD7MBRuOveQSiFOX1
To: <sip:0648390xxx@mv.fieb.nl>
Call-ID: 0a048d86-fb36-a76c-7219-a248bda3f23e
CSeq: 9451 INVITE
Content-Length: 0
User-Agent: mzfs
 
 
lib_softphone.js:35 webphone: [13:34:33.214] WRTC, EVENT, State machine: x0000_Any_2_Any_X_i1xx
lib_softphone.js:35 webphone: [13:34:33.217] WRTC, EVENT, __tsip_transport_ws_onmessage
lib_softphone.js:35 webphone: [13:34:33.223] WRTC, EVENT, recv=
SIP/2.0 404 Not Found
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=60251;received=145.129.85.151;branch=z9hG4bKtc0bNsvpATOusyeOqWSvhhqZ3hiZebLV
From: <sip:test@mv.fieb.nl>;tag=cqsTD7MBRuOveQSiFOX1
To: <sip:0648390xxx@mv.fieb.nl>;tag=H69vgZBSBmrSS
Call-ID: 0a048d86-fb36-a76c-7219-a248bda3f23e
CSeq: 9451 INVITE
Content-Length: 0
Max-Forwards: 69
User-Agent: mzfs
Accept: text/html
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,UPDATE,MESSAGE,INFO,OPTIONS,SUBSCRIBE,NOTIFY,REFER,PUBLISH
Supported: timer,path,replaces
Allow-Events: conference,hold,refer,message-summary,presence,presence.winfo,talk,dialog,call-info,sla,include-session-description,as-feature-event,line-seize
Reason: text="NO_ROUTE_DESTINATION";cause=3;text="NO_ROUTE_DESTINATION"
Remote-Party-ID: screen=no;party=calling;privacy=off;screen=no
 
 
lib_softphone.js:35 webphone: [13:34:33.225] WRTC, EVENT, SEND:
 ACK sip:0648390xxx@mv.fieb.nl SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKtc0bNsvpATOusyeOqWSvhhqZ3hiZebLV;rport
From: <sip:test@mv.fieb.nl>;tag=cqsTD7MBRuOveQSiFOX1
To: <sip:0648390xxx@mv.fieb.nl>;tag=H69vgZBSBmrSS
Call-ID: 0a048d86-fb36-a76c-7219-a248bda3f23e
CSeq: 9451 ACK
Content-Length: 0
Route: <sip:://:NaN;lr;MWWRTC;transport=udp>
Max-Forwards: 70
 
 
lib_softphone.js:35 webphone: [13:34:33.228] WRTC, EVENT, State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
lib_softphone.js:35 webphone: [13:34:33.230] WRTC, EVENT, === INVITE Dialog terminated ===
lib_softphone.js:35 webphone: [13:34:33.237] WRTC, EVENT, PeerConnection::stop()
lib_softphone.js:35 webphone: [13:34:33.245] WRTC, EVENT, ==session event = i_ao_request
lib_softphone.js:35 webphone: [13:34:33.247] EVENT, common: SaveCallSession endpoint updated, line: 1; destnr: 0648390xxx; isincoming: 1; callid: ; setuptime: 1521808473247; connecttime: -1; disctime: -1; setupsent: ; finishedsent: ; mutestate: ; holdstate: ; sipinvite:  
api_helper.js:1 webphone: onCallStateChange: callSetup,1,0648390xxx,0648390xxx,1
lib_softphone.js:35 webphone: [13:34:33.257] STATUS,1,CallSetup,0648390xxx,test,1,0648390xxx
lib_softphone.js:35 webphone: [13:34:33.258] EVENT, common: SaveCallSession endpoint updated, line: 1; destnr: 0648390xxx; isincoming: 1; callid: 3; setuptime: 1521808473258; connecttime: -1; disctime: -1; setupsent: true; finishedsent: ; mutestate: ; holdstate: ; sipinvite:  
lib_softphone.js:35 webphone: [13:34:33.261] WRTC, EVENT, ==session event = i_ao_request
lib_softphone.js:35 webphone: [13:34:33.262] WRTC, EVENT, ==session event = terminated
lib_softphone.js:35 webphone: [13:34:33.262] EVENT,EventSession call: terminated
lib_softphone.js:35 webphone: [13:34:33.263] EVENT, common: SaveCallSession endpoint updated, line: 1; destnr: 0648390xxx; isincoming: 1; callid: 3; setuptime: -1; connecttime: -1; disctime: 1521808473263; setupsent: ; finishedsent: true; mutestate: ; holdstate: ; sipinvite:  
api_helper.js:1 webphone: onCallStateChange: callDisconnected,1,0648390xxx,0648390xxx,1
lib_softphone.js:35 webphone: [13:34:33.265] EVENT, common: SaveCallSession endpoint updated, line: 1; destnr: 0648390xxx; isincoming: 1; callid: ; setuptime: -1; connecttime: -1; disctime: 1521808473265; setupsent: ; finishedsent: ; mutestate: ; holdstate: ; sipinvite:  
lib_softphone.js:35 webphone: [13:34:33.266] STATUS,1,CallDisconnect,0648390xxx,test,1,0648390xxx
lib_softphone.js:35 webphone: [13:34:33.267] EVENT, webrtcapi uiVideoDisplayEvent videoRemote is NULL
lib_softphone.js:35 webphone: [13:34:33.272] STATUS,1,Call Finished
lib_softphone.js:35 webphone: [13:34:33.276] EVENT, CanIUseService_1 true
lib_softphone.js:35 webphone: [13:34:33.279] STATUS,1,Finished
lib_softphone.js:35 webphone: [13:34:33.281] EVENT, common: GetEndpoint remove OLD endpoint_1: 1,[object Object],0648390xxx,1,1521808473258,-1,1521808473265,,3,false,true,true,[object Object],false,false,
lib_softphone.js:35 webphone: [13:34:33.285] EVENT, plhandler: GetSipHeader: X-reason; (webrtc)
lib_softphone.js:35 webphone: [13:34:33.286] EVENT, plhandler: GetSipHeader: Reason; (webrtc)
lib_softphone.js:35 webphone: [13:34:33.287] WRTC, WARNING, The FSM is in the final state
lib_softphone.js:35 webphone: [13:34:34.127] EVENT, notifications delete/remove endpoint on call disconnected for number: CallDisconnect; line: 1
lib_softphone.js:35 webphone: [13:34:34.132] EVENT, notifications: ProcessNotifications CDR received: CDR,1,0648390xxx,test,0648390xxx,NA,-1521808473259,0,NA,NO_ROUTE_DESTINATION,1
lib_softphone.js:35 webphone: [13:34:34.133] EVENT, plhandler: GetSipHeader: X-reason; (webrtc)
lib_softphone.js:35 webphone: [13:34:34.135] EVENT, plhandler: GetSipHeader: Reason; (webrtc)
lib_softphone.js:35 webphone: [13:34:34.136] EVENT, Disconnect reason: NO_ROUTE_DESTINATION
lib_softphone.js:35 webphone: [13:34:34.141] EVENT,notifications CDR 0 duration get presence status for: 0648390xxx
lib_softphone.js:35 webphone: [13:34:34.144] EVENT, NO_ROUTE_DESTINATION
lib_softphone.js:35 webphone: [13:34:34.145] EVENT, notifications CDR added; AddToCallLog(); number: 0648390xxx; type: 0; recorded: recfile; duration: 0; callDateTime = 0; line: 1
lib_softphone.js:35 webphone: [13:34:34.147] CDR,1,0648390xxx,test,0648390xxx,NA,-1521808473259,0,NA,NO_ROUTE_DESTINATION,1
lib_softphone.js:35 webphone: [13:34:34.788] STATUS,-1,Registered.
lib_softphone.js:35 webphone: [13:34:35.354] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.354] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.355] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.355] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.355] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.356] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.357] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.357] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.358] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.360] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.360] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.361] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.361] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.362] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.371] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.372] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.375] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.375] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.375] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.377] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.401] EVENT, SaveSettingsFile from timer Successfull
lib_softphone.js:35 webphone: [13:34:35.516] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.517] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:35.714] WRTC, EVENT, ICE GATHERING COMPLETED, mtimeout
lib_softphone.js:35 webphone: [13:34:35.716] WRTC, EVENT, onIceGatheringCompleted
lib_softphone.js:35 webphone: [13:34:36.008] WARNING, common GetActiveAccount no available accounts
lib_softphone.js:35 webphone: [13:34:36.015] EVENT, common GetActiveAccSettingsFilename cant get filename, probably doesn't exist
lib_softphone.js:35 webphone: [13:34:36.028] EVENT, SaveCallhistoryFile from timer Successfull
lib_softphone.js:35 webphone: [13:34:53.183] WRTC, EVENT, webrtc socket ping sent
lib_softphone.js:35 webphone: [13:34:53.302] WRTC, EVENT, __tsip_transport_ws_onmessage
lib_softphone.js:35 webphone: [13:34:53.305] WRTC, EVENT, __tsip_transport_ws_onmessage: mrtcpong
 
New Post
3/23/2018 10:55 AM
 
We don't provide support via forum or phone calls for the webphone (except for customers with dedicated support).
For the webphone we can provide support by email if you haven't reached your support subscription limits yet.
Please contact my colleagues at webphone@mizu-voip.com and I am sure that they will answer soon.



 
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