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View Code? Open in Web Editor NEWAutomatically exported from code.google.com/p/click-2-dial
License: GNU General Public License v2.0
Automatically exported from code.google.com/p/click-2-dial
License: GNU General Public License v2.0
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 15 Mar 2013 at 8:28
How can I use Clicl-2-Dial?
I gathered webrtc2sip, tested.
Just tested sipml5.
But there is no documentation on Clicl-2-Dial http://click2dial.org/doc.htm ! I
can not understand how I create settings in the database webrtc2sip that change
in the code c2c-api.js
Need step by step documentation launch.
Original issue reported on code.google.com by [email protected]
on 6 Feb 2014 at 11:04
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
What version of the product are you using? On what operating system?
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 22 Jan 2015 at 1:06
Hi everybody,
I'm trying register my extension from Chrome to FreeSwitch on local IP:
192.168.1.199, I don't see any logs about try to extension register, I got only
logs from chrome console:
SIPML5 API version = 1.2.185 SIPml-api.js:1
User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like
Gecko) Chrome/29.0.1510.0 Safari/537.36 SIPml-api.js:1
WebSocket supported = yes SIPml-api.js:1
Navigator friendly name = chrome SIPml-api.js:1
OS friendly name = windows SIPml-api.js:1
Have WebRTC = yes SIPml-api.js:1
Have GUM = yes SIPml-api.js:1
Engine initialized SIPml-api.js:1
s_websocket_server_url=(null) SIPml-api.js:1
s_sip_outboundproxy_url=(null) SIPml-api.js:1
b_rtcweb_breaker_enabled=yes SIPml-api.js:1
b_click2call_enabled=yes SIPml-api.js:1
SIP stack start: proxy='ns313841.ovh.net:11062', realm='<sip:192.168.1.199>',
impi='1001', impu='"1001"<sip:[email protected]>' SIPml-api.js:1
Connecting to 'wss://ns313841.ovh.net:11062' SIPml-api.js:1
==stack event = starting SIPml-api.js:1
__tsip_transport_ws_onopen SIPml-api.js:1
==stack event = started SIPml-api.js:1
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js:1
SEND: REGISTER sip:192.168.1.199 SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKPuUZ2iaKJwOuZAdUoghkckt3Sl44k39b;rport
From: "1001"<sip:[email protected]>;tag=WZe9xDOwDU1MqM4cy3fQ
To: "1001"<sip:[email protected]>
Contact:
"1001"<sips:[email protected];rtcweb-breaker=yes;transport=wss>;expires=
200;click2call=yes;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: aba41a90-f377-735e-420a-26d65e26fe12
CSeq: 33815 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.04.26
Organization: Doubango Telecom
Supported: path
SIPml-api.js:1
==session event = connecting SIPml-api.js:1
==session event = sent_request SIPml-api.js:1
State machine: tsip_dialog_register_Any_2_Terminated_X_transportError
SIPml-api.js:1
=== REGISTER Dialog terminated === SIPml-api.js:1
==session event = transport_error SIPml-api.js:1
==session event = terminated SIPml-api.js:1
The FSM is in the final state
Original issue reported on code.google.com by [email protected]
on 23 May 2013 at 4:43
What steps will reproduce the problem?
Using a Zultys IPBX a SIP trace shows the invite coming from the Caller client,
however the caller client never successfully registers to our server -- could
this be from an inadequate registration timeout?
Other success with Zultys? Epygi?
What is the expected output? What do you see instead?
8/9/2013 10:34:18 AM Notice Call attempt rejected because a device
or SIP server is not authorized: Contact
'sip:[email protected]:10060', IP 188.165.231.30:10060
What version of the product are you using? On what operating system?
Latest click2dial
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 9 Aug 2013 at 3:52
We use Https on site. How we can integrate button with https? It required to
carry the script to server?
Original issue reported on code.google.com by [email protected]
on 21 Jan 2014 at 7:14
What steps will reproduce the problem?
1. Attempt to use latest webrtc2sip with SIPml latest
What is the expected output? What do you see instead?
Expected response is an SSL/TLS/DTLS handshake, instead dies at initial Client
Hello
What version of the product are you using? On what operating system?
Latest webrtc2sip, latest Doubango SVN Branch 2.0, OpenSSL 1.0.1g, sipML 1.4
(working on upgrading to 1.5)
Please provide any additional information below.
We get a failure after Client HELLO in the SSL conversation, stating "no shared
ciphers" as the issue.....
Here's the debug output from SIPml:
Error: Failed to set remote offer sdp: Called with SDP without DTLS fingerprint.
We are using the latest webML5 1.4 codebase
State machine: s0000_Started_2_Ringing_X_iINVITE tsk_utils.js?svn=224:116
onSetRemoteDescriptionError tsk_utils.js?svn=224:116
Error: Failed to set remote offer sdp: Called with SDP without DTLS fingerprint.
Can provide more details if needed
Original issue reported on code.google.com by [email protected]
on 29 May 2014 at 6:39
Hello. I can't call.
The site shows this error then I click on button:
WebSocket connection to 'wss://ns313841.ovh.net:10062/' failed: Error in
connection establishment: net::ERR_CONNECTION_RESET
AND
Request URL:wss://ns313841.ovh.net:10062/
Request Headers
Provisional headers are shown
Original issue reported on code.google.com by [email protected]
on 10 Dec 2014 at 4:35
congrats to all developers group. Its very useful for us. I trying to call sip
client but arise some problem.
I am using 64 bit windows 7 operating system and Google chrome version 25. I
have already account in ekiga.net and sip2sip.info. I follow the all step the
registration on clicl-to-call, but I cant call or receive to any sip client and
have any palace for putting PSTN number or mobile number for calling. please
help me.
Original issue reported on code.google.com by [email protected]
on 12 Mar 2013 at 6:04
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
https://groups.google.com/forum/#!topic/doubango/Cw1t9dOir9M
Original issue reported on code.google.com by [email protected]
on 27 Jun 2013 at 4:20
What steps will reproduce the problem?
1. In My account section, add my SIP A/c
2. Says "Cannot add SIP A/c"
3.
What is the expected output? What do you see instead?
I expect the SIP A/c to be accepted and back to My account menu. Instead I see
a message saying not valid SIP A/c.
What version of the product are you using? On what operating system?
I am using on Google Chrome. Also I tried it on IE. I am trying the set up via
MY Account on the browser
Please provide any additional information below.
3-4 days back I tried to set up my SIP A/c, it accepted. Now I deleted my
previous SIP A/c and when I try to define a new SIP A/c to direct the Click to
Call action, it is saying NOT VALID SIP A/c.
Original issue reported on code.google.com by [email protected]
on 28 Oct 2013 at 2:28
What steps will reproduce the problem?
1. CLick on sign-up button
2. Enter your name and email
3. press ok
What is the expected output? What do you see instead?
I'm registered
What version of the product are you using? On what operating system?
I've got "time out" message
Please provide any additional information below.
In the browser console I see that HTTP OPTIONS request to
https://ns313841.ovh.net:10072/ fails with timeout
Original issue reported on code.google.com by [email protected]
on 30 May 2015 at 10:34
What steps will reproduce the problem?
1.dowloaded two boghe
2.called to each other
3.no audio
What is the expected output? What do you see instead?
No voive transmission on both sides
What version of the product are you using? On what operating system?
latest
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 14 Sep 2013 at 1:39
What steps will reproduce the problem?
1.SIP client send HTTP GET and webrtc2sip send back 101 Switching
2.Logs shows websocket connection accepted
3.When SIP over websocket packet is received by webrtc2sip it shows following
error and send TCP FIN to close connection:
After the handshaking (HTTP GET and HTTP 101), sipflex sends REGISTER message,
but webrtc2sip shows the Error message as follows:
***ERROR: function: "tsip_transport_layer_ws_cb()"
file: "src/transports/tsip_transport_layer.c"
line: "403"
MSG: WS handshaking not done yet
A new version same error but line #407
What is the expected output? What do you see instead?
SIP Over WebSocket should be accepted and there should not be TCP close
What version of the product are you using? On what operating system?
2.0 doubango
Please provide any additional information below.
Same issue has been discussed under following thread..
https://code.google.com/p/telepresence/issues/detail?id=22
https://groups.google.com/forum/#!searchin/doubango/WS$20handshaking$20not$20don
e$20yet$20/doubango/my1wDwTzf9Y/3K9uSnf2IfMJ
Original issue reported on code.google.com by [email protected]
on 19 Feb 2015 at 7:15
What is the expected output? What do you see instead?
conection timeout to socket on your webpage
What version of the product are you using? On what operating system?
windows 7 utimate, windows 8.1
Chrome 39.0.2171.99 dev-m
Please provide any additional information below.
Connecting to 'wss://ns313841.ovh.net:10062'
2SIPml-api.js:1 Not started
SIPml-api.js:1 [C2C] stack event = starting
SIPml-api.js:1 [C2C] stack event = stopped
SIPml-api.js:3 WebSocket connection to 'wss://ns313841.ovh.net:10062/' failed:
Error in connection establishment: net::ERR_TIMED_OUT
SIPml-api.js:1 __tsip_transport_ws_onerror
SIPml-api.js:1 __tsip_transport_ws_onclose
Original issue reported on code.google.com by [email protected]
on 4 May 2015 at 10:39
What version of the product are you using? On what operating system?
OS: Windows 7
Browser: Chrome Version 25.0.1364.152 m
Please provide any additional information below.
Call us button is not working.
I pasted the code generated for call us button. After that the button appeared
on the page but when I clicked on the button, it immediately goes to 'call
terminating' state. I have pasted the <script>...</script> code in the <body>
element.
But the link is working fine. It is calling my sip client.
What could be the reason?
Original issue reported on code.google.com by [email protected]
on 14 Mar 2013 at 6:11
What steps will reproduce the problem?
1.Choose the signup #
2. enter friendly name and email
3.
What is the expected output? What do you see instead?
should complete and send email instead see "failed to sign up: timeout"
What version of the product are you using? On what operating system?
Chrome, Firefox. IE and Chrome Canary
Please provide any additional information below.
-- this could be excellent!
Original issue reported on code.google.com by [email protected]
on 30 Jul 2013 at 2:54
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 15 Mar 2013 at 7:02
What steps will reproduce the problem?
1.fully implemented a sip account to call and call from.
2. using a thrid party sip client I verified that outboud call sip account works
3. using a third party sip client I verified that inboud call from external sip
account works
4. trying to use the test click to call link or url on my account page in
Google Chrome but I get:
Failed to load resource:
http://click2dial.org/u/assets/img/glyphicons-halflings.png the server
responded with a status of 404 (Not Found)
SIPml-api.js:3 WebSocket connection to 'wss://ns313841.ovh.net:10062/' failed:
Error in connection establishment: net::ERR_TIMED_OUT
What is the expected output? What do you see instead?
The call should go through but instead I in Chrome get the console message I
provided previously.
What version of the product are you using? On what operating system?
Latest version of your product. I am on a Mac 10.8.5, our SIP registrar is an
Asterisk installation.
Thanks in advance for any help.
Robert
Original issue reported on code.google.com by [email protected]
on 15 Jul 2015 at 4:03
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