naidu / click-2-dial Goto Github PK

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.

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

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!

We use Https on site. How we can integrate button with https? It required to 
carry the script to server?

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.

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

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 

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.

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

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.

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.

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.

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.

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?

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.

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