My name is Conrad de Wet, you can find more information about me at:
https://www.linkedin.com/in/conrad-de-wet-5488a79/
Since the beginning of my entrepreneurial career, back in 2004, my focus has been on developing technology that is commercially viable and customer-centric. Although I enjoy rolling my sleeves up, it gives me tremendous pleasure to see my highly committed team succeeding (even when the standards have been set near impossibly high).
The past 9 years (2010-2019) at Euphoria Telecom has been an amazing journey, however the most fulling moments that will stick with me the longest are the testimonials from customers where the product we delivered, and the service provided has made positive impact on them, their staff and their business.
My goal is to inspire, teach and generally give back to the community, that has given me so much over the years. In this channel we will dive into Asterisk, and what cool things we can do with it. This channel is for you if you are: interested in telephony, Linux, Asterisk, electronics, DIY, development, home automation, and cloud computing. Join me, and let’s have a ball.
How to Support these projections or make a donation: /support-and-donations/
See you soon.
Dear Conrad,
I am writing to you on behalf of the University of Applied Sciences Wiener Neustadt in Austria. We are participating currently in a research project and found your Browser-Phone github project.
We are wondering if you would be willing to license this under a different license than the AGPL because we could use some of your great code in our project but can’t open source the whole project.
I would really appriciate if you could get back to me to discuss possible terms.
Thanks and have a nice weekend,
Philipp Kolmann
Thanks for contacting me, i’ll reply privately.
Conrad, what you make is amazing!
Is there an option to have a simple authentication webpage asking only the user his/her sip username (optionally, with or without (predefined) sip server host name) and sip password. Upon successful auth, this webpage then redirects the BrowserPhone, populating (or, optionally, not populating) the rest of the user account’s info, and immediately connects to the sip server? Please, add this super amazing option. Thanks in advance!
The browser phone is very flexible in the way you can provision details, like server etc. You can also use the existing windowing system (provided by jquery UI) to prompt people for details.
Check the code for = ‘function OpenWindow(…)’ to open a window for example to ask for username and password. And then to save the details, you can just save them to the local storage like this:
https://github.com/InnovateAsterisk/Browser-Phone/blob/32c1cabced4a91b7300960bf24e95be060bba794/Phone/phone.js#L61
Hello sir, I really like what you do.
I am looking to make a Browser Phone with Ubuntu for Raspberry Pi but port 443 is not listening, what should I do?
Hi William, Thanks for you feedback
Port 443 is a system port (0 – 1024), so only if a service is running as root will it be able to listen in on that port and only if something else isn’t already attached.
Dear
I am professional web developer
I have some problems related with webrtc web sip phone development
If you have time, please contact me
I need your help really.
From Mikalai
Dear Conrad,
I found your Browser-Phone github project.
That solution is great. so I gonna use the solution in my platform.
but my platform is different from your platform.
About these, I would like to hear your suggestions.
Best Regards
Mikalai Saviski
Hi Mikalai, Glad to help!
Please will you make sure that in the future you post issues and questions with Browser Phone to https://github.com/InnovateAsterisk/Browser-Phone/issues
Dear Conrad,
That solution is great. so I gonna use the solution in my platform.
while integrated the solution with my platform, I can see some issues.
If you have time, please contact me in github.
About these, I would like to hear your suggestions.
Best Regards
Mikalai Saviski
Dear Conrad,
If you seperate Browser phone with server(so if run Browser phone and asterisk on other servers),
it will not affect to Browser phone working?
Best Regards
Mikalai Saviski
The Browser Phone code can run from or be hosted on any server.
hello sir thank you for the tutorial.
I have a question what do you mean by: “be sure you control a domain, and are able to add DNS entries.”?
I don’t quite understand this point at the moment I fail the cerbot verification.
can you help me ?
Hi Conrad,
I was trying to connect you over LinkedIn but it did not allow. So i am dropping a message here. Pls drop me a mail, if you happen to read this. Would like to connect with on to how to take raspberry pi device to a cheaper low cost, work desktop (preferably windows-ce) for contact center staffs, and what steps should be taken for mass production. Need a USB keyboard also.
Hi Abhijit, my LinkedIn account is a little bit neglected these days. I’ll try find you there.
would you be interested in doing some modifications to your BrowserPhone project, paid of course. Let me know, or if you know of someone. Trying to replace the old fonality HUD with something more up to date.
Hi, for the moment, I don’t have capacity to do custom development, but if you have suggestions that the community may benefit from, then it may be included in the master branch.
https://github.com/InnovateAsterisk/Browser-Phone/issues
Hi conrad
is it possible to create a wordpress plugin of the browser phone?
kindly contact me privately for further details
I’m sure it’s possible, but account management & security will be the main challenge.
Hi Conrad,
Thank you for this and currently following the asterisk installation.
However, since I am new to the PBX world, I am having a hard time starting.
We have an existing freepbx, working on microsip softphones, UCP is also working.
How can i point the existing configs we have to your webrtc dialer.
Hoping you can help me. Thanks
You can start with the hosted version for now. It will eliminate the need to host the html and JavaScript. Simply open the page in a browser:
https://www.innovateasterisk.com/phone/
In the settings that appear you can now enter the server details of your installation so long as you have correctly setup and enabled WebRTC endpoints on Asterisk/FreePBX.
When it comes to the server location please be sure to use a fully qualified url for the websocket server address. It is possible for this to be a .local address for your local lan, or a hosted server address. Either way it must be reachable via a TLS connection port.
Hi Conrad,
Your work is amazing!
Would you be able to support us? we are developing a click-to-dial service using an Angular web client app and encountered some issues with STUN/TURN – the client registers successfully and can place calls. Still, there’s 1-way audio and a long latency while connecting the call.
Hi Luis,
We will be launching a commercial WebRTC service around May 2024 called SIPERB: http://www.siperb.com/. We will offer paid for Technical Support as part of the paid for service. The service will be both the SIP proxy, transcoding, and a WebRTC client (Browser Phone). Feel tree to pop your mail in the mailing list – so we can keep you up to date on the progress.
Hello @Conrad,
I hope this message finds you well. I am currently working on a project involving Asterisk and am seeking expert guidance on tuning and optimization. Given your extensive experience in this area, I would greatly appreciate the opportunity to discuss my project with you. Could we arrange a time for a consultation or a brief chat?
Thank you in advance for your help.
Hi, I’m mostly focused on Siperb at the moment. (https://www.siperb.com/) Sorry, I can’t take on any consulting jobs right now.
Hi conrad
Thank you very much for this system (Browser Phone)
I installed Browser Phone. Works very well.
This system supports video conferencing?
I couldn’t find this function
Yes, just make a buddy using the conference number as the extension number, and use the video call button.
I added the following:
exten => 600,1,Confbridge(default_bridge)
Only audio works. Video not working. Also added video_mode=sfu in confbridge.conf
Can you post your issue on the gitHub Site, thanks:
https://github.com/InnovateAsterisk/Browser-Phone/issues
Hello Mr. Conrad,
I would like to connect your open-source browser phone to my FreePBX system. Would it be possible to create a tutorial explaining how to integrate it into FreePBX?
Thank you in advance!
Best regards,
Seteven hav
Thanks for you message! I’ll be sure to add this to the list of upcoming videos.