I've recently launched a free service to make WebRTC calls from a browser to phones running Lumicall
Just dial a number in the browser and if the phone has Lumicall then it will ring - nothing to pay. And the software is free too. The blog and the Lumicall site have had a tenfold increase in traffic since this was released.
I've explained it in more detail in my blog, any feedback would be very welcome:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256
On 09/12/13 21:29, Daniel Pocock wrote:
Just dial a number in the browser and if the phone has Lumicall then it will ring - nothing to pay. And the software is free too. The blog and the Lumicall site have had a tenfold increase in traffic since this was released.
Wow I'm not surprised it's generating traffic - sounds like an extremely interesting feature with some very practical features.
I'll be reading your explanation when I have a few minutes. I'm sure thousands of webmasters would love to integrate FS dialling from their sites and webapps.
Sam.
- -- Sam Tuke Campaign Manager Free Software Foundation Europe IM : samtuke@jabber.fsfe.org Latest UK Free Software news: uk.fsfe.org Is freedom important to you? Join the fellowship.fsfe.org
Daniel, Is there any possibility you would make a diagram of your known working servers available, along with the specific configurations you use? I find it difficult to understand what each line is supposed to do, what each configuration option does to the SIP connection. Certainly, you could redact any line that affects security of your servers.
I believe the several other people who have posted "It doesn't work for me" would also benefit from seeing a fully-formed, known-good configuration.
Thanks, Derek
On Tue, Dec 10, 2013 at 7:54 AM, Sam Tuke samtuke@fsfe.org wrote:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256
On 09/12/13 21:29, Daniel Pocock wrote:
Just dial a number in the browser and if the phone has Lumicall then it will ring - nothing to pay. And the software is free too. The blog and the Lumicall site have had a tenfold increase in traffic since this was released.
Wow I'm not surprised it's generating traffic - sounds like an extremely interesting feature with some very practical features.
I'll be reading your explanation when I have a few minutes. I'm sure thousands of webmasters would love to integrate FS dialling from their sites and webapps.
Sam.
Sam Tuke Campaign Manager Free Software Foundation Europe IM : samtuke@jabber.fsfe.org Latest UK Free Software news: uk.fsfe.org Is freedom important to you? Join the fellowship.fsfe.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.15 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
iF4EAREIAAYFAlKnDxMACgkQ1bR1Itj7YQUX+AD/XVb/SIwnQX6csq/0nT6Puitk 2zhfGbnOsiFdzUbvGqsBAKNC4keDO3knMV+wbwlSXioRVIuevhcMhdubLaVk3LP5 =qi20 -----END PGP SIGNATURE----- _______________________________________________ Free-RTC mailing list Free-RTC@lists.fsfe.org https://lists.fsfe.org/mailman/listinfo/free-rtc
On 10/12/13 15:47, Derek LaHousse wrote:
Daniel, Is there any possibility you would make a diagram of your known working servers available, along with the specific configurations you use? I find it difficult to understand what each line is supposed to do, what each configuration option does to the SIP connection. Certainly, you could redact any line that affects security of your servers.
Thanks for this feedback, I've put the config in a blog, please let me know if this helps you reproduce my own setup: