[asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Andrew Latham]

On Mon, Aug 20, 2012 at 10:52 AM, Joshua Colp <jcolp at digium.com> wrote:
> ----- Original Message -----
>> Joshua
>>
>> Can you copy and past into a wiki page for everyone's benefit?  Maybe
>> https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
>> like page would be good.
>
> If this thread has taught me anything it's that there needs to be a complete wiki page, just copying/pasting what I'm saying here isn't enough. It's on my list. I won't call it a demo setup though... since it won't actually work yet.
>
> --
> Joshua Colp
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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Agreed, but we need something and a place for comments.  The wiki is
great because we can rename and move things when they are no longer
relevant to our needs.

-- 
~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~