Open Stack

On Sun, Jun 30, 2013 at 12:43 PM, James E. Blair <jeblair at openstack.org> wrote:
> Hi,
>
> There seems to be more interest in having an Asterisk server for the use
> of user group meetings, conference calls, and summit telepresence.
>
> I'd like to get started setting one up, so this mail is to kick of a
> discussion of how to do that.  I think we need:
>
> 1) To decide on a version of Asterisk and package source.
> 2) Acquire or develop some puppet scripts to manage it.
> 3) At least one VOIP provider.
>
> I'm personally interested in using the WebRTC capabilities in newer
> versions of Asterisk, which I'm afraid may mean using something other
> than the version package in Ubuntu Precise (Version:
> 1:1.8.10.1~dfsg-1ubuntu1).  Does anyone have thoughts on what version to
> run and where to get it?  We could run this on Ubuntu Precise or CentOS
> 6.
>
Personally, I would stay with Asterisk 1.8, but that is just my
opinion. WebRTC support in Asterisk is still maturing and I wouldn't
count on using it for production for a little longer.

As for the Asterisk package, don't expect to see anything greater then
1.8 from Debian / Ubuntu until some newly embedded libraries are
removed.  I am not sure about REL, I'm sure Russell knows.  Other
option are compiling from source or rolling our own packages, but not
sure we'd want to take on that responsibility.

> Jesse Keating sent us the old puppet scripts that the Fedora project
> used to use.  They are likely to be somewhat outdated, but could be a
> starting point.  Does anyone have something more recent we should start
> with?
>
Here's the puppet modules I use for my asterisk deployments[1]. They
worked great for my needs, however some work on my side would be
needed to split them out.  I've been meaning to get around to doing
it, but sadly other things come up.

Managing Asterisk with Puppet works pretty well actually, I don't
think I have had any issue between both of them.  The real decision
point comes down to how you plan to configure asterisk, eg realtime vs
static files.  I prefer static files, which makes puppet happier.

> I contacted the Eclipse foundation and got a reference from them for a
> couple of good VOIP providers -- when we get closer to having something
> ready, I'll look into setting up an account.
>

I've used both http://voip.ms and http://flowroute.com in the past and
have had no real issues.

[1] https://github.com/kickstandproject/puppet-modules/tree/master/modules/asterisk

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