[Freeswitch-users] How FreeSWITCH is being used

Sun Apr 6 23:07:32 PDT 2008

Dale,

When I added the link for "how fs is being used" I was envisioning that
those posting would include their hardware and OS environments.  In
fact, I was kicking around having a small template for the info we'd all
like to see included in all of the examples.  

E.g.:
Number fs servers:
Operating system:
Hardware platform: (CPUs, RAM, RAID, NIC speed, etc.)
TDM: Y/N; if yes, what kind of TDM card(s)

Stuff like that.  If you have any ideas on what else would make for good
case studies or whatnot on "how fs is being used" then please let us
know.

Thanks!

-MC

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Dale Thatcher
> Sent: Saturday, April 05, 2008 10:51 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] How FreeSWITCH is being used
> 
> The "How FreeSWITCH is Being Used" is still coming soon on the wiki.
> I'd be curious to see what configurations of OS/hardware people have
and
> how they're using FreeSWITCH.  Think it's worth adding a page on the
> wiki for this?
> 
> - Dale
> 
> ----------------------------------------------
> http://myhelpa.com - The web when you need it.
> 
> On Sat, 2008-04-05 at 21:37 -0700, Michael Collins wrote:
> > Tim,
> >
> >
> >
> > If you're comfortable with Linux then I would suggest starting
there,
> > but you are by no means locked in.  Much of the development of FS is
> > done in CentOS, and I believe in a 64-bit hardware environment.
> > You'll also find that the current documentation is slanted toward a
> > Linux/Unix environment.  This is probably due to the fact that most
FS
> > users don't fall into the category of Microsoft fans. J  However, if
> > you're 100% VoIP then Windows should be fine.  (Please tell me
you're
> > not on Vista!!)  The PSTN/TDM stuff gets a bit tricky in Windows, so
> > be on the lookout if you are thinking about future PSTN
connectivity.
> >
> >
> >
> > Personally, I'm using CentOS 5.1 and I'm extremely happy with the
> > performance and setup.  For the record, if you choose Gentoo then
> > you're quite likely to get yelled at! J
> >
> >
> >
> > Let us know how it goes.  We'd be curious to see how FS stacks up in
a
> > head-to-head comparison against Asterisk in your application.
> >
> >
> >
> > Thanks again for checking it out,
> >
> > MC
> >
> >
> >
> >
> >
______________________________________________________________________
> > From:freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Tim Meade
> > Sent: Saturday, April 05, 2008 8:55 AM
> > To: 'freeswitch-users at lists.freeswitch.org'
> > Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
> >
> >
> >
> >
> > Thanks Michael.
> >
> >
> >
> > Well I'm going to give it a try.  I'm off to find an IRC program
> > now.
> >
> >
> >
> > I've identified the issues with asterisk and it's with my providers.
> >
> >
> >
> > Big question:
> >
> >
> >
> > I have a fedora and windows dev boxes.   What's the best bet for
> > someone new to the software?
> >
> >
> >
> > Thanks
> >
> >
> >
> > Tim
> >
> >
> >
> > From:freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Michael Collins
> > Sent: Saturday, April 05, 2008 3:04 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
> >
> >
> >
> >
> > Tim,
> >
> >
> >
> > I can attest to the fact that FreeSWITCH is a great project with a
> > really cool community.  Right now the IRC channel is the best place
to
> > get quick, specific information on how to handle the kinds of things
> > you're looking to do.  One of the important things about FS is that
> > the devs made extremely wise engineering decisions long before a
> > single line of code was written.  They made sure that FS would be
> > extremely flexible in what it can do.  One byproduct of that is that
> > we get visitors asking, "Hey, can FreeSWITCH do this?" when no one
> > here had thought of that before.  In many cases it can indeed "do
> > that," but it takes a little tweaking and some attention from the
> > experts.  Definitely hop on the IRC channel #freeswitch; the main
devs
> > generally are there during the day and some evenings.
> >
> >
> >
> > If you do get FS up and running, especially in a production
> > environment, we would ask that you consider documenting your setup
on
> > the wiki.  We hope to lower the barrier to entry for others with
> > similar needs by having good documentation, but of course we also
like
> > to brag about what FS can do! J
> >
> >
> >
> > Thanks for checking out FreeSWITCH!
> >
> >
> >
> > -Michael, aka mercutioviz
> >
> >
> >
> >
> >
______________________________________________________________________
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Tim Meade
> > Sent: Friday, April 04, 2008 5:34 PM
> > To: 'freeswitch-users at lists.freeswitch.org'
> > Subject: [Freeswitch-users] freeswitch as a SIP Bridge
> >
> >
> >
> >
> > Greetings all,
> >
> >
> >
> > I've just stumbled upon your project and it may solve an issue we
are
> > having.
> >
> >
> > I've just spent about 3 weeks getting to know asterisk just to
> > discover I don't think it can do what I need.
> >
> >
> > We have a project where we have incoming calls on a SIP channel.  We
> > need to do a direct forward of these calls to an outgoing channel
> > based to a number which is from our database.  Simple to do in
> > asterisk, but the problem is that we cannot have these calls
> > "connected" between the two lines.   They have an automated message
at
> > the beginning that is being activated when we do the answer before
the
> > dial of the second number in asterisk.
> >
> >
> >
> > Out first idea is to bridge the incoming call directly to the
outgoing
> > call.  The problem is that the incoming call cannot be "answered"
and
> > then we initiate the outgoing call.  It needs to be a seamless
bridge
> > between the two calls.   A nice feature would be to have a timer on
> > the call. I saw a bounty for the timer feature, so I'm guessing
> > (hoping) the bridging part can be done now.
> >
> >
> >
> > One other thought we are having is the ability to leave the incoming
> > line "ringing" and dial the outgoing line until it is answered.  At
> > that time, answer the incoming and then bridge them together.
> >
> >
> >
> > So my question is:  Can freeswitch do these things?
> >
> >
> >
> > Thanks and congratulations on the nice work!
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
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> 
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