[Freeswitch-users] How FreeSWITCH is being used

Dale Thatcher freeswitch at dalethatcher.com
Sat Apr 5 22:51:14 PDT 2008


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!
> 
> 
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