[Freeswitch-users] freeswitch as a SIP Bridge

[Michael Collins]

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. :-)  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! :-)

 

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! :-)

 

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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