If you're willing to get your hands dirty then that's great. I recommend the following resources:

* Our "bridge" book - it covers a lot of the basics of FreeSWITCH. There are links to Amazon and Packt Publishing from our main wiki page
* The forthcoming FreeSWITCH Cookbook. You can pre-order now. I'll throw some links up on the main page and wiki page, but if you go to PacktPub.com or Amazon.com just do a book search for FreeSWITCH and you'll find it. There's only two titles. :P
* You're already on the mailing list, so karma++ for you
* The IRC channel: #freeswitch on irc.freenode.net. This is, hands down, the best place to talk in real time about FreeSWITCH stuff
* The wiki - it's got TONS of information. Yes, I'll admit you may have to dig and you may find confusing or contradictory information but we do our best to keep it updated. Just let us know if you find that something is wrong/confusing/missing on the wiki
* The Wednesday conference calls - We do a weekly conference call for members of the community. You can interact with power users and occasionally even the developers themselves. We usually have a "feature presentation" after which we let folks do an open Q&A.

As for the choice between Asterisk and FreeSWITCH: we respect your decision, and I promise we won't cry if you choose Asterisk or PIAF or whatever. However, many members of the FS community are "Asterisk refugees" who've been burned by mysterious deadlocks, segfaults, scaling issues, etc. If you ask us about our Asterisk experiences be prepared for horror stories. :D That being said, lots of people use Asterisk with no apparent issues. It's up to you.

No matter which way you go, be prepared for aggravating problems and pulling your hair out. VoIP and telephony are always frustrating for the new user. Most of our community members remember their newbie pain and are quite willing to offer their knowledge and experience to help you keep at least some of your hair. ;)

Let us know what you decide!

-MC (IRC: mercutioviz)

On Thu, Feb 9, 2012 at 6:09 AM, Josh <mojo1736@privatedemail.net> wrote:

> Welcome to FreeSWITCH!
Pleasure.

> But calls will be sent to FreeSWITCH by some device, correct? If it's
> good old-fashioned SIP then FreeSWITCH will handle it just fine.
Yes, there are all softphones (PC machines with old-fashioned headset &
mic and smartphones running a sip client).

>
>     Could this be done relatively easily in FreeSWITCH?
>
> "Relatively?" Of course! It's relatively easy for someone with some
> experience. I highly recommend that you ask consulting@freeswitch.org
> <mailto:consulting@freeswitch.org> for professional assistance if you
> are not comfortable doing this all by yourself.
I'd rather do it by myself. For now though, I'd like to see whether what
I want to achieve is possible in FreeSWITCH. I haven't yet made a
decision what to use though - Asterisk or FreeSWITCH, it will all depend
on whether I could set it up properly and "relatively" easy. Having said
all that, as I developer with quite a bit of experience behind me, I am
not afraid to delve in and get my hands dirty, if needed. I just want to
make sure that what I want in terms of set up and functionality is possible.

> Yes, FreeSWITCH can bind to multiple interfaces. In FreeSWITCH lingo
> that would mean that you set up a separate SIP profile for each
> interface. (In fact, you can have more than one SIP profile on a given
> interface since the profile is a unique combo of IP addr and port number.)
I presume different profiles can "talk" to each other, right? In other
words calls/media can be routed/transferred from one interface to
another (eth1<->tun0 for example)?

> "Some assembly required." :D
> FreeSWITCH can do some stuff for you, but you definitely need to make
> sure that your NAT is not behaving badly, like having a SIP ALG.
This is what I am trying to figure out - do I rely entirely on
FreeSWITCH (if not, what is expected of me to set up so that FreeSWITCH
can do its job?), or do I have to do it all by myself with the kernel
module helpers (sip, h323 etc) and ip/iptables?

> I'll have to defer to Ken Rice on this one. I know he's working on
> RPMs for FreeSWITCH but I think it's all RedHat right now.
RedHat is good, all I need is a decent .spec file - I'll do the rest
myself, no problem.

One thing I seem to have forgotten from my newbie list of questions - I
take it FreeSWITCH can do call-recording (in both directions) right? If
so, how is this stored/implemented (I hope both ends are stored as a
single sound file)? I would like all calls to be recorded as a matter of
policy, so I do need this implemented if I am going to use FreeSWITCH.

Many thanks again.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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