[Freeswitch-users] [newbie] questions

[Michael Collins]

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 at 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 at freeswitch.org
> > <mailto:consulting at 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 at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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