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Comment by jessedhillon | Hacker News Reader

Can someone explain, or point the direction to a good explanation for noobs about how to go from setting up FreeSWITCH to actually making/receiving calls on the PSTN? And maybe, how to get phone numbers, or place calls with a certain caller ID?

I'd like to see a good primer on the topic but, frankly, I don't even know what to Google for.

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- You need to install FS on your server and run it. There's an installer for that (Check

http://www.plivo.org/get-started/).

- You need to install Plivo on the server as well. There's a separate installer for that.

So now you have your Twilio-like instance ready.

- To buy a number you can head to ipkall.com (free) or icall.com or flowroute.com (paid). There are several such

services actually.

- Once you have the number, you can just point it to your server that is running FreeSWITCH and Plivo.

- You can buy minutes for usage from a third party such as IVOX VoIP (ivoxvoip.com). Again, there are several such

providers out there.

- Now, go ahead and deploy your Plivo app (just as you would deploy a Twilio app).

I hope this makes it clear.

You know of any recommendations for something a little more in depth? Maybe a book recommendation, if not much is online?

Often times, the terminology is confusing and difficult to know where to start and ease in.

It's probably best to get yourself some books. The online documentation for both Asterisk and FreeSWITCH is shit, the IRC channels are mostly useless unless you already know a lot and the configuration files are just plain weird (Asterisk uses ini-style files with embedded programming, FreeSWITCH's configuration format pretends to be XML (which it certainly isn't)). Asterisk also has severe performance problems and AFAIR has some extremely bad code.

The entire idea with Plivo (much like Twilio and Tropo) is to simplify i.e. let web developers build telephony apps without having to learn much about a telephony engine.

http://wiki.freeswitch.org/wiki/Getting_Started_Guide has a good overview of getting FreeSWITCH going. However, if you have no experience with VoIP or SIP, things may be a little rougher.

If you don't want to muck with FreeSWITCH's XML configs, blue.box (http://wiki.2600hz.org/display/bluebox/Home) is a good GUI tool that will configure your switch for you. See the quick start guide (http://wiki.2600hz.org/display/bluebox/Quick+start+configura...) but know the docs are undergoing expansion and clarification. If you have questions, we're available on Freenode in #2600hz and have a pretty active community that likes to get folks up and running.

I work for 2600hz so I'm biased though :)

Is there a reason you guys went with FreeSWITCH and not SIP Servlets/MobiCents or some other proprietary stuff he backend SIP code? Asterisk sucks for SIP...

Several reasons, among which include our co-founder is a co-author of the FreeSWITCH book, we love FreeSWITCH's eventing model, we built Whistle in Erlang and Andrew Thompson (Vagabond) had already built mod_erlang_event so we can tap into the event system with native Erlang terms.

We believe strongly in open source wanted every piece of the puzzle to be open source.

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