Re: [PLUG] Asterisk advice

[Kevin McAllister <kevin@mcallister.ws>]

On Mar 22, 2011, at 8:38 PM, Eric at Lucii.org wrote:

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> Now I'm wondering if it would be better to set up my own asterisk box.  I know
> that I have to use a provider like Vitelity (or similar) to get my phone
> numbers.  I'll have to add a card to support the existing analog phone
> infrastructure in the house/office.  I have a couple of extra computers here and
> I'm reasonably certain that either one of them is up to the task.
> 
> Is that about it?  Is Asterisk a bear to set up?  I'm a bit of a hacker but have
> only minimal telephony knowledge.  I keep seeing trixbox community edition
> mentioned... is that a good way to get up and running?

Asterisk is relatively easy to use.  On ubuntu it's in one of their repositories.  Even if you install from source the steps are straight forward to follow.  A few points you may want to look into:

1. disable or block stuff you don't want to use.  (You can disable (noload) modules in the /etc/asterisk/modules.conf" By default lot's of different protocols are enabled.  Know what protocols are open to send calls into your phone system or someone will use it for you to make international calls.  On that topic also setup fail2ban because if you have SIP open someone will try and register.

http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk

2. You won't necessarily need an FXS card for your machine you could use an Analog Telephone Adapter (ATA) to drive your analog telephone network.  We've used the old SIPURA now the Cisco line of small business ATAs http://www.cisco.com/en/US/products/ps10024/prod_models_home.html

You can probably get an old one easy on ebay, or check telephonydepot.com they're in Manayunk.

3. I don't know anything about trixbox, but that probably makes it way easier to disable enable stuff through a web interface.  Probably worth a shot, I think they have a live CD.  Maybe Digium has one now called AsteriskNOW.

4. If you really want to hack you can hook it up with google talk (libjingle) and skype that will cost money a few bucks for the software license (http://www.digium.com/en/products/software/skypeforasterisk.php)

But basically you'll be receiving SIP traffic on the interface and routing them appropriately to either a SIP ATA or a DAHDI interface, if you go the digium/sangoma card route.

The other thing you'll want to be cautious of is backing up your config.  Trixbox might make this easy, but it can't hurt to keep a current tarball of /etc/asterisk or put it into some source control.

I don't know much about the SIP providers out there.  But know quite a bit about Asterisk, feel free to ask me questions.
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