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Tellme - www.Cognation.net/Asterisk/TellMe
Katrina

60 Second Asterisk Overview

 

Cognation doesn't provide consulting services for Asterisk.
This information is here for informational purposes only.
I'm a strong supporter of Asterisk and I couldn't run my business without it.

 

Asterisk is an open source Linux based ip-pabx. That means you can download for free one of the smartest and most flexible IP PABX solutions available today.

To start check these links out;

www.asterisk.org

www.digium.com 

 

There is also growing wealth of information on the wiki  http://www.voip-info.org/wiki/view/Asterisk   (although a little unorganized)

Asterisk includes features out of the box such as

  • Voicemail to email

  • Fax to email

  • Conference rooms with secured passwords

  • Gui web control for conference rooms

  • Agent and call centre stats

  • Web displayed operators console including drop and drag transfers

  • Dial from outlook/dial from html command.

  • Text to speech – when I dial extension *61 my asterisk box downloads a text file of the New York weather report from the bureau of meteorology and then reads it aloud to me –      (to get it to do this took only 40 lines of code)

  • Remote agents – you could answer your office extension from your home pc.

  • Auto/manual recording of calls (auto archive or email of calls to conversation participants)

 

For the non technical there is now Trixbox (originally called asterisk@home - don’t be put off by the name people run entire companies on this version)

The Trixbox solution the easiest way to get started. It is an .iso cd that you burn, load into a suitable PC (I run mine on a P3-1ghz) and this super smart scripting code automatically installs the following software;

  • Asterisk (the open source switching software)
  • AMP (an open source release of a gui configurator) they have their own separate sourceforge website https://sourceforge.net/projects/amportal
  • FOP (a graphical web page for transferring calls, monitoring who is online etc) http://www.asternic.org
  • Web meetme (a graphical web page for monitoring and controlling conference calls)

If you want to link multiple sites together and use a closed dialing group to connect your sites together via a vpn to save money with free inter office calls then checkout this great tutorial on setting up DUNDI.
http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf

 

Trixbox is available from www.Trixbox.org

There is an excellent setup guide here http://www.sureteq.com/asterisk/trixbox.htm

Or even better check out this step by step video that documents the entire process from start to finish.
http://www.asterisktutorials.com/videos/Getting_Started_with_TrixBox/Getting%20Started%20with%20TrixBox.html

BTW you might want to check out www.cognation.net/asterisk/katrina as an example of rapid deployment. 

Asterisk gives you for free what today the very best Cisco, Nortel or Avaya would charge you $40k+ for.

Asterisk also gives you the ability to intuitively customize your solution as much or as little as you want, allowing this project to be open sources means you have 1,000+ developers out there working for you on customizations you can implement when/if you choose and as this technology is gpl’d you never need to worry about increasing costs or license fees.

 

Dean Collins
 Cognation Pty Ltd
dean@cognation.net
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

 

 

How do I run Asterisk?

I've had a couple of people get in touch with me to ask how I run my own personal asterisk solution, so here is a run down on the hardware and voip providers I use.

HARDWARE
My server is an old P3 1ghz - 512mb - 20gb - intel motherboard with onboard video and 100mb nic in a rackmount case.
The great thing about asterisk is how low the processing requirement is. This is a pc I was using for another application years ago that I would probably have tossed by now but Asterisk never uses more than 10% of the processing power.

I use the http://www.digium.com/en/products/hardware/tdm400p.php card to terminate pstn onto this server.
I have two X100M modules but am only using 1 of them at the moment Packet8(in Australia I terminated my Telstra pstn line into the other).

 

Handsets
My main handsets are 3 x Polycom 500 with 3 individual line appearances, I'll probably update to a Polycom 601 with 6 line appearances soon but at the moment I have 2 on my desk and 1 in the other room of my apartment.

I dont use any cordless handsets at the moment however my GSM mobile the HTC TyTN has a wifi and gprs connection which I run a WM5 sip client on.

 

SOFTPHONES
 I use have used both firefly and windows messenger as sip clients in the past, currently I'm using IAXcomm as my laptop softphone.

 

VOIP CARRIERS
 My primary carrier in the USA is a www.Packet8.net service. For $19.95 a month I can receive unlimited incoming calls and unlimited outbound calls to both landlines and mobiles in the USA.
They also offer more expensive services that have unlimited outbound calls to Asia or Europe etc.


I also have a backup service from www.Stanaphone.com that I can receive  unlimited inbound calls as long as I have a positive balance (eg I put $5 on it a year ago and have been using it ever since). This number also acts as my fax service because when it receives a fax call it converts it to a pdf and emails it to me.


 My primary Australian number is provided by www.Faktortel.net.au
For $9.50 a month I can receive unlimited inbound calls from my Australian clients (eg a local call for them in Sydney) which is then delivered over the internet to anywhere in the world that my Asterisk server is located (e.g. here in NY).

So this means that when my clients in Australia want to ring me, they dial a local Sydney number and for 10c unlimited they could technically talk to me 24x7 at no more than this 10c charge.

The other advantage of this is that when I make outbound calls to Australia it costs me the same as if I was in Sydney (eg. 10c per call unlimited time frame).......are you starting to see why voip is cool :)


I also have 3 other voip services terminating on my server that are various USA numbers eg NY 212 or SF 415. Basically these are numbers purchased by clients I represent here in the USA.

Although I answer these calls anytime 8am to 6pm, outside of these hours they are automatically directed to a customized voicemail for each client answered in their name (the voicemails are then sent as wav files to my email address so even if I'm traveling I can still return these calls as soon as I get the email).

The cool part is that should I wrap up my contract with these clients, the voip service can be quickly and easily transferred to be answered somewhere else anywhere in the world.