[NMLUG] Fun with Asterisk

[niosop at msn.com (MATTHEW BOWIE)]

Hi everyone,

  I've been playing w/ Asterisk lately and I thought I'd post my experiences 
for those of you who haven't worked with it, and to see if any of you have 
had any experiences with it you would like to share.

  The whole thing started when a vendor wanted to sell us VoIP services.  
For our ~110 phones, they want to charge around $70K for 
hardware/software/phones.  Our current provider charges around $20K a year 
for leasing equipment/phones, so the $70K for purchase wasn't too 
rediculous.  In preparation for their presentation I figured I'd bone up on 
my VoIP and install a test Asterisk server.

  For those of you who are not familiar w/ Asterisk, it is an open source 
PBX.  PBX stands for Private Branch eXchange.  It's basically the hardware 
or software that routes calls in an office.  It's a smaller version of the 
system the phone company uses to route calls.

  Ok, so I started w/ my favorite distro, Gentoo, and emerged Asterisk.  
While it was compiling I looked for a decent softphone (software phone) for 
Linux and Windows.  I decided to use a SIP (Session Initiation Protocol) 
phone instead of one that supports IAX (Inter Asterisk eXchange) because 
most IP hardphones you can buy use SIP, so I wanted to make sure that 
portion worked well.  I finally ended up w/ Twinkle for Linux and either 
SJPhone or XLite for Windows.  I may end up writing a custom softphone if we 
put the system into production.

  Ok, so Asterisk finishes installing, and I have Twinkle installed on my 
laptop and SJPhone installed on my assistant's WinXP machine.  One thing 
that should be noted is that running a SIP softphone on the same machine as 
Asterisk doesn't work (at least out of the box).  So I add two entries to 
/etc/asterisk/sip.conf, assigning usernames/passwords.  Then I add assign 
extenstions (123 and 125) to /etc/asterisk/extensions.conf.  Ok, time for a 
test.  Log on to softphone on my laptop (123) and dial 125.  WinXP machine 
starts ringing.  Assistant picks it up, and just like that we're talking.  
Ok, basic phone to phone works.  Now for fun stuff.

  I add voice mail accounts for each extensions.  Then I install Festival 
(text to speech engine) and create another extenstion, that when called just 
has festival read out some random text.  Festival doesn't sound great, but 
it's understandable, and since it's not recorded, we can use it for dynamic 
stuff.  I test out call parking.  Everything seems to work ok.

  Ok, the really cool thing about Asterisk is the ability to interface with 
it from other programs.  To have Asterisk make calls and connect to 
extension, you can simply move a .call file into it's spool directory.  This 
means any cron job, bash script, or program in any language can initiate 
calls.  There's also the Manager system which will allow to connect and 
control asterisk via TCP.  It can make calls, transfer calls, keep your 
program notified of activity, etc.  Asterisk can also call programs (which 
communicate w/ Asterisk using stdin and stdout).

  So what's all this good for?  Ok, let's look at outgoing calls.  If 
students haven't made payment arrangements by a certain date, they are 
disenrolled from classes.  Every semester there is usually over a hundred 
students on the purge list.  Two days before purge, we could run a script 
that would query the database, find everyone who is going to be purged, get 
their phone number, connect to their phone and when answered play a 
recording letting them know they will be disenrolled unless they make 
arrangements.  If it's busy, they'll be moved to a busy list so it can try 
again later.

  How bout incomming calls?  If a student wants to know what their balance 
is, they can call in, choose "Balance" from the menu, enter their student 
id, and have the Asterisk server query the database for their balance and 
read it back to them.

  We also have satelite campuses in Springer and Santa Rosa.  We can use the 
same system for them, with additional benifits.  When someone at our main 
campus wants to call a student in Santa Rosa, the call can be routed over 
the network to our Santa Rosa campus and out their line.  So it's a free, 
local call.  We end up getting free calls to anywhere we have network 
connectivity and a phone line.  The same works in reverse.  Students in 
Santa Rosa could dial a local, Santa Rosa number, and be connected to our 
main campus over our network.  The call is free for the student, and because 
they don't have to use our 1-800 number, it's free for us as well.

  A few minutes ago I set up a tunnel between my laptop and work (needed 
because SIP doesn't transverse NAT).  Phone quality seems just fine.  So I 
can be sitting at the bar w/ my laptop, connected to their wireless, and 
someone at work can dial my extension.  They will be connected transparently 
to my laptop.

  Ok, this is all very nice in a small test enviornment, but how well will 
it scale?  I don't know yet.  Total cost to replace our leased equipment 
will be about $13K (less than a single year of leasing our existing system). 
  This includes buying phones, hardware, interface cards, etc.  So it makes 
sense financially, it gives us additional abilities that we don't have yet.

  I'm requesting $500 for some phones and FXO/FXS boxes (to convert between 
analog data and IP packets) to see how well it works together and give us a 
change to test out some different brands/models of phones to test for 
potential problems.  Whether or not we actually move to this system remains 
to be seen (getting any changes through takes an act of god), but it should 
be fun trying it out.

  If any of you are currently using an Asterisk system, I'd love to hear 
about your experiences with it.  And if anyone is interested in setting up 
their own system and has questions, feel free to contact me.

Matthew