For those wanting to dabble in VoIP to get free outgoing phone calls, setting up an Asterisk server and using your Gmail/Google Voice account for outgoing calls is a great start. If you don’t already have a linux box to mess around with a VPS can be had for less than $10/month. I’m currently using a VPS from Digital Ocean configured with Ubuntu 12.04

I prefer downloading the tarball and compiling the latest version of Asterisk (Available here). I prefer to download and unpack the tarball in /usr/src

cd /usr/src
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk 11-current.tar.gz
tar xvf asterisk-11-current.tar.gz
cd asterisk-11-current

Next we’ll run the configure program to check dependencies.

./configure

This will likely let you know that you are missing some packages. If starting from a newly installed VPS you’ll likely need the following packages first:

make
libncurses-dev
libxml2-dev
sqlite3
libsqlite3-dev
libiksemel-dev  (required for gtalk/jingle)
libssl-dev  (required for chan_sip)
subversion (if adding the mp3 addon)

Once ./configure has run without any errors we want to select the options and make sure we will have the required channel drivers (sip and gtalk):

make menuselect

Make sure that chan_sip, chan_gtalk and chan_jingle are available and selected:

 **************************************************
 Asterisk Module and Build Option Selection
 **************************************************
      Press 'h' for help.

      Add-ons (See README-addons.txt)
      Applications
      Bridging Modules
      Call Detail Recording
      Channel Event Logging
--->  Channel Drivers
      Codec Translators
      Format Interpreters
      Dialplan Functions
      PBX Modules
      Resource Modules
      Test Modules
      Compiler Flags
      Voicemail Build Options
      Utilities
      AGI Samples
      Module Embedding
      Core Sound Packages
      Music On Hold File Packages
      Extras Sound Packages

Now that we have selected our addons etc we can compile:

make
make install
make samples

You now have a working asterisk system and can begin configuring your sip.conf and externsions.conf files.