Asterisk provides many features like Voicemail services with Directory, Call Conferencing, Interactive Voice Response, and Call Queuing. It includes three-way calling, caller ID services and Skinny.

You will not need additional hardware to implement Asterisk. But it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism. Single or multiple VoIP Providers can be used for incoming and outgoing calls, though incoming calls can be handled by different VoIP Providers.

For use of digital and analog telephony equipment, Asterisk supports a number of equipment most notably is Asterisk sponsored hardware Digium. Their FXO and FXS cards are popular for small installations.

Full features of Asterisk Open Source Edition include Full PBX Capability, Switched or Packet Data, VoIP Capability, TDM Capability, Converged Telephony, Voice Mail, Unlimited-Seat Conference Bridge, SIP and IAX Support and Asterisk-centric Poundkey Linux.

Asterisk also includes the Business Edition, where it includes other features like Asterisk Technical Manual, Quick-start Guide, Commercial License with Legal Protection, 240 Simultaneous Calls, Tech Support for Install and Bug Resolution, Free Upgrades and Tech Support for One Year, Access to Business Edition Portal for One Year, Special Offers for Partner Products and many others.

Asterisk allows people to communicate using the internet. It is one of the cheapest ways to talk. The PC clients connect to each other through an Asterisk server which uses a Linux/Unix OS. Asterisk calls are passed through different channel protocols. The most popular forms of calls are H323, SIP, IAX or Zap. Depending on the type of channel you use, the user must have the appropriate software to handle the call type.

Let's say, if are a registered on an Asterisk PBX (or other PBX) as a SIP user, you are required to use a SIP phone client such as X-Lite or SIPPS. Or, if you are registered as an IAX user you must use an IAX phone client such as iaxcomm. SIP/IAX Client Configuration allows you to register SIP and IAX users. This includes the adjustment of all Asterisk and client phone settings.

To make calls with Asterisk telephone system you should connectivity to an Asterisk PBX for call handling and users using PCs with SoftPhones for calling internally and externally. All configuration information for the Asterisk PBX knows is written to the conf files located in /etc/asterisk.

All information for SIP users is stored in sip.conf and for IAX users in iax.conf. Numbers are read from extensions.conf. Voicemails are stored in voicemail.conf etc. To place a call through Asterisk, registration to Asterisk is not required. However, you do need a username and password which will allow you to call someone who is registered. To receive a call, you must be a registered user and have an extension.

When any change is made in conf files from /etc/asterisk/ or changes that relate with some of these files, you must type 'reload' in the Asterisk Command Line Interface (CLI) to make the changes effective.

It is possible to implement only the Open Source Edition for your corporate business. You will only need a Linux OS computer and Asterisk has to be installed.