Asterisk Today

From caller ID to long distance, anything your phone can do, Asterisk can do better and cheaper. Asterisk, an open source telephony project sponsored by Digium , greatly reduces the cost of traditional telecommunication technology and operation, and moves Voice over Internet Protocol (VoIP) into the mainstream.

With VoIP, telephone calls are transmitted over an Internet connection, eliminating long distance charges and the need for a traditional proprietary telephone service plan. If you own a telephone, heed the call to Asterisk.
 

 

Because Asterisk is based on VoIP, Asterisk provides an inexpensive telecommunication solution perfect for a small business, a home office, or even an entire household. And because Asterisk has the ability to communicate seamlessly between VoIP and the public switched telephone network using any of the most popular codecs and protocols, it allows for high voice quality at no toll costs. Also, Asterisk's freely and widely available open source code can replace a traditional hardware PBX. Finally, unlike most VoIP telephone systems, Asterisk integrates with a wide range of hardware and standards-based telephony equipment.

"Asterisk was designed to be able to do everything a traditional telephone system can do, and much more," said Mark Spencer, creator of Asterisk and founder of Digium.

The Linux configuration of Asterisk offers a myriad of calling features, including caller ID, call waiting, and voicemail. Moreover, it has all of the advanced capabilities of a professional-grade telephone system, such as conference call bridging, auto-attendant, interactive voice response (IVR), overhead paging, directory listing, and many more.

Along with the Asterisk code, the only equipment needed to set up a small PBX is a PC with a Linux operating system, an analog or digital telephone, an inexpensive Digium TDM400P card with Foreign Exchange Station (FXS) or Foreign Exchange Office (FXO) modules, and an Internet connection. [Foreign Exchange allows the user to have a number that doesn't originate from a local office. FXS supplies a ringing voltage to telephone lines, and FXO sends and receives phone calls through a central office switch.]

The Digium TDM400P card, starting at $125, can be used to connect to a conventional phone or phone line. Each card can terminate up to four telephones or telephone lines, or can service even more when used in conjunction with an IP telephone. A PC can hold several TDM400P cards, one in each PCI slot.

This Is Not Your Father's PBX

While Asterisk started out as a Open Source Software implementation of a standard PBX, it has grown into much more.

For example, when you dial into Digium's main telephone number (877-LINUX-ME), you'll hear the default Asterisk IVR (in a surprisingly sultry voice) say...

Thank you for calling Digium -- your Open Source telecommunications supplier. If you know your party's extension, you may dial it at any time. Otherwise, press one for sales, two for technical support, three for customer service, four for accounting, nine for a company directory, or zero for an operator.

Dial an extension number to be transferred directly to the person you need to speak with. Or, press nine and enter the first three digits of your contact's last name to be connected to that person, even if he or she is in a different city or country! The IVR can say whatever you'd like it to say, and can also access a database. All of these features are set up with a simple Asterisk configuration file.

The entire Asterisk application is licensed under the GPL with special exceptions for OpenH323 and G.729 code. New features are implemented weekly (if not daily) as needed, and bugs are eliminated by a team of "Bug Marshals" as they appear on Digium's Bug Tracker web site (http://bugs.digium.com).

For individual users, Asterisk open source telephony lowers cost, frees customers from proprietary solutions, and eliminates upgrade costs. For service providers, Asterisk fulfills the needs of all kinds of businesses, and can be customized as much as needed.

« Asterisk 1.0 | Main | CID/ANI spoofing on VoIP using Asterisk »