Asterisk is an open source/free software implementation of a telephone private
branch exchange (PBX) originally created by Mark Spencer of Digium. Like any
PBX, it allows a number of attached telephones to make calls to one another,
and to connect to other telephone services including the public switched
telephone network (PSTN). "Its name comes from the asterisk symbol, *, which in
Unix (and Unix-like operating systems such as Linux) and DOS environments
represents a wildcard, matching any sequence of characters in a filename."

Asterisk is released under a dual license scheme, the free software license
being the GNU General Public License (GPL), the other being a proprietary
software license as to allow proprietary/closed and patented code, such as the
G.729 codec to work with the system (although the G729 codec may work with
the free or proprietary versions).

However, due to free software/open source nature of the software, hundreds of
other programmers have contributed features and functionality and have
reported bugs. Originally designed for the Linux operating system, Asterisk now
also runs on NetBSD, OpenBSD, FreeBSD, Mac OS X, and Solaris, although as
the "native" platform, Linux is the most supported of these.

The basic
Asterisk software includes many features available in proprietary PBX
systems: voice mail, conference calling, interactive voice response (phone
menus), and automatic call distribution. Users can create new functionality by
writing dial plan scripts in Asterisk's own language, by adding custom modules
written in C, or by writing Asterisk Gateway Interface scripts in Perl or other
languages.





















To attach ordinary telephones to a Linux server running Asterisk, or to connect
to PSTN trunk lines, the server must be fitted with special hardware. (An
ordinary modem will not suffice.) Digium and a number of other firms sell PCI
cards to attach telephones, telephone lines, T1 and E1 lines, and other analog
and digital phone services to a server.

Perhaps of more interest to many deployers today, Asterisk also supports a wide
range of Voice over IP protocols, including SIP, MGCP and H.323. Asterisk can
interoperate with most SIP telephones, acting both as registrar and as a
gateway between IP phones and the PSTN. Asterisk developers have also
designed a new protocol, Inter-Asterisk eXchange, for efficient trunking of calls
among Asterisk PBXes.

By supporting a mix of traditional and VoIP telephony services, Asterisk allows
deployers to build new telephone systems, or gradually migrate existing systems
to new technologies. Some sites are using Asterisk servers to replace
proprietary PBXes; others to provide additional features (such as voice mail or
phone menus) or to cut costs by carrying long-distance calls over the Internet
(toll bypass).

VoIP telephone companies have begun to support Asterisk; many now offer IAX2
or SIP trunking direct to an Asterisk box as an alternative to providing the
customer with an ATA.

Asterisk was rated top in the PBX category out of 74 open source VoIP
resources. As of August 23, 2007, the current release version of Asterisk is
1.4.11.

Attributes and Credits
The information and facts supplied on this subject
derive from
http://en.wikipedia.org/wiki/Main_Page
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