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Abecedni seznam programov
Source Forge > Komunikacije > Internetna telefonija
Source Forge > Komunikacije > Telefonija

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{{Infobox Software
{{Infobox Software
| name = Asterisk
| name = Asterisk
-
| logo = [[Image:Asterisk-logo.png|right|150px]]
+
| logo = [[Image:Asterisk-logo.png|150px]]
| developer = [[Digium]]
| developer = [[Digium]]
| latest release version = 1.4.17
| latest release version = 1.4.17

Revision as of 14:23, 31 January 2008

Asterisk
Developer Digium
Latest release 1.4.17 / January 2, 2008
Preview release 1.6.0 Beta 1 / January 18, 2008
Programming language(s) C
OS Unix-like
Genre VoIP software
License GNU General Public License/Proprietary
Website www.asterisk.org


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."

Features

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 (AGI) 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. 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 (IAX2), for efficient trunking of calls among Asterisk PBXes, and to VoIP service providers who support it. Some telephones support the IAX2 protocol directly for communicating with an Asterisk server (see Comparison of VoIP software for examples).

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).