01.11.2024

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

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__NOTOC__
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{{Infobox Software
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<!--leva kolona start -->
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| name = Asterisk
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<div class="bodymaincoks" style="float:left;">
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| logo = [[Image:Asterisk-logo.png|150px]]
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<div>
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| developer = [[Digium]]
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| latest release version = 1.6
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| latest release date =
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[[image:Asterisk-logo.png|center|Asterisk]]<br>
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| latest preview version =
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'''Nagios sistem za nadzor naprav, strežnikov in storitev, ki omogoča proaktivno spremljanje in ukrepanje ob izpadih ali zmanjšanih odzivih sistemskih storitev.'''
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| latest preview date =
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| programming language = [[C (programming language)|C]]
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Nadzorni daemon v stalnih časovnih razmikih preverja delovanje naprav in storitev, ki omogočajo dostop preko standardnih ali nestandardnih vmesnikov ter posreduje informacije nadzornemu sistemu.
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| operating system = [[Unix-like]]
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| genre = VoIP software
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Ob odstopanju od pričakovanega stanja lahko sistem obvešča nadzornike na različne načine - e-pošta, SMS ... Trenutno stanje sistema in poročila so na voljo tudi preko spletnega vmesnika.
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| license = [[GNU General Public License]]/[[Proprietary software|Proprietary]]
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| website = [https://www.asterisk.org/ www.asterisk.org]
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'''Nadzor nad delovanjem ključnih elementov infrastrukture:'''
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*strežnikov; obremenjenost, prosti viri, odzivnost, razpoložljivost
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*posameznih storitev v omrežu; poštna storitev, spletni strežnik, SQL baza, SOA storitve, Web services
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* delovnih postaj; ključni parametri delovanja
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* omrežnih komponent - stikal, preklopnikov; obremenjenost, odzivnost
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*lokalnega omrežja in WAN povezav; obremenjenost, odzivnost, razpoložljivost
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<div class="coksnovice">
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<!--lokalna pomoč start -->
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=='''LOKALNA POMOČ'''==
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'''[https://www.agenda.si/ Agenda d.o.o.]'''
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----
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[mailto:uredniki@coks.si Prijavite] ponudbo lokalne pomoči
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</div>
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<div class="coksnovice">
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== Novice==
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<rss>https://www.nagios.org/backend/rss/newsitems.php|charset=UTF-8|short|max=5</rss>
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</div>
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<div class="uporabacoks">
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== Kako uporabljam Asterisk? ==
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----
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[mailto:uredniki@coks.si Prispevajte] svoje izkušnje o uporabi OK rešitve.
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==Študije primerov uporabe==
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* [https://www.nagioscommunity.org/ Uporaba Nagios-a (angleški)]
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</div>
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</div>
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<div class="rightboxesout">
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<div class="coksboxsw">
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{{infobox programi
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| image = [[image:Asterisk-logo.png|center|150px]]
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| name = Asterisk
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| kropis = Telefonska centrala (PBX).
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| izdaja = 1.4.17
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| razvijalec = [https://www.digium.com/ Digium]
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| os = [[Linux]]/[[Unix]]
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| stack = [[Unix]]
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| licenca = [[GPL]]
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| spletnastran = [https://www.asterisk.org www.asterisk.org]
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| lokalizacija = Ne
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}}
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</div>
 
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<div class="coksboxsw">
 
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==BREZPLAČNO SVETOVANJE==
 
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E-pošta:
 
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Forum:
 
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Tel.
 
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</div>
 
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<div class="coksdownloads">
 
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==Download==
 
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Zadnja stabilna različica: '''1.4.17'''
 
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Datum izida: '''02.01.2008'''
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'''Asterisk''' is an [[open-source software|open source]]/[[free software]] implementation of a telephone [[private branch exchange]] (PBX) originally created by [[Mark Spencer]] of [[Digium]].
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===Izvršne različice programa ===
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Like any PBX, it allows a number of attached [[telephone]]s 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 system]]s such as [[Linux]]) and [[DOS]] environments represents a wildcard, matching any sequence of characters in a filename."
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* [http://www.asterisk.org/downloads Asterisk prenosi]
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</div>
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<div class="coksdokumentacija">
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==Dokumentacija==
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==Features==
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* [http://www.asterisk.org/support Osnovna stran dokumentacije (angleška)]
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The basic Asterisk software includes many features available in proprietary PBX systems: [[voice mail]], [[conference call]]ing, [[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 (programming language)|C]], or by writing ''Asterisk Gateway Interface'' (AGI) scripts in [[Perl]] or other languages.
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</div>
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<div class="cokshelp">
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==Pomoč==
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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 [[Peripheral Component Interconnect|PCI]] cards to attach telephones,
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* [https://www.nagios.org/support/ Nagios pomoč]
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telephone lines, [[T-carrier|T1]] and [[E-carrier|E1]] lines, and other analog and digital phone services to a server.
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</div>
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<div class="cokslinks">
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==Hitre povezave==
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Perhaps of more interest to many deployers today, Asterisk also supports a wide range of [[Voice over IP]] protocols, including [[Session Initiation Protocol|SIP]], [[Media Gateway Control Protocol|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).
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* [https://www.nagios.org/products/ Nagios produkti]
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* [https://www.nagios.org/development/ Razvoj Nagios-a]
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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]]).
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===Povezave RSS===
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[[Category:Abecedni seznam programov]]
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* [http://sourceforge.net/export/rss2_project.php?group_id=26589 Nagios novice]
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[[Category:Telefonija]]
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</div>
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[[Category:Internetna telefonija]]
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<!--desni meni end -->
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</div>
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Current revision

Asterisk
Developer Digium
Latest release 1.6
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).

Retrieved from "https://en.coks.si/index.php5/Asterisk"