[ Bron: asterisk ]
Pakket: asterisk (1:1.2.16~dfsg-1ubuntu3.1) [security] [universe]
Verwijzigingen voor asterisk
Ubuntu bronnen:
Het bronpakket asterisk downloaden:
- [asterisk_1.2.16~dfsg-1ubuntu3.1.dsc]
- [asterisk_1.2.16~dfsg.orig.tar.gz]
- [asterisk_1.2.16~dfsg-1ubuntu3.1.diff.gz]
Beheerder:
Please consider filing a bug or asking a question via Launchpad before contacting the maintainer directly.
Original Maintainers (usually from Debian):
- Debian VoIP Team (Mailarchief)
- Mark Purcell
- Kilian Krause
- Jose Carlos Garcia Sogo
- Santiago Garcia Mantinan
- Simon Richter
- Tzafrir Cohen
It should generally not be necessary for users to contact the original maintainer.
Vergelijkbare pakketten:
Open Source Private Branch Exchange (PBX)
Asterisk is an Open Source PBX and telephony toolkit. It is, in a sense, middleware between Internet and telephony channels on the bottom, and Internet and telephony applications at the top.
Asterisk can be used with Voice over IP (SIP, H.323, IAX) standards, or the Public Switched Telephone Network (PSTN) through Supported Hardware.
Supported hardware:
* All Wildcard (tm) products from Digium (http://www.digium.com) * QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net) * Full Duplex Sound Card supported by Linux * Adtran Atlas 800 Plus * ISDN4Linux compatible ISDN card * Tormenta Dual T1 card (http://www.bsdtelephony.com.mx) * CAPI compatible ISDN cards can be run using the add-on package chan-capi
This Debian package includes the sample configuration, with demonstration extensions, etc
Website: http://www.asterisk.org.
Andere aan asterisk gerelateerde pakketten
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- dep: adduser (>= 3.63)
- Add and remove users and groups
-
- dep: asterisk-classic (>= 1:1.2.16~dfsg-1ubuntu3.1)
- Open Source Private Branch Exchange (PBX) - original Digium version
- of asterisk-bristuff (>= 1:1.2.16~dfsg-1ubuntu3.1)
- Open Source Private Branch Exchange (PBX) - BRIstuff-enabled version
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- sug: asterisk-dev
- development files for asterisk
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- sug: asterisk-doc
- documentation for asterisk
-
- sug: asterisk-rate-engine
- Asterisk least cost routing module
-
- sug: ekiga
- H.323 and SIP compatible VOIP client
-
- sug: gnomemeeting
- Package not available
-
- sug: kphone
- Voice over IP (VoIP) phone application
-
- sug: ohphone
- Command line H.323 client with X, SVGA and SDL support