http://directory.fsf.org/wiki?title=Asterisk_2&feed=atom&action=history 2013-05-25T16:11:16Z Revision history for this page on the wiki MediaWiki 1.20.2 http://directory.fsf.org/wiki?title=Asterisk_2&diff=311&oldid=prev 2011-04-12T12:22:29Z

<p>Created page with &quot;{{Entry |Name=Asterisk |Short description=Telephony system |Full description=Asterisk is a complete PBX in software. It does voice over IP in three protocols, and interoperates w...&quot;</p> <p><b>New page</b></p><div>{{Entry<br /> |Name=Asterisk<br /> |Short description=Telephony system<br /> |Full description=Asterisk is a complete PBX in software. It does voice over IP in three protocols, and interoperates with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, and Call Queueing. It also supports three-way calling, caller ID services, ADSI, MGCP, SIP and H.323 (as both client and gateway). The system needs no additional hardware for Voice over IP. Asterisk connects with digital and analog telephony equipment through a number of hardware devices, including that manufactured by Asterisk's sponsors, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks. An analog FXO card is available, and more analog interfaces are pending.<br /> |User level=none<br /> |Status=Vanished<br /> |Component programs=<br /> |Homepage URL=http://www.asterisk.org/<br /> |VCS checkout command=checkout zaptel zapata libpri asterisk gastman<br /> |Computer languages=C<br /> |Documentation note=http://www.digium.com/index.php?menu=documentation<br /> |Paid support=Technical support, consulting, and customization available from Digium, Inc. (http://www.digium.com/)<br /> |IRC help=irc://irc.freenode.net/asterisk<br /> |IRC general=<br /> |IRC development=irc://irc.freenode.net/asterisk<br /> |Related projects=ANT,Asterisk-oh323,Bayonne,Callweaver,Speak_Freely,email2fax,oMGCP,queXS,Yate-_Yet_Another_Telephony_Engine<br /> |Keywords=asterisk,pbx,voice over IP,telephony,telephone system<br /> |Is GNU=n<br /> |Last review by=Janet Casey<br /> |Last review date=2008-05-28<br /> |Submitted by=Database conversion<br /> |Submitted date=2011-04-01<br /> |Version identifier=1.2.0<br /> |Version date=2005-11-21<br /> |Version status=stable<br /> |Version download=ftp://ftp.asterisk.org/pub/telephony/asterisk/asterisk-1.2.0-tar.gz<br /> |License verified date=2003-02-14<br /> |Version comment=1.2.0 stable released 2005-11-21<br /> }}<br /> {{Person<br /> |Role=Maintainer<br /> |Real name=Mark Spencer<br /> |Email=markster@digium.com<br /> |Resource URL=<br /> }}<br /> {{Resource<br /> |Resource audience=Help<br /> |Resource kind=E-mail<br /> |Resource URL=mailto:asterisk-announce@lists.digium.com<br /> }}<br /> {{Resource<br /> |Resource audience=Developer<br /> |Resource kind=E-mail<br /> |Resource URL=mailto:asterisk-dev@lists.digium.com<br /> }}<br /> {{Resource<br /> |Resource audience=Support<br /> |Resource kind=E-mail<br /> |Resource URL=mailto:asterisk-users@lists.digium.com<br /> }}<br /> {{Software category<br /> |Business=productivity,telephony<br /> |Interface=command-line<br /> }}<br /> {{Project license<br /> |License=GPLv2orlater<br /> |License verified by=Janet Casey<br /> |License verified date=2003-02-14<br /> }}<br /> {{Software prerequisite<br /> |Prerequisite kind=Required to use<br /> |Prerequisite description=Linux kernel 2.4.x or later<br /> }}<br /> {{Software prerequisite<br /> |Prerequisite kind=Required to build<br /> |Prerequisite description=Linux Kernel Sources<br /> }}<br /> {{Software prerequisite<br /> |Prerequisite kind=Required to build<br /> |Prerequisite description=openssl<br /> }}<br /> {{Software prerequisite<br /> |Prerequisite kind=Weak prerequisite<br /> |Prerequisite description=libpri (for T1 or E1 interfaces)<br /> }}</div>

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