Extracted from asterisk-1.2.13~dfsg/debian/control:
===================================================

  asterisk - 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.

  asterisk-classic - Open Source Private Branch Exchange (PBX) - original Digium version
    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.

  asterisk-bristuff - Open Source Private Branch Exchange (PBX) - BRIstuff-enabled version
    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.

    The BRIstuff patch from www.junghanns.net enables support from certain
    hardware like HFC ISDN cards which are mainly consumer hardware.

    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.

  asterisk-doc - documentation for asterisk
    Asterisk is an Open Source PBX and telephony toolkit.

    This package contains the source documentation needed if you wish to
    extend the asterisk package.

  asterisk-dev - development files for asterisk
    Asterisk is an Open Source PBX and telephony toolkit.

    This package contains the include files used if you wish to compile a
    package which requires asterisk source file headers.

  asterisk-sounds-main - sound files for asterisk
    Asterisk is an Open Source PBX and telephony toolkit.

    This package contains the default sound files for operation of asterisk

  asterisk-web-vmail - Web-based (CGI) voice mail interface for Asterisk
    Asterisk is an Open Source PBX and telephony toolkit.

    This package contains a web-based interface for the voice-mail system

  asterisk-config - config files for asterisk
    Asterisk is an Open Source PBX and telephony toolkit.

    This package contains the default configuration files of Asterisk.


Extracted from asterisk-1.2.13~dfsg/debian/changelog:
=====================================================
  asterisk (1:1.2.13~dfsg-2.1) unstable; urgency=high
  
    * NMU
    * Backported to Sarge:
      - libreadline5 --> libreadline4
      - remove libopenh323-dev
      - Changed source:version in control to Source-Version
    * Added spa3000-504 patch.
  
   -- Russell Stuart <russell-debian@NOSPAM>  Mon, 18 Dec 2006 15:24:03 +1000
  
  asterisk (1:1.2.13~dfsg-2) unstable; urgency=low
  
    [ Tzafrir Cohen ]
    * less_docs.dpatch: remove unnecessary doxygen docs. asterisk-doc's size
      is now 1.6M (7.7M installed).
  
    [ Mark Purcell ]
    * asterisk-classic, asterisk-bristuff:
      /usr/lib/asterisk/modules/format_ogg_vorbis.so gone missing when
      rebuilt (Closes: #397147)
  
   -- Mark Purcell <msp@NOSPAM>  Mon,  6 Nov 2006 06:33:19 +0000


asterisk-1.2.13~dfsg/debian/copyright:
======================================

  This package was debianized by Mark Purcell <msp@NOSPAM> on
  Fri Apr 19 19:58:40 EST 2002
  
  Asterisk downloaded from http://www.asterisk.org
  Bristuff downloaded from http://www.junghanns.net/asterisk/
  
  Upstream Authors: Asterisk: Mark Spencer <markster@NOSPAM>
  		  Bristuff: Klaus-Peter Junghanns <kpj@NOSPAM> 
  
  Debian asterisk-xxx.dfsg.tar.gz
  ===============================
   * The Debian version of the upstream asterisk source has had the fpm Music
  on Hold removed as this music has only been licenced for use within 
  asterisk which is incompatible with the Debian Free Software Guildlines (DFSG)
  <http://www.debian.org/social_contract#guidelines>.
  
   * The iLBC codec library code has been removed from the Debian asterisk
  package as it does not conform with the DFSG.
  
   * The contrib/firmware/iax/iaxy.bin binary firmware without source has been removed.
  
  Copyright:
  
  Copyright (C) 2001-2004 Digium and others....
  
     This package is free software; you can redistribute it and/or modify
     it under the terms of the GNU General Public License as published by
     the Free Software Foundation; version 2 dated June, 1991.
  
     This package is distributed in the hope that it will be useful,
     but WITHOUT ANY WARRANTY; without even the implied warranty of
     MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
     GNU General Public License for more details.
  
     You should have received a copy of the GNU General Public License
     along with this package; if not, write to the Free Software
     Foundation, Inc., 51 Franklin St, Fifth Floor, Boston,
     MA 02110-1301, USA.
  
  On Debian GNU/Linux systems, the complete text of the GNU General
  Public License can be found in `/usr/share/common-licenses/GPL'.
  
      Asterisk is distributed under GNU General Public License.  The GPL also
    must apply to all loadable modules as well, except as defined below.
  
      Digium, Inc. (formerly Linux Support Services) retains copyright to all
    of the core Asterisk system, and therefore can grant, at its sole discretion,
    the ability for companies, individuals, or organizations to create proprietary
    or Open Source (but non-GPL'd) modules which may be dynamically linked at
    runtime with the portions of Asterisk which fall under our copyright
    umbrella, or are distributed under more flexible licenses than GPL
  
      If you wish to use our code in other GPL programs, don't worry -- there
    is no requirement that you provide the same exemption in your GPL'd
    products (although if you've written a module for Asterisk we would
    strongly encourage you to make the same excemption that we do).
  
      Specific permission is also granted to OpenSSL and OpenH323 to link to
    Asterisk.
  
      If you have any questions, whatsoever, regarding our licensing policy,
    please contact us.
  
      The 'Asterisk' name and logos are trademarks owned by Digium, Inc.,
     and use of them is subject to our trademark licensing policies. If you
     wish to use these trademarks for purposes other than simple
     redistribution of Asterisk source code obtained from Digium, you
     should contact our licensing department to determine the necessary
     steps you must take.
  
     If you have any questions regarding our licensing policy, please
     contact us:
  
     +1.877.546.8963 (via telephone in the USA)
     +1.256.428.6000 (via telephone outside the USA)
     +1.256.864.0464 (via FAX inside or outside the USA)
     IAX2/misery.digium.com/6000 (via IAX2)
     licensing@NOSPAM (via email)
  
     Digium, Inc.
     150 West Park Loop
     Suite 100
     Huntsville, AL  35806
     USA
  
  Other source code in Asterisk:
  
    GSM source:
  
      Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
      Technische Universitaet Berlin
  
      Any use of this software is permitted provided that this notice is not
      removed and that neither the authors nor the Technische Universitaet Berlin
      are deemed to have made any representations as to the suitability of this
      software for any purpose nor are held responsible for any defects of
      this software.  THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
  
      As a matter of courtesy, the authors request to be informed about uses
      this software has found, about bugs in this software, and about any
      improvements that may be of general interest.
  
      Berlin, 28.11.1994
      Jutta Degener
      Carsten Bormann
  
    ADPCM source:
  
      Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
      Netherlands.
  
                              All Rights Reserved
  
      Permission to use, copy, modify, and distribute this software and its
      documentation for any purpose and without fee is hereby granted,
      provided that the above copyright notice appear in all copies and that
      both that copyright notice and this permission notice appear in
      supporting documentation, and that the names of Stichting Mathematisch
      Centrum or CWI not be used in advertising or publicity pertaining to
      distribution of the software without specific, written prior permission.
  
      STICHTING MATHEMATISCH CENTRUM DISCLAIMS ALL WARRANTIES WITH REGARD TO
      THIS SOFTWARE, INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND
      FITNESS, IN NO EVENT SHALL STICHTING MATHEMATISCH CENTRUM BE LIABLE
      FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
      WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
      ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
      OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
  
   Bristuff:
  
      * Copyright (C) 2004, 2005 Junghanns.NET GmbH
      *
      * Klaus-Peter Junghanns <kpj@NOSPAM>
      *
      * This program is free software and may be modified and
      * distributed under the terms of the GNU Public License.
  
  Other portions of code:
  
  aescrypt.c: Copyright (c) 2003, Dr Brian Gladman <brg@NOSPAM>, Worcester, UK.
  aeskey.c: Copyright (c) 2003, Dr Brian Gladman <brg@NOSPAM>, Worcester, UK.
  dlfcn.c:Copyright (c) 2002 Jorge Acereda  <jacereda@NOSPAM> &
                     Peter O'Gorman <ogorman@NOSPAM>
  dns.c: * Copyright (C) 1999 - 2005 Thorsten Lockert
  dnsmgr.c: * Copyright (C) 2005, Kevin P. Fleming
  dsp.c:        Copyright (C) 2001  Steve Underwood <steveu@NOSPAM>
  indications.c: * Copyright (C) 2002, Pauline Middelink
  jitterbuf.c: * Copyright (C) 2004-2005, Horizon Wimba, Inc.
  jitterbuf.h: * Copyright (C) 2004-2005, Horizon Wimba, Inc.
  md5.c: * written by Colin Plumb in 1993, no copyright is claimed.
  plc.c: * Copyright (C) 2004 Steve Underwood
  poll.c:		Copyright (c) 1995-2002 Brian M. Clapper
  slinfactory.c: * Copyright (C) 2005, Anthony Minessale II.
  apps/app_alarmreceiver.c: * Copyright (C)  2004 - 2005 Steve Rodgers
  apps/app_chanspy.c: * Copyright (C) 2005 Anthony Minessale II (anthmct@NOSPAM)
  apps/app_controlplayback.c: * Copyright (C) 1999 - 2005, Digium, Inc.
  apps/app_curl.c: * Copyright (C)  2004 - 2005, Tilghman Lesher
  apps/app_cut.c: * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
  apps/app_db.c: * Copyright (C) 2003, Jefferson Noxon
  apps/app_dictate.c: * Copyright (C) 2005, Anthony Minessale II
  apps/app_directed_pickup.c: * Copyright (C) 2005, Joshua Colp
  apps/app_dumpchan.c: * Copyright (C) 2004 - 2005, Anthony Minessale II.
  apps/app_eval.c: * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
  apps/app_exec.c: * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
  apps/app_festival.c: * Copyright (C) 2002, Christos Ricudis
  apps/app_forkcdr.c: * Copyright (C) 1999 - 2005, Anthony Minessale anthmct@NOSPAM
  apps/app_hasnewvoicemail.c: * Changes Copyright (c) 2004 - 2005 Todd Freeman <freeman@NOSPAM>
  apps/app_hasnewvoicemail.c: * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
  apps/app_math.c: * Copyright (C) 2004 - 2005, Andy Powell 
  apps/app_md5.c: * Copyright (C) 2005, Olle E. Johansson, Edvina.net
  apps/app_mixmonitor.c: * Copyright (C) 2005, Anthony Minessale II
  apps/app_random.c: * Copyright (c) 2003 - 2005 Tilghman Lesher.  All rights reserved.
  apps/app_rpt.c: * Copyright (C) 2002-2005, Jim Dixon, WB6NIL
  apps/app_sayunixtime.c: * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
  apps/app_settransfercapability.c: * Copyright (C) 2005, Frank Sautter, levigo holding gmbh, www.levigo.de
  apps/app_skel.c: * Copyright (C) <Year>, <Your Name Here>
  apps/app_sms.c: * Copyright (C) 2004 - 2005, Adrian Kennard, rights assigned to Digium
  apps/app_sql_postgres.c: * Copyright (C) 2002, Christos Ricudis
  apps/app_verbose.c: * Copyright (c) 2004 - 2005 Tilghman Lesher.  All rights reserved.
  apps/app_while.c: * Copyright 2004 - 2005, Anthony Minessale <anthmct@NOSPAM>
  cdr/cdr_sqlite.c: * Copyright (C) 2004 - 2005, Holger Schurig
  cdr/cdr_tds.c:   Copyright (C) 1994, 1996, 1997, 2000, 2001 Free Software Foundation, Inc.
  channels/chan_misdn.c: * Copyright (C) 2004, Christian Richter
  channels/chan_misdn_config.c: * Copyright (C) 2005, Christian Richter
  channels/chan_modem_bestdata.c: * Copyright (C) 1999, Mark Spencer and 2001 Jim Dixon
  channels/chan_vpb.c: * Copyright (C) 2003, Paul Bagyenda
  channels/chan_vpb.c: * Copyright (C) 2004 - 2005, Ben Kramer
  channels/DialTone.h:  * Copyright (C) 1999, Mark Spencer
  channels/Makefile:# Copyright (C) 1999-2005, Mark Spencer
  codecs/codec_ilbc.c: * The iLBC code is from The IETF code base and is copyright The Internet Society (2004)
  doc/README.backtrace:Copyright 2004 Free Software Foundation, Inc.
  editline/: * Copyright (c) 1992, 1993
   *      The Regents of the University of California.  All rights reserved.
  editline/editline.3:.\" Copyright (c) 1997-1999 The NetBSD Foundation, Inc.
  editline/install-sh:# Copyright 1991 by the Massachusetts Institute of Technology
  formats/format_au.c: * Copyright (C) 2005, Andriy Pylypenko
  formats/format_g726.c: * Copyright (c) 2004 - 2005, inAccess Networks
  formats/format_ogg_vorbis.c: * Copyright (C) 2005, Jeff Ollie
  formats/format_sln.c: * Copyright (C) 1999 - 2005, Anthony Minessale
  funcs/func_cdr.c: * Portions Copyright (C) 2005, Anthony Minessale II
  funcs/func_db.c: * Copyright (C) 2005, Russell Bryant <russelb@NOSPAM> 
  funcs/func_db.c: * Copyright (C) 2003, Jefferson Noxon <jeff@NOSPAM>
  funcs/func_logic.c: * Portions Copyright (C) 2005, Anthony Minessale II
  funcs/func_math.c: * Copyright (C) 2004 - 2005, Andy Powell 
  funcs/func_md5.c: * Copyright (C) 2005, Olle E. Johansson, Edvina.net
  funcs/func_md5.c: * Copyright (C) 2005, Russell Bryant <russelb@NOSPAM> 
  funcs/func_strings.c: * Portions Copyright (C) 2005, Tilghman Lesher.  All rights reserved.
  funcs/func_strings.c: * Portions Copyright (C) 2005, Anthony Minessale II
  res/res_config_odbc.c: * Copyright (C) 2004 - 2005 Anthony Minessale II <anthmct@NOSPAM>
  res/res_indications.c: * Copyright (C) 2002, Pauline Middelink
  res/res_odbc.c: * Copyright (C) 2004 - 2005 Anthony Minessale II <anthmct@NOSPAM>
  utils/smsq.c: * Copyright (C) 2004 - 2005 Adrian Kennard
  utils/streamplayer.c:		fprintf(stderr, "Copyright (C) 2005 -- Russell Bryant -- Digium, Inc.\n\n");
  channels/misdn/: * Copyright (C) 2004, Christian Richter
  codecs/ilbc/:    Copyright (C) The Internet Society (2004). 
  contrib/scripts/managerproxy.pl:#  Copyright (c) 2004 David C. Troy &lt;dave@NOSPAM>
  contrib/scripts/retrieve_extensions_from_sql.pl:# Copyright:    2004 Peter Nixon <codemonkey@NOSPAM>
  db1-ast/*/*: * Copyright (c) 1990, 1993, 1994
   *      The Regents of the University of California.  All rights reserved.
  editline/np/: * Copyright (c) 1998 The NetBSD Foundation, Inc.
  editline/np/strlcat.c: * Copyright (c) 1998 Todd C. Miller <Todd.Miller@NOSPAM>
  include/asterisk/aes.h: Copyright (c) 2003, Dr Brian Gladman <brg@NOSPAM>, Worcester, UK.
  include/asterisk/dlfcn-compat.h:Copyright (c) 2002 Jorge Acereda  <jacereda@NOSPAM> &
                     Peter O'Gorman <ogorman@NOSPAM>
  include/asterisk/plc.h: * Copyright (C) 2004 Steve Underwood
  include/asterisk/poll-compat.h:		Copyright (c) 1995-2002 Brian M. Clapper
  include/asterisk/res_odbc.h: * Copyright (C) 2004 - 2005, Anthony Minessale II
  include/asterisk/slinfactory.h: * Copyright (C) 2005, Anthony Minessale II


asterisk-1.2.13~dfsg/debian/README.Debian:
==========================================

  Welcome to the world of telephone switch owners. Asterisk is a quite powerful
  and probably one of the feature-richest open-source PBXes out there. To get you
  a brief impression where to start tweaking, we've included this doc. (Thanks to
  Tzafrir Cohen for a proposal of this)
  
  File Locations
  ==============
  See /etc/asterisk/asterisk.conf for pathnames. All the configuration is done
  from the *.conf files there.
  
  Sounds reside at /usr/share/asterisk/sounds. Voicemail mailboxes are below
  /var/spool/asterisk/voicemail using symlinks.
  
  User, Groups, Permissions
  =========================
  By default the package will create user and group 'asterisk' and will run as
  them. It will optionally run with real-time priority (-p). It will also add
  itself to the group audio to enable the daemon to access sound cards. Running
  as root is disabled in /etc/init.d/asterisk for security reasons. If you use
  zaptel devices (like zaphfc too), you need to set permissions on /dev/zap to
  root.dialout or whatever other group you like and add the asterisk user to that
  group. Then make sure all the devicenodes below /dev/zap/ are 660 or asterisk
  will not have any access still. When no -G is given to the asterisk PARAMS,
  then asterisk will automatically initialize all groups of the asterisk user. 
  
  To change those values you can edit /etc/default/asterisk. You need to set
  RUNASTERISK=yes to make asterisk autostart upon boot.
  
  init.d Script Commands
  ======================
  In addition to reload, that runs the asterisk CLI command 'reload' there
  are also extensions-reload ('extensions reload') and logger-reload 
  ('logger-reload') to load only parts of the configuration file. 
  
  Asterisk supports a number of non-default paramters at startup.  For debugging
  it is often useful to run 'asterisk -U asterisk -vvvgc' from the console.  Add
  more -v for even higher verbosity. You can attach a remote console to a running
  asterisk daemon with 'asterisk -vcr'.  When testing, make sure to use the "-U
  asterisk", or asterisk may write files with root ownership. In that case you
  might not be able to restart the daemon without altering permissions to the
  files first.
  
  
  Getting started with config and more features
  =============================================
  Of course the default config is far from a plug'n'play for every environment.
  You need to setup a proper dialplan in extensions.conf, remove the demo entries
  and maybe remove FreeWorldDialup too in case you don't participate there. For
  more info on how to setup a proper dialplan, fax, enum and configuration of
  asterisk, please checkout
  	http://www.voip-info.org/wiki-Asterisk+quickstart
  	http://www.voip-info.org/wiki-Asterisk+config+files
  or the larger HowTos at:
  	http://www.automated.it/guidetoasterisk.htm
  or
  	http://www.asteriskdocs.org/ (Asterisk Handbook)
  
  
  Enjoy your PBX!
  
  Kilian Krause (for the pkg-voip team)


asterisk-1.2.13~dfsg/README:
============================

  The Asterisk Open Source PBX
  by Mark Spencer <markster@NOSPAM>
  and the Asterisk.org developer community
  
  Copyright (C) 2001-2005 Digium, Inc.
  and other copyright holders.
  ================================================================
  
  * SECURITY
    It is imperative that you read and fully understand the contents of
  the SECURITY file before you attempt to configure and run an Asterisk
  server.
  
  * WHAT IS ASTERISK ?
    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.  For more information
  on the project itself, please visit the Asterisk home page at:
  
             http://www.asterisk.org
  
  In addition you'll find lots of information compiled by the Asterisk
  community on this Wiki:
  
             http://www.voip-info.org/wiki-Asterisk
  
  There is a book on Asterisk published by O'Reilly under the
  Creative Commons License. It is available in book stores as well
  as in a downloadable version on the http://www.asteriskdocs.org
  web site.
  
  * SUPPORTED OPERATING SYSTEMS
  
  == Linux ==
    The Asterisk Open Source PBX is developed and tested primarily on the
  GNU/Linux operating system, and is supported on every major GNU/Linux
  distribution.
  
  == Others ==
    Asterisk has also been 'ported' and reportedly runs properly on other
  operating systems as well, including Sun Solaris, Apple's Mac OS X, and
  the BSD variants.
  
  * GETTING STARTED
  
    First, be sure you've got supported hardware (but note that you don't need
  ANY special hardware, not even a soundcard) to install and run Asterisk.
  
    Supported telephony hardware includes:
  
  	* All Wildcard (tm) products from Digium (www.digium.com)
  	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
  	* any full duplex sound card supported by ALSA or OSS
          * VoiceTronix OpenLine products
  
  The are several drivers for ISDN BRI cards available from third party sources.
  Check the voip-info.org wiki for more information on chan_capi, chan_misdn and 
  zaphfc.
  
  * UPGRADING FROM VERSION 1.0
  
    If you are updating from a previous version of Asterisk, make sure you
  read the UPGRADE.txt file in the source directory. There are some files
  and configuration options that you will have to change, even though we
  made every effort possible to maintain backwards compatibility.
  
    In order to discover new features to use, please check the configuration
  examples in the /configs directory of the source code distribution. 
  To discover the major new features of Asterisk 1.2, please visit 
  http://edvina.net/asterisk1-2/
  
  * NEW INSTALLATIONS
  
    Ensure that your system contains a compatible compiler and development
  libraries.  Asterisk requires either the GNU Compiler Collection (GCC) version
  3.0 or higher, or a compiler that supports the C99 specification and some of
  the gcc language extensions.  In addition, your system needs to have the C
  library headers available, and the headers and libraries for OpenSSL,
  ncurses and zlib.
  On many distributions, these files are installed by packages with names like
  'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
  
    So let's proceed:
  
  1) Run "make"
  
    Assuming the build completes successfully:
  
  2) Run "make install"
  
    Each time you update or checkout from CVS, you are strongly encouraged 
  to ensure all previous object files are removed to avoid internal 
  inconsistency in Asterisk. Normally, this is automatically done with 
  the presence of the file .cleancount, which increments each time a 'make clean'
  is required, and the file .lastclean, which contains the last .cleancount used. 
  
    If this is your first time working with Asterisk, you may wish to install
  the sample PBX, with demonstration extensions, etc.  If so, run:
  
  3) "make samples"
  
    Doing so will overwrite any existing config files you have.
  
    Finally, you can launch Asterisk in the foreground mode (not a daemon)
  with:
  
  # asterisk -vvvc
  
    You'll see a bunch of verbose messages fly by your screen as Asterisk
  initializes (that's the "very very verbose" mode).  When it's ready, if
  you specified the "c" then you'll get a command line console, that looks
  like this:
  
  *CLI>
  
    You can type "help" at any time to get help with the system.  For help
  with a specific command, type "help <command>".  To start the PBX using
  your sound card, you can type "dial" to dial the PBX.  Then you can use
  "answer", "hangup", and "dial" to simulate the actions of a telephone.
  Remember that if you don't have a full duplex sound card (and Asterisk
  will tell you somewhere in its verbose messages if you do/don't) then it
  won't work right (not yet).
  
    "man asterisk" at the Unix/Linux command prompt will give you detailed
  information on how to start and stop Asterisk, as well as all the command
  line options for starting Asterisk.
  
    Feel free to look over the configuration files in /etc/asterisk, where
  you'll find a lot of information about what you can do with Asterisk.
  
  * ABOUT CONFIGURATION FILES
  
    All Asterisk configuration files share a common format.  Comments are
  delimited by ';' (since '#' of course, being a DTMF digit, may occur in
  many places).  A configuration file is divided into sections whose names
  appear in []'s.  Each section typically contains two types of statements,
  those of the form 'variable = value', and those of the form 'object =>
  parameters'.  Internally the use of '=' and '=>' is exactly the same, so 
  they're used only to help make the configuration file easier to
  understand, and do not affect how it is actually parsed.
  
    Entries of the form 'variable=value' set the value of some parameter in
  asterisk.  For example, in zapata.conf, one might specify:
  
  	switchtype=national
  
  in order to indicate to Asterisk that the switch they are connecting to is
  of the type "national".  In general, the parameter will apply to
  instantiations which occur below its specification.  For example, if the
  configuration file read:
  
  	switchtype = national
  	channel => 1-4
  	channel => 10-12
  	switchtype = dms100
  	channel => 25-47
  
  the "national" switchtype would be applied to channels one through
  four and channels 10 through 12, whereas the "dms100" switchtype would
  apply to channels 25 through 47.
    
    The "object => parameters" instantiates an object with the given
  parameters.  For example, the line "channel => 25-47" creates objects for
  the channels 25 through 47 of the card, obtaining the settings
  from the variables specified above.
  
  * SPECIAL NOTE ON TIME
    
    Those using SIP phones should be aware that Asterisk is sensitive to
  large jumps in time.  Manually changing the system time using date(1)
  (or other similar commands) may cause SIP registrations and other
  internal processes to fail.  If your system cannot keep accurate time
  by itself use NTP (http://www.ntp.org/) to keep the system clock
  synchronized to "real time".  NTP is designed to keep the system clock
  synchronized by speeding up or slowing down the system clock until it
  is synchronized to "real time" rather than by jumping the time and
  causing discontinuities. Most Linux distributions include precompiled
  versions of NTP.  Beware of some time synchronization methods that get
  the correct real time periodically and then manually set the system
  clock.
  
    Apparent time changes due to daylight savings time are just that,
  apparent.  The use of daylight savings time in a Linux system is
  purely a user interface issue and does not affect the operation of the
  Linux kernel or Asterisk.  The system clock on Linux kernels operates
  on UTC.  UTC does not use daylight savings time.
  
    Also note that this issue is separate from the clocking of TDM
  channels, and is known to at least affect SIP registrations.
  
  * FILE DESCRIPTORS
  
    Depending on the size of your system and your configuration,
  Asterisk can consume a large number of file descriptors.  In UNIX,
  file descriptors are used for more than just files on disk.  File
  descriptors are also used for handling network communication
  (e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and
  digital trunk hardware).  Asterisk accesses many on-disk files for
  everything from configuration information to voicemail storage.
  
    Most systems limit the number of file descriptors that Asterisk can
  have open at one time.  This can limit the number of simultaneous
  calls that your system can handle.  For example, if the limit is set
  at 1024 (a common default value) Asterisk can handle approxiately 150
  SIP calls simultaneously.  To change the number of file descriptors
  follow the instructions for your system below:
  
  == PAM-based Linux System ==
  
    If your system uses PAM (Pluggable Authentication Modules) edit
  /etc/security/limits.conf.  Add these lines to the bottom of the file:
  
  root            soft    nofile          4096
  root            hard    nofile          8196
  asterisk        soft    nofile          4096
  asterisk        hard    nofile          8196
  
  (adjust the numbers to taste).  You may need to reboot the system for
  these changes to take effect.
  
  == Generic UNIX System ==
  
    If there are no instructions specifically adapted to your system
  above you can try adding the command "ulimit -n 8192" to the script
  that starts Asterisk.
  
  * MORE INFORMATION
  
    See the doc directory for more documentation on various features. Again,
  please read all the configuration samples that include documentation on
  the configuration options.
  
    Finally, you may wish to visit the web site and join the mailing list if
  you're interested in getting more information.
  
     http://www.asterisk.org/support
  
    Welcome to the growing worldwide community of Asterisk users!
  
  Mark Spencer
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