- Adding a doc/00README.1st with an INDEX over README files

- Moving files from / to /doc or /configs
- Renaming some documentation files

Thank you for the initiative, manxpower!


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9046 f38db490-d61c-443f-a65b-d21fe96a405b

8 files changed, 213 insertions, 0 deletions

diff --git a/doc/00README.1st b/doc/00README.1st
new file mode 100644
index 000000000..a530922e6
--- /dev/null
+++ b/doc/00README.1st
@@ -0,0 +1,68 @@
+Files in the /doc directory:
+----------------------------
+In addition to these files, there is a lot of documentation of various
+configuration options in the sample configuration files, in the /configs
+directory of your source code
+
+Start here
+----------
+README.security		IMPORTANT INFORMATION ABOUT ASTERISK SECURITY
+README.hardware		Hardware supported by Asterisk
+
+Configuration
+-------------
+README.configuration	Features in the configuration parser
+README.extensions	Basics about the dialplan
+README.extconfig	How to use databases for configuration of Asterisk (ARA)
+README.realtime		The Asterisk Realtime Architecture - database support
+README.tds		Information about the FreeTDS support
+README.ael		Information about the Asterisk Extension Language
+
+Misc
+----
+PEERING			The General Peering Agreement for Dundi
+README.app_sms		How to configure the SMS application
+README.asterisk.conf	Documentation of various options in asterisk.conf
+README.callingpres	Settings for Caller ID presentation
+README.cdr		Call Data Record information
+README.cliprompt	How to change the Asterisk CLI prompt
+README.dundi		Dundi - a discovery protocol
+README.enum		Enum support in Asterisk
+README.ices		Integrating ICEcast streaming in Asterisk
+README.jitterbuffer	About the IAX2 jitterbuffer implementation
+README.math		About the math() application
+README.mp3		About MP3 support in Asterisk
+README.musiconhold-fpm	Free Music On Hold music
+README.mysql		About MYSQL support in Asterisk
+README.odbcstorage	Voicemail storage of messages in UnixODBC
+README.privacy		Privacy enhancements in Asterisk
+README.queuelog		Agent and queue logging
+README.variables	Channel variables
+cdr.txt			About CDR storage in various databases (needs update)
+
+Channel drivers
+---------------
+README.misdn
+README.h323		How to compile and configure the H.323 channel
+README.iax		About the IAX2 protocol support in Asterisk
+README.localchannel	The local channel is a "gosub" in the dialplan
+
+Portability
+-----------
+README.cygwin		Compiling Asterisk on CygWin platforms (Windows)
+
+For developers
+--------------
+See http://www.asterisk.org/developers for more information
+
+README.manager		About the AMI - Asterisk Manager Interface
+			for third party call control and PBX management
+README.backtrace	How to produce a backtrace when Asterisk crashes
+CODING-GUIDELINES	Guidelines for developers
+README.channels		What is a channel?
+README.externalivr	Documentation of the protocol used in externalivr()
+README.linkedlists	How to develop linked lists in Asterisk (old)
+iax.txt			About the IAX protocol
+apps.txt		About application development
+model.txt		About the call model in Asterisk (old)
+modules.txt		How Asterisk modules work
diff --git a/doc/extensions.txt b/doc/README.extensions
index bab08d319..bab08d319 100644
--- a/doc/extensions.txt
+++ b/doc/README.extensions
diff --git a/doc/README.hardware b/doc/README.hardware
new file mode 100644
index 000000000..86b28bef7
--- /dev/null
+++ b/doc/README.hardware
@@ -0,0 +1,70 @@
+A PBX is only really useful if you can get calls into it.  Of course, you
+can use Asterisk with VoIP calls (SIP, H.323, IAX), but you can also talk
+to the real PSTN through various cards.
+
+Supported Hardware is divided into two general groups:  Zaptel devices and 
+non-zaptel devices.  The Zaptel compatible hardware supports pseudo-TDM 
+conferencing and all call features through chan_zap, whereas non-zaptel 
+compatible hardware may have different features.
+
+Zaptel compatible hardware
+==========================
+
+-- Digium (Primary author of Asterisk)
+        http://www.digium.com, http://store.yahoo.com/asteriskpbx
+
+   * Wildcard X100P - Single FXO interface connects to Loopstart phone 
+     line
+
+   * Wildcard T400P (obsolete) - Quad T1 interface connects to four T1/PRI 
+     interfaces.  Supports RBS and PRI voice and PPP, FR, and HDLC data.
+
+   * Wildcard E400P (obsolete)- Quad E1 interface connects to four E1/PRI 
+     (or PRA) interfaces.  Supports PRA/PRI, EuroISDN voice and data.
+
+   * Wildcard T100P - Single T1 interface connects to a single T1/PRI
+     interface.  Supports RBS and PRI voice and PPP, FR, and HDLC data.
+
+   * Wildcard E100P - Single E1 interface connects to a single E1/PRI (or PRA)
+     interface.  Supports PRA/PRI, EuroISDN voice and PPP, FR, HDLC data.
+
+   * Wildcard S100U - Single FXS interface connects to a standard analog
+     telephone.
+
+   * Wildcard TDM400P - Quad Modular FXS interface connects to standard
+     analog telephones.
+
+   * Wildcard TE410P - Quad T1/E1 switchable interface.  Supports PRI and 
+     RBS signalling, as well as PPP, FR, and HDLC data modes.
+
+Non-zaptel compatible hardware
+==============================
+
+-- QuickNet, Inc. 
+       http://www.quicknet.net
+
+   * Internet PhoneJack - Single FXS interface.  Supports Linux telephony
+     interface.  DSP compression built-in.
+
+   * Internet LineJack - Single FXS or FXO interface.  Supports Linux 
+     telephony interface.
+
+
+Miscellaneous other interfaces
+==============================
+
+-- ISDN4Linux
+       http://www.isdn4linux.de/
+
+   * Any ISDN terminal adapter supported by isdn4linux should provide
+     connectivity.
+
+-- ALSA
+       http://www.alsa-project.org
+
+   * Any ALSA compatible full-duplex sound card
+
+-- OSS
+       http://www.opensound.com
+
+   * Any OSS compatible full-duplex sound card
diff --git a/doc/localchannel.txt b/doc/README.localchannel
index f96ea15ec..f96ea15ec 100644
--- a/doc/localchannel.txt
+++ b/doc/README.localchannel
diff --git a/doc/manager.txt b/doc/README.manager
index 065d70a21..065d70a21 100644
--- a/doc/manager.txt
+++ b/doc/README.manager
diff --git a/doc/README.musiconhold-fpm b/doc/README.musiconhold-fpm
new file mode 100644
index 000000000..ad11c4815
--- /dev/null
+++ b/doc/README.musiconhold-fpm
@@ -0,0 +1,8 @@
+About Hold Music
+================
+Digium has licensed the music included with
+the Asterisk distribution From FreePlayMusic
+for use and distribution with Asterisk.  It
+is licensed ONLY for use as hold music within
+an Asterisk based PBX.
+
diff --git a/doc/queuelog.txt b/doc/README.queuelog
index 374b7a488..374b7a488 100644
--- a/doc/queuelog.txt
+++ b/doc/README.queuelog
diff --git a/doc/README.security b/doc/README.security
new file mode 100644
index 000000000..3290cba48
--- /dev/null
+++ b/doc/README.security
@@ -0,0 +1,67 @@
+==== Security Notes with Asterisk ====
+
+PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED INFORMATION.  
+IMPROPER CONFIGURATION OF ASTERISK COULD ALLOW UNAUTHORIZED USE OF YOUR
+FACILITIES, POTENTIALLY INCURRING SUBSTANTIAL CHARGES.
+
+Asterisk security involves both network security (encryption, authentication)
+as well as dialplan security (authorization - who can access services in 
+your pbx). If you are setting up Asterisk in production use, please make
+sure you understand the issues involved.
+
+* NETWORK SECURITY
+
+If you install Asterisk and use the "make samples" command to install
+a demonstration configuration, Asterisk will open a few ports for accepting
+VoIP calls. Check the channel configuration files for the ports and IP addresses.
+
+If you enable the manager interface in manager.conf, please make sure that
+you access manager in a safe environment or protect it with SSH or other
+VPN solutions.
+
+For all TCP/IP connections in Asterisk, you can set ACL lists that
+will permit or deny network access to Asterisk services. Please check
+the "permit" and "deny" configuration options in manager.conf and
+the VoIP channel configurations - i.e. sip.conf and iax.conf.
+
+The IAX2 protocol supports strong RSA key authentication as well as
+AES encryption of voice and signalling. The SIP channel does not
+support encryption in this version of Asterisk.
+
+* DIALPLAN SECURITY
+
+First and foremost remember this:
+
+USE THE EXTENSION CONTEXTS TO ISOLATE OUTGOING OR TOLL SERVICES FROM ANY
+INCOMING CONNECTIONS.
+
+You should consider that if any channel, incoming line, etc can enter an
+extension context that it has the capability of accessing any extension
+within that context. 
+
+Therefore, you should NOT allow access to outgoing or toll services in
+contexts that are accessible (especially without a password) from incoming
+channels, be they IAX channels, FX or other trunks, or even untrusted
+stations within you network.  In particular, never ever put outgoing toll
+services in the "default" context.  To make things easier, you can include
+the "default" context within other private contexts by using:
+
+	include => default
+
+in the appropriate section.  A well designed PBX might look like this:
+
+[longdistance]
+exten => _91NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
+include => local
+
+[local]
+exten => _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
+include => default
+
+[default]
+exten => 6123,Dial(Zap/1)
+
+
+DON'T FORGET TO TAKE THE DEMO CONTEXT OUT OF YOUR DEFAULT CONTEXT.  There
+isn't really a security reason, it just will keep people from wanting to 
+play with your Asterisk setup remotely.