[Asterisk-doc] docs introduction.xml,1.4,1.5
blitzrage
asterisk-doc@lists.digium.com
Mon, 31 May 2004 02:05:42 +0000
Comments:
Update of /cvsroot/asterisk/docs
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv23188/docs
Modified Files:
introduction.xml
Log Message:
blitzrage
- Asterisk is not a turnkey system
- Don't like it? Change it yourself
- Codecs and Conversions
- Asterisk software: Asterisk, Zaptel, Libpri
- Zaptel Hardware: Overview
Mike Preston
- The Big Picture
- Channels
- Codecs and Conversions
- Protocols
Index: introduction.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/introduction.xml,v
retrieving revision 1.4
retrieving revision 1.5
diff -C2 -d -r1.4 -r1.5
*** introduction.xml 7 May 2004 01:37:42 -0000 1.4
--- introduction.xml 31 May 2004 02:05:36 -0000 1.5
***************
*** 41,49 ****
<sect3>
<title>Asterisk is not a turnkey system</title>
! <para/>
</sect3>
<sect3>
<title>Don't like it? Change it yourself!</title>
! <para/>
</sect3>
<sect3>
--- 41,72 ----
<sect3>
<title>Asterisk is not a turnkey system</title>
! <para>
! The Asterisk PBX system is a complex peice of software. The learning curve
! is very steep and simply reading any single resource will not teach you
! everything that Asterisk is capable of. This resource is an attempt to gather
! some of the most common issues that new comers to Asterisk encounter.
! Learning how Asterisk works is very much like learning a new programming
! language. Many hours need to be spent with Asterisk in order to understand
! how all the configuration files work with each other to control the many
! interfaces. The ability to understand the dialplan is a key concept
! that those new to Asterisk need to fully comprehend. Once this is established
! the configuration of the many different kinds of interfaces that you would
! like Asterisk to communicate with work in tandem with the dialplan. This
! relationship extends througout Asterisk with many other modules that are
! not compiled by default. The term KISS (Keep It Super Simple) needs to be
! applied here with great emphasis. The mistake many people make when first
! discovering Asterisk is that they think they can have a working system
! in a couple of hours. This may be possible once all the concepts are learned,
! but few are able to do it their first time out.
! </para>
</sect3>
<sect3>
<title>Don't like it? Change it yourself!</title>
! <para>
! Asterisk is an open piece of software. The ability to read the source code
! is its power. Most (if not all) other PBX's are entirely closed source
! with only the abilities that have been provided to you. If something doesn't
! work quite the way you would expect it to, you are able to change it.
! </para>
</sect3>
<sect3>
***************
*** 57,73 ****
<sect2>
<title>The Big Picture</title>
! <para/>
</sect2>
<sect2>
<title>Channels</title>
! <para/>
</sect2>
<sect2>
<title>Codecs and Conversions</title>
! <para/>
</sect2>
<sect2>
! <title>Etc.</title>
! <para/>
</sect2>
</sect1>
--- 80,129 ----
<sect2>
<title>The Big Picture</title>
! <para>
! To summerise, a channel generally consists of either an analog signal running
! on POTS (or Plain Old Telephone System) or some combination of codec and
! signalling protocol, ie. GSM and SIP or ULAW and IAX.
! </para>
</sect2>
<sect2>
<title>Channels</title>
! <para>
! A channel is a voice path equivilent to a phone line between two points.
! There are many different ways they can be sent, but can be split into two
! groups -- analog and digital. Analog data is the type of signal that has
! been used on the phone system since it was invented. It can be prone to
! noise and echo and can not be sent as is over a digital network in a raw
! form. Digital data consist of ones and zeros. Analog data as picked up
! from a microphone can not be sent as is over a digital network and must
! be converted into a series of discrete levels, or quantised, to be able
! to form a digital signal. Once the data is in a digital state it will
! require a fair amount of bandwidth to send as is (64kbits/sec for
! uncompressed voice data sampled at 8KHz with 8bits resolution).
! </para>
</sect2>
<sect2>
<title>Codecs and Conversions</title>
! <para>
! Obviously it is desirable to fit as many calls as possible into a data network.
! This is done by encoding it into a form that takes up much less space
! using a codec (short for COder/DECoder). Some examples of these are ulaw,
! alaw, gsm, ilbc and g.729. Codecs determine the sustained data bit rate which
! is required for each channel. Different codecs have different advantages but
! are independent of the type of protocol that is used to establish the channel.
! The codec converts the analog voice signal to a digitally encoded one. The
! quality, databit rate required and the computational requirements vary from one
! codec to the next.
! </para>
</sect2>
<sect2>
! <title>Protocols</title>
! <para>
! Sending data to another phone would be easy if the data found its own way there
! and knew what to do at the other end. Unfortunately it doesn't which is
! why we use a signalling protocol to encapsulate the voice data. The common
! signalling protocol used today is SIP (an acronym for Session Initiation Protocol).
! Others that Asterisk supports include IAX, H.323 and CAPI. CAPI is a special
! case in that it is used within a computer system to deal with ISDN interfaces.
! </para>
</sect2>
</sect1>
***************
*** 78,90 ****
<sect3>
<title>Asterisk (Main PBX & Channels)</title>
! <para/>
</sect3>
<sect3>
<title>Zaptel (Drivers for Zaptel Hardware)</title>
! <para/>
</sect3>
<sect3>
<title>Libpri (ISDN PRI Drivers for Zaptel)</title>
! <para/>
</sect3>
</sect2>
--- 134,159 ----
<sect3>
<title>Asterisk (Main PBX & Channels)</title>
! <para>
! The Asterisk software is what gives a computer system the logic required
! to run a PBX system. IP based channels, dialplans, AGI scripting and
! timing insensitive parts of Asterisk are included.
! </para>
</sect3>
<sect3>
<title>Zaptel (Drivers for Zaptel Hardware)</title>
! <para>
! The drivers for Digium hardware can be obtained from the CVS server.
! These will allow you to integrate many types of legecy telephony
! equipment such as T1/E1, PSTN, FXO and FXS devices.
! </para>
</sect3>
<sect3>
<title>Libpri (ISDN PRI Drivers for Zaptel)</title>
! <para>
! Libpri will allow Asterisk to work with Primary Rate ISDN interfaces.
! It is based on the Bellcore specification SR-NWT-002343 for National
! ISDN. You will need to compile and configure these drivers if you need
! Asterisk to speak with an ISDN interface.
! </para>
</sect3>
</sect2>
***************
*** 93,110 ****
<sect3>
<title>Overview</title>
! <para/>
</sect3>
<sect3>
<title>X100P and X101P</title>
! <para>
! The X100P and X101P are Foreign Exchange Office (FXO) devices which
! allow you to connect the Asterisk PBX to a PSTN line. The only difference
! between an X100P and X101P is a slight difference in chips, but makes no difference
! to it's operation. The X100P uses a single PCI slot and supports FXS
! Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision).
! With the X100P Asterisk supports both incoming and outgoing calls and
! supports ring detection and remote hangup.
! </para>
! </sect3>
<sect3>
<title>S100U - Single Port FXS USB Interface</title>
--- 162,185 ----
<sect3>
<title>Overview</title>
! <para>
! Zaptel hardware is designed and built but Digium, the owners of Asterisk.
! The Asterisk PBX system is designed to work with these devices, so are fully
! supported. Drivers are provided to run the devices on a Linux based
! operating system.
! </para>
</sect3>
<sect3>
<title>X100P and X101P</title>
! <para>
! The X100P and X101P are Foreign Exchange Office (FXO) devices which
! allow you to connect the Asterisk PBX to a PSTN line. The only difference
! between an X100P and X101P is a slight difference in chips, but makes no difference
! to it's operation. The X100P uses a single PCI slot and supports FXS
! Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision).
! With the X100P Asterisk supports both incoming and outgoing calls and
! supports ring detection and remote hangup.
! </para>
! </sect3>
!
<sect3>
<title>S100U - Single Port FXS USB Interface</title>
***************
*** 116,132 ****
</sect3>
<sect3>
! <title>TDM400P</title>
! <para>
! The TDM400P is a half-length PCI 2.2 compliant card which allows you to connect
! standard analog telephones and analog lines to a computer. The card uses small
! modules to activate the 4 ports on the card. Depending on which daughter card is plugged
! onto the board will determine whether the port acts as an FXO or FXSinterface. The boards
! are not selectable between modes; the module used determines the type of interface.
! </para>
! <para>
! There is an alternate naming convention used as well to reference the type of modules attached
! to the TDM400P. This is in the form TDM##B where the first hash is the number of FXS (0-4) interfaces and the
! second hash is the number of FXO (0-4) interfaces.
! </para>
</sect3>
<sect3>
--- 191,208 ----
</sect3>
<sect3>
! <title>TDM400P</title>
! <para>
! The TDM400P is a half-length PCI 2.2 compliant card which allows you to connect
! standard analog telephones and analog lines to a computer. The card uses small
! modules to activate the 4 ports on the card. Depending on which daughter card is plugged
! onto the board will determine whether the port acts as an FXO or FXSinterface. The boards
! are not selectable between modes; the module used determines the type of interface.
! </para>
!
! <para>
! There is an alternate naming convention used as well to reference the type of modules attached
! to the TDM400P. This is in the form TDM##B where the first hash is the number of FXS (0-4) interfaces and the
! second hash is the number of FXO (0-4) interfaces.
! </para>
</sect3>
<sect3>
***************
*** 181,184 ****
--- 257,261 ----
</sect3>
</sect2>
+
<sect2>
<title>Applications</title>