[Asterisk-doc] introduction add-ons and spellcheck
Martin List-Petersen
asterisk-doc@lists.digium.com
Tue, 01 Jun 2004 01:42:06 +0100
--=-VfYQAzDe/geA2epEEBye
Content-Type: text/plain
Content-Transfer-Encoding: 7bit
New diff, that should cleanly apply after submitting Nicolas changes.
Kind regards,
Martin List-Petersen
martin at list-petersen dot net
On Mon, 2004-05-31 at 17:12, Martin List-Petersen wrote:
> I found out, that i could write some of the sections a bit better.
> Here is the corrected version of the patch.
>
> Kind regards,
> Martin List-Petersen
> martin at list-petersen dot dk
>
> On Mon, 2004-05-31 at 16:53, Martin List-Petersen wrote:
> > Here are some addon's for the introduction stuff
> >
> > (Zap, IAX, SIP, H.323).
> >
> > Would you have a look at it and submit it, if it is acceptable ?
> > I would like to ask for write access to the cvs repository, but you can
> > wait until i have
> > submitted more, if you feel better with that.
> >
> > The diff is based on today's anonymous cvs checkout.
> >
> > Kind regards,
> > Martin List-Petersen
> > martin at list-petersen dot net
--=-VfYQAzDe/geA2epEEBye
Content-Disposition: attachment; filename=introduction.xml.diff
Content-Type: text/x-patch; name=introduction.xml.diff; charset=ansi_x3.4-1968
Content-Transfer-Encoding: 7bit
--- docs-current/introduction.xml 2004-05-31 13:12:54.000000000 +0100
+++ docs/introduction.xml 2004-06-01 01:39:44.000000000 +0100
@@ -22,18 +22,47 @@
</para>
</sect2>
<sect2>
- <title>PBX, IVR, ACD</title>
- <para/>
- </sect2>
- <sect2>
- <title>Telephony 101</title>
+ <title>Prerequisite Knowledge and Skills</title>
+ <para>
+ This book assumes the reader has knowledge of both telephony and
+ Linux system administration.
+ </para>
<sect3>
- <title>Basic Concepts (FXO/FXS, loop/ground start/PRI, etc.)</title>
- <para/>
+ <title>Telephony</title>
+ <para>
+ Obviously, you should know enough about the telephony
+ technology that you are using to use to be able to set
+ up and debug it. As a general guideline, you should
+ understand the difference between FXS/FXO, and what ISDN,
+ PRI, BRI, POTS, PSTN, VoIP, signaling, and codecs are.
+ </para>
+ <para>
+ For the novice, a good introductory work is Noll's
+ <citetitle pubwork="book">Introduction to Telephones and
+ Telephone Systems</citetitle>. Another indispensable
+ resource for all skill levels is <citetitle pubwork="book">
+ Newton's Telecom Dictionary</citetitle>.
+ </para>
</sect3>
<sect3>
- <title>Telephony Resources: Newton's Telecom Dictionary, etc.</title>
- <para/>
+ <title>System Administration</title>
+ <para>
+ This book assumes that you have an i386 machine with Linux
+ installed ready for Asterisk. Unfamiliarity with Linux
+ administration will only compound the difficulties
+ inherent in getting Asterisk installed and working. There
+ are many resources that can help one learn about Linux.
+ On the Internet, <ulink url="http://tldp.org/">Linux
+ Documentation Project</ulink> provides many great
+ resources for beginners. In the bookstore, Frisch's
+ <citetitle pubwork="book">Essential System Administration
+ </citetitle> along with Nemeth, et al.'s
+ <citetitle pubwork="book"> Linux Administration Handbook
+ </citetitle> and <citetitle pubwork="book">Unix System
+ Administration Handbook</citetitle> are the best. Buying
+ one or two of these books can save a lot of headaches down
+ the road.
+ </para>
</sect3>
</sect2>
<sect2>
@@ -41,7 +70,7 @@
<sect3>
<title>Asterisk is not a turnkey system</title>
<para>
- The Asterisk PBX system is a complex peice of software. The learning curve
+ The Asterisk PBX system is a complex piece 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.
@@ -52,7 +81,7 @@
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
+ relationship extends throughout 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
@@ -64,7 +93,7 @@
<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
+ is its power. Most (if not all) other PBXs 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>
@@ -80,22 +109,22 @@
<sect2>
<title>The Big Picture</title>
<para>
- To summerise, a channel generally consists of either an analog signal running
+ To summarize, 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.
+ signaling 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.
+ A channel is a voice path equivalent 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
+ be converted into a series of discrete levels, or quantized, 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).
@@ -120,8 +149,8 @@
<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).
+ why we use a signaling protocol to encapsulate the voice data. The common
+ signaling 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>
@@ -143,7 +172,7 @@
<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
+ These will allow you to integrate many types of legacy telephony
equipment such as T1/E1, PSTN, FXO and FXS devices.
</para>
</sect3>
@@ -195,7 +224,7 @@
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
+ onto the board will determine whether the port acts as an FXO or FXS interface. The boards
are not selectable between modes; the module used determines the type of interface.
</para>
@@ -230,26 +259,81 @@
</sect3>
</sect2>
<sect2>
- <title>Channels</title>
+ <title>Protocols/Channels</title>
<sect3>
- <title>Zaptel Devices/Channels</title>
- <para/>
+ <title>Zap Devices/Channels</title>
+ <para>
+ Zap channels (Zapata/Zaptel) are the channel-type, that is used for FXO, FXS
+ and PRI-cards. There is also a third-party module, that implements zap
+ channels for certain BRI ISDN cards.
+ </para>
+ <para>
+ The Zap channels were originally given the name by the Zapata Telephony Project,
+ which is an effort to bring affordable computer telephony to the public domain.
+ This is happening because the commercial marked is ridiculous expensive and often
+ offers poor support. Besides that, it might be worth to mention that Zapata
+ was a Mexican revolutionary.
+ </para>
</sect3>
<sect3>
- <title>The IAX/IAX2 Protocol</title>
- <para/>
+ <title>The IAX Protocol</title>
+ <para>
+ IAX stands for InterAsterisk eXchange and was developed as an alternative to SIP
+ and H.323. Currently there exists 2 versions of IAX, where IAX2 is the most common
+ used. IAX is not submitted by any standards group, but is currently being adopted
+ by different manufacturers for both soft- and hard-phones.
+ </para>
+ <para>
+ The biggest advantage for IAX is that it only uses 1 UDP port and thus works
+ very well behind NAT firewalls. It allocates only the the minimum of bandwidth,
+ that is used at any time.
+ </para>
</sect3>
<sect3>
- <title>SIP</title>
- <para/>
+ <title>The SIP Protocol</title>
+ <para>
+ SIP, or the Session Initiation Protocol, is specified by the IETF. It allows text,
+ voice and multimedia sessions and uses port 5060 udp and tcp, but may use other
+ ports. Most VoIP devices on the marked currently support this protocol.
+ </para>
+ <para>
+ This protocol is not always easy to deploy in a firewalled environment, but with
+ the help of a STUN server not impossible.
+ </para>
</sect3>
<sect3>
<title>MGCP</title>
<para/>
</sect3>
<sect3>
- <title>H323</title>
- <para/>
+ <title>The H.323 Protocol</title>
+ <para>
+ H.323 is specified by the ITU-T (International Telecommunication Union Stanardization
+ Sector) and was meant for teleconferencing (Speech and Video). It basically should
+ enable you to terminate voice, video, fax and much more over IP, depending on what
+ features your client offers.
+ </para>
+ <para>
+ There are 2 implementations for H.323 that can be used with asterisk:
+ <orderedlist>
+ <listitem>
+ <para>
+ asterisk-oh323 - This was the first channel module available to asterisk, that
+ implemented H.323. It simulates a pseudo soundcard implementation to pass audio
+ from asterisk to the open h.323 stack.
+ </para>
+ </listitem>
+ <listitem>
+ <para>
+ chan_h323 - This channel module is part of asterisk now, it uses asterisk RTP stack and
+ implements H.323 in one shared library.
+ </para>
+ </listitem>
+ </orderedlist>
+ You might ask yourself now, what module you should choose and quite frankly, there is
+ no all-round answer to that. Implement whichever you have in hands and test it, if you
+ not are happy with that, try the other one.
+ </para>
</sect3>
<sect3>
<title>Skinny</title>
--=-VfYQAzDe/geA2epEEBye--