[Asterisk-doc] Patch for Introduction
Martin List-Petersen
asterisk-doc@lists.digium.com
Mon, 31 May 2004 23:37:55 +0100
Funny ... i had the same mispellings fixed in the patch i send earlier
today :o)
Kind regards,
Martin List-Petersen
martin at list-petersen dot net
On Mon, 2004-05-31 at 21:55, Nicholas Bachmann wrote:
> Howdy folks -
>
> I wrote a little bit of introduction that tells what we aren't going to
> cover in the book based on what everyone present (Jared and I) agreed in
> the first conference call. I also ran the spell check over what I
> submitted and it caught a few other misspellings in the file, which I
> fixed also.
>
> Hopefully I can get a chance to work on more this week, it's been a few
> months since I've offered up a meaty patch. :-) I plan to do some work
> on the agents/queues section, I've been doing some queue work at my job
> so some pitfalls are fresh in my mind...
>
> My cvs diff -u is attached for introduction.xml is attached.
>
> Nick
>
> ______________________________________________________________________
> Index: introduction.xml
> ===================================================================
> RCS file: /cvsroot/asterisk/docs/introduction.xml,v
> retrieving revision 1.5
> diff -u -r1.5 introduction.xml
> --- introduction.xml 31 May 2004 02:05:36 -0000 1.5
> +++ introduction.xml 31 May 2004 20:19:32 -0000
> @@ -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>
>