[Asterisk-doc] docs introduction.xml,1.7,1.8
blitzrage
asterisk-doc@lists.digium.com
Wed, 2 Jun 2004 01:20:56 +0000
Comments:
Update of /cvsroot/asterisk/docs
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv5139/docs
Modified Files:
introduction.xml
Log Message:
blitzrage
- a couple of spelling / grammer changes
- removed sections for VoIP providers and specific softphones
Index: introduction.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/introduction.xml,v
retrieving revision 1.7
retrieving revision 1.8
diff -C2 -d -r1.7 -r1.8
*** introduction.xml 1 Jun 2004 12:02:04 -0000 1.7
--- introduction.xml 2 Jun 2004 01:20:50 -0000 1.8
***************
*** 15,19 ****
<para>
! Since Asterisk can do so many things it.s not possible to cover them all in this book.
We will attempt to cover the most commonly used features of the system and attempt to answer many of the
commonly asked questions. This book contains quite a few appendices which became a bit of a necessity as
--- 15,19 ----
<para>
! Since Asterisk can do so many things it is not possible to cover them all in this book.
We will attempt to cover the most commonly used features of the system and attempt to answer many of the
commonly asked questions. This book contains quite a few appendices which became a bit of a necessity as
***************
*** 32,36 ****
<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,
--- 32,36 ----
<para>
Obviously, you should know enough about the telephony
! technology that you are using to be able to set
up and debug it. As a general guideline, you should
understand the difference between FXS/FXO, and what ISDN,
***************
*** 192,196 ****
<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
--- 192,196 ----
<title>Overview</title>
<para>
! Zaptel hardware is designed and built by 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
***************
*** 271,275 ****
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.
--- 271,275 ----
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 market is ridiculously expensive and often
offers poor support. Besides that, it might be worth to mention that Zapata
was a Mexican revolutionary.
***************
*** 279,283 ****
<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
--- 279,283 ----
<title>The IAX Protocol</title>
<para>
! IAX stands for Inter-Asterisk 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
***************
*** 285,289 ****
</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.
--- 285,289 ----
</para>
<para>
! The biggest advantage for IAX is that it uses only one UDP port and thus works
very well behind NAT firewalls. It allocates only the the minimum of bandwidth,
that is used at any time.
***************
*** 295,303 ****
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>
--- 295,305 ----
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 market 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. Asterisk is able to translate the
! information in the packet headers so that it is possible to run in a NAT'd
! environment. See chapter 8 for more information.
</para>
</sect3>
***************
*** 315,334 ****
</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
--- 317,339 ----
</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 the
! 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
***************
*** 379,383 ****
<sect3>
<title>Soft Phones</title>
! <sect4>
<title>Gnophone</title>
<para/>
--- 384,393 ----
<sect3>
<title>Soft Phones</title>
! <para>
! [blitzrage - removed the specific names of phones. This should be on the
! http://www.asteriskdocs.org website. Lets be more general here and explain
! what a softphone is and advantages and disadvantages of them]
! </para>
! <!-- <sect4>
<title>Gnophone</title>
<para/>
***************
*** 394,398 ****
<title>X-Lite/Pro</title>
<para/>
! </sect4>
</sect3>
<sect3>
--- 404,408 ----
<title>X-Lite/Pro</title>
<para/>
! </sect4> -->
</sect3>
<sect3>
***************
*** 449,452 ****
--- 459,465 ----
<sect3>
<title>Other hardware options</title>
+ <para>
+ [blitzrage - probably going to move this stuff to an appendix]
+ </para>
<sect4>
<title>VoiceTronix OpenLine and OpenSwitch Cards</title>
***************
*** 499,503 ****
</sect3>
</sect2>
! <sect2>
<title>VoIP Service Providers</title>
<sect3>
--- 512,517 ----
</sect3>
</sect2>
! <!-- This stuff can probably go on the website, or pointed at the wiki -->
! <!-- <sect2>
<title>VoIP Service Providers</title>
<sect3>
***************
*** 509,513 ****
<para/>
</sect3>
! </sect2>
</sect1>
</chapter>
--- 523,527 ----
<para/>
</sect3>
! </sect2> -->
</sect1>
</chapter>