[Asterisk-doc] docs channels.xml,1.1,1.2 introduction.xml,1.17,1.18
blitzrage
asterisk-doc@lists.digium.com
Fri, 23 Jul 2004 19:40:19 +0000
Comments:
Update of /cvsroot/asterisk/docs
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv6875/docs
Modified Files:
channels.xml introduction.xml
Log Message:
blitzrage
- minor changes to introduction
- move channel stuff from intro to channels chapter
Index: channels.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/channels.xml,v
retrieving revision 1.1
retrieving revision 1.2
diff -C2 -d -r1.1 -r1.2
*** channels.xml 21 Jul 2004 23:21:34 -0000 1.1
--- channels.xml 23 Jul 2004 19:40:13 -0000 1.2
***************
*** 17,20 ****
--- 17,21 ----
to IAX, etc.)]
</para>
+
<para>
In Asterisk a channel is a form of communications between some outside resource such as
***************
*** 24,27 ****
--- 25,29 ----
SIP (Session Initiation Protocol).
</para>
+
<para>
Different channels have their own set of features such as sending or receiving
***************
*** 32,35 ****
--- 34,53 ----
</para>
+ <sect2>
+ <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 market is ridiculously expensive and often
+ offers poor support. Besides that, it might be worth to mention that Zapata
+ was a Mexican revolutionary.
+ </para>
+ </sect2>
</sect1>
Index: introduction.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/introduction.xml,v
retrieving revision 1.17
retrieving revision 1.18
diff -C2 -d -r1.17 -r1.18
*** introduction.xml 21 Jul 2004 17:15:04 -0000 1.17
--- introduction.xml 23 Jul 2004 19:40:13 -0000 1.18
***************
*** 123,127 ****
Support for non-linux platforms is provided by third-parties, and as a
result there are various limitations on features, drivers or release dates
! vs Asterisk on Linux. As that support becomes integrated into Digium's
releases, these limitations will go away.
</para>
--- 123,127 ----
Support for non-linux platforms is provided by third-parties, and as a
result there are various limitations on features, drivers or release dates
! versus Asterisk on Linux. As that support becomes integrated into Digium's
releases, these limitations will go away.
</para>
***************
*** 147,153 ****
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>
--- 147,154 ----
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 presuming a production quality system is possible
! in only 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. However the intension of this book
! is to get you up to speed as quickly as possible.
</para>
</sect3>
***************
*** 155,161 ****
<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 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>
--- 156,162 ----
<title>Don't like it? Change it yourself!</title>
<para>
! Asterisk is open source software. The ability to read the source code
! is its power. Most (if not all) other PBX systems are entirely closed source
! with only the abilities that have been provided to you. If something does not
work quite the way you would expect it to, you are able to change it.
</para>
***************
*** 174,178 ****
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
! signaling protocol, ie. GSM and SIP or ULAW and IAX.
</para>
</sect2>
--- 175,179 ----
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
! signaling protocol, ie. GSM and SIP, or, ULAW and IAX.
</para>
</sect2>
***************
*** 181,194 ****
<para>
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 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).
</para>
</sect2>
--- 182,194 ----
<para>
A channel is a voice path equivalent to a phone line between two points.
! There are many different ways the information can be sent, but often is 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 its raw
form. Digital data consist of ones and zeros. Analog data as picked up
! from a microphone must 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 (relatively speaking) to send as-is;
! 64kbits/sec for uncompressed voice data sampled at 8KHz with 8bits resolution.
</para>
</sect2>
***************
*** 200,207 ****
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>
--- 200,207 ----
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>
***************
*** 214,219 ****
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>
</sect2>
--- 214,219 ----
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).
! Other protocols that Asterisk support 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>
***************
*** 228,232 ****
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>
--- 228,233 ----
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 can be used without any
! additional hardware.
</para>
</sect3>
***************
*** 255,259 ****
<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
operating system.
--- 256,260 ----
<para>
Zaptel hardware is designed and built by Digium, the owners of Asterisk.
! The Asterisk PBX system is designed to work with these devices and so are fully
supported. Drivers are provided to run the devices on a Linux based
operating system.
***************
*** 272,275 ****
--- 273,283 ----
supports ring detection and remote hangup.
</para>
+
+ <note>
+ <para>
+ These cards can not be used as an FXS device for attaching analog telephones
+ to the Asterisk PBX.
+ </para>
+ </note>
</sect3>
***************
*** 287,291 ****
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 FXS interface. The boards
are not selectable between modes; the module used determines the type of interface.
--- 295,299 ----
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. Which daughter card is plugged
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.
***************
*** 293,299 ****
<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>
--- 301,313 ----
<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>
!
! <para>
! With the development of the FXO module for the TDM400P it has become the
! preferred FXO interface device.
</para>
</sect3>
***************
*** 305,310 ****
telephony and data protocols, including both RBS and Primary Rate ISDN (PRI)
protocol families for voice and PPP, Cisco HDLC and Frame Relay modes.
! The T100P can also be connected via channels banks for use with Asterisk.
</para>
<para>
The E100P version is essentially the same card supporting the E-1 European
--- 319,325 ----
telephony and data protocols, including both RBS and Primary Rate ISDN (PRI)
protocol families for voice and PPP, Cisco HDLC and Frame Relay modes.
! The T100P can also be connected to channel banks for use with Asterisk.
</para>
+
<para>
The E100P version is essentially the same card supporting the E-1 European
***************
*** 316,320 ****
<para>
The T400P and E400P are 4 port versions of the T100P and E100P respectively.
! The TE410P is a quad T1 or E1 selectable per card or per port, allowing you
to do both signaling formats in a single card. This card only supports a
3.3v PCI bus available on newer machines or in 64-bit PCI bus architectures.
--- 331,335 ----
<para>
The T400P and E400P are 4 port versions of the T100P and E100P respectively.
! The TE410P is a quad T1 or E1, selectable per card or per port, allowing you
to do both signaling formats in a single card. This card only supports a
3.3v PCI bus available on newer machines or in 64-bit PCI bus architectures.
***************
*** 323,369 ****
</sect2>
<sect2>
! <title>Protocols/Channels</title>
! <sect3>
! <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 market is ridiculously 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 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
! by different manufacturers for both soft- and hard-phones.
! </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.
! </para>
</sect3>
<sect3>
<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 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>
<sect3>
--- 338,372 ----
</sect2>
<sect2>
! <title>Protocols</title>
<sect3>
<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
! by different manufacturers for both soft- and hard-phones.
! </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.
! </para>
</sect3>
+
<sect3>
<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 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>
<sect3>
***************
*** 374,406 ****
<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 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
! not are happy with that, try the other one.
! </para>
</sect3>
<sect3>
--- 377,410 ----
<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 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
! not are happy with that, try the other one.
! </para>
</sect3>
<sect3>
***************
*** 419,422 ****
--- 423,427 ----
dialplan scripting.
</para>
+
<para>
The voicemail system, called Comedian, will be extensivly covered. This will
***************
*** 424,427 ****
--- 429,433 ----
as well as the actual use of the Comedian mail system.
</para>
+
<para>
You can extend the capabilities of Asterisk through the AGI scripting interface.
***************
*** 430,462 ****
</para>
</sect2>
-
- <!-- <sect3>
- <title>Dial and Other Basics</title>
- <para/>
- </sect3>
- <sect3>
- <title>Voicemail</title>
- <para/>
- </sect3>
- <sect3>
- <title>Dial-Plan Scripting</title>
- <para/>
- </sect3>
- <sect3>
- <title>Call Detail Recording (CDR)</title>
- <para/>
- </sect3>
- </sect2>
- <sect2>
- <title>Extensibility</title>
- <sect3>
- <title>AGI</title>
- <para/>
- </sect3>
- <sect3>
- <title>Custom Applications</title>
- <para/>
- </sect3> -->
</sect1>
<sect1>
<title>Add-On/Optional Components</title>
--- 436,441 ----
</para>
</sect2>
</sect1>
+
<sect1>
<title>Add-On/Optional Components</title>
***************
*** 464,498 ****
<!-- <title>Software</title> -->
<!-- <sect3> -->
! <title>Soft Phones</title>
! <para>
! Softphones are software based interfaces on a modern PC. These softphones
! allow you to place and receive calls at your computer using a headset and
! microphone. Softphones have the advantage of being extremely portable,
! especially with a laptop. There are several freely available softphones
! working with a variety of protocols. Because of the number of softphones
! available and their configurations, they will not be covered extensively
! so you should consult the documentation of any softphone which you wish
! to use. We will however deal with the configuration of Asterisk for use
! with softphones.
! </para>
! <!-- <sect4>
! <title>Gnophone</title>
! <para/>
! </sect4>
! <sect4>
! <title>iaxClient/iaxComm</title>
! <para/>
! </sect4>
! <sect4>
! <title>DIAX</title>
! <para/>
! </sect4>
! <sect4>
! <title>X-Lite/Pro</title>
! <para/>
! </sect4> -->
! <!-- </sect3>
! <sect3> -->
</sect2>
<sect2>
<title>Management Tools</title>
--- 443,460 ----
<!-- <title>Software</title> -->
<!-- <sect3> -->
! <title>Soft Phones</title>
! <para>
! Softphones are software based interfaces on a modern PC. These softphones
! allow you to place and receive calls at your computer using a headset and
! microphone. Softphones have the advantage of being extremely portable,
! especially with a laptop. There are several freely available softphones
! working with a variety of protocols. Because of the number of softphones
! available and their configurations, they will not be covered extensively
! so you should consult the documentation of any softphone which you wish
! to use. We will however deal with the configuration of Asterisk for use
! with softphones.
! </para>
</sect2>
+
<sect2>
<title>Management Tools</title>
***************
*** 545,578 ****
with your hardware devices for their configuration.
</para>
- <!-- <sect3>
- <para/>
- <title>VoIP Hard Phones</title>
- <para/>
- </sect3>
- <sect3>
- <title>VoIP Gateways</title>
- <para/>
- </sect3>
- <sect3>
- <title>Channel Banks</title>
- <para/>
- </sect3>
- <sect3>
- <title>Legacy PBX Systems</title>
- <para/>
- </sect3> -->
</sect2>
- <!-- This stuff can probably go on the website, or pointed at the wiki -->
- <!-- <sect2>
- <title>VoIP Service Providers</title>
- <sect3>
- <title>IAX providers</title>
- <para/>
- </sect3>
- <sect3>
- <title>SIP providers</title>
- <para/>
- </sect3>
- </sect2> -->
</sect1>
</chapter>
--- 507,511 ----