[Asterisk-doc] bug report? Something that need to be fixed.
Steven Critchfield
asterisk-doc@lists.digium.com
Thu, 01 Jan 2004 10:03:26 -0600
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On Thu, 2004-01-01 at 09:53, Steven Critchfield wrote:
> BTW, here is another diff after doing some more spell checking to be
> looked over. I know I could submit it myself, but I am not comfortable
> with doing that yet. I'd rather be peer reviewed first.
would help to submit the diff
--
Steven Critchfield <critch@basesys.com>
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? words.list
Index: NOTES
===================================================================
RCS file: /cvsroot/asterisk/docs/NOTES,v
retrieving revision 1.2
diff -u -r1.2 NOTES
--- NOTES 30 Dec 2003 18:21:26 -0000 1.2
+++ NOTES 1 Jan 2004 09:25:13 -0000
@@ -47,7 +47,7 @@
functions: <function>Answer()</function>
<function>Background(<replaceable>filename</replaceable>)</function>
- simplelist: <simplelist>
+ simple list: <simplelist>
<member></member>
</simplelist>
Index: appendix03.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/appendix03.xml,v
retrieving revision 1.18
diff -u -r1.18 appendix03.xml
--- appendix03.xml 29 Dec 2003 23:39:10 -0000 1.18
+++ appendix03.xml 1 Jan 2004 09:25:15 -0000
@@ -76,7 +76,7 @@
<para>
The option string may contain zero or more of the following characters:
<simplelist>
- <member>'s' -- silent login - do not announce the login ok segment</member>
+ <member>'s' -- silent login - do not announce the login OK segment</member>
</simplelist>
</para>
@@ -90,8 +90,8 @@
Executes an Asterisk Gateway Interface compliant
program on a channel. AGI allows Asterisk to launch external programs
written in any language to control a telephony channel, play audio,
- read DTMF digits, etc. by communicating with the AGI protocol on stdin
- and stdout. Returns -1 on hangup or if application requested hangup, or
+ read DTMF digits, etc. by communicating with the AGI protocol on STDIN
+ and STDOUT. Returns -1 on hangup or if application requested hangup, or
0 on non-hangup exit. Using 'EAGI' provides enhanced AGI, with audio
available out of band on file descriptor 3.
</para>
@@ -1260,7 +1260,7 @@
</formalpara>
<para>
Requests client go to URL. If the client
- does not support html transport, and there exists a step with
+ does not support HTML transport, and there exists a step with
priority n + 101, then execution will continue at that step.
Otherwise, execution will continue at the next priority level.
</para>
Index: chapter01.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter01.xml,v
retrieving revision 1.3
diff -u -r1.3 chapter01.xml
--- chapter01.xml 29 Dec 2003 23:39:10 -0000 1.3
+++ chapter01.xml 1 Jan 2004 09:25:15 -0000
@@ -199,7 +199,7 @@
that I have found so far is at
<ulink url="http://sourceforge.net/projects/astguiclient/">http://sourceforge.net/projects/astguiclient/</ulink>
developed by Matt Florell. This program was designed as a GUI client for the
- Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as
+ Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or soft phones as
extensions, it could be adapted to other functions, but It was designed for
Zap/SIP users. The program will run on X and Win32.
</para>
Index: chapter02.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter02.xml,v
retrieving revision 1.5
diff -u -r1.5 chapter02.xml
--- chapter02.xml 29 Dec 2003 23:39:10 -0000 1.5
+++ chapter02.xml 1 Jan 2004 09:25:16 -0000
@@ -131,7 +131,7 @@
is part of the development packages.
</para>
<para>
- If you are using Devian simply install the required packages with
+ If you are using Debian simply install the required packages with
<command>apt-get install libreadline4-dev libssl-dev</command> .
</para>
<para>
@@ -245,7 +245,7 @@
<sect3>
<title>CAPI/ISDN</title>
<para>
- The complete source code is available from kapejods website
+ The complete source code is available from Kapejod website
</para>
<para>
<ulink url="http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.0.tar.gz" type="http">
@@ -272,7 +272,7 @@
</para>
<para>
Then you can set some build time configuration parameters like early B3
- connects, DEFLECT_ON_CIRCUITBUSY or software dtmf detection/generation.
+ connects, DEFLECT_ON_CIRCUITBUSY or software DTMF detection/generation.
If everything is done simply save the file.
</para>
<para>
@@ -301,7 +301,7 @@
</para>
<para>
After these steps your channel-module is available in * but it has to be
- configured. This is done in the main CAPI configfile capi.conf.
+ configured. This is done in the main CAPI config file capi.conf.
</para>
</sect3>
</sect2>
Index: chapter05.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter05.xml,v
retrieving revision 1.4
diff -u -r1.4 chapter05.xml
--- chapter05.xml 22 Dec 2003 03:48:48 -0000 1.4
+++ chapter05.xml 1 Jan 2004 09:25:16 -0000
@@ -187,7 +187,7 @@
</sect4>
<sect4>
- <title>Voice Over IPVoIP Provider</title>
+ <title>Voice Over IP/VoIP Provider</title>
<para/>
</sect4>
Index: chapter06.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter06.xml,v
retrieving revision 1.5
diff -u -r1.5 chapter06.xml
--- chapter06.xml 22 Dec 2003 03:48:48 -0000 1.5
+++ chapter06.xml 1 Jan 2004 09:25:17 -0000
@@ -3,9 +3,9 @@
<sect1>
<title>Agents and the Asterisk ACD</title>
<para>
- Asterisk provides a flexable call queueing system suitable for
- callcenter applications. The Asterisk ACD system utilizes several
- components, which work together to provide a very robust implemention.
+ Asterisk provides a flexible call queuing system suitable for
+ call center applications. The Asterisk ACD system utilizes several
+ components, which work together to provide a very robust implementation.
</para>
<sect2>
@@ -15,8 +15,8 @@
to the appropriate extensions. These extensions can be agents logged into
the system or any other type of channel supported by the system. Various
strategies can be used to determine how calls are routed from a queue,
- these strategies are used to imprement fair distribution of workload
- within a callcenter, and can be customised through the use of priority
+ these strategies are used to implement fair distribution of workload
+ within a call center, and can be customized through the use of priority
levels to fit an organization's policies.</para>
<para>Queues are configured using the queues.conf configuration file.</para>
@@ -74,7 +74,7 @@
; Queue members.
; Queue members can be any kind of channel supported by Asterisk.
- ; Agent channels are generally preferred, as they provide login/logout functunality.
+ ; Agent channels are generally preferred, as they provide login/logout functionality.
; Agent number 1000 (agents are defined in agents.conf)
member => Agent/1000
@@ -86,7 +86,7 @@
member => Agent/@1
; Agent 2000 is a supervisor that is capable of taking calls, but should only do so
- ; when no other agents are available, so we consider vith a penalty.
+ ; when no other agents are available, so we consider with a penalty.
member => 2000,4
@@ -160,7 +160,7 @@
To use TDMoE you MUST have a zaptel interface configured somewhere on the
network. It can be any zaptel interface, doesn't have to be a E400P, an
X100P will do. Why? Timing. Samples. Something like that. Just do it.
- Ofcourse a dummy ZAP interface like ztdummy or ztrtc might work, but I
+ Of course a dummy ZAP interface like ztdummy or ztrtc might work, but I
haven't tried it as yet. If somebody has please do update this.
</para>
@@ -169,7 +169,7 @@
</para>
<para>
- Well, we all know ethernet right? Its prolly the most popular network
+ Well, we all know ethernet right? Its probably the most popular network
infrastructure on Layer2 that the IP world knows. Time-division
multiplexing (TDM) puts multiple data streams in a single signal by
separating the signal into many segments, each of a short duration
@@ -264,7 +264,7 @@
<para>
Remember that TDMoE works at the ethernet layer, all you need to configure
is MAC addresses and ethernet interfaces.... so in theory you could TDMoE
- over 802.11 (low-cost last mile) or cipe (encrypted PRI), the possibilites
+ over 802.11 (low-cost last mile) or cipe (encrypted PRI), the possibilities
are limitless (well as limitless as csmacd can get)... IP does not come
into play here at all...
</para>
@@ -350,9 +350,9 @@
Hail * !
</para>
<para>
- Todo: multiple ethernet cards (local and remote), other signalling
+ TODO: multiple ethernet cards (local and remote), other signalling
examples, dummy eth driver to loopback test, caveats, benefits of TDMoE,
- comparision of various signalling, cook dinner
+ comparison of various signalling, cook dinner
</para>
</sect2>
@@ -535,7 +535,7 @@
<para>
(As a note, I believe that Asterisk only supports IAX2 and SIP as well as PSTN
for the URI's enum will return - I cannot be 100% sure of this so anyone in-the-know
- is free to correct me) In our examples, we will be using an iax2 user for
+ is free to correct me) In our examples, we will be using an IAX2 user for
receiving our calls.
</para>
Index: chapter07.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter07.xml,v
retrieving revision 1.7
diff -u -r1.7 chapter07.xml
--- chapter07.xml 29 Dec 2003 16:28:28 -0000 1.7
+++ chapter07.xml 1 Jan 2004 09:25:17 -0000
@@ -82,7 +82,7 @@
In this example, the class <command>default</command> plays MP3s from the directory
/var/lib/asterisk/mohmp3/ sequentially. Note the 'quietmp3' directive, which keeps the music
at an appropriate volume for most telephony Music on Hold applications. The class
- <command>nirvana</command> is similar, but uses the directory /usr/share/mp3/nivrana-music/ instead.
+ <command>nirvana</command> is similar, but uses the directory /usr/share/mp3/nirvana-music/ instead.
<command>random-nirvana</command> picks files in the directory randomly, instead of sequentially,
due to the '-z' option at the end of the line. The final class, <command>loud-nirvana</command>
does not reduce the volume of the output, due to 'quietmp3' being replaced by 'mp3'. The 'mp3'
@@ -186,7 +186,7 @@
<programlisting>
[general]
port=5060 <lineannotation>; make sure you have this line</lineannotation>
- externip=my.domain.com <lineannotation>; this can be either external ip address, or FQDN</lineannotation>
+ externip=my.domain.com <lineannotation>; this can be either external IP address, or FQDN</lineannotation>
localmask=255.255.255.0 <lineannotation>; subnet mask. This example is a /24 (class C)</lineannotation>
localnet=192.168.0.0 <lineannotation>; local network your Asterisk server is in.</lineannotation>
</programlisting>
@@ -254,7 +254,7 @@
<programlisting>
mailbox=1234
</programlisting>
- You can associate more than one mailbox with a SIP phone for a message waiting indication by seperating
+ You can associate more than one mailbox with a SIP phone for a message waiting indication by separating
the voice mail box numbers with commas:
<programlisting>
mailbox=1234,9999
@@ -313,14 +313,14 @@
<para>
In Figure 7-1, you can see there are several options for echo
cancellation. Commenting out all but one of these lines is required. If
- you'd like to use the MARK3 echo canceller, for instance, you'd comment
+ you'd like to use the MARK3 echo canceler, for instance, you'd comment
out the MARK2 line and uncomment the MARK3 line.
</para>
<para>
- All four of the echo cancellers will do a mediocre to good job of taking
+ All four of the echo cancelers will do a mediocre to good job of taking
care of echo, but it takes a little while for Asterisk to properly adjust.
- If you use the MARK2 canceller, there's an additional option:
+ If you use the MARK2 canceler, there's an additional option:
</para>
<para>
@@ -340,7 +340,7 @@
<para>
Now, thanks to the efforts of Brian West and the other Asterisk gang, we
now have a feature in Zaptel called Echo Training. Echo training, in my
- experience, works the best out of all of the echo cancellers.
+ experience, works the best out of all of the echo cancelers.
</para>
<figure id="echo-fig2"><title>zapata.conf echo training definition for FXO channel</title>
@@ -419,14 +419,14 @@
imagination), in which case the audio is doing something called "over
deviation" - it's the same thing that happens when people get too close
to a microphone and the audio is crackly. When this occurs, the echo
- canceller cannot compensate for the signal as well since it is busy
+ canceler cannot compensate for the signal as well since it is busy
receiving artifacts of the audio that "spill" back into the channel.
In this case, we want to lower the txgain level a bit.
</para>
<para>
Most people who configure echotraining correctly will never hear echo in
- their calls again. The echo canceller works nearly instantaneously in
+ their calls again. The echo canceler works nearly instantaneously in
echotraining mode.
</para>
</sect2>
@@ -486,7 +486,7 @@
discouraged by this; it is free market capitalization and is for the
good of the Asterisk community. If you do not wish to pay for quality
support, that is fine; many people will not answer your questions,
- however. Usually, if youre willing to pay for support, let it be
+ however. Usually, if you're willing to pay for support, let it be
known early on and your chances of receiving quality support will
increase.
</para>
Index: colophon.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/colophon.xml,v
retrieving revision 1.1
diff -u -r1.1 colophon.xml
--- colophon.xml 12 Dec 2003 05:14:15 -0000 1.1
+++ colophon.xml 1 Jan 2004 09:25:17 -0000
@@ -1,6 +1,6 @@
<colophon>
<para>
This document was written as an excuse to become more familiar with the
- Docbook format, and to contribute back to the Asterisk project.
+ DocBook format, and to contribute back to the Asterisk project.
</para>
</colophon>
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