[Asterisk-doc] bug report? Something that need to be fixed.
Steven Critchfield
asterisk-doc@lists.digium.com
Mon, 29 Dec 2003 17:28:50 -0600
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On Mon, 2003-12-29 at 14:47, Jared Smith wrote:
> On Mon, 2003-12-29 at 13:20, Steven Critchfield wrote:
> > Readline support is not needed anymore as editline is being distributed
> > with asterisk. This occurs in several locations.
> >
> > Also in sect2, 4th paragraph, the distribution being referenced is
> > Debian. It is a combination of Ian Murdock and his SO Debby(sp?).
> >
> > Hope this helps.
> It certainly does... thank you! We're always looking for feedback on
> what's in the book so far.
>
> >
> > BTW, what is the appropriate way of signing out a section to be written?
> > I might be moved to write some documentation this holiday and don't want
> > to conflict with already being written sections. Any section that you
> > may think would be good for me to write would also be appreciated.
> We're still trying to formalize that part... for now, the best thing to
> do is to ask on the mailing list or in the IRC channel (#asterisk-doc).
> Do you have a specific chapter or section you'd like to write?
> Personally, I'd like to get the outline finished for the section on
> extensions.conf so that someone like you can do the actual writing...
> but that's just my personal opinion.
>
> (Steven, while I have your attention, were you OK with the last revision
> of the outline I posted, or do you have suggestions on how you'd rather
> see it?)
I'll do some more looking at it later tonight.
For now, I have to point out it would be a good thing to build a common
aspell or ispell dictionary/word list so that spell checking could be
automated a bit more and that certain words that aren't part of the word
list could be presented the same all the time. As an example aspell
wants to break up hungup into 2 words. I'm not sure that is a bad thing,
but many people understand that term to be one word and referring to
phones. Also the same thing for dialtone and dialplan. A consensus
should be made as to which is best for each of these words, and then
stick with it for the entirety of the manual/book. Also capitalization
of terms like NIC, PRI, VoIP, TDMoE, and others should be decided and
stuck to.
As a bit more of a contribution, I have been working through the current
CVS with aspell and trying to correct spelling. Notice I am using a
tool, my spelling sucks worse than what I am checking. I just feel that
it is important for a manual to not have misspellings as it will be the
first thing that makes a user have doubts about the quality of the
manual.
Anyways here is a diff of spelling changes I see need to be fixed.
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Steven Critchfield <critch@basesys.com>
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Index: appendix03.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/appendix03.xml,v
retrieving revision 1.17
diff -u -r1.17 appendix03.xml
--- appendix03.xml 22 Dec 2003 18:01:08 -0000 1.17
+++ appendix03.xml 29 Dec 2003 23:19:21 -0000
@@ -42,7 +42,7 @@
If none is specified, the default is used.
</para>
<para>
- Returns 0 unless CPE is hungup.
+ Returns 0 unless CPE is hung up.
</para>
<formalpara><title>AgentCallbackLogin: Call agent callback login</title>
@@ -113,11 +113,11 @@
</para>
</formalpara>
<para>
- Requires a user to enter agiven
+ Requires a user to enter a given
password in order to continue execution. If the
password begins with the '/' character, it is interpreted as
a file which contains a list of valid passwords (1 per line).
- an optional set of opions may be provided by concatenating any
+ an optional set of option's may be provided by concatenating any
of the following letters:
<simplelist>
<member>a - Set account code to the password that is entered</member>
@@ -311,10 +311,10 @@
the user hangs up, or all channels return busy or error.
</para>
<para>
- In general, the dialler will return 0 if it was unable to place
+ In general, the dialer will return 0 if it was unable to place
the call, or the timeout expired. However, if all channels were
busy, and there exists an extension with priority n+101
- (where n is the priority of the dialler instance), then it will
+ (where n is the priority of the dialer instance), then it will
be the next executed extension (this allows you to setup different
behavior on busy from no-answer).
</para>
@@ -567,7 +567,7 @@
<formalpara><title>GetCPEID: Get ADSI CPE ID</title>
<para>
- <!-- TODO: Check to make sure that there really aren't any arguements -->
+ <!-- TODO: Check to make sure that there really aren't any arguments -->
<function>GetCPEID()</function>
</para>
</formalpara>
@@ -576,7 +576,7 @@
properly setup zapata.conf for on-hook operations.
</para>
<para>
- Returns -1 on hanup only.
+ Returns -1 on hangup only.
</para>
<formalpara><title>Goto: Goto a particular priority, extension, or context</title>
@@ -898,7 +898,7 @@
<varlistentry><term><replaceable>announce_template</replaceable></term>
<listitem>
<para>
- colon seperated list of files to announce, the word PARKED
+ colon separated list of files to announce, the word PARKED
will be replaced by a say_digits of the ext the call is parked in
</para>
</listitem>
@@ -943,7 +943,7 @@
<function>Playback(<replaceable>filename</replaceable>[|<replaceable>option</replaceable>])</function>
</para>
</formalpara>
-<!-- TODO: This says that 'skip' is the only uption, but then goes on to -->
+<!-- TODO: This says that 'skip' is the only option, but then goes on to -->
<!-- say that 'noanswer' is also an option -->
<para>
Plays back a given <replaceable>filename</replaceable>
@@ -1052,7 +1052,7 @@
<para>
This application returns -1 if the originating channel hangs up, or if the
call is bridged and either of the parties in the bridge terminate the call.
- Returns 0 if the queue is full, nonexistant, or has no members.
+ Returns 0 if the queue is full, nonexistent, or has no members.
</para>
<formalpara><title>Read: Read a variable</title>
@@ -1115,7 +1115,7 @@
<para>
Causes the Call Data Record to be reset, optionally
storing the current CDR before zeroing it out (if 'w' option is
- specifed the record <emphasis role="strong">WILL</emphasis> be stored.
+ specified the record <emphasis role="strong">WILL</emphasis> be stored.
</para>
<para>
Always returns 0.
@@ -1268,7 +1268,7 @@
<function>SendURL()</function> only returns 0 if the URL was sent correctly or if
the channel does not support HTML transport, and -1 otherwise.
If the option 'wait' is specified, execution will wait for an
- acknowledgement that the URL has been loaded before continuing
+ acknowledgment that the URL has been loaded before continuing
and will return -1 if the peer is unable to load the URL.
</para>
@@ -1551,11 +1551,11 @@
Leaves voicemail for a given <replaceable>extension</replaceable>
(must be configured in <filename>voicemail.conf</filename>). If the
extension is preceded by an <replaceable>s</replaceable> then instructions
- for leaving the message will be skipped. If the extension is preceeded
+ for leaving the message will be skipped. If the extension is preceded
by <replaceable>u</replaceable> then the "unavailable"
message will be played
(<filename>/var/lib/asterisk/sounds/vm/<replaceable>context</replaceable>/<replaceable>exten</replaceable>/unavail</filename>)
- if it exists. If the extension is preceeded by a <replaceable>b</replaceable>
+ if it exists. If the extension is preceded by a <replaceable>b</replaceable>
then the the busy message will be played (that is, busy instead of unavail).
If the requested mailbox does not exist, and there exists a priority
n + 101, then that priority will be taken next.
Index: chapter01.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter01.xml,v
retrieving revision 1.2
diff -u -r1.2 chapter01.xml
--- chapter01.xml 20 Dec 2003 18:43:57 -0000 1.2
+++ chapter01.xml 29 Dec 2003 23:19:21 -0000
@@ -13,7 +13,7 @@
<sect2>
<title>Telephony 101</title>
<sect3>
- <title>Basic Conecpts (FXO/FXS, loop/ground start/PRI, etc.)</title>
+ <title>Basic Concepts (FXO/FXS, loop/ground start/PRI, etc.)</title>
<para/>
</sect3>
<sect3>
@@ -246,7 +246,7 @@
<title>ISDN/CAPI Cards (Eicon, etc.)</title>
<para>Integrating ISDN channels to * can be done by several ways.
Basically isdn4linux support is implemented in Asterisk.
- So called chan_modem_i4l. Another way is trough the powerfull CAPI
+ So called chan_modem_i4l. Another way is trough the powerful CAPI
interface. chan_capi is developed under the terms of the GPL an maintained
by "Sir Kapejod". It is highly recommended to use chan_capi if your card
is supported, because chan_capi supports even more functions than
Index: chapter02.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter02.xml,v
retrieving revision 1.4
diff -u -r1.4 chapter02.xml
--- chapter02.xml 16 Dec 2003 18:43:35 -0000 1.4
+++ chapter02.xml 29 Dec 2003 23:19:21 -0000
@@ -62,7 +62,7 @@
<para>
ISDN Hardware must not be expensive. A basic AVM card
that comes with CAPI compatible kernel modules is available
- for about 40$. But there are several differents between the
+ for about 40$. But there are several differences between the
capacity of the cards (think of more than 2 B-channels) and
of different ISDN standards. chan_capi is programmed to work
even with multiple ISDN cards. To use chan_capi you must
@@ -161,7 +161,7 @@
<sect2>
<title>Using "make"</title>
<para>Now we need to compile Asterisk as root :
- <!-- We may need to clean up this section a bit as the way it is laid out is a little wierd -->
+ <!-- We may need to clean up this section a bit as the way it is laid out is a little weird -->
</para>
<para>
<literallayout>
@@ -271,7 +271,7 @@
</literallayout>
</para>
<para>
- Then you can set some buildtime configuration parameters like early B3
+ Then you can set some build time configuration parameters like early B3
connects, DEFLECT_ON_CIRCUITBUSY or software dtmf detection/generation.
If everything is done simply save the file.
</para>
@@ -355,7 +355,7 @@
</para>
</sect2>
<sect2>
- <title>Accssing the CLI when Asterisk is running</title>
+ <title>Accessing the CLI when Asterisk is running</title>
<para>
If your asterisk is already running, you can reattach with the <command>-r</command>
switch.
Index: hgta.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/hgta.xml,v
retrieving revision 1.21
diff -u -r1.21 hgta.xml
--- hgta.xml 17 Dec 2003 17:01:17 -0000 1.21
+++ hgta.xml 29 Dec 2003 23:19:21 -0000
@@ -6,10 +6,10 @@
<!ENTITY bookinfo SYSTEM "bookinfo.xml"> <!-- Book information and title -->
<!ENTITY preface SYSTEM "preface.xml"> <!-- Introductory letter -->
<!ENTITY chapter01 SYSTEM "chapter01.xml"> <!-- Introductions to Asterisk / General Concepts -->
-<!ENTITY chapter02 SYSTEM "chapter02.xml"> <!-- Installing and Compiling Asterisk and Componants -->
+<!ENTITY chapter02 SYSTEM "chapter02.xml"> <!-- Installing and Compiling Asterisk and Components -->
<!ENTITY chapter03 SYSTEM "chapter03.xml"> <!-- Basic configuration, sample.conf's -->
<!ENTITY chapter04 SYSTEM "chapter04.xml"> <!-- Scripting and AGI Extensions -->
-<!ENTITY chapter05 SYSTEM "chapter05.xml"> <!-- Connecting to Commong VoIP Providers -->
+<!ENTITY chapter05 SYSTEM "chapter05.xml"> <!-- Connecting to Common VoIP Providers -->
<!ENTITY chapter06 SYSTEM "chapter06.xml"> <!-- Advanced Asterisk Configuration -->
<!ENTITY chapter07 SYSTEM "chapter07.xml"> <!-- Common Issues / Troubleshooting -->
<!ENTITY chapter08 SYSTEM "chapter08.xml"> <!-- Creating Asterisk Applications in C -->
@@ -20,7 +20,7 @@
<!ENTITY appendix05 SYSTEM "appendix05.xml"> <!-- The Asterisk C API Reference -->
<!ENTITY appendix06 SYSTEM "appendix06.xml"> <!-- Other Open Source Telephony Systems -->
<!ENTITY glossary SYSTEM "glossary.xml"> <!-- Glossary of Terms -->
-<!ENTITY colophon SYSTEM "colophon.xml"> <!-- Colphone / Why we are doing this -->
+<!ENTITY colophon SYSTEM "colophon.xml"> <!-- Colophon / Why we are doing this -->
]>
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