Posted: Mon Dec 29, 2003 8:20 pm Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
Here are known problems with Chapter02;
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.
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.
--
Steven Critchfield <critch@basesys.com>
Posted: Mon Dec 29, 2003 8:47 pm Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
On Mon, 2003-12-29 at 13:20, Steven Critchfield wrote:
Quote:
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.
Quote:
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?)
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.
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 -->
]>
Posted: Mon Dec 29, 2003 11:40 pm Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
On Mon, 2003-12-29 at 16:28, Steven Critchfield wrote:
Quote:
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.
Thank you very much! I've applied your patch to the CVS. I'd be
interested in knowing more about using aspell and custom dictionaries...
if you could post a quick tutorial I'm sure we'd all appreciate it.
Posted: Tue Dec 30, 2003 3:23 am Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
On Mon, 2003-12-29 at 17:40, Jared Smith wrote:
Quote:
On Mon, 2003-12-29 at 16:28, Steven Critchfield wrote:
> 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.
>
Thank you very much! I've applied your patch to the CVS. I'd be
interested in knowing more about using aspell and custom dictionaries...
if you could post a quick tutorial I'm sure we'd all appreciate it.
aspell is pretty easy. The man page is pretty simple. The 2 basic
options to make this easy would be to use a person word list with -p,
and -c which hands aspell the file to check. Ideally the word list could
be in CVS so that it could be inspected from time to time for those of
us who don't spell so well and may have added a term that needs
correction. This also allows at some point for the spell check to happen
without any interaction unless there is real misspelling.
If need be, I could submit a start to the word list for the beginning.
--
Steven Critchfield <critch@basesys.com>
Posted: Thu Jan 01, 2004 3:53 pm Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
On Mon, 2003-12-29 at 21:23, Steven Critchfield wrote:
Quote:
On Mon, 2003-12-29 at 17:40, Jared Smith wrote:
> On Mon, 2003-12-29 at 16:28, Steven Critchfield wrote:
> > 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.
> >
>
> Thank you very much! I've applied your patch to the CVS. I'd be
> interested in knowing more about using aspell and custom dictionaries...
> if you could post a quick tutorial I'm sure we'd all appreciate it.
aspell is pretty easy. The man page is pretty simple. The 2 basic
options to make this easy would be to use a person word list with -p,
and -c which hands aspell the file to check. Ideally the word list could
be in CVS so that it could be inspected from time to time for those of
us who don't spell so well and may have added a term that needs
correction. This also allows at some point for the spell check to happen
without any interaction unless there is real misspelling.
If need be, I could submit a start to the word list for the beginning.
Since I now have CVS access, would it be a good idea for me to submit a
word list file for aspell, or is there a reason to use a different spell
checker?
I went through the current files and did some more spell checking. While
doing so, I created a word list file for terms, application names,
markup tags, names of people in the files, and a few other things that
occur often enough to be okay even if it is not proper spelling(eg.
newvar, and some directory path pieces.)
So with this word list file, I was able to do spell checking by typing
aspell -p ./word.list -c chapter01.xml
Luckily the word.list file is in plain text, and if you don't agree with
an inclusion, you just remove the entry from the list. If a entry needs
fixing, you just edit it.
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.
--
Steven Critchfield <critch@basesys.com>
On Thu, 2004-01-01 at 09:53, Steven Critchfield wrote:
Quote:
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.
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>
Posted: Fri Jan 02, 2004 1:35 am Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
On Thu, 2004-01-01 at 08:53, Steven Critchfield wrote:
Quote:
Since I now have CVS access, would it be a good idea for me to submit a
word list file for aspell, or is there a reason to use a different spell
checker?
I'm fine with aspell... as I understand it, it has the unique ability to
work around the XML tags, which is an added plus.
Quote:
I went through the current files and did some more spell checking. While
doing so, I created a word list file for terms, application names,
markup tags, names of people in the files, and a few other things that
occur often enough to be okay even if it is not proper spelling(eg.
newvar, and some directory path pieces.)
So with this word list file, I was able to do spell checking by typing
aspell -p ./word.list -c chapter01.xml
Well, that sounds easy enough... :-)
Quote:
Luckily the word.list file is in plain text, and if you don't agree with
an inclusion, you just remove the entry from the list. If a entry needs
fixing, you just edit it.
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.
It's been reviewed and checked into CVS... Please check the word list
into CVS and we'll make it a habit to run the spell-checker before
checking new stuff into CVS.
Posted: Fri Jan 02, 2004 6:56 am Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
On Thu, 2004-01-01 at 20:35, Jared Smith wrote:
Quote:
> Luckily the word.list file is in plain text, and if you don't agree with
> an inclusion, you just remove the entry from the list. If a entry needs
> fixing, you just edit it.
>
> 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.
It's been reviewed and checked into CVS... Please check the word list
into CVS and we'll make it a habit to run the spell-checker before
checking new stuff into CVS.
I'll even add it to my compile script (when I make one :)) before I run
my commit_htga script!
Posted: Fri Jan 02, 2004 7:05 am Post subject: [Asterisk-doc] bug report? Something that need to be fixed.
On Fri, 2004-01-02 at 00:56, Leif Madsen wrote:
Quote:
On Thu, 2004-01-01 at 20:35, Jared Smith wrote:
> > Luckily the word.list file is in plain text, and if you don't agree with
> > an inclusion, you just remove the entry from the list. If a entry needs
> > fixing, you just edit it.
> >
> > 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.
>
> It's been reviewed and checked into CVS... Please check the word list
> into CVS and we'll make it a habit to run the spell-checker before
> checking new stuff into CVS.
I'll even add it to my compile script (when I make one :)) before I run
my commit_htga script!
Thanks Steven!
Cool. That should help reduce later fixes to spelling.
--
Steven Critchfield <critch@basesys.com>
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