Author: mjordan
Date: Fri Aug 9 08:58:02 2013
New Revision: 396490
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=396490
Log:
Update documentation for ConfBridge with some additional markup
Add some additional markup for items that needed it, e.g.,
replaceable tags, literal tags, etc.
Modified:
trunk/apps/confbridge/conf_config_parser.c
Modified: trunk/apps/confbridge/conf_config_parser.c
URL:
http://svnview.digium.com/svn/asterisk/trunk/apps/confbridge/conf_config_parser.c?view=diff&rev=396490&r1=396489&r2=396490
==============================================================================
--- trunk/apps/confbridge/conf_config_parser.c (original)
+++ trunk/apps/confbridge/conf_config_parser.c Fri Aug 9 08:58:02 2013
@@ -123,8 +123,8 @@
<synopsis>Apply a denoise filter to the
audio before mixing</synopsis>
<description><para>Sets whether or not
a denoise filter should be applied
to the audio before mixing or not. Off
by default. Requires
- codec_speex to be built and installed.
Do not confuse this option
- with drop_silence. Denoise is useful
if there is a lot of background
+ <literal>codec_speex</literal> to be
built and installed. Do not confuse this option
+ with
<replaceable>drop_silence</replaceable>. Denoise is useful if there is a lot
of background
noise for a user as it attempts to
remove the noise while preserving
the speech. This option does NOT
remove silence from being mixed into
the conference and does come at the
cost of a slight performance hit.
@@ -158,7 +158,7 @@
during mid sentence.
</para>
<para>
- 2. The drop_silence option
depends on this value to
+ 2. The
<replaceable>drop_silence</replaceable> option depends on this value to
determine when the user's audio
should begin to be
dropped from the conference
bridge after the user
stops talking. If this value
is set too low the user's
@@ -200,7 +200,7 @@
room noise.
</para>
<para>
- 3. The drop_silence option
depends on this value to determine
+ 3. The
<replaceable>drop_silence</replaceable> option depends on this value to
determine
when the user's audio should be
mixed into the bridge
after periods of silence. If
this value is too loose
the beginning of a user's
speech will get cut off as they
@@ -274,15 +274,15 @@
Records the conference call
starting when the first user
enters the room, and ending
when the last user exits the room.
The default recorded filename is
- <filename>'confbridge-${name of
conference bridge}-${start time}.wav</filename>
+ <filename>'confbridge-${name of
conference bridge}-${start time}.wav'</filename>
and the default format is 8khz
slinear. This file will be
- located in the configured
monitoring directory in asterisk.conf.
+ located in the configured
monitoring directory in <filename>asterisk.conf</filename>.
</para></description>
</configOption>
<configOption name="record_file"
default="confbridge-${name of conference bridge}-${start time}.wav">
<synopsis>The filename of the
conference recording</synopsis>
<description><para>
- When record_conference is set
to yes, the specific name of the
+ When
<replaceable>record_conference</replaceable> is set to yes, the specific name
of the
record file can be set using
this option. Note that since multiple
conferences may use the same
bridge profile, this may cause issues
depending on the configuration.
It is recommended to only use this
@@ -295,9 +295,9 @@
<configOption name="record_file_append"
default="yes">
<synopsis>Append record file when
starting/stopping on same conference recording</synopsis>
<description><para>
- When record_file_append is set
to yes, stopping and starting recording on a
+ When
<replaceable>record_file_append</replaceable> is set to yes, stopping and
starting recording on a
conference adds the new portion
to end of current record_file. When this is
- set to no, a new record_file is
generated every time you start then stop recording
+ set to no, a new
<replaceable>record_file</replaceable> is generated every time you start then
stop recording
on a conference.
</para></description>
</configOption>
@@ -306,7 +306,7 @@
<description><para>
Sets how confbridge handles
video distribution to the conference participants.
Note that participants wanting
to view and be the source of a video feed
- _MUST_ be sharing the same
video codec. Also, using video in conjunction with
+ <emphasis>MUST</emphasis> be
sharing the same video codec. Also, using video in conjunction with
with the jitterbuffer currently
results in the audio being slightly out of sync
with the video. This is a
result of the jitterbuffer only working on the audio
stream. It is recommended to
disable the jitterbuffer when video is used.</para>
@@ -395,7 +395,7 @@
<configObject name="menu">
<synopsis>A conference user menu</synopsis>
<description>
- <para>Conference users, as defined by a
<literal>conf_user</literal>,
+ <para>Conference users, as defined by a
<replaceable>conf_user</replaceable>,
can have a DTMF menu assigned to their
profile when they enter the
<literal>ConfBridge</literal>
application.</para>
</description>
@@ -412,7 +412,7 @@
</configOption>
<configOption name="^[0-9A-D*#]+$">
<synopsis>DTMF sequences to assign
various confbridge actions to</synopsis>
- <description><para>--- ConfBridge Menu
Options ---</para>
+ <description>
<para>The ConfBridge application also
has the ability to apply custom DTMF menus to
each channel using the application.
Like the User and Bridge profiles a menu
is passed in to ConfBridge as an
argument in the dialplan.</para>
--
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