[asterisk-commits] mvanbaak: branch group/appdocsxml r143133 - /team/group/appdocsxml/apps/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Sep 15 13:34:21 CDT 2008
Author: mvanbaak
Date: Mon Sep 15 13:34:20 2008
New Revision: 143133
URL: http://svn.digium.com/view/asterisk?view=rev&rev=143133
Log:
some more XML cleanup.
Again, passes xmllint validation
Modified:
team/group/appdocsxml/apps/app_jack.c
team/group/appdocsxml/apps/app_page.c
Modified: team/group/appdocsxml/apps/app_jack.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_jack.c?view=diff&rev=143133&r1=143132&r2=143133
==============================================================================
--- team/group/appdocsxml/apps/app_jack.c (original)
+++ team/group/appdocsxml/apps/app_jack.c Mon Sep 15 13:34:20 2008
@@ -78,47 +78,41 @@
<synopsis>
Jack Audio Connection Kit
</synopsis>
+ <syntax>
+ <parameter name="options" required="false">
+ <optionlist>
+ <option name="s">
+ <argument name="name" required="true">
+ <para>Connect to the specified jack server name</para>
+ </argument>
+ </option>
+ <option name="i">
+ <argument name="name" required="true">
+ <para>Connect the output port that gets created to the specified jack input port</para>
+ </argument>
+ </option>
+ <option name="o">
+ <argument name="name" required="true">
+ <para>Connect the input port that gets created to the specified jack output port</para>
+ </argument>
+ </option>
+ <option name="c">
+ <argument name="name" required="true">
+ <para>By default, Asterisk will use the channel name for the jack client name.</para>
+ <para>Use this option to specify a custom client name.</para>
+ </argument>
+ </option>
+ </optionlist>
+ </parameter>
+ </syntax>
<description>
- When executing this application, two jack ports will be created;
+ <para>When executing this application, two jack ports will be created;
one input and one output. Other applications can be hooked up to
- these ports to access audio coming from, or being send to the channel.
+ these ports to access audio coming from, or being send to the channel.</para>
</description>
- <option name="s">
- <variable name="name" required="true">
- Connect to the specified jack server name
- </variable>
- </option>
- <option name="i">
- <variable name="name" required="true">
- Connect the output port that gets created to the specified jack input port
- </variable>
- </option>
- <option name="o">
- <variable name="name" required="true">
- Connect the input port that gets created to the specified jack output port
- </variable>
- </option>
- <option name="c">
- <variable name="name" required="true">
- By default, Asterisk will use the channel name for the jack client name.
- Use this option to specify a custom client name.
- </variable>
- </option>
</application>
***/
static char *jack_app = "JACK";
-/*
-static char *jack_synopsis =
-"JACK (Jack Audio Connection Kit) Application";
-static char *jack_desc =
-"JACK([options])\n"
-" When this application is executed, two jack ports will be created; one input\n"
-"and one output. Other applications can be hooked up to these ports to access\n"
-"the audio coming from, or being sent to the channel.\n"
-" Valid options:\n"
-COMMON_OPTIONS
-"";
-*/
struct jack_data {
AST_DECLARE_STRING_FIELDS(
Modified: team/group/appdocsxml/apps/app_page.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_page.c?view=diff&rev=143133&r1=143132&r2=143133
==============================================================================
--- team/group/appdocsxml/apps/app_page.c (original)
+++ team/group/appdocsxml/apps/app_page.c Mon Sep 15 13:34:20 2008
@@ -49,50 +49,39 @@
<synopsis>
Page series of phones
</synopsis>
+ <syntax>
+ <parameter name="Technology/Resource" required="true" argsep="&">
+ <argument name="Technology2/Resource2" multiple="true">
+ <para>Optional extra devices to dial inparallel</para>
+ <para>If you need more then one enter them asTechnology2/Resource2&Technology3/Resourse3&.....</para>
+ </argument>
+ <para>Specification of the device(s) to dial.
+ These must be in the format of <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
+ represents a particular channel driver, and <replaceable>Resource</replaceable>represents a resource available to that particular channel driver.</para>
+ </parameter>
+ <parameter name="d" required="false">
+ <para>Full duplex audio</para>
+ </parameter>
+ <parameter name="q" required="false">
+ <para>Quiet, do not play beep to caller</para>
+ </parameter>
+ <parameter name="r" required="false">
+ <para>record the page into file (meetme option <literal>r</literal>)</para>
+ </parameter>
+ <parameter name="s" required="false">
+ <para>Only dial channel if devicestate says its <literal>notinuse</literal></para>
+ </parameter>
+ </syntax>
<description>
- Places outbound calls to the given technology / resource and dumps
+ <para>Places outbound calls to the given <replaceable>technology</replaceable> / <replaceable>resource</replaceable> and dumps
them into a conference bridge as muted participants. The original
caller is dumped into conference as a speaker and the room is
- destroyed when the original callers leaves.
+ destroyed when the original callers leaves.</para>
</description>
- <option name="Technology/Resource" required="true" argsep="&">
- <argument name="Technology2/Resource2">
- Optional extra 'devices' to dial.
- If you need more then one enter them like this:
- Technology2/Resource2&Technology3/Resourse3&.....
- </argument>
- Device to dial
- </option>
- <option name="d">
- Full duplex audio
- </option>
- <option name="q">
- Quiet, do not play beep to caller
- </option>
- <option name="r">
- record the page into file (meetme option 'r')
- </option>
- <option name="s">
- Only dial channel if devicestate says its 'notinuse'
- </option>
</application>
***/
static const char *app_page= "Page";
-/*
-static const char *page_synopsis = "Pages phones";
-
-static const char *page_descrip =
-"Page(Technology/Resource&Technology2/Resource2[,options])\n"
-" Places outbound calls to the given technology / resource and dumps\n"
-"them into a conference bridge as muted participants. The original\n"
-"caller is dumped into the conference as a speaker and the room is\n"
-"destroyed when the original caller leaves. Valid options are:\n"
-" d - full duplex audio\n"
-" q - quiet, do not play beep to caller\n"
-" r - record the page into a file (see 'r' for app_meetme)\n"
-" s - only dial channel if devicestate says it is not in use\n";
-*/
enum {
PAGE_DUPLEX = (1 << 0),
PAGE_QUIET = (1 << 1),
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