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Merge changes from team/group/appdocsxml

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This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-11-01 21:10:07 +00:00
parent 1fef0f63bb
commit 5b168ee34b
111 changed files with 8063 additions and 2478 deletions
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67
apps/app_page.c
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@@ -44,24 +44,57 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/devicestate.h"
#include "asterisk/dial.h"
/*** DOCUMENTATION
<application name="Page" language="en_US">
<synopsis>
Page series of phones
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" argsep="&amp;">
<argument name="Technology/Resource" required="true">
<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>
</argument>
<argument name="Technology2/Resource2" multiple="true">
<para>Optional extra devices to dial inparallel</para>
<para>If you need more then one enter them as Technology2/Resource2&amp;
Technology3/Resourse3&amp;.....</para>
</argument>
</parameter>
<parameter name="options">
<optionlist>
<option name="d">
<para>Full duplex audio</para>
</option>
<option name="q">
<para>Quiet, do not play beep to caller</para>
</option>
<option name="r">
<para>Record the page into a file (meetme option <literal>r</literal>)</para>
</option>
<option name="s">
<para>Only dial channel if devicestate says its <literal>notinuse</literal></para>
</option>
</optionlist>
</parameter>
<parameter name="timeout">
<para>Specify the length of time that the system will attempt to connect a call.
After this duration, any intercom calls that have not been answered will be hung up by the
system.</para>
</parameter>
</syntax>
<description>
<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 the conference as a speaker and the room is
destroyed when the original callers leaves.</para>
</description>
</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][,timeout])\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"
"The timeout parameter specifies the length of time that the system\n"
"will attempt to connect a call. After this duration, any intercom\n"
"calls that have not been answered will be hung up by the system.\n";
enum {
PAGE_DUPLEX = (1 << 0),
PAGE_QUIET = (1 << 1),
@@ -216,7 +249,7 @@ static int unload_module(void)
static int load_module(void)
{
return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
return ast_register_application_xml(app_page, page_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");