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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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51
apps/app_transfer.c
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@@ -36,23 +36,44 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/app.h"
#include "asterisk/channel.h"
/*** DOCUMENTATION
<application name="Transfer" language="en_US">
<synopsis>
Transfer caller to remote extension.
</synopsis>
<syntax>
<parameter name="dest" required="true" argsep="/">
<argument name="Tech" />
<argument name="destination" required="true" />
</parameter>
</syntax>
<description>
<para>Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
an incoming call with the same channel technology will be transfered.
Note that for SIP, if you transfer before call is setup, a 302 redirect
SIP message will be returned to the caller.</para>
<para>The result of the application will be reported in the <variable>TRANSFERSTATUS</variable>
channel variable:</para>
<variablelist>
<variable name="TRANSFERSTATUS">
<value name="SUCCESS">
Transfer succeeded.
</value>
<value name="FAILURE">
Transfer failed.
</value>
<value name="UNSUPPORTED">
Transfer unsupported by channel driver.
</value>
</variable>
</variablelist>
</description>
</application>
***/
static const char *app = "Transfer";
static const char *synopsis = "Transfer caller to remote extension";
static const char *descrip =
" Transfer([Tech/]dest): Requests the remote caller be transferred\n"
"to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only\n"
"an incoming call with the same channel technology will be transfered.\n"
"Note that for SIP, if you transfer before call is setup, a 302 redirect\n"
"SIP message will be returned to the caller.\n"
"\nThe result of the application will be reported in the TRANSFERSTATUS\n"
"channel variable:\n"
" SUCCESS Transfer succeeded\n"
" FAILURE Transfer failed\n"
" UNSUPPORTED Transfer unsupported by channel driver\n";
static int transfer_exec(struct ast_channel *chan, void *data)
{
int res;
@@ -115,7 +136,7 @@ static int unload_module(void)
static int load_module(void)
{
return ast_register_application(app, transfer_exec, synopsis, descrip);
return ast_register_application_xml(app, transfer_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Transfers a caller to another extension");