mirror of
https://github.com/asterisk/asterisk.git
synced 2025-09-05 20:20:07 +00:00
e43bc6634d
As partly documented in loader.c and include/asterisk/module.h, modules are now expected to return all of their methods and flags into a structure 'mod_data', and are normally loaded with RTLD_NOW | RTLD_LOCAL, so symbols are resolved immediately and conflicts should be less likely. Only in a small number of cases (res_*, typically) modules are loaded RTLD_GLOBAL, so they can export symbols. The core of the change is only the two files loader.c and include/asterisk/module.h, all the rest is simply adaptation of the existing modules to the new API, a rather mechanical (but believe me, time and finger-consuming!) process whose detail you can figure out by svn diff'ing any single module. Expect some minor compilation issue after this change, please report it on mantis http://bugs.digium.com/view.php?id=6968 so we collect all the feedback in one place. I am just sorry that this change missed SVN version number 20000! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
165 lines
4.1 KiB
C
165 lines
4.1 KiB
C
/*
|
|
* Asterisk -- An open source telephony toolkit.
|
|
*
|
|
* Copyright (C) 1999 - 2005, Digium, Inc.
|
|
*
|
|
* Mark Spencer <markster@digium.com>
|
|
*
|
|
* See http://www.asterisk.org for more information about
|
|
* the Asterisk project. Please do not directly contact
|
|
* any of the maintainers of this project for assistance;
|
|
* the project provides a web site, mailing lists and IRC
|
|
* channels for your use.
|
|
*
|
|
* This program is free software, distributed under the terms of
|
|
* the GNU General Public License Version 2. See the LICENSE file
|
|
* at the top of the source tree.
|
|
*/
|
|
|
|
/*! \file
|
|
*
|
|
* \brief Trivial application to playback a sound file
|
|
*
|
|
* \author Mark Spencer <markster@digium.com>
|
|
*
|
|
* \ingroup applications
|
|
*/
|
|
|
|
#include <string.h>
|
|
#include <stdlib.h>
|
|
#include <stdio.h>
|
|
|
|
#include "asterisk.h"
|
|
|
|
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
|
|
|
#include "asterisk/lock.h"
|
|
#include "asterisk/file.h"
|
|
#include "asterisk/logger.h"
|
|
#include "asterisk/channel.h"
|
|
#include "asterisk/pbx.h"
|
|
#include "asterisk/module.h"
|
|
#include "asterisk/translate.h"
|
|
#include "asterisk/utils.h"
|
|
#include "asterisk/options.h"
|
|
#include "asterisk/app.h"
|
|
|
|
static char *tdesc = "Sound File Playback Application";
|
|
|
|
static char *app = "Playback";
|
|
|
|
static char *synopsis = "Play a file";
|
|
|
|
static char *descrip =
|
|
" Playback(filename[&filename2...][|option]): Plays back given filenames (do not put\n"
|
|
"extension). Options may also be included following a pipe symbol. The 'skip'\n"
|
|
"option causes the playback of the message to be skipped if the channel\n"
|
|
"is not in the 'up' state (i.e. it hasn't been answered yet). If 'skip' is \n"
|
|
"specified, the application will return immediately should the channel not be\n"
|
|
"off hook. Otherwise, unless 'noanswer' is specified, the channel will\n"
|
|
"be answered before the sound is played. Not all channels support playing\n"
|
|
"messages while still on hook. If 'j' is specified, the application\n"
|
|
"will jump to priority n+101 if present when a file specified to be played\n"
|
|
"does not exist.\n"
|
|
"This application sets the following channel variable upon completion:\n"
|
|
" PLAYBACKSTATUS The status of the playback attempt as a text string, one of\n"
|
|
" SUCCESS | FAILED\n"
|
|
;
|
|
|
|
LOCAL_USER_DECL;
|
|
|
|
static int playback_exec(struct ast_channel *chan, void *data)
|
|
{
|
|
int res = 0;
|
|
struct localuser *u;
|
|
char *tmp;
|
|
int option_skip=0;
|
|
int option_noanswer = 0;
|
|
int priority_jump = 0;
|
|
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(filenames);
|
|
AST_APP_ARG(options);
|
|
);
|
|
|
|
if (ast_strlen_zero(data)) {
|
|
ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
|
|
return -1;
|
|
}
|
|
|
|
if (!(tmp = ast_strdupa(data)))
|
|
return -1;
|
|
|
|
LOCAL_USER_ADD(u);
|
|
AST_STANDARD_APP_ARGS(args, tmp);
|
|
|
|
if (args.options) {
|
|
if (strcasestr(args.options, "skip"))
|
|
option_skip = 1;
|
|
if (strcasestr(args.options, "noanswer"))
|
|
option_noanswer = 1;
|
|
if (strchr(args.options, 'j'))
|
|
priority_jump = 1;
|
|
}
|
|
|
|
if (chan->_state != AST_STATE_UP) {
|
|
if (option_skip) {
|
|
/* At the user's option, skip if the line is not up */
|
|
LOCAL_USER_REMOVE(u);
|
|
return 0;
|
|
} else if (!option_noanswer)
|
|
/* Otherwise answer unless we're supposed to send this while on-hook */
|
|
res = ast_answer(chan);
|
|
}
|
|
if (!res) {
|
|
int mres = 0;
|
|
char *front;
|
|
|
|
ast_stopstream(chan);
|
|
while (!res && (front = strsep(&tmp, "&"))) {
|
|
res = ast_streamfile(chan, front, chan->language);
|
|
if (!res) {
|
|
res = ast_waitstream(chan, "");
|
|
ast_stopstream(chan);
|
|
} else {
|
|
ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
|
|
if (priority_jump || ast_opt_priority_jumping)
|
|
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
|
|
res = 0;
|
|
mres = 1;
|
|
}
|
|
}
|
|
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", mres ? "FAILED" : "SUCCESS");
|
|
}
|
|
LOCAL_USER_REMOVE(u);
|
|
return res;
|
|
}
|
|
|
|
static int unload_module(void *mod)
|
|
{
|
|
int res;
|
|
|
|
res = ast_unregister_application(app);
|
|
|
|
STANDARD_HANGUP_LOCALUSERS;
|
|
|
|
return res;
|
|
}
|
|
|
|
static int load_module(void *mod)
|
|
{
|
|
return ast_register_application(app, playback_exec, synopsis, descrip);
|
|
}
|
|
|
|
static const char *description(void)
|
|
{
|
|
return tdesc;
|
|
}
|
|
|
|
static const char *key(void)
|
|
{
|
|
return ASTERISK_GPL_KEY;
|
|
}
|
|
|
|
STD_MOD1;
|