asterisk/apps/app_playback.c

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/*
* 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
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
/* This file provides config-file based 'say' functions, and implenents
* some CLI commands.
*/
#include "asterisk/say.h" /*!< provides config-file based 'say' functions */
#include "asterisk/cli.h"
/*** DOCUMENTATION
<application name="Playback" language="en_US">
<synopsis>
Play a file.
</synopsis>
<syntax>
<parameter name="filenames" required="true" argsep="&amp;">
<argument name="filename" required="true" />
<argument name="filename2" multiple="true" />
</parameter>
<parameter name="options">
<para>Comma separated list of options</para>
<optionlist>
<option name="skip">
<para>Do not play if not answered</para>
</option>
<option name="noanswer">
<para>Playback without answering, otherwise the channel will
be answered before the sound is played.</para>
<note><para>Not all channel types support playing messages while still on hook.</para></note>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Plays back given filenames (do not put extension of wav/alaw etc).
The playback command answer the channel if no options are specified.
If the file is non-existant it will fail</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="PLAYBACKSTATUS">
<para>The status of the playback attempt as a text string.</para>
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
</variablelist>
<para>See Also: Background (application) -- for playing sound files that are interruptible</para>
<para>WaitExten (application) -- wait for digits from caller, optionally play music on hold</para>
</description>
</application>
***/
static char *app = "Playback";
static struct ast_config *say_cfg = NULL;
/*! \brief save the say' api calls.
* The first entry is NULL if we have the standard source,
* otherwise we are sourcing from here.
* 'say load [new|old]' will enable the new or old method, or report status
*/
static const void *say_api_buf[40];
static const char * const say_old = "old";
static const char * const say_new = "new";
static void save_say_mode(const void *arg)
{
int i = 0;
say_api_buf[i++] = arg;
say_api_buf[i++] = ast_say_number_full;
say_api_buf[i++] = ast_say_enumeration_full;
say_api_buf[i++] = ast_say_digit_str_full;
say_api_buf[i++] = ast_say_character_str_full;
say_api_buf[i++] = ast_say_phonetic_str_full;
say_api_buf[i++] = ast_say_datetime;
say_api_buf[i++] = ast_say_time;
say_api_buf[i++] = ast_say_date;
say_api_buf[i++] = ast_say_datetime_from_now;
say_api_buf[i++] = ast_say_date_with_format;
}
static void restore_say_mode(void *arg)
{
int i = 0;
say_api_buf[i++] = arg;
ast_say_number_full = say_api_buf[i++];
ast_say_enumeration_full = say_api_buf[i++];
ast_say_digit_str_full = say_api_buf[i++];
ast_say_character_str_full = say_api_buf[i++];
ast_say_phonetic_str_full = say_api_buf[i++];
ast_say_datetime = say_api_buf[i++];
ast_say_time = say_api_buf[i++];
ast_say_date = say_api_buf[i++];
ast_say_datetime_from_now = say_api_buf[i++];
ast_say_date_with_format = say_api_buf[i++];
}
/*! \brief
* Typical 'say' arguments in addition to the date or number or string
* to say. We do not include 'options' because they may be different
* in recursive calls, and so they are better left as an external
* parameter.
*/
typedef struct {
struct ast_channel *chan;
const char *ints;
const char *language;
int audiofd;
int ctrlfd;
} say_args_t;
static int s_streamwait3(const say_args_t *a, const char *fn)
{
int res = ast_streamfile(a->chan, fn, a->language);
if (res) {
ast_log(LOG_WARNING, "Unable to play message %s\n", fn);
return res;
}
res = (a->audiofd > -1 && a->ctrlfd > -1) ?
ast_waitstream_full(a->chan, a->ints, a->audiofd, a->ctrlfd) :
ast_waitstream(a->chan, a->ints);
ast_stopstream(a->chan);
return res;
}
/*! \brief
* the string is 'prefix:data' or prefix:fmt:data'
* with ':' being invalid in strings.
*/
static int do_say(say_args_t *a, const char *s, const char *options, int depth)
{
struct ast_variable *v;
char *lang, *x, *rule = NULL;
int ret = 0;
struct varshead head = { .first = NULL, .last = NULL };
struct ast_var_t *n;
ast_debug(2, "string <%s> depth <%d>\n", s, depth);
if (depth++ > 10) {
ast_log(LOG_WARNING, "recursion too deep, exiting\n");
return -1;
} else if (!say_cfg) {
ast_log(LOG_WARNING, "no say.conf, cannot spell '%s'\n", s);
return -1;
}
/* scan languages same as in file.c */
if (a->language == NULL)
a->language = "en"; /* default */
ast_debug(2, "try <%s> in <%s>\n", s, a->language);
lang = ast_strdupa(a->language);
for (;;) {
for (v = ast_variable_browse(say_cfg, lang); v ; v = v->next) {
if (ast_extension_match(v->name, s)) {
rule = ast_strdupa(v->value);
break;
}
}
if (rule)
break;
if ( (x = strchr(lang, '_')) )
*x = '\0'; /* try without suffix */
else if (strcmp(lang, "en"))
lang = "en"; /* last resort, try 'en' if not done yet */
else
break;
}
if (!rule)
return 0;
/* skip up to two prefixes to get the value */
if ( (x = strchr(s, ':')) )
s = x + 1;
if ( (x = strchr(s, ':')) )
s = x + 1;
ast_debug(2, "value is <%s>\n", s);
n = ast_var_assign("SAY", s);
AST_LIST_INSERT_HEAD(&head, n, entries);
/* scan the body, one piece at a time */
while ( !ret && (x = strsep(&rule, ",")) ) { /* exit on key */
char fn[128];
const char *p, *fmt, *data; /* format and data pointers */
/* prepare a decent file name */
x = ast_skip_blanks(x);
ast_trim_blanks(x);
/* replace variables */
pbx_substitute_variables_varshead(&head, x, fn, sizeof(fn));
ast_debug(2, "doing [%s]\n", fn);
/* locate prefix and data, if any */
fmt = strchr(fn, ':');
if (!fmt || fmt == fn) { /* regular filename */
ret = s_streamwait3(a, fn);
continue;
}
fmt++;
data = strchr(fmt, ':'); /* colon before data */
if (!data || data == fmt) { /* simple prefix-fmt */
ret = do_say(a, fn, options, depth);
continue;
}
/* prefix:fmt:data */
for (p = fmt; p < data && ret <= 0; p++) {
char fn2[sizeof(fn)];
if (*p == ' ' || *p == '\t') /* skip blanks */
continue;
if (*p == '\'') {/* file name - we trim them */
char *y;
strcpy(fn2, ast_skip_blanks(p+1)); /* make a full copy */
y = strchr(fn2, '\'');
if (!y) {
p = data; /* invalid. prepare to end */
break;
}
*y = '\0';
ast_trim_blanks(fn2);
p = strchr(p+1, '\'');
ret = s_streamwait3(a, fn2);
} else {
int l = fmt-fn;
strcpy(fn2, fn); /* copy everything */
/* after prefix, append the format */
fn2[l++] = *p;
strcpy(fn2 + l, data);
ret = do_say(a, fn2, options, depth);
}
if (ret) {
break;
}
}
}
ast_var_delete(n);
return ret;
}
static int say_full(struct ast_channel *chan, const char *string,
const char *ints, const char *lang, const char *options,
int audiofd, int ctrlfd)
{
say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
return do_say(&a, string, options, 0);
}
static int say_number_full(struct ast_channel *chan, int num,
const char *ints, const char *lang, const char *options,
int audiofd, int ctrlfd)
{
char buf[64];
say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
snprintf(buf, sizeof(buf), "num:%d", num);
return do_say(&a, buf, options, 0);
}
static int say_enumeration_full(struct ast_channel *chan, int num,
const char *ints, const char *lang, const char *options,
int audiofd, int ctrlfd)
{
char buf[64];
say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
snprintf(buf, sizeof(buf), "enum:%d", num);
return do_say(&a, buf, options, 0);
}
static int say_date_generic(struct ast_channel *chan, time_t t,
const char *ints, const char *lang, const char *format, const char *timezonename, const char *prefix)
{
char buf[128];
struct ast_tm tm;
struct timeval when = { t, 0 };
say_args_t a = { chan, ints, lang, -1, -1 };
if (format == NULL)
format = "";
ast_localtime(&when, &tm, NULL);
snprintf(buf, sizeof(buf), "%s:%s:%04d%02d%02d%02d%02d.%02d-%d-%3d",
prefix,
format,
tm.tm_year+1900,
tm.tm_mon+1,
tm.tm_mday,
tm.tm_hour,
tm.tm_min,
tm.tm_sec,
tm.tm_wday,
tm.tm_yday);
return do_say(&a, buf, NULL, 0);
}
static int say_date_with_format(struct ast_channel *chan, time_t t,
const char *ints, const char *lang, const char *format, const char *timezonename)
{
return say_date_generic(chan, t, ints, lang, format, timezonename, "datetime");
}
static int say_date(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
{
return say_date_generic(chan, t, ints, lang, "", NULL, "date");
}
static int say_time(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
{
return say_date_generic(chan, t, ints, lang, "", NULL, "time");
}
static int say_datetime(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
{
return say_date_generic(chan, t, ints, lang, "", NULL, "datetime");
}
/*! \brief
* remap the 'say' functions to use those in this file
*/
static int say_init_mode(const char *mode) {
if (!strcmp(mode, say_new)) {
if (say_cfg == NULL) {
ast_log(LOG_ERROR, "There is no say.conf file to use new mode\n");
return -1;
}
save_say_mode(say_new);
ast_say_number_full = say_number_full;
ast_say_enumeration_full = say_enumeration_full;
#if 0
/*! \todo XXX
These functions doesn't exist.
say.conf.sample indicates this is working...
*/
ast_say_digits_full = say_digits_full;
ast_say_digit_str_full = say_digit_str_full;
ast_say_character_str_full = say_character_str_full;
ast_say_phonetic_str_full = say_phonetic_str_full;
ast_say_datetime_from_now = say_datetime_from_now;
#endif
ast_say_datetime = say_datetime;
ast_say_time = say_time;
ast_say_date = say_date;
ast_say_date_with_format = say_date_with_format;
} else if (!strcmp(mode, say_old) && say_api_buf[0] == say_new) {
restore_say_mode(NULL);
} else if (strcmp(mode, say_old)) {
ast_log(LOG_WARNING, "unrecognized mode %s\n", mode);
return -1;
}
return 0;
}
static char *__say_cli_init(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
const char *old_mode = say_api_buf[0] ? say_new : say_old;
const char *mode;
switch (cmd) {
case CLI_INIT:
e->command = "say load [new|old]";
e->usage =
"Usage: say load [new|old]\n"
" say load\n"
" Report status of current say mode\n"
" say load new\n"
" Set say method, configured in say.conf\n"
" say load old\n"
" Set old say method, coded in asterisk core\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc == 2) {
ast_cli(a->fd, "say mode is [%s]\n", old_mode);
return CLI_SUCCESS;
} else if (a->argc != e->args)
return CLI_SHOWUSAGE;
mode = a->argv[2];
if (!strcmp(mode, old_mode))
ast_cli(a->fd, "say mode is %s already\n", mode);
else
if (say_init_mode(mode) == 0)
ast_cli(a->fd, "setting say mode from %s to %s\n", old_mode, mode);
return CLI_SUCCESS;
}
static struct ast_cli_entry cli_playback[] = {
AST_CLI_DEFINE(__say_cli_init, "Set or show the say mode"),
};
static int playback_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
int mres = 0;
char *tmp;
int option_skip=0;
int option_say=0;
int option_noanswer = 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;
}
tmp = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, tmp);
if (args.options) {
if (strcasestr(args.options, "skip"))
option_skip = 1;
if (strcasestr(args.options, "say"))
option_say = 1;
if (strcasestr(args.options, "noanswer"))
option_noanswer = 1;
}
if (chan->_state != AST_STATE_UP) {
if (option_skip) {
/* At the user's option, skip if the line is not up */
goto done;
Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04 14:02:34 +00:00
} else if (!option_noanswer) {
/* Otherwise answer unless we're supposed to send this while on-hook */
res = ast_answer(chan);
Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04 14:02:34 +00:00
}
}
if (!res) {
char *back = args.filenames;
char *front;
ast_stopstream(chan);
while (!res && (front = strsep(&back, "&"))) {
if (option_say)
res = say_full(chan, front, "", chan->language, NULL, -1, -1);
else
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);
res = 0;
mres = 1;
}
}
}
done:
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", mres ? "FAILED" : "SUCCESS");
return res;
}
static int reload(void)
{
struct ast_variable *v;
struct ast_flags config_flags = { CONFIG_FLAG_FILEUNCHANGED };
struct ast_config *newcfg;
if ((newcfg = ast_config_load("say.conf", config_flags)) == CONFIG_STATUS_FILEUNCHANGED) {
return 0;
} else if (newcfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file say.conf is in an invalid format. Aborting.\n");
return 0;
}
if (say_cfg) {
ast_config_destroy(say_cfg);
ast_log(LOG_NOTICE, "Reloading say.conf\n");
say_cfg = newcfg;
}
if (say_cfg) {
for (v = ast_variable_browse(say_cfg, "general"); v ; v = v->next) {
if (ast_extension_match(v->name, "mode")) {
say_init_mode(v->value);
break;
}
}
}
/*! \todo
* XXX here we should sort rules according to the same order
* we have in pbx.c so we have the same matching behaviour.
*/
return 0;
}
static int unload_module(void)
{
int res;
res = ast_unregister_application(app);
ast_cli_unregister_multiple(cli_playback, ARRAY_LEN(cli_playback));
if (say_cfg)
ast_config_destroy(say_cfg);
return res;
}
static int load_module(void)
{
struct ast_variable *v;
struct ast_flags config_flags = { 0 };
say_cfg = ast_config_load("say.conf", config_flags);
if (say_cfg && say_cfg != CONFIG_STATUS_FILEINVALID) {
for (v = ast_variable_browse(say_cfg, "general"); v ; v = v->next) {
if (ast_extension_match(v->name, "mode")) {
say_init_mode(v->value);
break;
}
}
}
ast_cli_register_multiple(cli_playback, ARRAY_LEN(cli_playback));
return ast_register_application_xml(app, playback_exec);
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Sound File Playback Application",
.load = load_module,
.unload = unload_module,
.reload = reload,
);