Files
asterisk/apps/app_record.c
George Joseph ce550fc1b0 docs: Add version information to application and function XML elements
* Do a git blame on the embedded XML application or function element.

* From the commit hash, grab the summary line.

* Do a git log --grep <summary> to find the cherry-pick commits in all
  branches that match.

* Do a git patch-id to ensure the commits are all related and didn't get
  a false match on the summary.

* Do a git tag --contains <commit> to find the tags that contain each
  commit.

* Weed out all tags not ..0.

* Sort and discard any .0.0 and following tags where the commit
  appeared in an earlier branch.

* The result is a single tag for each branch where the application or function
  was defined.

The applications and functions defined in the following files were done by
hand because the XML was extracted from the C source file relatively recently.
* channels/pjsip/dialplan_functions_doc.xml
* main/logger_doc.xml
* main/manager_doc.xml
* res/res_geolocation/geoloc_doc.xml
* res/res_stir_shaken/stir_shaken_doc.xml

(cherry picked from commit f6a193e87e)
2025-01-23 18:42:29 +00:00

545 lines
15 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Matthew Fredrickson <creslin@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 record a sound file
*
* \author Matthew Fredrickson <creslin@digium.com>
*
* \ingroup applications
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/file.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/dsp.h" /* use dsp routines for silence detection */
#include "asterisk/format_cache.h"
#include "asterisk/paths.h"
/*** DOCUMENTATION
<application name="Record" language="en_US">
<since><version>1.6.2.0</version></since>
<synopsis>
Record to a file.
</synopsis>
<syntax>
<parameter name="filename" required="true" argsep=".">
<argument name="filename" required="true" />
<argument name="format" required="true">
<para>Is the format of the file type to be recorded (wav, gsm, etc).</para>
</argument>
</parameter>
<parameter name="silence">
<para>Is the number of seconds of silence to allow before returning.</para>
</parameter>
<parameter name="maxduration">
<para>Is the maximum recording duration in seconds. If missing
or 0 there is no maximum.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Append to existing recording rather than replacing.</para>
</option>
<option name="n">
<para>Do not answer, but record anyway if line not yet answered.</para>
</option>
<option name="o">
<para>Exit when 0 is pressed, setting the variable <variable>RECORD_STATUS</variable>
to <literal>OPERATOR</literal> instead of <literal>DTMF</literal></para>
</option>
<option name="q">
<para>quiet (do not play a beep tone).</para>
</option>
<option name="s">
<para>skip recording if the line is not yet answered.</para>
</option>
<option name="t">
<para>use alternate '*' terminator key (DTMF) instead of default '#'</para>
</option>
<option name="u">
<para>Don't truncate recorded silence.</para>
</option>
<option name="x">
<para>Ignore all terminator keys (DTMF) and keep recording until hangup.</para>
</option>
<option name="k">
<para>Keep recorded file upon hangup.</para>
</option>
<option name="y">
<para>Terminate recording if *any* DTMF digit is received.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>If filename contains <literal>%d</literal>, these characters will be replaced with a number
incremented by one each time the file is recorded.
Use <astcli>core show file formats</astcli> to see the available formats on your system
User can press <literal>#</literal> to terminate the recording and continue to the next priority.
If the user hangs up during a recording, all data will be lost and the application will terminate.</para>
<variablelist>
<variable name="RECORDED_FILE">
<para>Will be set to the final filename of the recording, without an extension.</para>
</variable>
<variable name="RECORD_STATUS">
<para>This is the final status of the command</para>
<value name="DTMF">A terminating DTMF was received ('#' or '*', depending upon option 't')</value>
<value name="SILENCE">The maximum silence occurred in the recording.</value>
<value name="SKIP">The line was not yet answered and the 's' option was specified.</value>
<value name="TIMEOUT">The maximum length was reached.</value>
<value name="HANGUP">The channel was hung up.</value>
<value name="ERROR">An unrecoverable error occurred, which resulted in a WARNING to the logs.</value>
</variable>
</variablelist>
</description>
</application>
***/
#define OPERATOR_KEY '0'
static char *app = "Record";
enum {
OPTION_APPEND = (1 << 0),
OPTION_NOANSWER = (1 << 1),
OPTION_QUIET = (1 << 2),
OPTION_SKIP = (1 << 3),
OPTION_STAR_TERMINATE = (1 << 4),
OPTION_IGNORE_TERMINATE = (1 << 5),
OPTION_KEEP = (1 << 6),
OPTION_ANY_TERMINATE = (1 << 7),
OPTION_OPERATOR_EXIT = (1 << 8),
OPTION_NO_TRUNCATE = (1 << 9),
};
enum dtmf_response {
RESPONSE_NO_MATCH = 0,
RESPONSE_OPERATOR,
RESPONSE_DTMF,
};
AST_APP_OPTIONS(app_opts,{
AST_APP_OPTION('a', OPTION_APPEND),
AST_APP_OPTION('k', OPTION_KEEP),
AST_APP_OPTION('n', OPTION_NOANSWER),
AST_APP_OPTION('o', OPTION_OPERATOR_EXIT),
AST_APP_OPTION('q', OPTION_QUIET),
AST_APP_OPTION('s', OPTION_SKIP),
AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
AST_APP_OPTION('u', OPTION_NO_TRUNCATE),
AST_APP_OPTION('y', OPTION_ANY_TERMINATE),
AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
});
/*!
* \internal
* \brief Used to determine what action to take when DTMF is received while recording
* \since 13.0.0
*
* \param chan channel being recorded
* \param flags option flags in use by the record application
* \param dtmf_integer the integer value of the DTMF key received
* \param terminator key currently set to be pressed for normal termination
*
* \returns One of enum dtmf_response
*/
static enum dtmf_response record_dtmf_response(struct ast_channel *chan,
struct ast_flags *flags, int dtmf_integer, int terminator)
{
if ((dtmf_integer == OPERATOR_KEY) &&
(ast_test_flag(flags, OPTION_OPERATOR_EXIT))) {
return RESPONSE_OPERATOR;
}
if ((dtmf_integer == terminator) ||
(ast_test_flag(flags, OPTION_ANY_TERMINATE))) {
return RESPONSE_DTMF;
}
return RESPONSE_NO_MATCH;
}
static int create_destination_directory(const char *path)
{
int res;
char directory[PATH_MAX], *file_sep;
if (!(file_sep = strrchr(path, '/'))) {
/* No directory to create */
return 0;
}
/* Overwrite temporarily */
*file_sep = '\0';
/* Absolute path? */
if (path[0] == '/') {
res = ast_mkdir(path, 0777);
*file_sep = '/';
return res;
}
/* Relative path */
res = snprintf(directory, sizeof(directory), "%s/sounds/%s",
ast_config_AST_DATA_DIR, path);
*file_sep = '/';
if (res >= sizeof(directory)) {
/* We truncated, so we fail */
return -1;
}
return ast_mkdir(directory, 0777);
}
static int record_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
char *ext = NULL, *opts[0];
char *parse;
int i = 0;
char tmp[PATH_MAX];
struct ast_filestream *s = NULL;
struct ast_frame *f = NULL;
struct ast_dsp *sildet = NULL; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int silence = 0; /* amount of silence to allow */
int gotsilence = 0; /* did we timeout for silence? */
int truncate_silence = 1; /* truncate on complete silence recording */
int maxduration = 0; /* max duration of recording in milliseconds */
int gottimeout = 0; /* did we timeout for maxduration exceeded? */
int terminator = '#';
RAII_VAR(struct ast_format *, rfmt, NULL, ao2_cleanup);
int ioflags;
struct ast_silence_generator *silgen = NULL;
struct ast_flags flags = { 0, };
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
AST_APP_ARG(silence);
AST_APP_ARG(maxduration);
AST_APP_ARG(options);
);
int ms;
struct timeval start;
const char *status_response = "ERROR";
/* The next few lines of code parse out the filename and header from the input string */
if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (args.argc == 4)
ast_app_parse_options(app_opts, &flags, opts, args.options);
if (!ast_strlen_zero(args.filename)) {
ext = strrchr(args.filename, '.'); /* to support filename with a . in the filename, not format */
if (!ext)
ext = strchr(args.filename, ':');
if (ext) {
*ext = '\0';
ext++;
}
}
if (!ext) {
ast_log(LOG_WARNING, "No extension specified to filename!\n");
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
return -1;
}
if (args.silence) {
if ((sscanf(args.silence, "%30d", &i) == 1) && (i > -1)) {
silence = i * 1000;
} else if (!ast_strlen_zero(args.silence)) {
ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
}
}
if (ast_test_flag(&flags, OPTION_NO_TRUNCATE))
truncate_silence = 0;
if (args.maxduration) {
if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
/* Convert duration to milliseconds */
maxduration = i * 1000;
else if (!ast_strlen_zero(args.maxduration))
ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", args.maxduration);
}
if (ast_test_flag(&flags, OPTION_STAR_TERMINATE))
terminator = '*';
if (ast_test_flag(&flags, OPTION_IGNORE_TERMINATE))
terminator = '\0';
/*
If a '%d' is specified as part of the filename, we replace that token with
sequentially incrementing numbers until we find a unique filename.
*/
if (strchr(args.filename, '%')) {
size_t src, dst, count = 0;
size_t src_len = strlen(args.filename);
size_t dst_len = sizeof(tmp) - 1;
do {
for (src = 0, dst = 0; src < src_len && dst < dst_len; src++) {
if (!strncmp(&args.filename[src], "%d", 2)) {
int s = snprintf(&tmp[dst], PATH_MAX - dst, "%zu", count);
if (s >= PATH_MAX - dst) {
/* We truncated, so we need to bail */
ast_log(LOG_WARNING, "Failed to create unique filename from template: %s\n", args.filename);
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
return -1;
}
dst += s;
src++;
} else {
tmp[dst] = args.filename[src];
tmp[++dst] = '\0';
}
}
count++;
} while (ast_fileexists(tmp, ext, ast_channel_language(chan)) > 0);
} else
ast_copy_string(tmp, args.filename, sizeof(tmp));
pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
if (ast_channel_state(chan) != AST_STATE_UP) {
if (ast_test_flag(&flags, OPTION_SKIP)) {
/* At the user's option, skip if the line is not up */
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SKIP");
return 0;
} else if (!ast_test_flag(&flags, OPTION_NOANSWER)) {
/* Otherwise answer unless we're supposed to record while on-hook */
res = ast_answer(chan);
}
}
if (res) {
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", ast_channel_name(chan));
status_response = "ERROR";
goto out;
}
if (!ast_test_flag(&flags, OPTION_QUIET)) {
/* Some code to play a nice little beep to signify the start of the record operation */
res = ast_streamfile(chan, "beep", ast_channel_language(chan));
if (!res) {
res = ast_waitstream(chan, "");
} else {
ast_log(LOG_WARNING, "ast_streamfile(beep) failed on %s\n", ast_channel_name(chan));
res = 0;
}
ast_stopstream(chan);
}
/* The end of beep code. Now the recording starts */
if (silence > 0) {
rfmt = ao2_bump(ast_channel_readformat(chan));
res = ast_set_read_format(chan, ast_format_slin);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
return -1;
}
sildet = ast_dsp_new();
if (!sildet) {
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
return -1;
}
ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
}
if (create_destination_directory(tmp)) {
ast_log(LOG_WARNING, "Could not create directory for file %s\n", args.filename);
status_response = "ERROR";
goto out;
}
ioflags = ast_test_flag(&flags, OPTION_APPEND) ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
s = ast_writefile(tmp, ext, NULL, ioflags, 0, AST_FILE_MODE);
if (!s) {
ast_log(LOG_WARNING, "Could not create file %s\n", args.filename);
status_response = "ERROR";
goto out;
}
if (ast_opt_transmit_silence)
silgen = ast_channel_start_silence_generator(chan);
/* Request a video update */
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
if (maxduration <= 0)
maxduration = -1;
start = ast_tvnow();
while ((ms = ast_remaining_ms(start, maxduration))) {
ms = ast_waitfor(chan, ms);
if (ms < 0) {
break;
}
if (maxduration > 0 && ms == 0) {
break;
}
f = ast_read(chan);
if (!f) {
res = -1;
break;
}
if (f->frametype == AST_FRAME_VOICE) {
res = ast_writestream(s, f);
if (res) {
ast_log(LOG_WARNING, "Problem writing frame\n");
ast_frfree(f);
status_response = "ERROR";
break;
}
if (silence > 0) {
dspsilence = 0;
ast_dsp_silence(sildet, f, &dspsilence);
if (dspsilence) {
totalsilence = dspsilence;
} else {
totalsilence = 0;
}
if (totalsilence > silence) {
/* Ended happily with silence */
ast_frfree(f);
gotsilence = 1;
status_response = "SILENCE";
break;
}
}
} else if (f->frametype == AST_FRAME_VIDEO) {
res = ast_writestream(s, f);
if (res) {
ast_log(LOG_WARNING, "Problem writing frame\n");
status_response = "ERROR";
ast_frfree(f);
break;
}
} else if (f->frametype == AST_FRAME_DTMF) {
enum dtmf_response rc =
record_dtmf_response(chan, &flags, f->subclass.integer, terminator);
switch(rc) {
case RESPONSE_NO_MATCH:
break;
case RESPONSE_OPERATOR:
status_response = "OPERATOR";
ast_debug(1, "Got OPERATOR\n");
break;
case RESPONSE_DTMF:
status_response = "DTMF";
ast_debug(1, "Got DTMF\n");
break;
}
if (rc != RESPONSE_NO_MATCH) {
ast_frfree(f);
break;
}
}
ast_frfree(f);
}
if (maxduration > 0 && !ms) {
gottimeout = 1;
status_response = "TIMEOUT";
}
if (!f) {
ast_debug(1, "Got hangup\n");
res = -1;
status_response = "HANGUP";
if (!ast_test_flag(&flags, OPTION_KEEP)) {
ast_filedelete(args.filename, NULL);
}
}
if (gotsilence && truncate_silence) {
ast_stream_rewind(s, silence - 1000);
ast_truncstream(s);
} else if (!gottimeout && f) {
/*
* Strip off the last 1/4 second of it, if we didn't end because of a timeout,
* or a hangup. This must mean we ended because of a DTMF tone and while this
* 1/4 second stripping is very old code the most likely explanation is that it
* relates to stripping a partial DTMF tone.
*/
ast_stream_rewind(s, 250);
ast_truncstream(s);
}
ast_closestream(s);
if (silgen)
ast_channel_stop_silence_generator(chan, silgen);
out:
if ((silence > 0) && rfmt) {
res = ast_set_read_format(chan, rfmt);
if (res) {
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(chan));
}
}
if (sildet) {
ast_dsp_free(sildet);
}
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", status_response);
return res;
}
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
return ast_register_application_xml(app, record_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial Record Application");