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							545 lines
						
					
					
						
							15 KiB
						
					
					
				| /*
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|  * Asterisk -- An open source telephony toolkit.
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|  *
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|  * Copyright (C) 1999 - 2005, Digium, Inc.
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|  *
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|  * Matthew Fredrickson <creslin@digium.com>
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|  *
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|  * See http://www.asterisk.org for more information about
 | |
|  * the Asterisk project. Please do not directly contact
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|  * any of the maintainers of this project for assistance;
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|  * the project provides a web site, mailing lists and IRC
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|  * channels for your use.
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|  *
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|  * This program is free software, distributed under the terms of
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|  * the GNU General Public License Version 2. See the LICENSE file
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|  * at the top of the source tree.
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|  */
 | |
| 
 | |
| /*! \file
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|  *
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|  * \brief Trivial application to record a sound file
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|  *
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|  * \author Matthew Fredrickson <creslin@digium.com>
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|  *
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|  * \ingroup applications
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|  */
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| 
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| /*** MODULEINFO
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| 	<support_level>core</support_level>
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|  ***/
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| 
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| #include "asterisk.h"
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| 
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| ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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| 
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| #include "asterisk/file.h"
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| #include "asterisk/pbx.h"
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| #include "asterisk/module.h"
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| #include "asterisk/app.h"
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| #include "asterisk/channel.h"
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| #include "asterisk/dsp.h"	/* use dsp routines for silence detection */
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| #include "asterisk/format_cache.h"
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| #include "asterisk/paths.h"
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| 
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| /*** DOCUMENTATION
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| 	<application name="Record" language="en_US">
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| 		<synopsis>
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| 			Record to a file.
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| 		</synopsis>
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| 		<syntax>
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| 			<parameter name="filename" required="true" argsep=".">
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| 				<argument name="filename" required="true" />
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| 				<argument name="format" required="true">
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| 					<para>Is the format of the file type to be recorded (wav, gsm, etc).</para>
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| 				</argument>
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| 			</parameter>
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| 			<parameter name="silence">
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| 				<para>Is the number of seconds of silence to allow before returning.</para>
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| 			</parameter>
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| 			<parameter name="maxduration">
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| 				<para>Is the maximum recording duration in seconds. If missing
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| 				or 0 there is no maximum.</para>
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| 			</parameter>
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| 			<parameter name="options">
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| 				<optionlist>
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| 					<option name="a">
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| 						<para>Append to existing recording rather than replacing.</para>
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| 					</option>
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| 					<option name="n">
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| 						<para>Do not answer, but record anyway if line not yet answered.</para>
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| 					</option>
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| 					<option name="o">
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| 						<para>Exit when 0 is pressed, setting the variable <variable>RECORD_STATUS</variable>
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| 						to <literal>OPERATOR</literal> instead of <literal>DTMF</literal></para>
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| 					</option>
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| 					<option name="q">
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| 						<para>quiet (do not play a beep tone).</para>
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| 					</option>
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| 					<option name="s">
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| 						<para>skip recording if the line is not yet answered.</para>
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| 					</option>
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| 					<option name="t">
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| 						<para>use alternate '*' terminator key (DTMF) instead of default '#'</para>
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| 					</option>
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| 					<option name="u">
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| 						<para>Don't truncate recorded silence.</para>
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| 					</option>
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| 					<option name="x">
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| 						<para>Ignore all terminator keys (DTMF) and keep recording until hangup.</para>
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| 					</option>
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| 					<option name="k">
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| 					        <para>Keep recorded file upon hangup.</para>
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| 					</option>
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| 					<option name="y">
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| 					        <para>Terminate recording if *any* DTMF digit is received.</para>
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| 					</option>
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| 				</optionlist>
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| 			</parameter>
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| 		</syntax>
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| 		<description>
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| 			<para>If filename contains <literal>%d</literal>, these characters will be replaced with a number
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| 			incremented by one each time the file is recorded.
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| 			Use <astcli>core show file formats</astcli> to see the available formats on your system
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| 			User can press <literal>#</literal> to terminate the recording and continue to the next priority.
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| 			If the user hangs up during a recording, all data will be lost and the application will terminate.</para>
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| 			<variablelist>
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| 				<variable name="RECORDED_FILE">
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| 					<para>Will be set to the final filename of the recording, without an extension.</para>
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| 				</variable>
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| 				<variable name="RECORD_STATUS">
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| 					<para>This is the final status of the command</para>
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| 					<value name="DTMF">A terminating DTMF was received ('#' or '*', depending upon option 't')</value>
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| 					<value name="SILENCE">The maximum silence occurred in the recording.</value>
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| 					<value name="SKIP">The line was not yet answered and the 's' option was specified.</value>
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| 					<value name="TIMEOUT">The maximum length was reached.</value>
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| 					<value name="HANGUP">The channel was hung up.</value>
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| 					<value name="ERROR">An unrecoverable error occurred, which resulted in a WARNING to the logs.</value>
 | |
| 				</variable>
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| 			</variablelist>
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| 		</description>
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| 	</application>
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| 
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|  ***/
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| 
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| #define OPERATOR_KEY '0'
 | |
| 
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| static char *app = "Record";
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| 
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| enum {
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| 	OPTION_APPEND = (1 << 0),
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| 	OPTION_NOANSWER = (1 << 1),
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| 	OPTION_QUIET = (1 << 2),
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| 	OPTION_SKIP = (1 << 3),
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| 	OPTION_STAR_TERMINATE = (1 << 4),
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| 	OPTION_IGNORE_TERMINATE = (1 << 5),
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| 	OPTION_KEEP = (1 << 6),
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| 	OPTION_ANY_TERMINATE = (1 << 7),
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| 	OPTION_OPERATOR_EXIT = (1 << 8),
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| 	OPTION_NO_TRUNCATE = (1 << 9),
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| };
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| 
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| enum dtmf_response {
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| 	RESPONSE_NO_MATCH = 0,
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| 	RESPONSE_OPERATOR,
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| 	RESPONSE_DTMF,
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| };
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| 
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| AST_APP_OPTIONS(app_opts,{
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| 	AST_APP_OPTION('a', OPTION_APPEND),
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| 	AST_APP_OPTION('k', OPTION_KEEP),
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| 	AST_APP_OPTION('n', OPTION_NOANSWER),
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| 	AST_APP_OPTION('o', OPTION_OPERATOR_EXIT),
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| 	AST_APP_OPTION('q', OPTION_QUIET),
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| 	AST_APP_OPTION('s', OPTION_SKIP),
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| 	AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
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| 	AST_APP_OPTION('u', OPTION_NO_TRUNCATE),
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| 	AST_APP_OPTION('y', OPTION_ANY_TERMINATE),
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| 	AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
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| });
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| 
 | |
| /*!
 | |
|  * \internal
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|  * \brief Used to determine what action to take when DTMF is received while recording
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|  * \since 13.0.0
 | |
|  *
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|  * \param chan channel being recorded
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|  * \param flags option flags in use by the record application
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|  * \param dtmf_integer the integer value of the DTMF key received
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|  * \param terminator key currently set to be pressed for normal termination
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|  *
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|  * \returns One of enum dtmf_response
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|  */
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| static enum dtmf_response record_dtmf_response(struct ast_channel *chan,
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| 	struct ast_flags *flags, int dtmf_integer, int terminator)
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| {
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| 	if ((dtmf_integer == OPERATOR_KEY) &&
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| 		(ast_test_flag(flags, OPTION_OPERATOR_EXIT))) {
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| 		return RESPONSE_OPERATOR;
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| 	}
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| 
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| 	if ((dtmf_integer == terminator) ||
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| 		(ast_test_flag(flags, OPTION_ANY_TERMINATE))) {
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| 		return RESPONSE_DTMF;
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| 	}
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| 
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| 	return RESPONSE_NO_MATCH;
 | |
| }
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| 
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| static int create_destination_directory(const char *path)
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| {
 | |
| 	int res;
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| 	char directory[PATH_MAX], *file_sep;
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| 
 | |
| 	if (!(file_sep = strrchr(path, '/'))) {
 | |
| 		/* No directory to create */
 | |
| 		return 0;
 | |
| 	}
 | |
| 
 | |
| 	/* Overwrite temporarily */
 | |
| 	*file_sep = '\0';
 | |
| 
 | |
| 	/* Absolute path? */
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| 	if (path[0] == '/') {
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| 		res = ast_mkdir(path, 0777);
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| 		*file_sep = '/';
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| 		return res;
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| 	}
 | |
| 
 | |
| 	/* Relative path */
 | |
| 	res = snprintf(directory, sizeof(directory), "%s/sounds/%s",
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| 				   ast_config_AST_DATA_DIR, path);
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| 
 | |
| 	*file_sep = '/';
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| 
 | |
| 	if (res >= sizeof(directory)) {
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| 		/* We truncated, so we fail */
 | |
| 		return -1;
 | |
| 	}
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| 
 | |
| 	return ast_mkdir(directory, 0777);
 | |
| }
 | |
| 
 | |
| static int record_exec(struct ast_channel *chan, const char *data)
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| {
 | |
| 	int res = 0;
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| 	char *ext = NULL, *opts[0];
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| 	char *parse;
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| 	int i = 0;
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| 	char tmp[PATH_MAX];
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| 
 | |
| 	struct ast_filestream *s = NULL;
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| 	struct ast_frame *f = NULL;
 | |
| 
 | |
| 	struct ast_dsp *sildet = NULL;   	/* silence detector dsp */
 | |
| 	int totalsilence = 0;
 | |
| 	int dspsilence = 0;
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| 	int silence = 0;		/* amount of silence to allow */
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| 	int gotsilence = 0;		/* did we timeout for silence? */
 | |
| 	int truncate_silence = 1;	/* truncate on complete silence recording */
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| 	int maxduration = 0;		/* max duration of recording in milliseconds */
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| 	int gottimeout = 0;		/* did we timeout for maxduration exceeded? */
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| 	int terminator = '#';
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| 	RAII_VAR(struct ast_format *, rfmt, NULL, ao2_cleanup);
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| 	int ioflags;
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| 	struct ast_silence_generator *silgen = NULL;
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| 	struct ast_flags flags = { 0, };
 | |
| 	AST_DECLARE_APP_ARGS(args,
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| 		AST_APP_ARG(filename);
 | |
| 		AST_APP_ARG(silence);
 | |
| 		AST_APP_ARG(maxduration);
 | |
| 		AST_APP_ARG(options);
 | |
| 	);
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| 	int ms;
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| 	struct timeval start;
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| 	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 */
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| 		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 failed on %s\n", ast_channel_name(chan));
 | |
| 		}
 | |
| 		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");
 |