544 lines
15 KiB
C
544 lines
15 KiB
C
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/*
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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
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* 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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*/
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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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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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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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/*** 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>
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</variable>
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</variablelist>
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</description>
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</application>
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***/
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#define OPERATOR_KEY '0'
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static char *app = "Record";
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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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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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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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/*!
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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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*
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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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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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return RESPONSE_NO_MATCH;
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}
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static int create_destination_directory(const char *path)
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{
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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, '/'))) {
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/* No directory to create */
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return 0;
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}
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/* Overwrite temporarily */
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*file_sep = '\0';
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/* 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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}
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/* Relative path */
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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 */
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return -1;
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}
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return ast_mkdir(directory, 0777);
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}
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static int record_exec(struct ast_channel *chan, const char *data)
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{
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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;
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struct ast_dsp *sildet = NULL; /* silence detector dsp */
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int totalsilence = 0;
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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? */
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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, };
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(filename);
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AST_APP_ARG(silence);
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AST_APP_ARG(maxduration);
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AST_APP_ARG(options);
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);
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int ms;
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struct timeval start;
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const char *status_response = "ERROR";
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/* The next few lines of code parse out the filename and header from the input string */
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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");
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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return -1;
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}
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parse = ast_strdupa(data);
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AST_STANDARD_APP_ARGS(args, parse);
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if (args.argc == 4)
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ast_app_parse_options(app_opts, &flags, opts, args.options);
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if (!ast_strlen_zero(args.filename)) {
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ext = strrchr(args.filename, '.'); /* to support filename with a . in the filename, not format */
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if (!ext)
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ext = strchr(args.filename, ':');
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if (ext) {
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*ext = '\0';
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ext++;
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}
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}
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if (!ext) {
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ast_log(LOG_WARNING, "No extension specified to filename!\n");
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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return -1;
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}
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if (args.silence) {
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if ((sscanf(args.silence, "%30d", &i) == 1) && (i > -1)) {
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silence = i * 1000;
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} else if (!ast_strlen_zero(args.silence)) {
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ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
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}
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}
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if (ast_test_flag(&flags, OPTION_NO_TRUNCATE))
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truncate_silence = 0;
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if (args.maxduration) {
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if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
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/* Convert duration to milliseconds */
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maxduration = i * 1000;
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else if (!ast_strlen_zero(args.maxduration))
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ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", args.maxduration);
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}
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if (ast_test_flag(&flags, OPTION_STAR_TERMINATE))
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terminator = '*';
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if (ast_test_flag(&flags, OPTION_IGNORE_TERMINATE))
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terminator = '\0';
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/*
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If a '%d' is specified as part of the filename, we replace that token with
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sequentially incrementing numbers until we find a unique filename.
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*/
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if (strchr(args.filename, '%')) {
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size_t src, dst, count = 0;
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size_t src_len = strlen(args.filename);
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size_t dst_len = sizeof(tmp) - 1;
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do {
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for (src = 0, dst = 0; src < src_len && dst < dst_len; src++) {
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if (!strncmp(&args.filename[src], "%d", 2)) {
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int s = snprintf(&tmp[dst], PATH_MAX - dst, "%zu", count);
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if (s >= PATH_MAX - dst) {
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/* We truncated, so we need to bail */
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ast_log(LOG_WARNING, "Failed to create unique filename from template: %s\n", args.filename);
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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return -1;
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}
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dst += s;
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src++;
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} else {
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tmp[dst] = args.filename[src];
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tmp[++dst] = '\0';
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}
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}
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count++;
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} while (ast_fileexists(tmp, ext, ast_channel_language(chan)) > 0);
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} else
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ast_copy_string(tmp, args.filename, sizeof(tmp));
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pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
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if (ast_channel_state(chan) != AST_STATE_UP) {
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if (ast_test_flag(&flags, OPTION_SKIP)) {
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/* At the user's option, skip if the line is not up */
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SKIP");
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return 0;
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} else if (!ast_test_flag(&flags, OPTION_NOANSWER)) {
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/* Otherwise answer unless we're supposed to record while on-hook */
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res = ast_answer(chan);
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}
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}
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if (res) {
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ast_log(LOG_WARNING, "Could not answer channel '%s'\n", ast_channel_name(chan));
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status_response = "ERROR";
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goto out;
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}
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if (!ast_test_flag(&flags, OPTION_QUIET)) {
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/* Some code to play a nice little beep to signify the start of the record operation */
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res = ast_streamfile(chan, "beep", ast_channel_language(chan));
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if (!res) {
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res = ast_waitstream(chan, "");
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} else {
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ast_log(LOG_WARNING, "ast_streamfile(beep) failed on %s\n", ast_channel_name(chan));
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res = 0;
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}
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ast_stopstream(chan);
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}
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/* The end of beep code. Now the recording starts */
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if (silence > 0) {
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rfmt = ao2_bump(ast_channel_readformat(chan));
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res = ast_set_read_format(chan, ast_format_slin);
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if (res < 0) {
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ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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return -1;
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}
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sildet = ast_dsp_new();
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if (!sildet) {
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ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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return -1;
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}
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ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
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}
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if (create_destination_directory(tmp)) {
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ast_log(LOG_WARNING, "Could not create directory for file %s\n", args.filename);
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status_response = "ERROR";
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goto out;
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}
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ioflags = ast_test_flag(&flags, OPTION_APPEND) ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
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s = ast_writefile(tmp, ext, NULL, ioflags, 0, AST_FILE_MODE);
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if (!s) {
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ast_log(LOG_WARNING, "Could not create file %s\n", args.filename);
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status_response = "ERROR";
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goto out;
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}
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if (ast_opt_transmit_silence)
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silgen = ast_channel_start_silence_generator(chan);
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/* Request a video update */
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ast_indicate(chan, AST_CONTROL_VIDUPDATE);
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if (maxduration <= 0)
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maxduration = -1;
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start = ast_tvnow();
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while ((ms = ast_remaining_ms(start, maxduration))) {
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ms = ast_waitfor(chan, ms);
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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");
|