ef9ca8bc32
The Record dialplan function trims 1/4 of a second from the end of recordings in case they are terminated because of DTMF. When hanging up, however, you don't want this to happen. This change makes it so on hangup this does not occur. ASTERISK-24530 #close Reported by: Ben Smithurst patches: app_record_v2.diff submitted by Ben Smithurst (license 6529) Review: https://reviewboard.asterisk.org/r/4201/ ........ Merged revisions 428653 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428654 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428655 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
490 lines
14 KiB
C
490 lines
14 KiB
C
/*
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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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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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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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/*** 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="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.</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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FLAG_HAS_PERCENT = (1 << 7),
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OPTION_ANY_TERMINATE = (1 << 8),
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OPTION_OPERATOR_EXIT = (1 << 9),
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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('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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* \retval 0 do not exit
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* \retval -1 do exit
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*/
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static int record_dtmf_response(struct ast_channel *chan, 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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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "OPERATOR");
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return -1;
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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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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "DTMF");
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return -1;
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}
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return 0;
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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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int count = 0;
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char *ext = NULL, *opts[0];
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char *parse, *dir, *file;
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int i = 0;
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char tmp[256];
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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 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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/* 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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if (strstr(args.filename, "%d"))
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ast_set_flag(&flags, FLAG_HAS_PERCENT);
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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 (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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/* done parsing */
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/* these are to allow the use of the %d in the config file for a wild card of sort to
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create a new file with the inputed name scheme */
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if (ast_test_flag(&flags, FLAG_HAS_PERCENT)) {
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AST_DECLARE_APP_ARGS(fname,
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AST_APP_ARG(piece)[100];
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);
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char *tmp2 = ast_strdupa(args.filename);
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char countstring[15];
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int idx;
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/* Separate each piece out by the format specifier */
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AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%');
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do {
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int tmplen;
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/* First piece has no leading percent, so it's copied verbatim */
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ast_copy_string(tmp, fname.piece[0], sizeof(tmp));
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tmplen = strlen(tmp);
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for (idx = 1; idx < fname.argc; idx++) {
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if (fname.piece[idx][0] == 'd') {
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/* Substitute the count */
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snprintf(countstring, sizeof(countstring), "%d", count);
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ast_copy_string(tmp + tmplen, countstring, sizeof(tmp) - tmplen);
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tmplen += strlen(countstring);
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} else if (tmplen + 2 < sizeof(tmp)) {
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/* Unknown format specifier - just copy it verbatim */
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tmp[tmplen++] = '%';
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tmp[tmplen++] = fname.piece[idx][0];
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}
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/* Copy the remaining portion of the piece */
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ast_copy_string(tmp + tmplen, &(fname.piece[idx][1]), sizeof(tmp) - tmplen);
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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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pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
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} else
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ast_copy_string(tmp, args.filename, sizeof(tmp));
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/* end of routine mentioned */
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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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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "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 failed on %s\n", ast_channel_name(chan));
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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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/* Create the directory if it does not exist. */
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dir = ast_strdupa(tmp);
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if ((file = strrchr(dir, '/')))
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*file++ = '\0';
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ast_mkdir (dir, 0777);
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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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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "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) {
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break;
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}
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if (maxduration > 0 && ms == 0) {
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break;
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}
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f = ast_read(chan);
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if (!f) {
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res = -1;
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break;
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}
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if (f->frametype == AST_FRAME_VOICE) {
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res = ast_writestream(s, f);
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if (res) {
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ast_log(LOG_WARNING, "Problem writing frame\n");
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ast_frfree(f);
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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break;
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}
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if (silence > 0) {
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dspsilence = 0;
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ast_dsp_silence(sildet, f, &dspsilence);
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if (dspsilence) {
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totalsilence = dspsilence;
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} else {
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totalsilence = 0;
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}
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if (totalsilence > silence) {
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/* Ended happily with silence */
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ast_frfree(f);
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gotsilence = 1;
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SILENCE");
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break;
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}
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}
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} else if (f->frametype == AST_FRAME_VIDEO) {
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res = ast_writestream(s, f);
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if (res) {
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ast_log(LOG_WARNING, "Problem writing frame\n");
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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ast_frfree(f);
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break;
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}
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} else if (f->frametype == AST_FRAME_DTMF) {
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if (record_dtmf_response(chan, &flags, f->subclass.integer, terminator)) {
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ast_frfree(f);
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break;
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}
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}
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ast_frfree(f);
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}
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if (maxduration > 0 && !ms) {
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gottimeout = 1;
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "TIMEOUT");
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}
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if (!f) {
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ast_debug(1, "Got hangup\n");
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res = -1;
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "HANGUP");
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if (!ast_test_flag(&flags, OPTION_KEEP)) {
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ast_filedelete(args.filename, NULL);
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}
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}
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if (gotsilence) {
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ast_stream_rewind(s, silence - 1000);
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ast_truncstream(s);
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} else if (!gottimeout && f) {
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/*
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* Strip off the last 1/4 second of it, if we didn't end because of a timeout,
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* or a hangup. This must mean we ended because of a DTMF tone and while this
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* 1/4 second stripping is very old code the most likely explanation is that it
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* relates to stripping a partial DTMF tone.
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*/
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ast_stream_rewind(s, 250);
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ast_truncstream(s);
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}
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ast_closestream(s);
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if (silgen)
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ast_channel_stop_silence_generator(chan, silgen);
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|
|
|
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);
|
|
}
|
|
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");
|