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	As partly documented in loader.c and include/asterisk/module.h, modules are now expected to return all of their methods and flags into a structure 'mod_data', and are normally loaded with RTLD_NOW | RTLD_LOCAL, so symbols are resolved immediately and conflicts should be less likely. Only in a small number of cases (res_*, typically) modules are loaded RTLD_GLOBAL, so they can export symbols. The core of the change is only the two files loader.c and include/asterisk/module.h, all the rest is simply adaptation of the existing modules to the new API, a rather mechanical (but believe me, time and finger-consuming!) process whose detail you can figure out by svn diff'ing any single module. Expect some minor compilation issue after this change, please report it on mantis http://bugs.digium.com/view.php?id=6968 so we collect all the feedback in one place. I am just sorry that this change missed SVN version number 20000! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			244 lines
		
	
	
		
			6.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			244 lines
		
	
	
		
			6.2 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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|  * Mark Spencer <markster@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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| 
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| /*! \file
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|  *
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|  * \brief Playback a file with audio detect
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|  *
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|  * \author Mark Spencer <markster@digium.com>
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|  * 
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|  * \ingroup applications
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|  */
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|  
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| #include <stdlib.h>
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| #include <stdio.h>
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| #include <string.h>
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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/lock.h"
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| #include "asterisk/file.h"
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| #include "asterisk/logger.h"
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| #include "asterisk/channel.h"
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| #include "asterisk/pbx.h"
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| #include "asterisk/module.h"
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| #include "asterisk/translate.h"
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| #include "asterisk/utils.h"
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| #include "asterisk/dsp.h"
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| 
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| static char *tdesc = "Playback with Talk Detection";
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| 
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| static char *app = "BackgroundDetect";
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| 
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| static char *synopsis = "Background a file with talk detect";
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| 
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| static char *descrip = 
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| "  BackgroundDetect(filename[|sil[|min|[max]]]):  Plays  back  a  given\n"
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| "filename, waiting for interruption from a given digit (the digit must\n"
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| "start the beginning of a valid extension, or it will be ignored).\n"
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| "During the playback of the file, audio is monitored in the receive\n"
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| "direction, and if a period of non-silence which is greater than 'min' ms\n"
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| "yet less than 'max' ms is followed by silence for at least 'sil' ms then\n"
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| "the audio playback is aborted and processing jumps to the 'talk' extension\n"
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| "if available.  If unspecified, sil, min, and max default to 1000, 100, and\n"
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| "infinity respectively.\n";
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| 
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| LOCAL_USER_DECL;
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| 
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| static int background_detect_exec(struct ast_channel *chan, void *data)
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| {
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| 	int res = 0;
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| 	struct localuser *u;
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| 	char *tmp;
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| 	char *options;
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| 	char *stringp;
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| 	struct ast_frame *fr;
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| 	int notsilent=0;
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| 	struct timeval start = { 0, 0};
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| 	int sil = 1000;
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| 	int min = 100;
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| 	int max = -1;
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| 	int x;
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| 	int origrformat=0;
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| 	struct ast_dsp *dsp;
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| 	
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| 	if (ast_strlen_zero(data)) {
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| 		ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
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| 		return -1;
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| 	}
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| 
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| 	LOCAL_USER_ADD(u);
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| 
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| 	if (!(tmp = ast_strdupa(data))) {
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| 		LOCAL_USER_REMOVE(u);
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| 		return -1;
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| 	}	
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| 
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| 	stringp=tmp;
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| 	strsep(&stringp, "|");
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| 	options = strsep(&stringp, "|");
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| 	if (options) {
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| 		if ((sscanf(options, "%d", &x) == 1) && (x > 0))
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| 			sil = x;
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| 		options = strsep(&stringp, "|");
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| 		if (options) {
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| 			if ((sscanf(options, "%d", &x) == 1) && (x > 0))
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| 				min = x;
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| 			options = strsep(&stringp, "|");
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| 			if (options) {
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| 				if ((sscanf(options, "%d", &x) == 1) && (x > 0))
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| 					max = x;
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| 			}
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| 		}
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| 	}
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| 	ast_log(LOG_DEBUG, "Preparing detect of '%s', sil=%d,min=%d,max=%d\n", 
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| 						tmp, sil, min, max);
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| 	if (chan->_state != AST_STATE_UP) {
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| 		/* Otherwise answer unless we're supposed to send this while on-hook */
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| 		res = ast_answer(chan);
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| 	}
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| 	if (!res) {
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| 		origrformat = chan->readformat;
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| 		if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR))) 
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| 			ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
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| 	}
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| 	if (!(dsp = ast_dsp_new())) {
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| 		ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
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| 		res = -1;
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| 	}
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| 	if (!res) {
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| 		ast_stopstream(chan);
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| 		res = ast_streamfile(chan, tmp, chan->language);
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| 		if (!res) {
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| 			while(chan->stream) {
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| 				res = ast_sched_wait(chan->sched);
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| 				if ((res < 0) && !chan->timingfunc) {
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| 					res = 0;
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| 					break;
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| 				}
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| 				if (res < 0)
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| 					res = 1000;
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| 				res = ast_waitfor(chan, res);
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| 				if (res < 0) {
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| 					ast_log(LOG_WARNING, "Waitfor failed on %s\n", chan->name);
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| 					break;
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| 				} else if (res > 0) {
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| 					fr = ast_read(chan);
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| 					if (!fr) {
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| 						res = -1;
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| 						break;
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| 					} else if (fr->frametype == AST_FRAME_DTMF) {
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| 						char t[2];
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| 						t[0] = fr->subclass;
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| 						t[1] = '\0';
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| 						if (ast_canmatch_extension(chan, chan->context, t, 1, chan->cid.cid_num)) {
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| 							/* They entered a valid  extension, or might be anyhow */
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| 							res = fr->subclass;
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| 							ast_frfree(fr);
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| 							break;
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| 						}
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| 					} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass == AST_FORMAT_SLINEAR)) {
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| 						int totalsilence;
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| 						int ms;
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| 						res = ast_dsp_silence(dsp, fr, &totalsilence);
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| 						if (res && (totalsilence > sil)) {
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| 							/* We've been quiet a little while */
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| 							if (notsilent) {
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| 								/* We had heard some talking */
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| 								ms = ast_tvdiff_ms(ast_tvnow(), start);
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| 								ms -= sil;
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| 								if (ms < 0)
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| 									ms = 0;
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| 								if ((ms > min) && ((max < 0) || (ms < max))) {
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| 									char ms_str[10];
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| 									ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
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| 
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| 									/* Save detected talk time (in milliseconds) */ 
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| 									sprintf(ms_str, "%d", ms );	
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| 									pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
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| 									
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| 									ast_goto_if_exists(chan, chan->context, "talk", 1);
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| 									res = 0;
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| 									ast_frfree(fr);
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| 									break;
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| 								} else
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| 									ast_log(LOG_DEBUG, "Found unqualified token of %d ms\n", ms);
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| 								notsilent = 0;
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| 							}
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| 						} else {
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| 							if (!notsilent) {
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| 								/* Heard some audio, mark the begining of the token */
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| 								start = ast_tvnow();
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| 								ast_log(LOG_DEBUG, "Start of voice token!\n");
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| 								notsilent = 1;
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| 							}
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| 						}
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| 						
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| 					}
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| 					ast_frfree(fr);
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| 				}
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| 				ast_sched_runq(chan->sched);
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| 			}
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| 			ast_stopstream(chan);
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| 		} else {
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| 			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
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| 			res = 0;
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| 		}
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| 	}
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| 	if (res > -1) {
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| 		if (origrformat && ast_set_read_format(chan, origrformat)) {
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| 			ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n", 
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| 				chan->name, ast_getformatname(origrformat));
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| 		}
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| 	}
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| 	if (dsp)
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| 		ast_dsp_free(dsp);
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| 	LOCAL_USER_REMOVE(u);
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| 	return res;
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| }
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| 
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| static int unload_module(void *mod)
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| {
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| 	int res;
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| 
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| 	res = ast_unregister_application(app);
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| 	
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| 	STANDARD_HANGUP_LOCALUSERS;
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| 
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| 	return res;	
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| }
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| 
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| static int load_module(void *mod)
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| {
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| 	return ast_register_application(app, background_detect_exec, synopsis, descrip);
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| }
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| 
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| static const char *description(void)
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| {
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| 	return tdesc;
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| }
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| 
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| static const char *key(void)
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| {
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| 	return ASTERISK_GPL_KEY;
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| }
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| 
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| STD_MOD1;
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