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	(closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			242 lines
		
	
	
		
			6.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			242 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) 2011, Digium, Inc.
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 *
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 * Joshua Colp <jcolp@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 Technology independent volume control
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 *
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 * \author Joshua Colp <jcolp@digium.com> 
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 *
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 * \ingroup functions
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 *
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 */
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/*** MODULEINFO
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	<support_level>core</support_level>
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 ***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/utils.h"
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#include "asterisk/audiohook.h"
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#include "asterisk/app.h"
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/*** DOCUMENTATION
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	<function name="VOLUME" language="en_US">
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		<synopsis>
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			Set the TX or RX volume of a channel.
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		</synopsis>
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		<syntax>
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			<parameter name="direction" required="true">
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				<para>Must be <literal>TX</literal> or <literal>RX</literal>.</para>
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			</parameter>
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			<parameter name="options">
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				<optionlist>
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					<option name="p">
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						<para>Enable DTMF volume control</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>The VOLUME function can be used to increase or decrease the <literal>tx</literal> or
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			<literal>rx</literal> gain of any channel.</para>
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			<para>For example:</para>
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			<para>Set(VOLUME(TX)=3)</para>
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			<para>Set(VOLUME(RX)=2)</para>
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			<para>Set(VOLUME(TX,p)=3)</para>
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			<para>Set(VOLUME(RX,p)=3)</para>
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		</description>
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	</function>
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 ***/
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struct volume_information {
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	struct ast_audiohook audiohook;
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	int tx_gain;
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	int rx_gain;
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	unsigned int flags;
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};
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enum volume_flags {
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	VOLUMEFLAG_CHANGE = (1 << 1),
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};
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AST_APP_OPTIONS(volume_opts, {
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	AST_APP_OPTION('p', VOLUMEFLAG_CHANGE),
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});
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static void destroy_callback(void *data)
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{
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	struct volume_information *vi = data;
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	/* Destroy the audiohook, and destroy ourselves */
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	ast_audiohook_lock(&vi->audiohook);
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	ast_audiohook_detach(&vi->audiohook);
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	ast_audiohook_unlock(&vi->audiohook);
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	ast_audiohook_destroy(&vi->audiohook);
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	ast_free(vi);
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	return;
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}
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/*! \brief Static structure for datastore information */
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static const struct ast_datastore_info volume_datastore = {
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	.type = "volume",
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	.destroy = destroy_callback
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};
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static int volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
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{
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	struct ast_datastore *datastore = NULL;
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	struct volume_information *vi = NULL;
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	int *gain = NULL;
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	/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
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	if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
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		return 0;
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	/* Grab datastore which contains our gain information */
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	if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL)))
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		return 0;
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	vi = datastore->data;
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	/* If this is DTMF then allow them to increase/decrease the gains */
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	if (ast_test_flag(vi, VOLUMEFLAG_CHANGE)) {
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		if (frame->frametype == AST_FRAME_DTMF) {
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			/* Only use DTMF coming from the source... not going to it */
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			if (direction != AST_AUDIOHOOK_DIRECTION_READ)
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				return 0; 
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			if (frame->subclass.integer == '*') {
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				vi->tx_gain += 1;
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				vi->rx_gain += 1;
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			} else if (frame->subclass.integer == '#') {
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				vi->tx_gain -= 1;
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				vi->rx_gain -= 1;
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			}
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		}
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	}
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	if (frame->frametype == AST_FRAME_VOICE) {
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		/* Based on direction of frame grab the gain, and confirm it is applicable */
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		if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain)
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			return 0;
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		/* Apply gain to frame... easy as pi */
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		ast_frame_adjust_volume(frame, *gain);
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	}
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	return 0;
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}
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static int volume_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
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{
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	struct ast_datastore *datastore = NULL;
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	struct volume_information *vi = NULL;
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	int is_new = 0;
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	/* Separate options from argument */
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	AST_DECLARE_APP_ARGS(args,
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		AST_APP_ARG(direction);
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		AST_APP_ARG(options);
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	);
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	if (!chan) {
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		ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
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		return -1;
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	}
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	AST_STANDARD_APP_ARGS(args, data);
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	ast_channel_lock(chan);
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	if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) {
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		ast_channel_unlock(chan);
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		/* Allocate a new datastore to hold the reference to this volume and audiohook information */
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		if (!(datastore = ast_datastore_alloc(&volume_datastore, NULL)))
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			return 0;
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		if (!(vi = ast_calloc(1, sizeof(*vi)))) {
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			ast_datastore_free(datastore);
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			return 0;
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		}
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		ast_audiohook_init(&vi->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
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		vi->audiohook.manipulate_callback = volume_callback;
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		ast_set_flag(&vi->audiohook, AST_AUDIOHOOK_WANTS_DTMF);
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		is_new = 1;
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	} else {
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		ast_channel_unlock(chan);
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		vi = datastore->data;
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	}
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	/* Adjust gain on volume information structure */
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	if (ast_strlen_zero(args.direction)) {
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		ast_log(LOG_ERROR, "Direction must be specified for VOLUME function\n");
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		return -1;
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	}
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	if (!strcasecmp(args.direction, "tx")) { 
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		vi->tx_gain = atoi(value); 
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	} else if (!strcasecmp(args.direction, "rx")) {
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		vi->rx_gain = atoi(value);
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	} else {
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		ast_log(LOG_ERROR, "Direction must be either RX or TX\n");
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	}
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	if (is_new) {
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		datastore->data = vi;
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		ast_channel_lock(chan);
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		ast_channel_datastore_add(chan, datastore);
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		ast_channel_unlock(chan);
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		ast_audiohook_attach(chan, &vi->audiohook);
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	}
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	/* Add Option data to struct */
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	if (!ast_strlen_zero(args.options)) {
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		struct ast_flags flags = { 0 };
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		ast_app_parse_options(volume_opts, &flags, NULL, args.options);
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		vi->flags = flags.flags;
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	} else { 
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		vi->flags = 0; 
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	}
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	return 0;
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}
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static struct ast_custom_function volume_function = {
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	.name = "VOLUME",
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	.write = volume_write,
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};
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static int unload_module(void)
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{
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	return ast_custom_function_unregister(&volume_function);
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}
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static int load_module(void)
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{
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	return ast_custom_function_register(&volume_function);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Technology independent volume control");
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