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			236 lines
		
	
	
		
			6.0 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			236 lines
		
	
	
		
			6.0 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * Asterisk -- An open source telephony toolkit.
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 *
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 * Copyright (C) 2009, Olle E. Johansson
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 *
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 * Olle E. Johansson <oej@edvina.net>
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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 MUTESTREAM audiohooks
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 *
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 * \author Olle E. Johansson <oej@edvina.net>
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 *
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 *  \ingroup functions
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 *
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 * \note This module only handles audio streams today, but can easily be appended to also
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 * zero out text streams if there's an application for it.
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 * When we know and understand what happens if we zero out video, we can do that too.
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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/options.h"
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#include "asterisk/logger.h"
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/config.h"
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#include "asterisk/file.h"
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#include "asterisk/pbx.h"
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#include "asterisk/frame.h"
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#include "asterisk/utils.h"
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#include "asterisk/audiohook.h"
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#include "asterisk/manager.h"
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/*** DOCUMENTATION
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	<function name="MUTEAUDIO" language="en_US">
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		<synopsis>
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			Muting audio streams in the 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 one of </para>
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				<enumlist>
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					<enum name="in">
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						<para>Inbound stream (to the PBX)</para>
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					</enum>
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					<enum name="out">
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						<para>Outbound stream (from the PBX)</para>
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					</enum>
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					<enum name="all">
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						<para>Both streams</para>
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					</enum>
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				</enumlist>
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			</parameter>
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		</syntax>
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		<description>
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			<para>The MUTEAUDIO function can be used to mute inbound (to the PBX) or outbound audio in a call.</para>
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			<example title="Mute incoming audio">
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			exten => s,1,Set(MUTEAUDIO(in)=on)
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			</example>
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			<example title="Do not mute incoming audio">
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			exten => s,1,Set(MUTEAUDIO(in)=off)
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			</example>
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		</description>
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	</function>
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	<manager name="MuteAudio" language="en_US">
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		<synopsis>
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			Mute an audio stream.
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		</synopsis>
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		<syntax>
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			<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
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			<parameter name="Channel" required="true">
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				<para>The channel you want to mute.</para>
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			</parameter>
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			<parameter name="Direction" required="true">
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				<enumlist>
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					<enum name="in">
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						<para>Set muting on inbound audio stream. (to the PBX)</para>
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					</enum>
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					<enum name="out">
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						<para>Set muting on outbound audio stream. (from the PBX)</para>
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					</enum>
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					<enum name="all">
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						<para>Set muting on inbound and outbound audio streams.</para>
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					</enum>
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				</enumlist>
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			</parameter>
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			<parameter name="State" required="true">
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				<enumlist>
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					<enum name="on">
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						<para>Turn muting on.</para>
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					</enum>
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					<enum name="off">
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						<para>Turn muting off.</para>
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					</enum>
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				</enumlist>
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			</parameter>
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		</syntax>
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		<description>
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			<para>Mute an incoming or outgoing audio stream on a channel.</para>
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		</description>
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	</manager>
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 ***/
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static int mute_channel(struct ast_channel *chan, const char *direction, int mute)
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{
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	unsigned int mute_direction = 0;
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	enum ast_frame_type frametype = AST_FRAME_VOICE;
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	int ret = 0;
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	if (!strcmp(direction, "in")) {
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		mute_direction = AST_MUTE_DIRECTION_READ;
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	} else if (!strcmp(direction, "out")) {
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		mute_direction = AST_MUTE_DIRECTION_WRITE;
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	} else if (!strcmp(direction, "all")) {
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		mute_direction = AST_MUTE_DIRECTION_READ | AST_MUTE_DIRECTION_WRITE;
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	} else {
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		return -1;
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	}
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	ast_channel_lock(chan);
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	if (mute) {
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		ret = ast_channel_suppress(chan, mute_direction, frametype);
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	} else {
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		ret = ast_channel_unsuppress(chan, mute_direction, frametype);
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	}
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	ast_channel_unlock(chan);
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	return ret;
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}
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/*! \brief Mute dialplan function */
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static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
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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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	return mute_channel(chan, data, ast_true(value));
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}
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/* Function for debugging - might be useful */
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static struct ast_custom_function mute_function = {
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	.name = "MUTEAUDIO",
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	.write = func_mute_write,
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};
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static int manager_mutestream(struct mansession *s, const struct message *m)
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{
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	const char *channel = astman_get_header(m, "Channel");
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	const char *id = astman_get_header(m,"ActionID");
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	const char *state = astman_get_header(m,"State");
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	const char *direction = astman_get_header(m,"Direction");
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	char id_text[256];
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	struct ast_channel *c = NULL;
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	if (ast_strlen_zero(channel)) {
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		astman_send_error(s, m, "Channel not specified");
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		return 0;
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	}
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	if (ast_strlen_zero(state)) {
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		astman_send_error(s, m, "State not specified");
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		return 0;
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	}
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	if (ast_strlen_zero(direction)) {
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		astman_send_error(s, m, "Direction not specified");
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		return 0;
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	}
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	/* Ok, we have everything */
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	c = ast_channel_get_by_name(channel);
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	if (!c) {
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		astman_send_error(s, m, "No such channel");
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		return 0;
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	}
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	if (mute_channel(c, direction, ast_true(state))) {
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		astman_send_error(s, m, "Failed to mute/unmute stream");
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		ast_channel_unref(c);
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		return 0;
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	}
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	ast_channel_unref(c);
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	if (!ast_strlen_zero(id)) {
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		snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
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	} else {
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		id_text[0] = '\0';
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	}
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	astman_append(s, "Response: Success\r\n"
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		"%s"
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		"\r\n", id_text);
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	return 0;
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}
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static int load_module(void)
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{
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	int res;
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	res = ast_custom_function_register(&mute_function);
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	res |= ast_manager_register_xml("MuteAudio", EVENT_FLAG_SYSTEM, manager_mutestream);
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	return (res ? AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS);
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}
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static int unload_module(void)
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{
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	ast_custom_function_unregister(&mute_function);
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	/* Unregister AMI actions */
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	ast_manager_unregister("MuteAudio");
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	return 0;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mute audio stream resources");
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