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  • /*
     * Asterisk -- An open source telephony toolkit.
     *
     * Copyright (C) 2009, Olle E. Johansson
     *
     * Olle E. Johansson <oej@edvina.net>
     *
     * See http://www.asterisk.org for more information about
     * the Asterisk project. Please do not directly contact
     * any of the maintainers of this project for assistance;
     * the project provides a web site, mailing lists and IRC
     * channels for your use.
     *
     * This program is free software, distributed under the terms of
     * the GNU General Public License Version 2. See the LICENSE file
     * at the top of the source tree.
     */
    
    /*! \file
     *
     * \brief MUTESTREAM audiohooks
     *
     * \author Olle E. Johansson <oej@edvina.net>
     *
     *  \ingroup functions
     *
     * \note This module only handles audio streams today, but can easily be appended to also
     * zero out text streams if there's an application for it.
     * When we know and understands what happens if we zero out video, we can do that too.
     */
    
    
    /*** MODULEINFO
    	<support_level>core</support_level>
     ***/
    
    
    ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
    
    
    #include "asterisk/options.h"
    #include "asterisk/logger.h"
    #include "asterisk/channel.h"
    #include "asterisk/module.h"
    #include "asterisk/config.h"
    #include "asterisk/file.h"
    #include "asterisk/pbx.h"
    #include "asterisk/frame.h"
    #include "asterisk/utils.h"
    #include "asterisk/audiohook.h"
    #include "asterisk/manager.h"
    
    /*** DOCUMENTATION
    	<function name="MUTEAUDIO" language="en_US">
    		<synopsis>
    			Muting audio streams in the channel
    		</synopsis>
    		<syntax>
    			<parameter name="direction" required="true">
    				<para>Must be one of </para>
    				<enumlist>
    					<enum name="in">
    						<para>Inbound stream (to the PBX)</para>
    					</enum>
    					<enum name="out">
    						<para>Outbound stream (from the PBX)</para>
    					</enum>
    					<enum name="all">
    						<para>Both streams</para>
    					</enum>
    				</enumlist>
    			</parameter>
    		</syntax>
    		<description>
    			<para>The MUTEAUDIO function can be used to mute inbound (to the PBX) or outbound audio in a call.
    
    			</para>
    			<para>Examples:
    
    			MUTEAUDIO(in)=off
    			</para>
    		</description>
    	</function>
    
    	<manager name="MuteAudio" language="en_US">
    		<synopsis>
    			Mute an audio stream.
    		</synopsis>
    		<syntax>
    			<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
    			<parameter name="Channel" required="true">
    				<para>The channel you want to mute.</para>
    			</parameter>
    			<parameter name="Direction" required="true">
    				<enumlist>
    					<enum name="in">
    						<para>Set muting on inbound audio stream. (to the PBX)</para>
    					</enum>
    					<enum name="out">
    						<para>Set muting on outbound audio stream. (from the PBX)</para>
    					</enum>
    					<enum name="all">
    						<para>Set muting on inbound and outbound audio streams.</para>
    					</enum>
    				</enumlist>
    			</parameter>
    			<parameter name="State" required="true">
    				<enumlist>
    					<enum name="on">
    						<para>Turn muting on.</para>
    					</enum>
    					<enum name="off">
    						<para>Turn muting off.</para>
    					</enum>
    				</enumlist>
    			</parameter>
    		</syntax>
    		<description>
    			<para>Mute an incoming or outgoing audio stream on a channel.</para>
    		</description>
    	</manager>
    
    static int mute_channel(struct ast_channel *chan, const char *direction, int mute)
    
    	unsigned int mute_direction = 0;
    	enum ast_frame_type frametype = AST_FRAME_VOICE;
    	int ret = 0;
    
    	if (!strcmp(direction, "in")) {
    		mute_direction = AST_MUTE_DIRECTION_READ;
    	} else if (!strcmp(direction, "out")) {
    		mute_direction = AST_MUTE_DIRECTION_WRITE;
    	} else if (!strcmp(direction, "all")) {
    		mute_direction = AST_MUTE_DIRECTION_READ | AST_MUTE_DIRECTION_WRITE;
    	} else {
    		return -1;
    
    	if (mute) {
    		ret = ast_channel_suppress(chan, mute_direction, frametype);
    	} else {
    		ret = ast_channel_unsuppress(chan, mute_direction, frametype);
    
    	ast_channel_unlock(chan);
    
    }
    
    /*! \brief Mute dialplan function */
    static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
    {
    
    	if (!chan) {
    		ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
    		return -1;
    	}
    
    
    	return mute_channel(chan, data, ast_true(value));
    
    }
    
    /* Function for debugging - might be useful */
    static struct ast_custom_function mute_function = {
    
    	.name = "MUTEAUDIO",
    	.write = func_mute_write,
    
    };
    
    static int manager_mutestream(struct mansession *s, const struct message *m)
    {
    	const char *channel = astman_get_header(m, "Channel");
    	const char *id = astman_get_header(m,"ActionID");
    	const char *state = astman_get_header(m,"State");
    	const char *direction = astman_get_header(m,"Direction");
    
    	char id_text[256];
    
    	struct ast_channel *c = NULL;
    
    	if (ast_strlen_zero(channel)) {
    		astman_send_error(s, m, "Channel not specified");
    		return 0;
    	}
    	if (ast_strlen_zero(state)) {
    		astman_send_error(s, m, "State not specified");
    		return 0;
    	}
    	if (ast_strlen_zero(direction)) {
    		astman_send_error(s, m, "Direction not specified");
    		return 0;
    	}
    	/* Ok, we have everything */
    
    	c = ast_channel_get_by_name(channel);
    	if (!c) {
    		astman_send_error(s, m, "No such channel");
    		return 0;
    	}
    
    
    	if (mute_channel(c, direction, ast_true(state))) {
    		astman_send_error(s, m, "Failed to mute/unmute stream");
    		ast_channel_unref(c);
    		return 0;
    
    	if (!ast_strlen_zero(id)) {
    		snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
    	} else {
    		id_text[0] = '\0';
    	}
    
    	astman_append(s, "Response: Success\r\n"
    
    		"%s"
    		"\r\n", id_text);
    
    	res = ast_custom_function_register(&mute_function);
    	res |= ast_manager_register_xml("MuteAudio", EVENT_FLAG_SYSTEM, manager_mutestream);
    
    
    	return (res ? AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS);
    }
    
    static int unload_module(void)
    {
    	ast_custom_function_unregister(&mute_function);
    	/* Unregister AMI actions */
    
    	ast_manager_unregister("MuteAudio");
    
    
    	return 0;
    }
    
    AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mute audio stream resources");