res_mutestream.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2009, Olle E. Johansson
 *
 * Olle E. Johansson <[email protected]>
 *
 * 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 <[email protected]>
 *
 * \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>
 ***/

#include "asterisk.h"

#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:
 </para>
 <para>
 MUTEAUDIO(in)=on
 </para>
 <para>
 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;
 }

 ast_channel_lock(chan);

 if (mute) {
 ret = ast_channel_suppress(chan, mute_direction, frametype);
 } else {
 ret = ast_channel_unsuppress(chan, mute_direction, frametype);
 }

 ast_channel_unlock(chan);

 return ret;
}

/*! \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;
 }

 ast_channel_unref(c);

 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);
 return 0;
}


static int load_module(void)
{
 int res;

 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");