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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>
- ***/
- #include "asterisk.h"
- 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:
- </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");
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