res_mutestream.c

Go to the documentation of this file.

/*
 * 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");