app_mixmonitor.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2005, Anthony Minessale II
 * Copyright (C) 2005 - 2006, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 * Kevin P. Fleming <kpfleming@digium.com>
 *
 * Based on app_muxmon.c provided by
 * Anthony Minessale II <anthmct@yahoo.com>
 *
 * 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 MixMonitor() - Record a call and mix the audio during the recording
 * \ingroup applications
 *
 * \author Mark Spencer <markster@digium.com>
 * \author Kevin P. Fleming <kpfleming@digium.com>
 *
 * \note Based on app_muxmon.c provided by
 * Anthony Minessale II <anthmct@yahoo.com>
 */

/*** MODULEINFO
   <support_level>core</support_level>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 328209 $")

#include "asterisk/paths.h"   /* use ast_config_AST_MONITOR_DIR */
#include "asterisk/file.h"
#include "asterisk/audiohook.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/autochan.h"
#include "asterisk/manager.h"

/*** DOCUMENTATION
   <application name="MixMonitor" language="en_US">
      <synopsis>
         Record a call and mix the audio during the recording.  Use of StopMixMonitor is required
         to guarantee the audio file is available for processing during dialplan execution.
      </synopsis>
      <syntax>
         <parameter name="file" required="true" argsep=".">
            <argument name="filename" required="true">
               <para>If <replaceable>filename</replaceable> is an absolute path, uses that path, otherwise
               creates the file in the configured monitoring directory from <filename>asterisk.conf.</filename></para>
            </argument>
            <argument name="extension" required="true" />
         </parameter>
         <parameter name="options">
            <optionlist>
               <option name="a">
                  <para>Append to the file instead of overwriting it.</para>
               </option>
               <option name="b">
                  <para>Only save audio to the file while the channel is bridged.</para>
                  <note><para>Does not include conferences or sounds played to each bridged party</para></note>
                  <note><para>If you utilize this option inside a Local channel, you must make sure the Local
                  channel is not optimized away. To do this, be sure to call your Local channel with the
                  <literal>/n</literal> option. For example: Dial(Local/start@mycontext/n)</para></note>
               </option>
               <option name="v">
                  <para>Adjust the <emphasis>heard</emphasis> volume by a factor of <replaceable>x</replaceable>
                  (range <literal>-4</literal> to <literal>4</literal>)</para>
                  <argument name="x" required="true" />
               </option>
               <option name="V">
                  <para>Adjust the <emphasis>spoken</emphasis> volume by a factor
                  of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
                  <argument name="x" required="true" />
               </option>
               <option name="W">
                  <para>Adjust both, <emphasis>heard and spoken</emphasis> volumes by a factor
                  of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
                  <argument name="x" required="true" />
               </option>
            </optionlist>
         </parameter>
         <parameter name="command">
            <para>Will be executed when the recording is over.</para>
            <para>Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.</para>
            <para>All variables will be evaluated at the time MixMonitor is called.</para>
         </parameter>
      </syntax>
      <description>
         <para>Records the audio on the current channel to the specified file.</para>
         <variablelist>
            <variable name="MIXMONITOR_FILENAME">
               <para>Will contain the filename used to record.</para>
            </variable>
         </variablelist>   
      </description>
      <see-also>
         <ref type="application">Monitor</ref>
         <ref type="application">StopMixMonitor</ref>
         <ref type="application">PauseMonitor</ref>
         <ref type="application">UnpauseMonitor</ref>
      </see-also>
   </application>
   <application name="StopMixMonitor" language="en_US">
      <synopsis>
         Stop recording a call through MixMonitor, and free the recording's file handle.
      </synopsis>
      <syntax />
      <description>
         <para>Stops the audio recording that was started with a call to <literal>MixMonitor()</literal>
         on the current channel.</para>
      </description>
      <see-also>
         <ref type="application">MixMonitor</ref>
      </see-also>
   </application>
   <manager name="MixMonitorMute" language="en_US">
      <synopsis>
         Mute / unMute a Mixmonitor recording.
      </synopsis>
      <syntax>
         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
         <parameter name="Channel" required="true">
            <para>Used to specify the channel to mute.</para>
         </parameter>
         <parameter name="Direction">
            <para>Which part of the recording to mute:  read, write or both (from channel, to channel or both channels).</para>
         </parameter>
         <parameter name="State">
            <para>Turn mute on or off : 1 to turn on, 0 to turn off.</para>
         </parameter>
      </syntax>
      <description>
         <para>This action may be used to mute a MixMonitor recording.</para>
      </description>
   </manager>

 ***/

#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0

static const char * const app = "MixMonitor";

static const char * const stop_app = "StopMixMonitor";

static const char * const mixmonitor_spy_type = "MixMonitor";

struct mixmonitor {
   struct ast_audiohook audiohook;
   char *filename;
   char *post_process;
   char *name;
   unsigned int flags;
   struct ast_autochan *autochan;
   struct mixmonitor_ds *mixmonitor_ds;
};

enum mixmonitor_flags {
   MUXFLAG_APPEND = (1 << 1),
   MUXFLAG_BRIDGED = (1 << 2),
   MUXFLAG_VOLUME = (1 << 3),
   MUXFLAG_READVOLUME = (1 << 4),
   MUXFLAG_WRITEVOLUME = (1 << 5),
};

enum mixmonitor_args {
   OPT_ARG_READVOLUME = 0,
   OPT_ARG_WRITEVOLUME,
   OPT_ARG_VOLUME,
   OPT_ARG_ARRAY_SIZE,
};

AST_APP_OPTIONS(mixmonitor_opts, {
   AST_APP_OPTION('a', MUXFLAG_APPEND),
   AST_APP_OPTION('b', MUXFLAG_BRIDGED),
   AST_APP_OPTION_ARG('v', MUXFLAG_READVOLUME, OPT_ARG_READVOLUME),
   AST_APP_OPTION_ARG('V', MUXFLAG_WRITEVOLUME, OPT_ARG_WRITEVOLUME),
   AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
});

struct mixmonitor_ds {
   unsigned int destruction_ok;
   ast_cond_t destruction_condition;
   ast_mutex_t lock;

   /* The filestream is held in the datastore so it can be stopped
    * immediately during stop_mixmonitor or channel destruction. */
   int fs_quit;
   struct ast_filestream *fs;
   struct ast_audiohook *audiohook;
};

/*!
 * \internal
 * \pre mixmonitor_ds must be locked before calling this function
 */
static void mixmonitor_ds_close_fs(struct mixmonitor_ds *mixmonitor_ds)
{
   if (mixmonitor_ds->fs) {
      ast_closestream(mixmonitor_ds->fs);
      mixmonitor_ds->fs = NULL;
      mixmonitor_ds->fs_quit = 1;
      ast_verb(2, "MixMonitor close filestream\n");
   }
}

static void mixmonitor_ds_destroy(void *data)
{
   struct mixmonitor_ds *mixmonitor_ds = data;

   ast_mutex_lock(&mixmonitor_ds->lock);
   mixmonitor_ds->audiohook = NULL;
   mixmonitor_ds->destruction_ok = 1;
   ast_cond_signal(&mixmonitor_ds->destruction_condition);
   ast_mutex_unlock(&mixmonitor_ds->lock);
}

static struct ast_datastore_info mixmonitor_ds_info = {
   .type = "mixmonitor",
   .destroy = mixmonitor_ds_destroy,
};

static void destroy_monitor_audiohook(struct mixmonitor *mixmonitor)
{
   if (mixmonitor->mixmonitor_ds) {
      ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
      mixmonitor->mixmonitor_ds->audiohook = NULL;
      ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
   }
   /* kill the audiohook.*/
   ast_audiohook_lock(&mixmonitor->audiohook);
   ast_audiohook_detach(&mixmonitor->audiohook);
   ast_audiohook_unlock(&mixmonitor->audiohook);
   ast_audiohook_destroy(&mixmonitor->audiohook);
}

static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook) 
{
   struct ast_channel *peer = NULL;
   int res = 0;

   if (!chan)
      return -1;

   ast_audiohook_attach(chan, audiohook);

   if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
      ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);  

   return res;
}

#define SAMPLES_PER_FRAME 160

static void mixmonitor_free(struct mixmonitor *mixmonitor)
{
   if (mixmonitor) {
      if (mixmonitor->mixmonitor_ds) {
         ast_mutex_destroy(&mixmonitor->mixmonitor_ds->lock);
         ast_cond_destroy(&mixmonitor->mixmonitor_ds->destruction_condition);
         ast_free(mixmonitor->mixmonitor_ds);
      }
      ast_free(mixmonitor);
   }
}
static void *mixmonitor_thread(void *obj) 
{
   struct mixmonitor *mixmonitor = obj;
   struct ast_filestream **fs = NULL;
   unsigned int oflags;
   char *ext;
   char *last_slash;
   int errflag = 0;

   ast_verb(2, "Begin MixMonitor Recording %s\n", mixmonitor->name);

   fs = &mixmonitor->mixmonitor_ds->fs;

   /* The audiohook must enter and exit the loop locked */
   ast_audiohook_lock(&mixmonitor->audiohook);
   while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING && !mixmonitor->mixmonitor_ds->fs_quit) {
      struct ast_frame *fr = NULL;

      if (!(fr = ast_audiohook_read_frame(&mixmonitor->audiohook, SAMPLES_PER_FRAME, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR))) {
         ast_audiohook_trigger_wait(&mixmonitor->audiohook);

         if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
            break;
         }
         continue;
      }

      /* audiohook lock is not required for the next block.
       * Unlock it, but remember to lock it before looping or exiting */
      ast_audiohook_unlock(&mixmonitor->audiohook);

      if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || (mixmonitor->autochan->chan && ast_bridged_channel(mixmonitor->autochan->chan))) {
         ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
         /* Initialize the file if not already done so */
         if (!*fs && !errflag && !mixmonitor->mixmonitor_ds->fs_quit) {
            oflags = O_CREAT | O_WRONLY;
            oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;

            last_slash = strrchr(mixmonitor->filename, '/');
            if ((ext = strrchr(mixmonitor->filename, '.')) && (ext > last_slash))
               *(ext++) = '\0';
            else
               ext = "raw";

            if (!(*fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0666))) {
               ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
               errflag = 1;
            }
         }

         /* Write out the frame(s) */
         if (*fs) {
            struct ast_frame *cur;

            for (cur = fr; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
               ast_writestream(*fs, cur);
            }
         }
         ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
      }
      /* All done! free it. */
      ast_frame_free(fr, 0);

      ast_audiohook_lock(&mixmonitor->audiohook);
   }
   ast_audiohook_unlock(&mixmonitor->audiohook);

   ast_autochan_destroy(mixmonitor->autochan);

   /* Datastore cleanup.  close the filestream and wait for ds destruction */
   ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
   mixmonitor_ds_close_fs(mixmonitor->mixmonitor_ds);
   if (!mixmonitor->mixmonitor_ds->destruction_ok) {
      ast_cond_wait(&mixmonitor->mixmonitor_ds->destruction_condition, &mixmonitor->mixmonitor_ds->lock);
   }
   ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);

   /* kill the audiohook */
   destroy_monitor_audiohook(mixmonitor);

   if (mixmonitor->post_process) {
      ast_verb(2, "Executing [%s]\n", mixmonitor->post_process);
      ast_safe_system(mixmonitor->post_process);
   }

   ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
   mixmonitor_free(mixmonitor);
   return NULL;
}

static int setup_mixmonitor_ds(struct mixmonitor *mixmonitor, struct ast_channel *chan)
{
   struct ast_datastore *datastore = NULL;
   struct mixmonitor_ds *mixmonitor_ds;

   if (!(mixmonitor_ds = ast_calloc(1, sizeof(*mixmonitor_ds)))) {
      return -1;
   }

   ast_mutex_init(&mixmonitor_ds->lock);
   ast_cond_init(&mixmonitor_ds->destruction_condition, NULL);

   if (!(datastore = ast_datastore_alloc(&mixmonitor_ds_info, NULL))) {
      ast_mutex_destroy(&mixmonitor_ds->lock);
      ast_cond_destroy(&mixmonitor_ds->destruction_condition);
      ast_free(mixmonitor_ds);
      return -1;
   }

   mixmonitor_ds->audiohook = &mixmonitor->audiohook;
   datastore->data = mixmonitor_ds;

   ast_channel_lock(chan);
   ast_channel_datastore_add(chan, datastore);
   ast_channel_unlock(chan);

   mixmonitor->mixmonitor_ds = mixmonitor_ds;
   return 0;
}

static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
              int readvol, int writevol, const char *post_process) 
{
   pthread_t thread;
   struct mixmonitor *mixmonitor;
   char postprocess2[1024] = "";
   size_t len;

   len = sizeof(*mixmonitor) + strlen(chan->name) + strlen(filename) + 2;

   postprocess2[0] = 0;
   /* If a post process system command is given attach it to the structure */
   if (!ast_strlen_zero(post_process)) {
      char *p1, *p2;

      p1 = ast_strdupa(post_process);
      for (p2 = p1; *p2 ; p2++) {
         if (*p2 == '^' && *(p2+1) == '{') {
            *p2 = '$';
         }
      }
      pbx_substitute_variables_helper(chan, p1, postprocess2, sizeof(postprocess2) - 1);
      if (!ast_strlen_zero(postprocess2))
         len += strlen(postprocess2) + 1;
   }

   /* Pre-allocate mixmonitor structure and spy */
   if (!(mixmonitor = ast_calloc(1, len))) {
      return;
   }

   /* Setup the actual spy before creating our thread */
   if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) {
      mixmonitor_free(mixmonitor);
      return;
   }

   /* Copy over flags and channel name */
   mixmonitor->flags = flags;
   if (!(mixmonitor->autochan = ast_autochan_setup(chan))) {
      mixmonitor_free(mixmonitor);
      return;
   }

   if (setup_mixmonitor_ds(mixmonitor, chan)) {
      ast_autochan_destroy(mixmonitor->autochan);
      mixmonitor_free(mixmonitor);
      return;
   }
   mixmonitor->name = (char *) mixmonitor + sizeof(*mixmonitor);
   strcpy(mixmonitor->name, chan->name);
   if (!ast_strlen_zero(postprocess2)) {
      mixmonitor->post_process = mixmonitor->name + strlen(mixmonitor->name) + strlen(filename) + 2;
      strcpy(mixmonitor->post_process, postprocess2);
   }

   mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor) + strlen(chan->name) + 1;
   strcpy(mixmonitor->filename, filename);

   ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);

   if (readvol)
      mixmonitor->audiohook.options.read_volume = readvol;
   if (writevol)
      mixmonitor->audiohook.options.write_volume = writevol;

   if (startmon(chan, &mixmonitor->audiohook)) {
      ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
         mixmonitor_spy_type, chan->name);
      ast_audiohook_destroy(&mixmonitor->audiohook);
      mixmonitor_free(mixmonitor);
      return;
   }

   ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
}

static int mixmonitor_exec(struct ast_channel *chan, const char *data)
{
   int x, readvol = 0, writevol = 0;
   struct ast_flags flags = {0};
   char *parse, *tmp, *slash;
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(filename);
      AST_APP_ARG(options);
      AST_APP_ARG(post_process);
   );
   
   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
      return -1;
   }

   parse = ast_strdupa(data);

   AST_STANDARD_APP_ARGS(args, parse);
   
   if (ast_strlen_zero(args.filename)) {
      ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
      return -1;
   }

   if (args.options) {
      char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };

      ast_app_parse_options(mixmonitor_opts, &flags, opts, args.options);

      if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
         if (ast_strlen_zero(opts[OPT_ARG_READVOLUME])) {
            ast_log(LOG_WARNING, "No volume level was provided for the heard volume ('v') option.\n");
         } else if ((sscanf(opts[OPT_ARG_READVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
            ast_log(LOG_NOTICE, "Heard volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_READVOLUME]);
         } else {
            readvol = get_volfactor(x);
         }
      }
      
      if (ast_test_flag(&flags, MUXFLAG_WRITEVOLUME)) {
         if (ast_strlen_zero(opts[OPT_ARG_WRITEVOLUME])) {
            ast_log(LOG_WARNING, "No volume level was provided for the spoken volume ('V') option.\n");
         } else if ((sscanf(opts[OPT_ARG_WRITEVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
            ast_log(LOG_NOTICE, "Spoken volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_WRITEVOLUME]);
         } else {
            writevol = get_volfactor(x);
         }
      }
      
      if (ast_test_flag(&flags, MUXFLAG_VOLUME)) {
         if (ast_strlen_zero(opts[OPT_ARG_VOLUME])) {
            ast_log(LOG_WARNING, "No volume level was provided for the combined volume ('W') option.\n");
         } else if ((sscanf(opts[OPT_ARG_VOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
            ast_log(LOG_NOTICE, "Combined volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_VOLUME]);
         } else {
            readvol = writevol = get_volfactor(x);
         }
      }
   }

   /* if not provided an absolute path, use the system-configured monitoring directory */
   if (args.filename[0] != '/') {
      char *build;

      build = alloca(strlen(ast_config_AST_MONITOR_DIR) + strlen(args.filename) + 3);
      sprintf(build, "%s/%s", ast_config_AST_MONITOR_DIR, args.filename);
      args.filename = build;
   }

   tmp = ast_strdupa(args.filename);
   if ((slash = strrchr(tmp, '/')))
      *slash = '\0';
   ast_mkdir(tmp, 0777);

   pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
   launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process);

   return 0;
}

static int stop_mixmonitor_exec(struct ast_channel *chan, const char *data)
{
   struct ast_datastore *datastore = NULL;

   ast_channel_lock(chan);
   ast_audiohook_detach_source(chan, mixmonitor_spy_type);
   if ((datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, NULL))) {
      struct mixmonitor_ds *mixmonitor_ds = datastore->data;

      ast_mutex_lock(&mixmonitor_ds->lock);

      /* closing the filestream here guarantees the file is avaliable to the dialplan
       * after calling StopMixMonitor */
      mixmonitor_ds_close_fs(mixmonitor_ds);

      /* The mixmonitor thread may be waiting on the audiohook trigger.
       * In order to exit from the mixmonitor loop before waiting on channel
       * destruction, poke the audiohook trigger. */
      if (mixmonitor_ds->audiohook) {
         ast_audiohook_lock(mixmonitor_ds->audiohook);
         ast_cond_signal(&mixmonitor_ds->audiohook->trigger);
         ast_audiohook_unlock(mixmonitor_ds->audiohook);
         mixmonitor_ds->audiohook = NULL;
      }

      ast_mutex_unlock(&mixmonitor_ds->lock);

      /* Remove the datastore so the monitor thread can exit */
      if (!ast_channel_datastore_remove(chan, datastore)) {
         ast_datastore_free(datastore);
      }
   }
   ast_channel_unlock(chan);

   return 0;
}

static char *handle_cli_mixmonitor(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
   struct ast_channel *chan;

   switch (cmd) {
   case CLI_INIT:
      e->command = "mixmonitor {start|stop}";
      e->usage =
         "Usage: mixmonitor <start|stop> <chan_name> [args]\n"
         "       The optional arguments are passed to the MixMonitor\n"
         "       application when the 'start' command is used.\n";
      return NULL;
   case CLI_GENERATE:
      return ast_complete_channels(a->line, a->word, a->pos, a->n, 2);
   }

   if (a->argc < 3)
      return CLI_SHOWUSAGE;

   if (!(chan = ast_channel_get_by_name_prefix(a->argv[2], strlen(a->argv[2])))) {
      ast_cli(a->fd, "No channel matching '%s' found.\n", a->argv[2]);
      /* Technically this is a failure, but we don't want 2 errors printing out */
      return CLI_SUCCESS;
   }

   ast_channel_lock(chan);

   if (!strcasecmp(a->argv[1], "start")) {
      mixmonitor_exec(chan, a->argv[3]);
      ast_channel_unlock(chan);
   } else {
      ast_channel_unlock(chan);
      ast_audiohook_detach_source(chan, mixmonitor_spy_type);
   }

   chan = ast_channel_unref(chan);

   return CLI_SUCCESS;
}

/*! \brief  Mute / unmute  a MixMonitor channel */
static int manager_mute_mixmonitor(struct mansession *s, const struct message *m)
{
   struct ast_channel *c = NULL;

   const char *name = 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");

   int clearmute = 1;

   enum ast_audiohook_flags flag;

   if (ast_strlen_zero(direction)) {
      astman_send_error(s, m, "No direction specified. Must be read, write or both");
      return AMI_SUCCESS;
   }

   if (!strcasecmp(direction, "read")) {
      flag = AST_AUDIOHOOK_MUTE_READ;
   } else  if (!strcasecmp(direction, "write")) {
      flag = AST_AUDIOHOOK_MUTE_WRITE;
   } else  if (!strcasecmp(direction, "both")) {
      flag = AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE;
   } else {
      astman_send_error(s, m, "Invalid direction specified. Must be read, write or both");
      return AMI_SUCCESS;
   }

   if (ast_strlen_zero(name)) {
      astman_send_error(s, m, "No channel specified");
      return AMI_SUCCESS;
   }

   if (ast_strlen_zero(state)) {
      astman_send_error(s, m, "No state specified");
      return AMI_SUCCESS;
   }

   clearmute = ast_false(state);
   c = ast_channel_get_by_name(name);

   if (!c) {
      astman_send_error(s, m, "No such channel");
      return AMI_SUCCESS;
   }

   if (ast_audiohook_set_mute(c, mixmonitor_spy_type, flag, clearmute)) {
      c = ast_channel_unref(c);
      astman_send_error(s, m, "Cannot set mute flag");
      return AMI_SUCCESS;
   }

   astman_append(s, "Response: Success\r\n");

   if (!ast_strlen_zero(id)) {
      astman_append(s, "ActionID: %s\r\n", id);
   }

   astman_append(s, "\r\n");

   c = ast_channel_unref(c);

   return AMI_SUCCESS;
}

static struct ast_cli_entry cli_mixmonitor[] = {
   AST_CLI_DEFINE(handle_cli_mixmonitor, "Execute a MixMonitor command")
};

static int unload_module(void)
{
   int res;

   ast_cli_unregister_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
   res = ast_unregister_application(stop_app);
   res |= ast_unregister_application(app);
   res |= ast_manager_unregister("MixMonitorMute");
   
   return res;
}

static int load_module(void)
{
   int res;

   ast_cli_register_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
   res = ast_register_application_xml(app, mixmonitor_exec);
   res |= ast_register_application_xml(stop_app, stop_mixmonitor_exec);
   res |= ast_manager_register_xml("MixMonitorMute", 0, manager_mute_mixmonitor);

   return res;
}

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mixed Audio Monitoring Application");