236 lines
6.0 KiB
C
236 lines
6.0 KiB
C
|
/*
|
||
|
* 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 understand 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>
|
||
|
<example title="Mute incoming audio">
|
||
|
exten => s,1,Set(MUTEAUDIO(in)=on)
|
||
|
</example>
|
||
|
<example title="Do not mute incoming audio">
|
||
|
exten => s,1,Set(MUTEAUDIO(in)=off)
|
||
|
</example>
|
||
|
</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");
|