Asterisk - The Open Source Telephony Project GIT-master-f36a736
res_mutestream.c
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1/*
2 * Asterisk -- An open source telephony toolkit.
3 *
4 * Copyright (C) 2009, Olle E. Johansson
5 *
6 * Olle E. Johansson <oej@edvina.net>
7 *
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
13 *
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
17 */
18
19/*! \file
20 *
21 * \brief MUTESTREAM audiohooks
22 *
23 * \author Olle E. Johansson <oej@edvina.net>
24 *
25 * \ingroup functions
26 *
27 * \note This module only handles audio streams today, but can easily be appended to also
28 * zero out text streams if there's an application for it.
29 * When we know and understand what happens if we zero out video, we can do that too.
30 */
31
32/*** MODULEINFO
33 <support_level>core</support_level>
34 ***/
35
36#include "asterisk.h"
37
38#include "asterisk/options.h"
39#include "asterisk/logger.h"
40#include "asterisk/channel.h"
41#include "asterisk/module.h"
42#include "asterisk/config.h"
43#include "asterisk/file.h"
44#include "asterisk/pbx.h"
45#include "asterisk/frame.h"
46#include "asterisk/utils.h"
47#include "asterisk/audiohook.h"
48#include "asterisk/manager.h"
49
50/*** DOCUMENTATION
51 <function name="MUTEAUDIO" language="en_US">
52 <synopsis>
53 Muting audio streams in the channel
54 </synopsis>
55 <syntax>
56 <parameter name="direction" required="true">
57 <para>Must be one of </para>
58 <enumlist>
59 <enum name="in">
60 <para>Inbound stream (to the PBX)</para>
61 </enum>
62 <enum name="out">
63 <para>Outbound stream (from the PBX)</para>
64 </enum>
65 <enum name="all">
66 <para>Both streams</para>
67 </enum>
68 </enumlist>
69 </parameter>
70 </syntax>
71 <description>
72 <para>The MUTEAUDIO function can be used to mute inbound (to the PBX) or outbound audio in a call.</para>
73 <example title="Mute incoming audio">
74 exten => s,1,Set(MUTEAUDIO(in)=on)
75 </example>
76 <example title="Do not mute incoming audio">
77 exten => s,1,Set(MUTEAUDIO(in)=off)
78 </example>
79 </description>
80 </function>
81 <manager name="MuteAudio" language="en_US">
82 <synopsis>
83 Mute an audio stream.
84 </synopsis>
85 <syntax>
86 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
87 <parameter name="Channel" required="true">
88 <para>The channel you want to mute.</para>
89 </parameter>
90 <parameter name="Direction" required="true">
91 <enumlist>
92 <enum name="in">
93 <para>Set muting on inbound audio stream. (to the PBX)</para>
94 </enum>
95 <enum name="out">
96 <para>Set muting on outbound audio stream. (from the PBX)</para>
97 </enum>
98 <enum name="all">
99 <para>Set muting on inbound and outbound audio streams.</para>
100 </enum>
101 </enumlist>
102 </parameter>
103 <parameter name="State" required="true">
104 <enumlist>
105 <enum name="on">
106 <para>Turn muting on.</para>
107 </enum>
108 <enum name="off">
109 <para>Turn muting off.</para>
110 </enum>
111 </enumlist>
112 </parameter>
113 </syntax>
114 <description>
115 <para>Mute an incoming or outgoing audio stream on a channel.</para>
116 </description>
117 </manager>
118 ***/
119
120
121static int mute_channel(struct ast_channel *chan, const char *direction, int mute)
122{
123 unsigned int mute_direction = 0;
124 enum ast_frame_type frametype = AST_FRAME_VOICE;
125 int ret = 0;
126
127 if (!strcmp(direction, "in")) {
128 mute_direction = AST_MUTE_DIRECTION_READ;
129 } else if (!strcmp(direction, "out")) {
130 mute_direction = AST_MUTE_DIRECTION_WRITE;
131 } else if (!strcmp(direction, "all")) {
133 } else {
134 return -1;
135 }
136
137 ast_channel_lock(chan);
138
139 if (mute) {
140 ret = ast_channel_suppress(chan, mute_direction, frametype);
141 } else {
142 ret = ast_channel_unsuppress(chan, mute_direction, frametype);
143 }
144
145 ast_channel_unlock(chan);
146
147 return ret;
148}
149
150/*! \brief Mute dialplan function */
151static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
152{
153 if (!chan) {
154 ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
155 return -1;
156 }
157
158 return mute_channel(chan, data, ast_true(value));
159}
160
161/* Function for debugging - might be useful */
163 .name = "MUTEAUDIO",
164 .write = func_mute_write,
165};
166
167static int manager_mutestream(struct mansession *s, const struct message *m)
168{
169 const char *channel = astman_get_header(m, "Channel");
170 const char *id = astman_get_header(m,"ActionID");
171 const char *state = astman_get_header(m,"State");
172 const char *direction = astman_get_header(m,"Direction");
173 char id_text[256];
174 struct ast_channel *c = NULL;
175
176 if (ast_strlen_zero(channel)) {
177 astman_send_error(s, m, "Channel not specified");
178 return 0;
179 }
180 if (ast_strlen_zero(state)) {
181 astman_send_error(s, m, "State not specified");
182 return 0;
183 }
185 astman_send_error(s, m, "Direction not specified");
186 return 0;
187 }
188 /* Ok, we have everything */
189
190 c = ast_channel_get_by_name(channel);
191 if (!c) {
192 astman_send_error(s, m, "No such channel");
193 return 0;
194 }
195
197 astman_send_error(s, m, "Failed to mute/unmute stream");
199 return 0;
200 }
201
203
204 if (!ast_strlen_zero(id)) {
205 snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
206 } else {
207 id_text[0] = '\0';
208 }
209 astman_append(s, "Response: Success\r\n"
210 "%s"
211 "\r\n", id_text);
212 return 0;
213}
214
215
216static int load_module(void)
217{
218 int res;
219
222
224}
225
226static int unload_module(void)
227{
229 /* Unregister AMI actions */
230 ast_manager_unregister("MuteAudio");
231
232 return 0;
233}
234
235AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mute audio stream resources");
Asterisk main include file. File version handling, generic pbx functions.
#define ast_log
Definition: astobj2.c:42
Audiohooks Architecture.
General Asterisk PBX channel definitions.
#define ast_channel_lock(chan)
Definition: channel.h:2968
#define AST_MUTE_DIRECTION_READ
Definition: channel.h:4760
int ast_channel_suppress(struct ast_channel *chan, unsigned int direction, enum ast_frame_type frametype)
Suppress passing of a frame type on a channel.
Definition: channel.c:10799
#define ast_channel_unref(c)
Decrease channel reference count.
Definition: channel.h:3004
int ast_channel_unsuppress(struct ast_channel *chan, unsigned int direction, enum ast_frame_type frametype)
Stop suppressing of a frame type on a channel.
Definition: channel.c:10861
#define AST_MUTE_DIRECTION_WRITE
Definition: channel.h:4761
struct ast_channel * ast_channel_get_by_name(const char *name)
Find a channel by name.
Definition: channel.c:1473
#define ast_channel_unlock(chan)
Definition: channel.h:2969
Generic File Format Support. Should be included by clients of the file handling routines....
direction
void astman_send_error(struct mansession *s, const struct message *m, char *error)
Send error in manager transaction.
Definition: manager.c:1969
const char * astman_get_header(const struct message *m, char *var)
Get header from manager transaction.
Definition: manager.c:1630
void astman_append(struct mansession *s, const char *fmt,...)
Definition: manager.c:1890
int ast_manager_unregister(const char *action)
Unregister a registered manager command.
Definition: manager.c:7608
Configuration File Parser.
Asterisk internal frame definitions.
ast_frame_type
Frame types.
@ AST_FRAME_VOICE
Support for logging to various files, console and syslog Configuration in file logger....
#define LOG_WARNING
The AMI - Asterisk Manager Interface - is a TCP protocol created to manage Asterisk with third-party ...
#define EVENT_FLAG_SYSTEM
Definition: manager.h:75
#define ast_manager_register_xml(action, authority, func)
Register a manager callback using XML documentation to describe the manager.
Definition: manager.h:191
Asterisk module definitions.
#define AST_MODULE_INFO_STANDARD(keystr, desc)
Definition: module.h:581
#define ASTERISK_GPL_KEY
The text the key() function should return.
Definition: module.h:46
@ AST_MODULE_LOAD_SUCCESS
Definition: module.h:70
@ AST_MODULE_LOAD_DECLINE
Module has failed to load, may be in an inconsistent state.
Definition: module.h:78
Options provided by main asterisk program.
Core PBX routines and definitions.
#define ast_custom_function_register(acf)
Register a custom function.
Definition: pbx.h:1558
int ast_custom_function_unregister(struct ast_custom_function *acf)
Unregister a custom function.
static int mute_channel(struct ast_channel *chan, const char *direction, int mute)
static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
Mute dialplan function.
static struct ast_custom_function mute_function
static int load_module(void)
static int unload_module(void)
static int manager_mutestream(struct mansession *s, const struct message *m)
#define NULL
Definition: resample.c:96
int attribute_pure ast_true(const char *val)
Make sure something is true. Determine if a string containing a boolean value is "true"....
Definition: utils.c:2199
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition: strings.h:65
Main Channel structure associated with a channel.
Data structure associated with a custom dialplan function.
Definition: pbx.h:118
const char * name
Definition: pbx.h:119
In case you didn't read that giant block of text above the mansession_session struct,...
Definition: manager.c:326
int value
Definition: syslog.c:37
static struct test_val c
Utility functions.