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app_audiofork.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2019, Nadir Hamid
* Copyright (C) 2005 - 2006, Digium, Inc.
*
* Mark Spencer <[email protected]>
* Kevin P. Fleming <[email protected]>
*
* Based on app_muxmon.c provided by
* Anthony Minessale II <[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 AudioFork() - Offload Asterisk audio processing to a Websocket server.
* \ingroup applications
*
* \author Nadir Hamid <[email protected]>
*
* \note Based on app_mixmonitor.c provided by
* asterisk
*/
/*** MODULEINFO
<use type="module">func_periodic_hook</use>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/paths.h" /* use ast_config_AST_MONITOR_DIR */
#include "asterisk/stringfields.h"
#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"
#include "asterisk/callerid.h"
#include "asterisk/mod_format.h"
#include "asterisk/linkedlists.h"
#include "asterisk/test.h"
#include "asterisk/format_cache.h"
#include "asterisk/beep.h"
#include "asterisk/module.h"
#include "asterisk/astobj2.h"
#include "asterisk/pbx.h"
#include "asterisk/http_websocket.h"
#include "asterisk/tcptls.h"
/*** DOCUMENTATION
<application name="AudioFork" language="en_US">
<synopsis>
Forks a raw audio stream to a websocket server.
</synopsis>
<syntax>
<parameter name="wsserver" required="true" argsep=".">
<argument name="wsserver" required="true">
<para>the URL to the websocket server you want to send the audio to. </para>
</argument>
<argument name="extension" required="true" />
</parameter>
<parameter name="options">
<optionlist>
<option name="b">
<para>Only save audio to the file while the channel is bridged.</para>
<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="B">
<para>Play a periodic beep while this call is being recorded.</para>
<argument name="interval"><para>Interval, in seconds. Default is 15.</para></argument>
</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>
<option name="r">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>receive</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</option>
<option name="t">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>transmit</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</option>
<option name="S">
<para>When combined with the <replaceable>r</replaceable> or <replaceable>t</replaceable>
option, inserts silence when necessary to maintain synchronization between the receive
and transmit audio streams.</para>
</option>
<option name="i">
<argument name="chanvar" required="true" />
<para>Stores the AudioFork's ID on this channel variable.</para>
</option>
<option name="p">
<para>Play a beep on the channel that starts the recording.</para>
</option>
<option name="P">
<para>Play a beep on the channel that stops the recording.</para>
</option>
<option name="D">
<para>Direction of audiohook to process - supports in, out, and both</para>
</option>
<option name="T">
<para>comma separated TLS config for secure websocket connections</para>
</option>
</optionlist>
</parameter>
<parameter name="command">
<para>This is executed when the audio fork's hook finishes</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 AudioFork is called.</para>
<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para></warning>
</parameter>
</syntax>
<description>
<para>Forks raw audio to a remote websocket.</para>
<para>This application does not automatically answer and should be preceeded by
an application such as Answer or Progress().</para>
<note><para>AudioFork runs as an audiohook.</para></note>
<variablelist>
<variable name="AUDIOFORK_WSSERVER">
<para>The URL of the websocket server.</para>
</variable>
</variablelist>
<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of ANY of the application's
parameters. You risk a command injection attack executing arbitrary commands
if the untrusted strings aren't filtered to remove dangerous characters. See
function <variable>FILTER()</variable>.</para></warning>
</description>
<see-also>
<ref type="application">AudioFork</ref>
<ref type="application">StopAudioFork</ref>
<ref type="application">PauseMonitor</ref>
<ref type="application">UnpauseMonitor</ref>
<ref type="function">AUDIOHOOK_INHERIT</ref>
</see-also>
</application>
<application name="StopAudioFork" language="en_US">
<synopsis>
Cancels an ongoing audio fork and closes the websocket connection.
</synopsis>
<syntax>
<parameter name="AudioForkID" required="false">
<para>If a valid ID is provided, then this command will stop only that specific
AudioFork.</para>
</parameter>
</syntax>
<description>
<para>Stop an ongoing AudioFork created previously by <literal>AudioFork()</literal>
on the current channel.</para>
</description>
<see-also>
<ref type="application">AudioFork</ref>
</see-also>
</application>
<manager name="AudioForkMute" language="en_US">
<synopsis>
Mute / unMute a AudioFork session.
</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 audio fork 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 AudioFork session.</para>
</description>
</manager>
<manager name="AudioFork" language="en_US">
<synopsis>
Forks a raw audio stream to a websocket server.
</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 record.</para>
</parameter>
<parameter name="WsServer">
<para>The websocket server URL to fork audio to.</para>
</parameter>
<parameter name="options">
<para>Options that apply to the AudioFork in the same way as they
would apply if invoked from the AudioFork application. For a list of
available options, see the documentation for the audiofork application. </para>
</parameter>
<parameter name="Command">
<para>Will be executed when the audio fork has completed.
Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.
All variables will be evaluated at the time AudioFork is called.</para>
<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para></warning>
</parameter>
</syntax>
<description>
<para>This action will fork audio from an ongoing call to the designated websocket serrver.</para>
<variablelist>
<variable name="AUDIOFORK_WSSERVER">
<para>The websocket server URL.</para>
</variable>
</variablelist>
</description>
</manager>
<manager name="StopAudioFork" language="en_US">
<synopsis>
Stops an ongoing AudioFork() session
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The name of the channel monitored.</para>
</parameter>
<parameter name="AudioForkID" required="false">
<para>If a valid ID is provided, then this command will stop only that specific
AudioFork.</para>
</parameter>
</syntax>
<description>
<para>This command stops the audio fork that was created by the <literal>AudioFork</literal>
action.</para>
</description>
</manager>
<function name="AUDIOFORK" language="en_US">
<synopsis>
Retrieve data pertaining to specific instances of AudioFork on a channel.
</synopsis>
<syntax>
<parameter name="id" required="true">
<para>The unique ID of the AudioFork instance. The unique ID can be retrieved through the channel
variable used as an argument to the <replaceable>i</replaceable> option to AudioFork.</para>
</parameter>
<parameter name="key" required="true">
<para>The piece of data to retrieve from the AudioFork.</para>
<enumlist>
<enum name="filename" />
</enumlist>
</parameter>
</syntax>
</function>
***/
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
static const char *const app = "AudioFork";
static const char *const stop_app = "StopAudioFork";
static const char *const audiofork_spy_type = "AudioFork";
struct audiofork {
struct ast_audiohook audiohook;
struct ast_websocket *websocket;
char *wsserver;
struct ast_tls_config *tls_cfg;
char *tcert;
char *tcpvt;
char *tcipher;
char *tca;
char *tcpath;
enum ast_audiohook_direction direction;
char *post_process;
char *name;
ast_callid callid;
unsigned int flags;
struct ast_autochan *autochan;
struct audiofork_ds *audiofork_ds;
/* the below string fields describe data used for creating voicemails from the recording */
AST_DECLARE_STRING_FIELDS(AST_STRING_FIELD(call_context);
AST_STRING_FIELD(call_macrocontext);
AST_STRING_FIELD(call_extension);
AST_STRING_FIELD(call_callerchan);
AST_STRING_FIELD(call_callerid););
int call_priority;
int has_tls;
};
enum audiofork_flags {
MUXFLAG_APPEND = (1 << 1),
MUXFLAG_BRIDGED = (1 << 2),
MUXFLAG_VOLUME = (1 << 3),
MUXFLAG_READVOLUME = (1 << 4),
MUXFLAG_WRITEVOLUME = (1 << 5),
MUXFLAG_COMBINED = (1 << 8),
MUXFLAG_UID = (1 << 9),
MUXFLAG_BEEP = (1 << 11),
MUXFLAG_BEEP_START = (1 << 12),
MUXFLAG_BEEP_STOP = (1 << 13),
MUXFLAG_RWSYNC = (1 << 14),
MUXFLAG_DIRECTION = (1 << 15),
MUXFLAG_TLS = (1 << 16),
};
enum audiofork_args {
OPT_ARG_READVOLUME = 0,
OPT_ARG_WRITEVOLUME,
OPT_ARG_VOLUME,
OPT_ARG_UID,
OPT_ARG_BEEP_INTERVAL,
OPT_ARG_RWSYNC,
OPT_ARG_DIRECTION,
OPT_ARG_TLS,
OPT_ARG_ARRAY_SIZE, /* Always last element of the enum */
};
AST_APP_OPTIONS(audiofork_opts, {
AST_APP_OPTION('a', MUXFLAG_APPEND),
AST_APP_OPTION('b', MUXFLAG_BRIDGED),
AST_APP_OPTION_ARG('B', MUXFLAG_BEEP, OPT_ARG_BEEP_INTERVAL),
AST_APP_OPTION('p', MUXFLAG_BEEP_START),
AST_APP_OPTION('P', MUXFLAG_BEEP_STOP),
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),
AST_APP_OPTION_ARG('i', MUXFLAG_UID, OPT_ARG_UID),
AST_APP_OPTION_ARG('S', MUXFLAG_RWSYNC, OPT_ARG_RWSYNC),
AST_APP_OPTION_ARG('D', MUXFLAG_DIRECTION, OPT_ARG_DIRECTION),
AST_APP_OPTION_ARG('T', MUXFLAG_TLS, OPT_ARG_TLS),
});
struct audiofork_ds {
unsigned int destruction_ok;
ast_cond_t destruction_condition;
ast_mutex_t lock;
/**
* the audio hook we will use for sending raw audio
*/
struct ast_audiohook *audiohook;
unsigned int samp_rate;
char *wsserver;
char *beep_id;
struct ast_tls_config *tls_cfg;
};
static void audiofork_ds_destroy(void *data)
{
struct audiofork_ds *audiofork_ds = data;
ast_mutex_lock(&audiofork_ds->lock);
audiofork_ds->audiohook = NULL;
audiofork_ds->destruction_ok = 1;
ast_free(audiofork_ds->wsserver);
ast_free(audiofork_ds->beep_id);
ast_cond_signal(&audiofork_ds->destruction_condition);
ast_mutex_unlock(&audiofork_ds->lock);
}
static const struct ast_datastore_info audiofork_ds_info = {
.type = "audiofork",
.destroy = audiofork_ds_destroy,
};
static void destroy_monitor_audiohook(struct audiofork *audiofork)
{
if (audiofork->audiofork_ds) {
ast_mutex_lock(&audiofork->audiofork_ds->lock);
audiofork->audiofork_ds->audiohook = NULL;
ast_mutex_unlock(&audiofork->audiofork_ds->lock);
}
/* kill the audiohook. */
ast_audiohook_lock(&audiofork->audiohook);
ast_audiohook_detach(&audiofork->audiohook);
ast_audiohook_unlock(&audiofork->audiohook);
ast_audiohook_destroy(&audiofork->audiohook);
}
static int start_audiofork(struct ast_channel *chan,
struct ast_audiohook *audiohook)
{
if (!chan) {
return -1;
}
return ast_audiohook_attach(chan, audiohook);
}
#define SAMPLES_PER_FRAME 160
static void audiofork_free(struct audiofork *audiofork)
{
if (audiofork) {
if (audiofork->audiofork_ds) {
ast_mutex_destroy(&audiofork->audiofork_ds->lock);
ast_cond_destroy(&audiofork->audiofork_ds->destruction_condition);
ast_free(audiofork->audiofork_ds);
}
ast_free(audiofork->name);
ast_free(audiofork->post_process);
ast_free(audiofork->wsserver);
/* clean stringfields */
ast_string_field_free_memory(audiofork);
ast_free(audiofork);
}
}
static void *audiofork_thread(void *obj)
{
struct audiofork *audiofork = obj;
struct ast_format *format_slin;
enum ast_websocket_result result;
ast_verb(2, "Connecting websocket server at %s\n",
audiofork->audiofork_ds->wsserver);
//check if we're running with TLS
if (audiofork->has_tls == 1) {
audiofork->websocket =
ast_websocket_client_create(audiofork->audiofork_ds->wsserver, "echo", audiofork->tls_cfg,
&result);
} else {
audiofork->websocket =
ast_websocket_client_create(audiofork->audiofork_ds->wsserver, "echo", NULL,
&result);
}
if (result != WS_OK) {
ast_log(LOG_ERROR, "Could not connect to websocket on audio form %s\n",
audiofork->name);
return NULL;
}
/* Keep callid association before any log messages */
if (audiofork->callid) {
ast_callid_threadassoc_add(audiofork->callid);
}
ast_verb(2, "Begin AudioFork Recording %s\n", audiofork->name);
//fs = &audiofork->audiofork_ds->fs;
ast_mutex_lock(&audiofork->audiofork_ds->lock);
format_slin =
ast_format_cache_get_slin_by_rate(audiofork->audiofork_ds->samp_rate);
ast_mutex_unlock(&audiofork->audiofork_ds->lock);
/* The audiohook must enter and exit the loop locked */
ast_audiohook_lock(&audiofork->audiohook);
while (audiofork->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) {
//ast_verb(2, "reading audio hook frame...\n");
struct ast_frame *fr = NULL;
if (!
(fr =
ast_audiohook_read_frame(&audiofork->audiohook, SAMPLES_PER_FRAME,
audiofork->direction, format_slin))) {
ast_audiohook_trigger_wait(&audiofork->audiohook);
if (audiofork->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(&audiofork->audiohook);
struct ast_frame *cur;
//ast_mutex_lock(&audiofork->audiofork_ds->lock);
for (cur = fr; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
//ast_verb(2, "sending audio frame to websocket...\n");
//ast_mutex_lock(&audiofork->audiofork_ds->lock);
if (ast_websocket_write
(audiofork->websocket, AST_WEBSOCKET_OPCODE_BINARY, cur->data.ptr,
cur->datalen)) {
ast_log(LOG_ERROR, "could not write to websocket on audiofork %s.\n",
audiofork->name);
}
}
//ast_mutex_unlock(&audiofork->audiofork_ds->lock);
//
/* All done! free it. */
if (fr) {
ast_frame_free(fr, 0);
}
fr = NULL;
ast_audiohook_lock(&audiofork->audiohook);
}
ast_audiohook_unlock(&audiofork->audiohook);
if (ast_test_flag(audiofork, MUXFLAG_BEEP_STOP)) {
ast_autochan_channel_lock(audiofork->autochan);
ast_stream_and_wait(audiofork->autochan->chan, "beep", "");
ast_autochan_channel_unlock(audiofork->autochan);
}
ast_autochan_destroy(audiofork->autochan);
/* Datastore cleanup. close the filestream and wait for ds destruction */
ast_mutex_lock(&audiofork->audiofork_ds->lock);
if (!audiofork->audiofork_ds->destruction_ok) {
ast_cond_wait(&audiofork->audiofork_ds->destruction_condition,
&audiofork->audiofork_ds->lock);
}
ast_mutex_unlock(&audiofork->audiofork_ds->lock);
/* kill the audiohook */
destroy_monitor_audiohook(audiofork);
if (audiofork->post_process) {
ast_verb(2, "Executing [%s]\n", audiofork->post_process);
ast_safe_system(audiofork->post_process);
}
ast_verb(2, "End AudioFork Recording %s\n", audiofork->name);
ast_test_suite_event_notify("AUDIOFORK_END", "File: %s\r\n",
audiofork->wsserver);
audiofork_free(audiofork);
ast_module_unref(ast_module_info->self);
return NULL;
}
static int setup_audiofork_ds(struct audiofork *audiofork,
struct ast_channel *chan, char **datastore_id,
const char *beep_id)
{
struct ast_datastore *datastore = NULL;
struct audiofork_ds *audiofork_ds;
if (!(audiofork_ds = ast_calloc(1, sizeof(*audiofork_ds)))) {
return -1;
}
if (ast_asprintf(datastore_id, "%p", audiofork_ds) == -1) {
ast_log(LOG_ERROR, "Failed to allocate memory for AudioFork ID.\n");
ast_free(audiofork_ds);
return -1;
}
ast_mutex_init(&audiofork_ds->lock);
ast_cond_init(&audiofork_ds->destruction_condition, NULL);
if (!(datastore = ast_datastore_alloc(&audiofork_ds_info, *datastore_id))) {
ast_mutex_destroy(&audiofork_ds->lock);
ast_cond_destroy(&audiofork_ds->destruction_condition);
ast_free(audiofork_ds);
return -1;
}
if (ast_test_flag(audiofork, MUXFLAG_BEEP_START)) {
ast_autochan_channel_lock(audiofork->autochan);
ast_stream_and_wait(audiofork->autochan->chan, "beep", "");
ast_autochan_channel_unlock(audiofork->autochan);
}
audiofork_ds->samp_rate = 8000;
audiofork_ds->audiohook = &audiofork->audiohook;
audiofork_ds->wsserver = ast_strdup(audiofork->wsserver);
if (!ast_strlen_zero(beep_id)) {
audiofork_ds->beep_id = ast_strdup(beep_id);
}
datastore->data = audiofork_ds;
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
audiofork->audiofork_ds = audiofork_ds;
return 0;
}
static int launch_audiofork_thread(struct ast_channel *chan,
const char *wsserver, unsigned int flags,
enum ast_audiohook_direction direction,
char* tcert,
char* tcpvt,
char* tcipher,
char* tca,
char* tcpath,
int readvol, int writevol,
const char *post_process,
const char *uid_channel_var,
const char *beep_id)
{
pthread_t thread;
struct audiofork *audiofork;
char postprocess2[1024] = "";
char *datastore_id = NULL;
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 = '$';
}
}
ast_channel_lock(chan);
pbx_substitute_variables_helper(chan, p1, postprocess2,
sizeof(postprocess2) - 1);
ast_channel_unlock(chan);
}
/* Pre-allocate audiofork structure and spy */
if (!(audiofork = ast_calloc(1, sizeof(*audiofork)))) {
return -1;
}
/* Now that the struct has been calloced, go ahead and initialize the string fields. */
if (ast_string_field_init(audiofork, 512)) {
audiofork_free(audiofork);
return -1;
}
/* Setup the actual spy before creating our thread */
if (ast_audiohook_init
(&audiofork->audiohook, AST_AUDIOHOOK_TYPE_SPY, audiofork_spy_type, 0)) {
audiofork_free(audiofork);
return -1;
}
/* Copy over flags and channel name */
audiofork->flags = flags;
if (!(audiofork->autochan = ast_autochan_setup(chan))) {
audiofork_free(audiofork);
return -1;
}
if (!ast_strlen_zero(wsserver)) {
ast_verb(2, "setting wsserver to %s\r\n", wsserver);
audiofork->wsserver = ast_strdup(wsserver);
}
ast_verb(2, "setting direction to %d\r\n", direction);
audiofork->direction = direction;
audiofork->has_tls = 0;
if (!ast_strlen_zero(tcert)) {
ast_verb(2, "setting TLS Cert to %s\r\n", tcert);
struct ast_tls_config *ast_tls_config;
audiofork->tls_cfg = ast_calloc(1, sizeof(*ast_tls_config));
audiofork->has_tls = 1;
audiofork->tls_cfg->certfile = tcert;
}
if (!ast_strlen_zero(tcpvt)) {
ast_verb(2, "setting TLS Pvt to %s\r\n", tcpvt);
audiofork->tls_cfg->pvtfile = tcpvt;
}
if (!ast_strlen_zero(tcipher)) {
ast_verb(2, "setting TLS Cipher to %s\r\n", tcipher);
audiofork->tls_cfg->cafile = tca;
}
if (!ast_strlen_zero(tca)) {
ast_verb(2, "setting TLS CA to %s\r\n", tca);
audiofork->tls_cfg->capath= tcpath;
}
if (!ast_strlen_zero(tcpath)) {
ast_verb(2, "setting TLS CA Path to %s\r\n", tcpath);
audiofork->tcpath = tcpath;
}
if (setup_audiofork_ds(audiofork, chan, &datastore_id, beep_id)) {
ast_autochan_destroy(audiofork->autochan);
audiofork_free(audiofork);
ast_free(datastore_id);
return -1;
}
if (!ast_strlen_zero(uid_channel_var)) {
if (datastore_id) {
pbx_builtin_setvar_helper(chan, uid_channel_var, datastore_id);
}
}
ast_free(datastore_id);
audiofork->name = ast_strdup(ast_channel_name(chan));
if (!ast_strlen_zero(postprocess2)) {
audiofork->post_process = ast_strdup(postprocess2);
}
ast_set_flag(&audiofork->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
if ((ast_test_flag(audiofork, MUXFLAG_RWSYNC))) {
ast_set_flag(&audiofork->audiohook, AST_AUDIOHOOK_SUBSTITUTE_SILENCE);
}
if (readvol)
audiofork->audiohook.options.read_volume = readvol;
if (writevol)
audiofork->audiohook.options.write_volume = writevol;
if (start_audiofork(chan, &audiofork->audiohook)) {
ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
audiofork_spy_type, ast_channel_name(chan));
ast_audiohook_destroy(&audiofork->audiohook);
audiofork_free(audiofork);
return -1;
}
/* reference be released at audiofork destruction */
audiofork->callid = ast_read_threadstorage_callid();
return ast_pthread_create_detached_background(&thread, NULL, audiofork_thread,
audiofork);
}
static int audiofork_exec(struct ast_channel *chan, const char *data)
{
int x, readvol = 0, writevol = 0;
char *uid_channel_var = NULL;
char beep_id[64] = "";
unsigned int direction = 2;
struct ast_flags flags = { 0 };
char *parse;
char *tca = NULL;
char *tcert = NULL;
char *tcipher = NULL;
char *tcpvt = NULL;
char *tcpath = NULL;
char *certdata = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(wsserver);
AST_APP_ARG(options); AST_APP_ARG(post_process););
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "AudioFork requires an argument wsserver\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (args.options) {
char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
ast_app_parse_options(audiofork_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 (ast_test_flag(&flags, MUXFLAG_UID)) {
uid_channel_var = opts[OPT_ARG_UID];
}
if (ast_test_flag(&flags, MUXFLAG_BEEP)) {
const char *interval_str = S_OR(opts[OPT_ARG_BEEP_INTERVAL], "15");
unsigned int interval = 15;
if (sscanf(interval_str, "%30u", &interval) != 1) {
ast_log(LOG_WARNING,
"Invalid interval '%s' for periodic beep. Using default of %u\n",
interval_str, interval);
}
if (ast_beep_start(chan, interval, beep_id, sizeof(beep_id))) {
ast_log(LOG_WARNING,
"Unable to enable periodic beep, please ensure func_periodic_hook is loaded.\n");
return -1;
}
}
if (ast_test_flag(&flags, MUXFLAG_DIRECTION)) {
const char *direction_str = S_OR(opts[OPT_ARG_DIRECTION], "both"); //default to both directions
ast_verb(2, "Parsing direction %s\n",
direction_str);
if (strcmp(direction_str, "in")) {
direction = 0;
} else if (strcmp(direction_str, "out")) {
direction = 1;
} else if (strcmp(direction_str, "both")) {
direction = 2;
} else {
direction = 2;
ast_log(LOG_WARNING,
"Invalid direction '%s' given. Using default of %u\n",
direction_str, direction);
}
}
if (ast_test_flag(&flags, MUXFLAG_TLS)) {
certdata = S_OR(opts[OPT_ARG_TLS], "");
ast_verb(2, "Parsing TLS config %s\n",
certdata);
char* pt = strtok (certdata,",");
int pos = 0;
while (pt != NULL) {
if ( pos == 0 ) {
tcert = ast_strdup( pt );
} else if ( pos == 1 ) {
tcpvt = ast_strdup( pt );
} else if ( pos == 2 ) {
tcipher= ast_strdup( pt );
} else if ( pos == 3 ) {
tca= ast_strdup( pt );
} else if ( pos == 4 ) {
tcpath = ast_strdup( pt );
}
pt = strtok (NULL, ",");
pos = pos + 1;
}
ast_verb(2, "Parsing TLS result tcert: %s, tcpvt: %s, tcipher: %s, tca: %s, tcpath: %s\n",
tcert, tcpvt, tcipher, tca, tcpath);
}
}
/* If there are no file writing arguments/options for the mix monitor, send a warning message and return -1 */
if (ast_strlen_zero(args.wsserver)) {
ast_log(LOG_WARNING, "AudioFork requires an argument (wsserver)\n");
return -1;
}
/* If filename exists, try to create directories for it */
if (!(ast_strlen_zero(args.wsserver))) {
}
pbx_builtin_setvar_helper(chan, "AUDIOFORK_WSSERVER", args.wsserver);
/* If launch_monitor_thread works, the module reference must not be released until it is finished. */
ast_module_ref(ast_module_info->self);
if (launch_audiofork_thread(chan,
args.wsserver,
flags.flags,
direction,
tcert,
tcpvt,
tcipher,
tca,
tcpath,
readvol,
writevol,
args.post_process, uid_channel_var, beep_id)) {
ast_module_unref(ast_module_info->self);
}
return 0;
}
static int stop_audiofork_full(struct ast_channel *chan, const char *data)
{
struct ast_datastore *datastore = NULL;
char *parse = "";
struct audiofork_ds *audiofork_ds;
const char *beep_id = NULL;
AST_DECLARE_APP_ARGS(args, AST_APP_ARG(audioforkid););
if (!ast_strlen_zero(data)) {
parse = ast_strdupa(data);
}
AST_STANDARD_APP_ARGS(args, parse);
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &audiofork_ds_info,
S_OR(args.audioforkid, NULL));
if (!datastore) {
ast_channel_unlock(chan);
return -1;
}
audiofork_ds = datastore->data;
ast_mutex_lock(&audiofork_ds->lock);
/* The audiofork thread may be waiting on the audiohook trigger.
* In order to exit from the audiofork loop before waiting on channel
* destruction, poke the audiohook trigger. */
if (audiofork_ds->audiohook) {
if (audiofork_ds->audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
ast_audiohook_update_status(audiofork_ds->audiohook,
AST_AUDIOHOOK_STATUS_SHUTDOWN);
}
ast_audiohook_lock(audiofork_ds->audiohook);
ast_cond_signal(&audiofork_ds->audiohook->trigger);
ast_audiohook_unlock(audiofork_ds->audiohook);
audiofork_ds->audiohook = NULL;
}
if (!ast_strlen_zero(audiofork_ds->beep_id)) {
beep_id = ast_strdupa(audiofork_ds->beep_id);
}
ast_mutex_unlock(&audiofork_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);
if (!ast_strlen_zero(beep_id)) {
ast_beep_stop(chan, beep_id);
}
return 0;
}
static int stop_audiofork_exec(struct ast_channel *chan, const char *data)