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asterisk/res/res_tonedetect.c

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v18.15.1
2023-05-25 18:45:57 +00:00
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2021, Naveen Albert
*
* Naveen Albert <asterisk@phreaknet.org>
*
* 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 Tone detection module
*
* \author Naveen Albert <asterisk@phreaknet.org>
*
* \ingroup resources
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
#include <math.h>
#include "asterisk/module.h"
#include "asterisk/frame.h"
#include "asterisk/format_cache.h"
#include "asterisk/channel.h"
#include "asterisk/dsp.h"
#include "asterisk/pbx.h"
#include "asterisk/audiohook.h"
#include "asterisk/app.h"
#include "asterisk/indications.h"
#include "asterisk/conversions.h"
/*** DOCUMENTATION
<application name="WaitForTone" language="en_US">
<since>
<version>16.21.0</version>
<version>18.7.0</version>
<version>19.0.0</version>
</since>
<synopsis>
Wait for tone
</synopsis>
<syntax>
<parameter name="freq" required="true">
<para>Frequency of the tone to wait for.</para>
</parameter>
<parameter name="duration_ms" required="false">
<para>Minimum duration of tone, in ms. Default is 500ms.
Using a minimum duration under 50ms is unlikely to produce
accurate results.</para>
</parameter>
<parameter name="timeout" required="false">
<para>Maximum amount of time, in seconds, to wait for specified tone.
Default is forever.</para>
</parameter>
<parameter name="times" required="false">
<para>Number of times the tone should be detected (subject to the
provided timeout) before returning. Default is 1.</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="d">
<para>Custom decibel threshold to use. Default is 16.</para>
</option>
<option name="s">
<para>Squelch tone.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Waits for a single-frequency tone to be detected before dialplan execution continues.</para>
<variablelist>
<variable name="WAITFORTONESTATUS">
<para>This indicates the result of the wait.</para>
<value name="SUCCESS"/>
<value name="ERROR"/>
<value name="TIMEOUT"/>
<value name="HANGUP"/>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">PlayTones</ref>
</see-also>
</application>
<application name="ToneScan" language="en_US">
<since>
<version>16.23.0</version>
<version>18.9.0</version>
<version>19.1.0</version>
</since>
<synopsis>
Wait for period of time while scanning for call progress tones
</synopsis>
<syntax>
<parameter name="zone" required="false">
<para>Call progress zone. Default is the system default.</para>
</parameter>
<parameter name="timeout" required="false">
<para>Maximum amount of time, in seconds, to wait for call progress
or signal tones. Default is forever.</para>
</parameter>
<parameter name="threshold" required="false">
<para>DSP threshold required for a match. A higher number will
require a longer match and may reduce false positives, at the
expense of false negatives. Default is 1.</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="f">
<para>Enable fax machine detection. By default, this is disabled.</para>
</option>
<option name="v">
<para>Enable voice detection. By default, this is disabled.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Waits for a a distinguishable call progress tone and then exits.
Unlike a conventional scanner, this is not currently capable of
scanning for modem carriers.</para>
<variablelist>
<variable name="TONESCANSTATUS">
This indicates the result of the scan.
<value name="RINGING">
Audible ringback tone
</value>
<value name="BUSY">
Busy tone
</value>
<value name="SIT">
Special Information Tones
</value>
<value name="VOICE">
Human voice detected
</value>
<value name="DTMF">
DTMF digit
</value>
<value name="FAX">
Fax (answering)
</value>
<value name="MODEM">
Modem (answering)
</value>
<value name="DIALTONE">
Dial tone
</value>
<value name="NUT">
UK Number Unobtainable tone
</value>
<value name="TIMEOUT">
Timeout reached before any positive detection
</value>
<value name="HANGUP">
Caller hung up before any positive detection
</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">WaitForTone</ref>
</see-also>
</application>
<function name="TONE_DETECT" language="en_US">
<since>
<version>16.21.0</version>
<version>18.7.0</version>
<version>19.0.0</version>
</since>
<synopsis>
Asynchronously detects a tone
</synopsis>
<syntax>
<parameter name="freq" required="true">
<para>Frequency of the tone to detect. To disable frequency
detection completely (e.g. for signal detection only),
specify 0 for the frequency.</para>
</parameter>
<parameter name="duration_ms" required="false">
<para>Minimum duration of tone, in ms. Default is 500ms.
Using a minimum duration under 50ms is unlikely to produce
accurate results.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Match immediately on Special Information Tones, instead of or in addition
to a particular frequency.</para>
</option>
<option name="b">
<para>Match immediately on a busy signal, instead of or in addition to
a particular frequency.</para>
</option>
<option name="c">
<para>Match immediately on a dial tone, instead of or in addition to
a particular frequency.</para>
</option>
<option name="d">
<para>Custom decibel threshold to use. Default is 16.</para>
</option>
<option name="g">
<para>Go to the specified context,exten,priority if tone is received on this channel.
Detection will not end automatically.</para>
</option>
<option name="h">
<para>Go to the specified context,exten,priority if tone is transmitted on this channel.
Detection will not end automatically.</para>
</option>
<option name="n">
<para>Number of times the tone should be detected (subject to the
provided timeout) before going to the destination provided in the <literal>g</literal>
or <literal>h</literal> option. Default is 1.</para>
</option>
<option name="r">
<para>Apply to received frames only. Default is both directions.</para>
</option>
<option name="s">
<para>Squelch tone.</para>
</option>
<option name="t">
<para>Apply to transmitted frames only. Default is both directions.</para>
</option>
<option name="x">
<para>Destroy the detector (stop detection).</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The TONE_DETECT function detects a single-frequency tone and keeps
track of how many times the tone has been detected.</para>
<para>When reading this function (instead of writing), supply <literal>tx</literal>
to get the number of times a tone has been detected in the TX direction and
<literal>rx</literal> to get the number of times a tone has been detected in the
RX direction.</para>
<example title="intercept2600">
same => n,Set(TONE_DETECT(2600,1000,g(got-2600,s,1))=) ; detect 2600 Hz
same => n,Wait(15)
same => n,NoOp(${TONE_DETECT(rx)})
</example>
<example title="dropondialtone">
same => n,Set(TONE_DETECT(0,,bg(my-hangup,s,1))=) ; disconnect a call if we hear a busy signal
same => n,Goto(somewhere-else)
same => n(myhangup),Hangup()
</example>
<example title="removedetector">
same => n,Set(TONE_DETECT(0,,x)=) ; remove the detector from the channel
</example>
</description>
</function>
***/
struct detect_information {
struct ast_dsp *dsp;
struct ast_audiohook audiohook;
int freq1;
int freq2;
int duration;
int db;
char *gototx;
char *gotorx;
unsigned short int squelch;
unsigned short int tx;
unsigned short int rx;
int txcount;
int rxcount;
int hitsrequired;
int signalfeatures;
};
enum td_opts {
OPT_TX = (1 << 1),
OPT_RX = (1 << 2),
OPT_END_FILTER = (1 << 3),
OPT_GOTO_RX = (1 << 4),
OPT_GOTO_TX = (1 << 5),
OPT_DECIBEL = (1 << 6),
OPT_SQUELCH = (1 << 7),
OPT_HITS_REQ = (1 << 8),
OPT_SIT = (1 << 9),
OPT_BUSY = (1 << 10),
OPT_DIALTONE = (1 << 11),
};
enum {
OPT_ARG_DECIBEL,
OPT_ARG_GOTO_RX,
OPT_ARG_GOTO_TX,
OPT_ARG_HITS_REQ,
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE,
};
AST_APP_OPTIONS(td_opts, {
AST_APP_OPTION('a', OPT_SIT),
AST_APP_OPTION('b', OPT_BUSY),
AST_APP_OPTION('c', OPT_DIALTONE),
AST_APP_OPTION_ARG('d', OPT_DECIBEL, OPT_ARG_DECIBEL),
AST_APP_OPTION_ARG('g', OPT_GOTO_RX, OPT_ARG_GOTO_RX),
AST_APP_OPTION_ARG('h', OPT_GOTO_TX, OPT_ARG_GOTO_TX),
AST_APP_OPTION_ARG('n', OPT_HITS_REQ, OPT_ARG_HITS_REQ),
AST_APP_OPTION('s', OPT_SQUELCH),
AST_APP_OPTION('t', OPT_TX),
AST_APP_OPTION('r', OPT_RX),
AST_APP_OPTION('x', OPT_END_FILTER),
});
static void destroy_callback(void *data)
{
struct detect_information *di = data;
ast_dsp_free(di->dsp);
if (di->gotorx) {
ast_free(di->gotorx);
}
if (di->gototx) {
ast_free(di->gototx);
}
ast_audiohook_lock(&di->audiohook);
ast_audiohook_detach(&di->audiohook);
ast_audiohook_unlock(&di->audiohook);
ast_audiohook_destroy(&di->audiohook);
ast_free(di);
return;
}
static const struct ast_datastore_info detect_datastore = {
.type = "detect",
.destroy = destroy_callback
};
static int detect_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
{
struct ast_datastore *datastore = NULL;
struct detect_information *di = NULL;
int match = 0;
/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
return 0;
}
/* Grab datastore which contains our gain information */
if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
return 0;
}
di = datastore->data;
if (!frame || frame->frametype != AST_FRAME_VOICE) {
return 0;
}
if (!(direction == AST_AUDIOHOOK_DIRECTION_READ ? di->rx : di->tx)) {
return 0;
}
/* ast_dsp_process may free the frame and return a new one */
frame = ast_frdup(frame);
frame = ast_dsp_process(chan, di->dsp, frame);
if (frame->frametype == AST_FRAME_DTMF) {
char result = frame->subclass.integer;
if (result == 'q') {
int now;
match = 1;
if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
di->rxcount = di->rxcount + 1;
now = di->rxcount;
} else {
di->txcount = di->txcount + 1;
now = di->txcount;
}
ast_debug(1, "TONE_DETECT just got a hit (#%d in this direction, waiting for %d total)\n", now, di->hitsrequired);
if (now >= di->hitsrequired) {
if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
ast_async_parseable_goto(chan, di->gotorx);
} else if (di->gototx) {
ast_async_parseable_goto(chan, di->gototx);
}
}
}
}
if (di->signalfeatures && !match) { /* skip unless there are call progress/signal options */
int tstate, tcount;
tcount = ast_dsp_get_tcount(di->dsp);
tstate = ast_dsp_get_tstate(di->dsp);
if (tstate > 0) {
ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
switch (tstate) {
case DSP_TONE_STATE_DIALTONE:
if (di->signalfeatures & DSP_FEATURE_WAITDIALTONE) {
match = 1;
}
break;
case DSP_TONE_STATE_BUSY:
if (di->signalfeatures & DSP_PROGRESS_BUSY) {
match = 1;
}
break;
case DSP_TONE_STATE_SPECIAL3:
if (di->signalfeatures & DSP_PROGRESS_CONGESTION) {
match = 1;
}
break;
default: /* ignore */
break;
}
if (match) {
if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
ast_async_parseable_goto(chan, di->gotorx);
} else if (di->gototx) {
ast_async_parseable_goto(chan, di->gototx);
} else {
ast_debug(3, "Detected call progress signal, but don't know where to go\n");
}
}
}
}
/* this could be the duplicated frame or a new one, doesn't matter */
ast_frfree(frame);
return 0;
}
static int remove_detect(struct ast_channel *chan)
{
struct ast_datastore *datastore = NULL;
struct detect_information *data;
SCOPED_CHANNELLOCK(chan_lock, chan);
datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL);
if (!datastore) {
ast_log(AST_LOG_WARNING, "Cannot remove TONE_DETECT from %s: TONE_DETECT not currently enabled\n",
ast_channel_name(chan));
return -1;
}
data = datastore->data;
if (ast_audiohook_remove(chan, &data->audiohook)) {
ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT audiohook from channel %s\n", ast_channel_name(chan));
return -1;
}
if (ast_channel_datastore_remove(chan, datastore)) {
ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT datastore from channel %s\n",
ast_channel_name(chan));
return -1;
}
ast_datastore_free(datastore);
return 0;
}
static int freq_parser(char *freqs, int *freq1, int *freq2) {
char *f1, *f2, *f3;
if (ast_strlen_zero(freqs)) {
ast_log(LOG_ERROR, "No frequency specified\n");
return -1;
}
f3 = ast_strdupa(freqs);
f1 = strsep(&f3, "+");
f2 = strsep(&f3, "+");
if (!ast_strlen_zero(f3)) {
ast_log(LOG_WARNING, "Only up to 2 frequencies may be specified: %s\n", freqs);
return -1;
}
if (ast_str_to_int(f1, freq1)) {
ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f1);
return -1;
}
if (*freq1 < 0) {
ast_log(LOG_WARNING, "Sorry, no negative frequencies: %d\n", *freq1);
return -1;
}
if (!ast_strlen_zero(f2)) {
ast_log(LOG_WARNING, "Sorry, currently only 1 frequency is supported\n");
return -1;
/* not supported just yet, but possibly will be in the future */
if (ast_str_to_int(f2, freq2)) {
ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f2);
return -1;
}
if (*freq2 < 1) {
ast_log(LOG_WARNING, "Sorry, positive frequencies only: %d\n", *freq2);
return -1;
}
}
return 0;
}
static char* goto_parser(struct ast_channel *chan, char *loc) {
char *exten, *pri, *context, *parse;
char *dest;
int size;
parse = ast_strdupa(loc);
context = strsep(&parse, ",");
exten = strsep(&parse, ",");
pri = strsep(&parse, ",");
if (!exten) {
pri = context;
exten = NULL;
context = NULL;
} else if (!pri) {
pri = exten;
exten = context;
context = NULL;
}
ast_channel_lock(chan);
if (ast_strlen_zero(exten)) {
exten = ast_strdupa(ast_channel_exten(chan));
}
if (ast_strlen_zero(context)) {
context = ast_strdupa(ast_channel_context(chan));
}
ast_channel_unlock(chan);
/* size + 3: for 1 null terminator + 2 commas */
size = strlen(context) + strlen(exten) + strlen(pri) + 3;
dest = ast_malloc(size + 1);
if (!dest) {
ast_log(LOG_ERROR, "Failed to parse goto: %s,%s,%s\n", context, exten, pri);
return NULL;
}
snprintf(dest, size, "%s,%s,%s", context, exten, pri);
return dest;
}
static int detect_read(struct ast_channel *chan, const char *cmd, char *data, char *buffer, size_t buflen)
{
struct ast_datastore *datastore = NULL;
struct detect_information *di = NULL;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
ast_channel_unlock(chan);
return -1; /* function not initiated yet, so nothing to read */
} else {
ast_channel_unlock(chan);
di = datastore->data;
}
if (strchr(data, 't')) {
snprintf(buffer, buflen, "%d", di->txcount);
} else if (strchr(data, 'r')) {
snprintf(buffer, buflen, "%d", di->rxcount);
} else {
ast_log(LOG_WARNING, "Invalid direction: %s\n", data);
}
return 0;
}
static int parse_signal_features(struct ast_flags *flags)
{
int features = 0;
if (ast_test_flag(flags, OPT_SIT)) {
features |= DSP_PROGRESS_CONGESTION;
}
if (ast_test_flag(flags, OPT_BUSY)) {
features |= DSP_PROGRESS_BUSY;
}
if (ast_test_flag(flags, OPT_DIALTONE)) {
features |= DSP_FEATURE_WAITDIALTONE;
}
return features;
}
static int detect_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
char *parse;
struct ast_datastore *datastore = NULL;
struct detect_information *di = NULL;
struct ast_flags flags = { 0 };
char *opt_args[OPT_ARG_ARRAY_SIZE];
struct ast_dsp *dsp;
int freq1 = 0, freq2 = 0, duration = 500, db = 16, squelch = 0, hitsrequired = 1;
int signalfeatures = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(freqs);
AST_APP_ARG(duration);
AST_APP_ARG(options);
);
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.options)) {
ast_app_parse_options(td_opts, &flags, opt_args, args.options);
}
if (ast_test_flag(&flags, OPT_END_FILTER)) {
return remove_detect(chan);
}
if (freq_parser(args.freqs, &freq1, &freq2)) {
return -1;
}
if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
return -1;
}
if (ast_test_flag(&flags, OPT_HITS_REQ) && !ast_strlen_zero(opt_args[OPT_ARG_HITS_REQ])) {
if ((ast_str_to_int(opt_args[OPT_ARG_HITS_REQ], &hitsrequired) || hitsrequired < 1)) {
ast_log(LOG_WARNING, "Invalid number hits required: %s\n", opt_args[OPT_ARG_HITS_REQ]);
return -1;
}
}
if (ast_test_flag(&flags, OPT_DECIBEL) && !ast_strlen_zero(opt_args[OPT_ARG_DECIBEL])) {
if ((ast_str_to_int(opt_args[OPT_ARG_DECIBEL], &db) || db < 1)) {
ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_ARG_DECIBEL]);
return -1;
}
}
signalfeatures = parse_signal_features(&flags);
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
if (!(datastore = ast_datastore_alloc(&detect_datastore, NULL))) {
ast_channel_unlock(chan);
return 0;
}
if (!(di = ast_calloc(1, sizeof(*di)))) {
ast_datastore_free(datastore);
ast_channel_unlock(chan);
return 0;
}
ast_audiohook_init(&di->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Tone Detector", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
di->audiohook.manipulate_callback = detect_callback;
if (!(dsp = ast_dsp_new())) {
ast_datastore_free(datastore);
ast_channel_unlock(chan);
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
return -1;
}
di->signalfeatures = signalfeatures; /* we're not including freq detect */
if (freq1 > 0) {
signalfeatures |= DSP_FEATURE_FREQ_DETECT;
ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
}
ast_dsp_set_features(dsp, signalfeatures);
di->dsp = dsp;
di->txcount = 0;
di->rxcount = 0;
ast_debug(1, "Keeping our ears open for %s Hz, %d db\n", args.freqs, db);
datastore->data = di;
ast_channel_datastore_add(chan, datastore);
ast_audiohook_attach(chan, &di->audiohook);
} else {
di = datastore->data;
dsp = di->dsp;
di->signalfeatures = signalfeatures; /* we're not including freq detect */
if (freq1 > 0) {
signalfeatures |= DSP_FEATURE_FREQ_DETECT;
ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
}
ast_dsp_set_features(dsp, signalfeatures);
}
di->duration = duration;
di->gotorx = NULL;
di->gototx = NULL;
/* resolve gotos now, in case a full context,exten,pri wasn't specified */
if (ast_test_flag(&flags, OPT_GOTO_RX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_RX])) {
di->gotorx = goto_parser(chan, opt_args[OPT_ARG_GOTO_RX]);
}
if (ast_test_flag(&flags, OPT_GOTO_TX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_TX])) {
di->gototx = goto_parser(chan, opt_args[OPT_ARG_GOTO_TX]);
}
di->db = db;
di->hitsrequired = hitsrequired;
di->squelch = ast_test_flag(&flags, OPT_SQUELCH);
di->tx = 1;
di->rx = 1;
if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_TX)) {
di->tx = 1;
di->rx = 0;
}
if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_RX)) {
di->rx = 1;
di->tx = 0;
}
ast_channel_unlock(chan);
return 0;
}
enum {
OPT_APP_DECIBEL = (1 << 0),
OPT_APP_SQUELCH = (1 << 1),
};
enum {
OPT_APP_ARG_DECIBEL,
/* note: this entry _MUST_ be the last one in the enum */
OPT_APP_ARG_ARRAY_SIZE,
};
AST_APP_OPTIONS(wait_exec_options, BEGIN_OPTIONS
AST_APP_OPTION_ARG('d', OPT_APP_DECIBEL, OPT_APP_ARG_DECIBEL),
AST_APP_OPTION('s', OPT_APP_SQUELCH),
END_OPTIONS);
static int wait_exec(struct ast_channel *chan, const char *data)
{
char *appdata;
struct ast_flags flags = {0};
char *opt_args[OPT_APP_ARG_ARRAY_SIZE];
double timeoutf = 0;
int freq1 = 0, freq2 = 0, timeout = 0, duration = 500, times = 1, db = 16, squelch = 0;
struct ast_frame *frame = NULL;
struct ast_dsp *dsp;
struct timeval start;
int remaining_time = 0;
int hits = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(freqs);
AST_APP_ARG(duration);
AST_APP_ARG(timeout);
AST_APP_ARG(times);
AST_APP_ARG(options);
);
appdata = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, appdata);
if (!ast_strlen_zero(args.options)) {
ast_app_parse_options(wait_exec_options, &flags, opt_args, args.options);
}
if (freq_parser(args.freqs, &freq1, &freq2)) {
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
return -1;
}
if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeoutf < 0)) {
ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
return -1;
}
timeout = 1000 * timeoutf;
if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
return -1;
}
if (!ast_strlen_zero(args.times) && (ast_str_to_int(args.times, &times) || times < 1)) {
ast_log(LOG_WARNING, "Invalid number of times: %s\n", args.times);
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
return -1;
}
if (ast_test_flag(&flags, OPT_APP_DECIBEL) && !ast_strlen_zero(opt_args[OPT_APP_ARG_DECIBEL])) {
if ((ast_str_to_int(opt_args[OPT_APP_ARG_DECIBEL], &db) || db < 1)) {
ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_APP_ARG_DECIBEL]);
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
return -1;
}
}
squelch = ast_test_flag(&flags, OPT_APP_SQUELCH);
if (!(dsp = ast_dsp_new())) {
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
return -1;
}
ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
ast_debug(1, "Waiting for %s Hz, %d time(s), timeout %d ms, %d db\n", args.freqs, times, timeout, db);
start = ast_tvnow();
do {
if (timeout > 0) {
remaining_time = ast_remaining_ms(start, timeout);
if (remaining_time <= 0) {
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "TIMEOUT");
break;
}
}
if (ast_waitfor(chan, 1000) > 0) {
if (!(frame = ast_read(chan))) {
ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
break;
} else if (frame->frametype == AST_FRAME_VOICE) {
frame = ast_dsp_process(chan, dsp, frame);
if (frame->frametype == AST_FRAME_DTMF) {
char result = frame->subclass.integer;
if (result == 'q') {
hits++;
ast_debug(1, "We just detected %s Hz (hit #%d)\n", args.freqs, hits);
if (hits >= times) {
ast_frfree(frame);
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "SUCCESS");
break;
}
}
}
}
ast_frfree(frame);
} else {
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
}
} while (timeout == 0 || remaining_time > 0);
ast_dsp_free(dsp);
return 0;
}
static char *waitapp = "WaitForTone";
static char *scanapp = "ToneScan";
static int scan_exec(struct ast_channel *chan, const char *data)
{
char *appdata;
double timeoutf = 0;
int timeout = 0;
struct ast_frame *frame = NULL, *frame2 = NULL;
struct ast_dsp *dsp = NULL, *dsp2 = NULL;
struct timeval start;
int remaining_time = 0;
int features, match = 0, fax = 0, voice = 0, threshold = 1;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(zone);
AST_APP_ARG(timeout);
AST_APP_ARG(threshold);
AST_APP_ARG(options);
);
appdata = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, appdata);
if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeout < 0)) {
ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
return -1;
}
if (!ast_strlen_zero(args.threshold) && (ast_str_to_int(args.threshold, &threshold) || threshold < 1)) {
ast_log(LOG_WARNING, "Invalid threshold: %s\n", args.threshold);
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
return -1;
}
timeout = 1000 * timeoutf;
if (!ast_strlen_zero(args.options) && strchr(args.options, 'f')) {
fax = 1;
}
if (!ast_strlen_zero(args.options) && strchr(args.options, 'v')) {
voice = 1;
}
if (!(dsp = ast_dsp_new())) {
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
return -1;
}
if (!ast_strlen_zero(args.zone)) {
if (ast_dsp_set_call_progress_zone(dsp, args.zone)) {
ast_log(LOG_WARNING, "Invalid call progress zone: %s\n", args.zone);
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
ast_dsp_free(dsp);
return -1;
}
}
if (fax) {
if (!(dsp2 = ast_dsp_new())) {
ast_dsp_free(dsp);
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
return -1;
}
}
features = DSP_PROGRESS_RINGING; /* audible ringback tone */
features |= DSP_PROGRESS_BUSY; /* busy signal */
features |= DSP_PROGRESS_CONGESTION; /* SIT tones (not reorder!) */
features |= DSP_PROGRESS_TALK; /* voice. */
features |= DSP_FEATURE_WAITDIALTONE; /* dial tone */
features |= DSP_FEATURE_FREQ_DETECT; /* modem answer */
if (voice) {
features |= DSP_TONE_STATE_TALKING; /* voice */
}
ast_dsp_set_features(dsp, features);
/* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */
ast_dsp_set_freqmode(dsp, 2225, 400, 16, 0); /* this needs to be pretty short, or the progress tones code will think this is voice */
if (fax) { /* fax detect uses same tone detect internals as modem and causes things to not work as intended, so use a separate DSP if needed. */
ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */
ast_dsp_set_faxmode(dsp2, DSP_FAXMODE_DETECT_CED); /* we only care about the answering side (CED), not originating (CNG) */
}
ast_debug(1, "Starting tone scan, timeout: %d ms, threshold: %d\n", timeout, threshold);
start = ast_tvnow();
do {
if (timeout > 0) {
remaining_time = ast_remaining_ms(start, timeout);
if (remaining_time <= 0) {
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "TIMEOUT");
break;
}
}
if (ast_waitfor(chan, 1000) > 0) {
if (!(frame = ast_read(chan))) {
ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
break;
} else if (frame->frametype == AST_FRAME_VOICE) {
if (fax) {
frame2 = ast_frdup(frame);
}
frame = ast_dsp_process(chan, dsp, frame);
if (frame->frametype == AST_FRAME_DTMF) {
char result = frame->subclass.integer;
match = 1;
if (result == 'q') {
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "MODEM");
} else {
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DTMF");
}
} else if (fax) {
char result;
frame2 = ast_dsp_process(chan, dsp2, frame2);
result = frame2->subclass.integer;
if (frame2->frametype == AST_FRAME_DTMF) {
if (result == 'e') {
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "FAX");
match = 1;
} else {
ast_debug(1, "Ignoring inactionable event\n"); /* shouldn't happen */
}
}
ast_frfree(frame2);
}
if (!match) {
int tstate, tcount;
tcount = ast_dsp_get_tcount(dsp);
tstate = ast_dsp_get_tstate(dsp);
if (tstate > 0) {
ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
if (tcount >= threshold) {
match = 1;
switch (tstate) {
case DSP_TONE_STATE_RINGING:
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "RINGING");
break;
case DSP_TONE_STATE_DIALTONE:
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DIALTONE");
break;
case DSP_TONE_STATE_TALKING:
/* even if we don't specify this feature, it's still checked, so we always need to handle it.
Even if we are looking for it, we need to wait a while or tones will be interpreted
as voice, because this will match first (and this should match last). */
if (voice && tcount > 15 && tcount >= threshold) {
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "VOICE");
} else {
match = 0;
}
break;
case DSP_TONE_STATE_BUSY:
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "BUSY");
break;
case DSP_TONE_STATE_SPECIAL3:
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "SIT");
break;
case DSP_TONE_STATE_HUNGUP: /* UK only */
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "NUT");
break;
default:
match = 0;
ast_debug(1, "Something else we weren't expecting? tstate: %d, #%d\n", tstate, tcount);
}
}
}
}
}
ast_frfree(frame);
} else {
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
}
} while (!match && (timeout == 0 || remaining_time > 0));
ast_dsp_free(dsp);
if (dsp2) {
ast_dsp_free(dsp2);
}
return 0;
}
static struct ast_custom_function detect_function = {
.name = "TONE_DETECT",
.read = detect_read,
.write = detect_write,
};
static int unload_module(void)
{
int res;
res = ast_unregister_application(waitapp);
res |= ast_unregister_application(scanapp);
res |= ast_custom_function_unregister(&detect_function);
return res;
}
static int load_module(void)
{
int res;
res = ast_register_application_xml(waitapp, wait_exec);
res |= ast_register_application_xml(scanapp, scan_exec);
res |= ast_custom_function_register(&detect_function);
return res;
}
AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Tone detection module");