res_tonedetect.c

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/*
 * 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/stasis_channels.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="p">
                        <para>Match immediately on audible ringback tone, instead of or in addition to
                        a particular frequency.</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),
    OPT_RINGING = (1 << 12),
};
 
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('p', OPT_RINGING),
    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;
    struct stasis_message *message;
    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) {
                message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan), ast_channel_tone_detect(), NULL);
 
                if (!message) {
                    ast_log(LOG_ERROR, "Unable to publish tone detected event for ARI on channel '%s'", ast_channel_name(chan));
                    return 1;
                } else {
                    stasis_publish(ast_channel_topic(chan), message);
                    ao2_ref(message, -1);
                }
 
                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_RINGING:
                if (di->signalfeatures & DSP_PROGRESS_RINGING) {
                    ast_debug(1, "Detected ringing on %s in %s direction\n", ast_channel_name(chan),
                        direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
                    match = 1;
                }
                break;
            case DSP_TONE_STATE_DIALTONE:
                if (di->signalfeatures & DSP_FEATURE_WAITDIALTONE) {
                    ast_debug(1, "Detected dial tone on %s in %s direction\n", ast_channel_name(chan),
                        direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
                    match = 1;
                }
                break;
            case DSP_TONE_STATE_BUSY:
                if (di->signalfeatures & DSP_PROGRESS_BUSY) {
                    ast_debug(1, "Detected busy tone on %s in %s direction\n", ast_channel_name(chan),
                        direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
                    match = 1;
                }
                break;
            case DSP_TONE_STATE_SPECIAL3:
                if (di->signalfeatures & DSP_PROGRESS_CONGESTION) {
                    ast_debug(1, "Detected SIT on %s in %s direction\n", ast_channel_name(chan),
                        direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
                    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 in %s direction, but don't know where to go\n",
                        direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
                }
            }
        }
    }
    /* 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;
    }
    if (ast_test_flag(flags, OPT_RINGING)) {
        features |= DSP_PROGRESS_RINGING;
    }
 
    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");