--- /dev/null
+/*
+ * 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">
+ <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>
+ <function name="TONE_DETECT" language="en_US">
+ <synopsis>
+ Asynchronously detects a tone
+ </synopsis>
+ <syntax>
+ <parameter name="freq" required="true">
+ <para>Frequency of the tone to detect.</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="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))=)
+ same => n,Wait(15)
+ same => n,NoOp(${TONE_DETECT(rx)})
+ </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;
+};
+
+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),
+};
+
+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_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;
+
+ /* 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;
+ 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);
+ }
+ }
+ }
+ }
+ /* 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 < 1) {
+ ast_log(LOG_WARNING, "Sorry, positive frequencies only: %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 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;
+
+ 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_test_flag(&flags, OPT_END_FILTER)) {
+ return remove_detect(chan);
+ }
+ if (!ast_strlen_zero(args.options)) {
+ ast_app_parse_options(td_opts, &flags, opt_args, args.options);
+ }
+ 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;
+ }
+ }
+
+ 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;
+ }
+ ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
+ ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
+ 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;
+ ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
+ }
+ 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 || timeout < 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 < 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 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_custom_function_unregister(&detect_function);
+
+ return res;
+}
+
+static int load_module(void)
+{
+ int res;
+
+ res = ast_register_application_xml(waitapp, wait_exec);
+ res |= ast_custom_function_register(&detect_function);
+
+ return res;
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Tone detection module");
projects / thirdparty / asterisk.git / commitdiff
