Asterisk - The Open Source Telephony Project GIT-master-77d630f
res_tonedetect.c
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1/*
2 * Asterisk -- An open source telephony toolkit.
3 *
4 * Copyright (C) 2021, Naveen Albert
5 *
6 * Naveen Albert <asterisk@phreaknet.org>
7 *
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
13 *
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
17 */
18
19/*! \file
20 *
21 * \brief Tone detection module
22 *
23 * \author Naveen Albert <asterisk@phreaknet.org>
24 *
25 * \ingroup resources
26 */
27
28/*** MODULEINFO
29 <support_level>extended</support_level>
30 ***/
31
32#include "asterisk.h"
33
34#include <math.h>
35
36#include "asterisk/module.h"
37#include "asterisk/frame.h"
39#include "asterisk/channel.h"
40#include "asterisk/dsp.h"
41#include "asterisk/pbx.h"
43#include "asterisk/audiohook.h"
44#include "asterisk/app.h"
47
48/*** DOCUMENTATION
49 <application name="WaitForTone" language="en_US">
50 <since>
51 <version>16.21.0</version>
52 <version>18.7.0</version>
53 </since>
54 <synopsis>
55 Wait for tone
56 </synopsis>
57 <syntax>
58 <parameter name="freq" required="true">
59 <para>Frequency of the tone to wait for.</para>
60 </parameter>
61 <parameter name="duration_ms" required="false">
62 <para>Minimum duration of tone, in ms. Default is 500ms.
63 Using a minimum duration under 50ms is unlikely to produce
64 accurate results.</para>
65 </parameter>
66 <parameter name="timeout" required="false">
67 <para>Maximum amount of time, in seconds, to wait for specified tone.
68 Default is forever.</para>
69 </parameter>
70 <parameter name="times" required="false">
71 <para>Number of times the tone should be detected (subject to the
72 provided timeout) before returning. Default is 1.</para>
73 </parameter>
74 <parameter name="options" required="false">
75 <optionlist>
76 <option name="d">
77 <para>Custom decibel threshold to use. Default is 16.</para>
78 </option>
79 <option name="s">
80 <para>Squelch tone.</para>
81 </option>
82 </optionlist>
83 </parameter>
84 </syntax>
85 <description>
86 <para>Waits for a single-frequency tone to be detected before dialplan execution continues.</para>
87 <variablelist>
88 <variable name="WAITFORTONESTATUS">
89 <para>This indicates the result of the wait.</para>
90 <value name="SUCCESS"/>
91 <value name="ERROR"/>
92 <value name="TIMEOUT"/>
93 <value name="HANGUP"/>
94 </variable>
95 </variablelist>
96 </description>
97 <see-also>
98 <ref type="application">PlayTones</ref>
99 </see-also>
100 </application>
101 <application name="ToneScan" language="en_US">
102 <since>
103 <version>16.23.0</version>
104 <version>18.9.0</version>
105 <version>19.1.0</version>
106 </since>
107 <synopsis>
108 Wait for period of time while scanning for call progress tones
109 </synopsis>
110 <syntax>
111 <parameter name="zone" required="false">
112 <para>Call progress zone. Default is the system default.</para>
113 </parameter>
114 <parameter name="timeout" required="false">
115 <para>Maximum amount of time, in seconds, to wait for call progress
116 or signal tones. Default is forever.</para>
117 </parameter>
118 <parameter name="threshold" required="false">
119 <para>DSP threshold required for a match. A higher number will
120 require a longer match and may reduce false positives, at the
121 expense of false negatives. Default is 1.</para>
122 </parameter>
123 <parameter name="options" required="false">
124 <optionlist>
125 <option name="f">
126 <para>Enable fax machine detection. By default, this is disabled.</para>
127 </option>
128 <option name="v">
129 <para>Enable voice detection. By default, this is disabled.</para>
130 </option>
131 </optionlist>
132 </parameter>
133 </syntax>
134 <description>
135 <para>Waits for a a distinguishable call progress tone and then exits.
136 Unlike a conventional scanner, this is not currently capable of
137 scanning for modem carriers.</para>
138 <variablelist>
139 <variable name="TONESCANSTATUS">
140 This indicates the result of the scan.
141 <value name="RINGING">
142 Audible ringback tone
143 </value>
144 <value name="BUSY">
145 Busy tone
146 </value>
147 <value name="SIT">
148 Special Information Tones
149 </value>
150 <value name="VOICE">
151 Human voice detected
152 </value>
153 <value name="DTMF">
154 DTMF digit
155 </value>
156 <value name="FAX">
157 Fax (answering)
158 </value>
159 <value name="MODEM">
160 Modem (answering)
161 </value>
162 <value name="DIALTONE">
163 Dial tone
164 </value>
165 <value name="NUT">
166 UK Number Unobtainable tone
167 </value>
168 <value name="TIMEOUT">
169 Timeout reached before any positive detection
170 </value>
171 <value name="HANGUP">
172 Caller hung up before any positive detection
173 </value>
174 </variable>
175 </variablelist>
176 </description>
177 <see-also>
178 <ref type="application">WaitForTone</ref>
179 </see-also>
180 </application>
181 <function name="TONE_DETECT" language="en_US">
182 <since>
183 <version>16.21.0</version>
184 <version>18.7.0</version>
185 </since>
186 <synopsis>
187 Asynchronously detects a tone
188 </synopsis>
189 <syntax>
190 <parameter name="freq" required="true">
191 <para>Frequency of the tone to detect. To disable frequency
192 detection completely (e.g. for signal detection only),
193 specify 0 for the frequency.</para>
194 </parameter>
195 <parameter name="duration_ms" required="false">
196 <para>Minimum duration of tone, in ms. Default is 500ms.
197 Using a minimum duration under 50ms is unlikely to produce
198 accurate results.</para>
199 </parameter>
200 <parameter name="options">
201 <optionlist>
202 <option name="a">
203 <para>Match immediately on Special Information Tones, instead of or in addition
204 to a particular frequency.</para>
205 </option>
206 <option name="b">
207 <para>Match immediately on a busy signal, instead of or in addition to
208 a particular frequency.</para>
209 </option>
210 <option name="c">
211 <para>Match immediately on a dial tone, instead of or in addition to
212 a particular frequency.</para>
213 </option>
214 <option name="d">
215 <para>Custom decibel threshold to use. Default is 16.</para>
216 </option>
217 <option name="e">
218 <para>Automatically end detection after desired number of hits (as specified in the <literal>n</literal> option.</para>
219 </option>
220 <option name="g">
221 <para>Go to the specified context,exten,priority if tone is received on this channel.
222 Detection will not end automatically, unless the <literal>e</literal> option is also specified.</para>
223 </option>
224 <option name="h">
225 <para>Go to the specified context,exten,priority if tone is transmitted on this channel.
226 Detection will not end automatically, unless the <literal>e</literal> option is also specified.</para>
227 </option>
228 <option name="n">
229 <para>Number of times the tone should be detected (subject to the
230 provided timeout) before going to the destination provided in the <literal>g</literal>
231 or <literal>h</literal> option. Default is 1.</para>
232 </option>
233 <option name="p">
234 <para>Match immediately on audible ringback tone, instead of or in addition to
235 a particular frequency.</para>
236 </option>
237 <option name="r">
238 <para>Apply to received frames only. Default is both directions.</para>
239 </option>
240 <option name="s">
241 <para>Squelch tone.</para>
242 </option>
243 <option name="t">
244 <para>Apply to transmitted frames only. Default is both directions.</para>
245 </option>
246 <option name="x">
247 <para>Destroy the detector (stop detection).</para>
248 </option>
249 </optionlist>
250 </parameter>
251 </syntax>
252 <description>
253 <para>The TONE_DETECT function detects a single-frequency tone and keeps
254 track of how many times the tone has been detected.</para>
255 <para>When reading this function (instead of writing), supply <literal>tx</literal>
256 to get the number of times a tone has been detected in the TX direction and
257 <literal>rx</literal> to get the number of times a tone has been detected in the
258 RX direction.</para>
259 <example title="intercept2600">
260 same => n,Set(TONE_DETECT(2600,1000,g(got-2600,s,1))=) ; detect 2600 Hz
261 same => n,Wait(15)
262 same => n,NoOp(${TONE_DETECT(rx)})
263 </example>
264 <example title="dropondialtone">
265 same => n,Set(TONE_DETECT(0,,bg(my-hangup,s,1))=) ; disconnect a call if we hear a busy signal
266 same => n,Goto(somewhere-else)
267 same => n(myhangup),Hangup()
268 </example>
269 <example title="removedetector">
270 same => n,Set(TONE_DETECT(0,,x)=) ; remove the detector from the channel
271 </example>
272 </description>
273 </function>
274 ***/
275
277 struct ast_dsp *dsp;
279 int freq1;
280 int freq2;
282 int db;
283 char *gototx;
284 char *gotorx;
285 unsigned int squelch:1;
286 unsigned int tx:1;
287 unsigned int rx:1;
288 unsigned int autoend:1;
289 unsigned int matchmet:1;
294};
295
297 OPT_TX = (1 << 1),
298 OPT_RX = (1 << 2),
299 OPT_END_FILTER = (1 << 3),
300 OPT_GOTO_RX = (1 << 4),
301 OPT_GOTO_TX = (1 << 5),
302 OPT_DECIBEL = (1 << 6),
303 OPT_SQUELCH = (1 << 7),
304 OPT_HITS_REQ = (1 << 8),
305 OPT_SIT = (1 << 9),
306 OPT_BUSY = (1 << 10),
307 OPT_DIALTONE = (1 << 11),
308 OPT_RINGING = (1 << 12),
309 OPT_AUTOEND = (1 << 13),
310};
311
312enum {
317 /* note: this entry _MUST_ be the last one in the enum */
319};
320
335});
336
337static void destroy_callback(void *data)
338{
339 struct detect_information *di = data;
340 ast_dsp_free(di->dsp);
341 if (di->gotorx) {
342 ast_free(di->gotorx);
343 }
344 if (di->gototx) {
345 ast_free(di->gototx);
346 }
347 ast_audiohook_lock(&di->audiohook);
348 ast_audiohook_detach(&di->audiohook);
349 ast_audiohook_unlock(&di->audiohook);
350 ast_audiohook_destroy(&di->audiohook);
351 ast_free(di);
352 return;
353}
354
356 .type = "detect",
357 .destroy = destroy_callback
358};
359
360static int detect_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
361{
362 struct ast_datastore *datastore = NULL;
363 struct detect_information *di = NULL;
364 struct stasis_message *message;
365 int match = 0;
366
367 /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
368 if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
369 return 0;
370 }
371
372 /* Grab datastore which contains our gain information */
373 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
374 return 0;
375 }
376
377 di = datastore->data;
378
379 if (!frame || frame->frametype != AST_FRAME_VOICE) {
380 return 0;
381 }
382
383 /* If we've detected a match, it is very likely that we could detect additional matches,
384 * which is a problem if the 'g' or 'h' options were used to trigger a redirect,
385 * since if we detect again before disabling TONE_DETECT, we could redirect multiple times.
386 * If we don't need to detect anything further, just exit early until the user disables it. */
387 if (di->matchmet && di->autoend) {
388 return 0;
389 }
390
391 if (!(direction == AST_AUDIOHOOK_DIRECTION_READ ? di->rx : di->tx)) {
392 return 0;
393 }
394
395 /* ast_dsp_process may free the frame and return a new one */
396 frame = ast_frdup(frame);
397 frame = ast_dsp_process(chan, di->dsp, frame);
398 if (frame->frametype == AST_FRAME_DTMF) {
399 char result = frame->subclass.integer;
400 if (result == 'q') {
401 int now;
402 match = 1;
404 di->rxcount = di->rxcount + 1;
405 now = di->rxcount;
406 } else {
407 di->txcount = di->txcount + 1;
408 now = di->txcount;
409 }
410 ast_debug(1, "TONE_DETECT just got a hit (#%d in this direction, waiting for %d total)\n", now, di->hitsrequired);
411 if (now >= di->hitsrequired) {
413
414 if (!message) {
415 ast_log(LOG_ERROR, "Unable to publish tone detected event for ARI on channel '%s'", ast_channel_name(chan));
416 } else {
418 ao2_ref(message, -1);
419 }
420
421 di->matchmet = 1;
422 if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
423 ast_async_parseable_goto(chan, di->gotorx);
424 } else if (di->gototx) {
425 ast_async_parseable_goto(chan, di->gototx);
426 }
427 }
428 }
429 }
430 if (di->signalfeatures && !match) { /* skip unless there are call progress/signal options */
431 int tstate, tcount;
432 tcount = ast_dsp_get_tcount(di->dsp);
433 tstate = ast_dsp_get_tstate(di->dsp);
434 if (tstate > 0) {
435 ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
436 switch (tstate) {
438 if (di->signalfeatures & DSP_PROGRESS_RINGING) {
439 ast_debug(1, "Detected ringing on %s in %s direction\n", ast_channel_name(chan),
440 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
441 match = 1;
442 }
443 break;
445 if (di->signalfeatures & DSP_FEATURE_WAITDIALTONE) {
446 ast_debug(1, "Detected dial tone on %s in %s direction\n", ast_channel_name(chan),
447 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
448 match = 1;
449 }
450 break;
452 if (di->signalfeatures & DSP_PROGRESS_BUSY) {
453 ast_debug(1, "Detected busy tone on %s in %s direction\n", ast_channel_name(chan),
454 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
455 match = 1;
456 }
457 break;
459 if (di->signalfeatures & DSP_PROGRESS_CONGESTION) {
460 ast_debug(1, "Detected SIT on %s in %s direction\n", ast_channel_name(chan),
461 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
462 match = 1;
463 }
464 break;
465 default: /* ignore */
466 break;
467 }
468 if (match) {
469 if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
470 ast_async_parseable_goto(chan, di->gotorx);
471 } else if (di->gototx) {
472 ast_async_parseable_goto(chan, di->gototx);
473 } else {
474 ast_debug(3, "Detected call progress signal in %s direction, but don't know where to go\n",
475 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
476 }
477 }
478 }
479 }
480 /* this could be the duplicated frame or a new one, doesn't matter */
481 ast_frfree(frame);
482 return 0;
483}
484
485static int remove_detect(struct ast_channel *chan)
486{
487 struct ast_datastore *datastore = NULL;
488 struct detect_information *data;
489 SCOPED_CHANNELLOCK(chan_lock, chan);
490
492 if (!datastore) {
493 ast_log(AST_LOG_WARNING, "Cannot remove TONE_DETECT from %s: TONE_DETECT not currently enabled\n",
494 ast_channel_name(chan));
495 return -1;
496 }
497 data = datastore->data;
498
499 if (ast_audiohook_remove(chan, &data->audiohook)) {
500 ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT audiohook from channel %s\n", ast_channel_name(chan));
501 return -1;
502 }
503
504 if (ast_channel_datastore_remove(chan, datastore)) {
505 ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT datastore from channel %s\n",
506 ast_channel_name(chan));
507 return -1;
508 }
509 ast_datastore_free(datastore);
510
511 return 0;
512}
513
514static int freq_parser(char *freqs, int *freq1, int *freq2) {
515 char *f1, *f2, *f3;
516 if (ast_strlen_zero(freqs)) {
517 ast_log(LOG_ERROR, "No frequency specified\n");
518 return -1;
519 }
520 f3 = ast_strdupa(freqs);
521 f1 = strsep(&f3, "+");
522 f2 = strsep(&f3, "+");
523 if (!ast_strlen_zero(f3)) {
524 ast_log(LOG_WARNING, "Only up to 2 frequencies may be specified: %s\n", freqs);
525 return -1;
526 }
527 if (ast_str_to_int(f1, freq1)) {
528 ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f1);
529 return -1;
530 }
531 if (*freq1 < 0) {
532 ast_log(LOG_WARNING, "Sorry, no negative frequencies: %d\n", *freq1);
533 return -1;
534 }
535 if (!ast_strlen_zero(f2)) {
536 ast_log(LOG_WARNING, "Sorry, currently only 1 frequency is supported\n");
537 return -1;
538 /* not supported just yet, but possibly will be in the future */
539 if (ast_str_to_int(f2, freq2)) {
540 ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f2);
541 return -1;
542 }
543 if (*freq2 < 1) {
544 ast_log(LOG_WARNING, "Sorry, positive frequencies only: %d\n", *freq2);
545 return -1;
546 }
547 }
548 return 0;
549}
550
551static char* goto_parser(struct ast_channel *chan, char *loc) {
552 char *exten, *pri, *context, *parse;
553 char *dest;
554 int size;
555 parse = ast_strdupa(loc);
556 context = strsep(&parse, ",");
557 exten = strsep(&parse, ",");
558 pri = strsep(&parse, ",");
559 if (!exten) {
560 pri = context;
561 exten = NULL;
562 context = NULL;
563 } else if (!pri) {
564 pri = exten;
565 exten = context;
566 context = NULL;
567 }
568 ast_channel_lock(chan);
569 if (ast_strlen_zero(exten)) {
570 exten = ast_strdupa(ast_channel_exten(chan));
571 }
574 }
575 ast_channel_unlock(chan);
576
577 /* size + 3: for 1 null terminator + 2 commas */
578 size = strlen(context) + strlen(exten) + strlen(pri) + 3;
579 dest = ast_malloc(size + 1);
580 if (!dest) {
581 ast_log(LOG_ERROR, "Failed to parse goto: %s,%s,%s\n", context, exten, pri);
582 return NULL;
583 }
584 snprintf(dest, size, "%s,%s,%s", context, exten, pri);
585 return dest;
586}
587
588static int detect_read(struct ast_channel *chan, const char *cmd, char *data, char *buffer, size_t buflen)
589{
590 struct ast_datastore *datastore = NULL;
591 struct detect_information *di = NULL;
592
593 if (!chan) {
594 ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
595 return -1;
596 }
597
598 ast_channel_lock(chan);
599 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
600 ast_channel_unlock(chan);
601 return -1; /* function not initiated yet, so nothing to read */
602 } else {
603 ast_channel_unlock(chan);
604 di = datastore->data;
605 }
606
607 if (strchr(data, 't')) {
608 snprintf(buffer, buflen, "%d", di->txcount);
609 } else if (strchr(data, 'r')) {
610 snprintf(buffer, buflen, "%d", di->rxcount);
611 } else {
612 ast_log(LOG_WARNING, "Invalid direction: %s\n", data);
613 }
614
615 return 0;
616}
617
618static int parse_signal_features(struct ast_flags *flags)
619{
620 int features = 0;
621
622 if (ast_test_flag(flags, OPT_SIT)) {
623 features |= DSP_PROGRESS_CONGESTION;
624 }
625 if (ast_test_flag(flags, OPT_BUSY)) {
626 features |= DSP_PROGRESS_BUSY;
627 }
628 if (ast_test_flag(flags, OPT_DIALTONE)) {
629 features |= DSP_FEATURE_WAITDIALTONE;
630 }
631 if (ast_test_flag(flags, OPT_RINGING)) {
632 features |= DSP_PROGRESS_RINGING;
633 }
634
635 return features;
636}
637
638static int detect_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
639{
640 char *parse;
641 struct ast_datastore *datastore = NULL;
642 struct detect_information *di = NULL;
643 struct ast_flags flags = { 0 };
644 char *opt_args[OPT_ARG_ARRAY_SIZE];
645 struct ast_dsp *dsp;
646 int freq1 = 0, freq2 = 0, duration = 500, db = 16, squelch = 0, hitsrequired = 1;
647 int signalfeatures = 0;
648
651 AST_APP_ARG(duration);
653 );
654
655 if (!chan) {
656 ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
657 return -1;
658 }
659 parse = ast_strdupa(data);
661
662 if (!ast_strlen_zero(args.options)) {
663 ast_app_parse_options(td_opts, &flags, opt_args, args.options);
664 }
665 if (ast_test_flag(&flags, OPT_END_FILTER)) {
666 return remove_detect(chan);
667 }
668 if (freq_parser(args.freqs, &freq1, &freq2)) {
669 return -1;
670 }
671 if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
672 ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
673 return -1;
674 }
675 if (ast_test_flag(&flags, OPT_HITS_REQ) && !ast_strlen_zero(opt_args[OPT_ARG_HITS_REQ])) {
676 if ((ast_str_to_int(opt_args[OPT_ARG_HITS_REQ], &hitsrequired) || hitsrequired < 1)) {
677 ast_log(LOG_WARNING, "Invalid number hits required: %s\n", opt_args[OPT_ARG_HITS_REQ]);
678 return -1;
679 }
680 }
681 if (ast_test_flag(&flags, OPT_DECIBEL) && !ast_strlen_zero(opt_args[OPT_ARG_DECIBEL])) {
682 if ((ast_str_to_int(opt_args[OPT_ARG_DECIBEL], &db) || db < 1)) {
683 ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_ARG_DECIBEL]);
684 return -1;
685 }
686 }
687 signalfeatures = parse_signal_features(&flags);
688
689 ast_channel_lock(chan);
690 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
691 if (!(datastore = ast_datastore_alloc(&detect_datastore, NULL))) {
692 ast_channel_unlock(chan);
693 return 0;
694 }
695 if (!(di = ast_calloc(1, sizeof(*di)))) {
696 ast_datastore_free(datastore);
697 ast_channel_unlock(chan);
698 return 0;
699 }
701 di->audiohook.manipulate_callback = detect_callback;
702 if (!(dsp = ast_dsp_new())) {
703 ast_datastore_free(datastore);
704 ast_channel_unlock(chan);
705 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
706 return -1;
707 }
708 di->signalfeatures = signalfeatures; /* we're not including freq detect */
709 if (freq1 > 0) {
710 signalfeatures |= DSP_FEATURE_FREQ_DETECT;
711 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
712 }
713 ast_dsp_set_features(dsp, signalfeatures);
714 di->dsp = dsp;
715 di->txcount = 0;
716 di->rxcount = 0;
717 ast_debug(1, "Keeping our ears open for %s Hz, %d db\n", args.freqs, db);
718 datastore->data = di;
719 ast_channel_datastore_add(chan, datastore);
720 ast_audiohook_attach(chan, &di->audiohook);
721 } else {
722 di = datastore->data;
723 dsp = di->dsp;
724 di->signalfeatures = signalfeatures; /* we're not including freq detect */
725 if (freq1 > 0) {
726 signalfeatures |= DSP_FEATURE_FREQ_DETECT;
727 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
728 }
729 ast_dsp_set_features(dsp, signalfeatures);
730 }
731 di->duration = duration;
732 di->gotorx = NULL;
733 di->gototx = NULL;
734 /* resolve gotos now, in case a full context,exten,pri wasn't specified */
735 if (ast_test_flag(&flags, OPT_GOTO_RX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_RX])) {
736 di->gotorx = goto_parser(chan, opt_args[OPT_ARG_GOTO_RX]);
737 }
738 if (ast_test_flag(&flags, OPT_GOTO_TX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_TX])) {
739 di->gototx = goto_parser(chan, opt_args[OPT_ARG_GOTO_TX]);
740 }
741 di->db = db;
742 di->hitsrequired = hitsrequired;
743 di->squelch = ast_test_flag(&flags, OPT_SQUELCH) ? 1 : 0;
744 di->autoend = ast_test_flag(&flags, OPT_AUTOEND) ? 1 : 0;
745 di->tx = 1;
746 di->rx = 1;
747 if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_TX)) {
748 di->tx = 1;
749 di->rx = 0;
750 }
751 if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_RX)) {
752 di->rx = 1;
753 di->tx = 0;
754 }
755 ast_channel_unlock(chan);
756
757 return 0;
758}
759
760enum {
761 OPT_APP_DECIBEL = (1 << 0),
762 OPT_APP_SQUELCH = (1 << 1),
763};
764
765enum {
767 /* note: this entry _MUST_ be the last one in the enum */
769};
770
775
776static int wait_exec(struct ast_channel *chan, const char *data)
777{
778 char *appdata;
779 struct ast_flags flags = {0};
780 char *opt_args[OPT_APP_ARG_ARRAY_SIZE];
781 double timeoutf = 0;
782 int freq1 = 0, freq2 = 0, timeout = 0, duration = 500, times = 1, db = 16, squelch = 0;
783 struct ast_frame *frame = NULL;
784 struct ast_dsp *dsp;
785 struct timeval start;
786 int remaining_time = 0;
787 int hits = 0;
789 AST_APP_ARG(freqs);
790 AST_APP_ARG(duration);
791 AST_APP_ARG(timeout);
792 AST_APP_ARG(times);
794 );
795
796 appdata = ast_strdupa(data);
797 AST_STANDARD_APP_ARGS(args, appdata);
798
799 if (!ast_strlen_zero(args.options)) {
800 ast_app_parse_options(wait_exec_options, &flags, opt_args, args.options);
801 }
802 if (freq_parser(args.freqs, &freq1, &freq2)) {
803 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
804 return -1;
805 }
806 if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeoutf < 0)) {
807 ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
808 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
809 return -1;
810 }
811 timeout = 1000 * timeoutf;
812 if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
813 ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
814 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
815 return -1;
816 }
817 if (!ast_strlen_zero(args.times) && (ast_str_to_int(args.times, &times) || times < 1)) {
818 ast_log(LOG_WARNING, "Invalid number of times: %s\n", args.times);
819 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
820 return -1;
821 }
823 if ((ast_str_to_int(opt_args[OPT_APP_ARG_DECIBEL], &db) || db < 1)) {
824 ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_APP_ARG_DECIBEL]);
825 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
826 return -1;
827 }
828 }
829 squelch = ast_test_flag(&flags, OPT_APP_SQUELCH);
830 if (!(dsp = ast_dsp_new())) {
831 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
832 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
833 return -1;
834 }
836 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
837 ast_debug(1, "Waiting for %s Hz, %d time(s), timeout %d ms, %d db\n", args.freqs, times, timeout, db);
838 start = ast_tvnow();
839 do {
840 if (timeout > 0) {
841 remaining_time = ast_remaining_ms(start, timeout);
842 if (remaining_time <= 0) {
843 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "TIMEOUT");
844 break;
845 }
846 }
847 if (ast_waitfor(chan, 1000) > 0) {
848 if (!(frame = ast_read(chan))) {
849 ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
850 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
851 break;
852 } else if (frame->frametype == AST_FRAME_VOICE) {
853 frame = ast_dsp_process(chan, dsp, frame);
854 if (frame->frametype == AST_FRAME_DTMF) {
855 char result = frame->subclass.integer;
856 if (result == 'q') {
857 hits++;
858 ast_debug(1, "We just detected %s Hz (hit #%d)\n", args.freqs, hits);
859 if (hits >= times) {
860 ast_frfree(frame);
861 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "SUCCESS");
862 break;
863 }
864 }
865 }
866 }
867 ast_frfree(frame);
868 } else {
869 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
870 }
871 } while (timeout == 0 || remaining_time > 0);
872 ast_dsp_free(dsp);
873
874 return 0;
875}
876
877static char *waitapp = "WaitForTone";
878static char *scanapp = "ToneScan";
879
880static int scan_exec(struct ast_channel *chan, const char *data)
881{
882 char *appdata;
883 double timeoutf = 0;
884 int timeout = 0;
885 struct ast_frame *frame = NULL, *frame2 = NULL;
886 struct ast_dsp *dsp = NULL, *dsp2 = NULL;
887 struct timeval start;
888 int remaining_time = 0;
889 int features, match = 0, fax = 0, voice = 0, threshold = 1;
891 AST_APP_ARG(zone);
892 AST_APP_ARG(timeout);
895 );
896
897 appdata = ast_strdupa(data);
898 AST_STANDARD_APP_ARGS(args, appdata);
899
900 if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeout < 0)) {
901 ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
902 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
903 return -1;
904 }
905 if (!ast_strlen_zero(args.threshold) && (ast_str_to_int(args.threshold, &threshold) || threshold < 1)) {
906 ast_log(LOG_WARNING, "Invalid threshold: %s\n", args.threshold);
907 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
908 return -1;
909 }
910 timeout = 1000 * timeoutf;
911
912 if (!ast_strlen_zero(args.options) && strchr(args.options, 'f')) {
913 fax = 1;
914 }
915 if (!ast_strlen_zero(args.options) && strchr(args.options, 'v')) {
916 voice = 1;
917 }
918
919 if (!(dsp = ast_dsp_new())) {
920 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
921 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
922 return -1;
923 }
924
925 if (!ast_strlen_zero(args.zone)) {
926 if (ast_dsp_set_call_progress_zone(dsp, args.zone)) {
927 ast_log(LOG_WARNING, "Invalid call progress zone: %s\n", args.zone);
928 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
929 ast_dsp_free(dsp);
930 return -1;
931 }
932 }
933
934 if (fax) {
935 if (!(dsp2 = ast_dsp_new())) {
936 ast_dsp_free(dsp);
937 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
938 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
939 return -1;
940 }
941 }
942
943 features = DSP_PROGRESS_RINGING; /* audible ringback tone */
944 features |= DSP_PROGRESS_BUSY; /* busy signal */
945 features |= DSP_PROGRESS_CONGESTION; /* SIT tones (not reorder!) */
946 features |= DSP_PROGRESS_TALK; /* voice. */
947 features |= DSP_FEATURE_WAITDIALTONE; /* dial tone */
948 features |= DSP_FEATURE_FREQ_DETECT; /* modem answer */
949 if (voice) {
950 features |= DSP_TONE_STATE_TALKING; /* voice */
951 }
952 ast_dsp_set_features(dsp, features);
953 /* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */
954 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 */
955
956 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. */
957 ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */
958 ast_dsp_set_faxmode(dsp2, DSP_FAXMODE_DETECT_CED); /* we only care about the answering side (CED), not originating (CNG) */
959 }
960
961 ast_debug(1, "Starting tone scan, timeout: %d ms, threshold: %d\n", timeout, threshold);
962 start = ast_tvnow();
963 do {
964 if (timeout > 0) {
965 remaining_time = ast_remaining_ms(start, timeout);
966 if (remaining_time <= 0) {
967 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "TIMEOUT");
968 break;
969 }
970 }
971 if (ast_waitfor(chan, 1000) > 0) {
972 if (!(frame = ast_read(chan))) {
973 ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
974 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
975 break;
976 } else if (frame->frametype == AST_FRAME_VOICE) {
977 if (fax) {
978 frame2 = ast_frdup(frame);
979 }
980 frame = ast_dsp_process(chan, dsp, frame);
981 if (frame->frametype == AST_FRAME_DTMF) {
982 char result = frame->subclass.integer;
983 match = 1;
984 if (result == 'q') {
985 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "MODEM");
986 } else {
987 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DTMF");
988 }
989 } else if (fax) {
990 char result;
991 frame2 = ast_dsp_process(chan, dsp2, frame2);
992 result = frame2->subclass.integer;
993 if (frame2->frametype == AST_FRAME_DTMF) {
994 if (result == 'e') {
995 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "FAX");
996 match = 1;
997 } else {
998 ast_debug(1, "Ignoring inactionable event\n"); /* shouldn't happen */
999 }
1000 }
1001 ast_frfree(frame2);
1002 }
1003 if (!match) {
1004 int tstate, tcount;
1005 tcount = ast_dsp_get_tcount(dsp);
1006 tstate = ast_dsp_get_tstate(dsp);
1007 if (tstate > 0) {
1008 ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
1009 if (tcount >= threshold) {
1010 match = 1;
1011 switch (tstate) {
1013 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "RINGING");
1014 break;
1016 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DIALTONE");
1017 break;
1019 /* even if we don't specify this feature, it's still checked, so we always need to handle it.
1020 Even if we are looking for it, we need to wait a while or tones will be interpreted
1021 as voice, because this will match first (and this should match last). */
1022 if (voice && tcount > 15 && tcount >= threshold) {
1023 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "VOICE");
1024 } else {
1025 match = 0;
1026 }
1027 break;
1029 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "BUSY");
1030 break;
1032 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "SIT");
1033 break;
1034 case DSP_TONE_STATE_HUNGUP: /* UK only */
1035 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "NUT");
1036 break;
1037 default:
1038 match = 0;
1039 ast_debug(1, "Something else we weren't expecting? tstate: %d, #%d\n", tstate, tcount);
1040 }
1041 }
1042 }
1043 }
1044 }
1045 ast_frfree(frame);
1046 } else {
1047 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
1048 }
1049 } while (!match && (timeout == 0 || remaining_time > 0));
1050 ast_dsp_free(dsp);
1051 if (dsp2) {
1052 ast_dsp_free(dsp2);
1053 }
1054
1055 return 0;
1056}
1057
1059 .name = "TONE_DETECT",
1060 .read = detect_read,
1061 .write = detect_write,
1062};
1063
1064static int unload_module(void)
1065{
1066 int res;
1067
1071
1072 return res;
1073}
1074
1075static int load_module(void)
1076{
1077 int res;
1078
1082
1083 return res;
1084}
1085
char * strsep(char **str, const char *delims)
Asterisk main include file. File version handling, generic pbx functions.
#define ast_free(a)
Definition: astmm.h:180
#define ast_strdupa(s)
duplicate a string in memory from the stack
Definition: astmm.h:298
#define ast_calloc(num, len)
A wrapper for calloc()
Definition: astmm.h:202
#define ast_malloc(len)
A wrapper for malloc()
Definition: astmm.h:191
#define ast_log
Definition: astobj2.c:42
#define ao2_ref(o, delta)
Reference/unreference an object and return the old refcount.
Definition: astobj2.h:459
Audiohooks Architecture.
@ AST_AUDIOHOOK_MANIPULATE_ALL_RATES
Definition: audiohook.h:75
int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags flags)
Initialize an audiohook structure.
Definition: audiohook.c:100
int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
Remove an audiohook from a specified channel.
Definition: audiohook.c:758
ast_audiohook_direction
Definition: audiohook.h:48
@ AST_AUDIOHOOK_DIRECTION_READ
Definition: audiohook.h:49
#define ast_audiohook_lock(ah)
Lock an audiohook.
Definition: audiohook.h:313
int ast_audiohook_detach(struct ast_audiohook *audiohook)
Detach audiohook from channel.
Definition: audiohook.c:587
int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
Attach audiohook to channel.
Definition: audiohook.c:521
int ast_audiohook_destroy(struct ast_audiohook *audiohook)
Destroys an audiohook structure.
Definition: audiohook.c:124
#define ast_audiohook_unlock(ah)
Unlock an audiohook.
Definition: audiohook.h:318
@ AST_AUDIOHOOK_TYPE_MANIPULATE
Definition: audiohook.h:38
@ AST_AUDIOHOOK_STATUS_DONE
Definition: audiohook.h:45
static sqlite3 * db
static PGresult * result
Definition: cel_pgsql.c:84
static int match(struct ast_sockaddr *addr, unsigned short callno, unsigned short dcallno, const struct chan_iax2_pvt *cur, int check_dcallno)
Definition: chan_iax2.c:2388
General Asterisk PBX channel definitions.
const char * ast_channel_name(const struct ast_channel *chan)
int ast_channel_datastore_add(struct ast_channel *chan, struct ast_datastore *datastore)
Add a datastore to a channel.
Definition: channel.c:2355
int ast_channel_datastore_remove(struct ast_channel *chan, struct ast_datastore *datastore)
Remove a datastore from a channel.
Definition: channel.c:2364
struct stasis_topic * ast_channel_topic(struct ast_channel *chan)
A topic which publishes the events for a particular channel.
#define ast_channel_lock(chan)
Definition: channel.h:2972
int ast_waitfor(struct ast_channel *chan, int ms)
Wait for input on a channel.
Definition: channel.c:3132
const char * ast_channel_uniqueid(const struct ast_channel *chan)
const char * ast_channel_context(const struct ast_channel *chan)
struct ast_frame * ast_read(struct ast_channel *chan)
Reads a frame.
Definition: channel.c:4223
const char * ast_channel_exten(const struct ast_channel *chan)
#define ast_channel_unlock(chan)
Definition: channel.h:2973
struct ast_datastore * ast_channel_datastore_find(struct ast_channel *chan, const struct ast_datastore_info *info, const char *uid)
Find a datastore on a channel.
Definition: channel.c:2369
Conversion utility functions.
int ast_str_to_int(const char *str, int *res)
Convert the given string to a signed integer.
Definition: conversions.c:44
#define ast_datastore_alloc(info, uid)
Definition: datastore.h:85
int ast_datastore_free(struct ast_datastore *datastore)
Free a data store object.
Definition: datastore.c:68
Convenient Signal Processing routines.
void ast_dsp_free(struct ast_dsp *dsp)
Definition: dsp.c:1785
int ast_dsp_get_tcount(struct ast_dsp *dsp)
Get tcount (Threshold counter)
Definition: dsp.c:1918
threshold
Definition: dsp.h:71
#define DSP_PROGRESS_RINGING
Definition: dsp.h:40
#define DSP_TONE_STATE_SPECIAL3
Definition: dsp.h:59
#define DSP_FEATURE_WAITDIALTONE
Definition: dsp.h:44
#define DSP_TONE_STATE_DIALTONE
Definition: dsp.h:54
#define DSP_PROGRESS_TALK
Definition: dsp.h:39
#define DSP_TONE_STATE_BUSY
Definition: dsp.h:56
struct ast_frame * ast_dsp_process(struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *inf)
Return AST_FRAME_NULL frames when there is silence, AST_FRAME_BUSY on busies, and call progress,...
Definition: dsp.c:1501
#define DSP_PROGRESS_BUSY
Definition: dsp.h:41
#define DSP_FEATURE_FAX_DETECT
Definition: dsp.h:29
#define DSP_FAXMODE_DETECT_CED
Definition: dsp.h:48
#define DSP_PROGRESS_CONGESTION
Definition: dsp.h:42
#define DSP_TONE_STATE_HUNGUP
Definition: dsp.h:60
#define DSP_FEATURE_FREQ_DETECT
Definition: dsp.h:45
#define DSP_TONE_STATE_TALKING
Definition: dsp.h:55
int ast_dsp_get_tstate(struct ast_dsp *dsp)
Get tstate (Tone State)
Definition: dsp.c:1913
#define DSP_TONE_STATE_RINGING
Definition: dsp.h:53
int ast_dsp_set_faxmode(struct ast_dsp *dsp, int faxmode)
Set fax mode.
Definition: dsp.c:1885
int ast_dsp_set_freqmode(struct ast_dsp *dsp, int freq, int dur, int db, int squelch)
Set arbitrary frequency detection mode.
Definition: dsp.c:1874
void ast_dsp_set_features(struct ast_dsp *dsp, int features)
Select feature set.
Definition: dsp.c:1770
struct ast_dsp * ast_dsp_new(void)
Allocates a new dsp, assumes 8khz for internal sample rate.
Definition: dsp.c:1760
int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone)
Set zone for doing progress detection.
Definition: dsp.c:1894
Media Format Cache API.
direction
struct stasis_message_type * ast_channel_tone_detect(void)
Message type for a channel tone detection.
struct stasis_message * ast_channel_blob_create_from_cache(const char *uniqueid, struct stasis_message_type *type, struct ast_json *blob)
Create a ast_channel_blob message, pulling channel state from the cache.
Application convenience functions, designed to give consistent look and feel to Asterisk apps.
#define AST_APP_ARG(name)
Define an application argument.
#define END_OPTIONS
#define AST_APP_OPTIONS(holder, options...)
Declares an array of options for an application.
#define AST_APP_OPTION_ARG(option, flagno, argno)
Declares an application option that accepts an argument.
#define AST_DECLARE_APP_ARGS(name, arglist)
Declare a structure to hold an application's arguments.
#define BEGIN_OPTIONS
#define AST_STANDARD_APP_ARGS(args, parse)
Performs the 'standard' argument separation process for an application.
#define AST_APP_OPTION(option, flagno)
Declares an application option that does not accept an argument.
int ast_app_parse_options(const struct ast_app_option *options, struct ast_flags *flags, char **args, char *optstr)
Parses a string containing application options and sets flags/arguments.
Definition: main/app.c:3066
Asterisk internal frame definitions.
#define AST_FRAME_DTMF
#define ast_frdup(fr)
Copies a frame.
#define ast_frfree(fr)
@ AST_FRAME_VOICE
#define AST_LOG_WARNING
#define ast_debug(level,...)
Log a DEBUG message.
#define LOG_ERROR
#define LOG_WARNING
Tone Indication Support.
#define SCOPED_CHANNELLOCK(varname, chan)
scoped lock specialization for channels.
Definition: lock.h:626
Asterisk module definitions.
#define ASTERISK_GPL_KEY
The text the key() function should return.
Definition: module.h:46
int ast_unregister_application(const char *app)
Unregister an application.
Definition: pbx_app.c:392
#define ast_register_application_xml(app, execute)
Register an application using XML documentation.
Definition: module.h:640
Core PBX routines and definitions.
int pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value)
Add a variable to the channel variable stack, removing the most recently set value for the same name.
#define ast_custom_function_register(acf)
Register a custom function.
Definition: pbx.h:1562
int ast_custom_function_unregister(struct ast_custom_function *acf)
Unregister a custom function.
int ast_async_parseable_goto(struct ast_channel *chan, const char *goto_string)
Definition: pbx.c:8896
static int detect_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
static int wait_exec(struct ast_channel *chan, const char *data)
static int detect_read(struct ast_channel *chan, const char *cmd, char *data, char *buffer, size_t buflen)
@ OPT_APP_ARG_DECIBEL
@ OPT_APP_ARG_ARRAY_SIZE
static int remove_detect(struct ast_channel *chan)
static int scan_exec(struct ast_channel *chan, const char *data)
static const struct ast_datastore_info detect_datastore
static struct ast_custom_function detect_function
@ OPT_APP_DECIBEL
@ OPT_APP_SQUELCH
AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Tone detection module")
static char * waitapp
static int parse_signal_features(struct ast_flags *flags)
static void destroy_callback(void *data)
static int load_module(void)
static char * goto_parser(struct ast_channel *chan, char *loc)
static int freq_parser(char *freqs, int *freq1, int *freq2)
static int unload_module(void)
td_opts
@ OPT_GOTO_RX
@ OPT_SIT
@ OPT_END_FILTER
@ OPT_DIALTONE
@ OPT_SQUELCH
@ OPT_AUTOEND
@ OPT_DECIBEL
@ OPT_RX
@ OPT_GOTO_TX
@ OPT_TX
@ OPT_BUSY
@ OPT_RINGING
@ OPT_HITS_REQ
static const struct ast_app_option wait_exec_options[128]
static char * scanapp
static int detect_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
@ OPT_ARG_GOTO_TX
@ OPT_ARG_DECIBEL
@ OPT_ARG_HITS_REQ
@ OPT_ARG_GOTO_RX
@ OPT_ARG_ARRAY_SIZE
#define NULL
Definition: resample.c:96
void stasis_publish(struct stasis_topic *topic, struct stasis_message *message)
Publish a message to a topic's subscribers.
Definition: stasis.c:1578
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition: strings.h:65
enum ast_audiohook_status status
Definition: audiohook.h:108
Main Channel structure associated with a channel.
Data structure associated with a custom dialplan function.
Definition: pbx.h:118
const char * name
Definition: pbx.h:119
Structure for a data store type.
Definition: datastore.h:31
const char * type
Definition: datastore.h:32
Structure for a data store object.
Definition: datastore.h:64
void * data
Definition: datastore.h:66
Definition: dsp.c:407
goertzel_state_t freqs[FREQ_ARRAY_SIZE]
Definition: dsp.c:421
Structure used to handle boolean flags.
Definition: utils.h:217
unsigned int flags
Definition: utils.h:218
Data structure associated with a single frame of data.
struct ast_frame_subclass subclass
enum ast_frame_type frametype
unsigned int autoend
struct ast_audiohook audiohook
struct ast_dsp * dsp
unsigned int matchmet
unsigned int squelch
int value
Definition: syslog.c:37
static float di[4]
Definition: tdd.c:58
const char * args
static struct test_options options
int ast_remaining_ms(struct timeval start, int max_ms)
Calculate remaining milliseconds given a starting timestamp and upper bound.
Definition: utils.c:2317
struct timeval ast_tvnow(void)
Returns current timeval. Meant to replace calls to gettimeofday().
Definition: time.h:159
#define ast_test_flag(p, flag)
Definition: utils.h:63