Asterisk - The Open Source Telephony Project GIT-master-a358458
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"
42#include "asterisk/audiohook.h"
43#include "asterisk/app.h"
46
47/*** DOCUMENTATION
48 <application name="WaitForTone" language="en_US">
49 <since>
50 <version>16.21.0</version>
51 <version>18.7.0</version>
52 <version>19.0.0</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 <version>19.0.0</version>
186 </since>
187 <synopsis>
188 Asynchronously detects a tone
189 </synopsis>
190 <syntax>
191 <parameter name="freq" required="true">
192 <para>Frequency of the tone to detect. To disable frequency
193 detection completely (e.g. for signal detection only),
194 specify 0 for the frequency.</para>
195 </parameter>
196 <parameter name="duration_ms" required="false">
197 <para>Minimum duration of tone, in ms. Default is 500ms.
198 Using a minimum duration under 50ms is unlikely to produce
199 accurate results.</para>
200 </parameter>
201 <parameter name="options">
202 <optionlist>
203 <option name="a">
204 <para>Match immediately on Special Information Tones, instead of or in addition
205 to a particular frequency.</para>
206 </option>
207 <option name="b">
208 <para>Match immediately on a busy signal, instead of or in addition to
209 a particular frequency.</para>
210 </option>
211 <option name="c">
212 <para>Match immediately on a dial tone, instead of or in addition to
213 a particular frequency.</para>
214 </option>
215 <option name="d">
216 <para>Custom decibel threshold to use. Default is 16.</para>
217 </option>
218 <option name="g">
219 <para>Go to the specified context,exten,priority if tone is received on this channel.
220 Detection will not end automatically.</para>
221 </option>
222 <option name="h">
223 <para>Go to the specified context,exten,priority if tone is transmitted on this channel.
224 Detection will not end automatically.</para>
225 </option>
226 <option name="n">
227 <para>Number of times the tone should be detected (subject to the
228 provided timeout) before going to the destination provided in the <literal>g</literal>
229 or <literal>h</literal> option. Default is 1.</para>
230 </option>
231 <option name="p">
232 <para>Match immediately on audible ringback tone, instead of or in addition to
233 a particular frequency.</para>
234 </option>
235 <option name="r">
236 <para>Apply to received frames only. Default is both directions.</para>
237 </option>
238 <option name="s">
239 <para>Squelch tone.</para>
240 </option>
241 <option name="t">
242 <para>Apply to transmitted frames only. Default is both directions.</para>
243 </option>
244 <option name="x">
245 <para>Destroy the detector (stop detection).</para>
246 </option>
247 </optionlist>
248 </parameter>
249 </syntax>
250 <description>
251 <para>The TONE_DETECT function detects a single-frequency tone and keeps
252 track of how many times the tone has been detected.</para>
253 <para>When reading this function (instead of writing), supply <literal>tx</literal>
254 to get the number of times a tone has been detected in the TX direction and
255 <literal>rx</literal> to get the number of times a tone has been detected in the
256 RX direction.</para>
257 <example title="intercept2600">
258 same => n,Set(TONE_DETECT(2600,1000,g(got-2600,s,1))=) ; detect 2600 Hz
259 same => n,Wait(15)
260 same => n,NoOp(${TONE_DETECT(rx)})
261 </example>
262 <example title="dropondialtone">
263 same => n,Set(TONE_DETECT(0,,bg(my-hangup,s,1))=) ; disconnect a call if we hear a busy signal
264 same => n,Goto(somewhere-else)
265 same => n(myhangup),Hangup()
266 </example>
267 <example title="removedetector">
268 same => n,Set(TONE_DETECT(0,,x)=) ; remove the detector from the channel
269 </example>
270 </description>
271 </function>
272 ***/
273
275 struct ast_dsp *dsp;
277 int freq1;
278 int freq2;
280 int db;
281 char *gototx;
282 char *gotorx;
283 unsigned short int squelch;
284 unsigned short int tx;
285 unsigned short int rx;
290};
291
293 OPT_TX = (1 << 1),
294 OPT_RX = (1 << 2),
295 OPT_END_FILTER = (1 << 3),
296 OPT_GOTO_RX = (1 << 4),
297 OPT_GOTO_TX = (1 << 5),
298 OPT_DECIBEL = (1 << 6),
299 OPT_SQUELCH = (1 << 7),
300 OPT_HITS_REQ = (1 << 8),
301 OPT_SIT = (1 << 9),
302 OPT_BUSY = (1 << 10),
303 OPT_DIALTONE = (1 << 11),
304 OPT_RINGING = (1 << 12),
305};
306
307enum {
312 /* note: this entry _MUST_ be the last one in the enum */
314};
315
329});
330
331static void destroy_callback(void *data)
332{
333 struct detect_information *di = data;
334 ast_dsp_free(di->dsp);
335 if (di->gotorx) {
336 ast_free(di->gotorx);
337 }
338 if (di->gototx) {
339 ast_free(di->gototx);
340 }
341 ast_audiohook_lock(&di->audiohook);
342 ast_audiohook_detach(&di->audiohook);
343 ast_audiohook_unlock(&di->audiohook);
344 ast_audiohook_destroy(&di->audiohook);
345 ast_free(di);
346 return;
347}
348
350 .type = "detect",
351 .destroy = destroy_callback
352};
353
354static int detect_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
355{
356 struct ast_datastore *datastore = NULL;
357 struct detect_information *di = NULL;
358 int match = 0;
359
360 /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
362 return 0;
363 }
364
365 /* Grab datastore which contains our gain information */
366 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
367 return 0;
368 }
369
370 di = datastore->data;
371
372 if (!frame || frame->frametype != AST_FRAME_VOICE) {
373 return 0;
374 }
375
376 if (!(direction == AST_AUDIOHOOK_DIRECTION_READ ? di->rx : di->tx)) {
377 return 0;
378 }
379
380 /* ast_dsp_process may free the frame and return a new one */
381 frame = ast_frdup(frame);
382 frame = ast_dsp_process(chan, di->dsp, frame);
383 if (frame->frametype == AST_FRAME_DTMF) {
384 char result = frame->subclass.integer;
385 if (result == 'q') {
386 int now;
387 match = 1;
389 di->rxcount = di->rxcount + 1;
390 now = di->rxcount;
391 } else {
392 di->txcount = di->txcount + 1;
393 now = di->txcount;
394 }
395 ast_debug(1, "TONE_DETECT just got a hit (#%d in this direction, waiting for %d total)\n", now, di->hitsrequired);
396 if (now >= di->hitsrequired) {
397 if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
398 ast_async_parseable_goto(chan, di->gotorx);
399 } else if (di->gototx) {
400 ast_async_parseable_goto(chan, di->gototx);
401 }
402 }
403 }
404 }
405 if (di->signalfeatures && !match) { /* skip unless there are call progress/signal options */
406 int tstate, tcount;
407 tcount = ast_dsp_get_tcount(di->dsp);
408 tstate = ast_dsp_get_tstate(di->dsp);
409 if (tstate > 0) {
410 ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
411 switch (tstate) {
413 if (di->signalfeatures & DSP_PROGRESS_RINGING) {
414 ast_debug(1, "Detected ringing on %s in %s direction\n", ast_channel_name(chan),
415 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
416 match = 1;
417 }
418 break;
420 if (di->signalfeatures & DSP_FEATURE_WAITDIALTONE) {
421 ast_debug(1, "Detected dial tone on %s in %s direction\n", ast_channel_name(chan),
422 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
423 match = 1;
424 }
425 break;
427 if (di->signalfeatures & DSP_PROGRESS_BUSY) {
428 ast_debug(1, "Detected busy tone on %s in %s direction\n", ast_channel_name(chan),
429 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
430 match = 1;
431 }
432 break;
434 if (di->signalfeatures & DSP_PROGRESS_CONGESTION) {
435 ast_debug(1, "Detected SIT on %s in %s direction\n", ast_channel_name(chan),
436 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
437 match = 1;
438 }
439 break;
440 default: /* ignore */
441 break;
442 }
443 if (match) {
444 if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
445 ast_async_parseable_goto(chan, di->gotorx);
446 } else if (di->gototx) {
447 ast_async_parseable_goto(chan, di->gototx);
448 } else {
449 ast_debug(3, "Detected call progress signal in %s direction, but don't know where to go\n",
450 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
451 }
452 }
453 }
454 }
455 /* this could be the duplicated frame or a new one, doesn't matter */
456 ast_frfree(frame);
457 return 0;
458}
459
460static int remove_detect(struct ast_channel *chan)
461{
462 struct ast_datastore *datastore = NULL;
463 struct detect_information *data;
464 SCOPED_CHANNELLOCK(chan_lock, chan);
465
467 if (!datastore) {
468 ast_log(AST_LOG_WARNING, "Cannot remove TONE_DETECT from %s: TONE_DETECT not currently enabled\n",
469 ast_channel_name(chan));
470 return -1;
471 }
472 data = datastore->data;
473
474 if (ast_audiohook_remove(chan, &data->audiohook)) {
475 ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT audiohook from channel %s\n", ast_channel_name(chan));
476 return -1;
477 }
478
479 if (ast_channel_datastore_remove(chan, datastore)) {
480 ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT datastore from channel %s\n",
481 ast_channel_name(chan));
482 return -1;
483 }
484 ast_datastore_free(datastore);
485
486 return 0;
487}
488
489static int freq_parser(char *freqs, int *freq1, int *freq2) {
490 char *f1, *f2, *f3;
491 if (ast_strlen_zero(freqs)) {
492 ast_log(LOG_ERROR, "No frequency specified\n");
493 return -1;
494 }
495 f3 = ast_strdupa(freqs);
496 f1 = strsep(&f3, "+");
497 f2 = strsep(&f3, "+");
498 if (!ast_strlen_zero(f3)) {
499 ast_log(LOG_WARNING, "Only up to 2 frequencies may be specified: %s\n", freqs);
500 return -1;
501 }
502 if (ast_str_to_int(f1, freq1)) {
503 ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f1);
504 return -1;
505 }
506 if (*freq1 < 0) {
507 ast_log(LOG_WARNING, "Sorry, no negative frequencies: %d\n", *freq1);
508 return -1;
509 }
510 if (!ast_strlen_zero(f2)) {
511 ast_log(LOG_WARNING, "Sorry, currently only 1 frequency is supported\n");
512 return -1;
513 /* not supported just yet, but possibly will be in the future */
514 if (ast_str_to_int(f2, freq2)) {
515 ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f2);
516 return -1;
517 }
518 if (*freq2 < 1) {
519 ast_log(LOG_WARNING, "Sorry, positive frequencies only: %d\n", *freq2);
520 return -1;
521 }
522 }
523 return 0;
524}
525
526static char* goto_parser(struct ast_channel *chan, char *loc) {
527 char *exten, *pri, *context, *parse;
528 char *dest;
529 int size;
530 parse = ast_strdupa(loc);
531 context = strsep(&parse, ",");
532 exten = strsep(&parse, ",");
533 pri = strsep(&parse, ",");
534 if (!exten) {
535 pri = context;
536 exten = NULL;
537 context = NULL;
538 } else if (!pri) {
539 pri = exten;
540 exten = context;
541 context = NULL;
542 }
543 ast_channel_lock(chan);
544 if (ast_strlen_zero(exten)) {
545 exten = ast_strdupa(ast_channel_exten(chan));
546 }
549 }
550 ast_channel_unlock(chan);
551
552 /* size + 3: for 1 null terminator + 2 commas */
553 size = strlen(context) + strlen(exten) + strlen(pri) + 3;
554 dest = ast_malloc(size + 1);
555 if (!dest) {
556 ast_log(LOG_ERROR, "Failed to parse goto: %s,%s,%s\n", context, exten, pri);
557 return NULL;
558 }
559 snprintf(dest, size, "%s,%s,%s", context, exten, pri);
560 return dest;
561}
562
563static int detect_read(struct ast_channel *chan, const char *cmd, char *data, char *buffer, size_t buflen)
564{
565 struct ast_datastore *datastore = NULL;
566 struct detect_information *di = NULL;
567
568 if (!chan) {
569 ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
570 return -1;
571 }
572
573 ast_channel_lock(chan);
574 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
575 ast_channel_unlock(chan);
576 return -1; /* function not initiated yet, so nothing to read */
577 } else {
578 ast_channel_unlock(chan);
579 di = datastore->data;
580 }
581
582 if (strchr(data, 't')) {
583 snprintf(buffer, buflen, "%d", di->txcount);
584 } else if (strchr(data, 'r')) {
585 snprintf(buffer, buflen, "%d", di->rxcount);
586 } else {
587 ast_log(LOG_WARNING, "Invalid direction: %s\n", data);
588 }
589
590 return 0;
591}
592
593static int parse_signal_features(struct ast_flags *flags)
594{
595 int features = 0;
596
597 if (ast_test_flag(flags, OPT_SIT)) {
598 features |= DSP_PROGRESS_CONGESTION;
599 }
600 if (ast_test_flag(flags, OPT_BUSY)) {
601 features |= DSP_PROGRESS_BUSY;
602 }
603 if (ast_test_flag(flags, OPT_DIALTONE)) {
604 features |= DSP_FEATURE_WAITDIALTONE;
605 }
606 if (ast_test_flag(flags, OPT_RINGING)) {
607 features |= DSP_PROGRESS_RINGING;
608 }
609
610 return features;
611}
612
613static int detect_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
614{
615 char *parse;
616 struct ast_datastore *datastore = NULL;
617 struct detect_information *di = NULL;
618 struct ast_flags flags = { 0 };
619 char *opt_args[OPT_ARG_ARRAY_SIZE];
620 struct ast_dsp *dsp;
621 int freq1 = 0, freq2 = 0, duration = 500, db = 16, squelch = 0, hitsrequired = 1;
622 int signalfeatures = 0;
623
626 AST_APP_ARG(duration);
628 );
629
630 if (!chan) {
631 ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
632 return -1;
633 }
634 parse = ast_strdupa(data);
636
637 if (!ast_strlen_zero(args.options)) {
638 ast_app_parse_options(td_opts, &flags, opt_args, args.options);
639 }
640 if (ast_test_flag(&flags, OPT_END_FILTER)) {
641 return remove_detect(chan);
642 }
643 if (freq_parser(args.freqs, &freq1, &freq2)) {
644 return -1;
645 }
646 if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
647 ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
648 return -1;
649 }
650 if (ast_test_flag(&flags, OPT_HITS_REQ) && !ast_strlen_zero(opt_args[OPT_ARG_HITS_REQ])) {
651 if ((ast_str_to_int(opt_args[OPT_ARG_HITS_REQ], &hitsrequired) || hitsrequired < 1)) {
652 ast_log(LOG_WARNING, "Invalid number hits required: %s\n", opt_args[OPT_ARG_HITS_REQ]);
653 return -1;
654 }
655 }
656 if (ast_test_flag(&flags, OPT_DECIBEL) && !ast_strlen_zero(opt_args[OPT_ARG_DECIBEL])) {
657 if ((ast_str_to_int(opt_args[OPT_ARG_DECIBEL], &db) || db < 1)) {
658 ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_ARG_DECIBEL]);
659 return -1;
660 }
661 }
662 signalfeatures = parse_signal_features(&flags);
663
664 ast_channel_lock(chan);
665 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
666 if (!(datastore = ast_datastore_alloc(&detect_datastore, NULL))) {
667 ast_channel_unlock(chan);
668 return 0;
669 }
670 if (!(di = ast_calloc(1, sizeof(*di)))) {
671 ast_datastore_free(datastore);
672 ast_channel_unlock(chan);
673 return 0;
674 }
676 di->audiohook.manipulate_callback = detect_callback;
677 if (!(dsp = ast_dsp_new())) {
678 ast_datastore_free(datastore);
679 ast_channel_unlock(chan);
680 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
681 return -1;
682 }
683 di->signalfeatures = signalfeatures; /* we're not including freq detect */
684 if (freq1 > 0) {
685 signalfeatures |= DSP_FEATURE_FREQ_DETECT;
686 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
687 }
688 ast_dsp_set_features(dsp, signalfeatures);
689 di->dsp = dsp;
690 di->txcount = 0;
691 di->rxcount = 0;
692 ast_debug(1, "Keeping our ears open for %s Hz, %d db\n", args.freqs, db);
693 datastore->data = di;
694 ast_channel_datastore_add(chan, datastore);
695 ast_audiohook_attach(chan, &di->audiohook);
696 } else {
697 di = datastore->data;
698 dsp = di->dsp;
699 di->signalfeatures = signalfeatures; /* we're not including freq detect */
700 if (freq1 > 0) {
701 signalfeatures |= DSP_FEATURE_FREQ_DETECT;
702 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
703 }
704 ast_dsp_set_features(dsp, signalfeatures);
705 }
706 di->duration = duration;
707 di->gotorx = NULL;
708 di->gototx = NULL;
709 /* resolve gotos now, in case a full context,exten,pri wasn't specified */
710 if (ast_test_flag(&flags, OPT_GOTO_RX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_RX])) {
711 di->gotorx = goto_parser(chan, opt_args[OPT_ARG_GOTO_RX]);
712 }
713 if (ast_test_flag(&flags, OPT_GOTO_TX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_TX])) {
714 di->gototx = goto_parser(chan, opt_args[OPT_ARG_GOTO_TX]);
715 }
716 di->db = db;
717 di->hitsrequired = hitsrequired;
718 di->squelch = ast_test_flag(&flags, OPT_SQUELCH);
719 di->tx = 1;
720 di->rx = 1;
721 if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_TX)) {
722 di->tx = 1;
723 di->rx = 0;
724 }
725 if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_RX)) {
726 di->rx = 1;
727 di->tx = 0;
728 }
729 ast_channel_unlock(chan);
730
731 return 0;
732}
733
734enum {
735 OPT_APP_DECIBEL = (1 << 0),
736 OPT_APP_SQUELCH = (1 << 1),
737};
738
739enum {
741 /* note: this entry _MUST_ be the last one in the enum */
743};
744
749
750static int wait_exec(struct ast_channel *chan, const char *data)
751{
752 char *appdata;
753 struct ast_flags flags = {0};
754 char *opt_args[OPT_APP_ARG_ARRAY_SIZE];
755 double timeoutf = 0;
756 int freq1 = 0, freq2 = 0, timeout = 0, duration = 500, times = 1, db = 16, squelch = 0;
757 struct ast_frame *frame = NULL;
758 struct ast_dsp *dsp;
759 struct timeval start;
760 int remaining_time = 0;
761 int hits = 0;
763 AST_APP_ARG(freqs);
764 AST_APP_ARG(duration);
765 AST_APP_ARG(timeout);
766 AST_APP_ARG(times);
768 );
769
770 appdata = ast_strdupa(data);
771 AST_STANDARD_APP_ARGS(args, appdata);
772
773 if (!ast_strlen_zero(args.options)) {
774 ast_app_parse_options(wait_exec_options, &flags, opt_args, args.options);
775 }
776 if (freq_parser(args.freqs, &freq1, &freq2)) {
777 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
778 return -1;
779 }
780 if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeoutf < 0)) {
781 ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
782 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
783 return -1;
784 }
785 timeout = 1000 * timeoutf;
786 if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
787 ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
788 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
789 return -1;
790 }
791 if (!ast_strlen_zero(args.times) && (ast_str_to_int(args.times, &times) || times < 1)) {
792 ast_log(LOG_WARNING, "Invalid number of times: %s\n", args.times);
793 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
794 return -1;
795 }
797 if ((ast_str_to_int(opt_args[OPT_APP_ARG_DECIBEL], &db) || db < 1)) {
798 ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_APP_ARG_DECIBEL]);
799 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
800 return -1;
801 }
802 }
803 squelch = ast_test_flag(&flags, OPT_APP_SQUELCH);
804 if (!(dsp = ast_dsp_new())) {
805 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
806 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
807 return -1;
808 }
810 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
811 ast_debug(1, "Waiting for %s Hz, %d time(s), timeout %d ms, %d db\n", args.freqs, times, timeout, db);
812 start = ast_tvnow();
813 do {
814 if (timeout > 0) {
815 remaining_time = ast_remaining_ms(start, timeout);
816 if (remaining_time <= 0) {
817 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "TIMEOUT");
818 break;
819 }
820 }
821 if (ast_waitfor(chan, 1000) > 0) {
822 if (!(frame = ast_read(chan))) {
823 ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
824 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
825 break;
826 } else if (frame->frametype == AST_FRAME_VOICE) {
827 frame = ast_dsp_process(chan, dsp, frame);
828 if (frame->frametype == AST_FRAME_DTMF) {
829 char result = frame->subclass.integer;
830 if (result == 'q') {
831 hits++;
832 ast_debug(1, "We just detected %s Hz (hit #%d)\n", args.freqs, hits);
833 if (hits >= times) {
834 ast_frfree(frame);
835 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "SUCCESS");
836 break;
837 }
838 }
839 }
840 }
841 ast_frfree(frame);
842 } else {
843 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
844 }
845 } while (timeout == 0 || remaining_time > 0);
846 ast_dsp_free(dsp);
847
848 return 0;
849}
850
851static char *waitapp = "WaitForTone";
852static char *scanapp = "ToneScan";
853
854static int scan_exec(struct ast_channel *chan, const char *data)
855{
856 char *appdata;
857 double timeoutf = 0;
858 int timeout = 0;
859 struct ast_frame *frame = NULL, *frame2 = NULL;
860 struct ast_dsp *dsp = NULL, *dsp2 = NULL;
861 struct timeval start;
862 int remaining_time = 0;
863 int features, match = 0, fax = 0, voice = 0, threshold = 1;
865 AST_APP_ARG(zone);
866 AST_APP_ARG(timeout);
869 );
870
871 appdata = ast_strdupa(data);
872 AST_STANDARD_APP_ARGS(args, appdata);
873
874 if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeout < 0)) {
875 ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
876 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
877 return -1;
878 }
879 if (!ast_strlen_zero(args.threshold) && (ast_str_to_int(args.threshold, &threshold) || threshold < 1)) {
880 ast_log(LOG_WARNING, "Invalid threshold: %s\n", args.threshold);
881 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
882 return -1;
883 }
884 timeout = 1000 * timeoutf;
885
886 if (!ast_strlen_zero(args.options) && strchr(args.options, 'f')) {
887 fax = 1;
888 }
889 if (!ast_strlen_zero(args.options) && strchr(args.options, 'v')) {
890 voice = 1;
891 }
892
893 if (!(dsp = ast_dsp_new())) {
894 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
895 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
896 return -1;
897 }
898
899 if (!ast_strlen_zero(args.zone)) {
900 if (ast_dsp_set_call_progress_zone(dsp, args.zone)) {
901 ast_log(LOG_WARNING, "Invalid call progress zone: %s\n", args.zone);
902 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
903 ast_dsp_free(dsp);
904 return -1;
905 }
906 }
907
908 if (fax) {
909 if (!(dsp2 = ast_dsp_new())) {
910 ast_dsp_free(dsp);
911 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
912 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
913 return -1;
914 }
915 }
916
917 features = DSP_PROGRESS_RINGING; /* audible ringback tone */
918 features |= DSP_PROGRESS_BUSY; /* busy signal */
919 features |= DSP_PROGRESS_CONGESTION; /* SIT tones (not reorder!) */
920 features |= DSP_PROGRESS_TALK; /* voice. */
921 features |= DSP_FEATURE_WAITDIALTONE; /* dial tone */
922 features |= DSP_FEATURE_FREQ_DETECT; /* modem answer */
923 if (voice) {
924 features |= DSP_TONE_STATE_TALKING; /* voice */
925 }
926 ast_dsp_set_features(dsp, features);
927 /* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */
928 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 */
929
930 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. */
931 ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */
932 ast_dsp_set_faxmode(dsp2, DSP_FAXMODE_DETECT_CED); /* we only care about the answering side (CED), not originating (CNG) */
933 }
934
935 ast_debug(1, "Starting tone scan, timeout: %d ms, threshold: %d\n", timeout, threshold);
936 start = ast_tvnow();
937 do {
938 if (timeout > 0) {
939 remaining_time = ast_remaining_ms(start, timeout);
940 if (remaining_time <= 0) {
941 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "TIMEOUT");
942 break;
943 }
944 }
945 if (ast_waitfor(chan, 1000) > 0) {
946 if (!(frame = ast_read(chan))) {
947 ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
948 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
949 break;
950 } else if (frame->frametype == AST_FRAME_VOICE) {
951 if (fax) {
952 frame2 = ast_frdup(frame);
953 }
954 frame = ast_dsp_process(chan, dsp, frame);
955 if (frame->frametype == AST_FRAME_DTMF) {
956 char result = frame->subclass.integer;
957 match = 1;
958 if (result == 'q') {
959 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "MODEM");
960 } else {
961 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DTMF");
962 }
963 } else if (fax) {
964 char result;
965 frame2 = ast_dsp_process(chan, dsp2, frame2);
966 result = frame2->subclass.integer;
967 if (frame2->frametype == AST_FRAME_DTMF) {
968 if (result == 'e') {
969 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "FAX");
970 match = 1;
971 } else {
972 ast_debug(1, "Ignoring inactionable event\n"); /* shouldn't happen */
973 }
974 }
975 ast_frfree(frame2);
976 }
977 if (!match) {
978 int tstate, tcount;
979 tcount = ast_dsp_get_tcount(dsp);
980 tstate = ast_dsp_get_tstate(dsp);
981 if (tstate > 0) {
982 ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
983 if (tcount >= threshold) {
984 match = 1;
985 switch (tstate) {
987 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "RINGING");
988 break;
990 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DIALTONE");
991 break;
993 /* even if we don't specify this feature, it's still checked, so we always need to handle it.
994 Even if we are looking for it, we need to wait a while or tones will be interpreted
995 as voice, because this will match first (and this should match last). */
996 if (voice && tcount > 15 && tcount >= threshold) {
997 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "VOICE");
998 } else {
999 match = 0;
1000 }
1001 break;
1003 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "BUSY");
1004 break;
1006 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "SIT");
1007 break;
1008 case DSP_TONE_STATE_HUNGUP: /* UK only */
1009 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "NUT");
1010 break;
1011 default:
1012 match = 0;
1013 ast_debug(1, "Something else we weren't expecting? tstate: %d, #%d\n", tstate, tcount);
1014 }
1015 }
1016 }
1017 }
1018 }
1019 ast_frfree(frame);
1020 } else {
1021 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
1022 }
1023 } while (!match && (timeout == 0 || remaining_time > 0));
1024 ast_dsp_free(dsp);
1025 if (dsp2) {
1026 ast_dsp_free(dsp2);
1027 }
1028
1029 return 0;
1030}
1031
1033 .name = "TONE_DETECT",
1034 .read = detect_read,
1035 .write = detect_write,
1036};
1037
1038static int unload_module(void)
1039{
1040 int res;
1041
1045
1046 return res;
1047}
1048
1049static int load_module(void)
1050{
1051 int res;
1052
1056
1057 return res;
1058}
1059
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
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:721
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:550
int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
Attach audiohook to channel.
Definition: audiohook.c:484
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:2362
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:2385
int ast_channel_datastore_remove(struct ast_channel *chan, struct ast_datastore *datastore)
Remove a datastore from a channel.
Definition: channel.c:2394
#define ast_channel_lock(chan)
Definition: channel.h:2922
int ast_waitfor(struct ast_channel *chan, int ms)
Wait for input on a channel.
Definition: channel.c:3162
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:4257
const char * ast_channel_exten(const struct ast_channel *chan)
#define ast_channel_unlock(chan)
Definition: channel.h:2923
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:2399
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:1783
int ast_dsp_get_tcount(struct ast_dsp *dsp)
Get tcount (Threshold counter)
Definition: dsp.c:1916
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:1499
#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:1911
#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:1883
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:1872
void ast_dsp_set_features(struct ast_dsp *dsp, int features)
Select feature set.
Definition: dsp.c:1768
struct ast_dsp * ast_dsp_new(void)
Allocates a new dsp, assumes 8khz for internal sample rate.
Definition: dsp.c:1758
int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone)
Set zone for doing progress detection.
Definition: dsp.c:1892
Media Format Cache API.
direction
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:3056
char * strsep(char **str, const char *delims)
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:619
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:626
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:1558
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:8871
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_ARG_GOTO_TX
@ OPT_ARG_DECIBEL
@ OPT_ARG_HITS_REQ
@ OPT_ARG_GOTO_RX
@ OPT_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
@ OPT_APP_DECIBEL
@ OPT_APP_SQUELCH
static struct ast_custom_function detect_function
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)
@ OPT_APP_ARG_DECIBEL
@ OPT_APP_ARG_ARRAY_SIZE
td_opts
@ OPT_GOTO_RX
@ OPT_SIT
@ OPT_END_FILTER
@ OPT_DIALTONE
@ OPT_SQUELCH
@ 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)
#define NULL
Definition: resample.c:96
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:199
unsigned int flags
Definition: utils.h:200
Data structure associated with a single frame of data.
struct ast_frame_subclass subclass
enum ast_frame_type frametype
unsigned short int tx
struct ast_audiohook audiohook
unsigned short int squelch
struct ast_dsp * dsp
unsigned short int rx
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:2281
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