Asterisk - The Open Source Telephony Project GIT-master-0a46be9
Data Structures | Enumerations | Functions | Variables
res_tonedetect.c File Reference

Tone detection module. More...

#include "asterisk.h"
#include <math.h>
#include "asterisk/module.h"
#include "asterisk/frame.h"
#include "asterisk/format_cache.h"
#include "asterisk/channel.h"
#include "asterisk/dsp.h"
#include "asterisk/pbx.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/audiohook.h"
#include "asterisk/app.h"
#include "asterisk/indications.h"
#include "asterisk/conversions.h"
Include dependency graph for res_tonedetect.c:

Go to the source code of this file.

Data Structures

struct  detect_information
 

Enumerations

enum  {
  OPT_ARG_DECIBEL , OPT_ARG_GOTO_RX , OPT_ARG_GOTO_TX , OPT_ARG_HITS_REQ ,
  OPT_ARG_ARRAY_SIZE
}
 
enum  { OPT_APP_DECIBEL = (1 << 0) , OPT_APP_SQUELCH = (1 << 1) }
 
enum  { OPT_APP_ARG_DECIBEL , OPT_APP_ARG_ARRAY_SIZE }
 
enum  td_opts {
  OPT_TX = (1 << 1) , OPT_RX = (1 << 2) , OPT_END_FILTER = (1 << 3) , OPT_GOTO_RX = (1 << 4) ,
  OPT_GOTO_TX = (1 << 5) , OPT_DECIBEL = (1 << 6) , OPT_SQUELCH = (1 << 7) , OPT_HITS_REQ = (1 << 8) ,
  OPT_SIT = (1 << 9) , OPT_BUSY = (1 << 10) , OPT_DIALTONE = (1 << 11) , OPT_RINGING = (1 << 12) ,
  OPT_AUTOEND = (1 << 13)
}
 

Functions

 AST_MODULE_INFO_STANDARD_EXTENDED (ASTERISK_GPL_KEY, "Tone detection module")
 
static void destroy_callback (void *data)
 
static int detect_callback (struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
 
static int detect_read (struct ast_channel *chan, const char *cmd, char *data, char *buffer, size_t buflen)
 
static int detect_write (struct ast_channel *chan, const char *cmd, char *data, const char *value)
 
static int freq_parser (char *freqs, int *freq1, int *freq2)
 
static char * goto_parser (struct ast_channel *chan, char *loc)
 
static int load_module (void)
 
static int parse_signal_features (struct ast_flags *flags)
 
static int remove_detect (struct ast_channel *chan)
 
static int scan_exec (struct ast_channel *chan, const char *data)
 
static int unload_module (void)
 
static int wait_exec (struct ast_channel *chan, const char *data)
 

Variables

static const struct ast_datastore_info detect_datastore
 
static struct ast_custom_function detect_function
 
static char * scanapp = "ToneScan"
 
static const struct ast_app_option td_opts [128] = { [ 'a' ] = { .flag = OPT_SIT }, [ 'b' ] = { .flag = OPT_BUSY }, [ 'c' ] = { .flag = OPT_DIALTONE }, [ 'd' ] = { .flag = OPT_DECIBEL , .arg_index = OPT_ARG_DECIBEL + 1 }, [ 'e' ] = { .flag = OPT_AUTOEND }, [ 'g' ] = { .flag = OPT_GOTO_RX , .arg_index = OPT_ARG_GOTO_RX + 1 }, [ 'h' ] = { .flag = OPT_GOTO_TX , .arg_index = OPT_ARG_GOTO_TX + 1 }, [ 'n' ] = { .flag = OPT_HITS_REQ , .arg_index = OPT_ARG_HITS_REQ + 1 }, [ 'p' ] = { .flag = OPT_RINGING }, [ 's' ] = { .flag = OPT_SQUELCH }, [ 't' ] = { .flag = OPT_TX }, [ 'r' ] = { .flag = OPT_RX }, [ 'x' ] = { .flag = OPT_END_FILTER }, }
 
static const struct ast_app_option wait_exec_options [128] = { [ 'd' ] = { .flag = OPT_APP_DECIBEL , .arg_index = OPT_APP_ARG_DECIBEL + 1 }, [ 's' ] = { .flag = OPT_APP_SQUELCH }, }
 
static char * waitapp = "WaitForTone"
 

Detailed Description

Tone detection module.

Author
Naveen Albert aster.nosp@m.isk@.nosp@m.phrea.nosp@m.knet.nosp@m..org

Definition in file res_tonedetect.c.

Enumeration Type Documentation

◆ anonymous enum

anonymous enum
Enumerator
OPT_ARG_DECIBEL 
OPT_ARG_GOTO_RX 
OPT_ARG_GOTO_TX 
OPT_ARG_HITS_REQ 
OPT_ARG_ARRAY_SIZE 

Definition at line 312 of file res_tonedetect.c.

312 {
317 /* note: this entry _MUST_ be the last one in the enum */
319};
@ OPT_ARG_GOTO_TX
@ OPT_ARG_DECIBEL
@ OPT_ARG_HITS_REQ
@ OPT_ARG_GOTO_RX
@ OPT_ARG_ARRAY_SIZE

◆ anonymous enum

anonymous enum
Enumerator
OPT_APP_DECIBEL 
OPT_APP_SQUELCH 

Definition at line 760 of file res_tonedetect.c.

760 {
761 OPT_APP_DECIBEL = (1 << 0),
762 OPT_APP_SQUELCH = (1 << 1),
763};
@ OPT_APP_DECIBEL
@ OPT_APP_SQUELCH

◆ anonymous enum

anonymous enum
Enumerator
OPT_APP_ARG_DECIBEL 
OPT_APP_ARG_ARRAY_SIZE 

Definition at line 765 of file res_tonedetect.c.

765 {
767 /* note: this entry _MUST_ be the last one in the enum */
769};
@ OPT_APP_ARG_DECIBEL
@ OPT_APP_ARG_ARRAY_SIZE

◆ td_opts

enum td_opts
Enumerator
OPT_TX 
OPT_RX 
OPT_END_FILTER 
OPT_GOTO_RX 
OPT_GOTO_TX 
OPT_DECIBEL 
OPT_SQUELCH 
OPT_HITS_REQ 
OPT_SIT 
OPT_BUSY 
OPT_DIALTONE 
OPT_RINGING 
OPT_AUTOEND 

Definition at line 296 of file res_tonedetect.c.

296 {
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};
@ 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

Function Documentation

◆ AST_MODULE_INFO_STANDARD_EXTENDED()

AST_MODULE_INFO_STANDARD_EXTENDED ( ASTERISK_GPL_KEY  ,
"Tone detection module"   
)

◆ destroy_callback()

static void destroy_callback ( void *  data)
static

Definition at line 337 of file res_tonedetect.c.

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}
#define ast_free(a)
Definition: astmm.h:180
#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_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
void ast_dsp_free(struct ast_dsp *dsp)
Definition: dsp.c:1785
static float di[4]
Definition: tdd.c:58

References ast_audiohook_destroy(), ast_audiohook_detach(), ast_audiohook_lock, ast_audiohook_unlock, ast_dsp_free(), ast_free, and di.

◆ detect_callback()

static int detect_callback ( struct ast_audiohook audiohook,
struct ast_channel chan,
struct ast_frame frame,
enum ast_audiohook_direction  direction 
)
static

Definition at line 360 of file res_tonedetect.c.

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}
#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
@ AST_AUDIOHOOK_DIRECTION_READ
Definition: audiohook.h:49
@ AST_AUDIOHOOK_STATUS_DONE
Definition: audiohook.h:45
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
const char * ast_channel_name(const struct ast_channel *chan)
struct stasis_topic * ast_channel_topic(struct ast_channel *chan)
A topic which publishes the events for a particular channel.
const char * ast_channel_uniqueid(const struct ast_channel *chan)
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
int ast_dsp_get_tcount(struct ast_dsp *dsp)
Get tcount (Threshold counter)
Definition: dsp.c:1918
#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_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_PROGRESS_CONGESTION
Definition: dsp.h:42
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
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.
#define AST_FRAME_DTMF
#define ast_frdup(fr)
Copies a frame.
#define ast_frfree(fr)
@ AST_FRAME_VOICE
#define ast_debug(level,...)
Log a DEBUG message.
#define LOG_ERROR
int ast_async_parseable_goto(struct ast_channel *chan, const char *goto_string)
Definition: pbx.c:8896
static const struct ast_datastore_info detect_datastore
#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:1538
enum ast_audiohook_status status
Definition: audiohook.h:108
Structure for a data store object.
Definition: datastore.h:64
void * data
Definition: datastore.h:66
struct ast_frame_subclass subclass
enum ast_frame_type frametype

References ao2_ref, ast_async_parseable_goto(), AST_AUDIOHOOK_DIRECTION_READ, AST_AUDIOHOOK_STATUS_DONE, ast_channel_blob_create_from_cache(), ast_channel_datastore_find(), ast_channel_name(), ast_channel_tone_detect(), ast_channel_topic(), ast_channel_uniqueid(), ast_debug, ast_dsp_get_tcount(), ast_dsp_get_tstate(), ast_dsp_process(), AST_FRAME_DTMF, AST_FRAME_VOICE, ast_frdup, ast_frfree, ast_log, ast_datastore::data, detect_datastore, di, DSP_FEATURE_WAITDIALTONE, DSP_PROGRESS_BUSY, DSP_PROGRESS_CONGESTION, DSP_PROGRESS_RINGING, DSP_TONE_STATE_BUSY, DSP_TONE_STATE_DIALTONE, DSP_TONE_STATE_RINGING, DSP_TONE_STATE_SPECIAL3, ast_frame::frametype, ast_frame_subclass::integer, LOG_ERROR, match(), NULL, result, stasis_publish(), ast_audiohook::status, and ast_frame::subclass.

Referenced by detect_write().

◆ detect_read()

static int detect_read ( struct ast_channel chan,
const char *  cmd,
char *  data,
char *  buffer,
size_t  buflen 
)
static

Definition at line 588 of file res_tonedetect.c.

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}
#define ast_channel_lock(chan)
Definition: channel.h:2972
#define ast_channel_unlock(chan)
Definition: channel.h:2973
#define LOG_WARNING

References ast_channel_datastore_find(), ast_channel_lock, ast_channel_unlock, ast_log, ast_datastore::data, detect_datastore, di, LOG_WARNING, and NULL.

◆ detect_write()

static int detect_write ( struct ast_channel chan,
const char *  cmd,
char *  data,
const char *  value 
)
static

Definition at line 638 of file res_tonedetect.c.

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}
#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
@ 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_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
Attach audiohook to channel.
Definition: audiohook.c:521
@ AST_AUDIOHOOK_TYPE_MANIPULATE
Definition: audiohook.h:38
static sqlite3 * db
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_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
#define DSP_FEATURE_FREQ_DETECT
Definition: dsp.h:45
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
#define AST_APP_ARG(name)
Define an application argument.
#define AST_DECLARE_APP_ARGS(name, arglist)
Declare a structure to hold an application's arguments.
#define AST_STANDARD_APP_ARGS(args, parse)
Performs the 'standard' argument separation process for an application.
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
static int detect_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
static int remove_detect(struct ast_channel *chan)
static int parse_signal_features(struct ast_flags *flags)
static char * goto_parser(struct ast_channel *chan, char *loc)
static int freq_parser(char *freqs, int *freq1, int *freq2)
td_opts
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition: strings.h:65
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
const char * args
static struct test_options options
#define ast_test_flag(p, flag)
Definition: utils.h:63

References args, AST_APP_ARG, ast_app_parse_options(), ast_audiohook_attach(), ast_audiohook_init(), AST_AUDIOHOOK_MANIPULATE_ALL_RATES, AST_AUDIOHOOK_TYPE_MANIPULATE, ast_calloc, ast_channel_datastore_add(), ast_channel_datastore_find(), ast_channel_lock, ast_channel_unlock, ast_datastore_alloc, ast_datastore_free(), ast_debug, AST_DECLARE_APP_ARGS, ast_dsp_new(), ast_dsp_set_features(), ast_dsp_set_freqmode(), ast_log, AST_STANDARD_APP_ARGS, ast_str_to_int(), ast_strdupa, ast_strlen_zero(), ast_test_flag, ast_datastore::data, db, detect_callback(), detect_datastore, di, DSP_FEATURE_FREQ_DETECT, ast_flags::flags, freq_parser(), ast_dsp::freqs, goto_parser(), LOG_WARNING, NULL, OPT_ARG_ARRAY_SIZE, OPT_ARG_DECIBEL, OPT_ARG_GOTO_RX, OPT_ARG_GOTO_TX, OPT_ARG_HITS_REQ, OPT_AUTOEND, OPT_DECIBEL, OPT_END_FILTER, OPT_GOTO_RX, OPT_GOTO_TX, OPT_HITS_REQ, OPT_RX, OPT_SQUELCH, OPT_TX, options, parse_signal_features(), and remove_detect().

◆ freq_parser()

static int freq_parser ( char *  freqs,
int *  freq1,
int *  freq2 
)
static

Definition at line 514 of file res_tonedetect.c.

514 {
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}
char * strsep(char **str, const char *delims)

References ast_log, ast_str_to_int(), ast_strdupa, ast_strlen_zero(), detect_information::freq1, detect_information::freq2, LOG_ERROR, LOG_WARNING, and strsep().

Referenced by detect_write(), and wait_exec().

◆ goto_parser()

static char * goto_parser ( struct ast_channel chan,
char *  loc 
)
static

Definition at line 551 of file res_tonedetect.c.

551 {
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}
#define ast_malloc(len)
A wrapper for malloc()
Definition: astmm.h:191
const char * ast_channel_context(const struct ast_channel *chan)
const char * ast_channel_exten(const struct ast_channel *chan)

References ast_channel_context(), ast_channel_exten(), ast_channel_lock, ast_channel_unlock, ast_log, ast_malloc, ast_strdupa, ast_strlen_zero(), voicemailpwcheck::context, LOG_ERROR, NULL, and strsep().

Referenced by detect_write().

◆ load_module()

static int load_module ( void  )
static

Definition at line 1075 of file res_tonedetect.c.

1076{
1077 int res;
1078
1082
1083 return res;
1084}
#define ast_register_application_xml(app, execute)
Register an application using XML documentation.
Definition: module.h:640
#define ast_custom_function_register(acf)
Register a custom function.
Definition: pbx.h:1562
static int wait_exec(struct ast_channel *chan, const char *data)
static int scan_exec(struct ast_channel *chan, const char *data)
static struct ast_custom_function detect_function
static char * waitapp
static char * scanapp

References ast_custom_function_register, ast_register_application_xml, detect_function, scan_exec(), scanapp, wait_exec(), and waitapp.

◆ parse_signal_features()

static int parse_signal_features ( struct ast_flags flags)
static

Definition at line 618 of file res_tonedetect.c.

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}

References ast_test_flag, DSP_FEATURE_WAITDIALTONE, DSP_PROGRESS_BUSY, DSP_PROGRESS_CONGESTION, DSP_PROGRESS_RINGING, OPT_BUSY, OPT_DIALTONE, OPT_RINGING, and OPT_SIT.

Referenced by detect_write().

◆ remove_detect()

static int remove_detect ( struct ast_channel chan)
static

Definition at line 485 of file res_tonedetect.c.

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}
int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
Remove an audiohook from a specified channel.
Definition: audiohook.c:758
int ast_channel_datastore_remove(struct ast_channel *chan, struct ast_datastore *datastore)
Remove a datastore from a channel.
Definition: channel.c:2364
#define AST_LOG_WARNING
#define SCOPED_CHANNELLOCK(varname, chan)
scoped lock specialization for channels.
Definition: lock.h:626
struct ast_audiohook audiohook

References ast_audiohook_remove(), ast_channel_datastore_find(), ast_channel_datastore_remove(), ast_channel_name(), ast_datastore_free(), ast_log, AST_LOG_WARNING, detect_information::audiohook, ast_datastore::data, detect_datastore, NULL, and SCOPED_CHANNELLOCK.

Referenced by detect_write().

◆ scan_exec()

static int scan_exec ( struct ast_channel chan,
const char *  data 
)
static

Definition at line 880 of file res_tonedetect.c.

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}
int ast_waitfor(struct ast_channel *chan, int ms)
Wait for input on a channel.
Definition: channel.c:3132
struct ast_frame * ast_read(struct ast_channel *chan)
Reads a frame.
Definition: channel.c:4223
threshold
Definition: dsp.h:71
#define DSP_PROGRESS_TALK
Definition: dsp.h:39
#define DSP_FEATURE_FAX_DETECT
Definition: dsp.h:29
#define DSP_FAXMODE_DETECT_CED
Definition: dsp.h:48
#define DSP_TONE_STATE_HUNGUP
Definition: dsp.h:60
#define DSP_TONE_STATE_TALKING
Definition: dsp.h:55
int ast_dsp_set_faxmode(struct ast_dsp *dsp, int faxmode)
Set fax mode.
Definition: dsp.c:1885
int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone)
Set zone for doing progress detection.
Definition: dsp.c:1894
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.
Data structure associated with a single frame of data.
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

References args, AST_APP_ARG, ast_channel_name(), ast_debug, AST_DECLARE_APP_ARGS, ast_dsp_free(), ast_dsp_get_tcount(), ast_dsp_get_tstate(), ast_dsp_new(), ast_dsp_process(), ast_dsp_set_call_progress_zone(), ast_dsp_set_faxmode(), ast_dsp_set_features(), ast_dsp_set_freqmode(), AST_FRAME_DTMF, AST_FRAME_VOICE, ast_frdup, ast_frfree, ast_log, ast_read(), ast_remaining_ms(), AST_STANDARD_APP_ARGS, ast_str_to_int(), ast_strdupa, ast_strlen_zero(), ast_tvnow(), ast_waitfor(), DSP_FAXMODE_DETECT_CED, DSP_FEATURE_FAX_DETECT, DSP_FEATURE_FREQ_DETECT, DSP_FEATURE_WAITDIALTONE, DSP_PROGRESS_BUSY, DSP_PROGRESS_CONGESTION, DSP_PROGRESS_RINGING, DSP_PROGRESS_TALK, DSP_TONE_STATE_BUSY, DSP_TONE_STATE_DIALTONE, DSP_TONE_STATE_HUNGUP, DSP_TONE_STATE_RINGING, DSP_TONE_STATE_SPECIAL3, DSP_TONE_STATE_TALKING, ast_frame::frametype, ast_frame_subclass::integer, LOG_WARNING, match(), NULL, options, pbx_builtin_setvar_helper(), result, and ast_frame::subclass.

Referenced by load_module().

◆ unload_module()

static int unload_module ( void  )
static

Definition at line 1064 of file res_tonedetect.c.

1065{
1066 int res;
1067
1071
1072 return res;
1073}
int ast_unregister_application(const char *app)
Unregister an application.
Definition: pbx_app.c:392
int ast_custom_function_unregister(struct ast_custom_function *acf)
Unregister a custom function.

References ast_custom_function_unregister(), ast_unregister_application(), detect_function, scanapp, and waitapp.

◆ wait_exec()

static int wait_exec ( struct ast_channel chan,
const char *  data 
)
static

Definition at line 776 of file res_tonedetect.c.

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}
static const struct ast_app_option wait_exec_options[128]

References args, AST_APP_ARG, ast_app_parse_options(), ast_channel_name(), ast_debug, AST_DECLARE_APP_ARGS, ast_dsp_free(), ast_dsp_new(), ast_dsp_process(), ast_dsp_set_features(), ast_dsp_set_freqmode(), AST_FRAME_DTMF, AST_FRAME_VOICE, ast_frfree, ast_log, ast_read(), ast_remaining_ms(), AST_STANDARD_APP_ARGS, ast_str_to_int(), ast_strdupa, ast_strlen_zero(), ast_test_flag, ast_tvnow(), ast_waitfor(), db, DSP_FEATURE_FREQ_DETECT, ast_flags::flags, ast_frame::frametype, freq_parser(), ast_frame_subclass::integer, LOG_WARNING, NULL, OPT_APP_ARG_ARRAY_SIZE, OPT_APP_ARG_DECIBEL, OPT_APP_DECIBEL, OPT_APP_SQUELCH, options, pbx_builtin_setvar_helper(), result, ast_frame::subclass, and wait_exec_options.

Referenced by load_module().

Variable Documentation

◆ detect_datastore

const struct ast_datastore_info detect_datastore
static
Initial value:
= {
.type = "detect",
.destroy = destroy_callback
}
static void destroy_callback(void *data)

Definition at line 355 of file res_tonedetect.c.

Referenced by detect_callback(), detect_read(), detect_write(), and remove_detect().

◆ detect_function

struct ast_custom_function detect_function
static
Initial value:
= {
.name = "TONE_DETECT",
.read = detect_read,
.write = detect_write,
}
static int detect_read(struct ast_channel *chan, const char *cmd, char *data, char *buffer, size_t buflen)
static int detect_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)

Definition at line 1058 of file res_tonedetect.c.

Referenced by load_module(), and unload_module().

◆ scanapp

char* scanapp = "ToneScan"
static

Definition at line 878 of file res_tonedetect.c.

Referenced by load_module(), and unload_module().

◆ td_opts

const struct ast_app_option td_opts[128] = { [ 'a' ] = { .flag = OPT_SIT }, [ 'b' ] = { .flag = OPT_BUSY }, [ 'c' ] = { .flag = OPT_DIALTONE }, [ 'd' ] = { .flag = OPT_DECIBEL , .arg_index = OPT_ARG_DECIBEL + 1 }, [ 'e' ] = { .flag = OPT_AUTOEND }, [ 'g' ] = { .flag = OPT_GOTO_RX , .arg_index = OPT_ARG_GOTO_RX + 1 }, [ 'h' ] = { .flag = OPT_GOTO_TX , .arg_index = OPT_ARG_GOTO_TX + 1 }, [ 'n' ] = { .flag = OPT_HITS_REQ , .arg_index = OPT_ARG_HITS_REQ + 1 }, [ 'p' ] = { .flag = OPT_RINGING }, [ 's' ] = { .flag = OPT_SQUELCH }, [ 't' ] = { .flag = OPT_TX }, [ 'r' ] = { .flag = OPT_RX }, [ 'x' ] = { .flag = OPT_END_FILTER }, }
static

Definition at line 335 of file res_tonedetect.c.

◆ wait_exec_options

const struct ast_app_option wait_exec_options[128] = { [ 'd' ] = { .flag = OPT_APP_DECIBEL , .arg_index = OPT_APP_ARG_DECIBEL + 1 }, [ 's' ] = { .flag = OPT_APP_SQUELCH }, }
static

Definition at line 774 of file res_tonedetect.c.

Referenced by wait_exec().

◆ waitapp

char* waitapp = "WaitForTone"
static

Definition at line 877 of file res_tonedetect.c.

Referenced by load_module(), and unload_module().