Asterisk - The Open Source Telephony Project GIT-master-a358458
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/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)
}
 

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 }, [ '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 307 of file res_tonedetect.c.

307 {
312 /* note: this entry _MUST_ be the last one in the enum */
314};
@ 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 734 of file res_tonedetect.c.

734 {
735 OPT_APP_DECIBEL = (1 << 0),
736 OPT_APP_SQUELCH = (1 << 1),
737};
@ OPT_APP_DECIBEL
@ OPT_APP_SQUELCH

◆ anonymous enum

anonymous enum
Enumerator
OPT_APP_ARG_DECIBEL 
OPT_APP_ARG_ARRAY_SIZE 

Definition at line 739 of file res_tonedetect.c.

739 {
741 /* note: this entry _MUST_ be the last one in the enum */
743};
@ 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 

Definition at line 292 of file res_tonedetect.c.

292 {
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};
@ 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

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 331 of file res_tonedetect.c.

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}
#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:550
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:1783
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 354 of file res_tonedetect.c.

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}
@ 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:2362
const char * ast_channel_name(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:2399
int ast_dsp_get_tcount(struct ast_dsp *dsp)
Get tcount (Threshold counter)
Definition: dsp.c:1916
#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:1499
#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:1911
#define DSP_TONE_STATE_RINGING
Definition: dsp.h:53
direction
#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.
int ast_async_parseable_goto(struct ast_channel *chan, const char *goto_string)
Definition: pbx.c:8871
static const struct ast_datastore_info detect_datastore
#define NULL
Definition: resample.c:96
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
struct ast_audiohook audiohook

References ast_async_parseable_goto(), AST_AUDIOHOOK_DIRECTION_READ, AST_AUDIOHOOK_STATUS_DONE, ast_channel_datastore_find(), ast_channel_name(), ast_debug, ast_dsp_get_tcount(), ast_dsp_get_tstate(), ast_dsp_process(), AST_FRAME_DTMF, AST_FRAME_VOICE, ast_frdup, ast_frfree, detect_information::audiohook, 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, match(), NULL, result, 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 563 of file res_tonedetect.c.

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}
#define ast_log
Definition: astobj2.c:42
#define ast_channel_lock(chan)
Definition: channel.h:2922
#define ast_channel_unlock(chan)
Definition: channel.h:2923
#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 613 of file res_tonedetect.c.

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}
#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:484
@ 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:2385
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: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
#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:3056
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:199
unsigned int flags
Definition: utils.h:200
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_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 489 of file res_tonedetect.c.

489 {
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}
char * strsep(char **str, const char *delims)
#define LOG_ERROR

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 526 of file res_tonedetect.c.

526 {
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}
#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 1049 of file res_tonedetect.c.

1050{
1051 int res;
1052
1056
1057 return res;
1058}
#define ast_register_application_xml(app, execute)
Register an application using XML documentation.
Definition: module.h:626
#define ast_custom_function_register(acf)
Register a custom function.
Definition: pbx.h:1558
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 593 of file res_tonedetect.c.

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}

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 460 of file res_tonedetect.c.

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}
int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
Remove an audiohook from a specified channel.
Definition: audiohook.c:721
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_LOG_WARNING
#define SCOPED_CHANNELLOCK(varname, chan)
scoped lock specialization for channels.
Definition: lock.h:619

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 854 of file res_tonedetect.c.

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}
int ast_waitfor(struct ast_channel *chan, int ms)
Wait for input on a channel.
Definition: channel.c:3162
struct ast_frame * ast_read(struct ast_channel *chan)
Reads a frame.
Definition: channel.c:4257
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:1883
int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone)
Set zone for doing progress detection.
Definition: dsp.c:1892
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:2281
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 1038 of file res_tonedetect.c.

1039{
1040 int res;
1041
1045
1046 return res;
1047}
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 750 of file res_tonedetect.c.

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}
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 349 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 1032 of file res_tonedetect.c.

Referenced by load_module(), and unload_module().

◆ scanapp

char* scanapp = "ToneScan"
static

Definition at line 852 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 }, [ '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 329 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 748 of file res_tonedetect.c.

Referenced by wait_exec().

◆ waitapp

char* waitapp = "WaitForTone"
static

Definition at line 851 of file res_tonedetect.c.

Referenced by load_module(), and unload_module().