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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)
}
 

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 306 of file res_tonedetect.c.

306 {
311 /* note: this entry _MUST_ be the last one in the enum */
313};
@ 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 744 of file res_tonedetect.c.

744 {
745 OPT_APP_DECIBEL = (1 << 0),
746 OPT_APP_SQUELCH = (1 << 1),
747};
@ OPT_APP_DECIBEL
@ OPT_APP_SQUELCH

◆ anonymous enum

anonymous enum
Enumerator
OPT_APP_ARG_DECIBEL 
OPT_APP_ARG_ARRAY_SIZE 

Definition at line 749 of file res_tonedetect.c.

749 {
751 /* note: this entry _MUST_ be the last one in the enum */
753};
@ 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 291 of file res_tonedetect.c.

291 {
292 OPT_TX = (1 << 1),
293 OPT_RX = (1 << 2),
294 OPT_END_FILTER = (1 << 3),
295 OPT_GOTO_RX = (1 << 4),
296 OPT_GOTO_TX = (1 << 5),
297 OPT_DECIBEL = (1 << 6),
298 OPT_SQUELCH = (1 << 7),
299 OPT_HITS_REQ = (1 << 8),
300 OPT_SIT = (1 << 9),
301 OPT_BUSY = (1 << 10),
302 OPT_DIALTONE = (1 << 11),
303 OPT_RINGING = (1 << 12),
304};
@ 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 330 of file res_tonedetect.c.

331{
332 struct detect_information *di = data;
333 ast_dsp_free(di->dsp);
334 if (di->gotorx) {
335 ast_free(di->gotorx);
336 }
337 if (di->gototx) {
338 ast_free(di->gototx);
339 }
340 ast_audiohook_lock(&di->audiohook);
341 ast_audiohook_detach(&di->audiohook);
342 ast_audiohook_unlock(&di->audiohook);
343 ast_audiohook_destroy(&di->audiohook);
344 ast_free(di);
345 return;
346}
#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:578
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 353 of file res_tonedetect.c.

354{
355 struct ast_datastore *datastore = NULL;
356 struct detect_information *di = NULL;
357 struct stasis_message *message;
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 */
361 if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
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) {
398
399 if (!message) {
400 ast_log(LOG_ERROR, "Unable to publish tone detected event for ARI on channel '%s'", ast_channel_name(chan));
401 return 1;
402 } else {
404 ao2_ref(message, -1);
405 }
406
407 if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
408 ast_async_parseable_goto(chan, di->gotorx);
409 } else if (di->gototx) {
410 ast_async_parseable_goto(chan, di->gototx);
411 }
412 }
413 }
414 }
415 if (di->signalfeatures && !match) { /* skip unless there are call progress/signal options */
416 int tstate, tcount;
417 tcount = ast_dsp_get_tcount(di->dsp);
418 tstate = ast_dsp_get_tstate(di->dsp);
419 if (tstate > 0) {
420 ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
421 switch (tstate) {
423 if (di->signalfeatures & DSP_PROGRESS_RINGING) {
424 ast_debug(1, "Detected ringing on %s in %s direction\n", ast_channel_name(chan),
425 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
426 match = 1;
427 }
428 break;
430 if (di->signalfeatures & DSP_FEATURE_WAITDIALTONE) {
431 ast_debug(1, "Detected dial tone on %s in %s direction\n", ast_channel_name(chan),
432 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
433 match = 1;
434 }
435 break;
437 if (di->signalfeatures & DSP_PROGRESS_BUSY) {
438 ast_debug(1, "Detected busy tone on %s in %s direction\n", ast_channel_name(chan),
439 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
440 match = 1;
441 }
442 break;
444 if (di->signalfeatures & DSP_PROGRESS_CONGESTION) {
445 ast_debug(1, "Detected SIT on %s in %s direction\n", ast_channel_name(chan),
446 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
447 match = 1;
448 }
449 break;
450 default: /* ignore */
451 break;
452 }
453 if (match) {
454 if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
455 ast_async_parseable_goto(chan, di->gotorx);
456 } else if (di->gototx) {
457 ast_async_parseable_goto(chan, di->gototx);
458 } else {
459 ast_debug(3, "Detected call progress signal in %s direction, but don't know where to go\n",
460 direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write");
461 }
462 }
463 }
464 }
465 /* this could be the duplicated frame or a new one, doesn't matter */
466 ast_frfree(frame);
467 return 0;
468}
#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:2368
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
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:8886
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 573 of file res_tonedetect.c.

574{
575 struct ast_datastore *datastore = NULL;
576 struct detect_information *di = NULL;
577
578 if (!chan) {
579 ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
580 return -1;
581 }
582
583 ast_channel_lock(chan);
584 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
585 ast_channel_unlock(chan);
586 return -1; /* function not initiated yet, so nothing to read */
587 } else {
588 ast_channel_unlock(chan);
589 di = datastore->data;
590 }
591
592 if (strchr(data, 't')) {
593 snprintf(buffer, buflen, "%d", di->txcount);
594 } else if (strchr(data, 'r')) {
595 snprintf(buffer, buflen, "%d", di->rxcount);
596 } else {
597 ast_log(LOG_WARNING, "Invalid direction: %s\n", data);
598 }
599
600 return 0;
601}
#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 623 of file res_tonedetect.c.

624{
625 char *parse;
626 struct ast_datastore *datastore = NULL;
627 struct detect_information *di = NULL;
628 struct ast_flags flags = { 0 };
629 char *opt_args[OPT_ARG_ARRAY_SIZE];
630 struct ast_dsp *dsp;
631 int freq1 = 0, freq2 = 0, duration = 500, db = 16, squelch = 0, hitsrequired = 1;
632 int signalfeatures = 0;
633
636 AST_APP_ARG(duration);
638 );
639
640 if (!chan) {
641 ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
642 return -1;
643 }
644 parse = ast_strdupa(data);
646
647 if (!ast_strlen_zero(args.options)) {
648 ast_app_parse_options(td_opts, &flags, opt_args, args.options);
649 }
650 if (ast_test_flag(&flags, OPT_END_FILTER)) {
651 return remove_detect(chan);
652 }
653 if (freq_parser(args.freqs, &freq1, &freq2)) {
654 return -1;
655 }
656 if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
657 ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
658 return -1;
659 }
660 if (ast_test_flag(&flags, OPT_HITS_REQ) && !ast_strlen_zero(opt_args[OPT_ARG_HITS_REQ])) {
661 if ((ast_str_to_int(opt_args[OPT_ARG_HITS_REQ], &hitsrequired) || hitsrequired < 1)) {
662 ast_log(LOG_WARNING, "Invalid number hits required: %s\n", opt_args[OPT_ARG_HITS_REQ]);
663 return -1;
664 }
665 }
666 if (ast_test_flag(&flags, OPT_DECIBEL) && !ast_strlen_zero(opt_args[OPT_ARG_DECIBEL])) {
667 if ((ast_str_to_int(opt_args[OPT_ARG_DECIBEL], &db) || db < 1)) {
668 ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_ARG_DECIBEL]);
669 return -1;
670 }
671 }
672 signalfeatures = parse_signal_features(&flags);
673
674 ast_channel_lock(chan);
675 if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
676 if (!(datastore = ast_datastore_alloc(&detect_datastore, NULL))) {
677 ast_channel_unlock(chan);
678 return 0;
679 }
680 if (!(di = ast_calloc(1, sizeof(*di)))) {
681 ast_datastore_free(datastore);
682 ast_channel_unlock(chan);
683 return 0;
684 }
686 di->audiohook.manipulate_callback = detect_callback;
687 if (!(dsp = ast_dsp_new())) {
688 ast_datastore_free(datastore);
689 ast_channel_unlock(chan);
690 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
691 return -1;
692 }
693 di->signalfeatures = signalfeatures; /* we're not including freq detect */
694 if (freq1 > 0) {
695 signalfeatures |= DSP_FEATURE_FREQ_DETECT;
696 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
697 }
698 ast_dsp_set_features(dsp, signalfeatures);
699 di->dsp = dsp;
700 di->txcount = 0;
701 di->rxcount = 0;
702 ast_debug(1, "Keeping our ears open for %s Hz, %d db\n", args.freqs, db);
703 datastore->data = di;
704 ast_channel_datastore_add(chan, datastore);
705 ast_audiohook_attach(chan, &di->audiohook);
706 } else {
707 di = datastore->data;
708 dsp = di->dsp;
709 di->signalfeatures = signalfeatures; /* we're not including freq detect */
710 if (freq1 > 0) {
711 signalfeatures |= DSP_FEATURE_FREQ_DETECT;
712 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
713 }
714 ast_dsp_set_features(dsp, signalfeatures);
715 }
716 di->duration = duration;
717 di->gotorx = NULL;
718 di->gototx = NULL;
719 /* resolve gotos now, in case a full context,exten,pri wasn't specified */
720 if (ast_test_flag(&flags, OPT_GOTO_RX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_RX])) {
721 di->gotorx = goto_parser(chan, opt_args[OPT_ARG_GOTO_RX]);
722 }
723 if (ast_test_flag(&flags, OPT_GOTO_TX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_TX])) {
724 di->gototx = goto_parser(chan, opt_args[OPT_ARG_GOTO_TX]);
725 }
726 di->db = db;
727 di->hitsrequired = hitsrequired;
728 di->squelch = ast_test_flag(&flags, OPT_SQUELCH);
729 di->tx = 1;
730 di->rx = 1;
731 if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_TX)) {
732 di->tx = 1;
733 di->rx = 0;
734 }
735 if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_RX)) {
736 di->rx = 1;
737 di->tx = 0;
738 }
739 ast_channel_unlock(chan);
740
741 return 0;
742}
#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:512
@ 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:2354
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: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: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 499 of file res_tonedetect.c.

499 {
500 char *f1, *f2, *f3;
501 if (ast_strlen_zero(freqs)) {
502 ast_log(LOG_ERROR, "No frequency specified\n");
503 return -1;
504 }
505 f3 = ast_strdupa(freqs);
506 f1 = strsep(&f3, "+");
507 f2 = strsep(&f3, "+");
508 if (!ast_strlen_zero(f3)) {
509 ast_log(LOG_WARNING, "Only up to 2 frequencies may be specified: %s\n", freqs);
510 return -1;
511 }
512 if (ast_str_to_int(f1, freq1)) {
513 ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f1);
514 return -1;
515 }
516 if (*freq1 < 0) {
517 ast_log(LOG_WARNING, "Sorry, no negative frequencies: %d\n", *freq1);
518 return -1;
519 }
520 if (!ast_strlen_zero(f2)) {
521 ast_log(LOG_WARNING, "Sorry, currently only 1 frequency is supported\n");
522 return -1;
523 /* not supported just yet, but possibly will be in the future */
524 if (ast_str_to_int(f2, freq2)) {
525 ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f2);
526 return -1;
527 }
528 if (*freq2 < 1) {
529 ast_log(LOG_WARNING, "Sorry, positive frequencies only: %d\n", *freq2);
530 return -1;
531 }
532 }
533 return 0;
534}
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 536 of file res_tonedetect.c.

536 {
537 char *exten, *pri, *context, *parse;
538 char *dest;
539 int size;
540 parse = ast_strdupa(loc);
541 context = strsep(&parse, ",");
542 exten = strsep(&parse, ",");
543 pri = strsep(&parse, ",");
544 if (!exten) {
545 pri = context;
546 exten = NULL;
547 context = NULL;
548 } else if (!pri) {
549 pri = exten;
550 exten = context;
551 context = NULL;
552 }
553 ast_channel_lock(chan);
554 if (ast_strlen_zero(exten)) {
555 exten = ast_strdupa(ast_channel_exten(chan));
556 }
559 }
560 ast_channel_unlock(chan);
561
562 /* size + 3: for 1 null terminator + 2 commas */
563 size = strlen(context) + strlen(exten) + strlen(pri) + 3;
564 dest = ast_malloc(size + 1);
565 if (!dest) {
566 ast_log(LOG_ERROR, "Failed to parse goto: %s,%s,%s\n", context, exten, pri);
567 return NULL;
568 }
569 snprintf(dest, size, "%s,%s,%s", context, exten, pri);
570 return dest;
571}
#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 1059 of file res_tonedetect.c.

1060{
1061 int res;
1062
1066
1067 return res;
1068}
#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:1559
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 603 of file res_tonedetect.c.

604{
605 int features = 0;
606
607 if (ast_test_flag(flags, OPT_SIT)) {
608 features |= DSP_PROGRESS_CONGESTION;
609 }
610 if (ast_test_flag(flags, OPT_BUSY)) {
611 features |= DSP_PROGRESS_BUSY;
612 }
613 if (ast_test_flag(flags, OPT_DIALTONE)) {
614 features |= DSP_FEATURE_WAITDIALTONE;
615 }
616 if (ast_test_flag(flags, OPT_RINGING)) {
617 features |= DSP_PROGRESS_RINGING;
618 }
619
620 return features;
621}

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 470 of file res_tonedetect.c.

471{
472 struct ast_datastore *datastore = NULL;
473 struct detect_information *data;
474 SCOPED_CHANNELLOCK(chan_lock, chan);
475
477 if (!datastore) {
478 ast_log(AST_LOG_WARNING, "Cannot remove TONE_DETECT from %s: TONE_DETECT not currently enabled\n",
479 ast_channel_name(chan));
480 return -1;
481 }
482 data = datastore->data;
483
484 if (ast_audiohook_remove(chan, &data->audiohook)) {
485 ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT audiohook from channel %s\n", ast_channel_name(chan));
486 return -1;
487 }
488
489 if (ast_channel_datastore_remove(chan, datastore)) {
490 ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT datastore from channel %s\n",
491 ast_channel_name(chan));
492 return -1;
493 }
494 ast_datastore_free(datastore);
495
496 return 0;
497}
int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
Remove an audiohook from a specified channel.
Definition: audiohook.c:749
int ast_channel_datastore_remove(struct ast_channel *chan, struct ast_datastore *datastore)
Remove a datastore from a channel.
Definition: channel.c:2363
#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 864 of file res_tonedetect.c.

865{
866 char *appdata;
867 double timeoutf = 0;
868 int timeout = 0;
869 struct ast_frame *frame = NULL, *frame2 = NULL;
870 struct ast_dsp *dsp = NULL, *dsp2 = NULL;
871 struct timeval start;
872 int remaining_time = 0;
873 int features, match = 0, fax = 0, voice = 0, threshold = 1;
875 AST_APP_ARG(zone);
876 AST_APP_ARG(timeout);
879 );
880
881 appdata = ast_strdupa(data);
882 AST_STANDARD_APP_ARGS(args, appdata);
883
884 if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeout < 0)) {
885 ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
886 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
887 return -1;
888 }
889 if (!ast_strlen_zero(args.threshold) && (ast_str_to_int(args.threshold, &threshold) || threshold < 1)) {
890 ast_log(LOG_WARNING, "Invalid threshold: %s\n", args.threshold);
891 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
892 return -1;
893 }
894 timeout = 1000 * timeoutf;
895
896 if (!ast_strlen_zero(args.options) && strchr(args.options, 'f')) {
897 fax = 1;
898 }
899 if (!ast_strlen_zero(args.options) && strchr(args.options, 'v')) {
900 voice = 1;
901 }
902
903 if (!(dsp = ast_dsp_new())) {
904 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
905 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
906 return -1;
907 }
908
909 if (!ast_strlen_zero(args.zone)) {
910 if (ast_dsp_set_call_progress_zone(dsp, args.zone)) {
911 ast_log(LOG_WARNING, "Invalid call progress zone: %s\n", args.zone);
912 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
913 ast_dsp_free(dsp);
914 return -1;
915 }
916 }
917
918 if (fax) {
919 if (!(dsp2 = ast_dsp_new())) {
920 ast_dsp_free(dsp);
921 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
922 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
923 return -1;
924 }
925 }
926
927 features = DSP_PROGRESS_RINGING; /* audible ringback tone */
928 features |= DSP_PROGRESS_BUSY; /* busy signal */
929 features |= DSP_PROGRESS_CONGESTION; /* SIT tones (not reorder!) */
930 features |= DSP_PROGRESS_TALK; /* voice. */
931 features |= DSP_FEATURE_WAITDIALTONE; /* dial tone */
932 features |= DSP_FEATURE_FREQ_DETECT; /* modem answer */
933 if (voice) {
934 features |= DSP_TONE_STATE_TALKING; /* voice */
935 }
936 ast_dsp_set_features(dsp, features);
937 /* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */
938 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 */
939
940 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. */
941 ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */
942 ast_dsp_set_faxmode(dsp2, DSP_FAXMODE_DETECT_CED); /* we only care about the answering side (CED), not originating (CNG) */
943 }
944
945 ast_debug(1, "Starting tone scan, timeout: %d ms, threshold: %d\n", timeout, threshold);
946 start = ast_tvnow();
947 do {
948 if (timeout > 0) {
949 remaining_time = ast_remaining_ms(start, timeout);
950 if (remaining_time <= 0) {
951 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "TIMEOUT");
952 break;
953 }
954 }
955 if (ast_waitfor(chan, 1000) > 0) {
956 if (!(frame = ast_read(chan))) {
957 ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
958 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
959 break;
960 } else if (frame->frametype == AST_FRAME_VOICE) {
961 if (fax) {
962 frame2 = ast_frdup(frame);
963 }
964 frame = ast_dsp_process(chan, dsp, frame);
965 if (frame->frametype == AST_FRAME_DTMF) {
966 char result = frame->subclass.integer;
967 match = 1;
968 if (result == 'q') {
969 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "MODEM");
970 } else {
971 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DTMF");
972 }
973 } else if (fax) {
974 char result;
975 frame2 = ast_dsp_process(chan, dsp2, frame2);
976 result = frame2->subclass.integer;
977 if (frame2->frametype == AST_FRAME_DTMF) {
978 if (result == 'e') {
979 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "FAX");
980 match = 1;
981 } else {
982 ast_debug(1, "Ignoring inactionable event\n"); /* shouldn't happen */
983 }
984 }
985 ast_frfree(frame2);
986 }
987 if (!match) {
988 int tstate, tcount;
989 tcount = ast_dsp_get_tcount(dsp);
990 tstate = ast_dsp_get_tstate(dsp);
991 if (tstate > 0) {
992 ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
993 if (tcount >= threshold) {
994 match = 1;
995 switch (tstate) {
997 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "RINGING");
998 break;
1000 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DIALTONE");
1001 break;
1003 /* even if we don't specify this feature, it's still checked, so we always need to handle it.
1004 Even if we are looking for it, we need to wait a while or tones will be interpreted
1005 as voice, because this will match first (and this should match last). */
1006 if (voice && tcount > 15 && tcount >= threshold) {
1007 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "VOICE");
1008 } else {
1009 match = 0;
1010 }
1011 break;
1013 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "BUSY");
1014 break;
1016 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "SIT");
1017 break;
1018 case DSP_TONE_STATE_HUNGUP: /* UK only */
1019 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "NUT");
1020 break;
1021 default:
1022 match = 0;
1023 ast_debug(1, "Something else we weren't expecting? tstate: %d, #%d\n", tstate, tcount);
1024 }
1025 }
1026 }
1027 }
1028 }
1029 ast_frfree(frame);
1030 } else {
1031 pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
1032 }
1033 } while (!match && (timeout == 0 || remaining_time > 0));
1034 ast_dsp_free(dsp);
1035 if (dsp2) {
1036 ast_dsp_free(dsp2);
1037 }
1038
1039 return 0;
1040}
int ast_waitfor(struct ast_channel *chan, int ms)
Wait for input on a channel.
Definition: channel.c:3130
struct ast_frame * ast_read(struct ast_channel *chan)
Reads a frame.
Definition: channel.c:4214
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 1048 of file res_tonedetect.c.

1049{
1050 int res;
1051
1055
1056 return res;
1057}
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 760 of file res_tonedetect.c.

761{
762 char *appdata;
763 struct ast_flags flags = {0};
764 char *opt_args[OPT_APP_ARG_ARRAY_SIZE];
765 double timeoutf = 0;
766 int freq1 = 0, freq2 = 0, timeout = 0, duration = 500, times = 1, db = 16, squelch = 0;
767 struct ast_frame *frame = NULL;
768 struct ast_dsp *dsp;
769 struct timeval start;
770 int remaining_time = 0;
771 int hits = 0;
773 AST_APP_ARG(freqs);
774 AST_APP_ARG(duration);
775 AST_APP_ARG(timeout);
776 AST_APP_ARG(times);
778 );
779
780 appdata = ast_strdupa(data);
781 AST_STANDARD_APP_ARGS(args, appdata);
782
783 if (!ast_strlen_zero(args.options)) {
784 ast_app_parse_options(wait_exec_options, &flags, opt_args, args.options);
785 }
786 if (freq_parser(args.freqs, &freq1, &freq2)) {
787 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
788 return -1;
789 }
790 if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeoutf < 0)) {
791 ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
792 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
793 return -1;
794 }
795 timeout = 1000 * timeoutf;
796 if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
797 ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
798 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
799 return -1;
800 }
801 if (!ast_strlen_zero(args.times) && (ast_str_to_int(args.times, &times) || times < 1)) {
802 ast_log(LOG_WARNING, "Invalid number of times: %s\n", args.times);
803 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
804 return -1;
805 }
807 if ((ast_str_to_int(opt_args[OPT_APP_ARG_DECIBEL], &db) || db < 1)) {
808 ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_APP_ARG_DECIBEL]);
809 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
810 return -1;
811 }
812 }
813 squelch = ast_test_flag(&flags, OPT_APP_SQUELCH);
814 if (!(dsp = ast_dsp_new())) {
815 ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
816 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
817 return -1;
818 }
820 ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
821 ast_debug(1, "Waiting for %s Hz, %d time(s), timeout %d ms, %d db\n", args.freqs, times, timeout, db);
822 start = ast_tvnow();
823 do {
824 if (timeout > 0) {
825 remaining_time = ast_remaining_ms(start, timeout);
826 if (remaining_time <= 0) {
827 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "TIMEOUT");
828 break;
829 }
830 }
831 if (ast_waitfor(chan, 1000) > 0) {
832 if (!(frame = ast_read(chan))) {
833 ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
834 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
835 break;
836 } else if (frame->frametype == AST_FRAME_VOICE) {
837 frame = ast_dsp_process(chan, dsp, frame);
838 if (frame->frametype == AST_FRAME_DTMF) {
839 char result = frame->subclass.integer;
840 if (result == 'q') {
841 hits++;
842 ast_debug(1, "We just detected %s Hz (hit #%d)\n", args.freqs, hits);
843 if (hits >= times) {
844 ast_frfree(frame);
845 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "SUCCESS");
846 break;
847 }
848 }
849 }
850 }
851 ast_frfree(frame);
852 } else {
853 pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
854 }
855 } while (timeout == 0 || remaining_time > 0);
856 ast_dsp_free(dsp);
857
858 return 0;
859}
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 348 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 1042 of file res_tonedetect.c.

Referenced by load_module(), and unload_module().

◆ scanapp

char* scanapp = "ToneScan"
static

Definition at line 862 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 328 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 758 of file res_tonedetect.c.

Referenced by wait_exec().

◆ waitapp

char* waitapp = "WaitForTone"
static

Definition at line 861 of file res_tonedetect.c.

Referenced by load_module(), and unload_module().