main/app.c File Reference

Convenient Application Routines. More...

#include "asterisk.h"
#include <sys/stat.h>
#include <regex.h>
#include <sys/file.h>
#include <signal.h>
#include <stdlib.h>
#include <sys/types.h>
#include <sys/wait.h>
#include <dirent.h>
#include <sys/capability.h>
#include "asterisk/paths.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/file.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/indications.h"
#include "asterisk/linkedlists.h"
#include "asterisk/threadstorage.h"
#include "asterisk/test.h"
#include "asterisk/module.h"
#include "asterisk/astobj2.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/json.h"
#include "asterisk/format_cache.h"

Include dependency graph for main/app.c:

Go to the source code of this file.

Data Structures
struct	groups
struct	linear_state
struct	path_lock
struct	path_lock_list
struct	zombie
struct	zombies

Macros
#define	AST_MAX_FORMATS   10
#define	FMT   "%30Lf%9s"
#define	RES_EXIT   (1 << 17)
#define	RES_REPEAT   (1 << 18)
#define	RES_RESTART   ((1 << 19) | RES_REPEAT)
#define	RES_UPONE   (1 << 16)
#define	VM_API_CALL(res, api_call, api_parms)
#define	VM_GREETER_API_CALL(res, api_call, api_parms)

Enumerations
enum	control_tone_frame_response_result { CONTROL_TONE_RESPONSE_FAILED = -1, CONTROL_TONE_RESPONSE_NORMAL = 0, CONTROL_TONE_RESPONSE_FINISHED = 1 }

Functions
unsigned int	__ast_app_separate_args (char *buf, char delim, int remove_chars, char **array, int arraylen)
	Separate a string into arguments in an array. More...
static int	__ast_play_and_record (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence, const char *path, int prepend, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
int	__ast_vm_greeter_register (const struct ast_vm_greeter_functions *vm_table, struct ast_module *module)
	Set voicemail greeter function callbacks. More...
int	__ast_vm_register (const struct ast_vm_functions *vm_table, struct ast_module *module)
	Set voicemail function callbacks. More...
static	AO2_GLOBAL_OBJ_STATIC (vm_provider)
	The container for the voicemail provider. More...
static	AO2_GLOBAL_OBJ_STATIC (vm_provider_holder)
	Holding container for the voicemail provider used while testing. More...
static	AO2_GLOBAL_OBJ_STATIC (vm_greeter_provider)
	The container for the voicemail greeter provider. More...
static void	app_cleanup (void)
int	app_init (void)
	Initialize the application core. More...
int	ast_app_copy_recording_to_vm (struct ast_vm_recording_data *vm_rec_data)
	param[in] vm_rec_data Contains data needed to make the recording. retval 0 voicemail successfully created from recording. retval -1 Failure More...
int	ast_app_dtget (struct ast_channel *chan, const char *context, char *collect, size_t size, int maxlen, int timeout)
	This function presents a dialtone and reads an extension into 'collect' which must be a pointer to a pre-initialized array of char having a size of 'size' suitable for writing to. It will collect no more than the smaller of 'maxlen' or 'size' minus the original strlen() of collect digits. More...
int	ast_app_exec_macro (struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const char *macro_args)
	Run a macro on a channel, placing an optional second channel into autoservice. More...
int	ast_app_exec_sub (struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const char *sub_args, int ignore_hangup)
	Run a subroutine on a channel, placing an optional second channel into autoservice. More...
const char *	ast_app_expand_sub_args (struct ast_channel *chan, const char *args)
	Add missing context/exten to subroutine argument string. More...
enum ast_getdata_result	ast_app_getdata (struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout)
	ast_app_getdata More...
int	ast_app_getdata_full (struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout, int audiofd, int ctrlfd)
	Full version with audiofd and controlfd. NOTE: returns '2' on ctrlfd available, not '1' like other full functions. More...
int	ast_app_group_discard (struct ast_channel *chan)
	Discard all group counting for a channel. More...
int	ast_app_group_get_count (const char *group, const char *category)
	Get the current channel count of the specified group and category. More...
struct ast_group_info *	ast_app_group_list_head (void)
	Get the head of the group count list. More...
int	ast_app_group_list_rdlock (void)
	Read Lock the group count list. More...
int	ast_app_group_list_unlock (void)
	Unlock the group count list. More...
int	ast_app_group_list_wrlock (void)
	Write Lock the group count list. More...
int	ast_app_group_match_get_count (const char *groupmatch, const char *category)
	Get the current channel count of all groups that match the specified pattern and category. More...
int	ast_app_group_set_channel (struct ast_channel *chan, const char *data)
	Set the group for a channel, splitting the provided data into group and category, if specified. More...
int	ast_app_group_split_group (const char *data, char *group, int group_max, char *category, int category_max)
	Split a group string into group and category, returning a default category if none is provided. More...
int	ast_app_group_update (struct ast_channel *old, struct ast_channel *new)
	Update all group counting for a channel to a new one. More...
int	ast_app_has_voicemail (const char *mailboxes, const char *folder)
	Determine if a given mailbox has any voicemail If folder is NULL, defaults to "INBOX". If folder is "INBOX", includes the number of messages in the "Urgent" folder. More...
int	ast_app_inboxcount (const char *mailboxes, int *newmsgs, int *oldmsgs)
	Determine number of new/old messages in a mailbox. More...
int	ast_app_inboxcount2 (const char *mailboxes, int *urgentmsgs, int *newmsgs, int *oldmsgs)
	Determine number of urgent/new/old messages in a mailbox. More...
int	ast_app_messagecount (const char *mailbox_id, const char *folder)
	Get the number of messages in a given mailbox folder. More...
void	ast_app_options2str64 (const struct ast_app_option *options, struct ast_flags64 *flags, char *buf, size_t len)
	Given a list of options array, return an option string based on passed flags. More...
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. More...
int	ast_app_parse_options64 (const struct ast_app_option *options, struct ast_flags64 *flags, char **args, char *optstr)
	Parses a string containing application options and sets flags/arguments. More...
int	ast_app_parse_timelen (const char *timestr, int *result, enum ast_timelen unit)
	Common routine to parse time lengths, with optional time unit specifier. More...
int	ast_app_run_macro (struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const char *macro_name, const char *macro_args)
	Run a macro on a channel, placing an optional second channel into autoservice. More...
int	ast_app_run_sub (struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const char *sub_location, const char *sub_args, int ignore_hangup)
	Run a subroutine on a channel, placing an optional second channel into autoservice. More...
int	ast_app_sayname (struct ast_channel *chan, const char *mailbox_id)
	Play a recorded user name for the mailbox to the specified channel. More...
void	ast_close_fds_above_n (int n)
	Common routine for child processes, to close all fds prior to exec(2) More...
int	ast_control_streamfile (struct ast_channel *chan, const char *file, const char *fwd, const char *rev, const char *stop, const char *suspend, const char *restart, int skipms, long *offsetms)
	Stream a file with fast forward, pause, reverse, restart. More...
int	ast_control_streamfile_lang (struct ast_channel *chan, const char *file, const char *fwd, const char *rev, const char *stop, const char *suspend, const char *restart, int skipms, const char *lang, long *offsetms)
	Version of ast_control_streamfile() which allows the language of the media file to be specified. More...
int	ast_control_streamfile_w_cb (struct ast_channel *chan, const char *file, const char *fwd, const char *rev, const char *stop, const char *suspend, const char *restart, int skipms, long *offsetms, ast_waitstream_fr_cb cb)
	Stream a file with fast forward, pause, reverse, restart. More...
int	ast_control_tone (struct ast_channel *chan, const char *tone)
	Controls playback of a tone. More...
int	ast_dtmf_stream (struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between, unsigned int duration)
	Send a string of DTMF digits to a channel. More...
void	ast_dtmf_stream_external (struct ast_channel *chan, const char *digits, int between, unsigned int duration)
	Send a string of DTMF digits to a channel from an external thread. More...
int	ast_get_encoded_char (const char *stream, char *result, size_t *consumed)
	Decode an encoded control or extended ASCII character. More...
char *	ast_get_encoded_str (const char *stream, char *result, size_t result_size)
	Decode a stream of encoded control or extended ASCII characters. More...
void	ast_install_stack_functions (const struct ast_app_stack_funcs *funcs)
	Set stack application function callbacks. More...
void	ast_install_vm_test_functions (ast_vm_test_create_user_fn *vm_test_create_user_func, ast_vm_test_destroy_user_fn *vm_test_destroy_user_func)
int	ast_ivr_menu_run (struct ast_channel *chan, struct ast_ivr_menu *menu, void *cbdata)
	Runs an IVR menu. More...
static int	ast_ivr_menu_run_internal (struct ast_channel *chan, struct ast_ivr_menu *menu, void *cbdata)
int	ast_linear_stream (struct ast_channel *chan, const char *filename, int fd, int allowoverride)
	Stream a filename (or file descriptor) as a generator. More...
enum AST_LOCK_RESULT	ast_lock_path (const char *path)
	Lock a filesystem path. More...
static enum AST_LOCK_RESULT	ast_lock_path_flock (const char *path)
static enum AST_LOCK_RESULT	ast_lock_path_lockfile (const char *path)
int	ast_mf_stream (struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between, unsigned int duration, unsigned int durationkp, unsigned int durationst, int is_external)
	Send a string of MF digits to a channel. More...
int	ast_play_and_prepend (struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence)
	Record a file based on input frm a channel. Recording is performed in 'prepend' mode which works a little differently from normal recordings This function will not play a success message due to post-recording control in the application this was added for. More...
int	ast_play_and_record (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence, const char *path)
	Record a file based on input from a channel. Use default accept and cancel DTMF. This function will play "auth-thankyou" upon successful recording. More...
int	ast_play_and_record_full (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
	Record a file based on input from a channel This function will play "auth-thankyou" upon successful recording if skip_confirmation_sound is false. More...
int	ast_play_and_wait (struct ast_channel *chan, const char *fn)
	Play a stream and wait for a digit, returning the digit that was pressed. More...
struct stasis_topic *	ast_queue_topic (const char *queuename)
	Get the Stasis Message Bus API topic for queue messages for a particular queue name. More...
struct stasis_topic *	ast_queue_topic_all (void)
	Get the Stasis Message Bus API topic for queue messages. More...
char *	ast_read_textfile (const char *filename)
	Read a file into asterisk. More...
int	ast_record_review (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, const char *path)
	Allow to record message and have a review option. More...
int	ast_safe_fork (int stop_reaper)
	Common routine to safely fork without a chance of a signal handler firing badly in the child. More...
void	ast_safe_fork_cleanup (void)
	Common routine to cleanup after fork'ed process is complete (if reaping was stopped) More...
void	ast_set_lock_type (enum AST_LOCK_TYPE type)
	Set the type of locks used by ast_lock_path() More...
int	ast_str_get_encoded_str (struct ast_str **str, int maxlen, const char *stream)
	Decode a stream of encoded control or extended ASCII characters. More...
	AST_THREADSTORAGE_PUBLIC (ast_str_thread_global_buf)
void	ast_uninstall_vm_test_functions (void)
int	ast_unlock_path (const char *path)
	Unlock a path. More...
static int	ast_unlock_path_flock (const char *path)
static int	ast_unlock_path_lockfile (const char *path)
int	ast_vm_greeter_is_registered (void)
	Determine if a voicemail greeter provider is registered. More...
void	ast_vm_greeter_unregister (const char *module_name)
	Unregister the specified voicemail greeter provider. More...
const char *	ast_vm_index_to_foldername (int id)
	Return name of folder, given an id. More...
int	ast_vm_is_registered (void)
	Determine if a voicemail provider is registered. More...
struct ast_vm_mailbox_snapshot *	ast_vm_mailbox_snapshot_create (const char *mailbox, const char *context, const char *folder, int descending, enum ast_vm_snapshot_sort_val sort_val, int combine_INBOX_and_OLD)
	Create a snapshot of a mailbox which contains information about every msg. More...
struct ast_vm_mailbox_snapshot *	ast_vm_mailbox_snapshot_destroy (struct ast_vm_mailbox_snapshot *mailbox_snapshot)
	destroy a snapshot More...
int	ast_vm_msg_forward (const char *from_mailbox, const char *from_context, const char *from_folder, const char *to_mailbox, const char *to_context, const char *to_folder, size_t num_msgs, const char *msg_ids[], int delete_old)
	forward a message from one mailbox to another. More...
int	ast_vm_msg_move (const char *mailbox, const char *context, size_t num_msgs, const char *oldfolder, const char *old_msg_ids[], const char *newfolder)
	Move messages from one folder to another. More...
int	ast_vm_msg_play (struct ast_channel *chan, const char *mailbox, const char *context, const char *folder, const char *msg_num, ast_vm_msg_play_cb *cb)
	Play a voicemail msg back on a channel. More...
int	ast_vm_msg_remove (const char *mailbox, const char *context, size_t num_msgs, const char *folder, const char *msgs[])
	Remove/delete messages from a mailbox folder. More...
int	ast_vm_test_create_user (const char *context, const char *mailbox)
int	ast_vm_test_destroy_user (const char *context, const char *mailbox)
void	ast_vm_test_swap_table_in (const struct ast_vm_functions *vm_table)
	Swap out existing voicemail functions with a temporary set of functions for use with unit tests. More...
void	ast_vm_test_swap_table_out (void)
	Used after ast_vm_test_swap_table_in to restore the original set of voicemail functions. More...
void	ast_vm_unregister (const char *module_name)
	Unregister the specified voicemail provider. More...
static int	control_streamfile (struct ast_channel *chan, const char *file, const char *fwd, const char *rev, const char *stop, const char *suspend, const char *restart, int skipms, long *offsetms, const char *lang, ast_waitstream_fr_cb cb)
static enum control_tone_frame_response_result	control_tone_frame_response (struct ast_channel *chan, struct ast_frame *fr, struct ast_tone_zone_sound *ts, const char *tone, int *paused)
static int	dtmf_stream (struct ast_channel *chan, const char *digits, int between, unsigned int duration, int is_external)
static int	external_sleep (struct ast_channel *chan, int ms)
static int	ivr_dispatch (struct ast_channel *chan, struct ast_ivr_option *option, char *exten, void *cbdata)
static void *	linear_alloc (struct ast_channel *chan, void *params)
static int	linear_generator (struct ast_channel *chan, void *data, int len, int samples)
static void	linear_release (struct ast_channel *chan, void *params)
static struct ast_frame *	make_silence (const struct ast_frame *orig)
	Construct a silence frame of the same duration as orig. More...
static int	mf_stream (struct ast_channel *chan, const char *digits, int between, unsigned int duration, unsigned int durationkp, unsigned int durationst, int is_external)
static int	option_exists (struct ast_ivr_menu *menu, char *option)
static int	option_matchmore (struct ast_ivr_menu *menu, char *option)
static int	parse_options (const struct ast_app_option *options, void *_flags, char **args, char *optstr, int flaglen)
static int	parse_tone_uri (char *tone_parser, const char **tone_indication, const char **tone_zone)
static void	path_lock_destroy (struct path_lock *obj)
static int	read_newoption (struct ast_channel *chan, struct ast_ivr_menu *menu, char *exten, int maxexten)
static int	set_read_to_slin (struct ast_channel *chan, struct ast_format **orig_format)
	Sets a channel's read format to AST_FORMAT_SLINEAR, recording its original format. More...
static void *	shaun_of_the_dead (void *data)
static void	vm_greeter_warn_no_provider (void)
static void	vm_warn_no_provider (void)

Variables
static const struct ast_app_stack_funcs *	app_stack_callbacks
static enum AST_LOCK_TYPE	ast_lock_type = AST_LOCK_TYPE_LOCKFILE
static ast_vm_test_create_user_fn *	ast_vm_test_create_user_func = NULL
static ast_vm_test_destroy_user_fn *	ast_vm_test_destroy_user_func = NULL
static cap_t	child_cap
static const char	default_acceptdtmf [] = "#"
static const char	default_canceldtmf [] = ""
static int	global_maxsilence = 0
static int	global_silence_threshold = 128
static struct groups	groups = { .first = NULL, .last = NULL, .lock = { PTHREAD_RWLOCK_INITIALIZER , NULL, {1, 0} } , }
static struct ast_generator	linearstream
static struct path_lock_list	path_lock_list = { .first = NULL, .last = NULL, .lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, {1, 0} } , }
static int	provider_is_swapped = 0
static struct stasis_topic *	queue_topic_all
static struct stasis_topic_pool *	queue_topic_pool
static pthread_t	shaun_of_the_dead_thread = AST_PTHREADT_NULL
static int	vm_greeter_warnings
static int	vm_warnings
static struct zombies	zombies = { .first = NULL, .last = NULL, .lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, {1, 0} } , }

Detailed Description

Convenient Application Routines.

Author

Mark Spencer marks.nosp@m.ter@.nosp@m.digiu.nosp@m.m.co.nosp@m.m

Definition in file main/app.c.

Macro Definition Documentation

AST_MAX_FORMATS

#define AST_MAX_FORMATS   10

Definition at line 121 of file main/app.c.

Referenced by __ast_play_and_record().

FMT

#define FMT   "%30Lf%9s"

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Referenced by ast_app_parse_timelen().

RES_EXIT

#define RES_EXIT   (1 << 17)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2584 of file main/app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

RES_REPEAT

#define RES_REPEAT   (1 << 18)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2585 of file main/app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

RES_RESTART

#define RES_RESTART   ((1 << 19) | RES_REPEAT)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2586 of file main/app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

RES_UPONE

#define RES_UPONE   (1 << 16)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2583 of file main/app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

VM_API_CALL

#define VM_API_CALL	(		res,
			api_call,
			api_parms
	)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 620 of file main/app.c.

Referenced by ast_app_copy_recording_to_vm(), ast_app_has_voicemail(), ast_app_inboxcount(), ast_app_inboxcount2(), ast_app_messagecount(), ast_vm_index_to_foldername(), ast_vm_mailbox_snapshot_create(), ast_vm_mailbox_snapshot_destroy(), ast_vm_msg_forward(), ast_vm_msg_move(), ast_vm_msg_play(), and ast_vm_msg_remove().

VM_GREETER_API_CALL

#define VM_GREETER_API_CALL	(		res,
			api_call,
			api_parms
	)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 641 of file main/app.c.

Referenced by ast_app_sayname().

Enumeration Type Documentation

control_tone_frame_response_result

enum control_tone_frame_response_result

Enumerator
CONTROL_TONE_RESPONSE_FAILED
CONTROL_TONE_RESPONSE_NORMAL
CONTROL_TONE_RESPONSE_FINISHED

Definition at line 1334 of file main/app.c.

                                        {
   CONTROL_TONE_RESPONSE_FAILED = -1,
   CONTROL_TONE_RESPONSE_NORMAL = 0,
   CONTROL_TONE_RESPONSE_FINISHED = 1,
};

Function Documentation

__ast_app_separate_args()

unsigned int __ast_app_separate_args	(	char *	buf,
		char	delim,
		int	remove_chars,
		char **	array,
		int	arraylen
	)

Separate a string into arguments in an array.

Parameters

buf	The string to be parsed (this must be a writable copy, as it will be modified)
delim	The character to be used to delimit arguments
remove_chars	Remove backslashes and quote characters, while parsing
array	An array of 'char *' to be filled in with pointers to the found arguments
arraylen	The number of elements in the array (i.e. the number of arguments you will accept)

Note: if there are more arguments in the string than the array will hold, the last element of the array will contain the remaining arguments, not separated.

The array will be completely zeroed by this function before it populates any entries.

Returns

The number of arguments found, or zero if the function arguments are not valid.

Definition at line 2199 of file main/app.c.

References buf, NULL, paren, and quote().

{
   int argc;
   char *scan, *wasdelim = NULL;
   int paren = 0, quote = 0, bracket = 0;

   if (!array || !arraylen) {
      return 0;
   }

   memset(array, 0, arraylen * sizeof(*array));

   if (!buf) {
      return 0;
   }

   scan = buf;

   for (argc = 0; *scan && (argc < arraylen - 1); argc++) {
      array[argc] = scan;
      for (; *scan; scan++) {
         if (*scan == '(') {
            paren++;
         } else if (*scan == ')') {
            if (paren) {
               paren--;
            }
         } else if (*scan == '[') {
            bracket++;
         } else if (*scan == ']') {
            if (bracket) {
               bracket--;
            }
         } else if (*scan == '"' && delim != '"') {
            quote = quote ? 0 : 1;
            if (remove_chars) {
               /* Remove quote character from argument */
               memmove(scan, scan + 1, strlen(scan));
               scan--;
            }
         } else if (*scan == '\\') {
            if (remove_chars) {
               /* Literal character, don't parse */
               memmove(scan, scan + 1, strlen(scan));
            } else {
               scan++;
            }
         } else if ((*scan == delim) && !paren && !quote && !bracket) {
            wasdelim = scan;
            *scan++ = '\0';
            break;
         }
      }
   }

   /* If the last character in the original string was the delimiter, then
    * there is one additional argument. */
   if (*scan || (scan > buf && (scan - 1) == wasdelim)) {
      array[argc++] = scan;
   }

   return argc;
}

__ast_play_and_record()

static int __ast_play_and_record ( struct ast_channel *  chan, const char *  playfile, const char *  recordfile, int  maxtime, const char *  fmt, int *  duration, int *  sound_duration, int  beep, int  silencethreshold, int  maxsilence, const char *  path, int  prepend, const char *  acceptdtmf, const char *  canceldtmf, int  skip_confirmation_sound, enum ast_record_if_exists  if_exists  )

static

Optionally play a sound file or a beep, then record audio and video from the channel.

Parameters

chan	Channel to playback to/record from.
playfile	Filename of sound to play before recording begins.
recordfile	Filename to record to.
maxtime	Maximum length of recording (in seconds).
fmt	Format(s) to record message in. Multiple formats may be specified by separating them with a '|'.
duration	Where to store actual length of the recorded message (in milliseconds).
sound_duration	Where to store the length of the recorded message (in milliseconds), minus any silence
beep	Whether to play a beep before starting to record.
silencethreshold
maxsilence	Length of silence that will end a recording (in milliseconds).
path	Optional filesystem path to unlock.
prepend	If true, prepend the recorded audio to an existing file and follow prepend mode recording rules
acceptdtmf	DTMF digits that will end the recording.
canceldtmf	DTMF digits that will cancel the recording.
skip_confirmation_sound	If true, don't play auth-thankyou at end. Nice for custom recording prompts in apps.

Return values

-1	failure or hangup
'S'	Recording ended from silence timeout
't'	Recording ended from the message exceeding the maximum duration, or via DTMF in prepend mode
dtmfchar	Recording ended via the return value's DTMF character for either cancel or accept.

Note

Instead of asking how much time passed (end - start), calculate the number of seconds of audio which actually went into the file. This fixes a problem where audio is stopped up on the network and never gets to us.

Note that we still want to use the number of seconds passed for the max message, otherwise we could get a situation where this stream is never closed (which would create a resource leak).

Note

If we ended with silence, trim all but the first 200ms of silence off the recording. However, if we ended with '#', we don't want to trim ANY part of the recording.

Same logic as above.

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1580 of file main/app.c.

References ao2_cleanup, ast_assert, ast_channel_name(), ast_channel_start_silence_generator(), ast_channel_stop_silence_generator(), ast_closestream(), AST_CONTROL_RECORD_CANCEL, AST_CONTROL_RECORD_MUTE, AST_CONTROL_RECORD_STOP, AST_CONTROL_RECORD_SUSPEND, AST_CONTROL_VIDUPDATE, ast_copy_string(), ast_debug, ast_dsp_free(), ast_dsp_new(), ast_dsp_set_threshold(), ast_dsp_silence(), AST_FILE_MODE, ast_filedelete(), ast_filerename(), ast_format_get_name(), AST_FRAME_CONTROL, AST_FRAME_DTMF, ast_frame_dtor(), AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frfree, ast_indicate(), ast_log, AST_MAX_FORMATS, ast_opt_transmit_silence, ast_play_and_wait(), ast_read(), ast_readfile(), ast_readframe(), AST_RECORD_IF_EXISTS_APPEND, AST_RECORD_IF_EXISTS_ERROR, AST_RECORD_IF_EXISTS_FAIL, AST_RECORD_IF_EXISTS_OVERWRITE, ast_set_read_format(), ast_strdupa, ast_stream_and_wait(), ast_stream_rewind(), ast_tellstream(), ast_truncstream(), ast_unlock_path(), ast_verb, ast_waitfor(), ast_writefile(), ast_writestream(), comment, d, end, ast_frame::frametype, global_maxsilence, global_silence_threshold, ast_frame_subclass::integer, LOG_WARNING, make_silence(), muted, NULL, PATH_MAX, set_read_to_slin(), strsep(), ast_frame::subclass, and ast_dsp::totalsilence.

Referenced by ast_play_and_prepend(), ast_play_and_record(), and ast_play_and_record_full().

{
   int d = 0;
   char *fmts;
   char comment[256];
   int x, fmtcnt = 1, res = -1, outmsg = 0;
   struct ast_filestream *others[AST_MAX_FORMATS];
   const char *sfmt[AST_MAX_FORMATS];
   char *stringp = NULL;
   time_t start, end;
   struct ast_dsp *sildet = NULL;   /* silence detector dsp */
   int totalsilence = 0;
   int dspsilence = 0;
   int olddspsilence = 0;
   struct ast_format *rfmt = NULL;
   struct ast_silence_generator *silgen = NULL;
   char prependfile[PATH_MAX];
   int ioflags;   /* IO flags for writing output file */

   ioflags = O_CREAT|O_WRONLY;

   switch (if_exists) {
   case AST_RECORD_IF_EXISTS_FAIL:
      ioflags |= O_EXCL;
      break;
   case AST_RECORD_IF_EXISTS_OVERWRITE:
      ioflags |= O_TRUNC;
      break;
   case AST_RECORD_IF_EXISTS_APPEND:
      ioflags |= O_APPEND;
      break;
   case AST_RECORD_IF_EXISTS_ERROR:
      ast_assert(0);
      break;
   }

   if (silencethreshold < 0) {
      silencethreshold = global_silence_threshold;
   }

   if (maxsilence < 0) {
      maxsilence = global_maxsilence;
   }

   /* barf if no pointer passed to store duration in */
   if (!duration) {
      ast_log(LOG_WARNING, "Error play_and_record called without duration pointer\n");
      return -1;
   }

   ast_debug(1, "play_and_record: %s, %s, '%s'\n", playfile ? playfile : "<None>", recordfile, fmt);
   snprintf(comment, sizeof(comment), "Playing %s, Recording to: %s on %s\n", playfile ? playfile : "<None>", recordfile, ast_channel_name(chan));

   if (playfile || beep) {
      if (!beep) {
         d = ast_play_and_wait(chan, playfile);
      }
      if (d > -1) {
         d = ast_stream_and_wait(chan, "beep", "");
      }
      if (d < 0) {
         return -1;
      }
   }

   if (prepend) {
      ast_copy_string(prependfile, recordfile, sizeof(prependfile));
      strncat(prependfile, "-prepend", sizeof(prependfile) - strlen(prependfile) - 1);
   }

   fmts = ast_strdupa(fmt);

   stringp = fmts;
   strsep(&stringp, "|");
   ast_debug(1, "Recording Formats: sfmts=%s\n", fmts);
   sfmt[0] = ast_strdupa(fmts);

   while ((fmt = strsep(&stringp, "|"))) {
      if (fmtcnt > AST_MAX_FORMATS - 1) {
         ast_log(LOG_WARNING, "Please increase AST_MAX_FORMATS in file.h\n");
         break;
      }
      /*
       * Storage for 'fmt' is on the stack and held by 'fmts', which is maintained for
       * the rest of this function. So okay to not duplicate 'fmt' here, but only keep
       * a pointer to it.
       */
      sfmt[fmtcnt++] = fmt;
   }

   end = start = time(NULL);  /* pre-initialize end to be same as start in case we never get into loop */
   for (x = 0; x < fmtcnt; x++) {
      others[x] = ast_writefile(prepend ? prependfile : recordfile, sfmt[x], comment, ioflags, 0, AST_FILE_MODE);
      ast_verb(3, "x=%d, open writing:  %s format: %s, %p\n", x, prepend ? prependfile : recordfile, sfmt[x], others[x]);

      if (!others[x]) {
         break;
      }
   }

   if (path) {
      ast_unlock_path(path);
   }

   if (maxsilence > 0) {
      sildet = ast_dsp_new(); /* Create the silence detector */
      if (!sildet) {
         ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
         return -1;
      }
      ast_dsp_set_threshold(sildet, silencethreshold);
      res = set_read_to_slin(chan, &rfmt);
      if (res < 0) {
         ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
         ast_dsp_free(sildet);
         ao2_cleanup(rfmt);
         return -1;
      }
   }

   if (!prepend) {
      /* Request a video update */
      ast_indicate(chan, AST_CONTROL_VIDUPDATE);

      if (ast_opt_transmit_silence) {
         silgen = ast_channel_start_silence_generator(chan);
      }
   }

   if (x == fmtcnt) {
      /* Loop, writing the packets we read to the writer(s), until
       * we have reason to stop. */
      struct ast_frame *f;
      int paused = 0;
      int muted = 0;
      time_t pause_start = 0;
      int paused_secs = 0;
      int pausedsilence = 0;

      for (;;) {
         if (!(res = ast_waitfor(chan, 2000))) {
            ast_debug(1, "One waitfor failed, trying another\n");
            /* Try one more time in case of masq */
            if (!(res = ast_waitfor(chan, 2000))) {
               ast_log(LOG_WARNING, "No audio available on %s??\n", ast_channel_name(chan));
               res = -1;
            }
         }

         if (res < 0) {
            f = NULL;
            break;
         }
         if (!(f = ast_read(chan))) {
            break;
         }
         if (f->frametype == AST_FRAME_VOICE) {
            /* write each format */
            if (paused) {
               /* It's all good */
               res = 0;
            } else {
               struct ast_frame *silence = NULL;
               struct ast_frame *orig = f;

               if (muted) {
                  silence = make_silence(orig);
                  if (!silence) {
                     ast_log(LOG_WARNING, "Error creating silence\n");
                     break;
                  }
                  f = silence;
               }
               for (x = 0; x < fmtcnt; x++) {
                  if (prepend && !others[x]) {
                     break;
                  }
                  res = ast_writestream(others[x], f);
               }
               ast_frame_dtor(silence);
               f = orig;
            }

            /* Silence Detection */
            if (maxsilence > 0) {
               dspsilence = 0;
               ast_dsp_silence(sildet, f, &dspsilence);
               if (olddspsilence > dspsilence) {
                  totalsilence += olddspsilence;
               }
               olddspsilence = dspsilence;

               if (paused) {
                  /* record how much silence there was while we are paused */
                  pausedsilence = dspsilence;
               } else if (dspsilence > pausedsilence) {
                  /* ignore the paused silence */
                  dspsilence -= pausedsilence;
               } else {
                  /* dspsilence has reset, reset pausedsilence */
                  pausedsilence = 0;
               }

               if (dspsilence > maxsilence) {
                  /* Ended happily with silence */
                  ast_verb(3, "Recording automatically stopped after a silence of %d seconds\n", dspsilence/1000);
                  res = 'S';
                  outmsg = 2;
                  break;
               }
            }
            /* Exit on any error */
            if (res) {
               ast_log(LOG_WARNING, "Error writing frame\n");
               break;
            }
         } else if (f->frametype == AST_FRAME_VIDEO) {
            /* Write only once */
            ast_writestream(others[0], f);
         } else if (f->frametype == AST_FRAME_DTMF) {
            if (prepend) {
            /* stop recording with any digit */
               ast_verb(3, "User ended message by pressing %c\n", f->subclass.integer);
               res = 't';
               outmsg = 2;
               break;
            }
            if (strchr(acceptdtmf, f->subclass.integer)) {
               ast_verb(3, "User ended message by pressing %c\n", f->subclass.integer);
               res = f->subclass.integer;
               outmsg = 2;
               break;
            }
            if (strchr(canceldtmf, f->subclass.integer)) {
               ast_verb(3, "User canceled message by pressing %c\n", f->subclass.integer);
               res = f->subclass.integer;
               outmsg = 0;
               break;
            }
         } else if (f->frametype == AST_FRAME_CONTROL) {
            if (f->subclass.integer == AST_CONTROL_RECORD_CANCEL) {
               ast_verb(3, "Message canceled by control\n");
               outmsg = 0; /* cancels the recording */
               res = 0;
               break;
            } else if (f->subclass.integer == AST_CONTROL_RECORD_STOP) {
               ast_verb(3, "Message ended by control\n");
               res = 0;
               break;
            } else if (f->subclass.integer == AST_CONTROL_RECORD_SUSPEND) {
               paused = !paused;
               ast_verb(3, "Message %spaused by control\n",
                  paused ? "" : "un");
               if (paused) {
                  pause_start = time(NULL);
               } else {
                  paused_secs += time(NULL) - pause_start;
               }
            } else if (f->subclass.integer == AST_CONTROL_RECORD_MUTE) {
               muted = !muted;
               ast_verb(3, "Message %smuted by control\n",
                  muted ? "" : "un");
               /* We can only silence slin frames, so
                * set the mode, if we haven't already
                * for sildet
                */
               if (muted && !rfmt) {
                  ast_verb(3, "Setting read format to linear mode\n");
                  res = set_read_to_slin(chan, &rfmt);
                  if (res < 0) {
                     ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
                     break;
                  }
               }
            }
         }
         if (maxtime && !paused) {
            end = time(NULL);
            if (maxtime < (end - start - paused_secs)) {
               ast_verb(3, "Took too long, cutting it short...\n");
               res = 't';
               outmsg = 2;
               break;
            }
         }
         ast_frfree(f);
      }
      if (!f) {
         ast_verb(3, "User hung up\n");
         res = -1;
         outmsg = 1;
      } else {
         ast_frfree(f);
      }
   } else {
      ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", recordfile, sfmt[x]);
   }

   if (!prepend) {
      if (silgen) {
         ast_channel_stop_silence_generator(chan, silgen);
      }
   }

   /*!\note
    * Instead of asking how much time passed (end - start), calculate the number
    * of seconds of audio which actually went into the file.  This fixes a
    * problem where audio is stopped up on the network and never gets to us.
    *
    * Note that we still want to use the number of seconds passed for the max
    * message, otherwise we could get a situation where this stream is never
    * closed (which would create a resource leak).
    */
   *duration = others[0] ? ast_tellstream(others[0]) / 8000 : 0;
   if (sound_duration) {
      *sound_duration = *duration;
   }

   if (!prepend) {
      /* Reduce duration by a total silence amount */
      if (olddspsilence <= dspsilence) {
         totalsilence += dspsilence;
      }

      if (sound_duration) {
         if (totalsilence > 0) {
            *sound_duration -= (totalsilence - 200) / 1000;
         }
         if (*sound_duration < 0) {
            *sound_duration = 0;
         }
      }

      if (dspsilence > 0) {
         *duration -= (dspsilence - 200) / 1000;
      }

      if (*duration < 0) {
         *duration = 0;
      }

      for (x = 0; x < fmtcnt; x++) {
         if (!others[x]) {
            break;
         }
         /*!\note
          * If we ended with silence, trim all but the first 200ms of silence
          * off the recording.  However, if we ended with '#', we don't want
          * to trim ANY part of the recording.
          */
         if (res > 0 && dspsilence) {
            /* rewind only the trailing silence */
            ast_stream_rewind(others[x], dspsilence - 200);
         }
         ast_truncstream(others[x]);
         ast_closestream(others[x]);
      }
   } else if (prepend && outmsg) {
      struct ast_filestream *realfiles[AST_MAX_FORMATS];
      struct ast_frame *fr;

      for (x = 0; x < fmtcnt; x++) {
         snprintf(comment, sizeof(comment), "Opening the real file %s.%s\n", recordfile, sfmt[x]);
         realfiles[x] = ast_readfile(recordfile, sfmt[x], comment, O_RDONLY, 0, 0);
         if (!others[x]) {
            break;
         }
         if (!realfiles[x]) {
            ast_closestream(others[x]);
            continue;
         }
         /*!\note Same logic as above. */
         if (dspsilence) {
            ast_stream_rewind(others[x], dspsilence - 200);
         }
         ast_truncstream(others[x]);
         /* add the original file too */
         while ((fr = ast_readframe(realfiles[x]))) {
            ast_writestream(others[x], fr);
            ast_frfree(fr);
         }
         ast_closestream(others[x]);
         ast_closestream(realfiles[x]);
         ast_filerename(prependfile, recordfile, sfmt[x]);
         ast_verb(4, "Recording Format: sfmts=%s, prependfile %s, recordfile %s\n", sfmt[x], prependfile, recordfile);
         ast_filedelete(prependfile, sfmt[x]);
      }
   } else {
      for (x = 0; x < fmtcnt; x++) {
         if (!others[x]) {
            break;
         }
         ast_closestream(others[x]);
      }
   }

   if (rfmt && ast_set_read_format(chan, rfmt)) {
      ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_format_get_name(rfmt), ast_channel_name(chan));
   }
   ao2_cleanup(rfmt);
   if ((outmsg == 2) && (!skip_confirmation_sound)) {
      ast_stream_and_wait(chan, "auth-thankyou", "");
   }
   if (sildet) {
      ast_dsp_free(sildet);
   }
   return res;
}

__ast_vm_greeter_register()

int __ast_vm_greeter_register	(	const struct ast_vm_greeter_functions *	vm_table,
		struct ast_module *	module
	)

Set voicemail greeter function callbacks.

Since

13.0.0

Parameters

vm_table	Voicemail greeter function table to install.
module	Pointer to the module implementing the interface

Return values

0	on success.
-1	on error.
AST_MODULE_LOAD_DECLINE	if there's already another greeter registered.

Definition at line 552 of file main/app.c.

References AO2_ALLOC_OPT_LOCK_NOLOCK, ao2_alloc_options, ao2_cleanup, ao2_global_obj_ref, ao2_global_obj_replace_unref, ast_log, AST_MODULE_LOAD_DECLINE, LOG_ERROR, LOG_WARNING, ast_vm_greeter_functions::module, ast_vm_greeter_functions::module_name, ast_vm_greeter_functions::module_version, NULL, RAII_VAR, table, VM_GREETER_MODULE_VERSION, and vm_table.

{
   RAII_VAR(struct ast_vm_greeter_functions *, table, NULL, ao2_cleanup);

   if (!vm_table->module_name) {
      ast_log(LOG_ERROR, "Voicemail greeter provider missing required information.\n");
      return -1;
   }
   if (vm_table->module_version != VM_GREETER_MODULE_VERSION) {
      ast_log(LOG_ERROR, "Voicemail greeter provider '%s' has incorrect version\n",
         vm_table->module_name);
      return -1;
   }

   table = ao2_global_obj_ref(vm_greeter_provider);
   if (table) {
      ast_log(LOG_WARNING, "Voicemail greeter provider already registered by %s.\n",
         table->module_name);
      return AST_MODULE_LOAD_DECLINE;
   }

   table = ao2_alloc_options(sizeof(*table), NULL, AO2_ALLOC_OPT_LOCK_NOLOCK);
   if (!table) {
      return -1;
   }
   *table = *vm_table;
   table->module = module;

   ao2_global_obj_replace_unref(vm_greeter_provider, table);
   return 0;
}

__ast_vm_register()

int __ast_vm_register	(	const struct ast_vm_functions *	vm_table,
		struct ast_module *	module
	)

Set voicemail function callbacks.

Parameters

vm_table	Voicemail function table to install.
module	Pointer to the module implementing the interface

Return values

0	on success.
-1	on error.
AST_MODULE_LOAD_DECLINE	if there's already another provider registered.

Definition at line 441 of file main/app.c.

References AO2_ALLOC_OPT_LOCK_NOLOCK, ao2_alloc_options, ao2_cleanup, ao2_global_obj_ref, ao2_global_obj_replace_unref, ast_log, AST_MODULE_LOAD_DECLINE, LOG_ERROR, LOG_WARNING, ast_vm_functions::module, ast_vm_functions::module_name, ast_vm_functions::module_version, NULL, RAII_VAR, table, VM_MODULE_VERSION, and vm_table.

{
   RAII_VAR(struct ast_vm_functions *, table, NULL, ao2_cleanup);

   if (!vm_table->module_name) {
      ast_log(LOG_ERROR, "Voicemail provider missing required information.\n");
      return -1;
   }
   if (vm_table->module_version != VM_MODULE_VERSION) {
      ast_log(LOG_ERROR, "Voicemail provider '%s' has incorrect version\n",
         vm_table->module_name);
      return -1;
   }

   table = ao2_global_obj_ref(vm_provider);
   if (table) {
      ast_log(LOG_WARNING, "Voicemail provider already registered by %s.\n",
         table->module_name);
      return AST_MODULE_LOAD_DECLINE;
   }

   table = ao2_alloc_options(sizeof(*table), NULL, AO2_ALLOC_OPT_LOCK_NOLOCK);
   if (!table) {
      return -1;
   }
   *table = *vm_table;
   table->module = module;

   ao2_global_obj_replace_unref(vm_provider, table);
   return 0;
}

AO2_GLOBAL_OBJ_STATIC() [1/3]

static AO2_GLOBAL_OBJ_STATIC ( vm_provider  )

static

The container for the voicemail provider.

Referenced by ast_app_run_sub(), ast_vm_test_swap_table_out(), and ast_vm_unregister().

AO2_GLOBAL_OBJ_STATIC() [2/3]

static AO2_GLOBAL_OBJ_STATIC ( vm_provider_holder  )

static

Holding container for the voicemail provider used while testing.

AO2_GLOBAL_OBJ_STATIC() [3/3]

static AO2_GLOBAL_OBJ_STATIC ( vm_greeter_provider  )

static

The container for the voicemail greeter provider.

app_cleanup()

static void app_cleanup ( void  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 3194 of file main/app.c.

References ao2_cleanup, child_cap, and NULL.

Referenced by app_init().

{
#ifdef HAS_CAP
   cap_free(child_cap);
#endif
   ao2_cleanup(queue_topic_pool);
   queue_topic_pool = NULL;
   ao2_cleanup(queue_topic_all);
   queue_topic_all = NULL;
}

app_init()

int app_init

(

void

)

Initialize the application core.

Return values

0	Success
-1	Failure

Since

12

Definition at line 3205 of file main/app.c.

References app_cleanup(), ast_register_cleanup(), child_cap, stasis_topic_create(), and stasis_topic_pool_create().

Referenced by asterisk_daemon().

{
   ast_register_cleanup(app_cleanup);
#ifdef HAVE_CAP
   child_cap = cap_from_text("cap_net_admin-eip");
#endif
   queue_topic_all = stasis_topic_create("queue:all");
   if (!queue_topic_all) {
      return -1;
   }
   queue_topic_pool = stasis_topic_pool_create(queue_topic_all);
   if (!queue_topic_pool) {
      return -1;
   }
   return 0;
}

ast_app_copy_recording_to_vm()

int ast_app_copy_recording_to_vm

(

struct ast_vm_recording_data *

vm_rec_data

)

param[in] vm_rec_data Contains data needed to make the recording. retval 0 voicemail successfully created from recording. retval -1 Failure

Definition at line 669 of file main/app.c.

References VM_API_CALL.

Referenced by copy_to_voicemail().

{
   int res = -1;

   VM_API_CALL(res, copy_recording_to_vm, (vm_rec_data));
   return res;
}

ast_app_dtget()

int ast_app_dtget	(	struct ast_channel *	chan,
		const char *	context,
		char *	collect,
		size_t	size,
		int	maxlen,
		int	timeout
	)

This function presents a dialtone and reads an extension into 'collect' which must be a pointer to a pre-initialized array of char having a size of 'size' suitable for writing to. It will collect no more than the smaller of 'maxlen' or 'size' minus the original strlen() of collect digits.

Present a dialtone and collect a certain length extension.

Parameters

chan	struct.
context
collect
size
maxlen
timeout	timeout in milliseconds

Returns

0 if extension does not exist, 1 if extension exists

Definition at line 139 of file main/app.c.

References ast_channel_caller(), ast_channel_pbx(), ast_channel_zone(), ast_exists_extension(), ast_get_indication_tone(), ast_ignore_pattern(), ast_log, ast_matchmore_extension(), ast_playtones_start(), ast_playtones_stop(), ast_tone_zone_sound_unref(), ast_waitfordigit(), ast_tone_zone_sound::data, ast_pbx::dtimeoutms, LOG_NOTICE, NULL, and S_COR.

Referenced by grab_transfer().

{
   struct ast_tone_zone_sound *ts;
   int res = 0, x = 0;

   if (maxlen > size) {
      maxlen = size;
   }

   if (!timeout) {
      if (ast_channel_pbx(chan) && ast_channel_pbx(chan)->dtimeoutms) {
         timeout = ast_channel_pbx(chan)->dtimeoutms;
      } else {
         timeout = 5000;
      }
   }

   if ((ts = ast_get_indication_tone(ast_channel_zone(chan), "dial"))) {
      res = ast_playtones_start(chan, 0, ts->data, 0);
      ts = ast_tone_zone_sound_unref(ts);
   } else {
      ast_log(LOG_NOTICE, "Huh....? no dial for indications?\n");
   }

   for (x = strlen(collect); x < maxlen; ) {
      res = ast_waitfordigit(chan, timeout);
      if (!ast_ignore_pattern(context, collect)) {
         ast_playtones_stop(chan);
      }
      if (res < 1) {
         break;
      }
      if (res == '#') {
         break;
      }
      collect[x++] = res;
      if (!ast_matchmore_extension(chan, context, collect, 1,
         S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
         break;
      }
   }

   if (res >= 0) {
      res = ast_exists_extension(chan, context, collect, 1,
         S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL)) ? 1 : 0;
   }

   return res;
}

ast_app_exec_macro()

int ast_app_exec_macro	(	struct ast_channel *	autoservice_chan,
		struct ast_channel *	macro_chan,
		const char *	macro_args
	)

Run a macro on a channel, placing an optional second channel into autoservice.

Since

11.0

This is a shorthand method that makes it very easy to run a macro on any given channel. It is perfectly reasonable to supply a NULL autoservice_chan here in case there is no channel to place into autoservice.

Note

Absolutely NO channel locks should be held before calling this function.

Parameters

autoservice_chan	A channel to place into autoservice while the macro is run
macro_chan	Channel to execute macro on.
macro_args	Macro application argument string.

Return values

0	success
-1	on error

Definition at line 273 of file main/app.c.

References ast_autoservice_start(), ast_autoservice_stop(), ast_channel_context(), ast_channel_exten(), ast_channel_name(), ast_channel_priority(), ast_check_hangup_locked(), ast_debug, ast_log, ast_queue_hangup(), LOG_WARNING, pbx_exec(), and pbx_findapp().

Referenced by ast_app_run_macro(), dial_exec_full(), generic_recall(), run_app_helper(), and try_calling().

{
   struct ast_app *macro_app;
   int res;

   macro_app = pbx_findapp("Macro");
   if (!macro_app) {
      ast_log(LOG_WARNING,
         "Cannot run 'Macro(%s)'.  The application is not available.\n", macro_args);
      return -1;
   }
   if (autoservice_chan) {
      ast_autoservice_start(autoservice_chan);
   }

   ast_debug(4, "%s Original location: %s,%s,%d\n", ast_channel_name(macro_chan),
      ast_channel_context(macro_chan), ast_channel_exten(macro_chan),
      ast_channel_priority(macro_chan));

   res = pbx_exec(macro_chan, macro_app, macro_args);
   ast_debug(4, "Macro exited with status %d\n", res);

   /*
    * Assume anything negative from Macro is an error.
    * Anything else is success.
    */
   if (res < 0) {
      res = -1;
   } else {
      res = 0;
   }

   ast_debug(4, "%s Ending location: %s,%s,%d\n", ast_channel_name(macro_chan),
      ast_channel_context(macro_chan), ast_channel_exten(macro_chan),
      ast_channel_priority(macro_chan));

   if (autoservice_chan) {
      ast_autoservice_stop(autoservice_chan);
   }

   if (ast_check_hangup_locked(macro_chan)) {
      ast_queue_hangup(macro_chan);
   }

   return res;
}

ast_app_exec_sub()

int ast_app_exec_sub	(	struct ast_channel *	autoservice_chan,
		struct ast_channel *	sub_chan,
		const char *	sub_args,
		int	ignore_hangup
	)

Run a subroutine on a channel, placing an optional second channel into autoservice.

Since

11

This is a shorthand method that makes it very easy to run a subroutine on any given channel. It is perfectly reasonable to supply a NULL autoservice_chan here in case there is no channel to place into autoservice.

Note

Absolutely NO channel locks should be held before calling this function.

Parameters

autoservice_chan	A channel to place into autoservice while the subroutine is run
sub_chan	Channel to execute subroutine on.
sub_args	Gosub application argument string.
ignore_hangup	TRUE if a hangup does not stop execution of the routine.

Return values

0	success
-1	on error

Definition at line 370 of file main/app.c.

References app_stack_callbacks, ast_autoservice_start(), ast_autoservice_stop(), ast_check_hangup_locked(), ast_log, ast_module_running_ref, ast_module_unref, ast_queue_hangup(), LOG_WARNING, ast_app_stack_funcs::module, and ast_app_stack_funcs::run_sub.

Referenced by app_exec(), ast_app_run_sub(), ast_pbx_hangup_handler_run(), ast_pre_call(), dial_exec_full(), generic_recall(), originate_exec(), page_exec(), queue_exec(), run_app_helper(), and try_calling().

{
   const struct ast_app_stack_funcs *funcs;
   int res;

   funcs = app_stack_callbacks;
   if (!funcs || !funcs->run_sub || !ast_module_running_ref(funcs->module)) {
      ast_log(LOG_WARNING,
         "Cannot run 'Gosub(%s)'.  The app_stack module is not available.\n",
         sub_args);
      return -1;
   }

   if (autoservice_chan) {
      ast_autoservice_start(autoservice_chan);
   }

   res = funcs->run_sub(sub_chan, sub_args, ignore_hangup);
   ast_module_unref(funcs->module);

   if (autoservice_chan) {
      ast_autoservice_stop(autoservice_chan);
   }

   if (!ignore_hangup && ast_check_hangup_locked(sub_chan)) {
      ast_queue_hangup(sub_chan);
   }

   return res;
}

ast_app_expand_sub_args()

const char* ast_app_expand_sub_args	(	struct ast_channel *	chan,
		const char *	args
	)

Add missing context/exten to subroutine argument string.

Parameters

chan	Channel to obtain context/exten.
args	Gosub application argument string.

Fills in the optional context and exten from the given channel.

Return values

New-args	Gosub argument string on success. Must be freed.
NULL	on error.

Definition at line 351 of file main/app.c.

References app_stack_callbacks, ast_log, ast_module_running_ref, ast_module_unref, ast_app_stack_funcs::expand_sub_args, LOG_WARNING, ast_app_stack_funcs::module, and NULL.

Referenced by app_exec(), ast_pbx_hangup_handler_push(), dial_exec_full(), and page_exec().

{
   const struct ast_app_stack_funcs *funcs;
   const char *new_args;

   funcs = app_stack_callbacks;
   if (!funcs || !funcs->expand_sub_args || !ast_module_running_ref(funcs->module)) {
      ast_log(LOG_WARNING,
         "Cannot expand 'Gosub(%s)' arguments.  The app_stack module is not available.\n",
         args);
      return NULL;
   }

   new_args = funcs->expand_sub_args(chan, args);
   ast_module_unref(funcs->module);

   return new_args;
}

ast_app_getdata()

enum ast_getdata_result ast_app_getdata	(	struct ast_channel *	c,
		const char *	prompt,
		char *	s,
		int	maxlen,
		int	timeout
	)

ast_app_getdata

Plays a stream and gets DTMF data from a channel.

Parameters

c	The channel to read from
prompt	The file to stream to the channel
s	The string to read in to. Must be at least the size of your length
maxlen	How many digits to read (maximum)
timeout	set timeout to 0 for "standard" timeouts. Set timeout to -1 for "ludicrous time" (essentially never times out)

Definition at line 197 of file main/app.c.

References ast_channel_language(), ast_channel_pbx(), AST_GETDATA_EMPTY_END_TERMINATED, ast_readstring(), ast_strdupa, ast_streamfile(), ast_strlen_zero, ast_pbx::dtimeoutms, ast_pbx::rtimeoutms, strsep(), and timeout.

Referenced by auth_exec(), conf_exec(), conf_get_pin(), dictate_exec(), find_conf(), read_exec(), testclient_exec(), testserver_exec(), and vm_exec().

{
   int res = 0, to, fto;
   char *front, *filename;

   /* XXX Merge with full version? XXX */

   if (maxlen)
      s[0] = '\0';

   if (!prompt)
      prompt = "";

   filename = ast_strdupa(prompt);
   while ((front = strsep(&filename, "&"))) {
      if (!ast_strlen_zero(front)) {
         res = ast_streamfile(c, front, ast_channel_language(c));
         if (res)
            continue;
      }
      if (ast_strlen_zero(filename)) {
         /* set timeouts for the last prompt */
         fto = ast_channel_pbx(c) ? ast_channel_pbx(c)->rtimeoutms : 6000;
         to = ast_channel_pbx(c) ? ast_channel_pbx(c)->dtimeoutms : 2000;

         if (timeout > 0) {
            fto = to = timeout;
         }
         if (timeout < 0) {
            fto = to = 1000000000;
         }
      } else {
         /* there is more than one prompt, so
          * get rid of the long timeout between
          * prompts, and make it 50ms */
         fto = 50;
         to = ast_channel_pbx(c) ? ast_channel_pbx(c)->dtimeoutms : 2000;
      }
      res = ast_readstring(c, s, maxlen, to, fto, "#");
      if (res == AST_GETDATA_EMPTY_END_TERMINATED) {
         return res;
      }
      if (!ast_strlen_zero(s)) {
         return res;
      }
   }

   return res;
}

ast_app_getdata_full()

int ast_app_getdata_full	(	struct ast_channel *	c,
		const char *	prompt,
		char *	s,
		int	maxlen,
		int	timeout,
		int	audiofd,
		int	ctrlfd
	)

Full version with audiofd and controlfd. NOTE: returns '2' on ctrlfd available, not '1' like other full functions.

Definition at line 250 of file main/app.c.

References ast_channel_language(), ast_readstring_full(), ast_streamfile(), ast_strlen_zero, and timeout.

Referenced by handle_getdata().

{
   int res, to = 2000, fto = 6000;

   if (!ast_strlen_zero(prompt)) {
      res = ast_streamfile(c, prompt, ast_channel_language(c));
      if (res < 0) {
         return res;
      }
   }

   if (timeout > 0) {
      fto = to = timeout;
   }
   if (timeout < 0) {
      fto = to = 1000000000;
   }

   res = ast_readstring_full(c, s, maxlen, to, fto, "#", audiofd, ctrlfd);

   return res;
}

ast_app_group_discard()

int ast_app_group_discard

(

struct ast_channel *

chan

)

Discard all group counting for a channel.

Definition at line 2162 of file main/app.c.

References ast_free, AST_RWLIST_REMOVE_CURRENT, AST_RWLIST_TRAVERSE_SAFE_BEGIN, AST_RWLIST_TRAVERSE_SAFE_END, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_group_info::chan, ast_group_info::group_list, and NULL.

Referenced by ast_channel_destructor().

{
   struct ast_group_info *gi = NULL;

   AST_RWLIST_WRLOCK(&groups);
   AST_RWLIST_TRAVERSE_SAFE_BEGIN(&groups, gi, group_list) {
      if (gi->chan == chan) {
         AST_RWLIST_REMOVE_CURRENT(group_list);
         ast_free(gi);
      }
   }
   AST_RWLIST_TRAVERSE_SAFE_END;
   AST_RWLIST_UNLOCK(&groups);

   return 0;
}

ast_app_group_get_count()

int ast_app_group_get_count	(	const char *	group,
		const char *	category
	)

Get the current channel count of the specified group and category.

Definition at line 2083 of file main/app.c.

References AST_RWLIST_RDLOCK, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, ast_strlen_zero, ast_group_info::category, ast_group_info::group, ast_group_info::group_list, and NULL.

Referenced by group_count_function_read().

{
   struct ast_group_info *gi = NULL;
   int count = 0;

   if (ast_strlen_zero(group)) {
      return 0;
   }

   AST_RWLIST_RDLOCK(&groups);
   AST_RWLIST_TRAVERSE(&groups, gi, group_list) {
      if (!strcasecmp(gi->group, group) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category)))) {
         count++;
      }
   }
   AST_RWLIST_UNLOCK(&groups);

   return count;
}

ast_app_group_list_head()

struct ast_group_info* ast_app_group_list_head

(

void

)

Get the head of the group count list.

Definition at line 2189 of file main/app.c.

References AST_RWLIST_FIRST.

Referenced by group_count_function_read(), group_function_read(), group_list_function_read(), and group_show_channels().

{
   return AST_RWLIST_FIRST(&groups);
}

ast_app_group_list_rdlock()

int ast_app_group_list_rdlock

(

void

)

Read Lock the group count list.

Definition at line 2184 of file main/app.c.

References AST_RWLIST_RDLOCK.

Referenced by group_count_function_read(), group_function_read(), group_list_function_read(), and group_show_channels().

{
   return AST_RWLIST_RDLOCK(&groups);
}

ast_app_group_list_unlock()

int ast_app_group_list_unlock

(

void

)

Unlock the group count list.

Definition at line 2194 of file main/app.c.

References AST_RWLIST_UNLOCK.

Referenced by group_count_function_read(), group_function_read(), group_list_function_read(), and group_show_channels().

{
   return AST_RWLIST_UNLOCK(&groups);
}

ast_app_group_list_wrlock()

int ast_app_group_list_wrlock

(

void

)

Write Lock the group count list.

Definition at line 2179 of file main/app.c.

References AST_RWLIST_WRLOCK.

{
   return AST_RWLIST_WRLOCK(&groups);
}

ast_app_group_match_get_count()

int ast_app_group_match_get_count	(	const char *	groupmatch,
		const char *	category
	)

Get the current channel count of all groups that match the specified pattern and category.

Definition at line 2103 of file main/app.c.

References ast_log, AST_RWLIST_RDLOCK, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, ast_strlen_zero, ast_group_info::category, ast_group_info::group, ast_group_info::group_list, LOG_ERROR, LOG_NOTICE, and NULL.

Referenced by AST_TEST_DEFINE(), and group_match_count_function_read().

{
   struct ast_group_info *gi = NULL;
   regex_t regexbuf_group;
   regex_t regexbuf_category;
   int count = 0;

   if (ast_strlen_zero(groupmatch)) {
      ast_log(LOG_NOTICE, "groupmatch empty\n");
      return 0;
   }

   /* if regex compilation fails, return zero matches */
   if (regcomp(&regexbuf_group, groupmatch, REG_EXTENDED | REG_NOSUB)) {
      ast_log(LOG_ERROR, "Regex compile failed on: %s\n", groupmatch);
      return 0;
   }

   if (!ast_strlen_zero(category) && regcomp(&regexbuf_category, category, REG_EXTENDED | REG_NOSUB)) {
      ast_log(LOG_ERROR, "Regex compile failed on: %s\n", category);
      regfree(&regexbuf_group);
      return 0;
   }

   AST_RWLIST_RDLOCK(&groups);
   AST_RWLIST_TRAVERSE(&groups, gi, group_list) {
      if (!regexec(&regexbuf_group, gi->group, 0, NULL, 0) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !regexec(&regexbuf_category, gi->category, 0, NULL, 0)))) {
         count++;
      }
   }
   AST_RWLIST_UNLOCK(&groups);

   regfree(&regexbuf_group);
   if (!ast_strlen_zero(category)) {
      regfree(&regexbuf_category);
   }

   return count;
}

ast_app_group_set_channel()

int ast_app_group_set_channel	(	struct ast_channel *	chan,
		const char *	data
	)

Set the group for a channel, splitting the provided data into group and category, if specified.

Definition at line 2036 of file main/app.c.

References ast_app_group_split_group(), ast_calloc, ast_free, AST_RWLIST_INSERT_TAIL, AST_RWLIST_REMOVE_CURRENT, AST_RWLIST_TRAVERSE_SAFE_BEGIN, AST_RWLIST_TRAVERSE_SAFE_END, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_strlen_zero, ast_group_info::category, ast_group_info::chan, ast_group_info::group, ast_group_info::group_list, len(), and NULL.

Referenced by AST_TEST_DEFINE(), dial_exec_full(), and group_function_write().

{
   int res = 0;
   char group[80] = "", category[80] = "";
   struct ast_group_info *gi = NULL;
   size_t len = 0;

   if (ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category))) {
      return -1;
   }

   /* Calculate memory we will need if this is new */
   len = sizeof(*gi) + strlen(group) + 1;
   if (!ast_strlen_zero(category)) {
      len += strlen(category) + 1;
   }

   AST_RWLIST_WRLOCK(&groups);
   AST_RWLIST_TRAVERSE_SAFE_BEGIN(&groups, gi, group_list) {
      if ((gi->chan == chan) && ((ast_strlen_zero(category) && ast_strlen_zero(gi->category)) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category)))) {
         AST_RWLIST_REMOVE_CURRENT(group_list);
         ast_free(gi);
         break;
      }
   }
   AST_RWLIST_TRAVERSE_SAFE_END;

   if (ast_strlen_zero(group)) {
      /* Enable unsetting the group */
   } else if ((gi = ast_calloc(1, len))) {
      gi->chan = chan;
      gi->group = (char *) gi + sizeof(*gi);
      strcpy(gi->group, group);
      if (!ast_strlen_zero(category)) {
         gi->category = (char *) gi + sizeof(*gi) + strlen(group) + 1;
         strcpy(gi->category, category);
      }
      AST_RWLIST_INSERT_TAIL(&groups, gi, group_list);
   } else {
      res = -1;
   }

   AST_RWLIST_UNLOCK(&groups);

   return res;
}

ast_app_group_split_group()

int ast_app_group_split_group	(	const char *	data,
		char *	group,
		int	group_max,
		char *	category,
		int	category_max
	)

Split a group string into group and category, returning a default category if none is provided.

Definition at line 2009 of file main/app.c.

References ast_copy_string(), ast_strlen_zero, NULL, and tmp().

Referenced by ast_app_group_set_channel(), group_count_function_read(), and group_match_count_function_read().

{
   int res = 0;
   char tmp[256];
   char *grp = NULL, *cat = NULL;

   if (!ast_strlen_zero(data)) {
      ast_copy_string(tmp, data, sizeof(tmp));
      grp = tmp;
      if ((cat = strchr(tmp, '@'))) {
         *cat++ = '\0';
      }
   }

   if (!ast_strlen_zero(grp)) {
      ast_copy_string(group, grp, group_max);
   } else {
      *group = '\0';
   }

   if (!ast_strlen_zero(cat)) {
      ast_copy_string(category, cat, category_max);
   }

   return res;
}

ast_app_group_update()

int ast_app_group_update	(	struct ast_channel *	old,
		struct ast_channel *	new
	)

Update all group counting for a channel to a new one.

Definition at line 2143 of file main/app.c.

References ast_free, AST_RWLIST_REMOVE_CURRENT, AST_RWLIST_TRAVERSE_SAFE_BEGIN, AST_RWLIST_TRAVERSE_SAFE_END, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_group_info::chan, ast_group_info::group_list, and NULL.

Referenced by channel_do_masquerade().

{
   struct ast_group_info *gi = NULL;

   AST_RWLIST_WRLOCK(&groups);
   AST_RWLIST_TRAVERSE_SAFE_BEGIN(&groups, gi, group_list) {
      if (gi->chan == old) {
         gi->chan = new;
      } else if (gi->chan == new) {
         AST_RWLIST_REMOVE_CURRENT(group_list);
         ast_free(gi);
      }
   }
   AST_RWLIST_TRAVERSE_SAFE_END;
   AST_RWLIST_UNLOCK(&groups);

   return 0;
}

ast_app_has_voicemail()

int ast_app_has_voicemail	(	const char *	mailboxes,
		const char *	folder
	)

Determine if a given mailbox has any voicemail If folder is NULL, defaults to "INBOX". If folder is "INBOX", includes the number of messages in the "Urgent" folder.

Return values

1	Mailbox has voicemail
0	No new voicemail in specified mailbox
-1	Failure

Since

1.0

Definition at line 655 of file main/app.c.

References has_voicemail(), and VM_API_CALL.

Referenced by action_mailboxstatus(), AST_TEST_DEFINE(), has_voicemail(), mwi_update_cb(), notify_new_message(), play_dialtone(), poll_mailbox(), run_externnotify(), skinny_register(), and unistim_send_mwi_to_peer().

{
   int res = 0;

   VM_API_CALL(res, has_voicemail, (mailboxes, folder));
   return res;
}

ast_app_inboxcount()

int ast_app_inboxcount	(	const char *	mailboxes,
		int *	newmsgs,
		int *	oldmsgs
	)

Determine number of new/old messages in a mailbox.

Since

1.0

Parameters

[in]	mailboxes	Mailbox specification in the format /code mbox[@context][&mbox2[@context2]][...] /code
[out]	newmsgs	Number of messages in the "INBOX" folder. Includes number of messages in the "Urgent" folder, if any.
[out]	oldmsgs	Number of messages in the "Old" folder.

Return values

0	Success
-1	Failure

Definition at line 677 of file main/app.c.

References inboxcount(), and VM_API_CALL.

Referenced by AST_TEST_DEFINE(), sip_send_mwi_to_peer(), and update_registry().

{
   int res = 0;

   if (newmsgs) {
      *newmsgs = 0;
   }
   if (oldmsgs) {
      *oldmsgs = 0;
   }

   VM_API_CALL(res, inboxcount, (mailboxes, newmsgs, oldmsgs));
   return res;
}

ast_app_inboxcount2()

int ast_app_inboxcount2	(	const char *	mailboxes,
		int *	urgentmsgs,
		int *	newmsgs,
		int *	oldmsgs
	)

Determine number of urgent/new/old messages in a mailbox.

Parameters

[in]	mailboxes	the mailbox context to use
[out]	urgentmsgs	the urgent message count
[out]	newmsgs	the new message count
[out]	oldmsgs	the old message count

Returns

Returns 0 for success, negative upon error

Since

1.6.1

Definition at line 692 of file main/app.c.

References inboxcount2(), and VM_API_CALL.

Referenced by action_mailboxcount(), AST_TEST_DEFINE(), mwi_retrieve_then_create_state(), notify_new_message(), notify_new_state(), and vm_execmain().

{
   int res = 0;

   if (newmsgs) {
      *newmsgs = 0;
   }
   if (oldmsgs) {
      *oldmsgs = 0;
   }
   if (urgentmsgs) {
      *urgentmsgs = 0;
   }

   VM_API_CALL(res, inboxcount2, (mailboxes, urgentmsgs, newmsgs, oldmsgs));
   return res;
}

ast_app_messagecount()

int ast_app_messagecount	(	const char *	mailbox_id,
		const char *	folder
	)

Get the number of messages in a given mailbox folder.

Parameters

[in]	mailbox_id	Mailbox name
[in]	folder	The folder to look in. Default is INBOX if not provided.

Note

If requesting INBOX then the returned count is INBOX + Urgent.

Returns

The number of messages in the mailbox folder (zero or more).

Definition at line 718 of file main/app.c.

References messagecount(), and VM_API_CALL.

Referenced by acf_vmcount_exec(), and AST_TEST_DEFINE().

{
   int res = 0;

   VM_API_CALL(res, messagecount, (mailbox_id, folder));
   return res;
}

ast_app_options2str64()

void ast_app_options2str64	(	const struct ast_app_option *	options,
		struct ast_flags64 *	flags,
		char *	buf,
		size_t	len
	)

Given a list of options array, return an option string based on passed flags.

Parameters

options	The array of possible options declared with AST_APP_OPTIONS
flags	The flags of the options that you wish to populate the buffer with
buf	The buffer to fill with the string of options
len	The maximum length of buf

Definition at line 2916 of file main/app.c.

References ast_test_flag64, and len().

{
   unsigned int i, found = 0;
   for (i = 32; i < 128 && found < len; i++) {
      if (ast_test_flag64(flags, options[i].flag)) {
         buf[found++] = i;
      }
   }
   buf[found] = '\0';
}

ast_app_parse_options()

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.

Parameters

options	The array of possible options declared with AST_APP_OPTIONS
flags	The flag structure to have option flags set
args	The array of argument pointers to hold arguments found
optstr	The string containing the options to be parsed

Returns

zero for success, non-zero if an error occurs

See also

AST_APP_OPTIONS

Definition at line 2906 of file main/app.c.

References parse_options().

Referenced by agent_login_exec(), app_exec(), ast_multicast_rtp_create_options(), AST_TEST_DEFINE(), audiosocket_request(), auth_exec(), bridge_exec(), bridgewait_exec(), cdr_read_callback(), cdr_write_callback(), chanspy_exec(), connectedline_write(), controlplayback_exec(), detect_write(), directory_exec(), disa_exec(), dundi_query_read(), dundifunc_read(), extenspy_exec(), forkcdr_exec(), handle_options(), hint_read(), manager_mixmonitor(), minivm_accmess_exec(), minivm_greet_exec(), minivm_record_exec(), mixmonitor_exec(), page_exec(), park_app_parse_data(), pbx_builtin_background(), pbx_builtin_waitexten(), pickupchan_exec(), queue_exec(), read_exec(), read_mf_exec(), read_sf_exec(), readexten_exec(), realtime_common(), receivefax_exec(), record_exec(), redirecting_write(), resetcdr_exec(), sendfax_exec(), sendurl_exec(), sla_trunk_exec(), smdi_msg_retrieve_read(), sms_exec(), softhangup_exec(), speech_background(), start_monitor_exec(), unicast_rtp_request(), vm_exec(), vm_execmain(), volume_write(), and wait_exec().

{
   return parse_options(options, flags, args, optstr, 32);
}

ast_app_parse_options64()

int ast_app_parse_options64	(	const struct ast_app_option *	options,
		struct ast_flags64 *	flags,
		char **	args,
		char *	optstr
	)

Parses a string containing application options and sets flags/arguments.

Parameters

options	The array of possible options declared with AST_APP_OPTIONS
flags	The 64-bit flag structure to have option flags set
args	The array of argument pointers to hold arguments found
optstr	The string containing the options to be parsed

Returns

zero for success, non-zero if an error occurs

See also

AST_APP_OPTIONS

Definition at line 2911 of file main/app.c.

References parse_options().

Referenced by AST_TEST_DEFINE(), conf_exec(), dial_exec_full(), find_conf_realtime(), originate_exec(), and sayunixtime_exec().

{
   return parse_options(options, flags, args, optstr, 64);
}

ast_app_parse_timelen()

int ast_app_parse_timelen	(	const char *	timestr,
		int *	result,
		enum ast_timelen	defunit
	)

Common routine to parse time lengths, with optional time unit specifier.

Parameters

[in]	timestr	String to parse
[in]	defunit	Default unit type
[out]	result	Resulting value, specified in milliseconds

Return values

0	Success
-1	Failure

Since

1.8

Definition at line 3113 of file main/app.c.

References FMT, TIMELEN_HOURS, TIMELEN_MILLISECONDS, TIMELEN_MINUTES, and TIMELEN_SECONDS.

Referenced by ast_eivr_senddtmf(), ast_parse_arg(), new_realtime_sqlite3_db(), pbx_builtin_wait(), pbx_builtin_waitdigit(), pbx_builtin_waitexten(), senddtmf_exec(), and sendmf_exec().

{
   int res;
   char u[10];
#ifdef HAVE_LONG_DOUBLE_WIDER
   long double amount;
   #define FMT "%30Lf%9s"
#else
   double amount;
   #define FMT "%30lf%9s"
#endif
   if (!timestr) {
      return -1;
   }

   res = sscanf(timestr, FMT, &amount, u);

   if (res == 0 || res == EOF) {
#undef FMT
      return -1;
   } else if (res == 2) {
      switch (u[0]) {
      case 'h':
      case 'H':
         unit = TIMELEN_HOURS;
         if (u[1] != '\0') {
            return -1;
         }
         break;
      case 's':
      case 'S':
         unit = TIMELEN_SECONDS;
         if (u[1] != '\0') {
            return -1;
         }
         break;
      case 'm':
      case 'M':
         if (toupper(u[1]) == 'S') {
            unit = TIMELEN_MILLISECONDS;
            if (u[2] != '\0') {
               return -1;
            }
         } else if (u[1] == '\0') {
            unit = TIMELEN_MINUTES;
         } else {
            return -1;
         }
         break;
      default:
         return -1;
      }
   }

   switch (unit) {
   case TIMELEN_HOURS:
      amount *= 60;
      /* fall-through */
   case TIMELEN_MINUTES:
      amount *= 60;
      /* fall-through */
   case TIMELEN_SECONDS:
      amount *= 1000;
      /* fall-through */
   case TIMELEN_MILLISECONDS:
      ;
   }
   *result = amount > INT_MAX ? INT_MAX : (int) amount;
   return 0;
}

ast_app_run_macro()

int ast_app_run_macro	(	struct ast_channel *	autoservice_chan,
		struct ast_channel *	macro_chan,
		const char *	macro_name,
		const char *	macro_args
	)

Run a macro on a channel, placing an optional second channel into autoservice.

Since

1.8

This is a shorthand method that makes it very easy to run a macro on any given channel. It is perfectly reasonable to supply a NULL autoservice_chan here in case there is no channel to place into autoservice.

Note

Absolutely NO channel locks should be held before calling this function.

Parameters

autoservice_chan	A channel to place into autoservice while the macro is run
macro_chan	Channel to execute macro on.
macro_name	The name of the macro to run.
macro_args	The arguments to pass to the macro.

Return values

0	success
-1	on error

Definition at line 320 of file main/app.c.

References ast_app_exec_macro(), ast_free, ast_malloc, and ast_strlen_zero.

Referenced by ast_channel_connected_line_macro(), and ast_channel_redirecting_macro().

{
   int res;
   char *args_str;
   size_t args_len;

   if (ast_strlen_zero(macro_args)) {
      return ast_app_exec_macro(autoservice_chan, macro_chan, macro_name);
   }

   /* Create the Macro application argument string. */
   args_len = strlen(macro_name) + strlen(macro_args) + 2;
   args_str = ast_malloc(args_len);
   if (!args_str) {
      return -1;
   }
   snprintf(args_str, args_len, "%s,%s", macro_name, macro_args);

   res = ast_app_exec_macro(autoservice_chan, macro_chan, args_str);
   ast_free(args_str);
   return res;
}

ast_app_run_sub()

int ast_app_run_sub	(	struct ast_channel *	autoservice_chan,
		struct ast_channel *	sub_chan,
		const char *	sub_location,
		const char *	sub_args,
		int	ignore_hangup
	)

Run a subroutine on a channel, placing an optional second channel into autoservice.

Since

11

This is a shorthand method that makes it very easy to run a subroutine on any given channel. It is perfectly reasonable to supply a NULL autoservice_chan here in case there is no channel to place into autoservice.

Note

Absolutely NO channel locks should be held before calling this function.

Parameters

autoservice_chan	A channel to place into autoservice while the subroutine is run
sub_chan	Channel to execute subroutine on.
sub_location	The location of the subroutine to run.
sub_args	The arguments to pass to the subroutine.
ignore_hangup	TRUE if a hangup does not stop execution of the routine.

Return values

0	success
-1	on error

Definition at line 401 of file main/app.c.

References AO2_GLOBAL_OBJ_STATIC(), ast_app_exec_sub(), ast_free, ast_malloc, and ast_strlen_zero.

Referenced by ast_channel_connected_line_sub(), and ast_channel_redirecting_sub().

{
   int res;
   char *args_str;
   size_t args_len;

   if (ast_strlen_zero(sub_args)) {
      return ast_app_exec_sub(autoservice_chan, sub_chan, sub_location, ignore_hangup);
   }

   /* Create the Gosub application argument string. */
   args_len = strlen(sub_location) + strlen(sub_args) + 3;
   args_str = ast_malloc(args_len);
   if (!args_str) {
      return -1;
   }
   snprintf(args_str, args_len, "%s(%s)", sub_location, sub_args);

   res = ast_app_exec_sub(autoservice_chan, sub_chan, args_str, ignore_hangup);
   ast_free(args_str);
   return res;
}

ast_app_sayname()

int ast_app_sayname	(	struct ast_channel *	chan,
		const char *	mailbox_id
	)

Play a recorded user name for the mailbox to the specified channel.

Parameters

chan	Where to play the recorded name file.
mailbox_id	The mailbox name.

Return values

0	Name played without interruption
dtmf	ASCII value of the DTMF which interrupted playback.
-1	Unable to locate mailbox or hangup occurred.

Definition at line 710 of file main/app.c.

References sayname(), and VM_GREETER_API_CALL.

Referenced by play_mailbox_owner(), and spy_sayname().

{
   int res = -1;

   VM_GREETER_API_CALL(res, sayname, (chan, mailbox_id));
   return res;
}

ast_close_fds_above_n()

void ast_close_fds_above_n

(

int

n

)

Common routine for child processes, to close all fds prior to exec(2)

Parameters

[in]

n

starting file descriptor number for closing all higher file descriptors

Since

1.6.1

Definition at line 3042 of file main/app.c.

References closefrom().

Referenced by app_exec(), asterisk_daemon(), icesencode(), launch_script(), mp3play(), NBScatplay(), safe_exec_prep(), send_waveform_to_fd(), spawn_mp3(), spawn_ras(), and vm_check_password_shell().

{
   closefrom(n + 1);
}

ast_control_streamfile()

int ast_control_streamfile	(	struct ast_channel *	chan,
		const char *	file,
		const char *	fwd,
		const char *	rev,
		const char *	stop,
		const char *	pause,
		const char *	restart,
		int	skipms,
		long *	offsetms
	)

Stream a file with fast forward, pause, reverse, restart.

Parameters

chan	Channel
file	File to play.
fwd,rev,stop,pause,restart	DTMF keys for media control
skipms	Number of milliseconds to skip for fwd/rev.
offsetms	Number of milliseconds to skip when starting the media.

Before calling this function, set this to be the number of ms to start from the beginning of the file. When the function returns, it will be the number of ms from the beginning where the playback stopped. Pass NULL if you don't care.

Return values

0	on success
Non-zero	on failure

Definition at line 1319 of file main/app.c.

References control_streamfile(), and NULL.

Referenced by controlplayback_exec(), handle_controlstreamfile(), and wait_file().

{
   return control_streamfile(chan, file, fwd, rev, stop, suspend, restart, skipms, offsetms, NULL, NULL);
}

ast_control_streamfile_lang()

int ast_control_streamfile_lang	(	struct ast_channel *	chan,
		const char *	file,
		const char *	fwd,
		const char *	rev,
		const char *	stop,
		const char *	suspend,
		const char *	restart,
		int	skipms,
		const char *	lang,
		long *	offsetms
	)

Version of ast_control_streamfile() which allows the language of the media file to be specified.

Return values

0	on success
Non-zero	on failure

Definition at line 1327 of file main/app.c.

References control_streamfile(), and NULL.

Referenced by play_on_channel().

{
   return control_streamfile(chan, file, fwd, rev, stop, suspend, restart, skipms, offsetms, lang, NULL);
}

ast_control_streamfile_w_cb()

int ast_control_streamfile_w_cb	(	struct ast_channel *	chan,
		const char *	file,
		const char *	fwd,
		const char *	rev,
		const char *	stop,
		const char *	pause,
		const char *	restart,
		int	skipms,
		long *	offsetms,
		ast_waitstream_fr_cb	cb
	)

Stream a file with fast forward, pause, reverse, restart.

Parameters

chan
file	filename
fwd,rev,stop,pause,restart,skipms,offsetms
cb	waitstream callback to invoke when fastforward or rewind occurrs.

Before calling this function, set this to be the number of ms to start from the beginning of the file. When the function returns, it will be the number of ms from the beginning where the playback stopped. Pass NULL if you don't care.

Definition at line 1305 of file main/app.c.

References control_streamfile(), and NULL.

{
   return control_streamfile(chan, file, fwd, rev, stop, suspend, restart, skipms, offsetms, NULL, cb);
}

ast_control_tone()

int ast_control_tone	(	struct ast_channel *	chan,
		const char *	tone
	)

Controls playback of a tone.

Return values

0	on success
Non-zero	on failure

Definition at line 1399 of file main/app.c.

References ast_channel_zone(), AST_FRAME_CONTROL, ast_get_indication_tone(), ast_get_indication_zone(), ast_playtones_start(), ast_read_noaudio(), ast_strdupa, ast_strlen_zero, ast_tone_zone_sound_unref(), ast_tone_zone_unref(), ast_waitfor(), control_tone_frame_response(), CONTROL_TONE_RESPONSE_FAILED, CONTROL_TONE_RESPONSE_FINISHED, ast_tone_zone_sound::data, ast_frame::frametype, NULL, and parse_tone_uri().

Referenced by play_on_channel().

{
   struct ast_tone_zone *zone = NULL;
   struct ast_tone_zone_sound *ts;
   int paused = 0;
   int res = 0;

   const char *tone_indication = NULL;
   const char *tone_zone = NULL;
   char *tone_uri_parser;

   if (ast_strlen_zero(tone)) {
      return -1;
   }

   tone_uri_parser = ast_strdupa(tone);

   if (parse_tone_uri(tone_uri_parser, &tone_indication, &tone_zone)) {
      return -1;
   }

   if (tone_zone) {
      zone = ast_get_indication_zone(tone_zone);
   }

   ts = ast_get_indication_tone(zone ? zone : ast_channel_zone(chan), tone_indication);

   if (ast_playtones_start(chan, 0, ts ? ts->data : tone_indication, 0)) {
      res = -1;
   }

   while (!res) {
      struct ast_frame *fr;

      if (ast_waitfor(chan, -1) < 0) {
         res = -1;
         break;
      }

      fr = ast_read_noaudio(chan);

      if (!fr) {
         res = -1;
         break;
      }

      if (fr->frametype != AST_FRAME_CONTROL) {
         continue;
      }

      res = control_tone_frame_response(chan, fr, ts, tone_indication, &paused);
      if (res == CONTROL_TONE_RESPONSE_FINISHED) {
         res = 0;
         break;
      } else if (res == CONTROL_TONE_RESPONSE_FAILED) {
         res = -1;
         break;
      }
   }

   if (ts) {
      ast_tone_zone_sound_unref(ts);
   }

   if (zone) {
      ast_tone_zone_unref(zone);
   }

   return res;
}

ast_dtmf_stream()

int ast_dtmf_stream	(	struct ast_channel *	chan,
		struct ast_channel *	peer,
		const char *	digits,
		int	between,
		unsigned int	duration
	)

Send a string of DTMF digits to a channel.

Parameters

chan	The channel that will receive the DTMF frames
peer	(optional) Peer channel that will be autoserviced while the primary channel is receiving DTMF
digits	This is a string of characters representing the DTMF digits to be sent to the channel. Valid characters are "0123456789*#abcdABCD". Note: You can pass arguments 'f' or 'F', if you want to Flash the channel (if supported by the channel), or 'w' to add a 500 millisecond pause to the DTMF sequence.
between	This is the number of milliseconds to wait in between each DTMF digit. If zero milliseconds is specified, then the default value of 100 will be used.
duration	This is the duration that each DTMF digit should have.

Precondition

This must only be called by the channel's media handler thread.

Return values

0	on success.
-1	on failure or a channel hung up.

Definition at line 981 of file main/app.c.

References ast_autoservice_start(), ast_autoservice_stop(), and dtmf_stream().

Referenced by ast_eivr_senddtmf(), bridge_channel_dtmf_stream(), dial_exec_full(), dtmf_no_bridge(), handle_cli_misdn_send_digit(), senddtmf_exec(), testclient_exec(), testserver_exec(), and wait_for_answer().

{
   int res;

   if (peer && ast_autoservice_start(peer)) {
      return -1;
   }
   res = dtmf_stream(chan, digits, between, duration, 0);
   if (peer && ast_autoservice_stop(peer)) {
      res = -1;
   }

   return res;
}

ast_dtmf_stream_external()

void ast_dtmf_stream_external	(	struct ast_channel *	chan,
		const char *	digits,
		int	between,
		unsigned int	duration
	)

Send a string of DTMF digits to a channel from an external thread.

Parameters

chan	The channel that will receive the DTMF frames
digits	This is a string of characters representing the DTMF digits to be sent to the channel. Valid characters are "0123456789*#abcdABCD". Note: You can pass arguments 'f' or 'F', if you want to Flash the channel (if supported by the channel), or 'w' to add a 500 millisecond pause to the DTMF sequence.
between	This is the number of milliseconds to wait in between each DTMF digit. If zero milliseconds is specified, then the default value of 100 will be used.
duration	This is the duration that each DTMF digit should have.

Precondition

This must only be called by threads that are not the channel's media handler thread.

Returns

Nothing

Definition at line 996 of file main/app.c.

References dtmf_stream().

Referenced by dtmf_in_bridge().

{
   dtmf_stream(chan, digits, between, duration, 1);
}

ast_get_encoded_char()

int ast_get_encoded_char	(	const char *	stream,
		char *	result,
		size_t *	consumed
	)

Decode an encoded control or extended ASCII character.

Parameters

[in]	stream	String to decode
[out]	result	Decoded character
[out]	consumed	Number of characters used in stream to encode the character

Return values

-1	Stream is of zero length
0	Success

Definition at line 2927 of file main/app.c.

References ast_debug, ast_log, ast_strlen_zero, and LOG_ERROR.

Referenced by ast_get_encoded_str(), ast_str_get_encoded_str(), cut_internal(), filter(), function_fieldnum_helper(), function_fieldqty_helper(), replace(), shift_pop(), and unshift_push().

{
   int i;
   *consumed = 1;
   *result = 0;
   if (ast_strlen_zero(stream)) {
      *consumed = 0;
      return -1;
   }

   if (*stream == '\\') {
      *consumed = 2;
      switch (*(stream + 1)) {
      case 'n':
         *result = '\n';
         break;
      case 'r':
         *result = '\r';
         break;
      case 't':
         *result = '\t';
         break;
      case 'x':
         /* Hexadecimal */
         if (strchr("0123456789ABCDEFabcdef", *(stream + 2)) && *(stream + 2) != '\0') {
            *consumed = 3;
            if (*(stream + 2) <= '9') {
               *result = *(stream + 2) - '0';
            } else if (*(stream + 2) <= 'F') {
               *result = *(stream + 2) - 'A' + 10;
            } else {
               *result = *(stream + 2) - 'a' + 10;
            }
         } else {
            ast_log(LOG_ERROR, "Illegal character '%c' in hexadecimal string\n", *(stream + 2));
            return -1;
         }

         if (strchr("0123456789ABCDEFabcdef", *(stream + 3)) && *(stream + 3) != '\0') {
            *consumed = 4;
            *result <<= 4;
            if (*(stream + 3) <= '9') {
               *result += *(stream + 3) - '0';
            } else if (*(stream + 3) <= 'F') {
               *result += *(stream + 3) - 'A' + 10;
            } else {
               *result += *(stream + 3) - 'a' + 10;
            }
         }
         break;
      case '0':
         /* Octal */
         *consumed = 2;
         for (i = 2; ; i++) {
            if (strchr("01234567", *(stream + i)) && *(stream + i) != '\0') {
               (*consumed)++;
               ast_debug(5, "result was %d, ", *result);
               *result <<= 3;
               *result += *(stream + i) - '0';
               ast_debug(5, "is now %d\n", *result);
            } else {
               break;
            }
         }
         break;
      default:
         *result = *(stream + 1);
      }
   } else {
      *result = *stream;
      *consumed = 1;
   }
   return 0;
}

ast_get_encoded_str()

char* ast_get_encoded_str	(	const char *	stream,
		char *	result,
		size_t	result_len
	)

Decode a stream of encoded control or extended ASCII characters.

Parameters

[in]	stream	Encoded string
[out]	result	Decoded string
[in]	result_len	Maximum size of the result buffer

Returns

A pointer to the result string

Definition at line 3002 of file main/app.c.

References ast_get_encoded_char(), and result.

Referenced by chan_pjsip_new(), listfilter(), mgcp_new(), replace(), sip_addheader(), and sip_new().

{
   char *cur = result;
   size_t consumed;

   while (cur < result + result_size - 1 && !ast_get_encoded_char(stream, cur, &consumed)) {
      cur++;
      stream += consumed;
   }
   *cur = '\0';
   return result;
}

ast_install_stack_functions()

void ast_install_stack_functions

(

const struct ast_app_stack_funcs *

funcs

)

Set stack application function callbacks.

Since

11

Parameters

funcs

Stack applications callback functions.

Definition at line 346 of file main/app.c.

Referenced by load_module(), and unload_module().

{
   app_stack_callbacks = funcs;
}

ast_install_vm_test_functions()

void ast_install_vm_test_functions	(	ast_vm_test_create_user_fn *	vm_test_create_user_func,
		ast_vm_test_destroy_user_fn *	vm_test_destroy_user_func
	)

Definition at line 599 of file main/app.c.

References ast_vm_test_create_user_func, and ast_vm_test_destroy_user_func.

Referenced by load_module().

{
   ast_vm_test_create_user_func = vm_test_create_user_func;
   ast_vm_test_destroy_user_func = vm_test_destroy_user_func;
}

ast_ivr_menu_run()

int ast_ivr_menu_run	(	struct ast_channel *	c,
		struct ast_ivr_menu *	menu,
		void *	cbdata
	)

Runs an IVR menu.

Returns

returns 0 on successful completion, -1 on hangup, or -2 on user error in menu

Definition at line 2785 of file main/app.c.

References ast_ivr_menu_run_internal().

Referenced by skel_exec().

{
   int res = ast_ivr_menu_run_internal(chan, menu, cbdata);
   /* Hide internal coding */
   return res > 0 ? 0 : res;
}

ast_ivr_menu_run_internal()

static int ast_ivr_menu_run_internal ( struct ast_channel *  chan, struct ast_ivr_menu *  menu, void *  cbdata  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2700 of file main/app.c.

References ast_debug, AST_DIGIT_ANY, ast_log, AST_MAX_EXTENSION, exten, ivr_dispatch(), LOG_WARNING, maxretries, ast_ivr_option::option, option_exists(), ast_ivr_menu::options, read_newoption(), RES_EXIT, RES_REPEAT, RES_RESTART, RES_UPONE, and ast_ivr_menu::title.

Referenced by ast_ivr_menu_run(), and ivr_dispatch().

{
   /* Execute an IVR menu structure */
   int res = 0;
   int pos = 0;
   int retries = 0;
   char exten[AST_MAX_EXTENSION] = "s";
   if (option_exists(menu, "s") < 0) {
      strcpy(exten, "g");
      if (option_exists(menu, "g") < 0) {
         ast_log(LOG_WARNING, "No 's' nor 'g' extension in menu '%s'!\n", menu->title);
         return -1;
      }
   }
   while (!res) {
      while (menu->options[pos].option) {
         if (!strcasecmp(menu->options[pos].option, exten)) {
            res = ivr_dispatch(chan, menu->options + pos, exten, cbdata);
            ast_debug(1, "IVR Dispatch of '%s' (pos %d) yields %d\n", exten, pos, res);
            if (res < 0) {
               break;
            } else if (res & RES_UPONE) {
               return 0;
            } else if (res & RES_EXIT) {
               return res;
            } else if (res & RES_REPEAT) {
               int maxretries = res & 0xffff;
               if ((res & RES_RESTART) == RES_RESTART) {
                  retries = 0;
               } else {
                  retries++;
               }
               if (!maxretries) {
                  maxretries = 3;
               }
               if ((maxretries > 0) && (retries >= maxretries)) {
                  ast_debug(1, "Max retries %d exceeded\n", maxretries);
                  return -2;
               } else {
                  if (option_exists(menu, "g") > -1) {
                     strcpy(exten, "g");
                  } else if (option_exists(menu, "s") > -1) {
                     strcpy(exten, "s");
                  }
               }
               pos = 0;
               continue;
            } else if (res && strchr(AST_DIGIT_ANY, res)) {
               ast_debug(1, "Got start of extension, %c\n", res);
               exten[1] = '\0';
               exten[0] = res;
               if ((res = read_newoption(chan, menu, exten, sizeof(exten)))) {
                  break;
               }
               if (option_exists(menu, exten) < 0) {
                  if (option_exists(menu, "i")) {
                     ast_debug(1, "Invalid extension entered, going to 'i'!\n");
                     strcpy(exten, "i");
                     pos = 0;
                     continue;
                  } else {
                     ast_debug(1, "Aborting on invalid entry, with no 'i' option!\n");
                     res = -2;
                     break;
                  }
               } else {
                  ast_debug(1, "New existing extension: %s\n", exten);
                  pos = 0;
                  continue;
               }
            }
         }
         pos++;
      }
      ast_debug(1, "Stopping option '%s', res is %d\n", exten, res);
      pos = 0;
      if (!strcasecmp(exten, "s")) {
         strcpy(exten, "g");
      } else {
         break;
      }
   }
   return res;
}

ast_linear_stream()

int ast_linear_stream	(	struct ast_channel *	chan,
		const char *	filename,
		int	fd,
		int	allowoverride
	)

Stream a filename (or file descriptor) as a generator.

Definition at line 1089 of file main/app.c.

References linear_state::allowoverride, ast_activate_generator(), ast_calloc, ast_config_AST_DATA_DIR, ast_copy_string(), ast_log, ast_strlen_zero, linear_state::autoclose, errno, linear_state::fd, and LOG_WARNING.

{
   struct linear_state *lin;
   char tmpf[256];
   int autoclose = 0;

   if (fd < 0) {
      if (ast_strlen_zero(filename)) {
         return -1;
      }

      autoclose = 1;

      if (filename[0] == '/') {
         ast_copy_string(tmpf, filename, sizeof(tmpf));
      } else {
         snprintf(tmpf, sizeof(tmpf), "%s/%s/%s", ast_config_AST_DATA_DIR, "sounds", filename);
      }

      fd = open(tmpf, O_RDONLY);
      if (fd < 0) {
         ast_log(LOG_WARNING, "Unable to open file '%s': %s\n", tmpf, strerror(errno));
         return -1;
      }
   }

   lin = ast_calloc(1, sizeof(*lin));
   if (!lin) {
      if (autoclose) {
         close(fd);
      }

      return -1;
   }

   lin->fd = fd;
   lin->allowoverride = allowoverride;
   lin->autoclose = autoclose;

   return ast_activate_generator(chan, &linearstream, lin);
}

ast_lock_path()

enum AST_LOCK_RESULT ast_lock_path

(

const char *

path

)

Lock a filesystem path.

Parameters

path	the path to be locked

Returns

one of AST_LOCK_RESULT values

Definition at line 2457 of file main/app.c.

References AST_LOCK_FAILURE, ast_lock_path_flock(), ast_lock_path_lockfile(), AST_LOCK_TYPE_FLOCK, and AST_LOCK_TYPE_LOCKFILE.

Referenced by ast_module_reload(), test_vm_api_create_voicemail_files(), and vm_lock_path().

{
   enum AST_LOCK_RESULT r = AST_LOCK_FAILURE;

   switch (ast_lock_type) {
   case AST_LOCK_TYPE_LOCKFILE:
      r = ast_lock_path_lockfile(path);
      break;
   case AST_LOCK_TYPE_FLOCK:
      r = ast_lock_path_flock(path);
      break;
   }

   return r;
}

ast_lock_path_flock()

static enum AST_LOCK_RESULT ast_lock_path_flock ( const char *  path )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2337 of file main/app.c.

References ast_alloca, ast_calloc, ast_debug, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_UNLOCK, AST_LOCK_FAILURE, AST_LOCK_PATH_NOT_FOUND, AST_LOCK_SUCCESS, AST_LOCK_TIMEOUT, ast_log, ast_strdup, errno, path_lock::fd, LOG_WARNING, NULL, path_lock::path, and path_lock_destroy().

Referenced by ast_lock_path().

{
   char *fs;
   int res;
   int fd;
   time_t start;
   struct path_lock *pl;
   struct stat st, ost;

   fs = ast_alloca(strlen(path) + 20);

   snprintf(fs, strlen(path) + 19, "%s/lock", path);
   if (lstat(fs, &st) == 0) {
      if ((st.st_mode & S_IFMT) == S_IFLNK) {
         ast_log(LOG_WARNING, "Unable to create lock file "
               "'%s': it's already a symbolic link\n",
               fs);
         return AST_LOCK_FAILURE;
      }
      if (st.st_nlink > 1) {
         ast_log(LOG_WARNING, "Unable to create lock file "
               "'%s': %u hard links exist\n",
               fs, (unsigned int) st.st_nlink);
         return AST_LOCK_FAILURE;
      }
   }
   if ((fd = open(fs, O_WRONLY | O_CREAT, 0600)) < 0) {
      ast_log(LOG_WARNING, "Unable to create lock file '%s': %s\n",
            fs, strerror(errno));
      return AST_LOCK_PATH_NOT_FOUND;
   }
   if (!(pl = ast_calloc(1, sizeof(*pl)))) {
      /* We don't unlink the lock file here, on the possibility that
       * someone else created it - better to leave a little mess
       * than create a big one by destroying someone else's lock
       * and causing something to be corrupted.
       */
      close(fd);
      return AST_LOCK_FAILURE;
   }
   pl->fd = fd;
   pl->path = ast_strdup(path);

   time(&start);
   while (
      #ifdef SOLARIS
      ((res = fcntl(pl->fd, F_SETLK, fcntl(pl->fd, F_GETFL) | O_NONBLOCK)) < 0) &&
      #else
      ((res = flock(pl->fd, LOCK_EX | LOCK_NB)) < 0) &&
      #endif
         (errno == EWOULDBLOCK) &&
         (time(NULL) - start < 5))
      usleep(1000);
   if (res) {
      ast_log(LOG_WARNING, "Failed to lock path '%s': %s\n",
            path, strerror(errno));
      /* No unlinking of lock done, since we tried and failed to
       * flock() it.
       */
      path_lock_destroy(pl);
      return AST_LOCK_TIMEOUT;
   }

   /* Check for the race where the file is recreated or deleted out from
    * underneath us.
    */
   if (lstat(fs, &st) != 0 && fstat(pl->fd, &ost) != 0 &&
         st.st_dev != ost.st_dev &&
         st.st_ino != ost.st_ino) {
      ast_log(LOG_WARNING, "Unable to create lock file '%s': "
            "file changed underneath us\n", fs);
      path_lock_destroy(pl);
      return AST_LOCK_FAILURE;
   }

   /* Success: file created, flocked, and is the one we started with */
   AST_LIST_LOCK(&path_lock_list);
   AST_LIST_INSERT_TAIL(&path_lock_list, pl, le);
   AST_LIST_UNLOCK(&path_lock_list);

   ast_debug(1, "Locked path '%s'\n", path);

   return AST_LOCK_SUCCESS;
}

ast_lock_path_lockfile()

static enum AST_LOCK_RESULT ast_lock_path_lockfile ( const char *  path )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2263 of file main/app.c.

References ast_alloca, ast_debug, AST_FILE_MODE, AST_LOCK_PATH_NOT_FOUND, AST_LOCK_SUCCESS, AST_LOCK_TIMEOUT, ast_log, ast_random(), errno, LOG_ERROR, LOG_WARNING, and NULL.

Referenced by ast_lock_path().

{
   char *s;
   char *fs;
   int res;
   int fd;
   int lp = strlen(path);
   time_t start;

   s = ast_alloca(lp + 10);
   fs = ast_alloca(lp + 20);

   snprintf(fs, strlen(path) + 19, "%s/.lock-%08lx", path, (unsigned long)ast_random());
   fd = open(fs, O_WRONLY | O_CREAT | O_EXCL, AST_FILE_MODE);
   if (fd < 0) {
      ast_log(LOG_ERROR, "Unable to create lock file '%s': %s\n", path, strerror(errno));
      return AST_LOCK_PATH_NOT_FOUND;
   }
   close(fd);

   snprintf(s, strlen(path) + 9, "%s/.lock", path);
   start = time(NULL);
   while (((res = link(fs, s)) < 0) && (errno == EEXIST) && (time(NULL) - start < 5)) {
      sched_yield();
   }

   unlink(fs);

   if (res) {
      ast_log(LOG_WARNING, "Failed to lock path '%s': %s\n", path, strerror(errno));
      return AST_LOCK_TIMEOUT;
   } else {
      ast_debug(1, "Locked path '%s'\n", path);
      return AST_LOCK_SUCCESS;
   }
}

ast_mf_stream()

int ast_mf_stream	(	struct ast_channel *	chan,
		struct ast_channel *	peer,
		const char *	digits,
		int	between,
		unsigned int	duration,
		unsigned int	durationkp,
		unsigned int	durationst,
		int	is_external
	)

Send a string of MF digits to a channel.

Parameters

chan	The channel that will receive the MF frames
peer	(optional) Peer channel that will be autoserviced while the primary channel is receiving MF
digits	This is a string of characters representing the MF digits to be sent to the channel. Valid characters are "0123456789*#abcdABCD". Note: You can pass arguments 'f' or 'F', if you want to Flash the channel (if supported by the channel), or 'w' or 'W' to add a wink (if supported by the channel).
between	This is the number of milliseconds to wait in between each MF digit. If zero milliseconds is specified, then the default value of 50 will be used.
duration	This is the duration that each numeric MF digit should have. Default value is 55.
durationkp	This is the duration that each KP digit should have. Default is 120.
durationst	This is the duration that each ST, STP, ST2P, or ST3P digit should have. Default is 65.
is_external	1 if called by a thread that is not the channel's media handler thread, 0 if called by the channel's media handler thread.

Return values

0	on success.
-1	on failure or a channel hung up.

Definition at line 967 of file main/app.c.

References ast_autoservice_start(), ast_autoservice_stop(), and mf_stream().

Referenced by dial_exec_full(), manager_play_mf(), sendmf_exec(), and wait_for_answer().

{
   int res;
   if (!is_external && peer && ast_autoservice_start(peer)) {
      return -1;
   }
   res = mf_stream(chan, digits, between, duration, durationkp, durationst, is_external);
   if (!is_external && peer && ast_autoservice_stop(peer)) {
      res = -1;
   }
   return res;
}

ast_play_and_prepend()

int ast_play_and_prepend	(	struct ast_channel *	chan,
		char *	playfile,
		char *	recordfile,
		int	maxtime_sec,
		char *	fmt,
		int *	duration,
		int *	sound_duration,
		int	beep,
		int	silencethreshold,
		int	maxsilence_ms
	)

Record a file based on input frm a channel. Recording is performed in 'prepend' mode which works a little differently from normal recordings This function will not play a success message due to post-recording control in the application this was added for.

Parameters

chan	the channel being recorded
playfile	Filename of sound to play before recording begins
recordfile	Filename to save the recording
maxtime_sec	Longest possible message length in seconds
fmt	string containing all formats to be recorded delimited by '|'
duration	pointer to integer for storing length of the recording
sound_duration	pointer to integer for storing length of the recording minus all silence
beep	whether to play a beep to prompt the recording
silencethreshold	tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
maxsilence_ms	length of time in milliseconds which will trigger a timeout from silence, -1 for default.

Return values

-1	failure or hangup
'S'	Recording ended from silence timeout
't'	Recording either exceeded maximum duration or the call was ended via DTMF

Definition at line 2002 of file main/app.c.

References __ast_play_and_record(), AST_RECORD_IF_EXISTS_OVERWRITE, default_acceptdtmf, default_canceldtmf, and NULL.

Referenced by vm_forwardoptions().

{
   return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, NULL, 1, default_acceptdtmf, default_canceldtmf, 1, AST_RECORD_IF_EXISTS_OVERWRITE);
}

ast_play_and_record()

int ast_play_and_record	(	struct ast_channel *	chan,
		const char *	playfile,
		const char *	recordfile,
		int	maxtime_sec,
		const char *	fmt,
		int *	duration,
		int *	sound_duration,
		int	silencethreshold,
		int	maxsilence_ms,
		const char *	path
	)

Record a file based on input from a channel. Use default accept and cancel DTMF. This function will play "auth-thankyou" upon successful recording.

Parameters

chan	the channel being recorded
playfile	Filename of sound to play before recording begins
recordfile	Filename to save the recording
maxtime_sec	Longest possible message length in seconds
fmt	string containing all formats to be recorded delimited by '|'
duration	pointer to integer for storing length of the recording
sound_duration	pointer to integer for storing length of the recording minus all silence
silencethreshold	tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
maxsilence_ms	length of time in milliseconds which will trigger a timeout from silence, -1 for default
path	Optional filesystem path to unlock

Return values

-1	failure or hangup
'S'	Recording ended from silence timeout
't'	Recording ended from the message exceeding the maximum duration
dtmfchar	Recording ended via the return value's DTMF character for either cancel or accept.

Definition at line 1997 of file main/app.c.

References __ast_play_and_record(), AST_RECORD_IF_EXISTS_OVERWRITE, default_acceptdtmf, and default_canceldtmf.

Referenced by app_exec(), ast_record_review(), conf_rec_name(), conf_run(), and setup_privacy_args().

{
   return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold, maxsilence, path, 0, default_acceptdtmf, default_canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
}

ast_play_and_record_full()

int ast_play_and_record_full	(	struct ast_channel *	chan,
		const char *	playfile,
		const char *	recordfile,
		int	maxtime_sec,
		const char *	fmt,
		int *	duration,
		int *	sound_duration,
		int	beep,
		int	silencethreshold,
		int	maxsilence_ms,
		const char *	path,
		const char *	acceptdtmf,
		const char *	canceldtmf,
		int	skip_confirmation_sound,
		enum ast_record_if_exists	if_exists
	)

Record a file based on input from a channel This function will play "auth-thankyou" upon successful recording if skip_confirmation_sound is false.

Parameters

chan	the channel being recorded
playfile	Filename of sound to play before recording begins. A beep is also played when playfile completes, before the recording begins.
recordfile	Filename to save the recording
maxtime_sec	Longest possible message length in seconds
fmt	string containing all formats to be recorded delimited by '|'
duration	pointer to integer for storing length of the recording
beep	If true, play a beep before recording begins (and doesn't play playfile)
sound_duration	pointer to integer for storing length of the recording minus all silence
silencethreshold	tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
maxsilence_ms	Length of time in milliseconds which will trigger a timeout from silence, -1 for default
path	Optional filesystem path to unlock
acceptdtmf	Character of DTMF to end and accept the recording
canceldtmf	Character of DTMF to end and cancel the recording
skip_confirmation_sound	If true, don't play auth-thankyou at end. Nice for custom recording prompts in apps.
if_exists	Action to take if recording already exists.

Return values

-1	failure or hangup
'S'	Recording ended from silence timeout
't'	Recording ended from the message exceeding the maximum duration
dtmfchar	Recording ended via the return value's DTMF character for either cancel or accept.

Definition at line 1992 of file main/app.c.

References __ast_play_and_record(), default_canceldtmf, and S_OR.

Referenced by play_record_review(), and record_file().

{
   return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, ""), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
}

ast_play_and_wait()

int ast_play_and_wait	(	struct ast_channel *	chan,
		const char *	fn
	)

Play a stream and wait for a digit, returning the digit that was pressed.

Definition at line 1470 of file main/app.c.

References ast_channel_language(), AST_DIGIT_ANY, ast_stopstream(), ast_streamfile(), ast_waitstream(), and d.

Referenced by __ast_play_and_record(), advanced_options(), ast_record_review(), ast_say_counted_adjective(), ast_say_counted_noun(), dialout(), forward_message(), get_folder(), get_folder2(), get_folder_ja(), leave_voicemail(), minivm_greet_exec(), play_message_category(), play_message_duration(), play_record_review(), valid_priv_reply(), vm_allocate_dh(), vm_authenticate(), vm_browse_messages_en(), vm_browse_messages_es(), vm_browse_messages_gr(), vm_browse_messages_he(), vm_browse_messages_it(), vm_browse_messages_ja(), vm_browse_messages_pt(), vm_browse_messages_vi(), vm_browse_messages_zh(), vm_exec(), vm_execmain(), vm_forwardoptions(), vm_instructions_en(), vm_instructions_ja(), vm_instructions_zh(), vm_intro(), vm_intro_cs(), vm_intro_de(), vm_intro_en(), vm_intro_es(), vm_intro_fr(), vm_intro_gr(), vm_intro_he(), vm_intro_is(), vm_intro_it(), vm_intro_ja(), vm_intro_multilang(), vm_intro_nl(), vm_intro_no(), vm_intro_pl(), vm_intro_pt(), vm_intro_pt_BR(), vm_intro_se(), vm_intro_vi(), vm_intro_zh(), vm_newuser_setup(), vm_options(), vm_play_folder_name(), vm_play_folder_name_gr(), vm_play_folder_name_ja(), vm_play_folder_name_pl(), vm_play_folder_name_ua(), vm_tempgreeting(), and vmauthenticate().

{
   int d = 0;

   if ((d = ast_streamfile(chan, fn, ast_channel_language(chan)))) {
      return d;
   }

   d = ast_waitstream(chan, AST_DIGIT_ANY);

   ast_stopstream(chan);

   return d;
}

ast_queue_topic()

struct stasis_topic* ast_queue_topic

(

const char *

queuename

)

Get the Stasis Message Bus API topic for queue messages for a particular queue name.

Parameters

queuename

The name for which to get the topic

Return values

The	topic structure for queue messages for a given name
NULL	if it failed to be found or allocated

Since

12

Definition at line 3189 of file main/app.c.

References stasis_topic_pool_get_topic().

Referenced by send_agent_complete().

{
   return stasis_topic_pool_get_topic(queue_topic_pool, queuename);
}

ast_queue_topic_all()

struct stasis_topic* ast_queue_topic_all

(

void

)

Get the Stasis Message Bus API topic for queue messages.

Return values

The	topic structure for queue messages
NULL	if it has not been allocated

Since

12

Definition at line 3184 of file main/app.c.

References queue_topic_all.

Referenced by load_module().

{
   return queue_topic_all;
}

ast_read_textfile()

char* ast_read_textfile

(

const char *

filename

)

Read a file into asterisk.

Definition at line 2792 of file main/app.c.

References ast_free, ast_log, ast_malloc, errno, LOG_WARNING, and NULL.

Referenced by ast_tcptls_server_start().

{
   int fd, count = 0, res;
   char *output = NULL;
   struct stat filesize;

   if (stat(filename, &filesize) == -1) {
      ast_log(LOG_WARNING, "Error can't stat %s\n", filename);
      return NULL;
   }

   count = filesize.st_size + 1;

   if ((fd = open(filename, O_RDONLY)) < 0) {
      ast_log(LOG_WARNING, "Cannot open file '%s' for reading: %s\n", filename, strerror(errno));
      return NULL;
   }

   if ((output = ast_malloc(count))) {
      res = read(fd, output, count - 1);
      if (res == count - 1) {
         output[res] = '\0';
      } else {
         ast_log(LOG_WARNING, "Short read of %s (%d of %d): %s\n", filename, res, count - 1, strerror(errno));
         ast_free(output);
         output = NULL;
      }
   }

   close(fd);

   return output;
}

ast_record_review()

int ast_record_review	(	struct ast_channel *	chan,
		const char *	playfile,
		const char *	recordfile,
		int	maxtime,
		const char *	fmt,
		int *	duration,
		const char *	path
	)

Allow to record message and have a review option.

Definition at line 2489 of file main/app.c.

References AST_DIGIT_ANY, ast_dsp_get_threshold_from_settings(), ast_log, ast_play_and_record(), ast_play_and_wait(), ast_stream_and_wait(), ast_verb, ast_waitfordigit(), LOG_WARNING, maxsilence, NULL, silencethreshold, and THRESHOLD_SILENCE.

Referenced by conf_rec_name(), and conf_run().

{
   int silencethreshold;
   int maxsilence = 0;
   int res = 0;
   int cmd = 0;
   int max_attempts = 3;
   int attempts = 0;
   int recorded = 0;
   int message_exists = 0;
   /* Note that urgent and private are for flagging messages as such in the future */

   /* barf if no pointer passed to store duration in */
   if (!duration) {
      ast_log(LOG_WARNING, "Error ast_record_review called without duration pointer\n");
      return -1;
   }

   cmd = '3';   /* Want to start by recording */

   silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);

   while ((cmd >= 0) && (cmd != 't')) {
      switch (cmd) {
      case '1':
         if (!message_exists) {
            /* In this case, 1 is to record a message */
            cmd = '3';
            break;
         } else {
            ast_stream_and_wait(chan, "vm-msgsaved", "");
            cmd = 't';
            return res;
         }
      case '2':
         /* Review */
         ast_verb(3, "Reviewing the recording\n");
         cmd = ast_stream_and_wait(chan, recordfile, AST_DIGIT_ANY);
         break;
      case '3':
         message_exists = 0;
         /* Record */
         ast_verb(3, "R%secording\n", recorded == 1 ? "e-r" : "");
         recorded = 1;
         if ((cmd = ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, NULL, silencethreshold, maxsilence, path)) == -1) {
            /* User has hung up, no options to give */
            return cmd;
         }
         if (cmd == '0') {
            break;
         } else if (cmd == '*') {
            break;
         } else {
            /* If all is well, a message exists */
            message_exists = 1;
            cmd = 0;
         }
         break;
      case '4':
      case '5':
      case '6':
      case '7':
      case '8':
      case '9':
      case '*':
      case '#':
         cmd = ast_play_and_wait(chan, "vm-sorry");
         break;
      default:
         if (message_exists) {
            cmd = ast_play_and_wait(chan, "vm-review");
         } else {
            if (!(cmd = ast_play_and_wait(chan, "vm-torerecord"))) {
               cmd = ast_waitfordigit(chan, 600);
            }
         }

         if (!cmd) {
            cmd = ast_waitfordigit(chan, 6000);
         }
         if (!cmd) {
            attempts++;
         }
         if (attempts > max_attempts) {
            cmd = 't';
         }
      }
   }
   if (cmd == 't') {
      cmd = 0;
   }
   return cmd;
}

ast_safe_fork()

int ast_safe_fork

(

int

stop_reaper

)

Common routine to safely fork without a chance of a signal handler firing badly in the child.

Parameters

[in]

stop_reaper

flag to determine if sigchld handler is replaced or not

Since

1.6.1

Definition at line 3047 of file main/app.c.

References ast_calloc, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_UNLOCK, ast_log, ast_pthread_create_background, AST_PTHREADT_NULL, ast_replace_sigchld(), child_cap, errno, zombie::list, LOG_ERROR, LOG_WARNING, NULL, zombie::pid, shaun_of_the_dead(), and shaun_of_the_dead_thread.

Referenced by app_exec(), filestream_destructor(), icesencode(), launch_script(), mp3play(), NBScatplay(), send_waveform_to_fd(), spawn_mp3(), spawn_ras(), and vm_check_password_shell().

{
   sigset_t signal_set, old_set;
   int pid;

   /* Don't let the default signal handler for children reap our status */
   if (stop_reaper) {
      ast_replace_sigchld();
   }

   /* GCC 4.9 gives a bogus "right-hand operand of comma expression has
    * no effect" warning */
   (void) sigfillset(&signal_set);
   pthread_sigmask(SIG_BLOCK, &signal_set, &old_set);

   pid = fork();

   if (pid != 0) {
      /* Fork failed or parent */
      pthread_sigmask(SIG_SETMASK, &old_set, NULL);
      if (!stop_reaper && pid > 0) {
         struct zombie *cur = ast_calloc(1, sizeof(*cur));
         if (cur) {
            cur->pid = pid;
            AST_LIST_LOCK(&zombies);
            AST_LIST_INSERT_TAIL(&zombies, cur, list);
            AST_LIST_UNLOCK(&zombies);
            if (shaun_of_the_dead_thread == AST_PTHREADT_NULL) {
               if (ast_pthread_create_background(&shaun_of_the_dead_thread, NULL, shaun_of_the_dead, NULL)) {
                  ast_log(LOG_ERROR, "Shaun of the Dead wants to kill zombies, but can't?!!\n");
                  shaun_of_the_dead_thread = AST_PTHREADT_NULL;
               }
            }
         }
      }
      return pid;
   } else {
      /* Child */
#ifdef HAVE_CAP
      cap_set_proc(child_cap);
#endif

      /* Before we unblock our signals, return our trapped signals back to the defaults */
      signal(SIGHUP, SIG_DFL);
      signal(SIGCHLD, SIG_DFL);
      signal(SIGINT, SIG_DFL);
      signal(SIGURG, SIG_DFL);
      signal(SIGTERM, SIG_DFL);
      signal(SIGPIPE, SIG_DFL);
      signal(SIGXFSZ, SIG_DFL);

      /* unblock important signal handlers */
      if (pthread_sigmask(SIG_UNBLOCK, &signal_set, NULL)) {
         ast_log(LOG_WARNING, "unable to unblock signals: %s\n", strerror(errno));
         _exit(1);
      }

      return pid;
   }
}

ast_safe_fork_cleanup()

void ast_safe_fork_cleanup

(

void

)

Common routine to cleanup after fork'ed process is complete (if reaping was stopped)

Since

1.6.1

Definition at line 3108 of file main/app.c.

References ast_unreplace_sigchld().

Referenced by agi_exec_full(), and run_ras().

{
   ast_unreplace_sigchld();
}

ast_set_lock_type()

void ast_set_lock_type

(

enum AST_LOCK_TYPE

type

)

Set the type of locks used by ast_lock_path()

Parameters

type	the locking type to use

Definition at line 2452 of file main/app.c.

References type.

Referenced by load_asterisk_conf().

{
   ast_lock_type = type;
}

ast_str_get_encoded_str()

int ast_str_get_encoded_str	(	struct ast_str **	str,
		int	maxlen,
		const char *	stream
	)

Decode a stream of encoded control or extended ASCII characters.

Definition at line 3015 of file main/app.c.

References ast_get_encoded_char(), ast_str_buffer(), ast_str_make_space, ast_str_set(), ast_str_size(), ast_str_update(), buf, and zombie::next.

Referenced by sendtext_exec(), and system_exec_helper().

{
   char next, *buf;
   size_t offset = 0;
   size_t consumed;

   if (strchr(stream, '\\')) {
      while (!ast_get_encoded_char(stream, &next, &consumed)) {
         if (offset + 2 > ast_str_size(*str) && maxlen > -1) {
            ast_str_make_space(str, maxlen > 0 ? maxlen : (ast_str_size(*str) + 48) * 2 - 48);
         }
         if (offset + 2 > ast_str_size(*str)) {
            break;
         }
         buf = ast_str_buffer(*str);
         buf[offset++] = next;
         stream += consumed;
      }
      buf = ast_str_buffer(*str);
      buf[offset++] = '\0';
      ast_str_update(*str);
   } else {
      ast_str_set(str, maxlen, "%s", stream);
   }
   return 0;
}

AST_THREADSTORAGE_PUBLIC()

AST_THREADSTORAGE_PUBLIC

(

ast_str_thread_global_buf

)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

ast_uninstall_vm_test_functions()

void ast_uninstall_vm_test_functions

(

void

)

Definition at line 606 of file main/app.c.

References ast_vm_test_create_user_func, ast_vm_test_destroy_user_func, and NULL.

Referenced by unload_module().

{
   ast_vm_test_create_user_func = NULL;
   ast_vm_test_destroy_user_func = NULL;
}

ast_unlock_path()

int ast_unlock_path

(

const char *

path

)

Unlock a path.

Definition at line 2473 of file main/app.c.

References AST_LOCK_TYPE_FLOCK, AST_LOCK_TYPE_LOCKFILE, ast_unlock_path_flock(), and ast_unlock_path_lockfile().

Referenced by __ast_play_and_record(), access_counter_file(), ast_module_reload(), close_mailbox(), copy_message(), count_messages(), leave_voicemail(), msg_create_from_file(), open_mailbox(), resequence_mailbox(), save_to_folder(), and test_vm_api_create_voicemail_files().

{
   int r = 0;

   switch (ast_lock_type) {
   case AST_LOCK_TYPE_LOCKFILE:
      r = ast_unlock_path_lockfile(path);
      break;
   case AST_LOCK_TYPE_FLOCK:
      r = ast_unlock_path_flock(path);
      break;
   }

   return r;
}

ast_unlock_path_flock()

static int ast_unlock_path_flock ( const char *  path )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2422 of file main/app.c.

References ast_alloca, ast_debug, AST_LIST_LOCK, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_LIST_UNLOCK, path_lock::le, path_lock::path, and path_lock_destroy().

Referenced by ast_unlock_path().

{
   char *s;
   struct path_lock *p;

   s = ast_alloca(strlen(path) + 20);

   AST_LIST_LOCK(&path_lock_list);
   AST_LIST_TRAVERSE_SAFE_BEGIN(&path_lock_list, p, le) {
      if (!strcmp(p->path, path)) {
         AST_LIST_REMOVE_CURRENT(le);
         break;
      }
   }
   AST_LIST_TRAVERSE_SAFE_END;
   AST_LIST_UNLOCK(&path_lock_list);

   if (p) {
      snprintf(s, strlen(path) + 19, "%s/lock", path);
      unlink(s);
      path_lock_destroy(p);
      ast_debug(1, "Unlocked path '%s'\n", path);
   } else {
      ast_debug(1, "Failed to unlock path '%s': "
            "lock not found\n", path);
   }

   return 0;
}

ast_unlock_path_lockfile()

static int ast_unlock_path_lockfile ( const char *  path )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2300 of file main/app.c.

References ast_alloca, ast_debug, ast_log, errno, and LOG_ERROR.

Referenced by ast_unlock_path().

{
   char *s;
   int res;

   s = ast_alloca(strlen(path) + 10);

   snprintf(s, strlen(path) + 9, "%s/%s", path, ".lock");

   if ((res = unlink(s))) {
      ast_log(LOG_ERROR, "Could not unlock path '%s': %s\n", path, strerror(errno));
   } else {
      ast_debug(1, "Unlocked path '%s'\n", path);
   }

   return res;
}

ast_vm_greeter_is_registered()

int ast_vm_greeter_is_registered

(

void

)

Determine if a voicemail greeter provider is registered.

Since

13.0.0

Return values

0	if no provider registered.
1	if a provider is registered.

Definition at line 541 of file main/app.c.

References ao2_cleanup, ao2_global_obj_ref, and table.

{
   struct ast_vm_greeter_functions *table;
   int is_registered;

   table = ao2_global_obj_ref(vm_greeter_provider);
   is_registered = table ? 1 : 0;
   ao2_cleanup(table);
   return is_registered;
}

ast_vm_greeter_unregister()

void ast_vm_greeter_unregister

(

const char *

module_name

)

Unregister the specified voicemail greeter provider.

Since

13.0.0

Parameters

The	module name of the provider to unregister

Returns

Nothing

Definition at line 584 of file main/app.c.

References ao2_cleanup, ao2_global_obj_ref, ao2_global_obj_release, ast_vm_greeter_functions::module_name, and table.

Referenced by unload_module().

{
   struct ast_vm_greeter_functions *table;

   table = ao2_global_obj_ref(vm_greeter_provider);
   if (table && !strcmp(table->module_name, module_name)) {
      ao2_global_obj_release(vm_greeter_provider);
   }
   ao2_cleanup(table);
}

ast_vm_index_to_foldername()

const char* ast_vm_index_to_foldername

(

int

id

)

Return name of folder, given an id.

Parameters

[in]

id

Folder id

Returns

Name of folder

Definition at line 726 of file main/app.c.

References NULL, and VM_API_CALL.

{
   const char *res = NULL;

   VM_API_CALL(res, index_to_foldername, (id));
   return res;
}

ast_vm_is_registered()

int ast_vm_is_registered

(

void

)

Determine if a voicemail provider is registered.

Since

12.0.0

Return values

0	if no provider registered.
1	if a provider is registered.

Definition at line 430 of file main/app.c.

References ao2_cleanup, ao2_global_obj_ref, and table.

{
   struct ast_vm_functions *table;
   int is_registered;

   table = ao2_global_obj_ref(vm_provider);
   is_registered = table ? 1 : 0;
   ao2_cleanup(table);
   return is_registered;
}

ast_vm_mailbox_snapshot_create()

struct ast_vm_mailbox_snapshot* ast_vm_mailbox_snapshot_create	(	const char *	mailbox,
		const char *	context,
		const char *	folder,
		int	descending,
		enum ast_vm_snapshot_sort_val	sort_val,
		int	combine_INBOX_and_OLD
	)

Create a snapshot of a mailbox which contains information about every msg.

Parameters

mailbox,the	mailbox to look for
context,the	context to look for the mailbox in
folder,OPTIONAL.	When not NULL only msgs from the specified folder will be included.
descending,list	the msgs in descending order rather than ascending order.
combine_INBOX_and_OLD,When	this argument is set, The OLD folder will be represented in the INBOX folder of the snapshot. This allows the snapshot to represent the OLD and INBOX messages in sorted order merged together.

Return values

snapshot	on success
NULL	on failure

Definition at line 734 of file main/app.c.

References NULL, and VM_API_CALL.

Referenced by test_vm_api_remove_all_messages().

{
   struct ast_vm_mailbox_snapshot *res = NULL;

   VM_API_CALL(res, mailbox_snapshot_create, (mailbox, context, folder, descending,
      sort_val, combine_INBOX_and_OLD));
   return res;
}

ast_vm_mailbox_snapshot_destroy()

struct ast_vm_mailbox_snapshot* ast_vm_mailbox_snapshot_destroy

(

struct ast_vm_mailbox_snapshot *

mailbox_snapshot

)

destroy a snapshot

Parameters

mailbox_snapshot

The snapshot to destroy.

Return values

NULL

Definition at line 748 of file main/app.c.

References NULL, and VM_API_CALL.

Referenced by AST_TEST_DEFINE(), and test_vm_api_remove_all_messages().

{
   struct ast_vm_mailbox_snapshot *res = NULL;

   VM_API_CALL(res, mailbox_snapshot_destroy, (mailbox_snapshot));
   return res;
}

ast_vm_msg_forward()

int ast_vm_msg_forward	(	const char *	from_mailbox,
		const char *	from_context,
		const char *	from_folder,
		const char *	to_mailbox,
		const char *	to_context,
		const char *	to_folder,
		size_t	num_msgs,
		const char *	msg_ids[],
		int	delete_old
	)

forward a message from one mailbox to another.

from_mailbox The original mailbox the message is being forwarded from from_context The voicemail context of the from_mailbox from_folder The folder from which the message is being forwarded to_mailbox The mailbox to forward the message to to_context The voicemail context of the to_mailbox to_folder The folder to which the message is being forwarded num_msgs The number of messages being forwarded msg_ids The message IDs of the messages in from_mailbox to forward delete_old If non-zero, the forwarded messages are also deleted from from_mailbox. Otherwise, the messages will remain in the from_mailbox.

Return values

-1	Failure
0	Success

Definition at line 782 of file main/app.c.

References VM_API_CALL.

{
   int res = 0;

   VM_API_CALL(res, msg_forward, (from_mailbox, from_context, from_folder, to_mailbox,
      to_context, to_folder, num_msgs, msg_ids, delete_old));
   return res;
}

ast_vm_msg_move()

int ast_vm_msg_move	(	const char *	mailbox,
		const char *	context,
		size_t	num_msgs,
		const char *	oldfolder,
		const char *	old_msg_ids[],
		const char *	newfolder
	)

Move messages from one folder to another.

Parameters

mailbox	The mailbox to which the folders belong
context	The voicemail context for the mailbox
num_msgs	The number of messages to move
oldfolder	The folder from where messages should be moved
old_msg_ids	The message IDs of the messages to move
newfolder	The folder to which messages should be moved new folder. This array must be num_msgs sized.

Return values

-1	Failure
0	Success

Definition at line 756 of file main/app.c.

References VM_API_CALL.

{
   int res = 0;

   VM_API_CALL(res, msg_move, (mailbox, context, num_msgs, oldfolder, old_msg_ids,
      newfolder));
   return res;
}

ast_vm_msg_play()

int ast_vm_msg_play	(	struct ast_channel *	chan,
		const char *	mailbox,
		const char *	context,
		const char *	folder,
		const char *	msg_num,
		ast_vm_msg_play_cb *	cb
	)

Play a voicemail msg back on a channel.

Parameters

chan
mailbox	msg is in.
context	of mailbox.
folder	voicemail folder to look in.
msg_num	message number in the voicemailbox to playback to the channel.
cb

Return values

0	success
-1	failure

Definition at line 799 of file main/app.c.

References VM_API_CALL.

{
   int res = 0;

   VM_API_CALL(res, msg_play, (chan, mailbox, context, folder, msg_num, cb));
   return res;
}

ast_vm_msg_remove()

int ast_vm_msg_remove	(	const char *	mailbox,
		const char *	context,
		size_t	num_msgs,
		const char *	folder,
		const char *	msgs[]
	)

Remove/delete messages from a mailbox folder.

Parameters

mailbox	The mailbox from which to delete messages
context	The voicemail context for the mailbox
num_msgs	The number of messages to delete
folder	The folder from which to remove messages
msgs	The message IDs of the messages to delete

Return values

-1	Failure
0	Success

Definition at line 770 of file main/app.c.

References VM_API_CALL.

{
   int res = 0;

   VM_API_CALL(res, msg_remove, (mailbox, context, num_msgs, folder, msgs));
   return res;
}

ast_vm_test_create_user()

int ast_vm_test_create_user	(	const char *	context,
		const char *	mailbox
	)

Definition at line 813 of file main/app.c.

References ast_vm_test_create_user_func.

Referenced by test_vm_api_test_setup().

{
   if (ast_vm_test_create_user_func) {
      return ast_vm_test_create_user_func(context, mailbox);
   }
   return 0;
}

ast_vm_test_destroy_user()

int ast_vm_test_destroy_user	(	const char *	context,
		const char *	mailbox
	)

Definition at line 821 of file main/app.c.

References ast_vm_test_destroy_user_func.

Referenced by test_vm_api_test_teardown().

{
   if (ast_vm_test_destroy_user_func) {
      return ast_vm_test_destroy_user_func(context, mailbox);
   }
   return 0;
}

ast_vm_test_swap_table_in()

void ast_vm_test_swap_table_in

(

const struct ast_vm_functions *

vm_table

)

Swap out existing voicemail functions with a temporary set of functions for use with unit tests.

Parameters

vm_table

function table to use for testing

Note

ast_vm_test_swap_table_out should be called to restore the original set before testing concludes

Definition at line 489 of file main/app.c.

References AO2_ALLOC_OPT_LOCK_NOLOCK, ao2_alloc_options, ao2_cleanup, ao2_global_obj_ref, ao2_global_obj_replace_unref, ast_log, LOG_ERROR, NULL, provider_is_swapped, RAII_VAR, and vm_table.

{
   RAII_VAR(struct ast_vm_functions *, holding_table, NULL, ao2_cleanup);
   RAII_VAR(struct ast_vm_functions *, new_table, NULL, ao2_cleanup);

   if (provider_is_swapped) {
      ast_log(LOG_ERROR, "Attempted to swap in test function table without swapping out old test table.\n");
      return;
   }

   holding_table = ao2_global_obj_ref(vm_provider);

   if (holding_table) {
      ao2_global_obj_replace_unref(vm_provider_holder, holding_table);
   }

   new_table = ao2_alloc_options(sizeof(*new_table), NULL, AO2_ALLOC_OPT_LOCK_NOLOCK);
   if (!new_table) {
      return;
   }
   *new_table = *vm_table;

   ao2_global_obj_replace_unref(vm_provider, new_table);
   provider_is_swapped = 1;
}

ast_vm_test_swap_table_out()

void ast_vm_test_swap_table_out

(

void

)

Used after ast_vm_test_swap_table_in to restore the original set of voicemail functions.

Definition at line 515 of file main/app.c.

References ao2_cleanup, ao2_global_obj_ref, ao2_global_obj_release, ao2_global_obj_replace_unref, AO2_GLOBAL_OBJ_STATIC(), ast_log, LOG_ERROR, NULL, provider_is_swapped, and RAII_VAR.

{
   RAII_VAR(struct ast_vm_functions *, held_table, NULL, ao2_cleanup);

   if (!provider_is_swapped) {
      ast_log(LOG_ERROR, "Attempted to swap out test function table, but none is currently installed.\n");
      return;
   }

   held_table = ao2_global_obj_ref(vm_provider_holder);
   if (!held_table) {
      return;
   }

   ao2_global_obj_replace_unref(vm_provider, held_table);
   ao2_global_obj_release(vm_provider_holder);
   provider_is_swapped = 0;
}

ast_vm_unregister()

void ast_vm_unregister

(

const char *

module_name

)

Unregister the specified voicemail provider.

Parameters

The	module name of the provider to unregister

Returns

Nothing

Definition at line 473 of file main/app.c.

References ao2_cleanup, ao2_global_obj_ref, ao2_global_obj_release, AO2_GLOBAL_OBJ_STATIC(), ast_vm_functions::module_name, and table.

Referenced by unload_module().

{
   struct ast_vm_functions *table;

   table = ao2_global_obj_ref(vm_provider);
   if (table && !strcmp(table->module_name, module_name)) {
      ao2_global_obj_release(vm_provider);
   }
   ao2_cleanup(table);
}

control_streamfile()

static int control_streamfile ( struct ast_channel *  chan, const char *  file, const char *  fwd, const char *  rev, const char *  stop, const char *  suspend, const char *  restart, int  skipms, long *  offsetms, const char *  lang, ast_waitstream_fr_cb  cb  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1131 of file main/app.c.

References ast_alloca, ast_channel_language(), ast_channel_name(), ast_channel_start_silence_generator(), ast_channel_stop_silence_generator(), ast_channel_stream(), AST_CONTROL_STREAM_RESTART, AST_CONTROL_STREAM_STOP, AST_CONTROL_STREAM_SUSPEND, ast_debug, ast_opt_transmit_silence, ast_seekstream(), ast_stopstream(), ast_streamfile(), ast_tellstream(), ast_test_suite_event_notify, ast_verb, ast_waitfordigit(), ast_waitstream_fr(), ast_waitstream_fr_w_cb(), end, and NULL.

Referenced by ast_control_streamfile(), ast_control_streamfile_lang(), and ast_control_streamfile_w_cb().

{
   char *breaks = NULL;
   char *end = NULL;
   int blen = 2;
   int res;
   long pause_restart_point = 0;
   long offset = 0;
   struct ast_silence_generator *silgen = NULL;

   if (!file) {
      return -1;
   }
   if (offsetms) {
      offset = *offsetms * 8; /* XXX Assumes 8kHz */
   }
   if (lang == NULL) {
      lang = ast_channel_language(chan);
   }

   if (stop) {
      blen += strlen(stop);
   }
   if (suspend) {
      blen += strlen(suspend);
   }
   if (restart) {
      blen += strlen(restart);
   }

   if (blen > 2) {
      breaks = ast_alloca(blen + 1);
      breaks[0] = '\0';
      if (stop) {
         strcat(breaks, stop);
      }
      if (suspend) {
         strcat(breaks, suspend);
      }
      if (restart) {
         strcat(breaks, restart);
      }
   }

   if ((end = strchr(file, ':'))) {
      if (!strcasecmp(end, ":end")) {
         *end = '\0';
         end++;
      } else {
         end = NULL;
      }
   }

   for (;;) {
      ast_stopstream(chan);
      res = ast_streamfile(chan, file, lang);
      if (!res) {
         if (pause_restart_point) {
            ast_seekstream(ast_channel_stream(chan), pause_restart_point, SEEK_SET);
            pause_restart_point = 0;
         }
         else if (end || offset < 0) {
            if (offset == -8) {
               offset = 0;
            }
            ast_verb(3, "ControlPlayback seek to offset %ld from end\n", offset);

            ast_seekstream(ast_channel_stream(chan), offset, SEEK_END);
            end = NULL;
            offset = 0;
         } else if (offset) {
            ast_verb(3, "ControlPlayback seek to offset %ld\n", offset);
            ast_seekstream(ast_channel_stream(chan), offset, SEEK_SET);
            offset = 0;
         }
         if (cb) {
            res = ast_waitstream_fr_w_cb(chan, breaks, fwd, rev, skipms, cb);
         } else {
            res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
         }
      }

      if (res < 1) {
         break;
      }

      /* We go at next loop if we got the restart char */
      if ((restart && strchr(restart, res)) || res == AST_CONTROL_STREAM_RESTART) {
         ast_debug(1, "we'll restart the stream here at next loop\n");
         pause_restart_point = 0;
         ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
            "Control: %s\r\n",
            ast_channel_name(chan),
            "Restart");
         continue;
      }

      if ((suspend && strchr(suspend, res)) || res == AST_CONTROL_STREAM_SUSPEND) {
         pause_restart_point = ast_tellstream(ast_channel_stream(chan));

         if (ast_opt_transmit_silence) {
            silgen = ast_channel_start_silence_generator(chan);
         }
         ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
            "Control: %s\r\n",
            ast_channel_name(chan),
            "Pause");
         for (;;) {
            ast_stopstream(chan);
            if (!(res = ast_waitfordigit(chan, 1000))) {
               continue;
            } else if (res == -1 || (suspend && strchr(suspend, res)) || (stop && strchr(stop, res))
                  || res == AST_CONTROL_STREAM_SUSPEND || res == AST_CONTROL_STREAM_STOP) {
               break;
            }
         }
         if (silgen) {
            ast_channel_stop_silence_generator(chan, silgen);
            silgen = NULL;
         }

         if ((suspend && (res == *suspend)) || res == AST_CONTROL_STREAM_SUSPEND) {
            res = 0;
            ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
               "Control: %s\r\n",
               ast_channel_name(chan),
               "Unpause");
            continue;
         }
      }

      if (res == -1) {
         break;
      }

      /* if we get one of our stop chars, return it to the calling function */
      if ((stop && strchr(stop, res)) || res == AST_CONTROL_STREAM_STOP) {
         ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
            "Control: %s\r\n",
            ast_channel_name(chan),
            "Stop");
         break;
      }
   }

   if (pause_restart_point) {
      offset = pause_restart_point;
   } else {
      if (ast_channel_stream(chan)) {
         offset = ast_tellstream(ast_channel_stream(chan));
      } else {
         offset = -8;  /* indicate end of file */
      }
   }

   if (offsetms) {
      *offsetms = offset / 8; /* samples --> ms ... XXX Assumes 8 kHz */
   }

   ast_stopstream(chan);

   return res;
}

control_tone_frame_response()

static enum control_tone_frame_response_result control_tone_frame_response ( struct ast_channel *  chan, struct ast_frame *  fr, struct ast_tone_zone_sound *  ts, const char *  tone, int *  paused  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1340 of file main/app.c.

References AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_HANGUP, AST_CONTROL_STREAM_FORWARD, AST_CONTROL_STREAM_RESTART, AST_CONTROL_STREAM_REVERSE, AST_CONTROL_STREAM_STOP, AST_CONTROL_STREAM_SUSPEND, ast_log, ast_playtones_start(), ast_playtones_stop(), CONTROL_TONE_RESPONSE_FAILED, CONTROL_TONE_RESPONSE_FINISHED, CONTROL_TONE_RESPONSE_NORMAL, ast_tone_zone_sound::data, ast_frame_subclass::integer, LOG_NOTICE, and ast_frame::subclass.

Referenced by ast_control_tone().

{
   switch (fr->subclass.integer) {
   case AST_CONTROL_STREAM_STOP:
      ast_playtones_stop(chan);
      return CONTROL_TONE_RESPONSE_FINISHED;
   case AST_CONTROL_STREAM_SUSPEND:
      if (*paused) {
         *paused = 0;
         if (ast_playtones_start(chan, 0, ts ? ts->data : tone, 0)) {
            return CONTROL_TONE_RESPONSE_FAILED;
         }
      } else {
         *paused = 1;
         ast_playtones_stop(chan);
      }
      return CONTROL_TONE_RESPONSE_NORMAL;
   case AST_CONTROL_STREAM_RESTART:
      ast_playtones_stop(chan);
      if (ast_playtones_start(chan, 0, ts ? ts->data : tone, 0)) {
         return CONTROL_TONE_RESPONSE_FAILED;
      }
      return CONTROL_TONE_RESPONSE_NORMAL;
   case AST_CONTROL_STREAM_REVERSE:
      ast_log(LOG_NOTICE, "Media control operation 'reverse' not supported for media type 'tone'\n");
      return CONTROL_TONE_RESPONSE_NORMAL;
   case AST_CONTROL_STREAM_FORWARD:
      ast_log(LOG_NOTICE, "Media control operation 'forward' not supported for media type 'tone'\n");
      return CONTROL_TONE_RESPONSE_NORMAL;
   case AST_CONTROL_HANGUP:
   case AST_CONTROL_BUSY:
   case AST_CONTROL_CONGESTION:
      return CONTROL_TONE_RESPONSE_FINISHED;
   }

   return CONTROL_TONE_RESPONSE_NORMAL;
}

dtmf_stream()

static int dtmf_stream ( struct ast_channel *  chan, const char *  digits, int  between, unsigned int  duration, int  is_external  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 899 of file main/app.c.

References ast_channel_start_silence_generator(), ast_channel_stop_silence_generator(), AST_CONTROL_FLASH, ast_indicate(), ast_log, ast_opt_transmit_silence, ast_safe_sleep(), ast_senddigit(), ast_senddigit_external(), digit, external_sleep(), LOG_WARNING, and NULL.

Referenced by ast_dtmf_stream(), and ast_dtmf_stream_external().

{
   const char *ptr;
   int res;
   struct ast_silence_generator *silgen = NULL;
   int (*my_sleep)(struct ast_channel *chan, int ms);
   int (*my_senddigit)(struct ast_channel *chan, char digit, unsigned int duration);

   if (is_external) {
      my_sleep = external_sleep;
      my_senddigit = ast_senddigit_external;
   } else {
      my_sleep = ast_safe_sleep;
      my_senddigit = ast_senddigit;
   }

   if (!between) {
      between = 100;
   }

   /* Need a quiet time before sending digits. */
   if (ast_opt_transmit_silence) {
      silgen = ast_channel_start_silence_generator(chan);
   }
   res = my_sleep(chan, 100);
   if (res) {
      goto dtmf_stream_cleanup;
   }

   for (ptr = digits; *ptr; ptr++) {
      if (*ptr == 'w') {
         /* 'w' -- wait half a second */
         res = my_sleep(chan, 500);
         if (res) {
            break;
         }
      } else if (*ptr == 'W') {
         /* 'W' -- wait a second */
         res = my_sleep(chan, 1000);
         if (res) {
            break;
         }
      } else if (strchr("0123456789*#abcdfABCDF", *ptr)) {
         if (*ptr == 'f' || *ptr == 'F') {
            /* ignore return values if not supported by channel */
            ast_indicate(chan, AST_CONTROL_FLASH);
         } else {
            /* Character represents valid DTMF */
            my_senddigit(chan, *ptr, duration);
         }
         /* pause between digits */
         res = my_sleep(chan, between);
         if (res) {
            break;
         }
      } else {
         ast_log(LOG_WARNING, "Illegal DTMF character '%c' in string. (0-9*#aAbBcCdD allowed)\n", *ptr);
      }
   }

dtmf_stream_cleanup:
   if (silgen) {
      ast_channel_stop_silence_generator(chan, silgen);
   }

   return res;
}

external_sleep()

static int external_sleep ( struct ast_channel *  chan, int  ms  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 830 of file main/app.c.

Referenced by dtmf_stream(), and mf_stream().

{
   usleep(ms * 1000);
   return 0;
}

ivr_dispatch()

static int ivr_dispatch ( struct ast_channel *  chan, struct ast_ivr_option *  option, char *  exten, void *  cbdata  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2590 of file main/app.c.

References ast_ivr_option::action, ast_ivr_option::adata, AST_ACTION_BACKGROUND, AST_ACTION_BACKLIST, AST_ACTION_CALLBACK, AST_ACTION_EXIT, AST_ACTION_MENU, AST_ACTION_NOOP, AST_ACTION_PLAYBACK, AST_ACTION_PLAYLIST, AST_ACTION_REPEAT, AST_ACTION_RESTART, AST_ACTION_TRANSFER, AST_ACTION_UPONE, AST_ACTION_WAITOPTION, ast_channel_pbx(), AST_DIGIT_ANY, ast_ivr_menu_run_internal(), ast_log, ast_parseable_goto(), ast_stopstream(), ast_strdupa, ast_stream_and_wait(), ast_waitfordigit(), c, LOG_NOTICE, RES_EXIT, RES_REPEAT, RES_RESTART, RES_UPONE, and strsep().

Referenced by ast_ivr_menu_run_internal().

{
   int res;
   int (*ivr_func)(struct ast_channel *, void *);
   char *c;
   char *n;

   switch (option->action) {
   case AST_ACTION_UPONE:
      return RES_UPONE;
   case AST_ACTION_EXIT:
      return RES_EXIT | (((unsigned long)(option->adata)) & 0xffff);
   case AST_ACTION_REPEAT:
      return RES_REPEAT | (((unsigned long)(option->adata)) & 0xffff);
   case AST_ACTION_RESTART:
      return RES_RESTART ;
   case AST_ACTION_NOOP:
      return 0;
   case AST_ACTION_BACKGROUND:
      res = ast_stream_and_wait(chan, (char *)option->adata, AST_DIGIT_ANY);
      if (res < 0) {
         ast_log(LOG_NOTICE, "Unable to find file '%s'!\n", (char *)option->adata);
         res = 0;
      }
      return res;
   case AST_ACTION_PLAYBACK:
      res = ast_stream_and_wait(chan, (char *)option->adata, "");
      if (res < 0) {
         ast_log(LOG_NOTICE, "Unable to find file '%s'!\n", (char *)option->adata);
         res = 0;
      }
      return res;
   case AST_ACTION_MENU:
      if ((res = ast_ivr_menu_run_internal(chan, (struct ast_ivr_menu *)option->adata, cbdata)) == -2) {
         /* Do not pass entry errors back up, treat as though it was an "UPONE" */
         res = 0;
      }
      return res;
   case AST_ACTION_WAITOPTION:
      if (!(res = ast_waitfordigit(chan, ast_channel_pbx(chan) ? ast_channel_pbx(chan)->rtimeoutms : 10000))) {
         return 't';
      }
      return res;
   case AST_ACTION_CALLBACK:
      ivr_func = option->adata;
      res = ivr_func(chan, cbdata);
      return res;
   case AST_ACTION_TRANSFER:
      res = ast_parseable_goto(chan, option->adata);
      return 0;
   case AST_ACTION_PLAYLIST:
   case AST_ACTION_BACKLIST:
      res = 0;
      c = ast_strdupa(option->adata);
      while ((n = strsep(&c, ";"))) {
         if ((res = ast_stream_and_wait(chan, n,
               (option->action == AST_ACTION_BACKLIST) ? AST_DIGIT_ANY : ""))) {
            break;
         }
      }
      ast_stopstream(chan);
      return res;
   default:
      ast_log(LOG_NOTICE, "Unknown dispatch function %u, ignoring!\n", option->action);
      return 0;
   }
   return -1;
}

linear_alloc()

static void* linear_alloc ( struct ast_channel *  chan, void *  params  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1055 of file main/app.c.

References linear_state::allowoverride, ao2_bump, ao2_cleanup, ast_channel_flags(), ast_channel_name(), ast_channel_writeformat(), ast_clear_flag, AST_FLAG_WRITE_INT, ast_format_slin, ast_free, ast_log, ast_set_flag, ast_set_write_format(), LOG_WARNING, NULL, and linear_state::origwfmt.

{
   struct linear_state *ls = params;

   if (!params) {
      return NULL;
   }

   /* In this case, params is already malloc'd */
   if (ls->allowoverride) {
      ast_set_flag(ast_channel_flags(chan), AST_FLAG_WRITE_INT);
   } else {
      ast_clear_flag(ast_channel_flags(chan), AST_FLAG_WRITE_INT);
   }

   ls->origwfmt = ao2_bump(ast_channel_writeformat(chan));

   if (ast_set_write_format(chan, ast_format_slin)) {
      ast_log(LOG_WARNING, "Unable to set '%s' to linear format (write)\n", ast_channel_name(chan));
      ao2_cleanup(ls->origwfmt);
      ast_free(ls);
      ls = params = NULL;
   }

   return params;
}

linear_generator()

static int linear_generator ( struct ast_channel *  chan, void *  data, int  len, int  samples  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1025 of file main/app.c.

References ast_format_slin, AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log, ast_write(), buf, ast_frame::datalen, linear_state::fd, ast_frame_subclass::format, ast_frame::frametype, LOG_WARNING, ast_frame::samples, and ast_frame::subclass.

{
   short buf[2048 + AST_FRIENDLY_OFFSET / 2];
   struct linear_state *ls = data;
   struct ast_frame f = {
      .frametype = AST_FRAME_VOICE,
      .data.ptr = buf + AST_FRIENDLY_OFFSET / 2,
      .offset = AST_FRIENDLY_OFFSET,
   };
   int res;

   f.subclass.format = ast_format_slin;

   len = samples * 2;
   if (len > sizeof(buf) - AST_FRIENDLY_OFFSET) {
      ast_log(LOG_WARNING, "Can't generate %d bytes of data!\n" , len);
      len = sizeof(buf) - AST_FRIENDLY_OFFSET;
   }
   res = read(ls->fd, buf + AST_FRIENDLY_OFFSET/2, len);
   if (res > 0) {
      f.datalen = res;
      f.samples = res / 2;
      ast_write(chan, &f);
      if (res == len) {
         return 0;
      }
   }
   return -1;
}

linear_release()

static void linear_release ( struct ast_channel *  chan, void *  params  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1008 of file main/app.c.

References ao2_cleanup, ast_channel_name(), ast_format_get_name(), ast_free, ast_log, ast_set_write_format(), linear_state::autoclose, linear_state::fd, LOG_WARNING, and linear_state::origwfmt.

{
   struct linear_state *ls = params;

   if (ls->origwfmt && ast_set_write_format(chan, ls->origwfmt)) {
      ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%s'\n",
         ast_channel_name(chan), ast_format_get_name(ls->origwfmt));
   }
   ao2_cleanup(ls->origwfmt);

   if (ls->autoclose) {
      close(ls->fd);
   }

   ast_free(params);
}

make_silence()

static struct ast_frame* make_silence ( const struct ast_frame *  orig )

static

Construct a silence frame of the same duration as orig.

The orig frame must be AST_FORMAT_SLINEAR.

Parameters

orig	Frame as basis for silence to generate.

Returns

New frame of silence; free with ast_frfree().

NULL on error.

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1494 of file main/app.c.

References ao2_bump, ast_calloc, ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, ast_format_slin, AST_FRAME_VOICE, AST_LIST_NEXT, ast_log, AST_MALLOCD_HDR, ast_verb, ast_frame::data, ast_frame::datalen, ast_frame_subclass::format, ast_frame::frametype, LOG_WARNING, ast_frame::mallocd, NULL, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.

Referenced by __ast_play_and_record().

{
   struct ast_frame *silence;
   size_t size;
   size_t datalen;
   size_t samples = 0;

   if (!orig) {
      return NULL;
   }
   do {
      if (ast_format_cmp(orig->subclass.format, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
         ast_log(LOG_WARNING, "Attempting to silence non-slin frame\n");
         return NULL;
      }

      samples += orig->samples;

      orig = AST_LIST_NEXT(orig, frame_list);
   } while (orig);

   ast_verb(4, "Silencing %zu samples\n", samples);


   datalen = sizeof(short) * samples;
   size = sizeof(*silence) + datalen;
   silence = ast_calloc(1, size);
   if (!silence) {
      return NULL;
   }

   silence->mallocd = AST_MALLOCD_HDR;
   silence->frametype = AST_FRAME_VOICE;
   silence->data.ptr = (void *)(silence + 1);
   silence->samples = samples;
   silence->datalen = datalen;

   silence->subclass.format = ao2_bump(ast_format_slin);

   return silence;
}

mf_stream()

static int mf_stream ( struct ast_channel *  chan, const char *  digits, int  between, unsigned int  duration, unsigned int  durationkp, unsigned int  durationst, int  is_external  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 836 of file main/app.c.

References ast_channel_start_silence_generator(), ast_channel_stop_silence_generator(), AST_CONTROL_FLASH, AST_CONTROL_WINK, ast_indicate(), ast_log, ast_opt_transmit_silence, ast_playtones_start(), ast_safe_sleep(), ast_senddigit_mf(), ast_senddigit_mf_end(), external_sleep(), LOG_WARNING, and NULL.

Referenced by ast_mf_stream().

{
   const char *ptr;
   int res;
   struct ast_silence_generator *silgen = NULL;
   int (*my_sleep)(struct ast_channel *chan, int ms);

   if (is_external) {
      my_sleep = external_sleep;
   } else {
      my_sleep = ast_safe_sleep;
   }

   if (!between) {
      between = 100;
   }

   /* Need a quiet time before sending digits. */
   if (ast_opt_transmit_silence) {
      silgen = ast_channel_start_silence_generator(chan);
   }
   res = my_sleep(chan, 100);
   if (res) {
      goto mf_stream_cleanup;
   }

   for (ptr = digits; *ptr; ptr++) {
      if (strchr("0123456789*#ABCwWfF", *ptr)) {
         if (*ptr == 'f' || *ptr == 'F') {
            /* ignore return values if not supported by channel */
            ast_indicate(chan, AST_CONTROL_FLASH);
         } else if (*ptr == 'w' || *ptr == 'W') {
            /* ignore return values if not supported by channel */
            ast_indicate(chan, AST_CONTROL_WINK);
         } else {
            /* Character represents valid MF */
            ast_senddigit_mf(chan, *ptr, duration, durationkp, durationst, is_external);
         }
         /* pause between digits */
         /* The DSP code in Asterisk does not currently properly receive repeated tones
            if no audio is sent in the middle. Simply sending audio (even 0 Hz)
            works around this limitation and guarantees the correct behavior.
            */
         ast_playtones_start(chan, 0, "0", 0);
         res = my_sleep(chan, between);
         ast_senddigit_mf_end(chan);
         if (res) {
            break;
         }
      } else {
         ast_log(LOG_WARNING, "Illegal MF character '%c' in string. (0-9*#ABCwWfF allowed)\n", *ptr);
      }
   }

mf_stream_cleanup:
   if (silgen) {
      ast_channel_stop_silence_generator(chan, silgen);
   }

   return res;
}

option_exists()

static int option_exists ( struct ast_ivr_menu *  menu, char *  option  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2659 of file main/app.c.

References ast_ivr_option::option, ast_ivr_menu::options, and ast_channel::x.

Referenced by ast_ivr_menu_run_internal().

{
   int x;
   for (x = 0; menu->options[x].option; x++) {
      if (!strcasecmp(menu->options[x].option, option)) {
         return x;
      }
   }
   return -1;
}

option_matchmore()

static int option_matchmore ( struct ast_ivr_menu *  menu, char *  option  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2670 of file main/app.c.

References ast_ivr_option::option, ast_ivr_menu::options, and ast_channel::x.

Referenced by read_newoption().

{
   int x;
   for (x = 0; menu->options[x].option; x++) {
      if ((!strncasecmp(menu->options[x].option, option, strlen(option))) &&
            (menu->options[x].option[strlen(option)])) {
         return x;
      }
   }
   return -1;
}

parse_options()

static int parse_options ( const struct ast_app_option *  options, void *  _flags, char **  args, char *  optstr, int  flaglen  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2826 of file main/app.c.

References ast_app_option::arg_index, ast_clear_flag, ast_copy_string(), AST_FLAGS_ALL, ast_log, ast_set_flag, ast_set_flag64, ast_flags::flags, ast_flags64::flags, LOG_WARNING, paren, and quote().

Referenced by ast_app_parse_options(), and ast_app_parse_options64().

{
   char *s, *arg;
   int curarg, res = 0;
   unsigned int argloc;
   struct ast_flags *flags = _flags;
   struct ast_flags64 *flags64 = _flags;

   if (flaglen == 32) {
      ast_clear_flag(flags, AST_FLAGS_ALL);
   } else {
      flags64->flags = 0;
   }

   if (!optstr) {
      return 0;
   }

   s = optstr;
   while (*s) {
      curarg = *s++ & 0x7f;   /* the array (in app.h) has 128 entries */
      argloc = options[curarg].arg_index;
      if (*s == '(') {
         int paren = 1, quote = 0;
         int parsequotes = (s[1] == '"') ? 1 : 0;

         /* Has argument */
         arg = ++s;
         for (; *s; s++) {
            if (*s == '(' && !quote) {
               paren++;
            } else if (*s == ')' && !quote) {
               /* Count parentheses, unless they're within quotes (or backslashed, below) */
               paren--;
            } else if (*s == '"' && parsequotes) {
               /* Leave embedded quotes alone, unless they are the first character */
               quote = quote ? 0 : 1;
               ast_copy_string(s, s + 1, INT_MAX);
               s--;
            } else if (*s == '\\') {
               if (!quote) {
                  /* If a backslash is found outside of quotes, remove it */
                  ast_copy_string(s, s + 1, INT_MAX);
               } else if (quote && s[1] == '"') {
                  /* Backslash for a quote character within quotes, remove the backslash */
                  ast_copy_string(s, s + 1, INT_MAX);
               } else {
                  /* Backslash within quotes, keep both characters */
                  s++;
               }
            }

            if (paren == 0) {
               break;
            }
         }
         /* This will find the closing paren we found above, or none, if the string ended before we found one. */
         if ((s = strchr(s, ')'))) {
            if (argloc) {
               args[argloc - 1] = arg;
            }
            *s++ = '\0';
         } else {
            ast_log(LOG_WARNING, "Missing closing parenthesis for argument '%c' in string '%s'\n", curarg, arg);
            res = -1;
            break;
         }
      } else if (argloc) {
         args[argloc - 1] = "";
      }
      if (flaglen == 32) {
         ast_set_flag(flags, options[curarg].flag);
      } else {
         ast_set_flag64(flags64, options[curarg].flag);
      }
   }

   return res;
}

parse_tone_uri()

static int parse_tone_uri ( char *  tone_parser, const char **  tone_indication, const char **  tone_zone  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1378 of file main/app.c.

References ast_log, ast_strlen_zero, LOG_ERROR, and strsep().

Referenced by ast_control_tone().

{
   *tone_indication = strsep(&tone_parser, ";");

   if (ast_strlen_zero(tone_parser)) {
      /* Only the indication is included */
      return 0;
   }

   if (!(strncmp(tone_parser, "tonezone=", 9))) {
      *tone_zone = tone_parser + 9;
   } else {
      ast_log(LOG_ERROR, "Unexpected Tone URI component: %s\n", tone_parser);
      return -1;
   }

   return 0;
}

path_lock_destroy()

static void path_lock_destroy ( struct path_lock *  obj )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2326 of file main/app.c.

References ast_free, path_lock::fd, and path_lock::path.

Referenced by ast_lock_path_flock(), and ast_unlock_path_flock().

{
   if (obj->fd >= 0) {
      close(obj->fd);
   }
   if (obj->path) {
      ast_free(obj->path);
   }
   ast_free(obj);
}

read_newoption()

static int read_newoption ( struct ast_channel *  chan, struct ast_ivr_menu *  menu, char *  exten, int  maxexten  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 2682 of file main/app.c.

References ast_channel_pbx(), ast_waitfordigit(), ast_pbx::dtimeoutms, and option_matchmore().

Referenced by ast_ivr_menu_run_internal().

{
   int res = 0;
   int ms;
   while (option_matchmore(menu, exten)) {
      ms = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->dtimeoutms : 5000;
      if (strlen(exten) >= maxexten - 1) {
         break;
      }
      if ((res = ast_waitfordigit(chan, ms)) < 1) {
         break;
      }
      exten[strlen(exten) + 1] = '\0';
      exten[strlen(exten)] = res;
   }
   return res > 0 ? 0 : res;
}

set_read_to_slin()

static int set_read_to_slin ( struct ast_channel *  chan, struct ast_format **  orig_format  )

static

Sets a channel's read format to AST_FORMAT_SLINEAR, recording its original format.

Parameters

	chan	Channel to modify.
[out]	orig_format	Output variable to store channel's original read format.

Returns

0 on success.

-1 on error.

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1546 of file main/app.c.

References ao2_bump, ast_channel_readformat(), ast_format_slin, and ast_set_read_format().

Referenced by __ast_play_and_record().

{
   if (!chan || !orig_format) {
      return -1;
   }
   *orig_format = ao2_bump(ast_channel_readformat(chan));
   return ast_set_read_format(chan, ast_format_slin);
}

shaun_of_the_dead()

static void* shaun_of_the_dead ( void *  data )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 94 of file main/app.c.

References ast_free, AST_LIST_EMPTY, AST_LIST_FIRST, AST_LIST_LOCK, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_LIST_UNLOCK, ast_poll, zombie::list, NULL, zombie::pid, and status.

Referenced by ast_safe_fork().

{
   struct zombie *cur;
   int status;
   for (;;) {
      if (!AST_LIST_EMPTY(&zombies)) {
         /* Don't allow cancellation while we have a lock. */
         pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, NULL);
         AST_LIST_LOCK(&zombies);
         AST_LIST_TRAVERSE_SAFE_BEGIN(&zombies, cur, list) {
            if (waitpid(cur->pid, &status, WNOHANG) != 0) {
               AST_LIST_REMOVE_CURRENT(list);
               ast_free(cur);
            }
         }
         AST_LIST_TRAVERSE_SAFE_END
         AST_LIST_UNLOCK(&zombies);
         pthread_setcancelstate(PTHREAD_CANCEL_ENABLE, NULL);
      }
      pthread_testcancel();
      /* Wait for 60 seconds, without engaging in a busy loop. */
      ast_poll(NULL, 0, AST_LIST_FIRST(&zombies) ? 5000 : 60000);
   }
   return NULL;
}

vm_greeter_warn_no_provider()

static void vm_greeter_warn_no_provider ( void  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 634 of file main/app.c.

References ast_verb, and vm_greeter_warnings.

{
   if (vm_greeter_warnings++ % 10 == 0) {
      ast_verb(3, "No voicemail greeter provider registered.\n");
   }
}

vm_warn_no_provider()

static void vm_warn_no_provider ( void  )

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 613 of file main/app.c.

References ast_verb, and vm_warnings.

{
   if (vm_warnings++ % 10 == 0) {
      ast_verb(3, "No voicemail provider registered.\n");
   }
}

Variable Documentation

app_stack_callbacks

const struct ast_app_stack_funcs* app_stack_callbacks

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 344 of file main/app.c.

Referenced by ast_app_exec_sub(), and ast_app_expand_sub_args().

ast_lock_type

enum AST_LOCK_TYPE ast_lock_type = AST_LOCK_TYPE_LOCKFILE

static

Definition at line 248 of file main/app.c.

ast_vm_test_create_user_func

ast_vm_test_create_user_fn* ast_vm_test_create_user_func = NULL

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 596 of file main/app.c.

Referenced by ast_install_vm_test_functions(), ast_uninstall_vm_test_functions(), and ast_vm_test_create_user().

ast_vm_test_destroy_user_func

ast_vm_test_destroy_user_fn* ast_vm_test_destroy_user_func = NULL

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 597 of file main/app.c.

Referenced by ast_install_vm_test_functions(), ast_uninstall_vm_test_functions(), and ast_vm_test_destroy_user().

child_cap

cap_t child_cap

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 85 of file main/app.c.

Referenced by app_cleanup(), app_init(), and ast_safe_fork().

default_acceptdtmf

const char default_acceptdtmf[] = "#"

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1989 of file main/app.c.

Referenced by ast_play_and_prepend(), and ast_play_and_record().

default_canceldtmf

const char default_canceldtmf[] = ""

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1990 of file main/app.c.

Referenced by ast_play_and_prepend(), ast_play_and_record(), and ast_play_and_record_full().

global_maxsilence

int global_maxsilence = 0

static

Definition at line 1556 of file main/app.c.

Referenced by __ast_play_and_record().

global_silence_threshold

int global_silence_threshold = 128

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1555 of file main/app.c.

Referenced by __ast_play_and_record().

groups

struct groups groups = { .first = NULL, .last = NULL, .lock = { PTHREAD_RWLOCK_INITIALIZER , NULL, {1, 0} } , }

static

Referenced by ast_ref_namedgroups().

linearstream

struct ast_generator linearstream

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 1082 of file main/app.c.

path_lock_list

struct path_lock_list path_lock_list = { .first = NULL, .last = NULL, .lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, {1, 0} } , }

static

provider_is_swapped

int provider_is_swapped = 0

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 487 of file main/app.c.

Referenced by ast_vm_test_swap_table_in(), and ast_vm_test_swap_table_out().

queue_topic_all

struct stasis_topic* queue_topic_all

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 90 of file main/app.c.

Referenced by ast_queue_topic_all().

queue_topic_pool

struct stasis_topic_pool* queue_topic_pool

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 91 of file main/app.c.

shaun_of_the_dead_thread

pthread_t shaun_of_the_dead_thread = AST_PTHREADT_NULL

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 75 of file main/app.c.

Referenced by ast_safe_fork().

vm_greeter_warnings

int vm_greeter_warnings

static

Voicemail greeter not registered warning

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 539 of file main/app.c.

Referenced by vm_greeter_warn_no_provider().

vm_warnings

int vm_warnings

static

Voicemail not registered warning

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 428 of file main/app.c.

Referenced by vm_warn_no_provider().

zombies

struct zombies zombies = { .first = NULL, .last = NULL, .lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, {1, 0} } , }

static