Mon Oct 6 06:28:55 2008

Asterisk developer's documentation

---

app.c File Reference

Convenient Application Routines. More...

#include "asterisk.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/time.h>
#include <signal.h>
#include <errno.h>
#include <unistd.h>
#include <dirent.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <regex.h>
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/file.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/indications.h"
#include "asterisk/linkedlists.h"

Include dependency graph for app.c:

Go to the source code of this file.

Data Structures
struct	linear_state
Defines
#define	MAX_OTHER_FORMATS   10
#define	RES_EXIT   (1 << 17)
#define	RES_REPEAT   (1 << 18)
#define	RES_RESTART   ((1 << 19) | RES_REPEAT)
#define	RES_UPONE   (1 << 16)
Functions
static int	__ast_play_and_record (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int beep, int silencethreshold, int maxsilence, const char *path, int prepend, const char *acceptdtmf, const char *canceldtmf)
int	ast_app_dtget (struct ast_channel *chan, const char *context, char *collect, size_t size, int maxlen, int timeout)
	Present a dialtone and collect a certain length extension.
int	ast_app_getdata (struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout)
	Plays a stream and gets DTMF data from a channel.
int	ast_app_getdata_full (struct ast_channel *c, 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.
int	ast_app_group_discard (struct ast_channel *chan)
int	ast_app_group_get_count (const char *group, const char *category)
ast_group_info *	ast_app_group_list_head (void)
int	ast_app_group_list_lock (void)
int	ast_app_group_list_unlock (void)
int	ast_app_group_match_get_count (const char *groupmatch, const char *category)
int	ast_app_group_set_channel (struct ast_channel *chan, const char *data)
int	ast_app_group_split_group (const char *data, char *group, int group_max, char *category, int category_max)
int	ast_app_group_update (struct ast_channel *old, struct ast_channel *new)
int	ast_app_has_voicemail (const char *mailbox, const char *folder)
int	ast_app_inboxcount (const char *mailbox, int *newmsgs, int *oldmsgs)
int	ast_app_messagecount (const char *context, const char *mailbox, const char *folder)
void	ast_app_options2str (const struct ast_app_option *options, struct ast_flags *flags, char *buf, size_t len)
	Given a list of options array, return an option string based on passed flags.
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.
unsigned int	ast_app_separate_args (char *buf, char delim, char **array, int arraylen)
	Separate a string into arguments in an array.
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)
int	ast_dtmf_stream (struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between)
	Send DTMF to a channel.
void	ast_install_vm_functions (int(*has_voicemail_func)(const char *mailbox, const char *folder), int(*inboxcount_func)(const char *mailbox, int *newmsgs, int *oldmsgs), int(*messagecount_func)(const char *context, const char *mailbox, const char *folder))
int	ast_ivr_menu_run (struct ast_channel *chan, struct ast_ivr_menu *menu, void *cbdata)
	Runs an IVR menu.
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)
static	AST_LIST_HEAD_STATIC (groups, ast_group_info)
enum AST_LOCK_RESULT	ast_lock_path (const char *path)
	Lock a filesystem path.
int	ast_play_and_prepend (struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int beep, int silencethreshold, int maxsilence)
int	ast_play_and_record (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path)
int	ast_play_and_record_full (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf)
int	ast_play_and_wait (struct ast_channel *chan, const char *fn)
char *	ast_read_textfile (const char *filename)
int	ast_record_review (struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, const char *path)
void	ast_uninstall_vm_functions (void)
int	ast_unlock_path (const char *path)
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 int	option_exists (struct ast_ivr_menu *menu, char *option)
static int	option_matchmore (struct ast_ivr_menu *menu, char *option)
static int	read_newoption (struct ast_channel *chan, struct ast_ivr_menu *menu, char *exten, int maxexten)
Variables
static int(*)	ast_has_voicemail_func (const char *mailbox, const char *folder) = NULL
static int(*)	ast_inboxcount_func (const char *mailbox, int *newmsgs, int *oldmsgs) = NULL
static int(*)	ast_messagecount_func (const char *context, const char *mailbox, const char *folder) = NULL
static char	default_acceptdtmf [] = "#"
static char	default_canceldtmf [] = ""
static int	global_maxsilence = 0
static int	global_silence_threshold = 128
static struct ast_generator	linearstream

---

Detailed Description

Convenient Application Routines.

Author:

Mark Spencer <markster@digium.com>

Definition in file app.c.

---

Define Documentation

#define MAX_OTHER_FORMATS   10

Definition at line 54 of file app.c.

Referenced by __ast_play_and_record().

#define RES_EXIT   (1 << 17)

Definition at line 1163 of file app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

#define RES_REPEAT   (1 << 18)

Definition at line 1164 of file app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

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

Definition at line 1165 of file app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

#define RES_UPONE   (1 << 16)

Definition at line 1162 of file app.c.

Referenced by ast_ivr_menu_run_internal(), and ivr_dispatch().

---

Function Documentation

static int __ast_play_and_record	(	struct ast_channel *	chan,
		const char *	playfile,
		const char *	recordfile,
		int	maxtime,
		const char *	fmt,
		int *	duration,
		int	beep,
		int	silencethreshold,
		int	maxsilence,
		const char *	path,
		int	prepend,
		const char *	acceptdtmf,
		const char *	canceldtmf
	)			[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 milliseconds).
	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).
	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.
	acceptdtmf	DTMF digits that will end the recording.
	canceldtmf	DTMF digits that will cancel the recording.

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.

Definition at line 504 of file app.c.

References ast_channel_start_silence_generator(), ast_channel_stop_silence_generator(), ast_closestream(), AST_CONTROL_VIDUPDATE, ast_dsp_free(), ast_dsp_new(), ast_dsp_set_threshold(), ast_dsp_silence(), ast_filedelete(), ast_filerename(), AST_FORMAT_SLINEAR, AST_FRAME_DTMF, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frfree, ast_getformatname(), ast_indicate(), ast_log(), ast_opt_transmit_silence, ast_play_and_wait(), ast_read(), ast_readfile(), ast_readframe(), ast_set_read_format(), ast_strdupa, ast_stream_and_wait(), ast_stream_rewind(), ast_tellstream(), ast_truncstream(), ast_unlock_path(), ast_verbose(), ast_waitfor(), ast_writefile(), ast_writestream(), f, LOG_DEBUG, LOG_WARNING, MAX_OTHER_FORMATS, option_debug, option_verbose, ast_channel::readformat, strsep(), ast_dsp::totalsilence, VERBOSE_PREFIX_3, and VERBOSE_PREFIX_4.

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[MAX_OTHER_FORMATS];
   char *sfmt[MAX_OTHER_FORMATS];
   char *stringp = NULL;
   time_t start, end;
   struct ast_dsp *sildet = NULL;   /* silence detector dsp */
   int totalsilence = 0;
   int rfmt = 0;
   struct ast_silence_generator *silgen = NULL;
   char prependfile[80];

   if (silencethreshold < 0)
      silencethreshold = global_silence_threshold;

   if (maxsilence < 0)
      maxsilence = global_maxsilence;

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

   if (option_debug)
      ast_log(LOG_DEBUG,"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, chan->name);

   if (playfile || beep) {
      if (!beep)
         d = ast_play_and_wait(chan, playfile);
      if (d > -1)
         d = ast_stream_and_wait(chan, "beep", chan->language, "");
      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, "|");
   if (option_debug)
      ast_log(LOG_DEBUG, "Recording Formats: sfmts=%s\n", fmts);
   sfmt[0] = ast_strdupa(fmts);

   while ((fmt = strsep(&stringp, "|"))) {
      if (fmtcnt > MAX_OTHER_FORMATS - 1) {
         ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app.c\n");
         break;
      }
      sfmt[fmtcnt++] = ast_strdupa(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, O_TRUNC, 0, 0777);
      if (option_verbose > 2)
         ast_verbose(VERBOSE_PREFIX_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);
      rfmt = chan->readformat;
      res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
      if (res < 0) {
         ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
         ast_dsp_free(sildet);
         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 forever, writing the packets we read to the writer(s), until
         we read a digit or get a hangup */
      struct ast_frame *f;
      for (;;) {
         res = ast_waitfor(chan, 2000);
         if (!res) {
            if (option_debug)
               ast_log(LOG_DEBUG, "One waitfor failed, trying another\n");
            /* Try one more time in case of masq */
            res = ast_waitfor(chan, 2000);
            if (!res) {
               ast_log(LOG_WARNING, "No audio available on %s??\n", chan->name);
               res = -1;
            }
         }

         if (res < 0) {
            f = NULL;
            break;
         }
         f = ast_read(chan);
         if (!f)
            break;
         if (f->frametype == AST_FRAME_VOICE) {
            /* write each format */
            for (x = 0; x < fmtcnt; x++) {
               if (prepend && !others[x])
                  break;
               res = ast_writestream(others[x], f);
            }

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

               if (totalsilence > maxsilence) {
                  /* Ended happily with silence */
                  if (option_verbose > 2)
                     ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/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 */
               if (option_verbose > 2) 
                  ast_verbose(VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
               res = 't';
               outmsg = 2;
               break;
            }
            if (strchr(acceptdtmf, f->subclass)) {
               if (option_verbose > 2)
                  ast_verbose(VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
               res = f->subclass;
               outmsg = 2;
               break;
            }
            if (strchr(canceldtmf, f->subclass)) {
               if (option_verbose > 2)
                  ast_verbose(VERBOSE_PREFIX_3 "User cancelled message by pressing %c\n", f->subclass);
               res = f->subclass;
               outmsg = 0;
               break;
            }
         }
         if (maxtime) {
            end = time(NULL);
            if (maxtime < (end - start)) {
               if (option_verbose > 2)
                  ast_verbose(VERBOSE_PREFIX_3 "Took too long, cutting it short...\n");
               res = 't';
               outmsg = 2;
               break;
            }
         }
         ast_frfree(f);
      }
      if (!f) {
         if (option_verbose > 2)
            ast_verbose(VERBOSE_PREFIX_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 (!prepend) {
      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 && totalsilence)
            ast_stream_rewind(others[x], totalsilence - 200);
         ast_truncstream(others[x]);
         ast_closestream(others[x]);
      }
   }

   if (prepend && outmsg) {
      struct ast_filestream *realfiles[MAX_OTHER_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] || !realfiles[x])
            break;
         /*!\note Same logic as above. */
         if (totalsilence)
            ast_stream_rewind(others[x], totalsilence - 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]);
         if (option_verbose > 3)
            ast_verbose(VERBOSE_PREFIX_4 "Recording Format: sfmts=%s, prependfile %s, recordfile %s\n", sfmt[x], prependfile, recordfile);
         ast_filedelete(prependfile, sfmt[x]);
      }
   }
   if (rfmt && ast_set_read_format(chan, rfmt)) {
      ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(rfmt), chan->name);
   }
   if (outmsg == 2) {
      ast_stream_and_wait(chan, "auth-thankyou", chan->language, "");
   }
   if (sildet)
      ast_dsp_free(sildet);
   return res;
}

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

Present a dialtone and collect a certain length extension.

Returns:

Returns 1 on valid extension entered, -1 on hangup, or 0 on invalid extension.

Note:

Note that if 'collect' holds digits already, new digits will be appended, so be sure it's initialized properly

Definition at line 65 of file app.c.

References ast_exists_extension(), ast_get_indication_tone(), ast_ignore_pattern(), ast_log(), ast_matchmore_extension(), ast_playtones_start(), ast_playtones_stop(), ast_waitfordigit(), ast_channel::cid, ast_callerid::cid_num, ind_tone_zone_sound::data, ast_pbx::dtimeout, LOG_NOTICE, ast_channel::pbx, and ast_channel::zone.

Referenced by builtin_atxfer(), and builtin_blindtransfer().

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

   if (maxlen > size)
      maxlen = size;
   
   if (!timeout && chan->pbx)
      timeout = chan->pbx->dtimeout;
   else if (!timeout)
      timeout = 5;
   
   ts = ast_get_indication_tone(chan->zone,"dial");
   if (ts && ts->data[0])
      res = ast_playtones_start(chan, 0, ts->data, 0);
   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, chan->cid.cid_num))
         break;
   }
   if (res >= 0)
      res = ast_exists_extension(chan, context, collect, 1, chan->cid.cid_num) ? 1 : 0;
   return res;
}

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

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 107 of file app.c.

References ast_readstring(), ast_streamfile(), ast_pbx::dtimeout, ast_channel::pbx, and ast_pbx::rtimeout.

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

{
   int res,to,fto;
   /* XXX Merge with full version? XXX */
   if (maxlen)
      s[0] = '\0';
   if (prompt) {
      res = ast_streamfile(c, prompt, c->language);
      if (res < 0)
         return res;
   }
   fto = c->pbx ? c->pbx->rtimeout * 1000 : 6000;
   to = c->pbx ? c->pbx->dtimeout * 1000 : 2000;

   if (timeout > 0) 
      fto = to = timeout;
   if (timeout < 0) 
      fto = to = 1000000000;
   res = ast_readstring(c, s, maxlen, to, fto, "#");
   return res;
}

int ast_app_getdata_full	(	struct ast_channel *	c,
		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 130 of file app.c.

References ast_readstring_full(), and ast_streamfile().

Referenced by handle_getdata().

{
   int res, to, fto;
   if (prompt) {
      res = ast_streamfile(c, prompt, c->language);
      if (res < 0)
         return res;
   }
   fto = 6000;
   to = 2000;
   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;
}

int ast_app_group_discard

(

struct ast_channel *

chan

)

Discard all group counting for a channel

Definition at line 927 of file app.c.

References AST_LIST_LOCK, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_LIST_UNLOCK, ast_group_info::chan, and free.

Referenced by ast_channel_free().

{
   struct ast_group_info *gi = NULL;
   
   AST_LIST_LOCK(&groups);
   AST_LIST_TRAVERSE_SAFE_BEGIN(&groups, gi, list) {
      if (gi->chan == chan) {
         AST_LIST_REMOVE_CURRENT(&groups, list);
         free(gi);
      }
   }
        AST_LIST_TRAVERSE_SAFE_END
   AST_LIST_UNLOCK(&groups);
   
   return 0;
}

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 870 of file app.c.

References AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_strlen_zero(), ast_group_info::category, and ast_group_info::group.

Referenced by group_count_function_read().

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

   if (ast_strlen_zero(group))
      return 0;
   
   AST_LIST_LOCK(&groups);
   AST_LIST_TRAVERSE(&groups, gi, list) {
      if (!strcasecmp(gi->group, group) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category))))
         count++;
   }
   AST_LIST_UNLOCK(&groups);

   return count;
}

struct ast_group_info* ast_app_group_list_head

(

void

)

Get the head of the group count list

Definition at line 949 of file app.c.

References AST_LIST_FIRST.

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

{
   return AST_LIST_FIRST(&groups);
}

int ast_app_group_list_lock

(

void

)

Lock the group count list

Definition at line 944 of file app.c.

References AST_LIST_LOCK.

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

{
   return AST_LIST_LOCK(&groups);
}

int ast_app_group_list_unlock

(

void

)

Unlock the group count list

Definition at line 954 of file app.c.

References AST_LIST_UNLOCK.

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

{
   return AST_LIST_UNLOCK(&groups);
}

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 888 of file app.c.

References AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_strlen_zero(), ast_group_info::category, and ast_group_info::group.

Referenced by group_match_count_function_read().

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

   if (ast_strlen_zero(groupmatch))
      return 0;

   /* if regex compilation fails, return zero matches */
   if (regcomp(&regexbuf, groupmatch, REG_EXTENDED | REG_NOSUB))
      return 0;

   AST_LIST_LOCK(&groups);
   AST_LIST_TRAVERSE(&groups, gi, list) {
      if (!regexec(&regexbuf, gi->group, 0, NULL, 0) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category))))
         count++;
   }
   AST_LIST_UNLOCK(&groups);

   regfree(&regexbuf);

   return count;
}

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 825 of file app.c.

References ast_app_group_split_group(), AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_LIST_UNLOCK, ast_strlen_zero(), calloc, ast_group_info::category, ast_group_info::chan, free, group, ast_group_info::group, and len.

Referenced by 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_LIST_LOCK(&groups);
   AST_LIST_TRAVERSE_SAFE_BEGIN(&groups, gi, list) {
      if ((gi->chan == chan) && ((ast_strlen_zero(category) && ast_strlen_zero(gi->category)) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category)))) {
         AST_LIST_REMOVE_CURRENT(&groups, list);
         free(gi);
         break;
      }
   }
   AST_LIST_TRAVERSE_SAFE_END

   if (ast_strlen_zero(group)) {
      /* Enable unsetting the group */
   } else if ((gi = 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_LIST_INSERT_TAIL(&groups, gi, list);
   } else {
      res = -1;
   }
   
   AST_LIST_UNLOCK(&groups);
   
   return res;
}

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 798 of file app.c.

References ast_strlen_zero().

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;
      cat = strchr(tmp, '@');
      if (cat) {
         *cat = '\0';
         cat++;
      }
   }

   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;
}

int ast_app_group_update	(	struct ast_channel *	oldchan,
		struct ast_channel *	newchan
	)

Update all group counting for a channel to a new one

Definition at line 913 of file app.c.

References AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, and ast_group_info::chan.

Referenced by ast_do_masquerade().

{
   struct ast_group_info *gi = NULL;

   AST_LIST_LOCK(&groups);
   AST_LIST_TRAVERSE(&groups, gi, list) {
      if (gi->chan == old)
         gi->chan = new;
   }
   AST_LIST_UNLOCK(&groups);

   return 0;
}

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

Determine if a given mailbox has any voicemail

Definition at line 168 of file app.c.

References ast_has_voicemail_func, ast_verbose(), option_verbose, and VERBOSE_PREFIX_3.

Referenced by action_mailboxstatus(), has_voicemail(), notify_new_message(), play_dialtone(), poll_mailbox(), and run_externnotify().

{
   static int warned = 0;
   if (ast_has_voicemail_func)
      return ast_has_voicemail_func(mailbox, folder);

   if ((option_verbose > 2) && !warned) {
      ast_verbose(VERBOSE_PREFIX_3 "Message check requested for mailbox %s/folder %s but voicemail not loaded.\n", mailbox, folder ? folder : "INBOX");
      warned++;
   }
   return 0;
}

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

Determine number of new/old messages in a mailbox

Definition at line 182 of file app.c.

References ast_inboxcount_func, ast_verbose(), option_verbose, and VERBOSE_PREFIX_3.

Referenced by action_mailboxcount(), notify_new_message(), sip_send_mwi_to_peer(), and update_registry().

{
   static int warned = 0;
   if (newmsgs)
      *newmsgs = 0;
   if (oldmsgs)
      *oldmsgs = 0;
   if (ast_inboxcount_func)
      return ast_inboxcount_func(mailbox, newmsgs, oldmsgs);

   if (!warned && (option_verbose > 2)) {
      warned++;
      ast_verbose(VERBOSE_PREFIX_3 "Message count requested for mailbox %s but voicemail not loaded.\n", mailbox);
   }

   return 0;
}

int ast_app_messagecount	(	const char *	context,
		const char *	mailbox,
		const char *	folder
	)

Determine number of messages in a given mailbox and folder

Definition at line 200 of f