app.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 *
 * \brief Convenient Application Routines
 *
 * \author Mark Spencer <markster@digium.com>
 */

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#ifdef HAVE_SYS_STAT_H
#include <sys/stat.h>
#endif
#include <regex.h>          /* for regcomp(3) */
#include <sys/file.h>       /* for flock(2) */
#include <signal.h>         /* for pthread_sigmask(3) */
#include <stdlib.h>         /* for closefrom(3) */
#include <sys/types.h>
#include <sys/wait.h>       /* for waitpid(2) */
#ifndef HAVE_CLOSEFROM
#include <dirent.h>         /* for opendir(3)   */
#endif
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */

#include "asterisk/paths.h"	/* use ast_config_AST_DATA_DIR */
#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"

AST_THREADSTORAGE_PUBLIC(ast_str_thread_global_buf);

static pthread_t shaun_of_the_dead_thread = AST_PTHREADT_NULL;

struct zombie {
	pid_t pid;
	AST_LIST_ENTRY(zombie) list;
};

static AST_LIST_HEAD_STATIC(zombies, zombie);

static void *shaun_of_the_dead(void *data)
{
	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;
}


#define AST_MAX_FORMATS 10

static AST_RWLIST_HEAD_STATIC(groups, ast_group_info);

/*!
 * \brief 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.
 * \param chan struct.
 * \param context
 * \param collect
 * \param size
 * \param maxlen
 * \param timeout timeout in seconds
 *
 * \return 0 if extension does not exist, 1 if extension exists
*/
int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect, size_t size, int maxlen, int timeout)
{
	struct ast_tone_zone_sound *ts;
	int res = 0, x = 0;

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

	if (!timeout && chan->pbx) {
		timeout = chan->pbx->dtimeoutms / 1000.0;
	} else if (!timeout) {
		timeout = 5;
	}

	if ((ts = ast_get_indication_tone(chan->zone, "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(chan->caller.id.number.valid, chan->caller.id.number.str, NULL))) {
			break;
		}
	}

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

	return res;
}

/*!
 * \brief ast_app_getdata
 * \param c The channel to read from
 * \param prompt The file to stream to the channel
 * \param s The string to read in to.  Must be at least the size of your length
 * \param maxlen How many digits to read (maximum)
 * \param timeout set timeout to 0 for "standard" timeouts. Set timeout to -1 for 
 *      "ludicrous time" (essentially never times out) */
enum ast_getdata_result ast_app_getdata(struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout)
{
	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, c->language);
			if (res)
				continue;
		}
		if (ast_strlen_zero(filename)) {
			/* set timeouts for the last prompt */
			fto = c->pbx ? c->pbx->rtimeoutms : 6000;
			to = c->pbx ? c->pbx->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 = c->pbx ? c->pbx->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;
}

/* The lock type used by ast_lock_path() / ast_unlock_path() */
static enum AST_LOCK_TYPE ast_lock_type = AST_LOCK_TYPE_LOCKFILE;

int ast_app_getdata_full(struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout, int audiofd, int ctrlfd)
{
	int res, to = 2000, fto = 6000;

	if (!ast_strlen_zero(prompt)) {
		res = ast_streamfile(c, prompt, c->language);
		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;
}

int ast_app_run_macro(struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const char * const macro_name, const char * const macro_args)
{
	struct ast_app *macro_app;
	int res;
	char buf[1024];

	macro_app = pbx_findapp("Macro");
	if (!macro_app) {
		ast_log(LOG_WARNING, "Cannot run macro '%s' because the 'Macro' application in not available\n", macro_name);
		return -1;
	}
	snprintf(buf, sizeof(buf), "%s%s%s", macro_name, ast_strlen_zero(macro_args) ? "" : ",", S_OR(macro_args, ""));
	if (autoservice_chan) {
		ast_autoservice_start(autoservice_chan);
	}
	res = pbx_exec(macro_chan, macro_app, buf);
	if (autoservice_chan) {
		ast_autoservice_stop(autoservice_chan);
	}
	return res;
}

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_inboxcount2_func)(const char *mailbox, int *urgentmsgs, int *newmsgs, int *oldmsgs) = NULL;
static int (*ast_sayname_func)(struct ast_channel *chan, const char *mailbox, const char *context) = NULL;
static int (*ast_messagecount_func)(const char *context, const char *mailbox, const char *folder) = NULL;

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 (*inboxcount2_func)(const char *mailbox, int *urgentmsgs, int *newmsgs, int *oldmsgs),
			      int (*messagecount_func)(const char *context, const char *mailbox, const char *folder),
			      int (*sayname_func)(struct ast_channel *chan, const char *mailbox, const char *context))
{
	ast_has_voicemail_func = has_voicemail_func;
	ast_inboxcount_func = inboxcount_func;
	ast_inboxcount2_func = inboxcount2_func;
	ast_messagecount_func = messagecount_func;
	ast_sayname_func = sayname_func;
}

void ast_uninstall_vm_functions(void)
{
	ast_has_voicemail_func = NULL;
	ast_inboxcount_func = NULL;
	ast_inboxcount2_func = NULL;
	ast_messagecount_func = NULL;
	ast_sayname_func = NULL;
}

int ast_app_has_voicemail(const char *mailbox, const char *folder)
{
	static int warned = 0;
	if (ast_has_voicemail_func) {
		return ast_has_voicemail_func(mailbox, folder);
	}

	if (warned++ % 10 == 0) {
		ast_verb(3, "Message check requested for mailbox %s/folder %s but voicemail not loaded.\n", mailbox, folder ? folder : "INBOX");
	}
	return 0;
}


int ast_app_inboxcount(const char *mailbox, int *newmsgs, int *oldmsgs)
{
	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++ % 10 == 0) {
		ast_verb(3, "Message count requested for mailbox %s but voicemail not loaded.\n", mailbox);
	}

	return 0;
}

int ast_app_inboxcount2(const char *mailbox, int *urgentmsgs, int *newmsgs, int *oldmsgs)
{
	static int warned = 0;
	if (newmsgs) {
		*newmsgs = 0;
	}
	if (oldmsgs) {
		*oldmsgs = 0;
	}
	if (urgentmsgs) {
		*urgentmsgs = 0;
	}
	if (ast_inboxcount_func) {
		return ast_inboxcount2_func(mailbox, urgentmsgs, newmsgs, oldmsgs);
	}

	if (warned++ % 10 == 0) {
		ast_verb(3, "Message count requested for mailbox %s but voicemail not loaded.\n", mailbox);
	}

	return 0;
}

int ast_app_sayname(struct ast_channel *chan, const char *mailbox, const char *context)
{
	if (ast_sayname_func) {
		return ast_sayname_func(chan, mailbox, context);
	}
	return -1;
}

int ast_app_messagecount(const char *context, const char *mailbox, const char *folder)
{
	static int warned = 0;
	if (ast_messagecount_func) {
		return ast_messagecount_func(context, mailbox, folder);
	}

	if (!warned) {
		warned++;
		ast_verb(3, "Message count requested for mailbox %s@%s/%s but voicemail not loaded.\n", mailbox, context, folder);
	}

	return 0;
}

int ast_dtmf_stream(struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between, unsigned int duration)
{
	const char *ptr;
	int res = 0;
	struct ast_silence_generator *silgen = NULL;

	if (!between) {
		between = 100;
	}

	if (peer) {
		res = ast_autoservice_start(peer);
	}

	if (!res) {
		res = ast_waitfor(chan, 100);
	}

	/* ast_waitfor will return the number of remaining ms on success */
	if (res < 0) {
		if (peer) {
			ast_autoservice_stop(peer);
		}
		return res;
	}

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

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

	if (peer) {
		/* Stop autoservice on the peer channel, but don't overwrite any error condition
		   that has occurred previously while acting on the primary channel */
		if (ast_autoservice_stop(peer) && !res) {
			res = -1;
		}
	}

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

	return res;
}

struct linear_state {
	int fd;
	int autoclose;
	int allowoverride;
	int origwfmt;
};

static void linear_release(struct ast_channel *chan, void *params)
{
	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 '%d'\n", chan->name, ls->origwfmt);
	}

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

	ast_free(params);
}

static int linear_generator(struct ast_channel *chan, void *data, int len, int samples)
{
	short buf[2048 + AST_FRIENDLY_OFFSET / 2];
	struct linear_state *ls = data;
	struct ast_frame f = {
		.frametype = AST_FRAME_VOICE,
		.subclass.codec = AST_FORMAT_SLINEAR,
		.data.ptr = buf + AST_FRIENDLY_OFFSET / 2,
		.offset = AST_FRIENDLY_OFFSET,
	};
	int res;

	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;
}

static void *linear_alloc(struct ast_channel *chan, void *params)
{
	struct linear_state *ls = params;

	if (!params) {
		return NULL;
	}

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

	ls->origwfmt = chan->writeformat;

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

	return params;
}

static struct ast_generator linearstream =
{
	alloc: linear_alloc,
	release: linear_release,
	generate: linear_generator,
};

int ast_linear_stream(struct ast_channel *chan, const char *filename, int fd, int allowoverride)
{
	struct linear_state *lin;
	char tmpf[256];
	int res = -1;
	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);
		}
		if ((fd = open(tmpf, O_RDONLY)) < 0) {
			ast_log(LOG_WARNING, "Unable to open file '%s': %s\n", tmpf, strerror(errno));
			return -1;
		}
	}
	if ((lin = ast_calloc(1, sizeof(*lin)))) {
		lin->fd = fd;
		lin->allowoverride = allowoverride;
		lin->autoclose = autoclose;
		res = ast_activate_generator(chan, &linearstream, lin);
	}
	return res;
}

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

	if (offsetms) {
		offset = *offsetms * 8; /* XXX Assumes 8kHz */
	}

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

	if (blen > 2) {
		breaks = alloca(blen + 1);
		breaks[0] = '\0';
		if (stop) {
			strcat(breaks, stop);
		}
		if (suspend) {
			strcat(breaks, suspend);
		}
		if (restart) {
			strcat(breaks, restart);
		}
	}
	if (chan->_state != AST_STATE_UP) {
		res = ast_answer(chan);
	}

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

	for (;;) {
		ast_stopstream(chan);
		res = ast_streamfile(chan, file, chan->language);
		if (!res) {
			if (pause_restart_point) {
				ast_seekstream(chan->stream, 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(chan->stream, offset, SEEK_END);
				end = NULL;
				offset = 0;
			} else if (offset) {
				ast_verb(3, "ControlPlayback seek to offset %ld\n", offset);
				ast_seekstream(chan->stream, offset, SEEK_SET);
				offset = 0;
			}
			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)) {
			ast_debug(1, "we'll restart the stream here at next loop\n");
			pause_restart_point = 0;
			continue;
		}

		if (suspend && strchr(suspend, res)) {
			pause_restart_point = ast_tellstream(chan->stream);
			for (;;) {
				ast_stopstream(chan);
				if (!(res = ast_waitfordigit(chan, 1000))) {
					continue;
				} else if (res == -1 || strchr(suspend, res) || (stop && strchr(stop, res))) {
					break;
				}
			}
			if (res == *suspend) {
				res = 0;
				continue;
			}
		}

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

		/* if we get one of our stop chars, return it to the calling function */
		if (stop && strchr(stop, res)) {
			break;
		}
	}

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

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

	/* If we are returning a digit cast it as char */
	if (res > 0 || chan->stream) {
		res = (char)res;
	}

	ast_stopstream(chan);

	return res;
}

int ast_play_and_wait(struct ast_channel *chan, const char *fn)
{
	int d = 0;

	if ((d = ast_streamfile(chan, fn, chan->language))) {
		return d;
	}

	d = ast_waitstream(chan, AST_DIGIT_ANY);

	ast_stopstream(chan);

	return d;
}

static int global_silence_threshold = 128;
static int global_maxsilence = 0;

/*! Optionally play a sound file or a beep, then record audio and video from the channel.
 * \param chan Channel to playback to/record from.
 * \param playfile Filename of sound to play before recording begins.
 * \param recordfile Filename to record to.
 * \param maxtime Maximum length of recording (in milliseconds).
 * \param fmt Format(s) to record message in. Multiple formats may be specified by separating them with a '|'.
 * \param duration Where to store actual length of the recorded message (in milliseconds).
 * \param beep Whether to play a beep before starting to record.
 * \param silencethreshold
 * \param maxsilence Length of silence that will end a recording (in milliseconds).
 * \param path Optional filesystem path to unlock.
 * \param prepend If true, prepend the recorded audio to an existing file.
 * \param acceptdtmf DTMF digits that will end the recording.
 * \param canceldtmf DTMF digits that will cancel the recording.
 */

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 d = 0;
	char *fmts;
	char comment[256];
	int x, fmtcnt = 1, res = -1, outmsg = 0;
	struct ast_filestream *others[AST_MAX_FORMATS];
	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;
	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) {
		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, chan->name);

	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;
		}
		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, 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);
		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 (;;) {
			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", chan->name);
					res = -1;
				}
			}

			if (res < 0) {
				f = NULL;
				break;
			}
			if (!(f = ast_read(chan))) {
				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) {
					dspsilence = 0;
					ast_dsp_silence(sildet, f, &dspsilence);
					if (olddspsilence > dspsilence) {
						totalsilence += olddspsilence;
					}
					olddspsilence = dspsilence;

					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 cancelled message by pressing %c\n", f->subclass.integer);
					res = f->subclass.integer;
					outmsg = 0;
					break;
				}
			}
			if (maxtime) {
				end = time(NULL);
				if (maxtime < (end - start)) {
					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 (!prepend) {
		/* Reduce duration by a total silence amount */
		if (olddspsilence <= dspsilence) {
			totalsilence += dspsilence;
		}

		if (totalsilence > 0) {
			*duration -= (totalsilence - 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]);
		}
	}

	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] || !realfiles[x]) {
				break;
			}
			/*!\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);
			}