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
 * 
 */

#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.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#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"

#define MAX_OTHER_FORMATS 10


/* 
This function presents a dialtone and reads an extension into 'collect' 
which must be a pointer to a **pre-initilized** 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.
*/
int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect, size_t size, int maxlen, int timeout) 
{
	struct 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); strlen(collect) < maxlen; ) {
		res = ast_waitfordigit(chan, timeout);
		if (!ast_ignore_pattern(context, collect))
			ast_playtones_stop(chan);
		if (res < 1)
			break;
		collect[x++] = res;
		if (!ast_matchmore_extension(chan, context, collect, 1, chan->cid.cid_num)) {
			if (collect[x-1] == '#') {
				/* Not a valid extension, ending in #, assume the # was to finish dialing */
				collect[x-1] = '\0';
			}
			break;
		}
	}
	if (res >= 0) {
		if (ast_exists_extension(chan, context, collect, 1, chan->cid.cid_num))
			res = 1;
		else
			res = 0;
	}
	return res;
}



/* set timeout to 0 for "standard" timeouts. Set timeout to -1 for 
   "ludicrous time" (essentially never times out) */
int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout)
{
	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)
{
	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_getvoice(struct ast_channel *c, char *dest, char *dstfmt, char *prompt, int silence, int maxsec)
{
	int res;
	struct ast_filestream *writer;
	int rfmt;
	int totalms=0, total;
	
	struct ast_frame *f;
	struct ast_dsp *sildet;
	/* Play prompt if requested */
	if (prompt) {
		res = ast_streamfile(c, prompt, c->language);
		if (res < 0)
			return res;
		res = ast_waitstream(c,"");
		if (res < 0)
			return res;
	}
	rfmt = c->readformat;
	res = ast_set_read_format(c, AST_FORMAT_SLINEAR);
	if (res < 0) {
		ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
		return -1;
	}
	sildet = ast_dsp_new();
	if (!sildet) {
		ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
		return -1;
	}
	writer = ast_writefile(dest, dstfmt, "Voice file", 0, 0, 0666);
	if (!writer) {
		ast_log(LOG_WARNING, "Unable to open file '%s' in format '%s' for writing\n", dest, dstfmt);
		ast_dsp_free(sildet);
		return -1;
	}
	for(;;) {
		if ((res = ast_waitfor(c, 2000)) < 0) {
			ast_log(LOG_NOTICE, "Waitfor failed while recording file '%s' format '%s'\n", dest, dstfmt);
			break;
		}
		if (res) {
			f = ast_read(c);
			if (!f) {
				ast_log(LOG_NOTICE, "Hungup while recording file '%s' format '%s'\n", dest, dstfmt);
				break;
			}
			if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '#')) {
				/* Ended happily with DTMF */
				ast_frfree(f);
				break;
			} else if (f->frametype == AST_FRAME_VOICE) {
				ast_dsp_silence(sildet, f, &total);