Version 0.1.2 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_oss.c

@@ -0,0 +1,791 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Use /dev/dsp as a channel, and the console to command it :).
+ *
+ * The full-duplex "simulation" is pretty weak.  This is generally a 
+ * VERY BADLY WRITTEN DRIVER so please don't use it as a model for
+ * writing a driver.
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/frame.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/module.h>
+#include <asterisk/channel_pvt.h>
+#include <asterisk/options.h>
+#include <asterisk/pbx.h>
+#include <asterisk/config.h>
+#include <asterisk/cli.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+#include <sys/time.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <linux/soundcard.h>
+
+/* Which device to use */
+#define DEV_DSP "/dev/dsp"
+
+/* Lets use 160 sample frames, just like GSM.  */
+#define FRAME_SIZE 160
+
+/* When you set the frame size, you have to come up with
+   the right buffer format as well. */
+/* 5 64-byte frames = one frame */
+#define BUFFER_FMT ((buffersize * 5) << 16) | (0x0006);
+
+/* Don't switch between read/write modes faster than every 300 ms */
+#define MIN_SWITCH_TIME 600
+
+static struct timeval lasttime;
+
+static int usecnt;
+static int needanswer = 0;
+static int needhangup = 0;
+static int silencesuppression = 0;
+static int silencethreshold = 1000;
+
+static char digits[80] = "";
+
+static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
+
+static char *type = "Console";
+static char *desc = "OSS Console Channel Driver";
+static char *tdesc = "OSS Console Channel Driver";
+static char *config = "oss.conf";
+
+static char context[AST_MAX_EXTENSION] = "default";
+static char exten[AST_MAX_EXTENSION] = "s";
+
+/* Some pipes to prevent overflow */
+static int funnel[2];
+static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
+static pthread_t silly;
+
+static struct chan_oss_pvt {
+	/* We only have one OSS structure -- near sighted perhaps, but it
+	   keeps this driver as simple as possible -- as it should be. */
+	struct ast_channel *owner;
+	char exten[AST_MAX_EXTENSION];
+	char context[AST_MAX_EXTENSION];
+} oss;
+
+static int time_has_passed()
+{
+	struct timeval tv;
+	int ms;
+	gettimeofday(&tv, NULL);
+	ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
+			(tv.tv_usec - lasttime.tv_usec) / 1000;
+	if (ms > MIN_SWITCH_TIME)
+		return -1;
+	return 0;
+}
+
+/* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
+   with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
+   usually plenty. */
+
+
+#define MAX_BUFFER_SIZE 100
+static int buffersize = 3;
+
+static int full_duplex = 0;
+
+/* Are we reading or writing (simulated full duplex) */
+static int readmode = 1;
+
+/* File descriptor for sound device */
+static int sounddev = -1;
+
+static int autoanswer = 1;
+ 
+static int calc_loudness(short *frame)
+{
+	int sum = 0;
+	int x;
+	for (x=0;x<FRAME_SIZE;x++) {
+		if (frame[x] < 0)
+			sum -= frame[x];
+		else
+			sum += frame[x];
+	}
+	sum = sum/FRAME_SIZE;
+	return sum;
+}
+
+static int silence_suppress(short *buf)
+{
+#define SILBUF 3
+	int loudness;
+	static int silentframes = 0;
+	static char silbuf[FRAME_SIZE * 2 * SILBUF];
+	static int silbufcnt=0;
+	if (!silencesuppression)
+		return 0;
+	loudness = calc_loudness((short *)(buf));
+	if (option_debug)
+		ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
+	if (loudness < silencethreshold) {
+		silentframes++;
+		silbufcnt++;
+		/* Keep track of the last few bits of silence so we can play
+		   them as lead-in when the time is right */
+		if (silbufcnt >= SILBUF) {
+			/* Make way for more buffer */
+			memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1));
+			silbufcnt--;
+		}
+		memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2);
+		if (silentframes > 10) {
+			/* We've had plenty of silence, so compress it now */
+			return 1;
+		}
+	} else {
+		silentframes=0;
+		/* Write any buffered silence we have, it may have something
+		   important */
+		if (silbufcnt) {
+			write(funnel[1], silbuf, silbufcnt * FRAME_SIZE);
+			silbufcnt = 0;
+		}
+	}
+	return 0;
+}
+
+static void *silly_thread(void *ignore)
+{
+	char buf[FRAME_SIZE * 2];
+	int pos=0;
+	int res=0;
+	/* Read from the sound device, and write to the pipe. */
+	for (;;) {
+		/* Give the writer a better shot at the lock */
+#if 0
+		usleep(1000);
+#endif		
+		pthread_testcancel();
+		pthread_mutex_lock(&sound_lock);
+		res = read(sounddev, buf + pos, FRAME_SIZE * 2 - pos);
+		pthread_mutex_unlock(&sound_lock);
+		if (res > 0) {
+			pos += res;
+			if (pos == FRAME_SIZE * 2) {
+				if (needhangup || needanswer || strlen(digits) || 
+				    !silence_suppress((short *)buf)) {
+					res = write(funnel[1], buf, sizeof(buf));
+				}
+				pos = 0;
+			}
+		} else {
+			close(funnel[1]);
+			break;
+		}
+		pthread_testcancel();
+	}
+	return NULL;
+}
+
+static int setformat(void)
+{
+	int fmt, desired, res, fd = sounddev;
+	static int warnedalready = 0;
+	static int warnedalready2 = 0;
+	pthread_mutex_lock(&sound_lock);
+	fmt = AFMT_S16_LE;
+	res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
+		pthread_mutex_unlock(&sound_lock);
+		return -1;
+	}
+	res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+	if (res >= 0) {
+		if (option_verbose > 1) 
+			ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
+		full_duplex = -1;
+	}
+	fmt = 0;
+	res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		pthread_mutex_unlock(&sound_lock);
+		return -1;
+	}
+	/* 8000 Hz desired */
+	desired = 8000;
+	fmt = desired;
+	res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		pthread_mutex_unlock(&sound_lock);
+		return -1;
+	}
+	if (fmt != desired) {
+		if (!warnedalready++)
+			ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
+	}
+#if 1
+	fmt = BUFFER_FMT;
+	res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+	if (res < 0) {
+		if (!warnedalready2++)
+			ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
+	}
+#endif
+	pthread_mutex_unlock(&sound_lock);
+	return 0;
+}
+
+static int soundcard_setoutput(int force)
+{
+	/* Make sure the soundcard is in output mode.  */
+	int fd = sounddev;
+	if (full_duplex || (!readmode && !force))
+		return 0;
+	pthread_mutex_lock(&sound_lock);
+	readmode = 0;
+	if (force || time_has_passed()) {
+		ioctl(sounddev, SNDCTL_DSP_RESET);
+		/* Keep the same fd reserved by closing the sound device and copying stdin at the same
+		   time. */
+		/* dup2(0, sound); */ 
+		close(sounddev);
+		fd = open(DEV_DSP, O_WRONLY);
+		if (fd < 0) {
+			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		/* dup2 will close the original and make fd be sound */
+		if (dup2(fd, sounddev) < 0) {
+			ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		if (setformat()) {
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		pthread_mutex_unlock(&sound_lock);
+		return 0;
+	}
+	pthread_mutex_unlock(&sound_lock);
+	return 1;
+}
+
+static int soundcard_setinput(int force)
+{
+	int fd = sounddev;
+	if (full_duplex || (readmode && !force))
+		return 0;
+	pthread_mutex_lock(&sound_lock);
+	readmode = -1;
+	if (force || time_has_passed()) {
+		ioctl(sounddev, SNDCTL_DSP_RESET);
+		close(sounddev);
+		/* dup2(0, sound); */
+		fd = open(DEV_DSP, O_RDONLY);
+		if (fd < 0) {
+			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		/* dup2 will close the original and make fd be sound */
+		if (dup2(fd, sounddev) < 0) {
+			ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		if (setformat()) {
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		pthread_mutex_unlock(&sound_lock);
+		return 0;
+	}
+	pthread_mutex_unlock(&sound_lock);
+	return 1;
+}
+
+static int soundcard_init()
+{
+	/* Assume it's full duplex for starters */
+	int fd = open(DEV_DSP, 	O_RDWR);
+	if (fd < 0) {
+		ast_log(LOG_ERROR, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
+		return fd;
+	}
+	gettimeofday(&lasttime, NULL);
+	sounddev = fd;
+	setformat();
+	if (!full_duplex) 
+		soundcard_setinput(1);
+	return sounddev;
+}
+
+static int oss_digit(struct ast_channel *c, char digit)
+{
+	ast_verbose( " << Console Received digit %c >> \n", digit);
+	return 0;
+}
+
+static int oss_call(struct ast_channel *c, char *dest, int timeout)
+{
+	ast_verbose( " << Call placed to '%s' on console >> \n", dest);
+	if (autoanswer) {
+		ast_verbose( " << Auto-answered >> \n" );
+		needanswer = 1;
+	} else {
+		ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
+	}
+	return 0;
+}
+
+static int oss_answer(struct ast_channel *c)
+{
+	ast_verbose( " << Console call has been answered >> \n");
+	c->state = AST_STATE_UP;
+	return 0;
+}
+
+static int oss_hangup(struct ast_channel *c)
+{
+	c->pvt->pvt = NULL;
+	oss.owner = NULL;
+	ast_verbose( " << Hangup on console >> \n");
+	pthread_mutex_lock(&usecnt_lock);
+	usecnt--;
+	pthread_mutex_unlock(&usecnt_lock);
+	needhangup = 0;
+	needanswer = 0;
+	return 0;
+}
+
+static int soundcard_writeframe(short *data)
+{	
+	/* Write an exactly FRAME_SIZE sized of frame */
+	static int bufcnt = 0;
+	static char buffer[FRAME_SIZE * 2 * MAX_BUFFER_SIZE * 5];
+	struct audio_buf_info info;
+	int res;
+	int fd = sounddev;
+	static int warned=0;
+	pthread_mutex_lock(&sound_lock);
+	if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
+		if (!warned)
+			ast_log(LOG_WARNING, "Error reading output space\n");
+		bufcnt = buffersize;
+		warned++;
+	}
+	if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
+		/* We've run out of stuff, buffer again */
+		bufcnt = 0;
+	}
+	if (bufcnt == buffersize) {
+		/* Write sample immediately */
+		res = write(fd, ((void *)data), FRAME_SIZE * 2);
+	} else {
+		/* Copy the data into our buffer */
+		res = FRAME_SIZE * 2;
+		memcpy(buffer + (bufcnt * FRAME_SIZE * 2), data, FRAME_SIZE * 2);
+		bufcnt++;
+		if (bufcnt == buffersize) {
+			res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
+		}
+	}
+	pthread_mutex_unlock(&sound_lock);
+	return res;
+}
+
+
+static int oss_write(struct ast_channel *chan, struct ast_frame *f)
+{
+	int res;
+	static char sizbuf[8000];
+	static int sizpos = 0;
+	int len = sizpos;
+	int pos;
+	if (!full_duplex && (strlen(digits) || needhangup || needanswer)) {
+		/* If we're half duplex, we have to switch to read mode
+		   to honor immediate needs if necessary */
+		res = soundcard_setinput(1);
+		if (res < 0) {
+			ast_log(LOG_WARNING, "Unable to set device to input mode\n");
+			return -1;
+		}
+		return 0;
+	}
+	res = soundcard_setoutput(0);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set output device\n");
+		return -1;
+	} else if (res > 0) {
+		/* The device is still in read mode, and it's too soon to change it,
+		   so just pretend we wrote it */
+		return 0;
+	}
+	/* We have to digest the frame in 160-byte portions */
+	if (f->datalen > sizeof(sizbuf) - sizpos) {
+		ast_log(LOG_WARNING, "Frame too large\n");
+		return -1;
+	}
+	memcpy(sizbuf + sizpos, f->data, f->datalen);
+	len += f->datalen;
+	pos = 0;
+	while(len - pos > FRAME_SIZE * 2) {
+		soundcard_writeframe((short *)(sizbuf + pos));
+		pos += FRAME_SIZE * 2;
+	}
+	if (len - pos) 
+		memmove(sizbuf, sizbuf + pos, len - pos);
+	sizpos = len - pos;
+	return 0;
+}
+
+static struct ast_frame *oss_read(struct ast_channel *chan)
+{
+	static struct ast_frame f;
+	static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
+	static int readpos = 0;
+	int res;
+	
+#if 0
+	ast_log(LOG_DEBUG, "oss_read()\n");
+#endif
+	
+	f.frametype = AST_FRAME_NULL;
+	f.subclass = 0;
+	f.timelen = 0;
+	f.datalen = 0;
+	f.data = NULL;
+	f.offset = 0;
+	f.src = type;
+	f.mallocd = 0;
+	
+	if (needhangup) {
+		return NULL;
+	}
+	if (strlen(digits)) {
+		f.frametype = AST_FRAME_DTMF;
+		f.subclass = digits[0];
+		for (res=0;res<strlen(digits);res++)
+			digits[res] = digits[res + 1];
+		return &f;
+	}
+	
+	if (needanswer) {
+		needanswer = 0;
+		f.frametype = AST_FRAME_CONTROL;
+		f.subclass = AST_CONTROL_ANSWER;
+		chan->state = AST_STATE_UP;
+		return &f;
+	}
+	
+	res = soundcard_setinput(0);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set input mode\n");
+		return NULL;
+	}
+	if (res > 0) {
+		/* Theoretically shouldn't happen, but anyway, return a NULL frame */
+		return &f;
+	}
+	res = read(funnel[0], buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Error reading from sound device: %s\n", strerror(errno));
+		return NULL;
+	}
+	readpos += res;
+	
+	if (readpos == FRAME_SIZE * 2) {
+		/* A real frame */
+		readpos = 0;
+		f.frametype = AST_FRAME_VOICE;
+		f.subclass = AST_FORMAT_SLINEAR;
+		f.timelen = FRAME_SIZE / 8;
+		f.datalen = FRAME_SIZE * 2;
+		f.data = buf + AST_FRIENDLY_OFFSET;
+		f.offset = AST_FRIENDLY_OFFSET;
+		f.src = type;
+		f.mallocd = 0;
+	}
+	return &f;
+}
+
+static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
+{
+	struct ast_channel *tmp;
+	tmp = ast_channel_alloc();
+	if (tmp) {
+		snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
+		tmp->type = type;
+		tmp->fd = funnel[0];
+		tmp->format = AST_FORMAT_SLINEAR;
+		tmp->pvt->pvt = p;
+		tmp->pvt->send_digit = oss_digit;
+		tmp->pvt->hangup = oss_hangup;
+		tmp->pvt->answer = oss_answer;
+		tmp->pvt->read = oss_read;
+		tmp->pvt->write = oss_write;
+		if (strlen(p->context))
+			strncpy(tmp->context, p->context, sizeof(tmp->context));
+		if (strlen(p->exten))
+			strncpy(tmp->exten, p->exten, sizeof(tmp->exten));
+		p->owner = tmp;
+		tmp->state = state;
+		pthread_mutex_lock(&usecnt_lock);
+		usecnt++;
+		pthread_mutex_unlock(&usecnt_lock);
+		ast_update_use_count();
+		if (state != AST_STATE_DOWN) {
+			if (ast_pbx_start(tmp)) {
+				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
+				ast_hangup(tmp);
+				tmp = NULL;
+			}
+		}
+	}
+	return tmp;
+}
+
+static struct ast_channel *oss_request(char *type, int format, void *data)
+{
+	int oldformat = format;
+	format &= AST_FORMAT_SLINEAR;
+	if (!format) {
+		ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
+		return NULL;
+	}
+	if (oss.owner) {
+		ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+		return NULL;
+	}
+	return oss_new(&oss, AST_STATE_DOWN);
+}
+
+static int console_autoanswer(int fd, int argc, char *argv[])
+{
+	if ((argc != 1) && (argc != 2))
+		return RESULT_SHOWUSAGE;
+	if (argc == 1) {
+		ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
+		return RESULT_SUCCESS;
+	} else {
+		if (!strcasecmp(argv[1], "on"))
+			autoanswer = -1;
+		else if (!strcasecmp(argv[1], "off"))
+			autoanswer = 0;
+		else
+			return RESULT_SHOWUSAGE;
+	}
+	return RESULT_SUCCESS;
+}
+
+static char *autoanswer_complete(char *line, char *word, int pos, int state)
+{
+#ifndef MIN
+#define MIN(a,b) ((a) < (b) ? (a) : (b))
+#endif
+	switch(state) {
+	case 0:
+		if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
+			return strdup("on");
+	case 1:
+		if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
+			return strdup("off");
+	default:
+		return NULL;
+	}
+	return NULL;
+}
+
+static char autoanswer_usage[] =
+"Usage: autoanswer [on|off]\n"
+"       Enables or disables autoanswer feature.  If used without\n"
+"       argument, displays the current on/off status of autoanswer.\n"
+"       The default value of autoanswer is in 'oss.conf'.\n";
+
+static int console_answer(int fd, int argc, char *argv[])
+{
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	if (!oss.owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	needanswer++;
+	return RESULT_SUCCESS;
+}
+
+static char answer_usage[] =
+"Usage: answer\n"
+"       Answers an incoming call on the console (OSS) channel.\n";
+
+static int console_hangup(int fd, int argc, char *argv[])
+{
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	if (!oss.owner) {
+		ast_cli(fd, "No call to hangup up\n");
+		return RESULT_FAILURE;
+	}
+	needhangup++;
+	return RESULT_SUCCESS;
+}
+
+static char hangup_usage[] =
+"Usage: hangup\n"
+"       Hangs up any call currently placed on the console.\n";
+
+
+static int console_dial(int fd, int argc, char *argv[])
+{
+	char tmp[256], *tmp2;
+	char *mye, *myc;
+	if ((argc != 1) && (argc != 2))
+		return RESULT_SHOWUSAGE;
+	if (oss.owner) {
+		if (argc == 2)
+			strncat(digits, argv[1], sizeof(digits) - strlen(digits));
+		else {
+			ast_cli(fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
+			return RESULT_FAILURE;
+		}
+		return RESULT_SUCCESS;
+	}
+	mye = exten;
+	myc = context;
+	if (argc == 2) {
+		strncpy(tmp, argv[1], sizeof(tmp));
+		strtok(tmp, "@");
+		tmp2 = strtok(NULL, "@");
+		if (strlen(tmp))
+			mye = tmp;
+		if (tmp2 && strlen(tmp2))
+			myc = tmp2;
+	}
+	if (ast_exists_extension(NULL, myc, mye, 1)) {
+		strncpy(oss.exten, mye, sizeof(oss.exten));
+		strncpy(oss.context, myc, sizeof(oss.context));
+		oss_new(&oss, AST_STATE_UP);
+	} else
+		ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
+	return RESULT_SUCCESS;
+}
+
+static char dial_usage[] =
+"Usage: dial [extension[@context]]\n"
+"       Dials a given extensison (";
+
+
+static struct ast_cli_entry myclis[] = {
+	{ { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage },
+	{ { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage },
+	{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
+	{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
+};
+
+int load_module()
+{
+	int res;
+	int x;
+	int flags;
+	struct ast_config *cfg = ast_load(config);
+	struct ast_variable *v;
+	res = pipe(funnel);
+	if (res) {
+		ast_log(LOG_ERROR, "Unable to create pipe\n");
+		return -1;
+	}
+	/* We make the funnel so that writes to the funnel don't block...
+	   Our "silly" thread can read to its heart content, preventing
+	   recording overruns */
+	flags = fcntl(funnel[1], F_GETFL);
+#if 0
+	fcntl(funnel[0], F_SETFL, flags | O_NONBLOCK);
+#endif
+	fcntl(funnel[1], F_SETFL, flags | O_NONBLOCK);
+	res = soundcard_init();
+	if (res < 0) {
+		close(funnel[1]);
+		close(funnel[0]);
+		return -1;
+	}
+	if (!full_duplex)
+		ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
+	pthread_create(&silly, NULL, silly_thread, NULL);
+	res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
+	if (res < 0) {
+		ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
+		return -1;
+	}
+	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+		ast_cli_register(myclis + x);
+	if (cfg) {
+		v = ast_variable_browse(cfg, "general");
+		while(v) {
+			if (!strcasecmp(v->name, "autoanswer"))
+				autoanswer = ast_true(v->value);
+			else if (!strcasecmp(v->name, "silencesuppression"))
+				silencesuppression = ast_true(v->value);
+			else if (!strcasecmp(v->name, "silencethreshold"))
+				silencethreshold = atoi(v->value);
+			else if (!strcasecmp(v->name, "context"))
+				strncpy(context, v->value, sizeof(context));
+			else if (!strcasecmp(v->name, "extension"))
+				strncpy(exten, v->value, sizeof(exten));
+			v=v->next;
+		}
+		ast_destroy(cfg);
+	}
+	return 0;
+}
+
+
+
+int unload_module()
+{
+	int x;
+	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+		ast_cli_unregister(myclis + x);
+	close(sounddev);
+	if (funnel[0] > 0) {
+		close(funnel[0]);
+		close(funnel[1]);
+	}
+	if (silly) {
+		pthread_cancel(silly);
+		pthread_join(silly, NULL);
+	}
+	if (oss.owner)
+		ast_softhangup(oss.owner);
+	if (oss.owner)
+		return -1;
+	return 0;
+}
+
+char *description()
+{
+	return desc;
+}
+
+int usecount()
+{
+	int res;
+	pthread_mutex_lock(&usecnt_lock);
+	res = usecnt;
+	pthread_mutex_unlock(&usecnt_lock);
+	return res;
+}

configs/oss.conf.sample

@@ -0,0 +1,23 @@
+;
+; Open Sound System Console Driver Configuration File
+;
+[general]
+;
+; Automatically answer incoming calls on the console?  Choose yes if
+; for example you want to use this as an intercom.
+;
+autoanswer=yes
+;
+; Default context (is overridden with @context syntax)
+;
+;context=local
+;
+; Default extension to call
+;
+extension=s
+;
+; Silence supression can be enabled when sound is over a certain threshold.
+; The value for the threshold should probably be between 500 and 2000 or so,
+; but your mileage may vary.  Use the echo test to evaluate the best setting.
+;silencesuppression = yes
+;silencethreshold = 1000