Move to rizzo's new chan_oss, but leave the old one just in case... (bug #4379 with changes)

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6263 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 1120 insertions, 0 deletions

diff --git a/channels/chan_oss_old.c b/channels/chan_oss_old.c
new file mode 100755
index 000000000..8b61abf87
--- /dev/null
+++ b/channels/chan_oss_old.c
@@ -0,0 +1,1120 @@
+/*
+ * 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 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#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>
+
+#ifdef __linux
+#include <linux/soundcard.h>
+#elif defined(__FreeBSD__)
+#include <sys/soundcard.h>
+#else
+#include <soundcard.h>
+#endif
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/frame.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/module.h"
+#include "asterisk/options.h"
+#include "asterisk/pbx.h"
+#include "asterisk/config.h"
+#include "asterisk/cli.h"
+#include "asterisk/utils.h"
+#include "asterisk/causes.h"
+#include "asterisk/endian.h"
+
+#include "busy.h"
+#include "ringtone.h"
+#include "ring10.h"
+#include "answer.h"
+
+/* Which device to use */
+#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
+#define DEV_DSP "/dev/audio"
+#else
+#define DEV_DSP "/dev/dsp"
+#endif
+
+/* 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 * 10) << 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 silencesuppression = 0;
+static int silencethreshold = 1000;
+static int playbackonly = 0;
+
+
+AST_MUTEX_DEFINE_STATIC(usecnt_lock);
+
+static const char type[] = "Console";
+static const char desc[] = "OSS Console Channel Driver";
+static const char tdesc[] = "OSS Console Channel Driver";
+static const char config[] = "oss.conf";
+
+static char context[AST_MAX_CONTEXT] = "default";
+static char language[MAX_LANGUAGE] = "";
+static char exten[AST_MAX_EXTENSION] = "s";
+
+static int hookstate=0;
+
+static short silence[FRAME_SIZE] = {0, };
+
+struct sound {
+	int ind;
+	short *data;
+	int datalen;
+	int samplen;
+	int silencelen;
+	int repeat;
+};
+
+static struct sound sounds[] = {
+	{ AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
+	{ AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
+	{ AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
+	{ AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
+	{ AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
+};
+
+/* Sound command pipe */
+static int sndcmd[2];
+
+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_CONTEXT];
+} oss;
+
+static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
+static int oss_digit(struct ast_channel *c, char digit);
+static int oss_text(struct ast_channel *c, const char *text);
+static int oss_hangup(struct ast_channel *c);
+static int oss_answer(struct ast_channel *c);
+static struct ast_frame *oss_read(struct ast_channel *chan);
+static int oss_call(struct ast_channel *c, char *dest, int timeout);
+static int oss_write(struct ast_channel *chan, struct ast_frame *f);
+static int oss_indicate(struct ast_channel *chan, int cond);
+static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
+
+static const struct ast_channel_tech oss_tech = {
+	.type = type,
+	.description = tdesc,
+	.capabilities = AST_FORMAT_SLINEAR,
+	.requester = oss_request,
+	.send_digit = oss_digit,
+	.send_text = oss_text,
+	.hangup = oss_hangup,
+	.answer = oss_answer,
+	.read = oss_read,
+	.call = oss_call,
+	.write = oss_write,
+	.indicate = oss_indicate,
+	.fixup = oss_fixup,
+};
+
+static int time_has_passed(void)
+{
+	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. */
+
+static pthread_t sthread;
+
+#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;
+ 
+#if 0
+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;
+}
+#endif
+
+static int cursound = -1;
+static int sampsent = 0;
+static int silencelen=0;
+static int offset=0;
+static int nosound=0;
+
+static int send_sound(void)
+{
+	short myframe[FRAME_SIZE];
+	int total = FRAME_SIZE;
+	short *frame = NULL;
+	int amt=0;
+	int res;
+	int myoff;
+	audio_buf_info abi;
+	if (cursound > -1) {
+		res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
+		if (res) {
+			ast_log(LOG_WARNING, "Unable to read output space\n");
+			return -1;
+		}
+		/* Calculate how many samples we can send, max */
+		if (total > (abi.fragments * abi.fragsize / 2)) 
+			total = abi.fragments * abi.fragsize / 2;
+		res = total;
+		if (sampsent < sounds[cursound].samplen) {
+			myoff=0;
+			while(total) {
+				amt = total;
+				if (amt > (sounds[cursound].datalen - offset)) 
+					amt = sounds[cursound].datalen - offset;
+				memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
+				total -= amt;
+				offset += amt;
+				sampsent += amt;
+				myoff += amt;
+				if (offset >= sounds[cursound].datalen)
+					offset = 0;
+			}
+			/* Set it up for silence */
+			if (sampsent >= sounds[cursound].samplen) 
+				silencelen = sounds[cursound].silencelen;
+			frame = myframe;
+		} else {
+			if (silencelen > 0) {
+				frame = silence;
+				silencelen -= res;
+			} else {
+				if (sounds[cursound].repeat) {
+					/* Start over */
+					sampsent = 0;
+					offset = 0;
+				} else {
+					cursound = -1;
+					nosound = 0;
+				}
+			}
+		}
+		if (frame)
+			res = write(sounddev, frame, res * 2);
+		if (res > 0)
+			return 0;
+		return res;
+	}
+	return 0;
+}
+
+static void *sound_thread(void *unused)
+{
+	fd_set rfds;
+	fd_set wfds;
+	int max;
+	int res;
+	char ign[4096];
+	if (read(sounddev, ign, sizeof(sounddev)) < 0)
+		ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+	for(;;) {
+		FD_ZERO(&rfds);
+		FD_ZERO(&wfds);
+		max = sndcmd[0];
+		FD_SET(sndcmd[0], &rfds);
+		if (!oss.owner) {
+			FD_SET(sounddev, &rfds);
+			if (sounddev > max)
+				max = sounddev;
+		}
+		if (cursound > -1) {
+			FD_SET(sounddev, &wfds);
+			if (sounddev > max)
+				max = sounddev;
+		}
+		res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
+		if (res < 1) {
+			ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
+			continue;
+		}
+		if (FD_ISSET(sndcmd[0], &rfds)) {
+			read(sndcmd[0], &cursound, sizeof(cursound));
+			silencelen = 0;
+			offset = 0;
+			sampsent = 0;
+		}
+		if (FD_ISSET(sounddev, &rfds)) {
+			/* Ignore read */
+			if (read(sounddev, ign, sizeof(ign)) < 0)
+				ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+		}
+		if (FD_ISSET(sounddev, &wfds))
+			if (send_sound())
+				ast_log(LOG_WARNING, "Failed to write sound\n");
+	}
+	/* Never reached */
+	return NULL;
+}
+
+#if 0
+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(sounddev, silbuf, silbufcnt * FRAME_SIZE);
+			silbufcnt = 0;
+		}
+	}
+	return 0;
+}
+#endif
+
+static int setformat(void)
+{
+	int fmt, desired, res, fd = sounddev;
+	static int warnedalready = 0;
+	static int warnedalready2 = 0;
+
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+	fmt = AFMT_S16_LE;
+#else
+	fmt = AFMT_S16_BE;
+#endif
+
+	res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
+		return -1;
+	}
+	res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+	
+	/* Check to see if duplex set (FreeBSD Bug)*/
+	res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
+	
+	if ((fmt & DSP_CAP_DUPLEX) && !res) {
+		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");
+		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");
+		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
+	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;
+	readmode = 0;
+	if (force || time_has_passed()) {
+		ioctl(sounddev, SNDCTL_DSP_RESET, 0);
+		/* 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 |O_NONBLOCK);
+		if (fd < 0) {
+			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+			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));
+			return -1;
+		}
+		if (setformat()) {
+			return -1;
+		}
+		return 0;
+	}
+	return 1;
+}
+
+static int soundcard_setinput(int force)
+{
+	int fd = sounddev;
+	if (full_duplex || (readmode && !force))
+		return 0;
+	readmode = -1;
+	if (force || time_has_passed()) {
+		ioctl(sounddev, SNDCTL_DSP_RESET, 0);
+		close(sounddev);
+		/* dup2(0, sound); */
+		fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
+		if (fd < 0) {
+			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+			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));
+			return -1;
+		}
+		if (setformat()) {
+			return -1;
+		}
+		return 0;
+	}
+	return 1;
+}
+
+static int soundcard_init(void)
+{
+	/* Assume it's full duplex for starters */
+	int fd = open(DEV_DSP, 	O_RDWR | O_NONBLOCK);
+	if (fd < 0) {
+		ast_log(LOG_WARNING, "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_text(struct ast_channel *c, const char *text)
+{
+	ast_verbose( " << Console Received text %s >> \n", text);
+	return 0;
+}
+
+static int oss_call(struct ast_channel *c, char *dest, int timeout)
+{
+	int res = 3;
+	struct ast_frame f = { 0, };
+	ast_verbose( " << Call placed to '%s' on console >> \n", dest);
+	if (autoanswer) {
+		ast_verbose( " << Auto-answered >> \n" );
+		f.frametype = AST_FRAME_CONTROL;
+		f.subclass = AST_CONTROL_ANSWER;
+		ast_queue_frame(c, &f);
+	} else {
+		nosound = 1;
+		ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
+		f.frametype = AST_FRAME_CONTROL;
+		f.subclass = AST_CONTROL_RINGING;
+		ast_queue_frame(c, &f);
+		write(sndcmd[1], &res, sizeof(res));
+	}
+	return 0;
+}
+
+static void answer_sound(void)
+{
+	int res;
+	nosound = 1;
+	res = 4;
+	write(sndcmd[1], &res, sizeof(res));
+	
+}
+
+static int oss_answer(struct ast_channel *c)
+{
+	ast_verbose( " << Console call has been answered >> \n");
+	answer_sound();
+	ast_setstate(c, AST_STATE_UP);
+	cursound = -1;
+	nosound=0;
+	return 0;
+}
+
+static int oss_hangup(struct ast_channel *c)
+{
+	int res = 0;
+	cursound = -1;
+	c->tech_pvt = NULL;
+	oss.owner = NULL;
+	ast_verbose( " << Hangup on console >> \n");
+	ast_mutex_lock(&usecnt_lock);
+	usecnt--;
+	ast_mutex_unlock(&usecnt_lock);
+	if (hookstate) {
+		if (autoanswer) {
+			/* Assume auto-hangup too */
+			hookstate = 0;
+		} else {
+			/* Make congestion noise */
+			res = 2;
+			write(sndcmd[1], &res, sizeof(res));
+		}
+	}
+	return 0;
+}
+
+static int soundcard_writeframe(short *data)
+{	
+	/* Write an exactly FRAME_SIZE sized of frame */
+	static int bufcnt = 0;
+	static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5];
+	struct audio_buf_info info;
+	int res;
+	int fd = sounddev;
+	static int warned=0;
+	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), data, FRAME_SIZE * 2);
+		bufcnt++;
+		if (bufcnt == buffersize) {
+			res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
+		}
+	}
+	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;
+	/* Immediately return if no sound is enabled */
+	if (nosound)
+		return 0;
+	/* Stop any currently playing sound */
+	cursound = -1;
+	if (!full_duplex && !playbackonly) {
+		/* If we're half duplex, we have to switch to read mode
+		   to honor immediate needs if necessary.  But if we are in play
+		   back only mode, then we don't switch because the console
+		   is only being used one way -- just to playback something. */
+		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.samples = 0;
+	f.datalen = 0;
+	f.data = NULL;
+	f.offset = 0;
+	f.src = type;
+	f.mallocd = 0;
+	f.delivery.tv_sec = 0;
+	f.delivery.tv_usec = 0;
+	
+	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(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
+#if 0
+		CRASH;
+#endif		
+		return NULL;
+	}
+	readpos += res;
+	
+	if (readpos >= FRAME_SIZE * 2) {
+		/* A real frame */
+		readpos = 0;
+		if (chan->_state != AST_STATE_UP) {
+			/* Don't transmit unless it's up */
+			return &f;
+		}
+		f.frametype = AST_FRAME_VOICE;
+		f.subclass = AST_FORMAT_SLINEAR;
+		f.samples = FRAME_SIZE;
+		f.datalen = FRAME_SIZE * 2;
+		f.data = buf + AST_FRIENDLY_OFFSET;
+		f.offset = AST_FRIENDLY_OFFSET;
+		f.src = type;
+		f.mallocd = 0;
+		f.delivery.tv_sec = 0;
+		f.delivery.tv_usec = 0;
+#if 0
+		{ static int fd = -1;
+		  if (fd < 0)
+		  	fd = open("output.raw", O_RDWR | O_TRUNC | O_CREAT);
+		  write(fd, f.data, f.datalen);
+		}
+#endif		
+	}
+	return &f;
+}
+
+static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
+{
+	struct chan_oss_pvt *p = newchan->tech_pvt;
+	p->owner = newchan;
+	return 0;
+}
+
+static int oss_indicate(struct ast_channel *chan, int cond)
+{
+	int res;
+	switch(cond) {
+	case AST_CONTROL_BUSY:
+		res = 1;
+		break;
+	case AST_CONTROL_CONGESTION:
+		res = 2;
+		break;
+	case AST_CONTROL_RINGING:
+		res = 0;
+		break;
+	case -1:
+		cursound = -1;
+		return 0;
+	default:
+		ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
+		return -1;
+	}
+	if (res > -1) {
+		write(sndcmd[1], &res, sizeof(res));
+	}
+	return 0;	
+}
+
+static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
+{
+	struct ast_channel *tmp;
+	tmp = ast_channel_alloc(1);
+	if (tmp) {
+		tmp->tech = &oss_tech;
+		snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
+		tmp->type = type;
+		tmp->fds[0] = sounddev;
+		tmp->nativeformats = AST_FORMAT_SLINEAR;
+		tmp->readformat = AST_FORMAT_SLINEAR;
+		tmp->writeformat = AST_FORMAT_SLINEAR;
+		tmp->tech_pvt = p;
+		if (strlen(p->context))
+			strncpy(tmp->context, p->context, sizeof(tmp->context)-1);
+		if (strlen(p->exten))
+			strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1);
+		if (strlen(language))
+			strncpy(tmp->language, language, sizeof(tmp->language)-1);
+		p->owner = tmp;
+		ast_setstate(tmp, state);
+		ast_mutex_lock(&usecnt_lock);
+		usecnt++;
+		ast_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(const char *type, int format, void *data, int *cause)
+{
+	int oldformat = format;
+	struct ast_channel *tmp;
+	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");
+		*cause = AST_CAUSE_BUSY;
+		return NULL;
+	}
+	tmp= oss_new(&oss, AST_STATE_DOWN);
+	if (!tmp) {
+		ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
+	}
+	return tmp;
+}
+
+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[])
+{
+	struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	if (!oss.owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	hookstate = 1;
+	cursound = -1;
+	ast_queue_frame(oss.owner, &f);
+	answer_sound();
+	return RESULT_SUCCESS;
+}
+
+static char sendtext_usage[] =
+"Usage: send text <message>\n"
+"       Sends a text message for display on the remote terminal.\n";
+
+static int console_sendtext(int fd, int argc, char *argv[])
+{
+	int tmparg = 2;
+	char text2send[256] = "";
+	struct ast_frame f = { 0, };
+	if (argc < 2)
+		return RESULT_SHOWUSAGE;
+	if (!oss.owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	if (strlen(text2send))
+		ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
+	text2send[0] = '\0';
+	while(tmparg < argc) {
+		strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
+		strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
+	}
+	if (strlen(text2send)) {
+		f.frametype = AST_FRAME_TEXT;
+		f.subclass = 0;
+		f.data = text2send;
+		f.datalen = strlen(text2send);
+		ast_queue_frame(oss.owner, &f);
+	}
+	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;
+	cursound = -1;
+	if (!oss.owner && !hookstate) {
+		ast_cli(fd, "No call to hangup up\n");
+		return RESULT_FAILURE;
+	}
+	hookstate = 0;
+	if (oss.owner) {
+		ast_queue_hangup(oss.owner);
+	}
+	return RESULT_SUCCESS;
+}
+
+static int console_flash(int fd, int argc, char *argv[])
+{
+	struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	cursound = -1;
+	if (!oss.owner) {
+		ast_cli(fd, "No call to flash\n");
+		return RESULT_FAILURE;
+	}
+	hookstate = 0;
+	if (oss.owner) {
+		ast_queue_frame(oss.owner, &f);
+	}
+	return RESULT_SUCCESS;
+}
+
+static char hangup_usage[] =
+"Usage: hangup\n"
+"       Hangs up any call currently placed on the console.\n";
+
+
+static char flash_usage[] =
+"Usage: flash\n"
+"       Flashes the call currently placed on the console.\n";
+
+static int console_dial(int fd, int argc, char *argv[])
+{
+	char tmp[256], *tmp2;
+	char *mye, *myc;
+	int x;
+	struct ast_frame f = { AST_FRAME_DTMF, 0 };
+	if ((argc != 1) && (argc != 2))
+		return RESULT_SHOWUSAGE;
+	if (oss.owner) {
+		if (argc == 2) {
+			for (x=0;x<strlen(argv[1]);x++) {
+				f.subclass = argv[1][x];
+				ast_queue_frame(oss.owner, &f);
+			}
+		} 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) {
+		char *stringp=NULL;
+		strncpy(tmp, argv[1], sizeof(tmp)-1);
+		stringp=tmp;
+		strsep(&stringp, "@");
+		tmp2 = strsep(&stringp, "@");
+		if (strlen(tmp))
+			mye = tmp;
+		if (tmp2 && strlen(tmp2))
+			myc = tmp2;
+	}
+	if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
+		strncpy(oss.exten, mye, sizeof(oss.exten)-1);
+		strncpy(oss.context, myc, sizeof(oss.context)-1);
+		hookstate = 1;
+		oss_new(&oss, AST_STATE_RINGING);
+	} 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 (and context if specified)\n";
+
+static int console_transfer(int fd, int argc, char *argv[])
+{
+	char tmp[256];
+	char *context;
+	if (argc != 2)
+		return RESULT_SHOWUSAGE;
+	if (oss.owner && ast_bridged_channel(oss.owner)) {
+		strncpy(tmp, argv[1], sizeof(tmp) - 1);
+		context = strchr(tmp, '@');
+		if (context) {
+			*context = '\0';
+			context++;
+		} else
+			context = oss.owner->context;
+		if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) {
+			ast_cli(fd, "Whee, transferring %s to %s@%s.\n", 
+					ast_bridged_channel(oss.owner)->name, tmp, context);
+			if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1))
+				ast_cli(fd, "Failed to transfer :(\n");
+		} else {
+			ast_cli(fd, "No such extension exists\n");
+		}
+	} else {
+		ast_cli(fd, "There is no call to transfer\n");
+	}
+	return RESULT_SUCCESS;
+}
+
+static char transfer_usage[] =
+"Usage: transfer <extension>[@context]\n"
+"       Transfers the currently connected call to the given extension (and\n"
+"context if specified)\n";
+
+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 },
+	{ { "flash", NULL }, console_flash, "Flash a call on the console", flash_usage },
+	{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
+	{ { "transfer", NULL }, console_transfer, "Transfer a call to a different extension", transfer_usage },
+	{ { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
+	{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
+};
+
+int load_module()
+{
+	int res;
+	int x;
+	struct ast_config *cfg;
+	struct ast_variable *v;
+	res = pipe(sndcmd);
+	if (res) {
+		ast_log(LOG_ERROR, "Unable to create pipe\n");
+		return -1;
+	}
+	res = soundcard_init();
+	if (res < 0) {
+		if (option_verbose > 1) {
+			ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n");
+			ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
+		}
+		return 0;
+	}
+	if (!full_duplex)
+		ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
+	res = ast_channel_register(&oss_tech);
+	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 = ast_config_load(config))) {
+		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)-1);
+			else if (!strcasecmp(v->name, "language"))
+				strncpy(language, v->value, sizeof(language)-1);
+			else if (!strcasecmp(v->name, "extension"))
+				strncpy(exten, v->value, sizeof(exten)-1);
+			else if (!strcasecmp(v->name, "playbackonly"))
+				playbackonly = ast_true(v->value);
+			v=v->next;
+		}
+		ast_config_destroy(cfg);
+	}
+	ast_pthread_create(&sthread, NULL, sound_thread, NULL);
+	return 0;
+}
+
+
+
+int unload_module()
+{
+	int x;
+
+	ast_channel_unregister(&oss_tech);
+	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+		ast_cli_unregister(myclis + x);
+	close(sounddev);
+	if (sndcmd[0] > 0) {
+		close(sndcmd[0]);
+		close(sndcmd[1]);
+	}
+	if (oss.owner)
+		ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
+	if (oss.owner)
+		return -1;
+	return 0;
+}
+
+char *description()
+{
+	return (char *) desc;
+}
+
+int usecount()
+{
+	return usecnt;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}