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, 995 insertions, 709 deletions

diff --git a/channels/chan_oss.c b/channels/chan_oss.c
index 8b61abf87..b8f0fb6ef 100755
--- a/channels/chan_oss.c
+++ b/channels/chan_oss.c
@@ -1,28 +1,27 @@
 /*
  * 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
+ *
+ * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
+ * note-this code best seen with ts=8 (8-spaces tabs) in the editor
  */
 
+#include <stdio.h>
+#include <ctype.h>	/* for isalnum */
+#include <string.h>
 #include <unistd.h>
-#include <fcntl.h>
-#include <errno.h>
 #include <sys/ioctl.h>
+#include <fcntl.h>
 #include <sys/time.h>
-#include <string.h>
 #include <stdlib.h>
-#include <stdio.h>
+#include <errno.h>
+
 
 #ifdef __linux
 #include <linux/soundcard.h>
@@ -44,16 +43,132 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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"
 
+/* ringtones we use */
 #include "busy.h"
 #include "ringtone.h"
 #include "ring10.h"
 #include "answer.h"
 
+/*
+ * Basic mode of operation:
+ *
+ * we have one keyboard (which receives commands from the keyboard)
+ * and multiple headset's connected to audio cards.
+ * Cards/Headsets are named as the sections of oss.conf.
+ * The section called [general] contains the default parameters.
+ *
+ * At any time, the keyboard is attached to one card, and you
+ * can switch among them using the command 'console foo'
+ * where 'foo' is the name of the card you want.
+ *
+ * oss.conf parameters are
+
+[general]
+; general config options, default values are shown
+; all but debug can go also in the device-specific sections.
+; debug=0x0		; misc debug flags, default is 0
+
+[card1]
+; autoanswer = no	; no autoanswer on call
+; autohangup = yes	; hangup when other party closes
+; extension=s		; default extension to call
+; context=default	; default context
+; language=""		; default language
+; overridecontext=no	; the whole dial string is considered an extension.
+			; if yes, the last @ will start the context
+
+; device=/dev/dsp	; device to open
+; mixer="-f /dev/mixer0 pcm 80 ; mixer command to run on start
+; queuesize=10		; frames in device driver
+; frags=8		; argument to SETFRAGMENT
+
+.. and so on for the other cards.
+
+ */
+
+/*
+ * Helper macros to parse config arguments. They will go in a common
+ * header file if their usage is globally accepted. In the meantime,
+ * we define them here. Typical usage is as below.
+ * Remember to open a block right before M_START (as it declares
+ * some variables) and use the M_* macros WITHOUT A SEMICOLON:
+ *
+ *	{
+ *		M_START(v->name, v->value) 
+ *
+ *		M_BOOL("dothis", x->flag1)
+ *		M_STR("name", x->somestring)
+ *		M_F("bar", some_c_code)
+ *		M_END(some_final_statement)
+ *		... other code in the block
+ *	}
+ *
+ * XXX NOTE these macros should NOT be replicated in other parts of asterisk. 
+ * Likely we will come up with a better way of doing config file parsing.
+ */
+#define M_START(var, val) \
+        char *__s = var; char *__val = val;
+#define M_END(x)   x;
+#define M_F(tag, f)			if (!strcasecmp((__s), tag)) { f; } else
+#define M_BOOL(tag, dst)	M_F(tag, (dst) = ast_true(__val) )
+#define M_UINT(tag, dst)	M_F(tag, (dst) = strtoul(__val, NULL, 0) )
+#define M_STR(tag, dst)		M_F(tag, ast_copy_string(dst, __val, sizeof(dst)))
+
+/*
+ * The following parameters are used in the driver:
+ *
+ *  FRAME_SIZE	the size of an audio frame, in samples.
+ *		160 is used almost universally, so you should not change it.
+ *
+ *  FRAGS	the argument for the SETFRAGMENT ioctl.
+ *		Overridden by the 'frags' parameter in oss.conf
+ *
+ *		Bits 0-7 are the base-2 log of the device's block size,
+ *		bits 16-31 are the number of blocks in the driver's queue.
+ *		There are a lot of differences in the way this parameter
+ *		is supported by different drivers, so you may need to
+ *		experiment a bit with the value.
+ *		A good default for linux is 30 blocks of 64 bytes, which
+ *		results in 6 frames of 320 bytes (160 samples).
+ *		FreeBSD works decently with blocks of 256 or 512 bytes,
+ *		leaving the number unspecified.
+ *		Note that this only refers to the device buffer size,
+ *		this module will then try to keep the lenght of audio
+ *		buffered within small constraints.
+ *
+ *  QUEUE_SIZE	The max number of blocks actually allowed in the device
+ *		driver's buffer, irrespective of the available number.
+ *		Overridden by the 'queuesize' parameter in oss.conf
+ *
+ *		Should be >=2, and at most as large as the hw queue above
+ *		(otherwise it will never be full).
+ */
+
+#define FRAME_SIZE	160
+#define	QUEUE_SIZE	10
+
+#if defined(__FreeBSD__)
+#define	FRAGS	0x8
+#else
+#define	FRAGS	( ( (6 * 5) << 16 ) | 0x6 )
+#endif
+
+/*
+ * XXX text message sizes are probably 256 chars, but i am
+ * not sure if there is a suitable definition anywhere.
+ */
+#define TEXT_SIZE	256
+
+#if 0
+#define	TRYOPEN	1	/* try to open on startup */
+#endif
+#define	O_CLOSE	0x444	/* special 'close' mode for device */
 /* Which device to use */
 #if defined( __OpenBSD__ ) || defined( __NetBSD__ )
 #define DEV_DSP "/dev/audio"
@@ -61,42 +176,29 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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
+#ifndef MIN
+#define MIN(a,b) ((a) < (b) ? (a) : (b))
+#endif
+#ifndef MAX
+#define MAX(a,b) ((a) > (b) ? (a) : (b))
+#endif
 
-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 char *config = "oss.conf";	/* default config file */
 
-static int hookstate=0;
-
-static short silence[FRAME_SIZE] = {0, };
+static int oss_debug;
 
+/*
+ * Each sound is made of 'datalen' samples of sound, repeated as needed to
+ * generate 'samplen' samples of data, then followed by 'silencelen' samples
+ * of silence. The loop is repeated if 'repeat' is set.
+ */
 struct sound {
 	int ind;
+	char *desc;
 	short *data;
 	int datalen;
 	int samplen;
@@ -105,25 +207,99 @@ struct sound {
 };
 
 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 },
+	{ AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
+	{ AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
+	{ AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
+	{ AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
+	{ AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
+	{ -1, NULL, 0, 0, 0, 0 },	/* end marker */
 };
 
-/* 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. */
+/*
+ * descriptor for one of our channels.
+ * There is one used for 'default' values (from the [general] entry in
+ * the configuration file), and then one instance for each device
+ * (the default is cloned from [general], others are only created
+ * if the relevant section exists).
+ */
+struct chan_oss_pvt {
+	struct chan_oss_pvt *next;
+
+	char *type;	/* XXX maybe take the one from oss_tech */
+	char *name;
+	/*
+	 * cursound indicates which in struct sound we play. -1 means nothing,
+	 * any other value is a valid sound, in which case sampsent indicates
+	 * the next sample to send in [0..samplen + silencelen]
+	 * nosound is set to disable the audio data from the channel
+	 * (so we can play the tones etc.).
+	 */
+	int sndcmd[2]; /* Sound command pipe */
+	int cursound;	/* index of sound to send */
+	int sampsent;	/* # of sound samples sent	*/
+	int nosound;	/* set to block audio from the PBX */
+
+	int total_blocks;	/* total blocks in the output device */
+	int sounddev;
+	enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
+	int autoanswer;
+	int autohangup;
+	int hookstate;
+	char *mixer_cmd;		/* initial command to issue to the mixer */
+	unsigned int	queuesize;	/* max fragments in queue */
+	unsigned int	frags;		/* parameter for SETFRAGMENT */
+
+	int warned;		/* various flags used for warnings */
+#define WARN_used_blocks	1
+#define WARN_speed		2
+#define WARN_frag		4
+	int w_errors;	/* overfull in the write path */
+	struct timeval lastopen;
+
+	int overridecontext;
+	int mute;
+	char device[64];	/* device to open */
+
+	pthread_t sthread;
+
 	struct ast_channel *owner;
-	char exten[AST_MAX_EXTENSION];
-	char context[AST_MAX_CONTEXT];
-} oss;
+	char ext[AST_MAX_EXTENSION];
+	char ctx[AST_MAX_CONTEXT];
+	char language[MAX_LANGUAGE];
+
+	/* buffers used in oss_write */
+	char oss_write_buf[FRAME_SIZE*2];
+	int oss_write_dst;
+	/* buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
+	 * plus enough room for a full frame
+	 */
+	char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
+	int readpos; /* read position above */
+	struct ast_frame read_f;	/* returned by oss_read */
+};
+
+static struct chan_oss_pvt oss_default = {
+	.type = "Console",
+	.cursound = -1,
+	.sounddev = -1,
+	.duplex = M_UNSET, /* XXX check this */
+	.autoanswer = 1,
+	.autohangup = 1,
+	.queuesize = QUEUE_SIZE,
+	.frags = FRAGS,
+	.ext = "s",
+	.ctx = "default",
+	.readpos = AST_FRIENDLY_OFFSET,	/* start here on reads */
+	.lastopen = { 0, 0 },
+};
 
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
+static char *oss_active;	 /* the active device */
+
+static int setformat(struct chan_oss_pvt *o, int mode);
+
+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);
@@ -135,704 +311,649 @@ 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,
+	.type =			"Console",
+	.description =	"OSS Console Channel Driver",
+	.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)
+/*
+ * returns a pointer to the descriptor with the given name
+ */
+static struct chan_oss_pvt *find_desc(char *dev)
 {
-	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. */
+	struct chan_oss_pvt *o;
 
-static pthread_t sthread;
+	for (o = oss_default.next; o && strcmp(o->name, dev) != 0; o = o->next)
+		;
+	if (o == NULL)
+		ast_log(LOG_WARNING, "could not find <%s>\n", dev);
+	return o;
+}
 
-#define MAX_BUFFER_SIZE 100
-static int buffersize = 3;
+/*
+ * split a string in extension-context, returns pointers to malloc'ed
+ * strings.
+ * If we have 'overridecontext' then the last @ is considered as
+ * a context separator, and the context is overridden.
+ * This is usually not very necessary as you can play with the dialplan,
+ * and it is nice not to need it because you have '@' in SIP addresses.
+ * Return value is the buffer address.
+ */
+static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
+{
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
+	if (ext == NULL || ctx == NULL)
+		return NULL;	/* error */
+	*ext = *ctx = NULL;
+	if (src && *src != '\0')
+		*ext = strdup(src);
+	if (*ext == NULL)
+		return NULL;
+	if (!o->overridecontext) {
+		/* parse from the right */
+		*ctx = strrchr(*ext, '@');
+		if (*ctx)
+			*(*ctx)++ = '\0';
+	}
+	return *ext;
+}
 
-static int full_duplex = 0;
+/*
+ * Returns the number of blocks used in the audio output channel
+ */
+static int used_blocks(struct chan_oss_pvt *o)
+{
+	struct audio_buf_info info;
 
-/* Are we reading or writing (simulated full duplex) */
-static int readmode = 1;
+	if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
+		if (! (o->warned & WARN_used_blocks)) {
+			ast_log(LOG_WARNING, "Error reading output space\n");
+			o->warned |= WARN_used_blocks;
+		}
+		return 1;
+	}
+	if (o->total_blocks == 0) {
+		if (0) /* debugging */
+			ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n",
+			    info.fragstotal,
+			    info.fragsize,
+			    info.fragments);
+		o->total_blocks = info.fragments;
+	}
+	return o->total_blocks - info.fragments;
+}
 
-/* File descriptor for sound device */
-static int sounddev = -1;
+/* Write an exactly FRAME_SIZE sized frame */
+static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
+{	
+	int res;
 
-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];
+	if (o->sounddev < 0)
+		setformat(o, O_RDWR);
+	if (o->sounddev < 0)
+		return 0;	/* not fatal */
+	/*
+	 * Nothing complex to manage the audio device queue.
+	 * If the buffer is full just drop the extra, otherwise write.
+	 * XXX in some cases it might be useful to write anyways after
+	 * a number of failures, to restart the output chain.
+	 */
+	res = used_blocks(o);
+	if (res > o->queuesize) {	/* no room to write a block */
+		if (o->w_errors++ == 0 && (oss_debug & 0x4))
+			ast_log(LOG_WARNING, "write: used %d blocks (%d)\n",
+			    res, o->w_errors);
+		return 0;
 	}
-	sum = sum/FRAME_SIZE;
-	return sum;
+	o->w_errors = 0;
+	return write(o->sounddev, ((void *)data), FRAME_SIZE * 2);
 }
-#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)
+/*
+ * Handler for 'sound writable' events from the sound thread.
+ * Builds a frame from the high level description of the sounds,
+ * and passes it to the audio device.
+ * The actual sound is made of 1 or more sequences of sound samples
+ * (s->datalen, repeated to make s->samplen samples) followed by
+ * s->silencelen samples of silence. The position in the sequence is stored
+ * in o->sampsent, which goes between 0 .. s->samplen+s->silencelen.
+ * In case we fail to write a frame, don't update o->sampsent.
+ */
+static void send_sound(struct chan_oss_pvt *o)
 {
 	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;
+	int ofs, l, start;
+	int l_sampsent = o->sampsent;
+	struct sound *s;
+
+	if (o->cursound < 0)	/* no sound to send */
+		return;
+	s = &sounds[o->cursound];
+	for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
+		l = s->samplen - l_sampsent;	/* # of available samples */
+		if (l > 0) {
+			start = l_sampsent % s->datalen; /* source offset */
+			if (l > FRAME_SIZE - ofs)	/* don't overflow the frame */
+				l = FRAME_SIZE - ofs;
+			if (l > s->datalen - start)	/* don't overflow the source */
+				l = s->datalen - start;
+			bcopy(s->data + start, myframe + ofs, l*2);
+			if (0)
+				ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n",
+				    l_sampsent, l, s->samplen, ofs);
+			l_sampsent += l;
+		} else { /* end of samples, maybe some silence */
+			static const short silence[FRAME_SIZE] = {0, };
+
+			l += s->silencelen;
+			if (l > 0) {
+				if (l > FRAME_SIZE - ofs)
+					l = FRAME_SIZE - ofs;
+				bcopy(silence, myframe + ofs, l*2);
+				l_sampsent += l;
+			} else { /* silence is over, restart sound if loop */
+				if (s->repeat == 0) {	/* last block */
+					o->cursound = -1;
+					o->nosound = 0;	/* allow audio data */
+					if (ofs < FRAME_SIZE)	/* pad with silence */
+						bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs)*2);
 				}
+				l_sampsent = 0;
 			}
 		}
-		if (frame)
-			res = write(sounddev, frame, res * 2);
-		if (res > 0)
-			return 0;
-		return res;
 	}
-	return 0;
+	l = soundcard_writeframe(o, myframe);
+	if (l > 0)
+		o->sampsent = l_sampsent;	/* update status */
 }
 
-static void *sound_thread(void *unused)
+static void *sound_thread(void *arg)
 {
-	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(;;) {
+	struct chan_oss_pvt *o = (struct chan_oss_pvt *)arg;
+
+	/*
+	 * Just in case, kick the driver by trying to read from it.
+	 * Ignore errors - this read is almost guaranteed to fail.
+	 */
+	read(o->sounddev, ign, sizeof(ign));
+	for (;;) {
+		fd_set rfds, wfds;
+		int maxfd, res;
+
 		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;
+		FD_SET(o->sndcmd[0], &rfds);
+		maxfd = o->sndcmd[0];	/* pipe from the main process */
+		if (o->cursound > -1 && o->sounddev < 0)
+			setformat(o, O_RDWR);   /* need the channel, try to reopen */
+		else if (o->cursound == -1 && o->owner == NULL)
+			setformat(o, O_CLOSE);  /* can close */
+		if (o->sounddev > -1) {
+			if (!o->owner) { /* no one owns the audio, so we must drain it */
+				FD_SET(o->sounddev, &rfds);
+				maxfd = MAX(o->sounddev, maxfd);
+			}
+			if (o->cursound > -1) {
+				FD_SET(o->sounddev, &wfds);
+				maxfd = MAX(o->sounddev, maxfd);
+			}
 		}
-		res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
+		/* ast_select emulates linux behaviour in terms of timeout handling */
+		res = ast_select(maxfd + 1, &rfds, &wfds, NULL, NULL);
 		if (res < 1) {
 			ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
+			sleep(1);
 			continue;
 		}
-		if (FD_ISSET(sndcmd[0], &rfds)) {
-			read(sndcmd[0], &cursound, sizeof(cursound));
-			silencelen = 0;
-			offset = 0;
-			sampsent = 0;
+		if (FD_ISSET(o->sndcmd[0], &rfds)) {
+			/* read which sound to play from the pipe */
+			int i, what = -1;
+
+			read(o->sndcmd[0], &what, sizeof(what));
+			for (i = 0; sounds[i].ind != -1; i++) {
+				if (sounds[i].ind == what) {
+					o->cursound = i;
+					o->sampsent = 0;
+					o->nosound = 1; /* block audio from pbx */
+					break;
+				}
+			}
+			if (sounds[i].ind == -1)
+				ast_log(LOG_WARNING, "invalid sound index: %d\n", what);
 		}
-		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 (o->sounddev > -1) {
+			if (FD_ISSET(o->sounddev, &rfds)) /* read and ignore errors */
+				read(o->sounddev, ign, sizeof(ign));
+			if (FD_ISSET(o->sounddev, &wfds))
+				send_sound(o);
 		}
-		if (FD_ISSET(sounddev, &wfds))
-			if (send_sound())
-				ast_log(LOG_WARNING, "Failed to write sound\n");
 	}
-	/* Never reached */
-	return NULL;
+	return NULL; /* Never reached */
 }
 
-#if 0
-static int silence_suppress(short *buf)
+/*
+ * reset and close the device if opened,
+ * then open and initialize it in the desired mode,
+ * trigger reads and writes so we can start using it.
+ */
+static int setformat(struct chan_oss_pvt *o, int mode)
 {
-#define SILBUF 3
-	int loudness;
-	static int silentframes = 0;
-	static char silbuf[FRAME_SIZE * 2 * SILBUF];
-	static int silbufcnt=0;
-	if (!silencesuppression)
+	int fmt, desired, res, fd;
+
+	if (o->sounddev >= 0) {
+		ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
+		close(o->sounddev);
+		o->duplex = M_UNSET;
+		o->sounddev = -1;
+	}
+	if (mode == O_CLOSE)	/* we are done */
 		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;
-		}
+	if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000)
+		return -1;	/* don't open too often */
+	o->lastopen = ast_tvnow();
+	fd = o->sounddev = open(o->device, mode |O_NONBLOCK);
+	if (fd < 0) {
+		ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n",
+		    o->device, strerror(errno));
+		return -1;
 	}
-	return 0;
-}
-#endif
-
-static int setformat(void)
-{
-	int fmt, desired, res, fd = sounddev;
-	static int warnedalready = 0;
-	static int warnedalready2 = 0;
+	if (o->owner)
+		o->owner->fds[0] = fd;
 
 #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;
+	switch (mode) {
+	case O_RDWR:
+		res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+		/* Check to see if duplex set (FreeBSD Bug)*/
+		res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
+		if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
+			if (option_verbose > 1) 
+				ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
+			o->duplex = M_FULL;
+		};
+		break;
+	case O_WRONLY:
+		o->duplex = M_WRITE;
+		break;
+	case O_RDONLY:
+		o->duplex = M_READ;
+		break;
 	}
+
 	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;
+	fmt = desired = 8000; /* 8000 Hz 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;
+		if (!(o->warned & WARN_speed)) {
+			ast_log(LOG_WARNING,
+			    "Requested %d Hz, got %d Hz -- sound may be choppy\n",
+			    desired, fmt);
+			o->warned |= WARN_speed;
 		}
-		/* 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;
+	/*
+	 * on Freebsd, SETFRAGMENT does not work very well on some cards.
+	 * Default to use 256 bytes, let the user override
+	 */
+	if (o->frags) {
+		fmt = o->frags;
+		res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+		if (res < 0) {
+			if (!(o->warned & WARN_frag)) {
+				ast_log(LOG_WARNING,
+					"Unable to set fragment size -- sound may be choppy\n");
+				o->warned |= WARN_frag;
+			}
 		}
-		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;
+	/* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
+	res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
+	res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
+	/* it may fail if we are in half duplex, never mind */
+	return 0;
 }
 
+/*
+ * some of the standard methods supported by channels.
+ */
 static int oss_digit(struct ast_channel *c, char digit)
 {
+	/* no better use for received digits than print them */
 	ast_verbose( " << Console Received digit %c >> \n", digit);
 	return 0;
 }
 
 static int oss_text(struct ast_channel *c, const char *text)
 {
+	/* print received messages */
 	ast_verbose( " << Console Received text %s >> \n", text);
 	return 0;
 }
 
+/* Play ringtone 'x' on device 'o' */
+static void ring(struct chan_oss_pvt *o, int x)
+{
+	write(o->sndcmd[1], &x, sizeof(x));
+}
+
+
+/*
+ * handler for incoming calls. Either autoanswer, or start ringing
+ */
 static int oss_call(struct ast_channel *c, char *dest, int timeout)
 {
-	int res = 3;
+	struct chan_oss_pvt *o = c->tech_pvt;
 	struct ast_frame f = { 0, };
-	ast_verbose( " << Call placed to '%s' on console >> \n", dest);
-	if (autoanswer) {
+
+	ast_verbose(" << Call to '%s' on console from <%s><%s><%s> >>\n",
+		dest, c->cid.cid_dnid, c->cid.cid_num, c->cid.cid_name);
+	if (o->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");
+		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));
+		ring(o, AST_CONTROL_RING);
 	}
 	return 0;
 }
 
-static void answer_sound(void)
-{
-	int res;
-	nosound = 1;
-	res = 4;
-	write(sndcmd[1], &res, sizeof(res));
-	
-}
-
+/*
+ * remote side answered the phone
+ */
 static int oss_answer(struct ast_channel *c)
 {
+	struct chan_oss_pvt *o = c->tech_pvt;
+
 	ast_verbose( " << Console call has been answered >> \n");
-	answer_sound();
+#if 0
+	/* play an answer tone (XXX do we really need it ?) */
+	ring(o, AST_CONTROL_ANSWER);
+#endif
 	ast_setstate(c, AST_STATE_UP);
-	cursound = -1;
-	nosound=0;
+	o->cursound = -1;
+	o->nosound=0;
 	return 0;
 }
 
 static int oss_hangup(struct ast_channel *c)
 {
-	int res = 0;
-	cursound = -1;
+	struct chan_oss_pvt *o = c->tech_pvt;
+
+	o->cursound = -1;
+	o->nosound = 0;
 	c->tech_pvt = NULL;
-	oss.owner = NULL;
+	o->owner = NULL;
 	ast_verbose( " << Hangup on console >> \n");
-	ast_mutex_lock(&usecnt_lock);
+	ast_mutex_lock(&usecnt_lock);	/* XXX not sure why */
 	usecnt--;
 	ast_mutex_unlock(&usecnt_lock);
-	if (hookstate) {
-		if (autoanswer) {
+	if (o->hookstate) {
+		if (o->autoanswer || o->autohangup) {
 			/* Assume auto-hangup too */
-			hookstate = 0;
+			o->hookstate = 0;
+			setformat(o, O_CLOSE);
 		} else {
 			/* Make congestion noise */
-			res = 2;
-			write(sndcmd[1], &res, sizeof(res));
+			ring(o, AST_CONTROL_CONGESTION);
 		}
 	}
 	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)
+/* used for data coming from the network */
+static int oss_write(struct ast_channel *c, struct ast_frame *f)
 {
-	int res;
-	static char sizbuf[8000];
-	static int sizpos = 0;
-	int len = sizpos;
-	int pos;
+	int src;
+	struct chan_oss_pvt *o = c->tech_pvt;
+
 	/* Immediately return if no sound is enabled */
-	if (nosound)
+	if (o->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;
+	o->cursound = -1;
+	/*
+	 * we could receive a block which is not a multiple of our
+	 * FRAME_SIZE, so buffer it locally and write to the device
+	 * in FRAME_SIZE chunks.
+	 * Keep the residue stored for future use.
+	 */
+	src = 0; /* read position into f->data */
+	while ( src < f->datalen ) {
+		/* Compute spare room in the buffer */
+		int l = sizeof(o->oss_write_buf) - o->oss_write_dst;
+
+		if (f->datalen - src >= l) {	/* enough to fill a frame */
+			memcpy(o->oss_write_buf + o->oss_write_dst,
+				f->data + src, l);
+			soundcard_writeframe(o, (short *)o->oss_write_buf);
+			src += l;
+			o->oss_write_dst = 0;
+		} else { /* copy residue */
+			l = f->datalen - src;
+			memcpy(o->oss_write_buf + o->oss_write_dst,
+				f->data + src, l);
+			src += l;	/* but really, we are done */
+			o->oss_write_dst += l;
 		}
-		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 *oss_read(struct ast_channel *c)
 {
-	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;
+	struct chan_oss_pvt *o = c->tech_pvt;
+	struct ast_frame *f = &o->read_f;
+
+	/* prepare a NULL frame in case we don't have enough data to return */
+	bzero(f, sizeof(struct ast_frame));
+	f->frametype = AST_FRAME_NULL;
+	f->src = o->type;
+
+	res = read(o->sounddev, o->oss_read_buf + o->readpos,
+	sizeof(o->oss_read_buf) - o->readpos);
+	if (res < 0)	/* audio data not ready, return a NULL frame */
+		return f;
+
+	o->readpos += res;
+	if (o->readpos < sizeof(o->oss_read_buf))	/* not enough samples */
+		return f;
+
+	if (o->mute)
+		return f;
+
+	o->readpos = AST_FRIENDLY_OFFSET;	/* reset read pointer for next frame */
+	if (c->_state != AST_STATE_UP)	/* drop data if frame is not up */
+		return f;
+	/* ok we can build and deliver the frame to the caller */
+	f->frametype = AST_FRAME_VOICE;
+	f->subclass = AST_FORMAT_SLINEAR;
+	f->samples = FRAME_SIZE;
+	f->datalen = FRAME_SIZE * 2;
+	f->data = o->oss_read_buf + AST_FRIENDLY_OFFSET;
+	f->offset = AST_FRIENDLY_OFFSET;
+	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;
+	struct chan_oss_pvt *o = newchan->tech_pvt;
+	o->owner = newchan;
 	return 0;
 }
 
-static int oss_indicate(struct ast_channel *chan, int cond)
+static int oss_indicate(struct ast_channel *c, int cond)
 {
+	struct chan_oss_pvt *o = c->tech_pvt;
 	int res;
+
 	switch(cond) {
 	case AST_CONTROL_BUSY:
-		res = 1;
-		break;
 	case AST_CONTROL_CONGESTION:
-		res = 2;
-		break;
 	case AST_CONTROL_RINGING:
-		res = 0;
+		res = cond;
 		break;
+
 	case -1:
-		cursound = -1;
+		o->cursound = -1;
 		return 0;
+
 	default:
-		ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
+		ast_log(LOG_WARNING,
+		    "Don't know how to display condition %d on %s\n",
+		    cond, c->name);
 		return -1;
 	}
-	if (res > -1) {
-		write(sndcmd[1], &res, sizeof(res));
-	}
+	if (res > -1)
+		ring(o, res);
 	return 0;	
 }
 
-static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
+/*
+ * allocate a new channel.
+ */
+static struct ast_channel *oss_new(struct chan_oss_pvt *o,
+	char *ext, char *ctx, 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;
-			}
+	struct ast_channel *c;
+
+	c = ast_channel_alloc(1);
+	if (c == NULL)
+		return NULL;
+	c->tech = &oss_tech;
+	snprintf(c->name, sizeof(c->name), "OSS/%s", o->device + 5);
+	c->type = o->type;
+	c->fds[0] = o->sounddev; /* -1 if device closed, override later */
+	c->nativeformats = AST_FORMAT_SLINEAR;
+	c->readformat = AST_FORMAT_SLINEAR;
+	c->writeformat = AST_FORMAT_SLINEAR;
+	c->tech_pvt = o;
+
+	if (ctx && !ast_strlen_zero(ctx))
+		ast_copy_string(c->context, ctx, sizeof(c->context));
+	if (ext && !ast_strlen_zero(ext))
+		ast_copy_string(c->exten, ext, sizeof(c->exten));
+	if (o->language && !ast_strlen_zero(o->language))
+		ast_copy_string(c->language, o->language, sizeof(c->language));
+
+	o->owner = c;
+	ast_setstate(c, 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(c)) {
+			ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name);
+			ast_hangup(c);
+			o->owner = c = NULL;
+			/* XXX what about the channel itself ? */
+			/* XXX what about usecnt ? */
 		}
 	}
-	return tmp;
+	return c;
 }
 
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
+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);
+	struct ast_channel *c;
+	struct chan_oss_pvt *o = find_desc(data);
+
+	ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n",
+		type, data, (char *)data);
+	if (o == NULL) {
+		ast_log(LOG_NOTICE, "Device %s not found\n", (char *)data);
+		/* XXX we could default to 'dsp' perhaps ? */
 		return NULL;
 	}
-	if (oss.owner) {
-		ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+	if ((format & AST_FORMAT_SLINEAR) == 0) {
+		ast_log(LOG_NOTICE, "Format 0x%x unsupported\n", format);
+		return NULL;
+	}
+	if (o->owner) {
+		ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner);
 		*cause = AST_CAUSE_BUSY;
 		return NULL;
 	}
-	tmp= oss_new(&oss, AST_STATE_DOWN);
-	if (!tmp) {
+	c= oss_new(o, NULL, NULL, AST_STATE_DOWN);
+	if (c == NULL) {
 		ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
+		return NULL;
 	}
-	return tmp;
+	return c;
 }
 
 static int console_autoanswer(int fd, int argc, char *argv[])
 {
-	if ((argc != 1) && (argc != 2))
-		return RESULT_SHOWUSAGE;
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
 	if (argc == 1) {
-		ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
+		ast_cli(fd, "Auto answer is %s.\n", o->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;
 	}
+	if (argc != 2)
+		return RESULT_SHOWUSAGE;
+	if (o == NULL) {
+		ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
+		    oss_active);
+		return RESULT_FAILURE;
+	}
+	if (!strcasecmp(argv[1], "on"))
+		o->autoanswer = -1;
+	else if (!strcasecmp(argv[1], "off"))
+		o->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
+	int l = strlen(word);
+
 	switch(state) {
 	case 0:
-		if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
+		if (l && !strncasecmp(word, "on", MIN(l, 2)))
 			return strdup("on");
 	case 1:
-		if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
+		if (l && !strncasecmp(word, "off", MIN(l, 3)))
 			return strdup("off");
 	default:
 		return NULL;
@@ -846,19 +967,28 @@ static char autoanswer_usage[] =
 "       argument, displays the current on/off status of autoanswer.\n"
 "       The default value of autoanswer is in 'oss.conf'.\n";
 
+/*
+ * answer command from the console
+ */
 static int console_answer(int fd, int argc, char *argv[])
 {
 	struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
 	if (argc != 1)
 		return RESULT_SHOWUSAGE;
-	if (!oss.owner) {
+	if (!o->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();
+	o->hookstate = 1;
+	o->cursound = -1;
+	o->nosound = 0;
+	ast_queue_frame(o->owner, &f);
+#if 0
+	/* XXX do we really need it ? considering we shut down immediately... */
+	ring(o, AST_CONTROL_ANSWER);
+#endif
 	return RESULT_SUCCESS;
 }
 
@@ -866,30 +996,34 @@ static char sendtext_usage[] =
 "Usage: send text <message>\n"
 "       Sends a text message for display on the remote terminal.\n";
 
+/*
+ * concatenate all arguments into a single string
+ */
 static int console_sendtext(int fd, int argc, char *argv[])
 {
+	struct chan_oss_pvt *o = find_desc(oss_active);
 	int tmparg = 2;
-	char text2send[256] = "";
+	char text2send[TEXT_SIZE] = "";
 	struct ast_frame f = { 0, };
+
 	if (argc < 2)
 		return RESULT_SHOWUSAGE;
-	if (!oss.owner) {
-		ast_cli(fd, "No one is calling us\n");
+	if (!o->owner) {
+		ast_cli(fd, "Not in a call\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);
+	while (tmparg < argc) {
+		strncat(text2send, argv[tmparg++],
+			sizeof(text2send) - strlen(text2send) - 1);
+		strncat(text2send, " ",
+			sizeof(text2send) - strlen(text2send) - 1);
 	}
-	if (strlen(text2send)) {
+	if (!ast_strlen_zero(text2send)) {
 		f.frametype = AST_FRAME_TEXT;
 		f.subclass = 0;
 		f.data = text2send;
 		f.datalen = strlen(text2send);
-		ast_queue_frame(oss.owner, &f);
+		ast_queue_frame(o->owner, &f);
 	}
 	return RESULT_SUCCESS;
 }
@@ -900,86 +1034,91 @@ static char answer_usage[] =
 
 static int console_hangup(int fd, int argc, char *argv[])
 {
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
 	if (argc != 1)
 		return RESULT_SHOWUSAGE;
-	cursound = -1;
-	if (!oss.owner && !hookstate) {
+	o->cursound = -1;
+	o->nosound = 0;
+	if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
 		ast_cli(fd, "No call to hangup up\n");
 		return RESULT_FAILURE;
 	}
-	hookstate = 0;
-	if (oss.owner) {
-		ast_queue_hangup(oss.owner);
-	}
+	o->hookstate = 0;
+	if (o->owner)
+		ast_queue_hangup(o->owner);
+	setformat(o, O_CLOSE);
 	return RESULT_SUCCESS;
 }
 
+static char hangup_usage[] =
+"Usage: hangup\n"
+"       Hangs up any call currently placed on the console.\n";
+
+
 static int console_flash(int fd, int argc, char *argv[])
 {
 	struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
 	if (argc != 1)
 		return RESULT_SHOWUSAGE;
-	cursound = -1;
-	if (!oss.owner) {
+	o->cursound = -1;
+	if (!o->owner) { /* XXX maybe !o->hookstate too ? */
 		ast_cli(fd, "No call to flash\n");
 		return RESULT_FAILURE;
 	}
-	hookstate = 0;
-	if (oss.owner) {
-		ast_queue_frame(oss.owner, &f);
-	}
+	o->hookstate = 0;
+	if (o->owner) /* XXX must be true, right ? */
+		ast_queue_frame(o->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))
+	char *s = NULL, *mye = NULL, *myc = NULL;
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
+	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");
+	if (o->owner) {	/* already in a call */
+		int i;
+		struct ast_frame f = { AST_FRAME_DTMF, 0 };
+
+		if (argc == 1) {	/* argument is mandatory here */
+			ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n");
 			return RESULT_FAILURE;
 		}
+		s = argv[1];
+		/* send the string one char at a time */
+		for (i=0; i<strlen(s); i++) {
+			f.subclass = s[i];
+			ast_queue_frame(o->owner, &f);
+		}
 		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 we have an argument split it into extension and context */
+	if (argc == 2)
+		s = ast_ext_ctx(argv[1], &mye, &myc);
+	/* supply default values if needed */
+	if (mye == NULL)
+		mye = o->ext;
+	if (myc == NULL)
+		myc = o->ctx;
 	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);
+		o->hookstate = 1;
+		oss_new(o, mye, myc, AST_STATE_RINGING);
 	} else
 		ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
+	if (s)
+		free(s);
 	return RESULT_SUCCESS;
 }
 
@@ -987,31 +1126,60 @@ static char dial_usage[] =
 "Usage: dial [extension[@context]]\n"
 "       Dials a given extensison (and context if specified)\n";
 
+static char mute_usage[] =
+"Usage: mute\nMutes the microphone\n";
+
+static char unmute_usage[] =
+"Usage: unmute\nUnmutes the microphone\n";
+
+static int console_mute(int fd, int argc, char *argv[])
+{
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	o->mute = 1;
+	return RESULT_SUCCESS;
+}
+
+static int console_unmute(int fd, int argc, char *argv[])
+{
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	o->mute = 0;
+	return RESULT_SUCCESS;
+}
+
 static int console_transfer(int fd, int argc, char *argv[])
 {
-	char tmp[256];
-	char *context;
+	struct chan_oss_pvt *o = find_desc(oss_active);
+	struct ast_channel *b = NULL;
+	char *tmp, *ext, *ctx;
+
 	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 {
+	if (o == NULL)
+		return RESULT_FAILURE;
+	if (o->owner ==NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
 		ast_cli(fd, "There is no call to transfer\n");
+		return RESULT_SUCCESS;
+	}
+
+	tmp = ast_ext_ctx(argv[1], &ext, &ctx);
+	if (ctx == NULL)		/* supply default context if needed */
+		ctx = o->owner->context;
+	if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num))
+		ast_cli(fd, "No such extension exists\n");
+	else {
+		ast_cli(fd, "Whee, transferring %s to %s@%s.\n",
+			b->name, ext, ctx);
+		if (ast_async_goto(b, ctx, ext, 1))
+			ast_cli(fd, "Failed to transfer :(\n");
 	}
+	if (tmp)
+		free(tmp);
 	return RESULT_SUCCESS;
 }
 
@@ -1020,93 +1188,211 @@ static char transfer_usage[] =
 "       Transfers the currently connected call to the given extension (and\n"
 "context if specified)\n";
 
+static int console_active(int fd, int argc, char *argv[])
+{
+	if (argc == 1)
+		ast_cli(fd, "active console is [%s]\n", oss_active);
+	else if (argc != 2)
+		return RESULT_SHOWUSAGE;
+	else {
+		struct chan_oss_pvt *o;
+		if (strcmp(argv[1], "show") == 0) {
+			for (o = oss_default.next; o ; o = o->next)
+			    ast_cli(fd, "device [%s] exists\n", o->name);
+			return RESULT_SUCCESS;
+		}
+		o = find_desc(argv[1]);
+		if (o == NULL)
+			ast_cli(fd, "No device [%s] exists\n", argv[1]);
+		else
+			oss_active = o->name;
+	}
+	return RESULT_SUCCESS;
+}
+
 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 },
+	{ { "mute", NULL }, console_mute, "Disable mic input", mute_usage },
+	{ { "unmute", NULL }, console_unmute, "Enable mic input", unmute_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 }
+	{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete },
+	{ { "console", NULL }, console_active, "Sets/displays active console",
+		"console foo sets foo as the console"}
 };
 
-int load_module()
+/*
+ * store the mixer argument from the config file, filtering possibly
+ * invalid or dangerous values (the string is used as argument for
+ * system("mixer %s")
+ */
+static void store_mixer(struct chan_oss_pvt *o, char *s)
 {
-	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;
+	int i;
+
+	for (i=0; i < strlen(s); i++) {
+		if (!isalnum(s[i]) && index(" \t-/", s[i]) == NULL) {
+			ast_log(LOG_WARNING,
+				"Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s);
+			return;
+		}
 	}
-	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");
+	if (o->mixer_cmd)
+		free(o->mixer_cmd);
+	o->mixer_cmd = strdup(s);
+	ast_log(LOG_WARNING, "setting mixer %s\n", s);
+}
+
+/*
+ * grab fields from the config file, init the descriptor and open the device.
+ */
+static struct chan_oss_pvt * store_config(struct ast_config *cfg, char *ctg)
+{
+	struct ast_variable *v;
+	struct chan_oss_pvt *o;
+
+	if (ctg == NULL) {
+		o = &oss_default;
+		ctg = "general";
+	} else {
+		o = (struct chan_oss_pvt *)malloc(sizeof *o);
+		if (o == NULL)		/* fail */
+			return NULL;
+		*o = oss_default;
+		/* "general" is also the default thing */
+		if (strcmp(ctg, "general") == 0) {
+			o->name = strdup("dsp");
+			oss_active = o->name;
+			goto openit;
 		}
-		return 0;
+		o->name = strdup(ctg);
 	}
-	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;
+
+	/* fill other fields from configuration */
+	for (v = ast_variable_browse(cfg, ctg);v; v=v->next) {
+		M_START(v->name, v->value);
+
+		M_BOOL("autoanswer", o->autoanswer)
+		M_BOOL("autohangup", o->autohangup)
+		M_BOOL("overridecontext", o->overridecontext)
+		M_STR("device", o->device)
+		M_UINT("frags", o->frags)
+		M_UINT("debug", oss_debug)
+		M_UINT("queuesize", o->queuesize)
+		M_STR("context", o->ctx)
+		M_STR("language", o->language)
+		M_STR("extension", o->ext)
+		M_F("mixer", store_mixer(o, v->value))
+		M_END(;);
+	}
+	if (ast_strlen_zero(o->device))
+		ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
+	if (o->mixer_cmd) {
+		char *cmd;
+
+		asprintf(&cmd, "mixer %s", o->mixer_cmd);
+		ast_log(LOG_WARNING, "running [%s]\n", cmd);
+		system(cmd);
+		free(cmd);
 	}
-	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;
+	if (o == &oss_default)	/* we are done with the default */
+		return NULL;
+
+openit:
+#if TRYOPEN
+	if (setformat(o, O_RDWR) < 0) {	/* open device */
+		if (option_verbose > 0) {
+			ast_verbose(VERBOSE_PREFIX_2 "Device %s not detected\n", ctg);
+			ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding "
+			    "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
 		}
+		goto error;
+	}
+	if (o->duplex != M_FULL)
+		ast_log(LOG_WARNING, "XXX I don't work right with non "
+			"full-duplex sound cards XXX\n");
+#endif /* TRYOPEN */
+	if (pipe(o->sndcmd) != 0) {
+		ast_log(LOG_ERROR, "Unable to create pipe\n");
+		goto error;
+	}
+	ast_pthread_create(&o->sthread, NULL, sound_thread, o);
+	/* link into list of devices */
+	if (o != &oss_default) {
+		o->next = oss_default.next;
+		oss_default.next = o;
+	}
+	return o;
+
+error:
+	if (o != &oss_default)
+		free(o);
+	return NULL;
+}
+
+int load_module(void)
+{
+	int i;
+	struct ast_config *cfg;
+
+	/* load config file */
+	cfg = ast_config_load(config);
+	if (cfg != NULL) {
+		char *ctg = NULL;	/* first pass is 'general' */
+
+		do {
+			store_config(cfg, ctg);
+		} while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
 		ast_config_destroy(cfg);
 	}
-	ast_pthread_create(&sthread, NULL, sound_thread, NULL);
+	if (find_desc(oss_active) == NULL) {
+		ast_log(LOG_NOTICE, "Device %s not found\n", oss_active);
+		/* XXX we could default to 'dsp' perhaps ? */
+		/* XXX should cleanup allocated memory etc. */
+		return -1;
+	}
+	i = ast_channel_register(&oss_tech);
+	if (i < 0) {
+		ast_log(LOG_ERROR, "Unable to register channel class '%s'\n",
+			oss_default.type);
+		/* XXX should cleanup allocated memory etc. */
+		return -1;
+	}
+	ast_cli_register_multiple(myclis, sizeof(myclis)/sizeof(struct ast_cli_entry));
 	return 0;
 }
 
 
-
 int unload_module()
 {
-	int x;
+	struct chan_oss_pvt *o;
 
 	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]);
+	ast_cli_unregister_multiple(myclis,
+		sizeof(myclis)/sizeof(struct ast_cli_entry));
+
+	for (o = oss_default.next; o ; o = o->next) {
+		close(o->sounddev);
+		if (o->sndcmd[0] > 0) {
+			close(o->sndcmd[0]);
+			close(o->sndcmd[1]);
+		}
+		if (o->owner)
+			ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
+		if (o->owner) /* XXX how ??? */
+			return -1;
+		/* XXX what about the thread ? */
+		/* XXX what about the memory allocated ? */
 	}
-	if (oss.owner)
-		ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
-	if (oss.owner)
-		return -1;
 	return 0;
 }
 
 char *description()
 {
-	return (char *) desc;
+	return (char *)oss_tech.description;
 }
 
 int usecount()