5 files changed, 1370 insertions, 1178 deletions

diff --git a/net/asterisk-devel/Makefile b/net/asterisk-devel/Makefile
index bf9289fc189c..00cb6ce94061 100644
--- a/net/asterisk-devel/Makefile
+++ b/net/asterisk-devel/Makefile
@@ -7,7 +7,7 @@
 
 PORTNAME=	asterisk
 PORTVERSION=	1.0.7
-PORTREVISION=	2
+PORTREVISION=	3
 CATEGORIES=	net
 MASTER_SITES=	ftp://ftp.asterisk.org/pub/telephony/asterisk/ \
 		ftp://ftp.asterisk.org/pub/telephony/asterisk/old-releases/
@@ -20,12 +20,10 @@ PATCH_DIST_STRIP=	-p1
 MAINTAINER=	sobomax@FreeBSD.org
 COMMENT=	An Open Source PBX and telephony toolkit
 
-BUILD_DEPENDS=	${LOCALBASE}/lib/mysql/libmysqlclient.a:${PORTSDIR}/databases/mysql40-client \
-		mpg123:${PORTSDIR}/audio/mpg123
+BUILD_DEPENDS=	mpg123:${PORTSDIR}/audio/mpg123
 LIB_DEPENDS=	speex.3:${PORTSDIR}/audio/speex \
 		newt.51:${PORTSDIR}/devel/newt
-RUN_DEPENDS=	${LOCALBASE}/lib/mysql/libmysqlclient.a:${PORTSDIR}/databases/mysql40-client \
-		mpg123:${PORTSDIR}/audio/mpg123
+RUN_DEPENDS=	mpg123:${PORTSDIR}/audio/mpg123
 
 ONLY_FOR_ARCHS=	i386 sparc64
 
@@ -71,4 +69,12 @@ RUN_DEPENDS+=	${LOCALBASE}/include/zaptel.h:${PORTSDIR}/misc/zaptel
 PLIST_SUB+=	WITH_ZAPTEL=""
 .endif
 
+.if !defined(WITHOUT_MYSQL)
+BUILD_DEPENDS+=	${LOCALBASE}/lib/mysql/libmysqlclient.a:${PORTSDIR}/databases/mysql40-client
+RUN_DEPENDS+=	${LOCALBASE}/lib/mysql/libmysqlclient.a:${PORTSDIR}/databases/mysql40-client
+.endif
+
+post-patch:
+	${CP} ${FILESDIR}/chan_oss.c ${WRKSRC}/channels
+
 .include <bsd.port.post.mk>
diff --git a/net/asterisk-devel/files/chan_oss.c b/net/asterisk-devel/files/chan_oss.c
new file mode 100644
index 000000000000..aef0db9dca85
--- /dev/null
+++ b/net/asterisk-devel/files/chan_oss.c
@@ -0,0 +1,1320 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * 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
+ *
+ * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.04.26
+ * note-this code best seen with ts=8 (8-spaces tabs) in the editor
+ */
+
+#include <asterisk/lock.h>
+#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 <asterisk/utils.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 <ctype.h>	/* for isalnum */
+#ifdef __linux
+#include <linux/soundcard.h>
+#elif defined(__FreeBSD__)
+#include <sys/soundcard.h>
+#else
+#include <soundcard.h>
+#endif
+#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
+
+/*
+ * Basic mode of operation:
+ *
+ * we have one keyboard (which receives commands from the keyboard)
+ * and multiple headset's connected to audio cards. Headsets are named as
+ * the sections of oss.conf
+ *
+ * At any time, the keyboard is attached to one headset, and you
+ * can switch among them using the 'console' command.
+ *
+ * The following parameters are important for the configuration of
+ * the device:
+ *
+ *  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
+
+
+/* Don't switch between read/write modes faster than every 300 ms */
+#define MIN_SWITCH_TIME 300
+
+
+static int usecnt;
+AST_MUTEX_DEFINE_STATIC(usecnt_lock);
+
+static char *desc = "OSS Console Channel Driver";
+static char *tdesc = "OSS Console Channel Driver";
+static char *config = "oss.conf";	/* default config file */
+
+
+/*
+ * 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;
+	int silencelen;
+	int repeat;
+};
+
+static struct sound sounds[] = {
+    { 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 */
+};
+
+
+/*
+ * 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;
+    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;
+    struct timeval lasttime;	/* last setformat */
+    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 */
+
+    int silencesuppression;
+    int silencethreshold;
+	char device[64];	/* device to open */
+
+    pthread_t sthread;
+
+    struct ast_channel *owner;
+    char ext[AST_MAX_EXTENSION];
+    char ctx[AST_MAX_EXTENSION];
+    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,
+	.silencethreshold = 1000,	/* currently unused */
+	.ext = "s",
+	.ctx = "default",
+	.readpos = AST_FRIENDLY_OFFSET,	/* start here on reads */
+};
+
+static char *oss_active;	 /* the active device */
+
+/*
+ * returns true if too early to switch
+ */
+static int too_early(struct chan_oss_pvt *o)
+{
+	struct timeval tv;
+	int ms;
+	gettimeofday(&tv, NULL);
+	ms = (tv.tv_sec - o->lasttime.tv_sec) * 1000 +
+			(tv.tv_usec - o->lasttime.tv_usec) / 1000;
+	if (ms < MIN_SWITCH_TIME)
+		return -1;
+	return 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;
+
+    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;
+}
+
+static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
+{	
+    /* Write an exactly FRAME_SIZE sized frame */
+    int res;
+
+    /*
+     * 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 && 0)
+	    ast_log(LOG_WARNING, "write: used %d blocks (%d)\n",
+		res, o->w_errors);
+	return 0;
+    }
+    o->w_errors = 0;
+    res = write(o->sounddev, ((void *)data), FRAME_SIZE * 2);
+    return res;
+}
+
+/*
+ * 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 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;	/* sound available */
+	    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 { /* no sound, maybe some silence */
+		static 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;
+		}
+	    }
+	}
+	l = soundcard_writeframe(o, myframe);
+	if (l > 0)
+	    o->sampsent = l_sampsent;	/* update status */
+}
+
+static void *sound_thread(void *arg)
+{
+    char ign[4096];
+    struct chan_oss_pvt *o = (struct chan_oss_pvt *)arg;
+
+    /* kick the driver by trying to read from it. Ignore errors */
+    if (read(o->sounddev, ign, sizeof(ign)) < 0)
+	ast_log(LOG_WARNING, "Read error on sound device: %s\n",
+			strerror(errno));
+    for(;;) {
+	fd_set rfds, wfds;
+	int maxfd, res;
+
+	FD_ZERO(&rfds);
+	FD_ZERO(&wfds);
+	maxfd = o->sndcmd[0];	/* pipe from the main process */
+	FD_SET(o->sndcmd[0], &rfds);
+	if (!o->owner) { /* no one owns the audio, so we must drain it */
+	    FD_SET(o->sounddev, &rfds);
+	    if (o->sounddev > maxfd)
+		maxfd = o->sounddev;
+	}
+	if (o->cursound > -1) {
+	    FD_SET(o->sounddev, &wfds);
+	    if (o->sounddev > maxfd)
+		maxfd = o->sounddev;
+	}
+	/* 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));
+	    continue;
+	}
+	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(o->sounddev, &rfds)) { /* read and ignore errors */
+	    read(o->sounddev, ign, sizeof(ign));
+	}
+	if (FD_ISSET(o->sounddev, &wfds))
+	    send_sound(o);
+    }
+    /* Never reached */
+    return NULL;
+}
+
+#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;
+}
+
+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 (!oss.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(oss.sounddev, silbuf, silbufcnt * FRAME_SIZE);
+			silbufcnt = 0;
+		}
+	}
+	return 0;
+}
+#endif
+
+/*
+ * 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)
+{
+    int fmt, desired, res, fd;
+
+    if (o->sounddev >= 0) {
+	ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
+	close(o->sounddev);
+	o->duplex = M_UNSET;
+    }
+    fd = o->sounddev = open(o->device, mode |O_NONBLOCK);
+    if (o->sounddev < 0) {
+	ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n",
+		strerror(errno));
+	return -1;
+    }
+
+    gettimeofday(&o->lasttime, NULL);
+    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");
+	return -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;
+    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 (!(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;
+	}
+    }
+    /*
+     * 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;
+	    }
+	}
+    }
+    /* XXX 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;
+}
+
+/*
+ * make sure output mode is available. Returns 0 if done,
+ * 1 if too early to switch, -1 if error
+ */
+static int soundcard_setoutput(struct chan_oss_pvt *o, int force)
+{
+    if (o->duplex == M_FULL || (o->duplex == M_WRITE && !force))
+	return 0;
+    if (!force && too_early(o))
+	return 1;
+    if (setformat(o, O_WRONLY))
+	return -1;
+    return 0;
+}
+
+/*
+ * make sure input mode is available. Returns 0 if done
+ * 1 if too early to switch, -1 if error
+ */
+static int soundcard_setinput(struct chan_oss_pvt *o, int force)
+{
+    if (o->duplex == M_FULL || (o->duplex == M_READ && !force))
+	return 0;
+    if (!force && too_early(o))
+	return 1;
+    if (setformat(o, O_RDONLY))
+	return -1;
+    return 0;
+}
+
+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, char *text)
+{
+    ast_verbose( " << Console Received text %s >> \n", text);
+    return 0;
+}
+
+/* request to play a sound on the speaker XXX fix oss. */
+#define	RING(o, x) { int what = x; write((o)->sndcmd[1], &what, sizeof(what)); }
+
+static int oss_call(struct ast_channel *c, char *dest, int timeout)
+{
+    struct chan_oss_pvt *o = c->pvt->pvt;
+    struct ast_frame f = { 0, };
+
+    ast_verbose( " << Call placed to '%s' on console >> \n", dest);
+    if (o->autoanswer) {
+	ast_verbose( " << Auto-answered >> \n" );
+	f.frametype = AST_FRAME_CONTROL;
+	f.subclass = AST_CONTROL_ANSWER;
+	ast_queue_frame(c, &f);
+    } else {
+	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);
+	RING(o, AST_CONTROL_RING);
+    }
+    return 0;
+}
+
+static void answer_sound(struct chan_oss_pvt *o)
+{
+    RING(o, AST_CONTROL_ANSWER);
+}
+
+static int oss_answer(struct ast_channel *c)
+{
+    struct chan_oss_pvt *o = c->pvt->pvt;
+
+    ast_verbose( " << Console call has been answered >> \n");
+    answer_sound(o);	/* XXX do we really need it ? considering we shut down immediately... */
+    ast_setstate(c, AST_STATE_UP);
+    o->cursound = -1;
+    o->nosound=0;
+    return 0;
+}
+
+static int oss_hangup(struct ast_channel *c)
+{
+    struct chan_oss_pvt *o = c->pvt->pvt;
+
+    o->cursound = -1;
+    c->pvt->pvt = NULL;
+    o->owner = NULL;
+    ast_verbose( " << Hangup on console >> \n");
+    ast_mutex_lock(&usecnt_lock);	/* XXX not sure why */
+    usecnt--;
+    ast_mutex_unlock(&usecnt_lock);
+    if (o->hookstate) {
+	if (o->autoanswer || o->autohangup) {
+	    /* Assume auto-hangup too */
+	    o->hookstate = 0;
+	} else {
+	    /* Make congestion noise */
+	    RING(o, AST_CONTROL_CONGESTION);
+	}
+    }
+    return 0;
+}
+
+/* used for data coming from the network */
+static int oss_write(struct ast_channel *c, struct ast_frame *f)
+{
+    int res;
+    int src;
+    struct chan_oss_pvt *o = c->pvt->pvt;
+
+    /* Immediately return if no sound is enabled */
+    if (o->nosound)
+	return 0;
+    /* Stop any currently playing sound */
+    o->cursound = -1;
+    if (o->duplex != M_FULL) {
+	/* XXX check this, looks weird! */
+	/* If we're half duplex, we have to switch to read mode
+	   to honor immediate needs if necessary */
+	res = soundcard_setinput(o, 1); /* force set if not full_duplex */
+	if (res < 0) {
+	    ast_log(LOG_WARNING, "Unable to set device to input mode\n");
+	    return -1;
+	}
+	return 0;
+    }
+    res = soundcard_setoutput(o, 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 could receive a sample which is not a multiple of our FRAME_SIZE,
+     * so we buffer it locally and write to the device in FRAME_SIZE
+     * chunks, keeping 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;
+}
+
+static struct ast_frame *oss_read(struct ast_channel *c)
+{
+    /* XXX if we want multiple devices, should move these static vars
+     * into the device descriptor
+     */
+    int res;
+    struct chan_oss_pvt *o = c->pvt->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 = soundcard_setinput(o, 0);
+    if (res < 0) {
+	ast_log(LOG_WARNING, "Unable to set input mode\n");
+	return NULL;
+    } else if (res > 0) {	/* too early to switch ? */
+	/* Theoretically shouldn't happen, but anyway, return a NULL frame */
+	return f;
+    }
+
+    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;
+
+    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 *o = newchan->pvt->pvt;
+	o->owner = newchan;
+	return 0;
+}
+
+static int oss_indicate(struct ast_channel *c, int cond)
+{
+    struct chan_oss_pvt *o = c->pvt->pvt;
+    int res;
+
+    switch(cond) {
+    case AST_CONTROL_BUSY:
+    case AST_CONTROL_CONGESTION:
+    case AST_CONTROL_RINGING:
+	res = cond;
+	break;
+    case -1:
+	o->cursound = -1;
+	return 0;
+    default:
+	ast_log(LOG_WARNING,
+		"Don't know how to display condition %d on %s\n",
+		cond, c->name);
+	return -1;
+    }
+    if (res > -1)
+	RING(o, res);
+    return 0;	
+}
+
+static struct ast_channel *oss_new(struct chan_oss_pvt *o,
+	char *ext, char *ctx, int state)
+{
+    struct ast_channel *c;
+    struct ast_channel_pvt *pvt;
+
+    c = ast_channel_alloc(1);
+    if (c == NULL)
+	return NULL;
+    snprintf(c->name, sizeof(c->name), "OSS/%s", o->device + 5);
+    c->type = o->type;
+    c->fds[0] = o->sounddev;
+    c->nativeformats = AST_FORMAT_SLINEAR;
+    pvt = c->pvt;
+    pvt->pvt = o;
+
+    /* relevant callbacks */
+    pvt->send_digit = oss_digit;
+    pvt->send_text = oss_text;
+    pvt->hangup = oss_hangup;
+    pvt->answer = oss_answer;
+    pvt->read = oss_read;
+    pvt->call = oss_call;
+    pvt->write = oss_write;
+    pvt->indicate = oss_indicate;
+    pvt->fixup = oss_fixup;
+
+    if (strlen(ctx))
+	strncpy(c->context, ctx, sizeof(o->ctx)-1);
+    if (strlen(ext))
+	strncpy(c->exten, ext, sizeof(o->ext)-1);
+    if (strlen(o->language))
+	strncpy(c->language, o->language, sizeof(o->language)-1);
+    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 c;
+}
+
+/*
+ * returns a pointer to the descriptor with the given name
+ */
+static struct chan_oss_pvt *find_desc(char *dev)
+{
+    struct chan_oss_pvt *o;
+
+    for (o = oss_default.next; o && strcmp(o->name, dev) != 0; o = o->next)
+	;
+    if (o == NULL)
+	ast_log(LOG_WARNING, "%s could not find <%s>\n", __func__, dev);
+    return o;
+}
+
+static struct ast_channel *oss_request(char *type, int format, void *data)
+{
+    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 ((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 on the OSS channel\n");
+	return NULL;
+    }
+    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 c;
+}
+
+static int console_autoanswer(int fd, int argc, char *argv[])
+{
+    struct chan_oss_pvt *o = find_desc(oss_active);
+
+    if ((argc != 1) && (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 (argc == 1) {
+	ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
+	return RESULT_SUCCESS;
+    }
+    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 (l && !strncasecmp(word, "on", MIN(l, 2)))
+			return strdup("on");
+	case 1:
+		if (l && !strncasecmp(word, "off", MIN(l, 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 chan_oss_pvt *o = find_desc(oss_active);
+
+	struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	if (!o->owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	o->hookstate = 1;
+	o->cursound = -1;
+	ast_queue_frame(o->owner, &f);
+	answer_sound(o);
+	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[])
+{
+	struct chan_oss_pvt *o = find_desc(oss_active);
+	int tmparg = 2;
+	char text2send[256] = "";
+	struct ast_frame f = { 0, };
+
+	if (argc < 2)
+		return RESULT_SHOWUSAGE;
+	if (!o->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(o->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[])
+{
+	struct chan_oss_pvt *o = find_desc(oss_active);
+
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	o->cursound = -1;
+	if (!o->owner && !o->hookstate) {
+		ast_cli(fd, "No call to hangup up\n");
+		return RESULT_FAILURE;
+	}
+	o->hookstate = 0;
+	if (o->owner) {
+		ast_queue_hangup(o->owner);
+	}
+	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 = NULL, *mye = NULL, *myc = NULL;
+    int i;
+    struct ast_frame f = { AST_FRAME_DTMF, 0 };
+    struct chan_oss_pvt *o = find_desc(oss_active);
+
+    if ((argc != 1) && (argc != 2))
+	return RESULT_SHOWUSAGE;
+    if (o->owner) {	/* already in a call */
+	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;
+	}
+	mye = argv[1];
+	/* send the string one char at a time */
+	for (i=0; i<strlen(mye); i++) {
+	    f.subclass = mye[i];
+	    ast_queue_frame(o->owner, &f);
+	}
+	return RESULT_SUCCESS;
+    }
+    /* if we have an argument split it into extension and context */
+    if (argc == 2) {
+	tmp = myc = strdup(argv[1]); /* make a writable copy */
+	mye = strsep(&myc, "@");	/* set exten, advance to context */
+	myc = strsep(&myc, "@");	/* set context */
+    }
+    /* 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)) {
+	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);
+    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[])
+{
+    struct chan_oss_pvt *o = find_desc(oss_active);
+    struct ast_channel *b;
+
+    char *ext, *ctx;
+
+    if (argc != 2)
+	return RESULT_SHOWUSAGE;
+    if (o == NULL)
+	return RESULT_FAILURE;
+    if (! (o->owner && o->owner->bridge)) {
+	ast_cli(fd, "There is no call to transfer\n");
+	return RESULT_SUCCESS;
+    }
+    b = o->owner->bridge;
+
+    ext = ctx = strdup(argv[1]);	/* make a writable copy */
+    strsep(&ctx, "@");			/* set exten, advance to context */
+    ctx = strsep(&ctx, "@");		/* strip trailing @ and the rest */
+
+    if (ctx == NULL)			/* supply default context if needed */
+	ctx = o->owner->context;
+    if (!ast_exists_extension(b, ctx, ext, 1, b->callerid)) {
+	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");
+    }
+    free(ext);
+    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 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 },
+	{ { "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 },
+	{ { "console", NULL }, console_active, "Sets/displays active console",
+		"console foo sets foo as the console"}
+};
+
+/*
+ * 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 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;
+	}
+    }
+    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;
+	o->next = NULL; /* XXX needed ? */
+	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;
+	}
+	o->name = strdup(ctg);
+    }
+    ast_log(LOG_WARNING, "found category [%s]\n", ctg);
+
+    /* fill other fields from configuration */
+    v = ast_variable_browse(cfg, ctg);
+    while(v) {
+	if (!strcasecmp(v->name, "autoanswer"))
+	    o->autoanswer = ast_true(v->value);
+	else if (!strcasecmp(v->name, "autohangup"))
+	    o->autohangup = ast_true(v->value);
+	else if (!strcasecmp(v->name, "silencesuppression"))
+	    o->silencesuppression = ast_true(v->value);
+	else if (!strcasecmp(v->name, "silencethreshold"))
+	    o->silencethreshold = atoi(v->value);
+	else if (!strcasecmp(v->name, "device"))
+	    strncpy(o->device, v->value, sizeof(o->device)-1);
+	else if (!strcasecmp(v->name, "frags"))
+	    o->frags = strtoul(v->value, NULL, 0);
+	else if (!strcasecmp(v->name, "queuesize"))
+	    o->queuesize = strtoul(v->value, NULL, 0);
+	else if (!strcasecmp(v->name, "context"))
+	    strncpy(o->ctx, v->value, sizeof(o->ctx)-1);
+	else if (!strcasecmp(v->name, "language"))
+	    strncpy(o->language, v->value, sizeof(o->language)-1);
+	else if (!strcasecmp(v->name, "extension"))
+	    strncpy(o->ext, v->value, sizeof(o->ext)-1);
+	else if (!strcasecmp(v->name, "mixer"))
+	    store_mixer(o, v->value);
+	v=v->next;
+    }
+    if (!strlen(o->device))
+	strncpy(o->device, DEV_DSP, sizeof(o->device)-1);
+    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);
+    }
+    if (o == &oss_default)	/* we are done with the default */
+	return NULL;
+
+openit:
+    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;
+    }
+    soundcard_setinput(o, 1); /* force set if not full_duplex */
+    if (o->duplex != M_FULL)
+	ast_log(LOG_WARNING, "XXX I don't work right with non "
+		"full-duplex sound cards XXX\n");
+    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()
+{
+    int i;
+    struct ast_config *cfg;
+
+    /* load config file */
+    cfg = ast_load(config);
+    if (cfg != NULL) {
+	char *ctg;
+
+	store_config(cfg, NULL);	/* init general category */
+	ctg = ast_category_browse(cfg, NULL); /* initial category */
+	while (ctg != NULL) {
+	    store_config(cfg, ctg);
+	    ctg = ast_category_browse(cfg, ctg);
+	}
+	ast_destroy(cfg);
+    }
+    i = ast_channel_register(oss_default.type, tdesc,
+		AST_FORMAT_SLINEAR, oss_request);
+    if (i < 0) {
+	ast_log(LOG_ERROR, "Unable to register channel class '%s'\n",
+		oss_default.type);
+	return NULL;
+    }
+    for (i=0; i<sizeof(myclis)/sizeof(struct ast_cli_entry); i++)
+	    ast_cli_register(myclis + i);
+    return 0;
+}
+
+
+int unload_module()
+{
+    int x;
+    struct chan_oss_pvt *o;
+
+    for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+	ast_cli_unregister(myclis + x);
+
+    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 ? */
+    }
+    return 0;
+}
+
+char *description()
+{
+	return desc;
+}
+
+int usecount()	/* XXX is this per-device or global for the module ? */
+{
+	int res;
+	ast_mutex_lock(&usecnt_lock);
+	res = usecnt;
+	ast_mutex_unlock(&usecnt_lock);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
diff --git a/net/asterisk-devel/files/patch-channels::chan_oss.c b/net/asterisk-devel/files/patch-channels::chan_oss.c
deleted file mode 100644
index ef8cfc11d711..000000000000
--- a/net/asterisk-devel/files/patch-channels::chan_oss.c
+++ /dev/null
@@ -1,1167 +0,0 @@
-
-$FreeBSD$
-
---- channels/chan_oss.c
-+++ channels/chan_oss.c
-@@ -13,6 +13,8 @@
-  *
-  * This program is free software, distributed under the terms of
-  * the GNU General Public License
-+ *
-+ * FreeBSD changes by Luigi Rizzo, 2005.04.18
-  */
- 
- #include <asterisk/lock.h>
-@@ -54,21 +56,30 @@
- #endif
- 
- /* Lets use 160 sample frames, just like GSM.  */
--#define FRAME_SIZE 160
-+/* this corresponds to 20ms of audio. */
-+#define FRAME_SIZE 160 // was 160
- 
--/* When you set the frame size, you have to come up with
--   the right buffer format as well. */
-+/*
-+ * When you set the frame size, you have to come up with
-+ * the right buffer format as well.
-+ * OSS lets you define a 'block' size (which should be a power of 2,
-+ * which power is specified in the lower 16 bits) and the number of
-+ * blocks allowed in the buffer (to avoid that the queue grows too large).
-+ * The latter is specified in the top 16 bits.
-+ * We use a block of 64 bytes (0x6), 5 blocks make a frame each sample
-+ * being 2 bytes, and we make room to store two buffers.
-+ * XXX the '10' is magic
-+ */
-+
-+#define N_BLOCKS	(buffersize * 5 * 2)
- /* 5 64-byte frames = one frame */
--#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
-+#define BUFFER_FMT (N_BLOCKS << 16) | (0x0006);
- 
- /* Don't switch between read/write modes faster than every 300 ms */
--#define MIN_SWITCH_TIME 600
-+#define MIN_SWITCH_TIME 300
- 
--static struct timeval lasttime;
- 
- static int usecnt;
--static int silencesuppression = 0;
--static int silencethreshold = 1000;
- 
- 
- AST_MUTEX_DEFINE_STATIC(usecnt_lock);
-@@ -78,16 +89,15 @@
- static char *tdesc = "OSS Console Channel Driver";
- static char *config = "oss.conf";
- 
--static char context[AST_MAX_EXTENSION] = "default";
-+static char default_context[AST_MAX_EXTENSION] = "default";
- static char language[MAX_LANGUAGE] = "";
--static char exten[AST_MAX_EXTENSION] = "s";
-+static char oss_exten[AST_MAX_EXTENSION] = "s";
- 
--static int hookstate=0;
- 
--static short silence[FRAME_SIZE] = {0, };
- 
- struct sound {
- 	int ind;
-+	char *desc;
- 	short *data;
- 	int datalen;
- 	int samplen;
-@@ -96,136 +106,178 @@
- };
- 
- 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. */
-+    /*
-+     * 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;
-+
-+    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;
-+    struct timeval lasttime;	/* last setformat */
-+
-+    int silencesuppression;
-+    int silencethreshold;
-+	char device[64];	/* device to open */
-+
-+    pthread_t sthread;
-+
- 	struct ast_channel *owner;
- 	char exten[AST_MAX_EXTENSION];
- 	char context[AST_MAX_EXTENSION];
--} oss;
-+} oss = {
-+	.cursound = -1,
-+	.sounddev = -1,
-+	.duplex = M_UNSET, /* XXX check this */
-+	.autoanswer = 1,
-+	.autohangup = 1,
-+	.silencethreshold = 1000,
-+};
- 
--static int time_has_passed(void)
-+/*
-+ * returns true if too early to switch
-+ */
-+static int too_early(struct chan_oss_pvt *o)
- {
- 	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)
-+	ms = (tv.tv_sec - o->lasttime.tv_sec) * 1000 +
-+			(tv.tv_usec - o->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)
-+/*
-+ * Returns the number of blocks used in the audio output channel
-+ */
-+static int
-+used_blocks(struct chan_oss_pvt *o)
- {
--	int sum = 0;
--	int x;
--	for (x=0;x<FRAME_SIZE;x++) {
--		if (frame[x] < 0)
--			sum -= frame[x];
--		else
--			sum += frame[x];
-+    struct audio_buf_info info;
-+    static int warned=0;
-+    if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
-+	if (!warned) {
-+	    ast_log(LOG_WARNING, "Error reading output space\n");
-+	    warned++;
- 	}
--	sum = sum/FRAME_SIZE;
--	return sum;
-+	return 1;
-+    }
-+    if (o->total_blocks == 0) {
-+	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;
- }
--#endif
- 
--static int cursound = -1;
--static int sampsent = 0;
--static int silencelen=0;
--static int offset=0;
--static int nosound=0;
-+static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
-+{	
-+	/* Write an exactly FRAME_SIZE sized of frame */
-+	int res;
-+	static int errors = 0;
- 
--static int send_sound(void)
-+	/*
-+	 * nothing spectacular.
-+	 * If the buffer is full just drop the extra, otherwise write
-+	 */
-+	res = used_blocks(o);
-+	if (res > 10) {	/* no room to write a block */
-+	    errors ++;
-+	    if (errors == 0)
-+	    	ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, errors);
-+	    return 0;
-+	}
-+	errors = 0;
-+	res = write(o->sounddev, ((void *)data), FRAME_SIZE * 2);
-+	return res;
-+}
-+
-+/*
-+ * handler for 'sound writable' events from the sound thread.
-+ * Builds a frame from the high level description of the sounds,
-+ * (tone+silence) and passes it to the audio device.
-+ */
-+static int 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 = 0;
-+	int l_sampsent = o->sampsent;
-+	int l;
-+	struct sound *s;
-+
-+	if (o->cursound < 0)	/* no sound to send */
-+		return 0;
-+	s = &sounds[o->cursound];
-+	/*
-+	 * prepare a frame
-+	 */
-+	
-+	for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
-+	    /* take chunks of sound and data until the buffer is full */
-+	    l = s->samplen - l_sampsent;	/* sound available */
-+	    if (l > 0) {
-+		if (l > FRAME_SIZE - ofs)
-+		    l = FRAME_SIZE - ofs;
-+		if (0)
-+		ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n",
-+			l_sampsent, l, s->samplen, ofs);
-+		bcopy(s->data + l_sampsent, myframe + ofs, l*2);
-+		l_sampsent += l;
-+	    } else { /* no sound, maybe some silence */
-+		static short silence[FRAME_SIZE] = {0, };
-+
-+	        l += s->silencelen;
-+		if (l > 0) {
-+		    if (l > FRAME_SIZE - ofs)
-+			l = FRAME_SIZE - ofs;
-+		    if (0)
-+		    ast_log(LOG_WARNING, "send_sound silence %d/%d of %d into %d\n",
-+			l_sampsent - s->samplen, l, s->silencelen, ofs);
-+		    bcopy(silence, myframe + ofs, l*2);
-+		    l_sampsent += l;
-+		} else { /* silence is over, restart sound if loop */
-+		    if (s->repeat == 0) {	/* last block */
-+			ast_log(LOG_WARNING, "send_sound last block\n");
-+			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 */
-+	return 0;	/* fake success */
- }
- 
- static void *sound_thread(void *unused)
-@@ -235,41 +287,53 @@
- 	int max;
- 	int res;
- 	char ign[4096];
--	if (read(sounddev, ign, sizeof(sounddev)) < 0)
-+	if (read(oss.sounddev, ign, sizeof(ign)) < 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);
-+		max = oss.sndcmd[0];
-+		FD_SET(oss.sndcmd[0], &rfds);
- 		if (!oss.owner) {
--			FD_SET(sounddev, &rfds);
--			if (sounddev > max)
--				max = sounddev;
-+			FD_SET(oss.sounddev, &rfds);
-+			if (oss.sounddev > max)
-+				max = oss.sounddev;
- 		}
--		if (cursound > -1) {
--			FD_SET(sounddev, &wfds);
--			if (sounddev > max)
--				max = sounddev;
-+		if (oss.cursound > -1) {
-+			FD_SET(oss.sounddev, &wfds);
-+			if (oss.sounddev > max)
-+				max = oss.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(oss.sndcmd[0], &rfds)) { /* read which sound to play from the pipe */
-+			int i, what;
-+
-+			read(oss.sndcmd[0], &what, sizeof(what));
-+			for (i = 0; sounds[i].ind != -1; i++)
-+				if (sounds[i].ind == what) {
-+					oss.cursound = i;
-+					oss.sampsent = 0;
-+					oss.nosound = 1; /* block other audio */
-+					ast_log(LOG_WARNING, "play %s\n", sounds[i].desc);
-+					break;
-+				}
-+			if (sounds[i].ind == -1)
-+				oss.cursound = -1;
-+			ast_log(LOG_WARNING, "cursound %d samplen %d silencelen %d\n",
-+				oss.cursound, oss.cursound >=0 ? sounds[oss.cursound].samplen : -1,
-+				oss.cursound >=0 ?  sounds[oss.cursound].silencelen : -1);
- 		}
--		if (FD_ISSET(sounddev, &rfds)) {
-+		if (FD_ISSET(oss.sounddev, &rfds)) {
- 			/* Ignore read */
--			if (read(sounddev, ign, sizeof(ign)) < 0)
-+			if (read(oss.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())
-+		if (FD_ISSET(oss.sounddev, &wfds))
-+			if (send_sound(&oss) < 0)
- 				ast_log(LOG_WARNING, "Failed to write sound\n");
- 	}
- 	/* Never reached */
-@@ -277,6 +341,20 @@
- }
- 
- #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;
-+}
-+
- static int silence_suppress(short *buf)
- {
- #define SILBUF 3
-@@ -284,7 +362,7 @@
- 	static int silentframes = 0;
- 	static char silbuf[FRAME_SIZE * 2 * SILBUF];
- 	static int silbufcnt=0;
--	if (!silencesuppression)
-+	if (!oss.silencesuppression)
- 		return 0;
- 	loudness = calc_loudness((short *)(buf));
- 	if (option_debug)
-@@ -309,7 +387,7 @@
- 		/* Write any buffered silence we have, it may have something
- 		   important */
- 		if (silbufcnt) {
--			write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
-+			write(oss.sounddev, silbuf, silbufcnt * FRAME_SIZE);
- 			silbufcnt = 0;
- 		}
- 	}
-@@ -317,27 +395,55 @@
- }
- #endif
- 
--static int setformat(void)
-+/*
-+ * 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)
- {
--	int fmt, desired, res, fd = sounddev;
-+	int fmt, desired, res, fd;
- 	static int warnedalready = 0;
- 	static int warnedalready2 = 0;
-+
-+	if (o->sounddev >= 0) {
-+		ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
-+		close(o->sounddev);
-+		o->duplex = M_UNSET;
-+	}
-+	fd = o->sounddev = open(o->device, mode |O_NONBLOCK);
-+	if (o->sounddev < 0) {
-+		ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n",
-+			strerror(errno));
-+		return -1;
-+	}
-+
-+	gettimeofday(&o->lasttime, NULL);
- 	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");
- 		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) 
-+	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");
--		full_duplex = -1;
-+		    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) {
-@@ -348,6 +454,7 @@
- 	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;
-@@ -357,89 +464,54 @@
- 			ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
- 	}
- #if 1
--	fmt = BUFFER_FMT;
-+	/*
-+	 * on freebsd, SETFRAGMENT does not work very well on some cards.
-+	 * Better leave it out
-+	 */
-+	 
-+	// fmt = BUFFER_FMT;
-+	fmt = 0x8;	// 256-bytes fragment
- 	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
-+	/* XXX 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;
- }
- 
-+/*
-+ * make sure output mode is available. Returns 0 if done,
-+ * 1 if too early to switch, -1 if error
-+ */
- 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;
--		}
-+	if (oss.duplex == M_FULL || (oss.duplex == M_WRITE && !force))
- 		return 0;
--	}
--	return 1;
-+	if (!force && too_early(&oss))
-+		return 1;
-+	if (setformat(&oss, O_WRONLY))
-+		return -1;
-+	return 0;
- }
- 
-+/*
-+ * make sure input mode is available. Returns 0 if done
-+ * 1 if too early to switch, -1 if error
-+ */
- 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;
--		}
-+	if (oss.duplex == M_FULL || (oss.duplex == M_READ && !force))
- 		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;
-+	if (!force && too_early(&oss))
-+		return 1;
-+	if (setformat(&oss, O_RDONLY))
-+		return -1;
-+	return 0;
- }
- 
- static int oss_digit(struct ast_channel *c, char digit)
-@@ -454,120 +526,81 @@
- 	return 0;
- }
- 
-+/* request to play a sound on the speaker */
-+#define	RING(x) { int what = x; write(oss.sndcmd[1], &what, sizeof(what)); }
-+
- 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) {
-+	if (oss.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));
-+		RING(AST_CONTROL_RING);
- 	}
- 	return 0;
- }
- 
- static void answer_sound(void)
- {
--	int res;
--	nosound = 1;
--	res = 4;
--	write(sndcmd[1], &res, sizeof(res));
--	
-+	RING(AST_CONTROL_ANSWER);
- }
- 
- static int oss_answer(struct ast_channel *c)
- {
- 	ast_verbose( " << Console call has been answered >> \n");
--	answer_sound();
-+	answer_sound();	/* XXX do we really need it ? considering we shut down immediately... */
- 	ast_setstate(c, AST_STATE_UP);
--	cursound = -1;
--	nosound=0;
-+	oss.cursound = -1;
-+	oss.nosound=0;
- 	return 0;
- }
- 
- static int oss_hangup(struct ast_channel *c)
- {
--	int res = 0;
--	cursound = -1;
-+	oss.cursound = -1;
- 	c->pvt->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) {
-+	if (oss.hookstate) {
-+		if (oss.autoanswer || oss.autohangup) {
- 			/* Assume auto-hangup too */
--			hookstate = 0;
-+			oss.hookstate = 0;
- 		} else {
- 			/* Make congestion noise */
--			res = 2;
--			write(sndcmd[1], &res, sizeof(res));
-+			RING(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;
--}
--
--
-+/* used for data coming from the network */
- 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;
-+	int src;
-+
-+	// ast_log(LOG_WARNING, "oss_write size %d\n", f->datalen);
- 	/* Immediately return if no sound is enabled */
--	if (nosound)
-+	if (oss.nosound)
- 		return 0;
- 	/* Stop any currently playing sound */
--	cursound = -1;
--	if (!full_duplex) {
-+	oss.cursound = -1;
-+	if (oss.duplex != M_FULL) {
-+		/* XXX check this, looks weird! */
- 		/* If we're half duplex, we have to switch to read mode
- 		   to honor immediate needs if necessary */
--		res = soundcard_setinput(1);
-+		res = soundcard_setinput(1); /* force set if not full_duplex */
- 		if (res < 0) {
- 			ast_log(LOG_WARNING, "Unable to set device to input mode\n");
- 			return -1;
-@@ -583,21 +616,30 @@
- 		   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;
-+	/*
-+	 * we could receive a sample which is not a multiple of our FRAME_SIZE,
-+	 * so we buffer it locally and write to the device in FRAME_SIZE
-+	 * chunks, keeping the residue stored for future use.
-+	 */
-+
-+	src = 0; /* read position into f->data */
-+	while ( src < f->datalen ) {
-+	    static char buf[FRAME_SIZE*2];
-+	    static int dst = 0;
-+	    int l = sizeof(buf) - dst; /* how much room in the buffer */
-+
-+	    if (f->datalen - src >= l) {	/* enough to fill a frame */
-+		memcpy(buf + dst, f->data + src, l);
-+		soundcard_writeframe(&oss, (short *)buf);
-+		src += l;
-+		dst = 0;
-+	    } else { /* copy residue */
-+		l = f->datalen - src;
-+		memcpy(buf + dst, f->data + src, l);
-+		src += l;	/* but really, we are done */
-+		dst += l;
-+	    }
- 	}
--	if (len - pos) 
--		memmove(sizbuf, sizbuf + pos, len - pos);
--	sizpos = len - pos;
- 	return 0;
- }
- 
-@@ -628,18 +670,15 @@
- 		ast_log(LOG_WARNING, "Unable to set input mode\n");
- 		return NULL;
- 	}
--	if (res > 0) {
-+	if (res > 0) {	/* too early to switch ? */
- 		/* 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;
--	}
-+
-+	res = read(oss.sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
-+	// ast_log(LOG_WARNING, "oss_read() fd %d got %d\n", oss.sounddev, res);
-+	if (res < 0) /* audio data not ready, return a NULL frame */
-+		return &f;
- 	readpos += res;
- 	
- 	if (readpos >= FRAME_SIZE * 2) {
-@@ -682,64 +721,66 @@
- 	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;
-+		oss.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));
-+		RING(res);
- 	}
- 	return 0;	
- }
- 
--static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
-+static struct ast_channel *oss_new(struct chan_oss_pvt *oss, int state)
- {
- 	struct ast_channel *tmp;
-+	struct ast_channel_pvt *pvt;
-+
- 	tmp = ast_channel_alloc(1);
--	if (tmp) {
--		snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
--		tmp->type = type;
--		tmp->fds[0] = sounddev;
--		tmp->nativeformats = AST_FORMAT_SLINEAR;
--		tmp->pvt->pvt = p;
--		tmp->pvt->send_digit = oss_digit;
--		tmp->pvt->send_text = oss_text;
--		tmp->pvt->hangup = oss_hangup;
--		tmp->pvt->answer = oss_answer;
--		tmp->pvt->read = oss_read;
--		tmp->pvt->call = oss_call;
--		tmp->pvt->write = oss_write;
--		tmp->pvt->indicate = oss_indicate;
--		tmp->pvt->fixup = oss_fixup;
--		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;
--			}
-+	if (tmp == NULL)
-+		return NULL;
-+	snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", oss->device + 5);
-+	tmp->type = type;
-+	tmp->fds[0] = oss->sounddev;
-+	tmp->nativeformats = AST_FORMAT_SLINEAR;
-+	pvt = tmp->pvt;
-+	pvt->pvt = oss;
-+#if 1
-+	pvt->send_digit = oss_digit;
-+	pvt->send_text = oss_text;
-+	pvt->hangup = oss_hangup;
-+	pvt->answer = oss_answer;
-+	pvt->read = oss_read;
-+	pvt->call = oss_call;
-+	pvt->write = oss_write;
-+	pvt->indicate = oss_indicate;
-+	pvt->fixup = oss_fixup;
-+#endif
-+	if (strlen(oss->context))
-+		strncpy(tmp->context, oss->context, sizeof(tmp->context)-1);
-+	if (strlen(oss->exten))
-+		strncpy(tmp->exten, oss->exten, sizeof(tmp->exten)-1);
-+	if (strlen(language))
-+		strncpy(tmp->language, language, sizeof(tmp->language)-1);
-+	oss->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;
-+			/* XXX what about oss->owner and the channel itself ? */
- 		}
- 	}
- 	return tmp;
-@@ -770,13 +811,13 @@
- 	if ((argc != 1) && (argc != 2))
- 		return RESULT_SHOWUSAGE;
- 	if (argc == 1) {
--		ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
-+		ast_cli(fd, "Auto answer is %s.\n", oss.autoanswer ? "on" : "off");
- 		return RESULT_SUCCESS;
- 	} else {
- 		if (!strcasecmp(argv[1], "on"))
--			autoanswer = -1;
-+			oss.autoanswer = -1;
- 		else if (!strcasecmp(argv[1], "off"))
--			autoanswer = 0;
-+			oss.autoanswer = 0;
- 		else
- 			return RESULT_SHOWUSAGE;
- 	}
-@@ -788,12 +829,14 @@
- #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;
-@@ -816,8 +859,8 @@
- 		ast_cli(fd, "No one is calling us\n");
- 		return RESULT_FAILURE;
- 	}
--	hookstate = 1;
--	cursound = -1;
-+	oss.hookstate = 1;
-+	oss.cursound = -1;
- 	ast_queue_frame(oss.owner, &f);
- 	answer_sound();
- 	return RESULT_SUCCESS;
-@@ -863,12 +906,12 @@
- {
- 	if (argc != 1)
- 		return RESULT_SHOWUSAGE;
--	cursound = -1;
--	if (!oss.owner && !hookstate) {
-+	oss.cursound = -1;
-+	if (!oss.owner && !oss.hookstate) {
- 		ast_cli(fd, "No call to hangup up\n");
- 		return RESULT_FAILURE;
- 	}
--	hookstate = 0;
-+	oss.hookstate = 0;
- 	if (oss.owner) {
- 		ast_queue_hangup(oss.owner);
- 	}
-@@ -900,8 +943,8 @@
- 		}
- 		return RESULT_SUCCESS;
- 	}
--	mye = exten;
--	myc = context;
-+	mye = oss_exten;
-+	myc = default_context;
- 	if (argc == 2) {
- 		char *stringp=NULL;
- 		strncpy(tmp, argv[1], sizeof(tmp)-1);
-@@ -916,7 +959,7 @@
- 	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.hookstate = 1;
- 		oss_new(&oss, AST_STATE_RINGING);
- 	} else
- 		ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
-@@ -974,21 +1017,47 @@
- 	int res;
- 	int x;
- 	struct ast_config *cfg;
--	struct ast_variable *v;
--	res = pipe(sndcmd);
-+
-+	res = pipe(oss.sndcmd);
- 	if (res) {
- 		ast_log(LOG_ERROR, "Unable to create pipe\n");
- 		return -1;
- 	}
--	res = soundcard_init();
--	if (res < 0) {
-+	/* load config file */
-+	if ((cfg = ast_load(config))) {
-+		struct ast_variable *v = ast_variable_browse(cfg, "general");
-+		while(v) {
-+			if (!strcasecmp(v->name, "autoanswer"))
-+				oss.autoanswer = ast_true(v->value);
-+			else if (!strcasecmp(v->name, "autohangup"))
-+				oss.autohangup = ast_true(v->value);
-+			else if (!strcasecmp(v->name, "oss.silencesuppression"))
-+				oss.silencesuppression = ast_true(v->value);
-+			else if (!strcasecmp(v->name, "silencethreshold"))
-+				oss.silencethreshold = atoi(v->value);
-+			else if (!strcasecmp(v->name, "device"))
-+				strncpy(oss.device, v->value, sizeof(oss.device)-1);
-+			else if (!strcasecmp(v->name, "context"))
-+				strncpy(default_context, v->value, sizeof(default_context)-1);
-+			else if (!strcasecmp(v->name, "language"))
-+				strncpy(language, v->value, sizeof(language)-1);
-+			else if (!strcasecmp(v->name, "extension"))
-+				strncpy(oss_exten, v->value, sizeof(oss_exten)-1);
-+			v=v->next;
-+		}
-+		ast_destroy(cfg);
-+	}
-+	if (!strlen(oss.device))
-+		strncpy(oss.device, DEV_DSP, sizeof(oss.device)-1);
-+	if (setformat(&oss, O_RDWR) < 0) {	/* open device */
- 		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)
-+	soundcard_setinput(1); /* force set if not full_duplex */
-+	if (oss.duplex != M_FULL)
- 		ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
- 	res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
- 	if (res < 0) {
-@@ -997,26 +1066,7 @@
- 	}
- 	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
- 		ast_cli_register(myclis + x);
--	if ((cfg = ast_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);
--			v=v->next;
--		}
--		ast_destroy(cfg);
--	}
--	ast_pthread_create(&sthread, NULL, sound_thread, NULL);
-+	ast_pthread_create(&oss.sthread, NULL, sound_thread, NULL);
- 	return 0;
- }
- 
-@@ -1027,15 +1077,16 @@
- 	int x;
- 	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]);
-+	close(oss.sounddev);
-+	if (oss.sndcmd[0] > 0) {
-+		close(oss.sndcmd[0]);
-+		close(oss.sndcmd[1]);
- 	}
- 	if (oss.owner)
- 		ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
- 	if (oss.owner)
- 		return -1;
-+	/* XXX what about the thread ? */
- 	return 0;
- }
- 
diff --git a/net/asterisk-devel/files/patch-pbx::pbx_wilcalu.c b/net/asterisk-devel/files/patch-pbx::pbx_wilcalu.c
new file mode 100644
index 000000000000..41722c65568d
--- /dev/null
+++ b/net/asterisk-devel/files/patch-pbx::pbx_wilcalu.c
@@ -0,0 +1,14 @@
+--- pbx/pbx_wilcalu.c.orig	Tue Apr 26 10:00:28 2005
++++ pbx/pbx_wilcalu.c	Tue Apr 26 10:03:42 2005
+@@ -82,6 +82,11 @@
+ 		fds[0].events = POLLIN;
+ 		poll(fds, 1, -1);
+ 		bytes=read(fd,buf,256);
++		if (bytes <= 0) {
++			/* XXX error on device, sleep a bit before retrying */
++			sleep(1);
++			continue;
++		}
+ 		buf[(int)bytes]=0;
+ 
+ 		if(bytes>0){
diff --git a/net/asterisk-devel/files/patch-rtp.c b/net/asterisk-devel/files/patch-rtp.c
index 06289f357208..36c4bea2f7ea 100644
--- a/net/asterisk-devel/files/patch-rtp.c
+++ b/net/asterisk-devel/files/patch-rtp.c
@@ -1,8 +1,5 @@
-
-$FreeBSD$
-
---- rtp.c.orig	Sat Sep 18 16:56:28 2004
-+++ rtp.c	Sun Oct 10 15:57:22 2004
+--- rtp.c.orig	Tue Apr 26 10:00:28 2005
++++ rtp.c	Tue Apr 26 10:06:35 2005
 @@ -127,7 +127,7 @@
  {
  	switch(buf & TYPE_MASK) {
@@ -12,7 +9,29 @@ $FreeBSD$
  		break;
  	case TYPE_SILENCE:
  		return 4;
-@@ -841,8 +841,10 @@
+@@ -351,9 +351,7 @@
+ 					0, (struct sockaddr *)&sin, &len);
+ 	
+ 	if (res < 0) {
+-		if (errno == EAGAIN)
+-			ast_log(LOG_NOTICE, "RTP: Received packet with bad UDP checksum\n");
+-		else
++		if (errno != EAGAIN)
+ 			ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
+ 		if (errno == EBADF)
+ 			CRASH;
+@@ -431,9 +429,7 @@
+ 
+ 	rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
+ 	if (res < 0) {
+-		if (errno == EAGAIN)
+-			ast_log(LOG_NOTICE, "RTP: Received packet with bad UDP checksum\n");
+-		else
++		if (errno != EAGAIN)
+ 			ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
+ 		if (errno == EBADF)
+ 			CRASH;
+@@ -862,8 +858,10 @@
  		/* Must be an even port number by RTP spec */
  		rtp->us.sin_port = htons(x);
  		rtp->us.sin_addr = addr;