pbx_wilcalu.c:

	new patch for this file, smooths the effects of
	an unhandled error Cexiting from poll() and resulting
	otherwise in this process taking 100% of the CPU

rtp.c:

	updated patch for rtp.c, removes a misleading 'checksum error'
	message when in reality the recvfrom() just returned no data;

chan_oss.c:

	replacement for the old chan_oss.c - the changes are
	so massive that having a patch would be completely
	unreadable.

	Among other things this lets you change many /dev/dsp
	parameters from the config file, to ease adapting to
	the idiosincracies of various sound cards and drivers.
	It also supports multiple soundcards on the same box,
	which might be useful in some cases.

Submitted by:	luigi

Add WITHOUT_MYSQL knob.

Suggested by:	phantom

5 files changed, 1370 insertions, 1178 deletions

diff --git a/net/asterisk12/Makefile b/net/asterisk12/Makefile
index bf9289fc189c..00cb6ce94061 100644
--- a/net/asterisk12/Makefile
+++ b/net/asterisk12/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/asterisk12/files/chan_oss.c b/net/asterisk12/files/chan_oss.c
new file mode 100644
index 000000000000..aef0db9dca85
--- /dev/null
+++ b/net/asterisk12/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/asterisk12/files/patch-channels::chan_oss.c b/net/asterisk12/files/patch-channels::chan_oss.c
deleted file mode 100644
index ef8cfc11d711..000000000000
--- a/net/asterisk12/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/asterisk12/files/patch-pbx::pbx_wilcalu.c b/net/asterisk12/files/patch-pbx::pbx_wilcalu.c
new file mode 100644
index 000000000000..41722c65568d
--- /dev/null
+++ b/net/asterisk12/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/asterisk12/files/patch-rtp.c b/net/asterisk12/files/patch-rtp.c
index 06289f357208..36c4bea2f7ea 100644
--- a/net/asterisk12/files/patch-rtp.c
+++ b/net/asterisk12/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;