diff --git a/formats/format_mp3.c b/formats/format_mp3.c
new file mode 100755
index 0000000000000000000000000000000000000000..811acbe149a98d9af6c6778f395114803673ef20
--- /dev/null
+++ b/formats/format_mp3.c
@@ -0,0 +1,268 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Everybody's favorite format: MP3 Files!  Yay!
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/channel.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/sched.h>
+#include <asterisk/module.h>
+#include <arpa/inet.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+#include <pthread.h>
+#include <sys/time.h>
+#include "../channels/adtranvofr.h"
+
+
+#define MP3_MAX_SIZE 1400
+
+struct ast_filestream {
+	/* First entry MUST be reserved for the channel type */
+	void *reserved[AST_RESERVED_POINTERS];
+	/* This is what a filestream means to us */
+	int fd; /* Descriptor */
+	struct ast_channel *owner;
+	struct ast_filestream *next;
+	struct ast_frame *fr;	/* Frame representation of buf */
+	char buf[sizeof(struct ast_frame) + MP3_MAX_SIZE + AST_FRIENDLY_OFFSET];	/* Buffer for sending frames, etc */
+	int pos;
+};
+
+
+static struct ast_filestream *glist = NULL;
+static pthread_mutex_t mp3_lock = PTHREAD_MUTEX_INITIALIZER;
+static int glistcnt = 0;
+
+static char *name = "mp3";
+static char *desc = "MPEG-2 Layer 3 File Format Support";
+static char *exts = "mp3|mpeg3";
+
+#if 0
+#define MP3_FRAMELEN 417
+#else
+#define MP3_FRAMELEN 400
+#endif
+#define MP3_OUTPUTLEN 2304	/* Bytes */
+#define MP3_TIMELEN ((MP3_OUTPUTLEN * 1000 / 16000) )
+
+static struct ast_filestream *mp3_open(int fd)
+{
+	/* We don't have any header to read or anything really, but
+	   if we did, it would go here.  We also might want to check
+	   and be sure it's a valid file.  */
+	struct ast_filestream *tmp;
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
+		if (pthread_mutex_lock(&mp3_lock)) {
+			ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->fr = (struct ast_frame *)tmp->buf;
+		tmp->fr->data = tmp->buf + sizeof(struct ast_frame);
+		tmp->fr->frametype = AST_FRAME_VOICE;
+		tmp->fr->subclass = AST_FORMAT_MP3;
+		/* datalen will vary for each frame */
+		tmp->fr->src = name;
+		tmp->fr->mallocd = 0;
+		glistcnt++;
+		pthread_mutex_unlock(&mp3_lock);
+		ast_update_use_count();
+	}
+	return tmp;
+}
+
+static struct ast_filestream *mp3_rewrite(int fd, char *comment)
+{
+	/* We don't have any header to read or anything really, but
+	   if we did, it would go here.  We also might want to check
+	   and be sure it's a valid file.  */
+	struct ast_filestream *tmp;
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
+		if (pthread_mutex_lock(&mp3_lock)) {
+			ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->fr = NULL;
+		glistcnt++;
+		pthread_mutex_unlock(&mp3_lock);
+		ast_update_use_count();
+	} else
+		ast_log(LOG_WARNING, "Out of memory\n");
+	return tmp;
+}
+
+static struct ast_frame *mp3_read(struct ast_filestream *s)
+{
+	return NULL;
+}
+
+static void mp3_close(struct ast_filestream *s)
+{
+	struct ast_filestream *tmp, *tmpl = NULL;
+	if (pthread_mutex_lock(&mp3_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
+		return;
+	}
+	tmp = glist;
+	while(tmp) {
+		if (tmp == s) {
+			if (tmpl)
+				tmpl->next = tmp->next;
+			else
+				glist = tmp->next;
+			break;
+		}
+		tmpl = tmp;
+		tmp = tmp->next;
+	}
+	glistcnt--;
+	if (s->owner) {
+		s->owner->stream = NULL;
+		if (s->owner->streamid > -1)
+			ast_sched_del(s->owner->sched, s->owner->streamid);
+		s->owner->streamid = -1;
+	}
+	pthread_mutex_unlock(&mp3_lock);
+	ast_update_use_count();
+	if (!tmp) 
+		ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
+	close(s->fd);
+	free(s);
+}
+
+static int ast_read_callback(void *data)
+{
+	/* XXX Don't assume frames are this size XXX */
+	u_int16_t size=MP3_FRAMELEN;
+	u_int32_t delay = -1;
+	int res;
+	struct ast_filestream *s = data;
+	/* Send a frame from the file to the appropriate channel */
+	/* Read the data into the buffer */
+	s->fr->offset = AST_FRIENDLY_OFFSET;
+	s->fr->datalen = size;
+	s->fr->data = s->buf + sizeof(struct ast_frame) + AST_FRIENDLY_OFFSET;
+	if ((res = read(s->fd, s->fr->data , size)) != size) {
+		ast_log(LOG_WARNING, "Short read (%d of %d bytes) (%s)!\n", res, size, strerror(errno));
+		s->owner->streamid = -1;
+		return 0;
+	}
+	delay = MP3_TIMELEN;
+	s->fr->timelen = delay;
+	/* Lastly, process the frame */
+	if (ast_write(s->owner, s->fr)) {
+		ast_log(LOG_WARNING, "Failed to write frame\n");
+		s->owner->streamid = -1;
+		return 0;
+	}
+	return -1;
+}
+
+static int mp3_apply(struct ast_channel *c, struct ast_filestream *s)
+{
+	/* Select our owner for this stream, and get the ball rolling. */
+	s->owner = c;
+	s->owner->streamid = ast_sched_add(s->owner->sched, MP3_TIMELEN, ast_read_callback, s);
+	ast_read_callback(s);
+	return 0;
+}
+
+static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	int res;
+	if (fs->fr) {
+		ast_log(LOG_WARNING, "Asked to write on a read stream??\n");
+		return -1;
+	}
+	if (f->frametype != AST_FRAME_VOICE) {
+		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
+		return -1;
+	}
+	if (f->subclass != AST_FORMAT_MP3) {
+		ast_log(LOG_WARNING, "Asked to write non-mp3 frame!\n");
+		return -1;
+	}
+	if ((res = write(fs->fd, f->data, f->datalen)) != f->datalen) {
+		ast_log(LOG_WARNING, "Unable to write frame: res=%d (%s)\n", res, strerror(errno));
+		return -1;
+	}	
+	return 0;
+}
+
+char *mp3_getcomment(struct ast_filestream *s)
+{
+	return NULL;
+}
+
+int load_module()
+{
+	return ast_format_register(name, exts, AST_FORMAT_MP3,
+								mp3_open,
+								mp3_rewrite,
+								mp3_apply,
+								mp3_write,
+								mp3_read,
+								mp3_close,
+								mp3_getcomment);
+								
+								
+}
+
+int unload_module()
+{
+	struct ast_filestream *tmp, *tmpl;
+	if (pthread_mutex_lock(&mp3_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
+		return -1;
+	}
+	tmp = glist;
+	while(tmp) {
+		if (tmp->owner)
+			ast_softhangup(tmp->owner);
+		tmpl = tmp;
+		tmp = tmp->next;
+		free(tmpl);
+	}
+	pthread_mutex_unlock(&mp3_lock);
+	return ast_format_unregister(name);
+}	
+
+int usecount()
+{
+	int res;
+	if (pthread_mutex_lock(&mp3_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
+		return -1;
+	}
+	res = glistcnt;
+	pthread_mutex_unlock(&mp3_lock);
+	return res;
+}
+
+char *description()
+{
+	return desc;
+}
+
diff --git a/frame.c b/frame.c
new file mode 100755
index 0000000000000000000000000000000000000000..ba0e48e925411a1811225d1c764c30beb4925a81
--- /dev/null
+++ b/frame.c
@@ -0,0 +1,89 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Frame manipulation routines
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/frame.h>
+#include <asterisk/logger.h>
+#include <stdlib.h>
+#include <string.h>
+
+/*
+ * Important: I should be made more efficient.  Frame headers should
+ * most definitely be cached
+ */
+
+void ast_frfree(struct ast_frame *fr)
+{
+	if (fr->mallocd & AST_MALLOCD_DATA) {
+		if (fr->data) 
+			free(fr->data - fr->offset);
+	}
+	if (fr->mallocd & AST_MALLOCD_SRC) {
+		if (fr->src)
+			free(fr->src);
+	}
+	if (fr->mallocd & AST_MALLOCD_HDR) {
+		free(fr);
+	}
+}
+
+void ast_frchain(struct ast_frame_chain *fc)
+{
+	struct ast_frame_chain *last;
+	while(fc) {
+		last = fc;
+		fc = fc->next;
+		if (last->fr)
+			ast_frfree(last->fr);
+		free(last);
+	}
+}
+
+struct ast_frame *ast_frisolate(struct ast_frame *fr)
+{
+	struct ast_frame *out;
+	if (!(fr->mallocd & AST_MALLOCD_HDR)) {
+		/* Allocate a new header if needed */
+		out = malloc(sizeof(struct ast_frame));
+		if (!out) {
+			ast_log(LOG_WARNING, "Out of memory\n");
+			return NULL;
+		}
+		out->frametype = fr->frametype;
+		out->subclass = fr->subclass;
+		out->datalen = 0;
+		out->timelen = fr->timelen;
+		out->offset = 0;
+		out->src = NULL;
+		out->data = NULL;
+	} else {
+		out = fr;
+	}
+	if (!(fr->mallocd & AST_MALLOCD_SRC)) {
+		if (fr->src)
+			out->src = strdup(fr->src);
+	} else
+		out->src = fr->src;
+	if (!(fr->mallocd & AST_MALLOCD_DATA))  {
+		out->data = malloc(fr->datalen + fr->offset);
+		out->data += fr->offset;
+		out->offset = fr->offset;
+		out->datalen = fr->datalen;
+		memcpy(out->data, fr->data, fr->datalen);
+		if (!out->data) {
+			ast_log(LOG_WARNING, "Out of memory\n");
+			return NULL;
+		}
+	}
+	out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
+	return out;
+}
diff --git a/include/asterisk/file.h b/include/asterisk/file.h
new file mode 100755
index 0000000000000000000000000000000000000000..da4ffaa42a023285a79de22a02de2a485cd5af58
--- /dev/null
+++ b/include/asterisk/file.h
@@ -0,0 +1,84 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Generic File Format Support.
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#ifndef _ASTERISK_FILE_H
+#define _ASTERISK_FILE_H
+
+#include <asterisk/channel.h>
+#include <asterisk/frame.h>
+#include <fcntl.h>
+
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+
+/* Convenient for waiting */
+#define AST_DIGIT_ANY "0123456789#*"
+
+/* Defined by individual formats.  First item MUST be a
+   pointer for use by the stream manager */
+struct ast_filestream;
+
+/* Register a new file format capability */
+int ast_format_register(char *name, char *exts, int format,
+						struct ast_filestream * (*open)(int fd),
+						struct ast_filestream * (*rewrite)(int fd, char *comment),
+						int (*apply)(struct ast_channel *, struct ast_filestream *),
+						int (*write)(struct ast_filestream *, struct ast_frame *),
+						struct ast_frame * (*read)(struct ast_filestream *),
+						void (*close)(struct ast_filestream *),
+						char * (*getcomment)(struct ast_filestream *));
+	
+int ast_format_unregister(char *name);
+
+/* Start streaming a file */
+int ast_streamfile(struct ast_channel *c, char *filename);
+
+/* Stop playback of a stream */
+int ast_stopstream(struct ast_channel *c);
+
+/* See if a given file exists in a given format.  If fmt is NULL,  any format is accepted.*/
+int ast_fileexists(char *filename, char *fmt);
+
+/* Rename a given file in a given format, or if fmt is NULL, then do so for all */
+int ast_filerename(char *oldname, char *newname, char *fmt);
+
+/* Delete a given file in a given format, or if fmt is NULL, then do so for all */
+int ast_filedelete(char *filename, char *fmt);
+
+/* Wait for a stream to stop or for any one of a given digit to arrive,  Returns
+   0 if the stream finishes, the character if it was interrupted, and -1 on error */
+char ast_waitstream(struct ast_channel *c, char *breakon);
+
+/* Create an outgoing file stream.  oflags are flags for the open() command, and
+   if check is non-zero, then it will not write a file if there are any files that
+   start with that name and have an extension */
+struct ast_filestream *ast_writefile(char *filename, char *type, char *comment, int oflags, int check, mode_t mode);
+
+/* Send a frame to a filestream -- note: does NOT free the frame, call ast_frfree manually */
+int ast_writestream(struct ast_filestream *fs, struct ast_frame *f);
+
+/* Close a playback or recording stream */
+int ast_closestream(struct ast_filestream *f);
+
+#define AST_RESERVED_POINTERS 4
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+
+
+#endif
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
new file mode 100755
index 0000000000000000000000000000000000000000..d9fb440f0afb3ec538b793ef6622324aa95f532c
--- /dev/null
+++ b/include/asterisk/frame.h
@@ -0,0 +1,101 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Asterisk internal frame definitions.
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#ifndef _ASTERISK_FRAME_H
+#define _ASTERISK_FRAME_H
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+/* A frame of data read used to communicate between 
+   between channels and applications */
+struct ast_frame {
+	int frametype;				/* Kind of frame */
+	int subclass;				/* Subclass, frame dependent */
+	int datalen;				/* Length of data */
+	int timelen;				/* Amount of time associated with this frame */
+	int mallocd;				/* Was the data malloc'd?  i.e. should we
+								   free it when we discard the frame? */
+	int offset;					/* How far into "data" the data really starts */
+	char *src;					/* Optional source of frame for debugging */
+	void *data;					/* Pointer to actual data */
+};
+
+struct ast_frame_chain {
+	/* XXX Should ast_frame chain's be just prt of frames, i.e. should they just link? XXX */
+	struct ast_frame *fr;
+	struct ast_frame_chain *next;
+};
+
+#define AST_FRIENDLY_OFFSET 	64		/* It's polite for a a new frame to
+										   have at least this number of bytes
+										   of offset before your real frame data
+										   so that additional headers can be
+										   added. */
+
+#define AST_MALLOCD_HDR		(1 << 0)	/* Need the header be free'd? */
+#define AST_MALLOCD_DATA	(1 << 1)	/* Need the data be free'd? */
+#define AST_MALLOCD_SRC		(1 << 2)	/* Need the source be free'd? (haha!) */
+
+/* Frame types */
+#define AST_FRAME_DTMF		1		/* A DTMF digit, subclass is the digit */
+#define AST_FRAME_VOICE		2		/* Voice data, subclass is AST_FORMAT_* */
+#define AST_FRAME_VIDEO		3		/* Video frame, maybe?? :) */
+#define AST_FRAME_CONTROL	4		/* A control frame, subclass is AST_CONTROL_* */
+#define AST_FRAME_NULL		5		/* An empty, useless frame */
+
+/* Data formats for capabilities and frames alike */
+#define AST_FORMAT_G723_1	(1 << 0)	/* G.723.1 compression */
+#define AST_FORMAT_GSM		(1 << 1)	/* GSM compression */
+#define AST_FORMAT_ULAW		(1 << 2)	/* Raw mu-law data (G.711) */
+#define AST_FORMAT_ALAW		(1 << 3)	/* Raw A-law data (G.711) */
+#define AST_FORMAT_MP3		(1 << 4)	/* MPEG-2 layer 3 */
+#define AST_FORMAT_ADPCM	(1 << 5)	/* ADPCM */
+#define AST_FORMAT_SLINEAR	(1 << 6)	/* Raw 16-bit Signed Linear (8000 Hz) PCM */
+#define AST_FORMAT_MAX_AUDIO (1 << 15)	/* Maximum audio format */
+#define AST_FORMAT_JPEG		(1 << 16)	/* JPEG Images */
+#define AST_FORMAT_PNG		(1 << 17)	/* PNG Images */
+#define AST_FORMAT_H261		(1 << 18)	/* H.261 Video */
+#define AST_FORMAT_H263		(1 << 19)	/* H.263 Video */
+
+/* Control frame types */
+#define AST_CONTROL_HANGUP		1			/* Other end has hungup */
+#define AST_CONTROL_RING		2			/* Local ring */
+#define AST_CONTROL_RINGING 	3			/* Remote end is ringing */
+#define AST_CONTROL_ANSWER		4			/* Remote end has answered */
+#define AST_CONTROL_BUSY		5			/* Remote end is busy */
+#define AST_CONTROL_TAKEOFFHOOK 6			/* Make it go off hook */
+#define AST_CONTROL_OFFHOOK		7			/* Line is off hook */
+
+/* Request a frame be allocated.  source is an optional source of the frame, 
+   len is the requested length, or "0" if the caller will supply the buffer */
+struct ast_frame *ast_fralloc(char *source, int len);
+
+/* Free a frame, and the memory it used if applicable */
+void ast_frfree(struct ast_frame *fr);
+
+/* Take a frame, and if it's not been malloc'd, make a malloc'd copy
+   and if the data hasn't been malloced then make the
+   data malloc'd.  If you need to store frames, say for queueing, then
+   you should call this function. */
+struct ast_frame *ast_frisolate(struct ast_frame *fr);
+
+void ast_frchain(struct ast_frame_chain *fc);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+
+#endif
diff --git a/include/asterisk/translate.h b/include/asterisk/translate.h
new file mode 100755
index 0000000000000000000000000000000000000000..68edc046dca6c755fe3a2ccab054552584cf022d
--- /dev/null
+++ b/include/asterisk/translate.h
@@ -0,0 +1,76 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Translate via the use of pseudo channels
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#ifndef _ASTERISK_TRANSLATE_H
+#define _ASTERISK_TRANSLATE_H
+
+#define MAX_FORMAT 32
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+#include <asterisk/frame.h>
+
+/* Declared by individual translators */
+struct ast_translator_pvt;
+
+struct ast_translator {
+	char name[80];
+	int srcfmt;
+	int dstfmt;
+	struct ast_translator_pvt *(*new)();
+	int (*framein)(struct ast_translator_pvt *pvt, struct ast_frame *in);
+	struct ast_frame * (*frameout)(struct ast_translator_pvt *pvt);
+	void (*destroy)(struct ast_translator_pvt *pvt);
+	/* For performance measurements */
+	/* Generate an example frame */
+	struct ast_frame * (*sample)(void);
+	/* Cost in milliseconds for encoding/decoding 1 second of sound */
+	int cost;
+	/* For linking, not to be modified by the translator */
+	struct ast_translator *next;
+};
+
+struct ast_trans_pvt;
+
+/* Create a pseudo channel which translates from a real channel into our
+   desired format.  When a translator is installed, you should not use the
+   sub channel until you have stopped the translator.  For all other
+   actions, use the real channel. Generally, translators should be created 
+   when needed and immediately destroyed when no longer needed.  */
+
+/* Directions */
+#define AST_DIRECTION_OUT  1
+#define AST_DIRECTION_IN   2
+#define AST_DIRECTION_BOTH 3
+
+extern struct ast_channel *ast_translator_create(struct ast_channel *real, int format, int direction);
+extern void ast_translator_destroy(struct ast_channel *tran);
+/* Register a Codec translator */
+extern int ast_register_translator(struct ast_translator *t);
+/* Unregister same */
+extern int ast_unregister_translator(struct ast_translator *t);
+/* Given a list of sources, and a designed destination format, which should
+   I choose? */
+extern int ast_translator_best_choice(int dst, int srcs);
+extern struct ast_trans_pvt *ast_translator_build_path(int source, int dest);
+extern void ast_translator_free_path(struct ast_trans_pvt *tr);
+extern struct ast_frame_chain *ast_translate(struct ast_trans_pvt *tr, struct ast_frame *f);
+
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif