main/asterisk: upgrade to 10.0.0

... and delete the testing version.

5 files changed, 603 insertions, 4 deletions

diff --git a/main/asterisk/APKBUILD b/main/asterisk/APKBUILD
index db6362ac9..b6bb6e6b4 100644
--- a/main/asterisk/APKBUILD
+++ b/main/asterisk/APKBUILD
@@ -1,7 +1,7 @@
 # Contributor: Timo Teras <timo.teras@iki.fi>
 # Maintainer: Timo Teras <timo.teras@iki.fi>
 pkgname=asterisk
-pkgver=1.8.8.0_rc5
+pkgver=10.0.0
 pkgrel=0
 pkgdesc="Asterisk: A Module Open Source PBX System"
 url="http://www.asterisk.org/"
@@ -11,7 +11,7 @@ depends=
 makedepends="autoconf automake libtool ncurses-dev popt-dev newt-dev zlib-dev
 	postgresql-dev unixodbc-dev dahdi-tools-dev libpri-dev tar
 	freetds-dev openssl-dev lua-dev alsa-lib-dev spandsp-dev tiff-dev
-	libresample sqlite-dev wget speex-dev"
+	libresample sqlite-dev wget speex-dev libogg-dev"
 install="$pkgname.pre-install $pkgname.pre-upgrade $pkgname.post-install"
 subpackages="$pkgname-dev $pkgname-doc $pkgname-pgsql $pkgname-odbc
 	$pkgname-tds $pkgname-fax $pkgname-sample-config:sample $pkgname-sqlite
@@ -20,6 +20,10 @@ source="http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-$
 	100-uclibc-daemon.patch
 	101-caps-uclibc.patch
 	400-bug-227.patch
+	ASTERISK-18976.patch
+	ASTERISK-18977.patch
+	ASTERISK-18994.patch
+	ASTERISK-18995.patch
 	asterisk.initd
 	asterisk.confd
 	asterisk.logrotate"
@@ -28,7 +32,7 @@ _builddir="$srcdir/$pkgname-${pkgver/_/-}"
 
 prepare() {
 	cd "$_builddir"
-	for i in ../[1-9]*.patch; do
+	for i in ../[1-9A]*.patch; do
 		msg "Apply $i"
 		patch -p1 < $i || return 1
 	done
@@ -159,10 +163,14 @@ sound_en() {
 	_move_dir var/lib/asterisk/sounds/en
 }
 
-md5sums="737c309d970112b39c44ea246e4c9b21  asterisk-1.8.8.0-rc5.tar.gz
+md5sums="6b52336b9dc01eeecb9de2eb2818e127  asterisk-10.0.0.tar.gz
 b00c9d98ce2ad445501248a197c6e436  100-uclibc-daemon.patch
 6e1129e30c4fd2c25c86c81685a485a9  101-caps-uclibc.patch
 79e9634b5054bceb3b8dc246654bb243  400-bug-227.patch
+da5a3c500192dee4275aae5235d25f97  ASTERISK-18976.patch
+1ddadef41aa7120e168738b6f3ed8917  ASTERISK-18977.patch
+0af5e797f0a99d0f81f95e3710baf5b6  ASTERISK-18994.patch
+bc6713f5434e07b79d3afdd155461d72  ASTERISK-18995.patch
 0ce3219e05f55af884a7dc6b99c2b276  asterisk.initd
 ed31d7ba37bcf8b0346dcf8593c395f0  asterisk.confd
 3e65172275684373e1a25c8a11224411  asterisk.logrotate"
diff --git a/main/asterisk/ASTERISK-18976.patch b/main/asterisk/ASTERISK-18976.patch
new file mode 100644
index 000000000..36c613ff4
--- /dev/null
+++ b/main/asterisk/ASTERISK-18976.patch
@@ -0,0 +1,142 @@
+--- a/pbx/pbx_lua.c.orig	2011-12-08 08:28:31.000000000 +0200
++++ b/pbx/pbx_lua.c	2011-12-08 08:54:22.000000000 +0200
+@@ -94,7 +94,6 @@
+ static void lua_create_application_metatable(lua_State *L);
+ static void lua_create_autoservice_functions(lua_State *L);
+ static void lua_create_hangup_function(lua_State *L);
+-static void lua_detect_goto(lua_State *L);
+ static void lua_concat_args(lua_State *L, int start, int nargs);
+ 
+ static void lua_state_destroy(void *data);
+@@ -213,19 +212,10 @@
+ 	chan = lua_touserdata(L, -1);
+ 	lua_pop(L, 1);
+ 	
++	context = ast_strdupa(chan->context);
++	exten = ast_strdupa(chan->exten);
++	priority = chan->priority;
+ 	
+-	lua_getfield(L, LUA_REGISTRYINDEX, "context");
+-	context = ast_strdupa(lua_tostring(L, -1));
+-	lua_pop(L, 1);
+-	
+-	lua_getfield(L, LUA_REGISTRYINDEX, "exten");
+-	exten = ast_strdupa(lua_tostring(L, -1));
+-	lua_pop(L, 1);
+-	
+-	lua_getfield(L, LUA_REGISTRYINDEX, "priority");
+-	priority = lua_tointeger(L, -1);
+-	lua_pop(L, 1);
+-
+ 	lua_concat_args(L, 2, nargs);
+ 	data = lua_tostring(L, -1);
+ 
+@@ -256,75 +246,51 @@
+ 		return lua_error(L);
+ 	}
+ 
+-	lua_detect_goto(L);
+-
+-	return 0;
+-}
+-
+-/*!
+- * \brief Detect if a Goto or other dialplan jump has been executed and return
+- * control to the pbx engine.
+- */
+-static void lua_detect_goto(lua_State *L)
+-{
+-	struct ast_channel *chan;
+-
+-	lua_getfield(L, LUA_REGISTRYINDEX, "channel");
+-	chan = lua_touserdata(L, -1);
+-	lua_pop(L, 1);
+-
+-	/* check context */
+-	lua_getfield(L, LUA_REGISTRYINDEX, "context");
+-	lua_pushstring(L, chan->context);
+-	if (!lua_equal(L, -1, -2)) {
++	if (strcmp(context, chan->context)) {
++		lua_pushstring(L, context);
++		lua_pushstring(L, chan->context);
+ 		lua_pushliteral(L, "context");
+-		goto e_goto_detected;
+-	}
+-	lua_pop(L, 2);
+-
+-	/* check exten */
+-	lua_getfield(L, LUA_REGISTRYINDEX, "exten");
+-	lua_pushstring(L, chan->exten);
+-	if (!lua_equal(L, -1, -2)) {
++	} else if (strcmp(exten, chan->exten)) {
++		lua_pushstring(L, exten);
++		lua_pushstring(L, chan->exten);
+ 		lua_pushliteral(L, "exten");
+-		goto e_goto_detected;
+-	}
+-	lua_pop(L, 2);
+-
+-	/* check priority */
+-	lua_getfield(L, LUA_REGISTRYINDEX, "priority");
+-	lua_pushinteger(L, chan->priority);
+-	if (!lua_equal(L, -1, -2)) {
++	} else if (priority != chan->priority) {
++		lua_pushinteger(L, priority);
++		lua_pushinteger(L, chan->priority);
+ 		lua_pushliteral(L, "priority");
+-		goto e_goto_detected;
++	} else {
++		/* no goto - restore the original position back
++		 * to lua state, in case this was a recursive dialplan
++		 * call (a dialplan application re-entering dialplan) */
++		lua_update_registry(L, context, exten, priority);
++		return 0;
+ 	}
+-	lua_pop(L, 2);
+-	return;
+-
+-e_goto_detected:
+-	/* format our debug message */
+-	lua_insert(L, -3);
+ 
+-	lua_pushliteral(L, " changed from ");
++	/* goto detected - construct error message */
+ 	lua_insert(L, -3);
+-
+-	lua_pushliteral(L, " to ");
+-	lua_insert(L, -2);
+-
+-	lua_concat(L, 5);
+-
+-	ast_debug(2, "Goto detected: %s\n", lua_tostring(L, -1));
+-	lua_pop(L, 1);
+-
+-	/* let the lua engine know it needs to return control to the pbx */
+-	lua_pushinteger(L, LUA_GOTO_DETECTED);
++													
++	lua_pushliteral(L, " changed from ");							    
++	lua_insert(L, -3);									       
++													 
++	lua_pushliteral(L, " to ");								      
++	lua_insert(L, -2);									       
++													 
++	lua_concat(L, 5);										
++													 
++	ast_debug(2, "Goto detected: %s\n", lua_tostring(L, -1));					
++	lua_pop(L, 1);										   
++													 
++	/* let the lua engine know it needs to return control to the pbx */			      
++	lua_pushinteger(L, LUA_GOTO_DETECTED);							   
+ 	lua_error(L);
++
++	return 0;
+ }
+ 
+ /*!
+  * \brief [lua_CFunction] Used to get the value of a variable or dialplan
+  * function (for access from lua, don't call directly)
+- * 
++ *
+  * The value of the variable or function is returned.  This function is the
+  * 'get()' function in the following example as would be seen in
+  * extensions.lua.
diff --git a/main/asterisk/ASTERISK-18977.patch b/main/asterisk/ASTERISK-18977.patch
new file mode 100644
index 000000000..fffce032c
--- /dev/null
+++ b/main/asterisk/ASTERISK-18977.patch
@@ -0,0 +1,63 @@
+--- a/res/res_musiconhold.c.orig
++++ b/res/res_musiconhold.c
+@@ -158,6 +158,7 @@
+ 	char name[MAX_MUSICCLASS];
+ 	struct ast_format origwfmt;
+ 	struct ast_format mohwfmt;
++	int announcement;
+ 	int samples;
+ 	int sample_queue;
+ 	int pos;
+@@ -173,6 +174,7 @@
+ #define MOH_SORTALPHA		(1 << 4)
+ 
+ #define MOH_CACHERTCLASSES      (1 << 5)        /*!< Should we use a separate instance of MOH for each user or not */
++#define MOH_ANNOUNCEMENT	(1 << 6)
+ 
+ /* Custom astobj2 flag */
+ #define MOH_NOTDELETED          (1 << 30)       /*!< Find only records that aren't deleted? */
+@@ -183,6 +185,7 @@
+ 	char name[MAX_MUSICCLASS];
+ 	char dir[256];
+ 	char args[256];
++	char announcement[80];
+ 	char mode[80];
+ 	char digit;
+ 	/*! A dynamically sized array to hold the list of filenames in "files" mode */
+@@ -278,6 +281,7 @@
+ 	}
+ 
+ 	state->save_pos = state->pos;
++	state->announcement = 0;
+ 
+ 	state->class = mohclass_unref(state->class, "Unreffing channel's music class upon deactivation of generator");
+ }
+@@ -293,6 +297,16 @@
+ 		chan->stream = NULL;
+ 	}
+ 
++	if (ast_test_flag(state->class, MOH_ANNOUNCEMENT) && state->announcement == 0) {
++		state->announcement = 1;
++		if (ast_openstream_full(chan, state->class->announcement, chan->language, 1)) {
++			ast_debug(1, "%s Opened announcement '%s'\n", chan->name, state->class->announcement);
++			return 0;
++		}
++	} else {
++		state->announcement = 0;
++	}
++
+ 	if (!state->class->total_files) {
+ 		ast_log(LOG_WARNING, "No files available for class '%s'\n", state->class->name);
+ 		return -1;
+@@ -1731,7 +1745,10 @@
+ 				ast_copy_string(class->dir, var->value, sizeof(class->dir));
+ 			else if (!strcasecmp(var->name, "application"))
+ 				ast_copy_string(class->args, var->value, sizeof(class->args));
+-			else if (!strcasecmp(var->name, "digit") && (isdigit(*var->value) || strchr("*#", *var->value)))
++			else if (!strcasecmp(var->name, "announcement")) {
++				ast_copy_string(class->announcement, var->value, sizeof(class->announcement));
++				ast_set_flag(class, MOH_ANNOUNCEMENT);
++			} else if (!strcasecmp(var->name, "digit") && (isdigit(*var->value) || strchr("*#", *var->value)))
+ 				class->digit = *var->value;
+ 			else if (!strcasecmp(var->name, "random"))
+ 				ast_set2_flag(class, ast_true(var->value), MOH_RANDOMIZE);
diff --git a/main/asterisk/ASTERISK-18994.patch b/main/asterisk/ASTERISK-18994.patch
new file mode 100644
index 000000000..702e91375
--- /dev/null
+++ b/main/asterisk/ASTERISK-18994.patch
@@ -0,0 +1,28 @@
+--- a/main/file.c.orig	2011-12-09 11:32:05.000000000 +0200
++++ b/main/file.c	2011-12-09 11:35:22.000000000 +0200
+@@ -1005,6 +1005,7 @@
+ 	struct ast_filestream *fs;
+ 	struct ast_filestream *vfs=NULL;
+ 	char fmt[256];
++	off_t pos;
+ 	int seekattempt;
+ 	int res;
+ 
+@@ -1017,12 +1018,14 @@
+ 	/* check to see if there is any data present (not a zero length file),
+ 	 * done this way because there is no where for ast_openstream_full to
+ 	 * return the file had no data. */
+-	seekattempt = fseek(fs->f, -1, SEEK_END);
++	pos = ftello(fs->f);
++	seekattempt = fseeko(fs->f, -1, SEEK_END);
+ 	if (seekattempt && errno == EINVAL) {
+ 		/* Zero-length file, as opposed to a pipe */
+ 		return 0;
+-	} else {
+-		ast_seekstream(fs, 0, SEEK_SET);
++	}
++	if (seekattempt == 0) {
++		fseeko(fs->f, pos, SEEK_SET);
+ 	}
+ 
+ 	vfs = ast_openvstream(chan, filename, preflang);
diff --git a/main/asterisk/ASTERISK-18995.patch b/main/asterisk/ASTERISK-18995.patch
new file mode 100644
index 000000000..cd144847b
--- /dev/null
+++ b/main/asterisk/ASTERISK-18995.patch
@@ -0,0 +1,358 @@
+--- /dev/null	2011-11-29 09:02:40.279581283 +0200
++++ b/formats/format_ogg_speex.c	2011-12-08 15:57:12.000000000 +0200
+@@ -0,0 +1,355 @@
++/*
++ * Asterisk -- An open source telephony toolkit.
++ *
++ * Copyright (C) 2011, Timo Teräs
++ *
++ * See http://www.asterisk.org for more information about
++ * the Asterisk project. Please do not directly contact
++ * any of the maintainers of this project for assistance;
++ * the project provides a web site, mailing lists and IRC
++ * channels for your use.
++ *
++ * This program is free software, distributed under the terms of
++ * the GNU General Public License Version 2. See the LICENSE file
++ * at the top of the source tree.
++ */
++
++/*! \file
++ *
++ * \brief OGG/Speex streams.
++ * \arg File name extension: spx
++ * \ingroup formats
++ */
++
++/*** MODULEINFO
++	<depend>speex</depend>
++	<depend>ogg</depend>
++	<support_level>extended</support_level>
++ ***/
++
++#include "asterisk.h"
++
++ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
++
++#include "asterisk/mod_format.h"
++#include "asterisk/module.h"
++
++#include <speex/speex_header.h>
++#include <ogg/ogg.h>
++
++#define BLOCK_SIZE	4096		/* buffer size for feeding OGG routines */
++#define	BUF_SIZE	200
++
++struct speex_desc {	/* format specific parameters */
++	/* structures for handling the Ogg container */
++	ogg_sync_state oy;
++	ogg_stream_state os;
++	ogg_page og;
++	ogg_packet op;
++
++	int format_id;
++	int serialno;
++
++	/*! \brief Indicates whether an End of Stream condition has been detected. */
++	int eos;
++};
++
++static int read_packet(struct ast_filestream *fs)
++{
++	struct speex_desc *s = (struct speex_desc *)fs->_private;
++	char *buffer;
++	int result;
++	size_t bytes;
++
++	while (1) {
++		/* Get one packet */
++		result = ogg_stream_packetout(&s->os, &s->op);
++		if (result > 0) {
++			if (s->op.bytes>=5 && !memcmp(s->op.packet, "Speex", 5))
++				s->serialno = s->os.serialno;
++			if (s->serialno == -1 || s->os.serialno != s->serialno)
++				continue;
++			return 0;
++		}
++
++		if (result < 0)
++			ast_log(LOG_WARNING,
++				"Corrupt or missing data at this page position; continuing...\n");
++
++		/* No more packets left in the current page... */
++		if (s->eos) {
++			/* No more pages left in the stream */
++			return -1;
++		}
++
++		while (!s->eos) {
++			/* See if OGG has any pages in it's internal buffers */
++			result = ogg_sync_pageout(&s->oy, &s->og);
++			if (result > 0) {
++				/* Read all streams. */
++				if (ogg_page_serialno(&s->og) != s->os.serialno)
++					ogg_stream_reset_serialno(&s->os, ogg_page_serialno(&s->og));
++				/* Yes, OGG has more pages in it's internal buffers,
++				   add the page to the stream state */
++				result = ogg_stream_pagein(&s->os, &s->og);
++				if (result == 0) {
++					/* Yes, got a new,valid page */
++					if (ogg_page_eos(&s->og) &&
++					    ogg_page_serialno(&s->og) == s->serialno)
++						s->eos = 1;
++					break;
++				}
++				ast_log(LOG_WARNING,
++					"Invalid page in the bitstream; continuing...\n");
++			}
++
++			if (result < 0)
++				ast_log(LOG_WARNING,
++					"Corrupt or missing data in bitstream; continuing...\n");
++
++			/* No, we need to read more data from the file descrptor */
++			/* get a buffer from OGG to read the data into */
++			buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
++			bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
++			ogg_sync_wrote(&s->oy, bytes);
++			if (bytes == 0)
++				s->eos = 1;
++		}
++	}
++}
++
++/*!
++ * \brief Create a new OGG/Speex filestream and set it up for reading.
++ * \param fs File that points to on disk storage of the OGG/Speex data.
++ * \param expected_rate The expected Speex format (sampling rate).
++ * \return The new filestream.
++ */
++static int ogg_speex_open(struct ast_filestream *fs, int format_id, int expected_rate)
++{
++	char *buffer;
++	size_t bytes;
++	struct speex_desc *s = (struct speex_desc *)fs->_private;
++	SpeexHeader *hdr = NULL;
++	int i, result;
++
++	s->format_id = format_id;
++	s->serialno = -1;
++	ogg_sync_init(&s->oy);
++
++	buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
++	bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
++	ogg_sync_wrote(&s->oy, bytes);
++
++	result = ogg_sync_pageout(&s->oy, &s->og);
++	if (result != 1) {
++		if(bytes < BLOCK_SIZE) {
++			ast_log(LOG_ERROR, "Run out of data...\n");
++		} else {
++			ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n");
++		}
++		ogg_sync_clear(&s->oy);
++		return -1;
++	}
++
++	ogg_stream_init(&s->os, ogg_page_serialno(&s->og));
++	if (ogg_stream_pagein(&s->os, &s->og) < 0) { 
++		ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
++		goto error;
++	}
++
++	if (read_packet(fs) < 0) {
++		ast_log(LOG_ERROR, "Error reading initial header packet.\n");
++		goto error;
++	}
++
++	hdr = speex_packet_to_header((char*)s->op.packet, s->op.bytes);
++	if (memcmp(hdr->speex_string, "Speex   ", 8)) {
++		ast_log(LOG_ERROR, "OGG container does not contain Speex audio!\n");
++		goto error;
++	}
++	if (hdr->frames_per_packet != 1) {
++		ast_log(LOG_ERROR, "Only one frame-per-packet OGG/Speex files are currently supported!\n");
++		goto error;
++	}
++	if (hdr->nb_channels != 1) {
++		ast_log(LOG_ERROR, "Only monophonic OGG/Speex files are currently supported!\n");
++		goto error;
++	}
++	if (hdr->rate != expected_rate) {
++		ast_log(LOG_ERROR, "Unexpected sampling rate (%d != %d)!\n",
++			hdr->rate, expected_rate);
++		goto error;
++	}
++
++	/* this packet is the comment */
++	if (read_packet(fs) < 0) {
++		ast_log(LOG_ERROR, "Error reading comment packet.\n");
++		goto error;
++	}
++	for (i = 0; i < hdr->extra_headers; i++) {
++		if (read_packet(fs) < 0) {
++			ast_log(LOG_ERROR, "Error reading extra header packet %d.\n", i+1);
++			goto error;
++		}
++	}
++	free(hdr);
++
++	return 0;
++error:
++	if (hdr)
++		free(hdr);
++	ogg_stream_clear(&s->os);
++	ogg_sync_clear(&s->oy);
++	return -1;
++}
++
++/*!
++ * \brief Close a OGG/Speex filestream.
++ * \param fs A OGG/Speex filestream.
++ */
++static void ogg_speex_close(struct ast_filestream *fs)
++{
++	struct speex_desc *s = (struct speex_desc *)fs->_private;
++
++	ogg_stream_clear(&s->os);
++	ogg_sync_clear(&s->oy);
++}
++
++/*!
++ * \brief Read a frame full of audio data from the filestream.
++ * \param fs The filestream.
++ * \param whennext Number of sample times to schedule the next call.
++ * \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
++ */
++static struct ast_frame *ogg_speex_read(struct ast_filestream *fs,
++					 int *whennext)
++{
++	struct speex_desc *s = (struct speex_desc *)fs->_private;
++
++	if (read_packet(fs) < 0)
++		return NULL;
++
++	fs->fr.frametype = AST_FRAME_VOICE;
++	ast_format_set(&fs->fr.subclass.format, s->format_id, 0);
++	fs->fr.mallocd = 0;
++	AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
++	memcpy(fs->fr.data.ptr, s->op.packet, s->op.bytes);
++	fs->fr.datalen = s->op.bytes;
++	fs->fr.samples = *whennext = ast_codec_get_samples(&fs->fr);
++
++	return &fs->fr;
++}
++
++/*!
++ * \brief Trucate an OGG/Speex filestream.
++ * \param s The filestream to truncate.
++ * \return 0 on success, -1 on failure.
++ */
++
++static int ogg_speex_trunc(struct ast_filestream *s)
++{
++	ast_log(LOG_WARNING, "Truncation is not supported on OGG/Speex streams!\n");
++	return -1;
++}
++
++/*!
++ * \brief Seek to a specific position in an OGG/Speex filestream.
++ * \param s The filestream to truncate.
++ * \param sample_offset New position for the filestream, measured in 8KHz samples.
++ * \param whence Location to measure 
++ * \return 0 on success, -1 on failure.
++ */
++static int ogg_speex_seek(struct ast_filestream *s, off_t sample_offset, int whence)
++{
++	ast_log(LOG_WARNING, "Seeking is not supported on OGG/Speex streams!\n");
++	return -1;
++}
++
++static off_t ogg_speex_tell(struct ast_filestream *s)
++{
++	ast_log(LOG_WARNING, "Telling is not supported on OGG/Speex streams!\n");
++	return -1;
++}
++
++static int ogg_speex_open_nb(struct ast_filestream *fs) 
++{
++	return ogg_speex_open(fs, AST_FORMAT_SPEEX, 8000);
++}
++
++static struct ast_format_def speex_f = {
++	.name = "ogg_speex",
++	.exts = "spx",
++	.open = ogg_speex_open_nb,
++	.seek = ogg_speex_seek,
++	.trunc = ogg_speex_trunc,
++	.tell = ogg_speex_tell,
++	.read = ogg_speex_read,
++	.close = ogg_speex_close,
++	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
++	.desc_size = sizeof(struct speex_desc),
++};
++
++static int ogg_speex_open_wb(struct ast_filestream *fs) 
++{
++	return ogg_speex_open(fs, AST_FORMAT_SPEEX16, 16000);
++}
++
++static struct ast_format_def speex16_f = {
++	.name = "ogg_speex16",
++	.exts = "spx16",
++	.open = ogg_speex_open_wb,
++	.seek = ogg_speex_seek,
++	.trunc = ogg_speex_trunc,
++	.tell = ogg_speex_tell,
++	.read = ogg_speex_read,
++	.close = ogg_speex_close,
++	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
++	.desc_size = sizeof(struct speex_desc),
++};
++
++static int ogg_speex_open_uwb(struct ast_filestream *fs) 
++{
++	return ogg_speex_open(fs, AST_FORMAT_SPEEX32, 32000);
++}
++
++static struct ast_format_def speex32_f = {
++	.name = "ogg_speex32",
++	.exts = "spx32",
++	.open = ogg_speex_open_uwb,
++	.seek = ogg_speex_seek,
++	.trunc = ogg_speex_trunc,
++	.tell = ogg_speex_tell,
++	.read = ogg_speex_read,
++	.close = ogg_speex_close,
++	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
++	.desc_size = sizeof(struct speex_desc),
++};
++
++static int load_module(void)
++{
++	ast_format_set(&speex_f.format, AST_FORMAT_SPEEX, 0);
++	ast_format_set(&speex16_f.format, AST_FORMAT_SPEEX16, 0);
++	ast_format_set(&speex32_f.format, AST_FORMAT_SPEEX32, 0);
++
++	if (ast_format_def_register(&speex_f) ||
++	    ast_format_def_register(&speex16_f) ||
++	    ast_format_def_register(&speex32_f))
++		return AST_MODULE_LOAD_FAILURE;
++
++	return AST_MODULE_LOAD_SUCCESS;
++}
++
++static int unload_module(void)
++{
++	int res = 0;
++	res |= ast_format_def_unregister(speex_f.name);
++	res |= ast_format_def_unregister(speex16_f.name);
++	res |= ast_format_def_unregister(speex32_f.name);
++	return res;
++}
++
++AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Speex audio",
++	.load = load_module,
++	.unload = unload_module,
++	.load_pri = AST_MODPRI_APP_DEPEND
++);