main/asterisk: upgrade to 11.0.0 from testing/asterisk

- re-enable libasteriskssl as the uclibc bug should be fixed now

12 files changed, 0 insertions, 1787 deletions

diff --git a/testing/asterisk/100-uclibc-daemon.patch b/testing/asterisk/100-uclibc-daemon.patch
deleted file mode 100644
index 4956791d4d..0000000000
--- a/testing/asterisk/100-uclibc-daemon.patch
+++ /dev/null
@@ -1,44 +0,0 @@
-diff -Nru asterisk-1.6.1-beta4.org/main/asterisk.c asterisk-1.6.1-beta4/main/asterisk.c
---- asterisk-1.6.1-beta4.org/main/asterisk.c	2008-12-12 23:05:58.000000000 +0100
-+++ asterisk-1.6.1-beta4/main/asterisk.c	2008-12-23 15:28:21.000000000 +0100
-@@ -3295,9 +3295,40 @@
- #if HAVE_WORKING_FORK
- 	if (ast_opt_always_fork || !ast_opt_no_fork) {
- #ifndef HAVE_SBIN_LAUNCHD
-+#ifndef __UCLIBC__
- 		if (daemon(1, 0) < 0) {
- 			ast_log(LOG_ERROR, "daemon() failed: %s\n", strerror(errno));
- 		}
-+#else
-+		/*
-+		 * workaround for uClibc-0.9.29 mipsel bug:
-+		 * recursive mutexes do not work if uClibc daemon() function has been called,
-+		 * if parent thread locks a mutex
-+		 * the child thread cannot acquire a lock with the same name
-+		 * (same code works if daemon() is not called)
-+		 * but duplication of uClibc daemon.c code in here does work.
-+		 */
-+		int fd;
-+		switch (fork()) {
-+			case -1:
-+				exit(1);
-+			case 0:
-+				break;
-+			default:
-+				_exit(0);
-+		}
-+		if (setsid() == -1)
-+			exit(1);
-+		if (fork())
-+			_exit(0);
-+		if ((fd = open("/dev/null", O_RDWR, 0)) != -1) {
-+			dup2(fd, STDIN_FILENO);
-+			dup2(fd, STDOUT_FILENO);
-+			dup2(fd, STDERR_FILENO);
-+			if (fd > 2)
-+				close(fd);
-+		}
-+#endif
- 		ast_mainpid = getpid();
- 		/* Blindly re-write pid file since we are forking */
- 		unlink(ast_config_AST_PID);
diff --git a/testing/asterisk/101-caps-uclibc.patch b/testing/asterisk/101-caps-uclibc.patch
deleted file mode 100644
index bb32d1ece1..0000000000
--- a/testing/asterisk/101-caps-uclibc.patch
+++ /dev/null
@@ -1,17 +0,0 @@
---- asterisk-1.6.0.18/configure.ac.orig	Mon Oct 26 23:13:28 2009
-+++ asterisk-1.6.0.18/configure.ac	Fri Nov 27 21:42:36 2009
-@@ -627,9 +627,11 @@
- 
- AST_EXT_LIB_CHECK([CURSES], [curses], [initscr], [curses.h])
- 
--if test "x${OSARCH}" = "xlinux-gnu" ; then
--  AST_EXT_LIB_CHECK([CAP], [cap], [cap_from_text], [sys/capability.h])
--fi
-+case "${OSARCH}" in
-+    linux*)         
-+    AST_EXT_LIB_CHECK([CAP], [cap], [cap_from_text], [sys/capability.h])
-+    ;;                                                                  
-+esac  
- 
- AST_C_DEFINE_CHECK([DAHDI], [DAHDI_CODE], [dahdi/user.h])
- 
diff --git a/testing/asterisk/102-gsm-pic.patch b/testing/asterisk/102-gsm-pic.patch
deleted file mode 100644
index 71370ec0b7..0000000000
--- a/testing/asterisk/102-gsm-pic.patch
+++ /dev/null
@@ -1,54 +0,0 @@
----  a/codecs/gsm/Makefile.org	2008-03-29 11:33:09.000000000 +0100
-+++  b/codecs/gsm/Makefile	2008-03-29 11:44:40.000000000 +0100
-@@ -37,23 +37,6 @@
- ######### ppro's, etc, as well as the AMD K6 and K7.  The compile will
- ######### probably require gcc. 
- 
--ifeq (, $(findstring $(OSARCH) , Darwin SunOS ))
--ifeq (, $(findstring $(PROC) , x86_64 amd64 ultrasparc sparc64 arm armv5b armeb ppc powerpc ppc64 ia64 s390 bfin mipsel mips))
--ifeq (, $(findstring $(shell uname -m) , ppc ppc64 alpha armv4l s390 ))
--OPTIMIZE+=-march=$(PROC)
--endif
--endif
--endif
--
--#The problem with sparc is the best stuff is in newer versions of gcc (post 3.0) only.
--#This works for even old (2.96) versions of gcc and provides a small boost either way.
--#A ultrasparc cpu is really v9 but the stock debian stable 3.0 gcc doesn't support it.
--#So we go lowest common available by gcc and go a step down, still a step up from
--#the default as we now have a better instruction set to work with. - Belgarath
--ifeq ($(PROC),ultrasparc)
--OPTIMIZE+=-mcpu=v8 -mtune=$(PROC) -O3 
--endif
--
- PG =
- #PG = -g -pg
- ######### Profiling flags.  If you don't know what that means, leave it blank.
-@@ -208,12 +191,10 @@
- # XXX Keep a space after each findstring argument
- # XXX should merge with GSM_OBJECTS
- ifeq ($(OSARCH),linux-gnu)
--ifeq (,$(findstring $(shell uname -m) , x86_64 amd64 ppc ppc64 alpha armv4l sparc64 parisc s390 ))
--ifeq (,$(findstring $(PROC) , arm armv5b armeb powerpc ia64 s390 bfin mipsel mips ))
-+ifneq ($(K6OPT),)
- GSM_SOURCES+= $(SRC)/k6opt.s
- endif
- endif
--endif
- 
- TOAST_SOURCES = $(SRC)/toast.c 		\
- 		$(SRC)/toast_lin.c	\
-@@ -260,12 +241,10 @@
- 		$(SRC)/table.o
- 
- ifeq ($(OSARCH),linux-gnu)
--ifeq (,$(findstring $(shell uname -m) , x86_64 amd64 ppc ppc64 alpha armv4l sparc64 parisc ))
--ifeq (,$(findstring $(PROC) , arm armv5b armeb powerpc ia64 bfin mipsel mips ))
-+ifneq ($(K6OPT),)
- GSM_OBJECTS+= $(SRC)/k6opt.o
- endif
- endif
--endif
- 
- TOAST_OBJECTS =	$(SRC)/toast.o 		\
- 		$(SRC)/toast_lin.o	\
diff --git a/testing/asterisk/APKBUILD b/testing/asterisk/APKBUILD
deleted file mode 100644
index 5220f32649..0000000000
--- a/testing/asterisk/APKBUILD
+++ /dev/null
@@ -1,193 +0,0 @@
-# Contributor: Timo Teras <timo.teras@iki.fi>
-# Maintainer: Timo Teras <timo.teras@iki.fi>
-pkgname=asterisk
-pkgver=11.0.0
-pkgrel=1
-pkgdesc="Asterisk: A Module Open Source PBX System"
-pkgusers="asterisk"
-pkggroups="asterisk"
-url="http://www.asterisk.org/"
-arch="all"
-license="GPL"
-depends=
-makedepends="autoconf automake libtool ncurses-dev popt-dev newt-dev zlib-dev
-	postgresql-dev unixodbc-dev dahdi-tools-dev libpri-dev tar
-	freetds-dev openssl-dev lua-dev alsa-lib-dev spandsp-dev tiff-dev
-	libresample sqlite-dev wget speex-dev libogg-dev bluez-dev"
-install="$pkgname.pre-install $pkgname.pre-upgrade"
-subpackages="$pkgname-dev $pkgname-doc $pkgname-pgsql $pkgname-odbc
-	$pkgname-tds $pkgname-fax $pkgname-sample-config:sample
-	$pkgname-sounds-moh:sound_moh $pkgname-sounds-en:sound_en
-	$pkgname-mobile"
-source="http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-${pkgver/_/-}.tar.gz
-	100-uclibc-daemon.patch
-	101-caps-uclibc.patch
-	ASTERISK-18995.patch
-	ASTERISK-19109.patch
-	ASTERISK-20527.patch
-	asterisk.initd
-	asterisk.confd
-	asterisk.logrotate"
-
-_builddir="$srcdir/$pkgname-${pkgver/_/-}"
-
-prepare() {
-	cd "$_builddir"
-	for i in $source; do
-		case $i in
-		*.patch) msg "$i"; patch -p1 -i "$srcdir"/$i || return 1;;
-		esac
-	done
-
-	sed -i -e 's:lua5.1/::' pbx/pbx_lua.c
-	sed -i -e 's/PBX_ICONV=1/PBX_ICONV=0/g' configure.ac
-	sed -i -e 's/int foo = res_ninit(NULL);/res_ninit_is_not_really_here();/g' configure.ac
-
-	./bootstrap.sh
-}
-
-build() {
-	cd "$_builddir"
-	SHA1SUM="$PWD"/build_tools/sha1sum-sh ./configure --prefix=/usr \
-		--sysconfdir=/etc \
-		--mandir=/usr/share/man \
-		--infodir=/usr/share/info \
-		--libdir=/usr/lib \
-		--localstatedir=/var \
-		--disable-xmldoc --with-gsm=internal \
-		--without-iconv --with-popt --with-z --with-newt \
-		--with-unixodbc --with-postgres --with-tds \
-		--with-dahdi --with-pri --with-tonezone \
-		--with-resample \
-		--with-sqlite3 \
-		--with-speex \
-		--with-asound \
-		--without-x11 \
-		--with-spandsp \
-		--with-bluetooth \
-		--disable-asteriskssl \
-		|| return 1
-
-	# and figure out which modules to build
-	rm menuselect.makeopts
-	make menuselect.makeopts
-	# enable chan_mobile
-	sed -i -e '/^MENUSELECT_ADDONS=/s/chan_mobile//' menuselect.makeopts
-	make ASTCFLAGS="$CFLAGS" ASTLDFLAGS="$LDFLAGS" || return 1
-}
-
-package() {
-	cd "$_builddir"
-	make -j1 DESTDIR="$pkgdir" install
-
-	install -d "$pkgdir"/var/run/asterisk
-	install -d "$pkgdir"/var/lib/asterisk
-
-	install -m755 -D "$srcdir"/$pkgname.initd "$pkgdir"/etc/init.d/$pkgname
-	install -m644 -D "$srcdir"/$pkgname.confd "$pkgdir"/etc/conf.d/$pkgname
-	install -m644 -D "$srcdir"/$pkgname.logrotate \
-		"$pkgdir"/etc/logrotate.d/$pkgname
-
-	chown -R asterisk:asterisk "$pkgdir"/var/*/asterisk
-	chown -R asterisk:asterisk "$pkgdir"/etc/asterisk
-	chmod -R u=rwX,g=rX,o= "$pkgdir"/etc/asterisk
-}
-
-_move_dir() {
-	for DIR in "$@"; do
-		local dest=`dirname "$subpkgdir/$DIR"`
-		echo mkdir -p $dest
-		mkdir -p "$dest"
-		echo mv "$pkgdir"/$DIR $dest
-		mv "$pkgdir"/"$DIR" "$dest"
-	done
-}
-
-_find_and_move() {
-	local pattern="$1"
-	cd "$pkgdir" || return 1
-	find -name "$pattern" -type f | while read f; do
-		local dest="$subpkgdir/${f%/*}"
-		mkdir -p "$dest"
-		mv "$f" "$dest"
-	done
-}
-
-doc() {
-	default_doc
-}
-
-dev() {
-	default_dev
-	depends="asterisk"
-}
-
-pgsql() {
-	depends=
-	install=
-	_find_and_move '*_pgsql*'
-}
-
-odbc() {
-	depends=
-	install=
-	_find_and_move '*odbc*'
-}
-
-tds() {
-	depends=
-	install=
-	_find_and_move '*_tds*'
-}
-
-fax() {
-        depends=
-        install=
-        _find_and_move '*_fax*'
-}
-
-mobile() {
-        depends=
-        install=
-        _find_and_move '*_mobile*'
-}
-
-sample() {
-	arch="noarch"
-	pkgdesc="Sample configuration files for asterisk"
-	cd "$_builddir"
-	mkdir -p "$subpkgdir"/var/lib/asterisk/phoneprov
-	make -j1 samples DESTDIR="$subpkgdir"
-
-	chown -R asterisk:asterisk "$subpkgdir"/var/*/asterisk
-	chown -R asterisk:asterisk "$subpkgdir"/etc/asterisk
-	chmod -R u=rwX,g=rX,o= "$subpkgdir"/etc/asterisk
-}
-
-sound_moh() {
-	arch="noarch"
-	pkgdesc="Default on-hold music files for asterisk"
-	depends=
-	install=
-	_move_dir var/lib/asterisk/moh
-	chown -R asterisk:asterisk "$subpkgdir"/var/*/asterisk
-}
-
-sound_en() {
-	arch="noarch"
-	pkgdesc="English sound files for asterisk"
-	depends=
-	install=
-	_move_dir var/lib/asterisk/sounds/en
-	chown -R asterisk:asterisk "$subpkgdir"/var/*/asterisk
-}
-
-md5sums="e23c8535a425253764bdddeee49d1778  asterisk-11.0.0.tar.gz
-b00c9d98ce2ad445501248a197c6e436  100-uclibc-daemon.patch
-6e1129e30c4fd2c25c86c81685a485a9  101-caps-uclibc.patch
-bc6713f5434e07b79d3afdd155461d72  ASTERISK-18995.patch
-146befabe95798a67c58d8ac00d397a6  ASTERISK-19109.patch
-676ca42ee1859d8a7bae4345ede5eb89  ASTERISK-20527.patch
-74cd25a5638a94ef51e9f4ede2fd28f2  asterisk.initd
-ed31d7ba37bcf8b0346dcf8593c395f0  asterisk.confd
-3e65172275684373e1a25c8a11224411  asterisk.logrotate"
diff --git a/testing/asterisk/ASTERISK-18995.patch b/testing/asterisk/ASTERISK-18995.patch
deleted file mode 100644
index cd144847be..0000000000
--- a/testing/asterisk/ASTERISK-18995.patch
+++ /dev/null
@@ -1,358 +0,0 @@
---- /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
-+);
diff --git a/testing/asterisk/ASTERISK-19109.patch b/testing/asterisk/ASTERISK-19109.patch
deleted file mode 100644
index cd45b42731..0000000000
--- a/testing/asterisk/ASTERISK-19109.patch
+++ /dev/null
@@ -1,724 +0,0 @@
-From 806946c35cf0560248e63fea53c4d82426a2034a Mon Sep 17 00:00:00 2001
-From: =?UTF-8?q?Timo=20Ter=C3=A4s?= <timo.teras@iki.fi>
-Date: Wed, 5 Sep 2012 10:07:05 +0300
-Subject: [PATCH] ASTERISK-19109: Implement deaf participant support for
- ConfBridge
-
----
- CHANGES                                |   3 +
- apps/app_confbridge.c                  | 218 ++++++++++++++++++++++++++++++---
- apps/confbridge/conf_config_parser.c   |  14 +++
- apps/confbridge/include/confbridge.h   |   6 +
- bridges/bridge_multiplexed.c           |   2 +-
- bridges/bridge_simple.c                |   2 +-
- bridges/bridge_softmix.c               |  38 +++---
- configs/confbridge.conf.sample         |   5 +
- include/asterisk/bridging_features.h   |   2 +
- include/asterisk/bridging_technology.h |  15 +++
- main/bridging.c                        |  22 ++++
- 11 files changed, 294 insertions(+), 33 deletions(-)
-
-diff --git a/CHANGES b/CHANGES
-index c3c9891..2efa17b 100644
---- a/CHANGES
-+++ b/CHANGES
-@@ -66,6 +66,9 @@ ConfBridge
-    file will be played to the user, and only the user, upon joining the
-    conference bridge.
- 
-+ * Added support for deaf participants with CLI commands, manager actions
-+   and ConfBridge DTMF actions to toggle the deaf state.
-+
- 
- Dial
- -------------------
-diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c
-index 90954b8..0455be4 100644
---- a/apps/app_confbridge.c
-+++ b/apps/app_confbridge.c
-@@ -185,6 +185,30 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
- 		<description>
- 		</description>
- 	</manager>
-+	<manager name="ConfbridgeDeafen" language="en_US">
-+		<synopsis>
-+			Deafen a Confbridge user.
-+		</synopsis>
-+		<syntax>
-+			<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
-+			<parameter name="Conference" required="true" />
-+			<parameter name="Channel" required="true" />
-+		</syntax>
-+		<description>
-+		</description>
-+	</manager>
-+	<manager name="ConfbridgeUndeafen" language="en_US">
-+		<synopsis>
-+			Undeafen a Confbridge user.
-+		</synopsis>
-+		<syntax>
-+			<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
-+			<parameter name="Conference" required="true" />
-+			<parameter name="Channel" required="true" />
-+		</syntax>
-+		<description>
-+		</description>
-+	</manager>
- 	<manager name="ConfbridgeKick" language="en_US">
- 		<synopsis>
- 			Kick a Confbridge user.
-@@ -273,6 +297,13 @@ static const char app[] = "ConfBridge";
- /* Number of buckets our conference bridges container can have */
- #define CONFERENCE_BRIDGE_BUCKETS 53
- 
-+enum confbridge_feature_action {
-+	CONFBRIDGE_FEATURE_MUTE,
-+	CONFBRIDGE_FEATURE_UNMUTE,
-+	CONFBRIDGE_FEATURE_DEAFEN,
-+	CONFBRIDGE_FEATURE_UNDEAFEN
-+};
-+
- /*! \brief Container to hold all conference bridges in progress */
- static struct ao2_container *conference_bridges;
- 
-@@ -311,6 +342,10 @@ const char *conf_get_sound(enum conf_sounds sound, struct bridge_profile_sounds
- 		return S_OR(custom_sounds->muted, "conf-muted");
- 	case CONF_SOUND_UNMUTED:
- 		return S_OR(custom_sounds->unmuted, "conf-unmuted");
-+	case CONF_SOUND_DEAFENED:
-+		return S_OR(custom_sounds->deafened, "conf-deafened");
-+	case CONF_SOUND_UNDEAFENED:
-+		return S_OR(custom_sounds->undeafened, "conf-undeafened");
- 	case CONF_SOUND_ONLY_ONE:
- 		return S_OR(custom_sounds->onlyone, "conf-onlyone");
- 	case CONF_SOUND_THERE_ARE:
-@@ -1504,10 +1539,13 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
- 	volume_adjustments[0] = ast_audiohook_volume_get(chan, AST_AUDIOHOOK_DIRECTION_READ);
- 	volume_adjustments[1] = ast_audiohook_volume_get(chan, AST_AUDIOHOOK_DIRECTION_WRITE);
- 
--	/* If the caller should be joined already muted, make it so */
-+	/* If the caller should be joined already muted or deaf, make it so */
- 	if (ast_test_flag(&conference_bridge_user.u_profile, USER_OPT_STARTMUTED)) {
- 		conference_bridge_user.features.mute = 1;
- 	}
-+	if (ast_test_flag(&conference_bridge_user.u_profile, USER_OPT_STARTDEAF)) {
-+		conference_bridge_user.features.deaf = 1;
-+	}
- 
- 	if (ast_test_flag(&conference_bridge_user.u_profile, USER_OPT_DROP_SILENCE)) {
- 		conference_bridge_user.tech_args.drop_silence = 1;
-@@ -1668,6 +1706,20 @@ static int action_toggle_mute_participants(struct conference_bridge *conference_
- 	return 0;
- }
- 
-+static int action_toggle_deaf(struct conference_bridge *conference_bridge,
-+	struct conference_bridge_user *conference_bridge_user,
-+	struct ast_channel *chan)
-+{
-+	/* Deafen or undeafen yourself */
-+	conference_bridge_user->features.deaf = (!conference_bridge_user->features.deaf ? 1 : 0);
-+	ast_test_suite_event_notify("CONF_DEAF", "Message: participant %s %s\r\nConference: %s\r\nChannel: %s", chan->name, conference_bridge_user->features.deaf ? "deafened" : "undeafened", conference_bridge_user->b_profile.name, chan->name);
-+
-+	return ast_stream_and_wait(chan, (conference_bridge_user->features.deaf ?
-+		conf_get_sound(CONF_SOUND_DEAFENED, conference_bridge_user->b_profile.sounds) :
-+		conf_get_sound(CONF_SOUND_UNDEAFENED, conference_bridge_user->b_profile.sounds)),
-+		"");
-+}
-+
- static int action_playback(struct ast_bridge_channel *bridge_channel, const char *playback_file)
- {
- 	char *file_copy = ast_strdupa(playback_file);
-@@ -1856,6 +1908,11 @@ static int execute_menu_entry(struct conference_bridge *conference_bridge,
- 		case MENU_ACTION_PARTICIPANT_COUNT:
- 			announce_user_count(conference_bridge, conference_bridge_user);
- 			break;
-+		case MENU_ACTION_TOGGLE_DEAF:
-+			res |= action_toggle_deaf(conference_bridge,
-+				conference_bridge_user,
-+				bridge_channel->chan);
-+			break;
- 		case MENU_ACTION_PLAYBACK:
- 			if (!stop_prompts) {
- 				res |= action_playback(bridge_channel, menu_action->data.playback_file);
-@@ -2119,13 +2176,13 @@ static int generic_lock_unlock_helper(int lock, const char *conference)
- }
- 
- /* \internal
-- * \brief finds a conference user by channel name and mutes/unmutes them.
-+ * \brief finds a conference user by channel name and changes feature bits on it.
-  *
-  * \retval 0 success
-  * \retval -1 conference not found
-  * \retval -2 user not found
-  */
--static int generic_mute_unmute_helper(int mute, const char *conference, const char *user)
-+static int generic_feature_action_helper(enum confbridge_feature_action action, const char *conference, const char *user)
- {
- 	struct conference_bridge *bridge = NULL;
- 	struct conference_bridge tmp;
-@@ -2143,10 +2200,44 @@ static int generic_mute_unmute_helper(int mute, const char *conference, const ch
- 		}
- 	}
- 	if (participant) {
--		participant->features.mute = mute;
--		ast_test_suite_event_notify("CONF_MUTE", "Message: participant %s %s\r\nConference: %s\r\nChannel: %s", ast_channel_name(participant->chan), participant->features.mute ? "muted" : "unmuted", bridge->b_profile.name, ast_channel_name(participant->chan));
-+		const char *state, *verb;
-+
-+		switch (action) {
-+		case CONFBRIDGE_FEATURE_DEAFEN:
-+			participant->features.deaf = 1;
-+			state = "CONF_DEAF";
-+			verb = "deafened";
-+			break;
-+		case CONFBRIDGE_FEATURE_UNDEAFEN:
-+			participant->features.deaf = 0;
-+			state = "CONF_DEAF";
-+			verb = "undeafened";
-+			break;
-+		case CONFBRIDGE_FEATURE_MUTE:
-+			participant->features.mute = 1;
-+			state = "CONF_MUTE";
-+			verb = "muted";
-+			break;
-+		case CONFBRIDGE_FEATURE_UNMUTE:
-+		default:
-+			participant->features.mute = 0;
-+			state = "CONF_MUTE";
-+			verb = "unmuted";
-+			break;
-+		}
-+
-+		if (state != NULL && verb != NULL) {
-+			ast_test_suite_event_notify(state,
-+				"Message: participant %s %s\r\n"
-+				"Conference: %s\r\n"
-+				"Channel: %s",
-+				ast_channel_name(participant->chan),
-+				verb,
-+				bridge->b_profile.name,
-+				ast_channel_name(participant->chan));
-+		}
- 	} else {
--		res = -2;;
-+		res = -2;
- 	}
- 	ao2_unlock(bridge);
- 	ao2_ref(bridge, -1);
-@@ -2154,9 +2245,10 @@ static int generic_mute_unmute_helper(int mute, const char *conference, const ch
- 	return res;
- }
- 
--static int cli_mute_unmute_helper(int mute, struct ast_cli_args *a)
-+static int cli_feature_action_helper(enum confbridge_feature_action action, struct ast_cli_args *a)
- {
--	int res = generic_mute_unmute_helper(mute, a->argv[2], a->argv[3]);
-+	const char *verb;
-+	int res = generic_feature_action_helper(action, a->argv[2], a->argv[3]);
- 
- 	if (res == -1) {
- 		ast_cli(a->fd, "No conference bridge named '%s' found!\n", a->argv[2]);
-@@ -2165,7 +2257,24 @@ static int cli_mute_unmute_helper(int mute, struct ast_cli_args *a)
- 		ast_cli(a->fd, "No channel named '%s' found in conference %s\n", a->argv[3], a->argv[2]);
- 		return -1;
- 	}
--	ast_cli(a->fd, "%s %s from confbridge %s\n", mute ? "Muting" : "Unmuting", a->argv[3], a->argv[2]);
-+
-+	switch (action) {
-+	case CONFBRIDGE_FEATURE_DEAFEN:
-+		verb = "Deafening";
-+		break;
-+	case CONFBRIDGE_FEATURE_UNDEAFEN:
-+		verb = "Undeafening";
-+		break;
-+	case CONFBRIDGE_FEATURE_MUTE:
-+		verb = "Muting";
-+		break;
-+	case CONFBRIDGE_FEATURE_UNMUTE:
-+	default:
-+		verb = "Unmuting";
-+		break;
-+	}
-+
-+	ast_cli(a->fd, "%s %s from confbridge %s\n", verb, a->argv[3], a->argv[2]);
- 	return 0;
- }
- 
-@@ -2187,7 +2296,7 @@ static char *handle_cli_confbridge_mute(struct ast_cli_entry *e, int cmd, struct
- 		return CLI_SHOWUSAGE;
- 	}
- 
--	cli_mute_unmute_helper(1, a);
-+	cli_feature_action_helper(CONFBRIDGE_FEATURE_MUTE, a);
- 
- 	return CLI_SUCCESS;
- }
-@@ -2210,7 +2319,53 @@ static char *handle_cli_confbridge_unmute(struct ast_cli_entry *e, int cmd, stru
- 		return CLI_SHOWUSAGE;
- 	}
- 
--	cli_mute_unmute_helper(0, a);
-+	cli_feature_action_helper(CONFBRIDGE_FEATURE_UNMUTE, a);
-+
-+	return CLI_SUCCESS;
-+}
-+
-+static char *handle_cli_confbridge_deafen(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-+{
-+	switch (cmd) {
-+	case CLI_INIT:
-+		e->command = "confbridge deafen";
-+		e->usage =
-+			"Usage: confbridge deafen <conference> <channel>\n";
-+		return NULL;
-+	case CLI_GENERATE:
-+		if (a->pos == 2) {
-+			return complete_confbridge_name(a->line, a->word, a->pos, a->n);
-+		}
-+		return NULL;
-+	}
-+	if (a->argc != 4) {
-+		return CLI_SHOWUSAGE;
-+	}
-+
-+	cli_feature_action_helper(CONFBRIDGE_FEATURE_DEAFEN, a);
-+
-+	return CLI_SUCCESS;
-+}
-+
-+static char *handle_cli_confbridge_undeafen(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-+{
-+	switch (cmd) {
-+	case CLI_INIT:
-+		e->command = "confbridge undeafen";
-+		e->usage =
-+			"Usage: confbridge undeafen <conference> <channel>\n";
-+		return NULL;
-+	case CLI_GENERATE:
-+		if (a->pos == 2) {
-+			return complete_confbridge_name(a->line, a->word, a->pos, a->n);
-+		}
-+		return NULL;
-+	}
-+	if (a->argc != 4) {
-+		return CLI_SHOWUSAGE;
-+	}
-+
-+	cli_feature_action_helper(CONFBRIDGE_FEATURE_UNDEAFEN, a);
- 
- 	return CLI_SUCCESS;
- }
-@@ -2358,6 +2513,8 @@ static struct ast_cli_entry cli_confbridge[] = {
- 	AST_CLI_DEFINE(handle_cli_confbridge_kick, "Kick participants out of conference bridges."),
- 	AST_CLI_DEFINE(handle_cli_confbridge_mute, "Mute a participant."),
- 	AST_CLI_DEFINE(handle_cli_confbridge_unmute, "Unmute a participant."),
-+	AST_CLI_DEFINE(handle_cli_confbridge_deafen, "Deafen a participant."),
-+	AST_CLI_DEFINE(handle_cli_confbridge_undeafen, "Undeafen a participant."),
- 	AST_CLI_DEFINE(handle_cli_confbridge_lock, "Lock a conference."),
- 	AST_CLI_DEFINE(handle_cli_confbridge_unlock, "Unlock a conference."),
- 	AST_CLI_DEFINE(handle_cli_confbridge_start_record, "Start recording a conference"),
-@@ -2492,10 +2649,11 @@ static int action_confbridgelistrooms(struct mansession *s, const struct message
- 	return 0;
- }
- 
--static int action_mute_unmute_helper(struct mansession *s, const struct message *m, int mute)
-+static int action_feature_action_helper(struct mansession *s, const struct message *m, enum confbridge_feature_action action)
- {
- 	const char *conference = astman_get_header(m, "Conference");
- 	const char *channel = astman_get_header(m, "Channel");
-+	char *verb;
- 	int res = 0;
- 
- 	if (ast_strlen_zero(conference)) {
-@@ -2511,7 +2669,7 @@ static int action_mute_unmute_helper(struct mansession *s, const struct message
- 		return 0;
- 	}
- 
--	res = generic_mute_unmute_helper(mute, conference, channel);
-+	res = generic_feature_action_helper(action, conference, channel);
- 
- 	if (res == -1) {
- 		astman_send_error(s, m, "No Conference by that name found.");
-@@ -2521,17 +2679,41 @@ static int action_mute_unmute_helper(struct mansession *s, const struct message
- 		return 0;
- 	}
- 
--	astman_send_ack(s, m, mute ? "User muted" : "User unmuted");
-+	switch (action) {
-+	case CONFBRIDGE_FEATURE_DEAFEN:
-+		verb = "User deafened";
-+		break;
-+	case CONFBRIDGE_FEATURE_UNDEAFEN:
-+		verb = "User undeafened";
-+		break;
-+	case CONFBRIDGE_FEATURE_MUTE:
-+		verb = "User muted";
-+		break;
-+	case CONFBRIDGE_FEATURE_UNMUTE:
-+	default:
-+		verb = "User unmuted";
-+		break;
-+	}
-+
-+	astman_send_ack(s, m, verb);
- 	return 0;
- }
- 
- static int action_confbridgeunmute(struct mansession *s, const struct message *m)
- {
--	return action_mute_unmute_helper(s, m, 0);
-+	return action_feature_action_helper(s, m, CONFBRIDGE_FEATURE_UNMUTE);
- }
- static int action_confbridgemute(struct mansession *s, const struct message *m)
- {
--	return action_mute_unmute_helper(s, m, 1);
-+	return action_feature_action_helper(s, m, CONFBRIDGE_FEATURE_MUTE);
-+}
-+static int action_confbridgeundeafen(struct mansession *s, const struct message *m)
-+{
-+	return action_feature_action_helper(s, m, CONFBRIDGE_FEATURE_UNDEAFEN);
-+}
-+static int action_confbridgedeafen(struct mansession *s, const struct message *m)
-+{
-+	return action_feature_action_helper(s, m, CONFBRIDGE_FEATURE_DEAFEN);
- }
- 
- static int action_lock_unlock_helper(struct mansession *s, const struct message *m, int lock)
-@@ -2818,6 +3000,8 @@ static int unload_module(void)
- 	res |= ast_manager_unregister("ConfbridgeListRooms");
- 	res |= ast_manager_unregister("ConfbridgeMute");
- 	res |= ast_manager_unregister("ConfbridgeUnmute");
-+	res |= ast_manager_unregister("ConfbridgeDeafen");
-+	res |= ast_manager_unregister("ConfbridgeUndeafen");
- 	res |= ast_manager_unregister("ConfbridgeKick");
- 	res |= ast_manager_unregister("ConfbridgeUnlock");
- 	res |= ast_manager_unregister("ConfbridgeLock");
-@@ -2860,6 +3044,8 @@ static int load_module(void)
- 	res |= ast_manager_register_xml("ConfbridgeListRooms", EVENT_FLAG_REPORTING, action_confbridgelistrooms);
- 	res |= ast_manager_register_xml("ConfbridgeMute", EVENT_FLAG_CALL, action_confbridgemute);
- 	res |= ast_manager_register_xml("ConfbridgeUnmute", EVENT_FLAG_CALL, action_confbridgeunmute);
-+	res |= ast_manager_register_xml("ConfbridgeDeafen", EVENT_FLAG_CALL, action_confbridgedeafen);
-+	res |= ast_manager_register_xml("ConfbridgeUndeafen", EVENT_FLAG_CALL, action_confbridgeundeafen);
- 	res |= ast_manager_register_xml("ConfbridgeKick", EVENT_FLAG_CALL, action_confbridgekick);
- 	res |= ast_manager_register_xml("ConfbridgeUnlock", EVENT_FLAG_CALL, action_confbridgeunlock);
- 	res |= ast_manager_register_xml("ConfbridgeLock", EVENT_FLAG_CALL, action_confbridgelock);
-diff --git a/apps/confbridge/conf_config_parser.c b/apps/confbridge/conf_config_parser.c
-index f4a9604..8a02de7 100644
---- a/apps/confbridge/conf_config_parser.c
-+++ b/apps/confbridge/conf_config_parser.c
-@@ -279,6 +279,10 @@ static int set_sound(const char *sound_name, const char *sound_file, struct brid
- 		ast_string_field_set(sounds, muted, sound_file);
- 	} else if (!strcasecmp(sound_name, "sound_unmuted")) {
- 		ast_string_field_set(sounds, unmuted, sound_file);
-+	} else if (!strcasecmp(sound_name, "sound_deafened")) {
-+		ast_string_field_set(sounds, deafened, sound_file);
-+	} else if (!strcasecmp(sound_name, "sound_undeafened")) {
-+		ast_string_field_set(sounds, undeafened, sound_file);
- 	} else if (!strcasecmp(sound_name, "sound_there_are")) {
- 		ast_string_field_set(sounds, thereare, sound_file);
- 	} else if (!strcasecmp(sound_name, "sound_other_in_party")) {
-@@ -418,6 +422,7 @@ static int add_action_to_menu_entry(struct conf_menu_entry *menu_entry, enum con
- 	switch (id) {
- 	case MENU_ACTION_NOOP:
- 	case MENU_ACTION_TOGGLE_MUTE:
-+	case MENU_ACTION_TOGGLE_DEAF:
- 	case MENU_ACTION_INCREASE_LISTENING:
- 	case MENU_ACTION_DECREASE_LISTENING:
- 	case MENU_ACTION_INCREASE_TALKING:
-@@ -708,6 +713,9 @@ static char *handle_cli_confbridge_show_user_profile(struct ast_cli_entry *e, in
- 	ast_cli(a->fd,"Start Muted:             %s\n",
- 		u_profile.flags & USER_OPT_STARTMUTED?
- 		"true" : "false");
-+	ast_cli(a->fd,"Start Deaf:              %s\n",
-+		u_profile.flags & USER_OPT_STARTDEAF?
-+		"true" : "false");
- 	ast_cli(a->fd,"MOH When Empty:          %s\n",
- 		u_profile.flags & USER_OPT_MUSICONHOLD ?
- 		"enabled" : "disabled");
-@@ -896,6 +904,8 @@ static char *handle_cli_confbridge_show_bridge_profile(struct ast_cli_entry *e,
- 	ast_cli(a->fd,"sound_kicked:         %s\n", conf_get_sound(CONF_SOUND_KICKED, b_profile.sounds));
- 	ast_cli(a->fd,"sound_muted:          %s\n", conf_get_sound(CONF_SOUND_MUTED, b_profile.sounds));
- 	ast_cli(a->fd,"sound_unmuted:        %s\n", conf_get_sound(CONF_SOUND_UNMUTED, b_profile.sounds));
-+	ast_cli(a->fd,"sound_deafened:       %s\n", conf_get_sound(CONF_SOUND_DEAFENED, b_profile.sounds));
-+	ast_cli(a->fd,"sound_undeafened:     %s\n", conf_get_sound(CONF_SOUND_UNDEAFENED, b_profile.sounds));
- 	ast_cli(a->fd,"sound_there_are:      %s\n", conf_get_sound(CONF_SOUND_THERE_ARE, b_profile.sounds));
- 	ast_cli(a->fd,"sound_other_in_party: %s\n", conf_get_sound(CONF_SOUND_OTHER_IN_PARTY, b_profile.sounds));
- 	ast_cli(a->fd,"sound_place_into_conference: %s\n", conf_get_sound(CONF_SOUND_PLACE_IN_CONF, b_profile.sounds));
-@@ -1021,6 +1031,9 @@ static char *handle_cli_confbridge_show_menu(struct ast_cli_entry *e, int cmd, s
- 			case MENU_ACTION_TOGGLE_MUTE:
- 				ast_cli(a->fd, "toggle_mute");
- 				break;
-+			case MENU_ACTION_TOGGLE_DEAF:
-+				ast_cli(a->fd, "toggle_deaf");
-+				break;
- 			case MENU_ACTION_NOOP:
- 				ast_cli(a->fd, "no_op");
- 				break;
-@@ -1268,6 +1281,7 @@ int conf_load_config(int reload)
- 	aco_option_register(&cfg_info, "admin", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_ADMIN);
- 	aco_option_register(&cfg_info, "marked", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_MARKEDUSER);
- 	aco_option_register(&cfg_info, "startmuted", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_STARTMUTED);
-+	aco_option_register(&cfg_info, "startdeaf", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_STARTDEAF);
- 	aco_option_register(&cfg_info, "music_on_hold_when_empty", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_MUSICONHOLD);
- 	aco_option_register(&cfg_info, "quiet", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_QUIET);
- 	aco_option_register_custom(&cfg_info, "announce_user_count_all", ACO_EXACT, user_types, "no", announce_user_count_all_handler, 0);
-diff --git a/apps/confbridge/include/confbridge.h b/apps/confbridge/include/confbridge.h
-index d3ead35..3d773c5 100644
---- a/apps/confbridge/include/confbridge.h
-+++ b/apps/confbridge/include/confbridge.h
-@@ -57,6 +57,7 @@ enum user_profile_flags {
- 	USER_OPT_DTMF_PASS    =  (1 << 13), /*!< Sets if dtmf should be passed into the conference or not */
- 	USER_OPT_ANNOUNCEUSERCOUNTALL = (1 << 14), /*!< Sets if the number of users should be announced to everyone. */
- 	USER_OPT_JITTERBUFFER =  (1 << 15), /*!< Places a jitterbuffer on the user. */
-+	USER_OPT_STARTDEAF =     (1 << 16), /*!< Set if the caller should be initially set deaf */
- };
- 
- enum bridge_profile_flags {
-@@ -68,6 +69,7 @@ enum bridge_profile_flags {
- 
- enum conf_menu_action_id {
- 	MENU_ACTION_TOGGLE_MUTE = 1,
-+	MENU_ACTION_TOGGLE_DEAF,
- 	MENU_ACTION_PLAYBACK,
- 	MENU_ACTION_PLAYBACK_AND_CONTINUE,
- 	MENU_ACTION_INCREASE_LISTENING,
-@@ -142,6 +144,8 @@ enum conf_sounds {
- 	CONF_SOUND_KICKED,
- 	CONF_SOUND_MUTED,
- 	CONF_SOUND_UNMUTED,
-+	CONF_SOUND_DEAFENED,
-+	CONF_SOUND_UNDEAFENED,
- 	CONF_SOUND_ONLY_ONE,
- 	CONF_SOUND_THERE_ARE,
- 	CONF_SOUND_OTHER_IN_PARTY,
-@@ -168,6 +172,8 @@ struct bridge_profile_sounds {
- 		AST_STRING_FIELD(kicked);
- 		AST_STRING_FIELD(muted);
- 		AST_STRING_FIELD(unmuted);
-+		AST_STRING_FIELD(deafened);
-+		AST_STRING_FIELD(undeafened);
- 		AST_STRING_FIELD(onlyone);
- 		AST_STRING_FIELD(thereare);
- 		AST_STRING_FIELD(otherinparty);
-diff --git a/bridges/bridge_multiplexed.c b/bridges/bridge_multiplexed.c
-index cd30266..190f790 100644
---- a/bridges/bridge_multiplexed.c
-+++ b/bridges/bridge_multiplexed.c
-@@ -386,7 +386,7 @@ static enum ast_bridge_write_result multiplexed_bridge_write(struct ast_bridge *
- 	}
- 
- 	if (other->state == AST_BRIDGE_CHANNEL_STATE_WAIT) {
--		ast_write(other->chan, frame);
-+		ast_bridge_handle_channel_write(bridge, other, frame);
- 	}
- 
- 	return AST_BRIDGE_WRITE_SUCCESS;
-diff --git a/bridges/bridge_simple.c b/bridges/bridge_simple.c
-index 69e4114..1623ce0 100644
---- a/bridges/bridge_simple.c
-+++ b/bridges/bridge_simple.c
-@@ -81,7 +81,7 @@ static enum ast_bridge_write_result simple_bridge_write(struct ast_bridge *bridg
- 
- 	/* Write the frame out if they are in the waiting state... don't worry about freeing it, the bridging core will take care of it */
- 	if (other->state == AST_BRIDGE_CHANNEL_STATE_WAIT) {
--		ast_write(other->chan, frame);
-+		ast_bridge_handle_channel_write(bridge, other, frame);
- 	}
- 
- 	return AST_BRIDGE_WRITE_SUCCESS;
-diff --git a/bridges/bridge_softmix.c b/bridges/bridge_softmix.c
-index 52e5551..5754e41 100644
---- a/bridges/bridge_softmix.c
-+++ b/bridges/bridge_softmix.c
-@@ -435,7 +435,7 @@ static void softmix_pass_dtmf(struct ast_bridge *bridge, struct ast_bridge_chann
- 		if (tmp == bridge_channel) {
- 			continue;
- 		}
--		ast_write(tmp->chan, frame);
-+		ast_bridge_handle_channel_write(bridge, tmp, frame);
- 	}
- }
- 
-@@ -447,7 +447,7 @@ static void softmix_pass_video_top_priority(struct ast_bridge *bridge, struct as
- 			continue;
- 		}
- 		if (ast_bridge_is_video_src(bridge, tmp->chan) == 1) {
--			ast_write(tmp->chan, frame);
-+			ast_bridge_handle_channel_write(bridge, tmp, frame);
- 			break;
- 		}
- 	}
-@@ -463,7 +463,7 @@ static void softmix_pass_video_all(struct ast_bridge *bridge, struct ast_bridge_
- 		if ((tmp->chan == bridge_channel->chan) && !echo) {
- 			continue;
- 		}
--		ast_write(tmp->chan, frame);
-+		ast_bridge_handle_channel_write(bridge, tmp, frame);
- 	}
- }
- 
-@@ -563,7 +563,7 @@ static enum ast_bridge_write_result softmix_bridge_write(struct ast_bridge *brid
- 
- 	/* If a frame is ready to be written out, do so */
- 	if (sc->have_frame) {
--		ast_write(bridge_channel->chan, &sc->write_frame);
-+		ast_bridge_handle_channel_write(bridge, bridge_channel, &sc->write_frame);
- 		sc->have_frame = 0;
- 	}
- 
-@@ -582,7 +582,7 @@ bridge_write_cleanup:
- 	 * the conference to the channel. */
- 	ast_mutex_lock(&sc->lock);
- 	if (sc->have_frame) {
--		ast_write(bridge_channel->chan, &sc->write_frame);
-+		ast_bridge_handle_channel_write(bridge, bridge_channel, &sc->write_frame);
- 		sc->have_frame = 0;
- 	}
- 	ast_mutex_unlock(&sc->lock);
-@@ -598,7 +598,7 @@ static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_chan
- 	ast_mutex_lock(&sc->lock);
- 
- 	if (sc->have_frame) {
--		ast_write(bridge_channel->chan, &sc->write_frame);
-+		ast_bridge_handle_channel_write(bridge, bridge_channel, &sc->write_frame);
- 		sc->have_frame = 0;
- 	}
- 
-@@ -850,16 +850,24 @@ static int softmix_bridge_thread(struct ast_bridge *bridge)
- 
- 			ast_mutex_lock(&sc->lock);
- 
--			/* Make SLINEAR write frame from local buffer */
--			if (sc->write_frame.subclass.format.id != cur_slin_id) {
--				ast_format_set(&sc->write_frame.subclass.format, cur_slin_id, 0);
-+			if (bridge->features.deaf ||
-+			    (bridge_channel->features && bridge_channel->features->deaf)) {
-+				/* For deaf channels post a null frame */
-+				sc->write_frame.frametype = AST_FRAME_NULL;
-+			} else {
-+				/* Make SLINEAR write frame from local buffer */
-+				sc->write_frame.frametype = AST_FRAME_VOICE;
-+				if (sc->write_frame.subclass.format.id != cur_slin_id) {
-+					ast_format_set(&sc->write_frame.subclass.format, cur_slin_id, 0);
-+				}
-+				sc->write_frame.datalen = softmix_datalen;
-+				sc->write_frame.samples = softmix_samples;
-+				memcpy(sc->final_buf, buf, softmix_datalen);
-+
-+				/* process the softmix channel's new write audio */
-+				softmix_process_write_audio(&trans_helper,
-+					ast_channel_rawwriteformat(bridge_channel->chan), sc);
- 			}
--			sc->write_frame.datalen = softmix_datalen;
--			sc->write_frame.samples = softmix_samples;
--			memcpy(sc->final_buf, buf, softmix_datalen);
--
--			/* process the softmix channel's new write audio */
--			softmix_process_write_audio(&trans_helper, ast_channel_rawwriteformat(bridge_channel->chan), sc);
- 
- 			/* The frame is now ready for use... */
- 			sc->have_frame = 1;
-diff --git a/configs/confbridge.conf.sample b/configs/confbridge.conf.sample
-index 7484b28..3b0ce85 100644
---- a/configs/confbridge.conf.sample
-+++ b/configs/confbridge.conf.sample
-@@ -16,6 +16,7 @@ type=user
- ;admin=yes     ; Sets if the user is an admin or not. Off by default.
- ;marked=yes    ; Sets if this is a marked user or not. Off by default.
- ;startmuted=yes; Sets if all users should start out muted. Off by default
-+;startdeaf=yes ; Sets if all users should start out deaf. Off by default.
- ;music_on_hold_when_empty=yes  ; Sets whether MOH should be played when only
-                                ; one person is in the conference or when the
-                                ; the user is waiting on a marked user to enter
-@@ -210,6 +211,8 @@ type=bridge
- ;sound_kicked ; The sound played to a user who has been kicked from the conference.
- ;sound_muted  ; The sound played when the mute option it toggled on.
- ;sound_unmuted  ; The sound played when the mute option it toggled off.
-+;sound_deafened  ; The sound played when the deaf option is toggled on.
-+;sound_undeafened  ; The sound played when the deaf option is toggled off.
- ;sound_only_person ; The sound played when the user is the only person in the conference.
- ;sound_only_one ; The sound played to a user when there is only one other
-                 ; person is in the conference.
-@@ -264,6 +267,8 @@ type=bridge
- ; toggle_mute      ; Toggle turning on and off mute.  Mute will make the user silent
-                    ; to everyone else, but the user will still be able to listen in.
-                    ; continue to collect the dtmf sequence.
-+; toggle_deaf      ; Toggle turning on and off deaf.  Deaf will make the user to hear
-+                   ; only silence, but the user will still be able to talk.
- ; no_op ; This action does nothing (No Operation). Its only real purpose exists for
-         ; being able to reserve a sequence in the config as a menu exit sequence.
- ; decrease_listening_volume ; Decreases the channel's listening volume.
-diff --git a/include/asterisk/bridging_features.h b/include/asterisk/bridging_features.h
-index e377ca6..5ce3d56 100644
---- a/include/asterisk/bridging_features.h
-+++ b/include/asterisk/bridging_features.h
-@@ -127,6 +127,8 @@ struct ast_bridge_features {
- 	unsigned int usable:1;
- 	/*! Bit to indicate whether the channel/bridge is muted or not */
- 	unsigned int mute:1;
-+	/*! Bit to indicate whether the channel/bridge is deaf or not */
-+	unsigned int deaf:1;
- 	/*! Bit to indicate whether DTMF should be passed into the bridge tech or not.  */
- 	unsigned int dtmf_passthrough:1;
- 
-diff --git a/include/asterisk/bridging_technology.h b/include/asterisk/bridging_technology.h
-index 3d2e870..1ecb4c1 100644
---- a/include/asterisk/bridging_technology.h
-+++ b/include/asterisk/bridging_technology.h
-@@ -143,6 +143,21 @@ int ast_bridge_technology_unregister(struct ast_bridge_technology *technology);
-  */
- void ast_bridge_handle_trip(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_channel *chan, int outfd);
- 
-+/*! \brief Used by bridging technologies to hand off a frame to be written to a bridge_channel.
-+ *
-+ * \param bridge The bridge that the channel is part of.
-+ * \param bridge_channel The bridge channel to which the frame is written to.
-+ * \param frame The frame to write.
-+ *
-+ * \retval 0 on success
-+ * \retval -1 on failure
-+ *
-+ * \note This function is essentially a wrapper for ast_write(). The bridging core has some features associated with it
-+ * that requires it to have control over how frames are written into a channel.  For these features to be available, the bridging
-+ * technology must use this wrapper function over ast_write when pushing a frame out a channel.
-+ */
-+int ast_bridge_handle_channel_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame);
-+
- /*! \brief Lets the bridging indicate when a bridge channel has stopped or started talking.
-  *
-  * \note All DSP functionality on the bridge has been pushed down to the lowest possible
-diff --git a/main/bridging.c b/main/bridging.c
-index 465d033..4f67e90 100644
---- a/main/bridging.c
-+++ b/main/bridging.c
-@@ -337,6 +337,28 @@ void ast_bridge_handle_trip(struct ast_bridge *bridge, struct ast_bridge_channel
- 	return;
- }
- 
-+int ast_bridge_handle_channel_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
-+{
-+	if (frame->frametype == AST_FRAME_VOICE &&
-+	    (bridge->features.deaf ||
-+	     (bridge_channel->features && bridge_channel->features->deaf))) {
-+		short buf[frame->samples];
-+		struct ast_frame sframe = {
-+			.frametype = AST_FRAME_VOICE,
-+			.data.ptr = buf,
-+			.samples = frame->samples,
-+			.datalen = sizeof(buf),
-+		};
-+		ast_format_set(&sframe.subclass.format, AST_FORMAT_SLINEAR, 0);
-+		memset(buf, 0, sizeof(buf));
-+
-+		return ast_write(bridge_channel->chan, &sframe);
-+	}
-+
-+	return ast_write(bridge_channel->chan, frame);
-+}
-+
-+
- /*! \brief Generic thread loop, TODO: Rethink this/improve it */
- static int generic_thread_loop(struct ast_bridge *bridge)
- {
--- 
-1.7.12
-
diff --git a/testing/asterisk/ASTERISK-20527.patch b/testing/asterisk/ASTERISK-20527.patch
deleted file mode 100644
index 42a81d8ca5..0000000000
--- a/testing/asterisk/ASTERISK-20527.patch
+++ /dev/null
@@ -1,26 +0,0 @@
-diff --git a/channels/chan_sip.c b/channels/chan_sip.c
-index 7569bba..24a8cec 100644
---- a/channels/chan_sip.c
-+++ b/channels/chan_sip.c
-@@ -30189,7 +30189,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
- 	olddirectmediaacl = ast_free_acl_list(olddirectmediaacl);
- 	if (!ast_strlen_zero(peer->callback)) { /* build string from peer info */
- 		char *reg_string;
--		if (ast_asprintf(&reg_string, "%s?%s:%s@%s/%s", peer->name, peer->username, !ast_strlen_zero(peer->remotesecret) ? peer->remotesecret : peer->secret, peer->tohost, peer->callback) >= 0) {
-+		if (ast_asprintf(&reg_string, "%s?%s:%s:%s@%s/%s", peer->name, S_OR(peer->fromuser, peer->username), S_OR(peer->remotesecret, peer->secret), peer->username, peer->tohost, peer->callback) >= 0) {
- 			sip_register(reg_string, 0); /* XXX TODO: count in registry_count */
- 			ast_free(reg_string);
- 		}
-diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
-index 81ca998..812be7b 100644
---- a/configs/sip.conf.sample
-+++ b/configs/sip.conf.sample
-@@ -711,7 +711,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
- ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
- ; this is equivalent to having the following line in the general section:
- ;
--;        register => username:secret@host/callbackextension
-+;        register => fromuser:secret:username@host/callbackextension
- ;
- ; and more readable because you don't have to write the parameters in two places
- ; (note that the "port" is ignored - this is a bug that should be fixed).
diff --git a/testing/asterisk/asterisk.confd b/testing/asterisk/asterisk.confd
deleted file mode 100644
index fe9f138ab7..0000000000
--- a/testing/asterisk/asterisk.confd
+++ /dev/null
@@ -1,91 +0,0 @@
-#
-# Additional options for asterisk
-#
-# see "asterisk -h" for a list of options
-#
-ASTERISK_OPTS=""
-
-#
-# User and group to run asterisk as
-#
-# Value: double-colon separated list of user and group, or empty to run as root:
-#
-#
-# "asterisk:asterisk" to run as user "asterisk" and group "asterisk"
-#          "asterisk" to run as user "asterisk" and all groups that user "asterisk" is a member of
-#         ":asterisk" to run as user "root" and group "asterisk"
-#                  "" to run as user "root" and group "root"
-#
-ASTERISK_USER="asterisk"
-
-#
-# Nicelevel
-#
-# Set the priority of the asterisk process
-#
-# Value: (highest) -20..19 (lowest)
-#
-#ASTERISK_NICE="19"
-
-#
-# Wrapper script
-#
-# Value: yes or no/empty
-#
-ASTERISK_WRAPPER="no"
-
-############# Wrapper script settings #############
-
-#
-# Send crash notifications emails to this address
-# (needs a working mail service and /usr/sbin/sendmail to do so (e.g. ssmtp))
-#
-# Value: Email address or empty to disable
-#
-#ASTERISK_NOTIFY_EMAIL="root"
-
-#
-# Send asterisk's output to this terminal
-#
-# Value: Full path to device node or a number
-#
-#ASTERISK_TTY="/dev/tty9"
-
-#
-# Start an asterisk console on the terminal specified by ASTERISK_TTY
-#
-# Warning! Use only for debugging, this is a potential security issue!
-#
-# Value: yes or no/empty
-#
-ASTERISK_CONSOLE="no"
-
-#
-# Maximum size of core files. 
-#
-# Value: Size in bytes, unlimited for no limit or empty to disable.
-# 
-#ASTERISK_CORE_SIZE="unlimited"
-
-#
-# ASTERISK_CORE_DIR
-#
-# Value: Directory (will be created if non-existant), default is /tmp
-#
-ASTERISK_CORE_DIR="/var/lib/asterisk/coredump"
-
-#
-# Max number of filedescriptors
-#
-# Value: Number of descriptors
-#
-#ASTERISK_MAX_FD="1024"
-
-#
-# Kill these tasks after asterisk crashed (ASTERISK_WRAPPER=yes only!)
-#
-# Warning! This will kill _ALL_ tasks with the specified names!
-#
-# Value: Space separated list of names in double quotes (e.g. "mpg123 mad")
-#
-#ASTERISK_CLEANUP_ON_CRASH="mpg123 asterisk-mpg123 mad"
diff --git a/testing/asterisk/asterisk.initd b/testing/asterisk/asterisk.initd
deleted file mode 100644
index 9b6c93ecc1..0000000000
--- a/testing/asterisk/asterisk.initd
+++ /dev/null
@@ -1,251 +0,0 @@
-#!/sbin/runscript
-
-extra_started_commands="forcestop reload"
-
-depend() {
-	need net
-	after firewall
-	use nscd dns zaptel mysql postgresql slapd capi
-}
-
-is_running() {
-	if [ -z "$(pidof asterisk)" ]; then
-		return 1
-	else
-		PID="$(cat /var/run/asterisk/asterisk.pid 2>/dev/null)"
-		for x in $(pidof asterisk); do
-			if [ "${x}" = "${PID}" ]; then
-				return 0
-			fi	
-		done
-	fi
-
-	return 1
-}
-
-asterisk_run_loop() {
-	local OPTS ARGS MSG NICE=""
-	local result=0 signal=0
-
-	# default options
-	OPTS="-f"  # don't fork / detach breaks wrapper script...
-
-	# filter (redundant) arguments
-	ARGS="$(echo "${@}" | sed -e "s:-c\|-f::g")"
-
-	# mangle yes/no options
-	ASTERISK_CONSOLE="$(echo ${ASTERISK_CONSOLE} | tr '[:lower:]' '[:upper:]')"
-
-	if [ -n "${ASTERISK_CORE_SIZE}" ] &&
-	   [ "${ASTERISK_CORE_SIZE}" != "0" ]; then
-		ulimit -c ${ASTERISK_CORE_SIZE}
-
-		if [ -n "${ASTERISK_CORE_DIR}" ] && \
-		   [ ! -d "${ASTERISK_CORE_DIR}" ]
-		then
-			mkdir -m750 -p "${ASTERISK_CORE_DIR}"
-
-			if [ -n "${ASTERISK_USER}" ]; then
-				chown -R "${ASTERISK_USER}" "${ASTERISK_CORE_DIR}"
-			fi
-		fi
-		ASTERISK_CORE_DIR="${ASTERISK_CORE_DIR:-/tmp}"
-
-		cd "${ASTERISK_CORE_DIR}"
-		echo "   Core dump size            : ${ASTERISK_CORE_SIZE}"
-		echo "   Core dump location        : ${ASTERISK_CORE_DIR}"
-	fi
-
-	if [ -n "${ASTERISK_MAX_FD}" ]; then
-		ulimit -n ${ASTERISK_MAX_FD}
-		echo "   Max open filedescriptors  : ${ASTERISK_MAX_FD}"
-	fi
-
-	if [ -n "${ASTERISK_NICE}" ]; then
-		echo "   Nice level                : ${ASTERISK_NICE}"
-		NICE="nice -n ${ASTERISK_NICE} --"
-	fi
-
-	if [ -n "${ASTERISK_NOTIFY_EMAIL}" ]; then
-		if [ -x /usr/sbin/sendmail ]; then
-			echo "   Email notifications go to : ${ASTERISK_NOTIFY_EMAIL}"
-		else
-			echo "   Notifications disabled, /usr/sbin/sendmail doesn't exist or is not executable!"
-			unset ASTERISK_NOTIFY_EMAIL
-		fi
-	fi
-
-	if [ -n "${ASTERISK_TTY}" ]; then
-		for x in ${ASTERISK_TTY} \
-			 /dev/tty${ASTERISK_TTY} \
-			 /dev/vc/${ASTERISK_TTY}
-		do
-			if [ -c "${x}" ]; then
-				TTY="${x}"
-			fi
-		done
-		[ -n "${TTY}" ] && \
-			echo "   Messages are sent to      : ${TTY}"
-	fi
-
-	if [ "${ASTERISK_CONSOLE}" = "YES" ] && [ -n "${TTY}" ]; then
-		echo "   Starting Asterisk console : ${ASTERISK_CONSOLE}"
-		OPTS="${OPTS} -c"
-	fi
-
-	OPTS="${OPTS} ${ARGS}"
-
-	while :; do
-
-		if [ -n "${TTY}" ]; then
-			/usr/bin/stty -F ${TTY} sane
-			${NICE} /usr/sbin/asterisk ${OPTS} >${TTY} 2>&1 <${TTY}
-			result=$?
-		else
-			${NICE} /usr/sbin/asterisk ${OPTS} &>/dev/null
-			result=$?
-		fi		
-
-		if [ $result -eq 0 ]; then
-			echo "Asterisk terminated normally"
-			break
-		else
-			if [ $result -gt 128 ]; then
-				signal=$((result -  128))
-				MSG="Asterisk terminated with Signal: $signal"
-
-				CORE_TARGET="core-$(date "+%Y%m%d-%h%M%s")"
-
-				local CORE_DUMPED=0
-				if [ -f "${ASTERISK_CORE_DIR}/core" ]; then
-					mv "${ASTERISK_CORE_DIR}/core" \
-					   "${ASTERISK_CORE_DIR}/${CORE_TARGET}"
-					CORE_DUMPED=1
-
-				elif [ -f "${ASTERISK_CORE_DIR}/core.${PID}" ]; then
-					mv "${ASTERISK_CORE_DIR}/core.${PID}" \
-					   "${ASTERISK_CORE_DIR}/${CORE_TARGET}"
-					CORE_DUMPED=1
-
-				fi
-
-				[ $CORE_DUMPED -eq 1 ] && \
-					MSG="${MSG}\n\rCore dumped: ${ASTERISK_CORE_DIR}/${CORE_TARGET}"
-			else
-				MSG="Asterisk terminated with return code: $result"
-			fi
-
-			# kill left-over tasks
-			for X in ${ASTERISK_CLEANUP_ON_CRASH}; do
-				kill -9 $(pidof ${X});
-			done
-		fi
-
-		[ -n "${TTY}" ] \
-			&& echo "${MSG}" >${TTY} \
-			|| echo "${MSG}"
-
-
-		if [ -n "${ASTERISK_NOTIFY_EMAIL}" ] && \
-		   [ -x /usr/sbin/sendmail ]; then
-			echo -e -n "Subject: Asterisk crashed\n\r${MSG}\n\r" |\
-				 /usr/sbin/sendmail "${ASTERISK_NOTIFY_EMAIL}"
-		fi
-		sleep 5
-		echo "Restarting Asterisk..."
-	done
-	return 0
-}
-
-start() {
-	local OPTS USER GROUP PID
-	local tmp x
-
-	if [ -n "${ASTERISK_NICE}" ]; then
-		if [ ${ASTERISK_NICE} -ge -20 ] && \
-		   [ ${ASTERISK_NICE} -le  19 ]; then 
-			OPTS="--nicelevel ${ASTERISK_NICE}"
-		else
-			eerror "Nice value must be between -20 and 19"
-			return 1
-		fi
-	fi
-
-	if [ -n "${ASTERISK_USER}" ]; then
-		USER=$(echo $ASTERISK_USER | sed 's/:.*//')
-		GROUP=$(echo $ASTERISK_USER | awk -F: '/.*:.*/ { print $2 }')
-		if [ -n "${USER}" ]; then
-			ASTERISK_OPTS="${ASTERISK_OPTS} -U ${USER}"
-		fi
-		if [ -n "${GROUP}" ]; then
-			ASTERISK_OPTS="${ASTERISK_OPTS} -G ${GROUP}"
-			GROUP=":${GROUP}"	# make it look nice...
-		fi
-		ebegin "Starting asterisk PBX (as ${USER}${GROUP})"
-	else
-		ebegin "Starting asterisk PBX (as root)"
-	fi
-
-	if [ "$(echo ${ASTERISK_WRAPPER} | tr '[:upper:]' '[:lower:]')" != "yes" ]; then
-		start-stop-daemon --start --exec /usr/sbin/asterisk \
-			${OPTS} -- ${ASTERISK_OPTS}
-		result=$?
-	else
-		asterisk_run_loop ${ASTERISK_OPTS} 2>/dev/null &
-		result=$?
-	fi
-
-	if [ $result -eq 0 ]; then
-		# 2 seconds should be enough for asterisk to start
-		sleep 2 
-		is_running
-		result=$?
-	fi
-
-	eend $result
-}
-
-forcestop() {
-	ebegin "Stopping asterisk PBX"
-	start-stop-daemon --stop --pidfile /var/run/asterisk/asterisk.pid
-	eend $?
-}
-
-stop() {
-	if ! is_running; then
-		eerror "Asterisk is not running!"
-		return 0
-	fi
-
-	ebegin "Stopping asterisk PBX now"
-	/usr/sbin/asterisk -r -x "core stop now" &>/dev/null
-	# Now we have to wait until asterisk has _really_ stopped.
-	sleep 1
-	if is_running; then
-		einfon "Waiting for asterisk to shutdown ."
-		local cnt=0
-		while is_running; do
-			cnt=`expr $cnt + 1`
-			if [ $cnt -gt 60 ] ; then
-				# Waited 120 seconds now. Fail.
-				echo
-				eend 1 "Failed."
-				return
-			fi
-			sleep 2
-			echo -n "."
-		done
-		echo
-	fi
-	eend 0
-}
-
-reload() {
-	if is_running; then
-		ebegin "Reloading asterisk configuration"
-		/usr/sbin/asterisk -r -x "core reload" &>/dev/null
-		eend $?
-	else
-		eerror "Asterisk is not running!"
-	fi
-}
diff --git a/testing/asterisk/asterisk.logrotate b/testing/asterisk/asterisk.logrotate
deleted file mode 100644
index 30836c5c11..0000000000
--- a/testing/asterisk/asterisk.logrotate
+++ /dev/null
@@ -1,17 +0,0 @@
-/var/log/asterisk/messages /var/log/asterisk/*log {
-	missingok
-	rotate 5
-	weekly
-	create 0640 asterisk asterisk
-	postrotate
-		/usr/sbin/asterisk -rx 'logger reload' > /dev/null 2> /dev/null
-	endscript
-}
-
-/var/log/asterisk/cdr-csv/*csv {
-	missingok
-	rotate 5
-	monthly
-	create 0640 asterisk asterisk
-}
-
diff --git a/testing/asterisk/asterisk.pre-install b/testing/asterisk/asterisk.pre-install
deleted file mode 100644
index 6c2984ae4e..0000000000
--- a/testing/asterisk/asterisk.pre-install
+++ /dev/null
@@ -1,6 +0,0 @@
-#!/bin/sh
-
-adduser -S -h /var/lib/asterisk -s /bin/false -D asterisk 2>/dev/null
-addgroup -S dialout 2>/dev/null
-addgroup asterisk dialout 2>/dev/null
-exit 0
diff --git a/testing/asterisk/asterisk.pre-upgrade b/testing/asterisk/asterisk.pre-upgrade
deleted file mode 100644
index 6c2984ae4e..0000000000
--- a/testing/asterisk/asterisk.pre-upgrade
+++ /dev/null
@@ -1,6 +0,0 @@
-#!/bin/sh
-
-adduser -S -h /var/lib/asterisk -s /bin/false -D asterisk 2>/dev/null
-addgroup -S dialout 2>/dev/null
-addgroup asterisk dialout 2>/dev/null
-exit 0