main/asterisk: upgrade to 14.0.2

5 files changed, 6 insertions, 522 deletions

diff --git a/main/asterisk/APKBUILD b/main/asterisk/APKBUILD
index 836d368935..1c58eb9fda 100644
--- a/main/asterisk/APKBUILD
+++ b/main/asterisk/APKBUILD
@@ -2,7 +2,7 @@
 # Contributor: Timo Teras <timo.teras@iki.fi>
 # Maintainer: Timo Teras <timo.teras@iki.fi>
 pkgname=asterisk
-pkgver=13.11.1
+pkgver=14.0.2
 pkgrel=0
 pkgdesc="Asterisk: A Module Open Source PBX System"
 pkgusers="asterisk"
@@ -13,7 +13,7 @@ license="GPL2"
 depends=
 makedepends="autoconf automake libtool findutils tar wget bsd-compat-headers
 	ncurses-dev popt-dev newt-dev zlib-dev libedit-dev tiff-dev curl-dev
-	openssl-dev lua-dev libcap-dev jansson-dev util-linux-dev
+	libressl-dev lua-dev libcap-dev jansson-dev util-linux-dev
 	sqlite-dev postgresql-dev unixodbc-dev freetds-dev mariadb-dev
 	alsa-lib-dev pjproject-dev dahdi-tools-dev libpri-dev libsrtp-dev
 	spandsp-dev libresample speex-dev speexdsp-dev libogg-dev
@@ -29,11 +29,7 @@ _download="http://downloads.asterisk.org/pub/telephony/asterisk/releases"
 source="$_download/asterisk-$pkgver.tar.gz
 	http://dev.alpinelinux.org/~tteras/asterisk-addon-mp3-r201.patch.gz
 
-	musl-libcap.patch
-	musl-glob-compat.patch
 	musl-mutex-init.patch
-	ASTERISK-18995.patch
-	ASTERISK-20527.patch
 	asterisk.initd
 	asterisk.confd
 	asterisk.logrotate"
@@ -60,15 +56,12 @@ prepare() {
 
 	sed -i -e 's:lua5.1/::' pbx/pbx_lua.c
 	sed -i -e 's/int foo = res_ninit(NULL);/res_ninit_is_not_really_here();/g' configure.ac
-	update_config_sub || return 1
 
 	./bootstrap.sh
 }
 
 build() {
 	cd "$_builddir"
-	# cannot build with imap on aarch64
-	[ "$CARCH" = "aarch64" ] || _imap="--with-imap"
 	SHA1SUM="$PWD"/build_tools/sha1sum-sh ./configure \
 		--build=$CBUILD \
 		--host=$CHOST \
@@ -99,7 +92,7 @@ build() {
 		--with-libcurl \
 		--with-libedit \
 		--with-srtp \
-		$_imap \
+		--with-imap=system \
 		|| return 1
 
 	# get default modules to build
@@ -240,33 +233,21 @@ sound_en() {
 	chown -R asterisk:asterisk "$subpkgdir"/var/*/asterisk
 }
 
-md5sums="7c27d278533c5936348a412e9df5982d  asterisk-13.11.1.tar.gz
+md5sums="51eae0af8eea2ee42f48e0132cac617f  asterisk-14.0.2.tar.gz
 126dd4fba66f4cf9aa94dfd7034e0ec4  asterisk-addon-mp3-r201.patch.gz
-7b699961171a93d7788420f518c5931e  musl-libcap.patch
-9b3e6cb75464a6ef4c40d191bdfdf7ea  musl-glob-compat.patch
 1ca5e8326dc03c963a7ce5455d0f21ba  musl-mutex-init.patch
-120deda3ccd21c905ac51fda1f696931  ASTERISK-18995.patch
-a91bf5c4fea24b3e04f02e41751aaa88  ASTERISK-20527.patch
 4bdc82ba3d6bdfdedc71e5da2fde5ec2  asterisk.initd
 ed31d7ba37bcf8b0346dcf8593c395f0  asterisk.confd
 3e65172275684373e1a25c8a11224411  asterisk.logrotate"
-sha256sums="d1dc61b78b5e2c51a3c7b65300f2cb29470c88552fc6c30aa9ea426fef385748  asterisk-13.11.1.tar.gz
+sha256sums="22482606ed5efed5f67813369e1a304bd2a5ec44e7b08727710a9baaeb8fe7de  asterisk-14.0.2.tar.gz
 d32a5a695cee1699011d0e9ad02eb43af612def06f92017627194099edf98e3f  asterisk-addon-mp3-r201.patch.gz
-f23aa2c03f9be1eb3c4c0ceac79ddcce7381aae46d6e6853ad20b1de5ac125d9  musl-libcap.patch
-c6a5c32646b767f09ebffccc49cac6a4ff9181498499e4453300775e75284b52  musl-glob-compat.patch
 a5205ed44b57a72934baf8cde543ddb179f277494181818905110bfdcccfe7d9  musl-mutex-init.patch
-76ba70d0df5f4a788294854cf3a40c3adea711ea70fdb7750ee208afc0b13b12  ASTERISK-18995.patch
-790af610fe147d8add270e7e7e0d998e265ed78374f597844beaedd270d44b07  ASTERISK-20527.patch
 10454553733e6cc52c6e9db508768a638655d99c095c5b39fb043858c088f21f  asterisk.initd
 d221148583b57f9c37d7160f2493f0d204ad11f7abb17e3a3534e108ad5452d7  asterisk.confd
 77b253b6db71460acf9a51e87ad4c8582027a46db01a4c50fb048bada58c19d1  asterisk.logrotate"
-sha512sums="23bb5b8997426ae952017bdc6c7630ff1ca4ca401879557f14b26fd5fb83014577b487e2e0a52cc46d15b691c4e5a732aaf7baba47d566bfb676e9c5f7f472c0  asterisk-13.11.1.tar.gz
+sha512sums="aa51d7ddaf4945b67aa28a8456834f33e75e0274310b4237443b576bf40eee15d6a3e4e5baad7f28ae299d34b749554518c1c17e364b90ece9aead732392627f  asterisk-14.0.2.tar.gz
 aacef3f4796fb1abd33266998b53909cb4b36e7cc5ad2f7bac68bdc43e9a9072d9a4e2e7e681bddfa31f3d04575eb248afe6ea95da780c67e4829c1e22adfe1b  asterisk-addon-mp3-r201.patch.gz
-31bdf1323155a9fbedf8c05183167903ff83bef4d408848368d3fab78e6c6265228ceece54375d68df6a99b9c1879431033d0a8cec875244c61ccc1f8e37fc5d  musl-libcap.patch
-edf65eba44e8f1f0e78671aba312bcff2090134cf94e89ebb4b6beef7b2d57b24381ae9511cbf9fffe6b555e695a3dbe2cedcd04f1d237df822fa742bd5092ac  musl-glob-compat.patch
 f72c2e04de80d3ed9ce841308101383a1655e6da7a3c888ad31fffe63d1280993e08aefcf8e638316d439c68b38ee05362c87503fca1f36343976a01af9d6eb1  musl-mutex-init.patch
-e6becba58f73ffeac7831092135e9cfd5ca9029b9471061b6f20d6e98c9c4d4dcaea7d69a53c079fb5baf03963809740e5feaf8b595072229a04912bff9ff41a  ASTERISK-18995.patch
-0c9f643b2b5548753bf5228ea17a2a55753ea4fdcd32c3ef105f4f27a762a0f93a4a02ba0b88d969fad8729048d357959d3f7d0fa166c2935170f0753b08d8a4  ASTERISK-20527.patch
 cd5bd1c1d7db0a44b14eb10e6d098af0c6474c8fe1a57395090d6795ac00e9243d004b7d24eba2cfd5bd6d6407c271913e794551a8dfcf3cf93e89fc91349e12  asterisk.initd
 ab6b6f08ff43268cbb1abb7ed7d678949991ba495682a644bbaeb017d6adbff0a43297905fd73ae8db1786a28d5b5904f1bc253209a0e388c8a27f26c6ce14ed  asterisk.confd
 7591d2faf539d05d9ee4e431c78a5e20686721fd79221ad94dffeeaff9282220b09cb9aec214bd7a8d12affaec0276c9c91e6e21af8b6712c0a9502b60b02f2b  asterisk.logrotate"
diff --git a/main/asterisk/ASTERISK-18995.patch b/main/asterisk/ASTERISK-18995.patch
deleted file mode 100644
index 03f796dd79..0000000000
--- a/main/asterisk/ASTERISK-18995.patch
+++ /dev/null
@@ -1,386 +0,0 @@
-From 56bdf048d2c873d0ddfad3672a07e7a08f0b706e Mon Sep 17 00:00:00 2001
-From: =?UTF-8?q?Timo=20Ter=C3=A4s?= <timo.teras@iki.fi>
-Date: Fri, 3 Jun 2016 09:20:39 +0300
-Subject: [PATCH] Add support for OGG/Speex file format
-
-ASTERISK-18995 #close
-
-Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a
----
- CHANGES                    |   7 +
- formats/format_ogg_speex.c | 345 +++++++++++++++++++++++++++++++++++++++++++++
- 2 files changed, 352 insertions(+)
- create mode 100644 formats/format_ogg_speex.c
-
-diff --git a/CHANGES b/CHANGES
-index 43dc18f..175138a 100644
---- a/CHANGES
-+++ b/CHANGES
-@@ -249,6 +249,13 @@ Functions
-  * The func_odbc global option "single_db_connection" default value has been
-    changed to 'no'.
- 
-+
-+Formats
-+------------------
-+ * New module format_ogg_speex added which supports Speex codec inside
-+   Ogg containers (filename extension .spx).
-+
-+
- CHANNEL
- ------------------
-  * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
-diff --git a/formats/format_ogg_speex.c b/formats/format_ogg_speex.c
-new file mode 100644
-index 0000000..6152e9c
---- /dev/null
-+++ b/formats/format_ogg_speex.c
-@@ -0,0 +1,345 @@
-+/*
-+ * Asterisk -- An open source telephony toolkit.
-+ *
-+ * Copyright (C) 2011-2016, 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_REGISTER_FILE()
-+
-+#include "asterisk/mod_format.h"
-+#include "asterisk/module.h"
-+#include "asterisk/format_cache.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 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.
-+ * \return The new filestream.
-+ */
-+static int ogg_speex_open(struct ast_filestream *fs)
-+{
-+	char *buffer;
-+	size_t bytes;
-+	struct speex_desc *s = (struct speex_desc *)fs->_private;
-+	SpeexHeader *hdr = NULL;
-+	int i, result, expected_rate;
-+
-+	expected_rate = ast_format_get_sample_rate(fs->fmt->format);
-+	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;
-+		}
-+	}
-+	speex_header_free(hdr);
-+
-+	return 0;
-+error:
-+	if (hdr) {
-+		speex_header_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;
-+	}
-+
-+	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_samples_count(&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 struct ast_format_def speex_f = {
-+	.name = "ogg_speex",
-+	.exts = "spx",
-+	.open = ogg_speex_open,
-+	.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 struct ast_format_def speex16_f = {
-+	.name = "ogg_speex16",
-+	.exts = "spx16",
-+	.open = ogg_speex_open,
-+	.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 struct ast_format_def speex32_f = {
-+	.name = "ogg_speex32",
-+	.exts = "spx32",
-+	.open = ogg_speex_open,
-+	.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)
-+{
-+	speex_f.format = ast_format_speex;
-+	speex16_f.format = ast_format_speex16;
-+	speex32_f.format = ast_format_speex32;
-+
-+	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
-+);
--- 
-2.10.0
-
diff --git a/main/asterisk/ASTERISK-20527.patch b/main/asterisk/ASTERISK-20527.patch
deleted file mode 100644
index e2095bed41..0000000000
--- a/main/asterisk/ASTERISK-20527.patch
+++ /dev/null
@@ -1,43 +0,0 @@
-From 538c6415c6d255876a808d69acd9f5ebd603b286 Mon Sep 17 00:00:00 2001
-From: =?UTF-8?q?Timo=20Ter=C3=A4s?= <timo.teras@iki.fi>
-Date: Fri, 3 Jun 2016 09:33:08 +0300
-Subject: [PATCH 1/1] chan_sip: Support auth username for callbackextension
- feature
-
-ASTERISK-20527 #close
-
-Change-Id: I659cf7f00836a09d09d146ad226a40477d731239
----
- channels/chan_sip.c             | 2 +-
- configs/samples/sip.conf.sample | 2 +-
- 2 files changed, 2 insertions(+), 2 deletions(-)
-
-diff --git a/channels/chan_sip.c b/channels/chan_sip.c
-index 19f8aa3..03cba92 100644
---- a/channels/chan_sip.c
-+++ b/channels/chan_sip.c
-@@ -31811,7 +31811,7 @@ 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/samples/sip.conf.sample b/configs/samples/sip.conf.sample
-index 8f28e26..a7b74df 100644
---- a/configs/samples/sip.conf.sample
-+++ b/configs/samples/sip.conf.sample
-@@ -786,7 +786,7 @@ 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).
--- 
-2.10.0
-
diff --git a/main/asterisk/musl-glob-compat.patch b/main/asterisk/musl-glob-compat.patch
deleted file mode 100644
index 429babb573..0000000000
--- a/main/asterisk/musl-glob-compat.patch
+++ /dev/null
@@ -1,31 +0,0 @@
---- asterisk-11.7.0.orig/res/ael/ael.flex
-+++ asterisk-11.7.0/res/ael/ael.flex
-@@ -79,6 +79,12 @@
- #if !defined(GLOB_ABORTED)
- #define GLOB_ABORTED GLOB_ABEND
- #endif
-+#if !defined(GLOB_BRACE)
-+#define GLOB_BRACE 0
-+#endif
-+#if !defined(GLOB_NOMAGIC)
-+#define GLOB_NOMAGIC 0
-+#endif
- 
- #include "asterisk/logger.h"
- #include "asterisk/utils.h"
-Only in asterisk-11.7.0: res/ael/ael.tab.o
---- asterisk-11.7.0.orig/res/ael/ael_lex.c
-+++ asterisk-11.7.0/res/ael/ael_lex.c
-@@ -838,6 +838,12 @@
- #if !defined(GLOB_ABORTED)
- #define GLOB_ABORTED GLOB_ABEND
- #endif
-+#if !defined(GLOB_BRACE)
-+#define GLOB_BRACE 0
-+#endif
-+#if !defined(GLOB_NOMAGIC)
-+#define GLOB_NOMAGIC 0
-+#endif
- 
- #include "asterisk/logger.h"
- #include "asterisk/utils.h"
diff --git a/main/asterisk/musl-libcap.patch b/main/asterisk/musl-libcap.patch
deleted file mode 100644
index a414e05fd3..0000000000
--- a/main/asterisk/musl-libcap.patch
+++ /dev/null
@@ -1,37 +0,0 @@
---- asterisk-13.3.2.orig/configure.ac
-+++ asterisk-13.3.2/configure.ac
-@@ -181,6 +181,9 @@
-      linux-gnueabi* |  linux-gnuspe)
-      OSARCH=linux-gnu
-      ;;
-+     linux-musl*)
-+     OSARCH=linux-musl
-+     ;;
-      kfreebsd*-gnu)
-      OSARCH=kfreebsd-gnu
-      ;;
-@@ -1328,9 +1331,11 @@
-   AST_EXT_LIB_CHECK([BFD], [bfd], [bfd_check_format], [bfd.h], [-ldl -liberty -lz])
- fi
- 
--if test "x${OSARCH}" = "xlinux-gnu" ; then
-+case "${OSARCH}" in
-+linux*)
-   AST_EXT_LIB_CHECK([CAP], [cap], [cap_from_text], [sys/capability.h])
--fi
-+  ;;
-+esac
- 
- AST_C_DEFINE_CHECK([DAHDI], [DAHDI_RESET_COUNTERS], [dahdi/user.h], [230])
- AST_C_DEFINE_CHECK([DAHDI], [DAHDI_DEFAULT_MTU_MRU], [dahdi/user.h], [220])
---- asterisk-13.3.2.orig/main/Makefile
-+++ asterisk-13.3.2/main/Makefile
-@@ -42,7 +42,7 @@
- AST_LIBS+=$(CRYPT_LIB)
- AST_LIBS+=$(AST_CLANG_BLOCKS_LIBS)
- 
--ifneq ($(findstring $(OSARCH), linux-gnu uclinux linux-uclibc kfreebsd-gnu),)
-+ifneq ($(findstring $(OSARCH), linux-gnu uclinux linux-uclibc linux-musl kfreebsd-gnu),)
-   ifneq ($(findstring LOADABLE_MODULES,$(MENUSELECT_CFLAGS)),)
-   AST_LIBS+=-ldl
-   endif