ChangeLog: Updated for 14.0.1

Updated codecs/codecs.xml to add codec_opus to the external
download list.

ASTERISK-26409

Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4
(cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c)

.version

@@ -1 +1 @@
-14.0.0
+14.0.1

CHANGES

@@ -8,6 +8,21 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.0.1 ----------
+------------------------------------------------------------------------------
+
+Build System
+------------------
+ * The res_digium_phone, codec_g729a, codec_silk, codec_siren7 and
+   codec_siren14 binary modules hosted at downloads.digium.com can now be
+   automatically downloaded and installed during the Asterisk install
+   process.  If selected in menuselect, when 'make install' is run, the
+   script will check the downloads site for a new version and download
+   and install it if needed.  The '--with-externals-cache' option to
+   ./configure can be used to specify a location to cache the latest
+   tarballs so they don't have to be re-downloaded for every install.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
 ------------------------------------------------------------------------------

ChangeLog

@@ -1,3 +1,165 @@
+2016-09-27 18:43 +0000  Asterisk Development Team <asteriskteam@digium.com>
+
+	* asterisk 14.0.1 Released.
+
+2016-09-27 13:43 +0000 [80b3a952e0]  Kevin Harwell <kharwell@digium.com>
+
+	* Release summaries: Remove previous versions
+
+2016-09-27 13:43 +0000 [404c4a6820]  Kevin Harwell <kharwell@digium.com>
+
+	* .version: Update for 14.0.1
+
+2016-09-27 13:43 +0000 [8e0f992505]  Kevin Harwell <kharwell@digium.com>
+
+	* .lastclean: Update for 14.0.1
+
+2016-09-27 13:43 +0000 [801966afa9]  Kevin Harwell <kharwell@digium.com>
+
+	* realtime: Add database scripts for 14.0.1
+
+2016-09-22 09:49 +0000 [d3d8257106]  gtjoseph <gjoseph@digium.com>
+
+	* codec_opus: Add download ability to menuselect
+
+	  Updated codecs/codecs.xml to add codec_opus to the external
+	  download list.
+
+	  ASTERISK-26409
+
+	  Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4
+	  (cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c)
+
+2016-07-23 14:50 +0000 [1e98c8d5c8]  gtjoseph <gjoseph@digium.com>
+
+	* codec_opus: Replace res_format_attr_opus with the one from codec_opus
+
+	  Preparation
+
+	  ASTERISK-26409
+
+	  Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3
+
+2016-07-23 15:56 +0000 [a091e15985]  gtjoseph <gjoseph@digium.com>
+
+	* format_ogg_opus: New format
+
+	  Add Ogg/Opus playback support.
+
+	  This uses libopusfile in order to be able to read .opus files and play
+	  them back.
+
+	  Writing/recording support is not present at this time.
+
+	  ASTERISK-26409
+
+	  Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955
+
+2016-09-24 19:05 +0000 [a13b54df83]  gtjoseph <gjoseph@digium.com>
+
+	* build_tools:  Add ability to download variants to download_externals
+
+	  Some external packages have multiple variants that apply to different
+	  builds of asterisk.  The DPMA for instance has a "bundled" variant that
+	  needs to be downloaded if asterisk was configured with
+	  --with-pjproject-bundled.
+
+	  There are 2 ways to specify variants:
+
+	  If you need the user to make the decision about which variant to
+	  download, simply create multiple menuselect "member" entries like so...
+
+	  <member name="res_digium_phone" displayname="..snipped..">
+	    <support_level>external</support_level>
+	    <depend>xmlstarlet</depend>
+	    <depend>bash</depend>
+	    <defaultenabled>no</defaultenabled>
+	  </member>
+
+	  <member name="res_digium_phone-bundled" displayname="..snipped..">
+	    <support_level>external</support_level>
+	    <depend>xmlstarlet</depend>
+	    <depend>bash</depend>
+	    <defaultenabled>no</defaultenabled>
+	  </member>
+
+	  Note that the second entry has "-<variant>" appended to the name.
+	  You can then use the existing menuselect facilities to restrict which
+	  members to enable or disable.  Youy probably don't want the user to
+	  enable multiple at the same time.
+
+	  If you want to hide the details of the variants, the better way to
+	  do it is to create 1 member with "variant" elements.
+
+	  <member name="res_digium_phone" displayname="..snipped..">
+	    <support_level>external</support_level>
+	    <depend>xmlstarlet</depend>
+	    <depend>bash</depend>
+	    <defaultenabled>no</defaultenabled>
+	    <member_data>
+	      <downloader>
+	        <variants>
+	          <variant tag="bundled"
+	            condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/>
+	        </variants>
+	      </downloader>
+	    </member_data>
+	  </member>
+
+	  The condition must be a bash expression suitable for use with an "if"
+	  statement.  Any environment variable can be used plus those available
+	  in makeopts.
+
+	  In this case, if asterisk was configured with --with-pjproject-bundled
+	  the bundled variant will be automatically downloaded.  Otherwise the
+	  normal version will be downloaded.
+
+	  Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e
+
+2016-08-01 20:55 +0000 [13cae45bb5]  gtjoseph <gjoseph@digium.com>
+
+	* build: Add download capability for external packages
+
+	  The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
+	  http://downloads.digium.com/pub/telephony/ are now listed in the
+	  "External" sections of the "Resource Modules" and "Codec Translators"
+	  pages in menuselect.  Any that are selected will automatically be
+	  downloaded and installed when "make install" is run.  Their LICENSE and
+	  README (if avaialble) files will be installed to
+	  ASTVARLIBDIR/documentation/thirdparty/<product_name>.
+
+	  Example use with codecs:
+
+	  The codecs/codecs.xml file is a menuselect style xml file that lists
+	  the codecs to be included.  Their support levels are 'external', which
+	  triggers the download and install, and defaultenabled is no.  Also
+	  because codec_g729a is actually in a directory named codec_g729 on the
+	  download server, the newly added 'member_data' element is used to
+	  override the default of the directory name being the package name.  You
+	  can use the 'directory_name' attribute to keep default base URL
+	  (http://downloads.digium.com/pub/telephony/) but use the new directory,
+	  or you use the 'remote_url' attribute to specify a full URL to the
+	  download directory.  In this case, you must still follow the same
+	  subdirectory naming conventions as that used for the packages located
+	  at 'http://downloads.digium.com/pub/telephony'.
+
+	  A new configure option '--with-externals-cache' was added and like
+	  '--with-sounds-cache' it allows the installer to cache tarballs so
+	  they're not downloaded every time.
+
+	  To assist with the download and install process, each external package
+	  now has a manifest.xml file that, among other things, contains a package
+	  version and checksums for each file in the tarball.  The manifest is
+	  saved to both the cache directory and ASTMODDIR and together with the
+	  manifest.xml on the downloads site, tells the install scripts whether
+	  a download and/or update is needed.
+
+	  bash and xmlstarlet are required for downloader operation.  If they're
+	  not installed, the external items in menuselect will be unavailable.
+
+	  Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a
+	  (cherry picked from commit 8b6f9dde14d5ff30dd63c83ed20fb04dcda07e3e)
+
 2016-09-23 20:38 +0000  Asterisk Development Team <asteriskteam@digium.com>
 
 	* asterisk 14.0.0 Released.

Makefile

@@ -618,9 +618,10 @@ $(SUBDIRS_INSTALL):
 
 NEWMODS:=$(foreach d,$(MOD_SUBDIRS),$(notdir $(wildcard $(d)/*.so)))
 OLDMODS=$(filter-out $(NEWMODS) $(notdir $(DESTDIR)$(ASTMODDIR)),$(notdir $(wildcard $(DESTDIR)$(ASTMODDIR)/*.so)))
+BADMODS=$(strip $(filter-out $(shell ./build_tools/list_valid_installed_externals),$(OLDMODS)))
 
 oldmodcheck:
-	@if [ -n "$(OLDMODS)" ]; then \
+	@if [ -n "$(BADMODS)" ]; then \
 		echo " WARNING WARNING WARNING" ;\
 		echo "" ;\
 		echo " Your Asterisk modules directory, located at" ;\
@@ -630,7 +631,7 @@ oldmodcheck:
 		echo " modules are compatible with this version before" ;\
 		echo " attempting to run Asterisk." ;\
 		echo "" ;\
-		for f in $(OLDMODS); do \
+		for f in $(BADMODS); do \
 			echo "    $$f" ;\
 		done ;\
 		echo "" ;\
@@ -980,7 +981,7 @@ menuselect/nmenuselect: menuselect/makeopts .lastclean
 menuselect/makeopts: makeopts .lastclean
 	+$(MAKE_MENUSELECT) makeopts
 
-menuselect-tree: $(foreach dir,$(filter-out main,$(MOD_SUBDIRS)),$(wildcard $(dir)/*.c) $(wildcard $(dir)/*.cc)) build_tools/cflags.xml build_tools/cflags-devmode.xml sounds/sounds.xml build_tools/embed_modules.xml utils/utils.xml agi/agi.xml configure makeopts
+menuselect-tree: $(foreach dir,$(filter-out main,$(MOD_SUBDIRS)),$(wildcard $(dir)/*.c) $(wildcard $(dir)/*.cc) $(wildcard $(dir)/*.xml)) build_tools/cflags.xml build_tools/cflags-devmode.xml sounds/sounds.xml build_tools/embed_modules.xml utils/utils.xml agi/agi.xml configure makeopts
 	@echo "Generating input for menuselect ..."
 	@echo "<?xml version=\"1.0\"?>" > $@
 	@echo >> $@

Makefile.moddir_rules

@@ -146,6 +146,18 @@ clean::
 install:: all
 	@echo "Installing modules from `basename $(CURDIR)`..."
 	@for x in $(LOADABLE_MODS:%=%.so); do $(INSTALL) -m 755 $$x "$(DESTDIR)$(ASTMODDIR)" ; done
+ifneq ($(findstring :,$(XMLSTARLET)$(BASH)),:)
+	@if [ -f .moduleinfo ] ; then \
+		declare -A DISABLED_MODS ;\
+		for x in $(MENUSELECT_$(MENUSELECT_CATEGORY)) ; do DISABLED_MODS[$${x}]=1 ; done ;\
+		EXTERNAL_MODS=$$(xmlstarlet sel -t -m "/category/member[support_level = 'external']" -v "@name" -n .moduleinfo) ;\
+		for x in $${EXTERNAL_MODS} ; do \
+			if [ -z "$${DISABLED_MODS[$${x}]}" ] ; then \
+				$(ASTTOPDIR)/build_tools/download_externals $${x} ;\
+			fi ;\
+		done ;\
+	fi
+endif
 
 uninstall::
 

asterisk-14.0.1-summary.html

@@ -0,0 +1,41 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-14.0.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-14.0.1</h3><h3 align="center">Date: 2016-09-27</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
+<li><a href="#summary">Summary</a></li>
+<li><a href="#contributors">Contributors</a></li>
+<li><a href="#open_issues">Open Issues</a></li>
+<li><a href="#commits">Other Changes</a></li>
+<li><a href="#diffstat">Diffstat</a></li>
+</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-14.0.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
+<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
+<tr valign="top"><td width="33%">5 gtjoseph <gjoseph@digium.com><br/>4 Kevin Harwell <kharwell@digium.com><br/></td><td width="33%"><td width="33%">3 Kevin Harwell <kharwell@digium.com><br/></td></tr>
+</table><hr><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Improvement</h3><h4>Category: Resources/res_format_attr_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26409">ASTERISK-26409</a>: codec_opus: Update Asterisk to support the translation codec.<br/>Reported by: Kevin Harwell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3d82571068f6830604201c2b7aff86305e97760">[d3d8257106]</a> gtjoseph -- codec_opus: Add download ability to menuselect</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e98c8d5c868e433efd2c0521bd3eb9e0162baaa">[1e98c8d5c8]</a> gtjoseph -- codec_opus: Replace res_format_attr_opus with the one from codec_opus</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a091e15985794539d120795e7b4dbf2bc7f449ea">[a091e15985]</a> gtjoseph -- format_ogg_opus: New format</li>
+</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
+<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=80b3a952e038be199abd6de0a7f5cb5a524e4b2b">80b3a952e0</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=404c4a6820ce1748389832927a429eb3c655174a">404c4a6820</a></td><td>Kevin Harwell</td><td>.version: Update for 14.0.1</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e0f992505d58f7dec99bb828c9be5af8530d602">8e0f992505</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 14.0.1</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=801966afa9d5b188deaeaa29e6e1fe3e5cbe912e">801966afa9</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 14.0.1</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a13b54df83a352dece6f2750c61b9399b78c8968">a13b54df83</a></td><td>gtjoseph</td><td>build_tools:  Add ability to download variants to download_externals</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13cae45bb5aa072780bb6645f2146fce7edb9935">13cae45bb5</a></td><td>gtjoseph</td><td>build: Add download capability for external packages</td></tr>
+</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-14.0.0-summary.html                 | 4318 -----------
+asterisk-14.0.0-summary.txt                  |10112 ---------------------------
+b/.version                                   |    2
+b/CHANGES                                    |   15
+b/Makefile                                   |    7
+b/Makefile.moddir_rules                      |   12
+b/build_tools/download_externals             |  224
+b/build_tools/list_valid_installed_externals |   55
+b/build_tools/make_version                   |    4
+b/build_tools/menuselect-deps.in             |    3
+b/codecs/codecs.xml                          |   32
+b/configure                                  |  234
+b/configure.ac                               |   24
+b/formats/format_ogg_opus.c                  |  229
+b/include/asterisk/autoconfig.h.in           |    3
+b/include/asterisk/opus.h                    |   51
+b/makeopts.in                                |    5
+b/res/res.xml                                |   13
+b/res/res_format_attr_opus.c                 |  304
+19 files changed, 1040 insertions(+), 14607 deletions(-)</pre><br></html>

asterisk-14.0.1-summary.txt

@@ -0,0 +1,134 @@
+                                Release Summary
+
+                                asterisk-14.0.1
+
+                                Date: 2016-09-27
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Open Issues
+    4. Other Changes
+    5. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This release is a point release of an existing major version. The changes
+   included were made to address problems that have been identified in this
+   release series, or are minor, backwards compatible new features or
+   improvements. Users should be able to safely upgrade to this version if
+   this release series is already in use. Users considering upgrading from a
+   previous version are strongly encouraged to review the UPGRADE.txt
+   document as well as the CHANGES document for information about upgrading
+   to this release series.
+
+   The data in this summary reflects changes that have been made since the
+   previous release, asterisk-14.0.0.
+
+     ----------------------------------------------------------------------
+
+                                  Contributors
+
+                                 [Back to Top]
+
+   This table lists the people who have submitted code, those that have
+   tested patches, as well as those that reported issues on the issue tracker
+   that were resolved in this release. For coders, the number is how many of
+   their patches (of any size) were committed into this release. For testers,
+   the number is the number of times their name was listed as assisting with
+   testing a patch. Finally, for reporters, the number is the number of
+   issues that they reported that were affected by commits that went into
+   this release.
+
+   Coders                   Testers                  Reporters                
+   5 gtjoseph                                        3 Kevin Harwell          
+   4 Kevin Harwell          
+
+     ----------------------------------------------------------------------
+
+                                  Open Issues
+
+                                 [Back to Top]
+
+   This is a list of all open issues from the issue tracker that were
+   referenced by changes that went into this release.
+
+  Improvement
+
+    Category: Resources/res_format_attr_opus
+
+   ASTERISK-26409: codec_opus: Update Asterisk to support the translation
+   codec.
+   Reported by: Kevin Harwell
+     * [d3d8257106] gtjoseph -- codec_opus: Add download ability to
+       menuselect
+     * [1e98c8d5c8] gtjoseph -- codec_opus: Replace res_format_attr_opus with
+       the one from codec_opus
+     * [a091e15985] gtjoseph -- format_ogg_opus: New format
+
+     ----------------------------------------------------------------------
+
+                      Commits Not Associated with an Issue
+
+                                 [Back to Top]
+
+   This is a list of all changes that went into this release that did not
+   reference a JIRA issue.
+
+   +------------------------------------------------------------------------+
+   | Revision   | Author        | Summary                                   |
+   |------------+---------------+-------------------------------------------|
+   | 80b3a952e0 | Kevin Harwell | Release summaries: Remove previous        |
+   |            |               | versions                                  |
+   |------------+---------------+-------------------------------------------|
+   | 404c4a6820 | Kevin Harwell | .version: Update for 14.0.1               |
+   |------------+---------------+-------------------------------------------|
+   | 8e0f992505 | Kevin Harwell | .lastclean: Update for 14.0.1             |
+   |------------+---------------+-------------------------------------------|
+   | 801966afa9 | Kevin Harwell | realtime: Add database scripts for 14.0.1 |
+   |------------+---------------+-------------------------------------------|
+   | a13b54df83 | gtjoseph      | build_tools: Add ability to download      |
+   |            |               | variants to download_externals            |
+   |------------+---------------+-------------------------------------------|
+   | 13cae45bb5 | gtjoseph      | build: Add download capability for        |
+   |            |               | external packages                         |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                Diffstat Results
+
+                                 [Back to Top]
+
+   This is a summary of the changes to the source code that went into this
+   release that was generated using the diffstat utility.
+
+ asterisk-14.0.0-summary.html                 | 4318 -----------
+ asterisk-14.0.0-summary.txt                  |10112 ---------------------------
+ b/.version                                   |    2
+ b/CHANGES                                    |   15
+ b/Makefile                                   |    7
+ b/Makefile.moddir_rules                      |   12
+ b/build_tools/download_externals             |  224
+ b/build_tools/list_valid_installed_externals |   55
+ b/build_tools/make_version                   |    4
+ b/build_tools/menuselect-deps.in             |    3
+ b/codecs/codecs.xml                          |   32
+ b/configure                                  |  234
+ b/configure.ac                               |   24
+ b/formats/format_ogg_opus.c                  |  229
+ b/include/asterisk/autoconfig.h.in           |    3
+ b/include/asterisk/opus.h                    |   51
+ b/makeopts.in                                |    5
+ b/res/res.xml                                |   13
+ b/res/res_format_attr_opus.c                 |  304
+ 19 files changed, 1040 insertions(+), 14607 deletions(-)

build_tools/download_externals

@@ -0,0 +1,224 @@
+#!/usr/bin/env bash
+
+if [[ ( ${BASH_VERSINFO[0]} == 4 && ${BASH_VERSINFO[1]} > 1 ) || ${BASH_VERSINFO[0]} > 4 ]] ; then
+	shopt -s compat41
+fi
+set -e
+
+ASTTOPDIR=${ASTTOPDIR:-.}
+
+module_name=${1%%-*}
+variant=${1##*-}
+
+if [[ "${variant}" = "${module_name}" ]] ; then
+	unset variant
+fi
+
+if [[ -z ${module_name} ]] ; then
+	echo "You must supply a module name."
+	exit 64
+fi
+
+tmpdir=$(mktemp -d)
+if [[ -z "${tmpdir}" ]] ; then
+	echo "${module_name}: Unable to create temporary directory."
+	exit 1
+fi
+trap "rm -rf ${tmpdir}" EXIT
+
+sed -r -e "s/^([^ =]+)\s*=\s*(.*)$/\1=\"\2\"/g" ${ASTTOPDIR}/makeopts >${tmpdir}/makeopts
+source ${tmpdir}/makeopts
+if [[ -z "${ASTMODDIR}" ]] ; then
+	echo "${module_name}: Unable to parse ${ASTTOPDIR}/makeopts."
+	exit 1
+fi
+
+XMLSTARLET=${XMLSTARLET:-xmlstarlet}
+if [[ "${XMLSTARLET}" = ":" ]] ; then
+	echo "${module_name}: The externals downloader requires xmlstarlet to be installed."
+	exit 1
+fi
+
+cache_dir="${EXTERNALS_CACHE_DIR}"
+if [[ -z ${cache_dir} ]] ; then
+	cache_dir=${tmpdir}
+fi
+
+version=$(${ASTTOPDIR}/build_tools/make_version ${ASTTOPDIR})
+if [[ ! ${version} =~ ^(GIT-)?([^.-]+)[.-].* ]] ; then
+	echo "${module_name}: Couldn't parse version ${version}"
+	exit 1
+fi
+major_version=${BASH_REMATCH[2]}
+
+if [[ "${major_version}" == "master" ]] ; then
+	echo "${module_name}: External module downloading is not available in the 'master' git branch.  Please disable in menuselect and download manually."
+	exit 1
+fi
+
+major_version=${major_version}.0
+
+if [[ "${HOST_CPU}" = "x86_64" ]] ; then
+	host_bits=64
+else
+	host_bits=32
+fi
+
+if [[ -z "${variant}" ]] ; then
+	variants=$(${XMLSTARLET} sel -t -m "/menu/category/member[@name = '${module_name}']/member_data/downloader/variants/variant" -v "@tag" -n ${ASTTOPDIR}/menuselect-tree || :)
+	member_name=${module_name}
+	for tag in ${variants} ; do
+		condition=$(${XMLSTARLET} sel -t -v "/menu/category/member[@name = '${module_name}']/member_data/downloader/variants/variant[@tag = '${tag}']/@condition" ${ASTTOPDIR}/menuselect-tree || :)
+		variant=$(eval "if $condition ; then echo $tag ; fi")
+		if [[ -n "${variant}" ]] ; then
+			break
+		fi
+	done
+else
+	member_name=${module_name}${variant:+-${variant}}
+fi
+
+full_name=${module_name}${variant:+-${variant}}
+variant_manifest=manifest${variant:+-${variant}}.xml
+
+# Override the remote base for all packages
+# useful for testing
+remote_url=${REMOTE_BASE:+${REMOTE_BASE}/asterisk-${major_version}/x86-${host_bits}}
+
+if [[ -z "${remote_url}" ]] ; then
+	remote_url=$(${XMLSTARLET} sel -t -v "/menu/category/member[@name = '${member_name}']/member_data/downloader/@remote_url" ${ASTTOPDIR}/menuselect-tree || :)
+	if [[ -n "${remote_url}" ]] ; then
+		remote_url="${remote_url}/asterisk-${major_version}/x86-${host_bits}"
+	else
+		directory_name=$(${XMLSTARLET} sel -t -v "/menu/category/member[@name = '${member_name}']/member_data/downloader/@directory_name" ${ASTTOPDIR}/menuselect-tree || :)
+		remote_url="http://downloads.digium.com/pub/telephony/${directory_name:-${module_name}}/asterisk-${major_version}/x86-${host_bits}"
+	fi
+fi
+
+version_convert() {
+	local v=${1##*_}
+	if [[ ${v} =~ ([0-9]+)[.]([0-9]+)[.]([0-9]+) ]] ; then
+		v=$(( ${BASH_REMATCH[1]}<<18 | ${BASH_REMATCH[2]}<<9 | ${BASH_REMATCH[3]} ))
+	fi
+	echo ${v}
+}
+
+${WGET} -q -O ${tmpdir}/${variant_manifest} ${remote_url}/${variant_manifest} || {
+	echo "${full_name}: Unable to fetch ${remote_url}/${variant_manifest}"
+	exit 1
+}
+
+rpv=$(${XMLSTARLET} sel -t -v "/package/@version" ${tmpdir}/${variant_manifest})
+rpvi=$(version_convert ${rpv})
+echo "${full_name}: Remote package version ${rpv} (${rpvi})"
+
+module_dir=${full_name}-${rpv}-x86_${host_bits}
+tarball=${module_dir}.tar.gz
+
+export need_install=0
+
+if [[ -f ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml ]] ; then
+	package_arch=$(${XMLSTARLET} sel -t -v "/package/@arch" ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml)
+	ipv=$(${XMLSTARLET} sel -t -v "/package/@version" ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml)
+	package_variant=$(${XMLSTARLET} sel -t -v "/package/@variant" ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml || :)
+	ipvi=$(version_convert ${ipv})
+	ip_major=${ipv%_*}
+	echo "${full_name}: Installed package version ${ipv} (${ipvi})"
+	if [[ "${ip_major}" != "${major_version}" || "${package_arch}" != "x86_${host_bits}" || "${package_variant}" != "${variant}" ]] ; then
+		echo "${full_name}: The installed package is not for this version of Asterisk.  Reinstalling."
+		need_install=1
+	elif [[ ${rpvi} > ${ipvi} ]] ; then
+		echo "${full_name}: A newer package is available"
+		need_install=1
+	else
+		sums=$(${XMLSTARLET} sel -t -m "//file" -v "@md5sum" -n ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml)
+		for sum in ${sums} ; do
+			install_path=$(${XMLSTARLET} sel -t -v "//file[@md5sum = '${sum}']/@install_path" ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml )
+			executable=$(${XMLSTARLET} sel -t -v "//file[@md5sum = '${sum}']/@executable" ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml )
+			f=${DESTDIR}$(eval echo ${install_path})
+			if [[ ! -f ${f} ]] ; then
+				echo Not found: ${f}
+				need_install=1
+				break
+			else
+				if [[ "$executable" = "yes" ]] ; then
+					# There are easier ways of doing this (objcopy --dump-section) but not in older bunutils
+					length_offset=$(objdump -h $f | sed -n -r -e "s/^\s+[0-9]+\s+.ast_manifest\s+([0-9a-fA-F]+)\s+[0-9a-fA-F]+\s+[0-9a-fA-F]+\s+([0-9a-fA-F]+)\s+.*$/0x\1 0x\2/p")
+					tags=$($(eval 'printf "dd if=$f bs=1 count=%d skip=%d\n" $length_offset') 2>/dev/null)
+					if [[ -n "${tags}" && "${tags}" != "${module_name},${variant},${rpv}" ]] ; then
+						echo Tag mismatch: ${f} File: "${tags}" Manifest: "${module_name},${variant},${rpv}"
+						need_install=1
+						break
+					fi
+				fi
+
+				cs=$(md5sum ${f} | cut -b1-32)
+				if [[ "${cs}" !=  "${sum}" ]] ; then
+					echo Checksum mismatch: ${f}
+					need_install=1
+					break
+				fi
+			fi
+		done
+	fi
+else
+	need_install=1
+fi
+
+if [[ ${need_install} == 1 ]] ; then
+	if [[ ( -n "${ipvi}" ) && ${ipvi} > ${rpvi} ]] ; then
+		echo "${full_name}: Installed package is newer than that available for download."
+		exit 0
+	fi
+else
+	echo "${full_name} is up to date."
+	exit 0;
+fi
+
+need_download=1
+if [[ -f ${cache_dir}/${full_name}.manifest.xml ]] ; then
+	cpv=$(${XMLSTARLET} sel -t -v "/package/@version" ${cache_dir}/${full_name}.manifest.xml)
+	cpvi=$(version_convert ${cpv})
+	echo "${full_name}: Cached package version ${cpv} (${cpvi})"
+	if [[ ${cpvi} == ${rpvi} && ( -f ${cache_dir}/${tarball} ) ]] ; then
+		echo "${full_name}: Cached version is available."
+		need_download=0
+	fi
+fi
+
+if [[ ${need_download} = 1 ]] ; then
+	echo "${full_name}: Downloading ${remote_url}/${tarball}"
+	${WGET} -q -O ${cache_dir}/${tarball} ${remote_url}/${tarball} || {
+		echo "${full_name}: Unable to fetch ${remote_url}/${tarball}"
+		exit 1
+	}
+	cp ${tmpdir}/${variant_manifest}  ${cache_dir}/${full_name}.manifest.xml
+fi
+
+tar -xzf ${cache_dir}/${tarball} -C ${cache_dir}
+trap "rm -rf ${cache_dir}/${module_dir} ; rm -rf ${tmpdir}" EXIT
+
+echo "${full_name}: Installing."
+
+if [[ $EUID == 0 ]] ; then
+	install_params="--group=0 --owner=0"
+fi
+
+names=$(${XMLSTARLET} sel -t -m "//file" -v "@name" -n ${cache_dir}/${module_dir}/manifest.xml)
+for name in ${names} ; do
+	source_path=${cache_dir}/${module_dir}/${name}
+	install_path=$(${XMLSTARLET} sel -t -v "//file[@name = '${name}']/@install_path" ${cache_dir}/${module_dir}/manifest.xml)
+	install_path=${DESTDIR}$(eval echo ${install_path})
+	executable=$(${XMLSTARLET} sel -t -v "//file[@name = '${name}']/@executable" ${cache_dir}/${module_dir}/manifest.xml || :)
+	if [[ "${executable}" = "yes" ]] ; then
+		mode=0755
+	else
+		mode=0644
+	fi
+
+	${INSTALL} -Dp ${install_params} --mode=${mode} ${source_path} ${install_path}
+
+done
+${INSTALL} -Dp ${install_params} --mode=0644 ${cache_dir}/${module_dir}/manifest.xml ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml
+
+echo "${full_name}: Installed."

build_tools/list_valid_installed_externals

@@ -0,0 +1,55 @@
+#!/usr/bin/env bash
+
+if [[ ( ${BASH_VERSINFO[0]} == 4 && ${BASH_VERSINFO[1]} > 1 ) || ${BASH_VERSINFO[0]} > 4 ]] ; then
+	shopt -s compat41
+fi
+set -e
+
+ASTTOPDIR=${ASTTOPDIR:-.}
+
+tmpdir=$(mktemp -d)
+if [[ -z "${tmpdir}" ]] ; then
+	echo "${module_name}: Unable to create temporary directory."
+	exit 1
+fi
+trap "rm -rf ${tmpdir}" EXIT
+
+sed -r -e "s/^([^ =]+)\s*=\s*(.*)$/\1=\"\2\"/g" ${ASTTOPDIR}/makeopts >${tmpdir}/makeopts
+source ${tmpdir}/makeopts
+if [[ -z "${ASTMODDIR}" ]] ; then
+	echo "${module_name}: Unable to parse ${ASTTOPDIR}/makeopts."
+	exit 1
+fi
+
+XMLSTARLET=${XMLSTARLET:-xmlstarlet}
+if [[ "${XMLSTARLET}" = ":" ]] ; then
+	echo "${module_name}: The externals downloader requires xmlstarlet to be installed."
+	exit 1
+fi
+
+version=$(${ASTTOPDIR}/build_tools/make_version ${ASTTOPDIR})
+if [[ ! ${version} =~ ^(GIT-)?([^.-]+)[.-].* ]] ; then
+	echo "${module_name}: Couldn't parse version ${version}"
+	exit 1
+fi
+major_version=${BASH_REMATCH[2]}.0
+
+if [[ "${HOST_CPU}" = "x86_64" ]] ; then
+	host_bits=64
+else
+	host_bits=32
+fi
+
+names=""
+for manifest in ${DESTDIR}${ASTMODDIR}/*.manifest.xml ; do
+	if [ ! -f "$manifest" ] ; then
+		break
+	fi
+	package_version=$(${XMLSTARLET} sel -t -v "/package/@version" ${manifest})
+	package_major_version=${package_version%_*}
+	package_arch=$(${XMLSTARLET} sel -t -v "/package/@arch" ${manifest})
+	if [[ "$package_major_version" = "$major_version" && "${package_arch}" = "x86_${host_bits}" ]] ; then
+		names+=$(${XMLSTARLET} sel -t -m "//file[@executable = 'yes']" -v "concat(@name, ' ')" ${manifest})
+	fi
+done
+echo $names

build_tools/make_version

@@ -16,6 +16,8 @@ elif [ -d ${1}/.git ]; then
         echo "UNKNOWN__and_probably_unsupported"
         exit 1
     fi
+	cd ${1} 
+    
     # If the first log commit messages indicates that this is checked into
     # subversion, we'll just use the SVN- form of the revision.
     MODIFIED=""

build_tools/menuselect-deps.in

@@ -30,6 +30,8 @@ KQUEUE=@PBX_KQUEUE@
 LDAP=@PBX_LDAP@
 LIBEDIT=@PBX_LIBEDIT@
 LIBXML2=@PBX_LIBXML2@
+XMLSTARLET=@PBX_XMLSTARLET@
+BASH=@PBX_BASH@
 LTDL=@PBX_LTDL@
 LUA=@PBX_LUA@
 MISDN=@PBX_MISDN@
@@ -42,6 +44,7 @@ NEON29=@PBX_NEON29@
 OGG=@PBX_OGG@
 OPENH323=@PBX_OPENH323@
 OPUS=@PBX_OPUS@
+OPUSFILE=@PBX_OPUSFILE@
 OSPTK=@PBX_OSPTK@
 OSS=@PBX_OSS@
 PGSQL=@PBX_PGSQL@

codecs/codecs.xml

@@ -0,0 +1,32 @@
+<member name="codec_opus" displayname="Download the Opus codec from Digium.  See http://downloads.digium.com/pub/telephony/codec_opus/README.">
+	<support_level>external</support_level>
+	<depend>xmlstarlet</depend>
+	<depend>bash</depend>
+	<depend>res_format_attr_opus</depend>
+	<defaultenabled>no</defaultenabled>
+</member>
+<member name="codec_silk" displayname="Download the SILK codec from Digium.  See http://downloads.digium.com/pub/telephony/codec_silk/README.">
+	<support_level>external</support_level>
+	<depend>xmlstarlet</depend>
+	<depend>bash</depend>
+	<defaultenabled>no</defaultenabled>
+</member>
+<member name="codec_siren7" displayname="Download the Siren7 codec from Digium.  See http://downloads.digium.com/pub/telephony/codec_siren7/README.">
+	<support_level>external</support_level>
+	<depend>xmlstarlet</depend>
+	<depend>bash</depend>
+	<defaultenabled>no</defaultenabled>
+</member>
+<member name="codec_siren14" displayname="Download the Siren14 codec from Digium.  See http://downloads.digium.com/pub/telephony/codec_siren14/README.">
+	<support_level>external</support_level>
+	<depend>xmlstarlet</depend>
+	<depend>bash</depend>
+	<defaultenabled>no</defaultenabled>
+</member>
+<member name="codec_g729a" displayname="Download the g729a codec from Digium.  A license must be purchased for this codec.  See http://downloads.digium.com/pub/telephony/codec_g729/README.">
+	<support_level>external</support_level>
+	<depend>xmlstarlet</depend>
+	<depend>bash</depend>
+	<defaultenabled>no</defaultenabled>
+	<member_data><downloader directory_name="codec_g729"/></member_data>
+</member>

configure

@@ -822,6 +822,7 @@ PBX_SPANDSP
 SPANDSP_DIR
 SPANDSP_INCLUDE
 SPANDSP_LIB
+EXTERNALS_CACHE_DIR
 SOUNDS_CACHE_DIR
 PBX_SDL_IMAGE
 SDL_IMAGE_DIR
@@ -984,6 +985,10 @@ PBX_OSPTK
 OSPTK_DIR
 OSPTK_INCLUDE
 OSPTK_LIB
+PBX_OPUSFILE
+OPUSFILE_DIR
+OPUSFILE_INCLUDE
+OPUSFILE_LIB
 PBX_OPUS
 OPUS_DIR
 OPUS_INCLUDE
@@ -1198,6 +1203,8 @@ PTHREAD_CC
 ax_pthread_config
 MD5
 SOXMIX
+PBX_BASH
+PBX_XMLSTARLET
 PBX_FLEX
 PBX_BISON
 OPENSSL
@@ -1207,6 +1214,7 @@ DOWNLOAD
 FETCH
 ALEMBIC
 GIT
+BASH
 XMLSTARLET
 XMLLINT
 KPATHSEA
@@ -1379,6 +1387,7 @@ with_newt
 with_ogg
 with_openr2
 with_opus
+with_opusfile
 with_osptk
 with_oss
 with_postgres
@@ -1393,6 +1402,7 @@ with_resample
 with_sdl
 with_SDL_image
 with_sounds_cache
+with_externals_cache
 with_spandsp
 with_ss7
 with_speex
@@ -2122,6 +2132,7 @@ Optional Packages:
   --with-ogg=PATH         use OGG files in PATH
   --with-openr2=PATH      use MFR2 files in PATH
   --with-opus=PATH        use Opus files in PATH
+  --with-opusfile=PATH    use Opusfile files in PATH
   --with-osptk=PATH       use OSP Toolkit files in PATH
   --with-oss=PATH         use Open Sound System files in PATH
   --with-postgres=PATH    use PostgreSQL files in PATH
@@ -2138,6 +2149,8 @@ Optional Packages:
   --with-SDL_image=PATH   use Sdl Image files in PATH
   --with-sounds-cache=PATH
                           use cached sound tarfiles in PATH
+  --with-externals-cache=PATH
+                          use cached external module tarfiles in PATH
   --with-spandsp=PATH     use SPANDSP files in PATH
   --with-ss7=PATH         use ISDN SS7 files in PATH
   --with-speex=PATH       use Speex files in PATH
@@ -7483,6 +7496,47 @@ $as_echo "no" >&6; }
 fi
 
 
+# Extract the first word of "bash", so it can be a program name with args.
+set dummy bash; ac_word=$2
+{ $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5
+$as_echo_n "checking for $ac_word... " >&6; }
+if ${ac_cv_path_BASH+:} false; then :
+  $as_echo_n "(cached) " >&6
+else
+  case $BASH in
+  [\\/]* | ?:[\\/]*)
+  ac_cv_path_BASH="$BASH" # Let the user override the test with a path.
+  ;;
+  *)
+  as_save_IFS=$IFS; IFS=$PATH_SEPARATOR
+for as_dir in $PATH
+do
+  IFS=$as_save_IFS
+  test -z "$as_dir" && as_dir=.
+    for ac_exec_ext in '' $ac_executable_extensions; do
+  if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then
+    ac_cv_path_BASH="$as_dir/$ac_word$ac_exec_ext"
+    $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5
+    break 2
+  fi
+done
+  done
+IFS=$as_save_IFS
+
+  test -z "$ac_cv_path_BASH" && ac_cv_path_BASH=":"
+  ;;
+esac
+fi
+BASH=$ac_cv_path_BASH
+if test -n "$BASH"; then
+  { $as_echo "$as_me:${as_lineno-$LINENO}: result: $BASH" >&5
+$as_echo "$BASH" >&6; }
+else
+  { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5
+$as_echo "no" >&6; }
+fi
+
+
 # Extract the first word of "git", so it can be a program name with args.
 set dummy git; ac_word=$2
 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5
@@ -7790,6 +7844,20 @@ else
 fi
 
 
+if test "x${XMLSTARLET}" = "x:" ; then
+	PBX_XMLSTARLET=0
+else
+	PBX_XMLSTARLET=1
+fi
+
+
+if test "x${BASH}" = "x:" ; then
+	PBX_BASH=0
+else
+	PBX_BASH=1
+fi
+
+
 if test -n "$ac_tool_prefix"; then
   # Extract the first word of "${ac_tool_prefix}soxmix", so it can be a program name with args.
 set dummy ${ac_tool_prefix}soxmix; ac_word=$2
@@ -10715,6 +10783,38 @@ fi
 
 
 
+    OPUSFILE_DESCRIP="Opusfile"
+    OPUSFILE_OPTION="opusfile"
+    PBX_OPUSFILE=0
+
+# Check whether --with-opusfile was given.
+if test "${with_opusfile+set}" = set; then :
+  withval=$with_opusfile;
+	case ${withval} in
+	n|no)
+	USE_OPUSFILE=no
+	# -1 is a magic value used by menuselect to know that the package
+	# was disabled, other than 'not found'
+	PBX_OPUSFILE=-1
+	;;
+	y|ye|yes)
+	ac_mandatory_list="${ac_mandatory_list} OPUSFILE"
+	;;
+	*)
+	OPUSFILE_DIR="${withval}"
+	ac_mandatory_list="${ac_mandatory_list} OPUSFILE"
+	;;
+	esac
+
+fi
+
+
+
+
+
+
+
+
     OSPTK_DESCRIP="OSP Toolkit"
     OSPTK_OPTION="osptk"
     PBX_OSPTK=0
@@ -11488,6 +11588,30 @@ fi
 
 
 
+
+# Check whether --with-externals-cache was given.
+if test "${with_externals_cache+set}" = set; then :
+  withval=$with_externals_cache;
+	case ${withval} in
+	n|no)
+		unset EXTERNALS_CACHE_DIR
+		;;
+	*)
+		if test "x${withval}" = "x"; then
+			:
+		else
+			EXTERNALS_CACHE_DIR="${withval}"
+		fi
+		;;
+	esac
+
+else
+  :
+fi
+
+
+
+
     SPANDSP_DESCRIP="SPANDSP"
     SPANDSP_OPTION="spandsp"
     PBX_SPANDSP=0
@@ -29242,6 +29366,116 @@ _ACEOF
 fi
 
 
+# opusfile.h includes <opus_multistream.h> so we need to make sure that
+# either $OPUS_INCLUDE or /usr/include/opus is added to the search path.
+__opus_include=${OPUS_INCLUDE}
+if test -z "$__opus_include" -o x"$__opus_include" = x" " ; then
+	__opus_include=-I/usr/include/opus
+fi
+
+if test "x${PBX_OPUSFILE}" != "x1" -a "${USE_OPUSFILE}" != "no"; then
+   pbxlibdir=""
+   # if --with-OPUSFILE=DIR has been specified, use it.
+   if test "x${OPUSFILE_DIR}" != "x"; then
+      if test -d ${OPUSFILE_DIR}/lib; then
+         pbxlibdir="-L${OPUSFILE_DIR}/lib"
+      else
+         pbxlibdir="-L${OPUSFILE_DIR}"
+      fi
+   fi
+   pbxfuncname="op_open_callbacks"
+   if test "x${pbxfuncname}" = "x" ; then   # empty lib, assume only headers
+      AST_OPUSFILE_FOUND=yes
+   else
+      ast_ext_lib_check_save_CFLAGS="${CFLAGS}"
+      CFLAGS="${CFLAGS} $__opus_include"
+      as_ac_Lib=`$as_echo "ac_cv_lib_opusfile_${pbxfuncname}" | $as_tr_sh`
+{ $as_echo "$as_me:${as_lineno-$LINENO}: checking for ${pbxfuncname} in -lopusfile" >&5
+$as_echo_n "checking for ${pbxfuncname} in -lopusfile... " >&6; }
+if eval \${$as_ac_Lib+:} false; then :
+  $as_echo_n "(cached) " >&6
+else
+  ac_check_lib_save_LIBS=$LIBS
+LIBS="-lopusfile ${pbxlibdir}  $LIBS"
+cat confdefs.h - <<_ACEOF >conftest.$ac_ext
+/* end confdefs.h.  */
+
+/* Override any GCC internal prototype to avoid an error.
+   Use char because int might match the return type of a GCC
+   builtin and then its argument prototype would still apply.  */
+#ifdef __cplusplus
+extern "C"
+#endif
+char ${pbxfuncname} ();
+int
+main ()
+{
+return ${pbxfuncname} ();
+  ;
+  return 0;
+}
+_ACEOF
+if ac_fn_c_try_link "$LINENO"; then :
+  eval "$as_ac_Lib=yes"
+else
+  eval "$as_ac_Lib=no"
+fi
+rm -f core conftest.err conftest.$ac_objext \
+    conftest$ac_exeext conftest.$ac_ext
+LIBS=$ac_check_lib_save_LIBS
+fi
+eval ac_res=\$$as_ac_Lib
+	       { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_res" >&5
+$as_echo "$ac_res" >&6; }
+if eval test \"x\$"$as_ac_Lib"\" = x"yes"; then :
+  AST_OPUSFILE_FOUND=yes
+else
+  AST_OPUSFILE_FOUND=no
+fi
+
+      CFLAGS="${ast_ext_lib_check_save_CFLAGS}"
+   fi
+
+   # now check for the header.
+   if test "${AST_OPUSFILE_FOUND}" = "yes"; then
+      OPUSFILE_LIB="${pbxlibdir} -lopusfile "
+      # if --with-OPUSFILE=DIR has been specified, use it.
+      if test "x${OPUSFILE_DIR}" != "x"; then
+         OPUSFILE_INCLUDE="-I${OPUSFILE_DIR}/include"
+      fi
+      OPUSFILE_INCLUDE="${OPUSFILE_INCLUDE} $__opus_include"
+      if test "xopus/opusfile.h" = "x" ; then	# no header, assume found
+         OPUSFILE_HEADER_FOUND="1"
+      else				# check for the header
+         ast_ext_lib_check_saved_CPPFLAGS="${CPPFLAGS}"
+         CPPFLAGS="${CPPFLAGS} ${OPUSFILE_INCLUDE}"
+         ac_fn_c_check_header_mongrel "$LINENO" "opus/opusfile.h" "ac_cv_header_opus_opusfile_h" "$ac_includes_default"
+if test "x$ac_cv_header_opus_opusfile_h" = xyes; then :
+  OPUSFILE_HEADER_FOUND=1
+else
+  OPUSFILE_HEADER_FOUND=0
+fi
+
+
+         CPPFLAGS="${ast_ext_lib_check_saved_CPPFLAGS}"
+      fi
+      if test "x${OPUSFILE_HEADER_FOUND}" = "x0" ; then
+         OPUSFILE_LIB=""
+         OPUSFILE_INCLUDE=""
+      else
+         if test "x${pbxfuncname}" = "x" ; then		# only checking headers -> no library
+            OPUSFILE_LIB=""
+         fi
+         PBX_OPUSFILE=1
+         cat >>confdefs.h <<_ACEOF
+#define HAVE_OPUSFILE 1
+_ACEOF
+
+      fi
+   fi
+fi
+
+
 
 if test "${USE_PWLIB}" != "no"; then
 	if test -n "${PWLIB_DIR}"; then

configure.ac

@@ -281,6 +281,7 @@ AC_PATH_PROG([CATDVI], [catdvi], :)
 AC_PATH_PROG([KPATHSEA], [kpsewhich], :)
 AC_PATH_PROG([XMLLINT], [xmllint], :)
 AC_PATH_PROG([XMLSTARLET], [xmlstarlet], :)
+AC_PATH_PROG([BASH], [bash], :)
 AC_PATH_PROG([GIT], [git], :)
 AC_PATH_PROG([ALEMBIC], [alembic], :)
 if test "${WGET}" != ":" ; then
@@ -340,6 +341,20 @@ else
 fi
 AC_SUBST(PBX_FLEX)
 
+if test "x${XMLSTARLET}" = "x:" ; then
+	PBX_XMLSTARLET=0
+else
+	PBX_XMLSTARLET=1
+fi
+AC_SUBST(PBX_XMLSTARLET)
+
+if test "x${BASH}" = "x:" ; then
+	PBX_BASH=0
+else
+	PBX_BASH=1
+fi
+AC_SUBST(PBX_BASH)
+
 AC_CHECK_TOOL([SOXMIX], [soxmix], [:])
 if test "${SOXMIX}" != ":" ; then
 	AC_DEFINE([HAVE_SOXMIX], 1, [Define to 1 if your system has soxmix application.])
@@ -452,6 +467,7 @@ AST_EXT_LIB_SETUP([NEWT], [newt], [newt])
 AST_EXT_LIB_SETUP([OGG], [OGG], [ogg])
 AST_EXT_LIB_SETUP([OPENR2], [MFR2], [openr2])
 AST_EXT_LIB_SETUP([OPUS], [Opus], [opus])
+AST_EXT_LIB_SETUP([OPUSFILE], [Opusfile], [opusfile])
 AST_EXT_LIB_SETUP([OSPTK], [OSP Toolkit], [osptk])
 AST_EXT_LIB_SETUP([OSS], [Open Sound System], [oss])
 AST_EXT_LIB_SETUP([PGSQL], [PostgreSQL], [postgres])
@@ -520,6 +536,7 @@ AST_EXT_LIB_SETUP([RESAMPLE], [LIBRESAMPLE], [resample])
 AST_EXT_LIB_SETUP([SDL], [Sdl], [sdl])
 AST_EXT_LIB_SETUP([SDL_IMAGE], [Sdl Image], [SDL_image])
 AST_OPTION_ONLY([sounds-cache], [SOUNDS_CACHE_DIR], [cached sound tarfiles], [])
+AST_OPTION_ONLY([externals-cache], [EXTERNALS_CACHE_DIR], [cached external module tarfiles], [])
 AST_EXT_LIB_SETUP([SPANDSP], [SPANDSP], [spandsp])
 AST_EXT_LIB_SETUP([SS7], [ISDN SS7], [ss7])
 AST_EXT_LIB_SETUP([SPEEX], [Speex], [speex])
@@ -2274,6 +2291,13 @@ AST_EXT_LIB_CHECK([SS7], [ss7], [ss7_set_isup_timer], [libss7.h])
 AST_EXT_LIB_CHECK([OPENR2], [openr2], [openr2_chan_new], [openr2.h])
 
 AST_EXT_LIB_CHECK([OPUS], [opus], [opus_encoder_create], [opus/opus.h])
+# opusfile.h includes <opus_multistream.h> so we need to make sure that
+# either $OPUS_INCLUDE or /usr/include/opus is added to the search path.
+__opus_include=${OPUS_INCLUDE}
+if test -z "$__opus_include" -o x"$__opus_include" = x" " ; then
+	__opus_include=-I/usr/include/opus
+fi
+AST_EXT_LIB_CHECK([OPUSFILE], [opusfile], [op_open_callbacks], [opus/opusfile.h], [], [$__opus_include])
 
 if test "${USE_PWLIB}" != "no"; then
 	if test -n "${PWLIB_DIR}"; then

formats/format_ogg_opus.c

@@ -0,0 +1,229 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2016, Digium, Inc.
+ *
+ * Mark Michelson <mmichelson@digium.com>
+ *
+ * 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.
+ */
+
+/*** MODULEINFO
+	<depend>opusfile</depend>
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <opus/opus.h>
+#include <opus/opusfile.h>
+#include "asterisk/mod_format.h"
+#include "asterisk/utils.h"
+#include "asterisk/module.h"
+#include "asterisk/format_cache.h"
+
+/* 120ms of 48KHz audio */
+#define SAMPLES_MAX 5760
+#define BUF_SIZE (2 * SAMPLES_MAX)
+
+struct ogg_opus_desc {
+	OggOpusFile *of;
+};
+
+static int fread_wrapper(void *_stream, unsigned char *_ptr, int _nbytes)
+{
+	FILE *stream = _stream;
+	size_t bytes_read;
+
+	if (!stream || _nbytes < 0) {
+		return -1;
+	}
+
+	bytes_read = fread(_ptr, 1, _nbytes, stream);
+
+	return bytes_read > 0 || feof(stream) ? (int) bytes_read : OP_EREAD;
+}
+
+static int fseek_wrapper(void *_stream, opus_int64 _offset, int _whence)
+{
+	FILE *stream = _stream;
+
+	return fseeko(stream, (off_t) _offset, _whence);
+}
+
+static opus_int64 ftell_wrapper(void *_stream)
+{
+	FILE *stream = _stream;
+
+	return ftello(stream);
+}
+
+static int ogg_opus_open(struct ast_filestream *s)
+{
+	struct ogg_opus_desc *desc = (struct ogg_opus_desc *) s->_private;
+	OpusFileCallbacks cb = {
+		.read = fread_wrapper,
+		.seek = fseek_wrapper,
+		.tell = ftell_wrapper,
+		.close = NULL,
+	};
+
+	memset(desc, 0, sizeof(*desc));
+	desc->of = op_open_callbacks(s->f, &cb, NULL, 0, NULL);
+	if (!desc->of) {
+		return -1;
+	}
+
+	return 0;
+}
+
+static int ogg_opus_rewrite(struct ast_filestream *s, const char *comment)
+{
+	/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
+	ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
+	return -1;
+}
+
+static int ogg_opus_write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
+	ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
+	return -1;
+}
+
+static int ogg_opus_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
+{
+	int seek_result = -1;
+	off_t relative_pcm_pos;
+	struct ogg_opus_desc *desc = fs->_private;
+
+	switch (whence) {
+	case SEEK_SET:
+		seek_result = op_pcm_seek(desc->of, sample_offset);
+		break;
+	case SEEK_CUR:
+		if ((relative_pcm_pos = op_pcm_tell(desc->of)) < 0) {
+			seek_result = -1;
+			break;
+		}
+		seek_result = op_pcm_seek(desc->of, relative_pcm_pos + sample_offset);
+		break;
+	case SEEK_END:
+		if ((relative_pcm_pos = op_pcm_total(desc->of, -1)) < 0) {
+			seek_result = -1;
+			break;
+		}
+		seek_result = op_pcm_seek(desc->of, relative_pcm_pos - sample_offset);
+		break;
+	default:
+		ast_log(LOG_WARNING, "Unknown *whence* to seek on OGG/Opus streams!\n");
+		break;
+	}
+
+	/* normalize error value to -1,0 */
+	return (seek_result == 0) ? 0 : -1;
+}
+
+static int ogg_opus_trunc(struct ast_filestream *fs)
+{
+	/* XXX Unimplemented. This is only used when recording, and we don't support that right now. */
+	ast_log(LOG_ERROR, "Truncation is not supported on OGG/Opus streams!\n");
+	return -1;
+}
+
+static off_t ogg_opus_tell(struct ast_filestream *fs)
+{
+	struct ogg_opus_desc *desc = fs->_private;
+	off_t pos;
+
+	pos = (off_t) op_pcm_tell(desc->of);
+	if (pos < 0) {
+		return -1;
+	}
+	return pos;
+}
+
+static struct ast_frame *ogg_opus_read(struct ast_filestream *fs, int *whennext)
+{
+	struct ogg_opus_desc *desc = fs->_private;
+	int hole = 1;
+	int samples_read;
+	opus_int16 *out_buf;
+
+	AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+
+	out_buf = (opus_int16 *) fs->fr.data.ptr;
+
+	while (hole) {
+		samples_read = op_read(
+			desc->of,
+			out_buf,
+			SAMPLES_MAX,
+			NULL);
+
+		if (samples_read != OP_HOLE) {
+			hole = 0;
+		}
+	}
+
+	if (samples_read <= 0) {
+		return NULL;
+	}
+
+	fs->fr.datalen = samples_read * 2;
+	fs->fr.samples = samples_read;
+	*whennext = fs->fr.samples;
+
+	return &fs->fr;
+}
+
+static void ogg_opus_close(struct ast_filestream *fs)
+{
+	struct ogg_opus_desc *desc = fs->_private;
+
+	op_free(desc->of);
+}
+
+static struct ast_format_def opus_f = {
+	.name = "ogg_opus",
+	.exts = "opus",
+	.open = ogg_opus_open,
+	.rewrite = ogg_opus_rewrite,
+	.write = ogg_opus_write,
+	.seek = ogg_opus_seek,
+	.trunc = ogg_opus_trunc,
+	.tell = ogg_opus_tell,
+	.read = ogg_opus_read,
+	.close = ogg_opus_close,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+	.desc_size = sizeof(struct ogg_opus_desc),
+};
+
+static int load_module(void)
+{
+	opus_f.format = ast_format_slin48;
+	if (ast_format_def_register(&opus_f)) {
+		return AST_MODULE_LOAD_FAILURE;
+	}
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	return ast_format_def_unregister(opus_f.name);
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Opus audio",
+	.support_level = AST_MODULE_SUPPORT_CORE,
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_APP_DEPEND
+);

include/asterisk/autoconfig.h.in

@@ -556,6 +556,9 @@
 /* Define to 1 if you have the Opus library. */
 #undef HAVE_OPUS
 
+/* Define to 1 if you have the Opusfile library. */
+#undef HAVE_OPUSFILE
+
 /* Define this to indicate the ${OSPTK_DESCRIP} library */
 #undef HAVE_OSPTK
 

include/asterisk/opus.h

@@ -18,24 +18,47 @@
 
 /*!
  * \file
- * \brief Opus Format Attributes (http://tools.ietf.org/html/draft-ietf-payload-rtp-opus)
+ * \brief Codec opus externals and format attributes
  *
- * \author Lorenzo Miniero <lorenzo@meetecho.com>
+ * RFC - https://tools.ietf.org/rfc/rfc7587.txt
  */
 #ifndef _AST_FORMAT_OPUS_H_
 #define _AST_FORMAT_OPUS_H_
 
-/*! Opus format attribute key value pairs, all are accessible through ast_format_get_value()*/
-enum opus_attr_keys {
-	OPUS_ATTR_KEY_MAX_BITRATE, /*! value is an int (6000-510000 in spec). */
-	OPUS_ATTR_KEY_MAX_PLAYRATE, /*! value is an int (8000-48000), maximum output rate the receiver can render. */
-	OPUS_ATTR_KEY_MINPTIME, /*! value is an int (3-120 in spec, 10-60 in format.c), decoder's minimum length of time in milliseconds. */
-	OPUS_ATTR_KEY_STEREO, /*! value is an int, 1 prefer receiving stereo, 0 prefer mono. */
-	OPUS_ATTR_KEY_CBR, /*! value is an int, 1 use constant bitrate, 0 use variable bitrate. */
-	OPUS_ATTR_KEY_FEC, /*! value is an int, 1 encode with FEC, 0 do not use FEC. */
-	OPUS_ATTR_KEY_DTX, /*! value is an int, 1 dtx is enabled, 0 dtx not enabled. */
-	OPUS_ATTR_KEY_SPROP_CAPTURE_RATE, /*! value is an int (8000-48000), likely input rate we're going to produce. */
-	OPUS_ATTR_KEY_SPROP_STEREO, /*! value is an int, 1 likely to send stereo, 0 likely to send mono. */
-};
+/*! \brief Maximum sampling rate an endpoint is capable of receiving */
+#define CODEC_OPUS_ATTR_MAX_PLAYBACK_RATE "maxplaybackrate"
+/*! \brief An alias for maxplaybackrate (used in older versions) */
+#define CODEC_OPUS_ATTR_MAX_CODED_AUDIO_BANDWIDTH "maxcodedaudiobandwidth"
+/*! \brief Maximum sampling rate an endpoint is capable of sending */
+#define CODEC_OPUS_ATTR_SPROP_MAX_CAPTURE_RATE "sprop-maxcapturerate"
+/*! \brief Maximum duration of packet (in milliseconds) */
+#define CODEC_OPUS_ATTR_MAX_PTIME "maxptime"
+/*! \brief Duration of packet (in milliseconds) */
+#define CODEC_OPUS_ATTR_PTIME "ptime"
+/*! \brief Maximum average received bit rate (in bits per second) */
+#define CODEC_OPUS_ATTR_MAX_AVERAGE_BITRATE "maxaveragebitrate"
+/*! \brief Decode stereo (1) vs mono (0) */
+#define CODEC_OPUS_ATTR_STEREO "stereo"
+/*! \brief Likeliness of sender producing stereo (1) vs mono (0) */
+#define CODEC_OPUS_ATTR_SPROP_STEREO "sprop-stereo"
+/*! \brief Decoder prefers a constant (1) vs variable (0) bitrate */
+#define CODEC_OPUS_ATTR_CBR "cbr"
+/*! \brief Use forward error correction (1) or not (0) */
+#define CODEC_OPUS_ATTR_FEC "useinbandfec"
+/*! \brief Use discontinuous transmission (1) or not (0) */
+#define CODEC_OPUS_ATTR_DTX "usedtx"
+/*! \brief Custom data object */
+#define CODEC_OPUS_ATTR_DATA "data"
+
+/*! \brief Default attribute values */
+#define CODEC_OPUS_DEFAULT_SAMPLE_RATE 48000
+#define CODEC_OPUS_DEFAULT_MAX_PLAYBACK_RATE 48000
+#define CODEC_OPUS_DEFAULT_MAX_PTIME 120
+#define CODEC_OPUS_DEFAULT_PTIME 20
+#define CODEC_OPUS_DEFAULT_BITRATE -1000 /* OPUS_AUTO */
+#define CODEC_OPUS_DEFAULT_CBR 0
+#define CODEC_OPUS_DEFAULT_FEC 0
+#define CODEC_OPUS_DEFAULT_DTX 0
+#define CODEC_OPUS_DEFAULT_STEREO 0
 
 #endif /* _AST_FORMAT_OPUS_H */

makeopts.in

@@ -28,11 +28,13 @@ WGET=@WGET@
 FETCH=@FETCH@
 DOWNLOAD=@DOWNLOAD@
 SOUNDS_CACHE_DIR=@SOUNDS_CACHE_DIR@
+EXTERNALS_CACHE_DIR=@EXTERNALS_CACHE_DIR@
 RUBBER=@RUBBER@
 CATDVI=@CATDVI@
 KPATHSEA=@KPATHSEA@
 XMLLINT=@XMLLINT@
 XMLSTARLET=@XMLSTARLET@
+BASH=@BASH@
 MD5=@MD5@
 SHA1SUM=@SHA1SUM@
 OPENSSL=@OPENSSL@
@@ -218,6 +220,9 @@ OGG_LIB=@OGG_LIB@
 OPUS_INCLUDE=@OPUS_INCLUDE@
 OPUS_LIB=@OPUS_LIB@
 
+OPUSFILE_INCLUDE=@OPUSFILE_INCLUDE@
+OPUSFILE_LIB=@OPUSFILE_LIB@
+
 OSPTK_INCLUDE=@OSPTK_INCLUDE@
 OSPTK_LIB=@OSPTK_LIB@
 

res/res.xml

@@ -0,0 +1,13 @@
+<member name="res_digium_phone" displayname="Download the Digium Phone Module for Asterisk.  See http://downloads.digium.com/pub/telephony/res_digium_phone/README.">
+	<support_level>external</support_level>
+	<depend>xmlstarlet</depend>
+	<depend>bash</depend>
+	<defaultenabled>no</defaultenabled>
+	<member_data>
+		<downloader>
+			<variants>
+				<variant tag="bundled" condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/>
+			</variants>
+		</downloader>
+	</member_data>
+</member>

res/res_format_attr_opus.c

@@ -33,9 +33,10 @@ ASTERISK_REGISTER_FILE()
 
 #include "asterisk/module.h"
 #include "asterisk/format.h"
-#include "asterisk/logger.h"            /* for ast_log, LOG_WARNING */
-#include "asterisk/strings.h"           /* for ast_str_append */
-#include "asterisk/utils.h"             /* for MIN, ast_malloc, ast_free */
+#include "asterisk/logger.h"
+#include "asterisk/strings.h"
+#include "asterisk/utils.h"
+#include "asterisk/opus.h"
 
 /*!
  * \brief Opus attribute structure.
@@ -43,32 +44,42 @@ ASTERISK_REGISTER_FILE()
  * \note http://tools.ietf.org/html/rfc7587#section-6
  */
 struct opus_attr {
-	unsigned int maxbitrate;
-	unsigned int maxplayrate;
-	unsigned int unused; /* was minptime, kept for binary compatibility */
-	unsigned int stereo;
-	unsigned int cbr;
-	unsigned int fec;
-	unsigned int dtx;
-	unsigned int spropmaxcapturerate;
-	unsigned int spropstereo;
+	int maxbitrate;
+	int maxplayrate;
+	int ptime;
+	int stereo;
+	int cbr;
+	int fec;
+	int dtx;
+	int spropmaxcapturerate;
+	int spropstereo;
+	int maxptime;
+	/* Note data is expected to be an ao2_object type */
+	void *data;
 };
 
 static struct opus_attr default_opus_attr = {
-	.maxplayrate         = 48000,
-	.spropmaxcapturerate = 48000,
-	.maxbitrate          = 510000,
-	.stereo              = 0,
-	.spropstereo         = 0,
-	.cbr                 = 0,
-	.fec                 = 1,
-	.dtx                 = 0,
+	.maxbitrate = CODEC_OPUS_DEFAULT_BITRATE,
+	.maxplayrate = CODEC_OPUS_DEFAULT_SAMPLE_RATE,
+	.ptime = CODEC_OPUS_DEFAULT_PTIME,
+	.stereo = CODEC_OPUS_DEFAULT_STEREO,
+	.cbr = CODEC_OPUS_DEFAULT_CBR,
+	.fec = CODEC_OPUS_DEFAULT_FEC,
+	.dtx = CODEC_OPUS_DEFAULT_DTX,
+	.spropmaxcapturerate = CODEC_OPUS_DEFAULT_SAMPLE_RATE,
+	.spropstereo = CODEC_OPUS_DEFAULT_STEREO,
+	.maxptime = CODEC_OPUS_DEFAULT_MAX_PTIME
 };
 
 static void opus_destroy(struct ast_format *format)
 {
 	struct opus_attr *attr = ast_format_get_attribute_data(format);
 
+	if (!attr) {
+		return;
+	}
+
+	ao2_cleanup(attr->data);
 	ast_free(attr);
 }
 
@@ -81,81 +92,65 @@ static int opus_clone(const struct ast_format *src, struct ast_format *dst)
 		return -1;
 	}
 
-	if (original) {
-		*attr = *original;
-	} else {
-		*attr = default_opus_attr;
-	}
+	*attr = original ? *original : default_opus_attr;
+	ao2_bump(attr->data);
 
 	ast_format_set_attribute_data(dst, attr);
 
 	return 0;
 }
 
+static void sdp_fmtp_get(const char *attributes, const char *name, int *attr)
+{
+	const char *kvp = "";
+	int val;
+
+	if (attributes && !(kvp = strstr(attributes, name))) {
+		return;
+	}
+
+	/*
+	 * If the named attribute is not at the start of the given attributes, and
+	 * the preceding character is not a space or semicolon then it's not the
+	 * attribute we are looking for. It's an attribute with the name embedded
+	 * within it (e.g. ptime in maxptime, stereo in sprop-stereo).
+	 */
+	if (kvp != attributes && *(kvp - 1) != ' ' && *(kvp - 1) != ';') {
+		/* Keep searching as it might still be in the attributes string */
+		sdp_fmtp_get(strchr(kvp, ';'), name, attr);
+	/*
+	 * Otherwise it's a match, so retrieve the value and set the attribute.
+	 */
+	} else if (sscanf(kvp, "%*[^=]=%30d", &val) == 1) {
+		*attr = val;
+	}
+}
+
 static struct ast_format *opus_parse_sdp_fmtp(const struct ast_format *format, const char *attributes)
 {
 	struct ast_format *cloned;
 	struct opus_attr *attr;
-	const char *kvp;
-	unsigned int val;
 
 	cloned = ast_format_clone(format);
 	if (!cloned) {
 		return NULL;
 	}
+
 	attr = ast_format_get_attribute_data(cloned);
 
-	if ((kvp = strstr(attributes, "maxplaybackrate")) && sscanf(kvp, "maxplaybackrate=%30u", &val) == 1) {
-		attr->maxplayrate = val;
-	} else {
-		attr->maxplayrate = 48000;
-	}
-
-	if ((kvp = strstr(attributes, "sprop-maxcapturerate")) && sscanf(kvp, "sprop-maxcapturerate=%30u", &val) == 1) {
-		attr->spropmaxcapturerate = val;
-	} else {
-		attr->spropmaxcapturerate = 48000;
-	}
-
-	if ((kvp = strstr(attributes, "maxaveragebitrate")) && sscanf(kvp, "maxaveragebitrate=%30u", &val) == 1) {
-		attr->maxbitrate = val;
-	} else {
-		attr->maxbitrate = 510000;
-	}
-
-	if (!strncmp(attributes, "stereo=1", 8)) {
-		attr->stereo = 1;
-	} else if (strstr(attributes, " stereo=1")) {
-		attr->stereo = 1;
-	} else if (strstr(attributes, ";stereo=1")) {
-		attr->stereo = 1;
-	} else {
-		attr->stereo = 0;
-	}
-
-	if (strstr(attributes, "sprop-stereo=1")) {
-		attr->spropstereo = 1;
-	} else {
-		attr->spropstereo = 0;
-	}
-
-	if (strstr(attributes, "cbr=1")) {
-		attr->cbr = 1;
-	} else {
-		attr->cbr = 0;
-	}
-
-	if (strstr(attributes, "useinbandfec=1")) {
-		attr->fec = 1;
-	} else {
-		attr->fec = 0;
-	}
-
-	if (strstr(attributes, "usedtx=1")) {
-		attr->dtx = 1;
-	} else {
-		attr->dtx = 0;
-	}
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_MAX_PLAYBACK_RATE, &attr->maxplayrate);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_MAX_CODED_AUDIO_BANDWIDTH,
+		&attr->maxplayrate);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_SPROP_MAX_CAPTURE_RATE,
+		&attr->spropmaxcapturerate);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_MAX_PTIME, &attr->maxptime);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_PTIME, &attr->ptime);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_MAX_AVERAGE_BITRATE, &attr->maxbitrate);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_STEREO, &attr->stereo);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_SPROP_STEREO, &attr->spropstereo);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_CBR, &attr->cbr);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_FEC, &attr->fec);
+	sdp_fmtp_get(attributes, CODEC_OPUS_ATTR_DTX, &attr->dtx);
 
 	return cloned;
 }
@@ -163,7 +158,7 @@ static struct ast_format *opus_parse_sdp_fmtp(const struct ast_format *format, c
 static void opus_generate_sdp_fmtp(const struct ast_format *format, unsigned int payload, struct ast_str **str)
 {
 	struct opus_attr *attr = ast_format_get_attribute_data(format);
-	int added = 0;
+	int size;
 
 	if (!attr) {
 		/*
@@ -174,79 +169,52 @@ static void opus_generate_sdp_fmtp(const struct ast_format *format, unsigned int
 		attr = &default_opus_attr;
 	}
 
-	if (48000 != attr->maxplayrate) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "maxplaybackrate=%u", attr->maxplayrate);
+	size = ast_str_append(str, 0, "a=fmtp:%u ", payload);
+
+	if (CODEC_OPUS_DEFAULT_SAMPLE_RATE != attr->maxplayrate) {
+		ast_str_append(str, 0, "%s=%d;",
+			CODEC_OPUS_ATTR_MAX_PLAYBACK_RATE, attr->maxplayrate);
 	}
 
-	if (48000 != attr->spropmaxcapturerate) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "sprop-maxcapturerate=%u", attr->spropmaxcapturerate);
+	if (CODEC_OPUS_DEFAULT_SAMPLE_RATE != attr->spropmaxcapturerate) {
+		ast_str_append(str, 0, "%s=%d;",
+			CODEC_OPUS_ATTR_SPROP_MAX_CAPTURE_RATE, attr->spropmaxcapturerate);
 	}
 
-	if (510000 != attr->maxbitrate) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "maxaveragebitrate=%u", attr->maxbitrate);
+	if (CODEC_OPUS_DEFAULT_BITRATE != attr->maxbitrate || attr->maxbitrate > 0) {
+		ast_str_append(str, 0, "%s=%d;",
+			CODEC_OPUS_ATTR_MAX_AVERAGE_BITRATE, attr->maxbitrate);
 	}
 
-	if (0 != attr->stereo) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "stereo=%u", attr->stereo);
+	if (CODEC_OPUS_DEFAULT_STEREO != attr->stereo) {
+		ast_str_append(str, 0, "%s=%d;",
+			CODEC_OPUS_ATTR_STEREO, attr->stereo);
 	}
 
-	if (0 != attr->spropstereo) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "sprop-stereo=%u", attr->spropstereo);
+	if (CODEC_OPUS_DEFAULT_STEREO != attr->spropstereo) {
+		ast_str_append(str, 0, "%s=%d;",
+			CODEC_OPUS_ATTR_SPROP_STEREO, attr->spropstereo);
 	}
 
-	if (0 != attr->cbr) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "cbr=%u", attr->cbr);
+	if (CODEC_OPUS_DEFAULT_CBR != attr->cbr) {
+		ast_str_append(str, 0, "%s=%d;",
+			CODEC_OPUS_ATTR_CBR, attr->cbr);
 	}
 
-	if (0 != attr->fec) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "useinbandfec=%u", attr->fec);
+	if (CODEC_OPUS_DEFAULT_FEC!= attr->fec) {
+		ast_str_append(str, 0, "%s=%d;",
+		       CODEC_OPUS_ATTR_FEC, attr->fec);
 	}
 
-	if (0 != attr->dtx) {
-		if (added) {
-			ast_str_append(str, 0, ";");
-		} else if (0 < ast_str_append(str, 0, "a=fmtp:%u ", payload)) {
-			added = 1;
-		}
-		ast_str_append(str, 0, "usedtx=%u", attr->dtx);
+	if (CODEC_OPUS_DEFAULT_DTX != attr->dtx) {
+		ast_str_append(str, 0, "%s=%d;",
+			CODEC_OPUS_ATTR_DTX, attr->dtx);
 	}
 
-	if (added) {
+	if (size == ast_str_strlen(*str)) {
+		ast_str_reset(*str);
+	} else {
+		ast_str_truncate(*str, -1);
 		ast_str_append(str, 0, "\r\n");
 	}
 }
@@ -285,49 +253,68 @@ static struct ast_format *opus_getjoint(const struct ast_format *format1, const
 	 * to receive stereo signals, it may be a waste of bandwidth. */
 	attr_res->stereo = attr1->stereo && attr2->stereo ? 1 : 0;
 
+	if (attr1->maxbitrate < 0) {
+		attr_res->maxbitrate = attr2->maxbitrate;
+	} else if (attr2->maxbitrate < 0) {
+		attr_res->maxbitrate = attr1->maxbitrate;
+	} else {
 		attr_res->maxbitrate = MIN(attr1->maxbitrate, attr2->maxbitrate);
+	}
+
 	attr_res->spropmaxcapturerate = MIN(attr1->spropmaxcapturerate, attr2->spropmaxcapturerate);
 	attr_res->maxplayrate = MIN(attr1->maxplayrate, attr2->maxplayrate);
 
 	return jointformat;
 }
 
-static struct ast_format *opus_set(const struct ast_format *format, const char *name, const char *value)
+static struct ast_format *opus_set(const struct ast_format *format,
+	const char *name, const char *value)
 {
 	struct ast_format *cloned;
 	struct opus_attr *attr;
-	unsigned int val;
+	int val;
 
-	if (sscanf(value, "%30u", &val) != 1) {
-		ast_log(LOG_WARNING, "Unknown value '%s' for attribute type '%s'\n",
-			value, name);
+	if (!(cloned = ast_format_clone(format))) {
 		return NULL;
 	}
 
-	cloned = ast_format_clone(format);
-	if (!cloned) {
-		return NULL;
-	}
 	attr = ast_format_get_attribute_data(cloned);
 
-	if (!strcasecmp(name, "max_bitrate")) {
-		attr->maxbitrate = val;
-	} else if (!strcasecmp(name, "max_playrate")) {
+	if (!strcmp(name, CODEC_OPUS_ATTR_DATA)) {
+		ao2_cleanup(attr->data);
+		attr->data = ao2_bump((void*)value);
+		return cloned;
+	}
+
+	if (sscanf(value, "%30d", &val) != 1) {
+		ast_log(LOG_WARNING, "Unknown value '%s' for attribute type '%s'\n",
+			value, name);
+		ao2_ref(cloned, -1);
+		return NULL;
+	}
+
+	if (!strcasecmp(name, CODEC_OPUS_ATTR_MAX_PLAYBACK_RATE)) {
 		attr->maxplayrate = val;
-	} else if (!strcasecmp(name, "minptime")) {
-		attr->unused = val;
-	} else if (!strcasecmp(name, "stereo")) {
-		attr->stereo = val;
-	} else if (!strcasecmp(name, "cbr")) {
-		attr->cbr = val;
-	} else if (!strcasecmp(name, "fec")) {
-		attr->fec = val;
-	} else if (!strcasecmp(name, "dtx")) {
-		attr->dtx = val;
-	} else if (!strcasecmp(name, "sprop_capture_rate")) {
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_MAX_CODED_AUDIO_BANDWIDTH)) {
+		attr->maxplayrate = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_SPROP_MAX_CAPTURE_RATE)) {
 		attr->spropmaxcapturerate = val;
-	} else if (!strcasecmp(name, "sprop_stereo")) {
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_MAX_PTIME)) {
+		attr->maxptime = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_PTIME)) {
+		attr->ptime = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_MAX_AVERAGE_BITRATE)) {
+		attr->maxbitrate = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_STEREO)) {
+		attr->stereo = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_SPROP_STEREO)) {
 		attr->spropstereo = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_CBR)) {
+		attr->cbr = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_FEC)) {
+		attr->fec = val;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_DTX)) {
+		attr->dtx = val;
 	} else {
 		ast_log(LOG_WARNING, "unknown attribute type %s\n", name);
 	}
@@ -335,6 +322,44 @@ static struct ast_format *opus_set(const struct ast_format *format, const char *
 	return cloned;
 }
 
+static const void *opus_get(const struct ast_format *format, const char *name)
+{
+	struct opus_attr *attr = ast_format_get_attribute_data(format);
+	int *val = NULL;
+
+	if (!attr) {
+		return NULL;
+	}
+
+	if (!strcasecmp(name, CODEC_OPUS_ATTR_DATA)) {
+		return ao2_bump(attr->data);
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_MAX_PLAYBACK_RATE)) {
+		val = &attr->maxplayrate;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_SPROP_MAX_CAPTURE_RATE)) {
+		val = &attr->spropmaxcapturerate;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_MAX_PTIME)) {
+		val = &attr->maxptime;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_PTIME)) {
+		val = &attr->ptime;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_MAX_AVERAGE_BITRATE)) {
+		val = &attr->maxbitrate;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_STEREO)) {
+		val = &attr->stereo;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_SPROP_STEREO)) {
+		val = &attr->spropstereo;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_CBR)) {
+		val = &attr->cbr;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_FEC)) {
+		val = &attr->fec;
+	} else if (!strcasecmp(name, CODEC_OPUS_ATTR_DTX)) {
+		val = &attr->dtx;
+	} else {
+		ast_log(LOG_WARNING, "unknown attribute type %s\n", name);
+	}
+
+	return val;
+}
+
 static struct ast_format_interface opus_interface = {
 	.format_destroy = opus_destroy,
 	.format_clone = opus_clone,
@@ -342,11 +367,12 @@ static struct ast_format_interface opus_interface = {
 	.format_attribute_set = opus_set,
 	.format_parse_sdp_fmtp = opus_parse_sdp_fmtp,
 	.format_generate_sdp_fmtp = opus_generate_sdp_fmtp,
+	.format_attribute_get = opus_get
 };
 
 static int load_module(void)
 {
-	if (ast_format_interface_register("opus", &opus_interface)) {
+	if (__ast_format_interface_register("opus", &opus_interface, ast_module_info->self)) {
 		return AST_MODULE_LOAD_DECLINE;
 	}
 
@@ -358,9 +384,9 @@ static int unload_module(void)
 	return 0;
 }
 
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Opus Format Attribute Module",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Opus Format Attribute Module",
 	.support_level = AST_MODULE_SUPPORT_CORE,
 	.load = load_module,
 	.unload = unload_module,
-	.load_pri = AST_MODPRI_CHANNEL_DEPEND,
+	.load_pri = AST_MODPRI_REALTIME_DRIVER /* Needs to load before codec_opus */
 );