Update for 18.17.1

There are two main parts of the change associated with this
commit. These are driven by the change in call order of
pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject
when an in-dialog SUBSCRIBE is received.

First, the previous behavior was for pjproject to call
pubsub_on_rx_refresh before calling pubsub_on_evsub_state
when an in-dialog SUBSCRIBE was received that changes the
subscription state.

If that change was a termination due to a re-SUBSCRIBE with
an expires of 0, we used to use the call to pubsub_on_rx_refresh
to set the substate of the evsub to TERMINATE_PENDING before
pjproject could call pubsub_on_evsub_state.

This substate let pubsub_on_evsub_state know that the
subscription TERMINATED event could be ignored as there was
still a subsequent NOTIFY that needed to be generated and
another call to pubsub_on_evsub_state to come with it.

That NOTIFY was sent via serialized_pubsub_on_refresh_timeout
which would see the TERMINATE_PENDING state and transition it
to TERMINATE_IN_PROGRESS before triggering another call to
pubsub_on_evsub_state (which now would clean up the evsub.)

The new pjproject behavior is to call pubsub_on_evsub_state
before pubsub_on_rx_refresh. This means we no longer can set
the state to TERMINATE_PENDING to tell pubsub_on_evsub_state
that it can ignore the first TERMINATED event.

To handle this, we now look directly at the event type,
method type and the expires value to determine whether we
want to ignore the event or use it to trigger the evsub
cleanup.

Second, pjproject now expects the NOTIFY to actually be sent
during pubsub_on_rx_refresh and avoids the protocol violation
inherent in sending a NOTIFY before the SUBSCRIBE is
acknowledged by caching the sent NOTIFY then sending it
after responding to the SUBSCRIBE.

This requires we send the NOTIFY using the non-serialized
pubsub_on_refresh_timeout directly and let pjproject handle
the protocol violation.

ASTERISK-30469

Change-Id: I05c1d91a44fe28244ae93faa4a2268a3332b5fd7

.gitignore

@@ -37,4 +37,4 @@ doxygen.log
 out/
 *.orig
 tests/CI/output
-
+.develvars

.gitreview

@@ -1,5 +1,6 @@
 [gerrit]
-defaultbranch=master
+defaultbranch=18
+basebranch=18
 #
 # Intentional padding to ensure it is possible to point a commit
 # to an alternative gerrit server/repository without breaking

.lastclean

@@ -0,0 +1 @@
+40

.version

@@ -0,0 +1 @@
+18.17.1

CREDITS

@@ -40,7 +40,7 @@
 		rewrite of SIP transfers
 
 
-=== WISHLIST CONTRIBUTERS ===
+=== WISHLIST CONTRIBUTORS ===
 
  We'd like to thank the following for contributing to our wishlist
 

LICENSE

@@ -45,7 +45,7 @@ redistribution of Asterisk source code obtained from Digium, you
 should contact our licensing department to determine the necessary
 steps you must take. For more information on this policy, please read:
 
-http://www.digium.com/en/company/profile/trademarkpolicy.php
+https://www.sangoma.com/wp-content/uploads/Sangoma-Trademark-Policy.pdf
 
 If you have any questions regarding our licensing policy, please
 contact us:
@@ -53,7 +53,6 @@ contact us:
 +1.877.344.4861 (via telephone in the USA)
 +1.256.428.6000 (via telephone outside the USA)
 +1.256.864.0464 (via FAX inside or outside the USA)
-IAX2/pbx.digium.com (via IAX2)
 licensing@digium.com (via email)
 
 Digium, Inc.

Makefile

@@ -21,7 +21,7 @@
 #      on a single object just for that object
 # SOLINK - linker flags used only for creating dynamically loadable modules
 #          as .so files
-# DYLINK - linker flags used only for creating shared libaries
+# DYLINK - linker flags used only for creating shared libraries
 #          (.so files on Unix-type platforms, .dylib on Darwin)
 #
 # Values for ASTCFLAGS and ASTLDFLAGS can be specified in the
@@ -49,6 +49,7 @@ DESTDIR?=$(INSTALL_PATH)
 export DESTDIR
 
 export INSTALL_PATH       # Additional prefix for the following paths
+export ASTCACHEDIR
 export ASTETCDIR          # Path for config files
 export ASTVARRUNDIR
 export ASTSPOOLDIR
@@ -100,6 +101,11 @@ export TAR
 export PATCH
 export SED
 export NM
+export FIND
+export BASENAME
+export DIRNAME
+export XMLLINT
+export XMLSTARLET
 
 # makeopts is required unless the goal is just {dist{-}}clean
 ifeq ($(MAKECMDGOALS),clean)
@@ -134,7 +140,7 @@ empty:=
 space:=$(empty) $(empty)
 ASTTOPDIR:=$(subst $(space),\$(space),$(CURDIR))
 
-# Overwite config files on "make samples" or other config installation targets
+# Overwrite config files on "make samples" or other config installation targets
 OVERWRITE=y
 
 # Include debug and macro symbols in the executables (-g) and profiling info (-pg)
@@ -249,7 +255,7 @@ else
 endif
 ifneq ($(AWK),)
   ifneq ($(wildcard .version),)
-    ASTERISKVERSIONNUM:=$(shell $(AWK) -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}' .version)
+    ASTERISKVERSIONNUM:=$(shell $(SED) -e 's/^certified\///' -e 's/-cert/./' .version | $(AWK) -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}')
   endif
 endif
 
@@ -321,6 +327,9 @@ else
 SUBMAKE:=$(MAKE) --quiet --no-print-directory
 endif
 
+mkfile_path := $(abspath $(lastword $(MAKEFILE_LIST)))
+mkfile_dir := $(dir $(mkfile_path))
+
 # $(MAKE) is printed in several places, and we want it to be a
 # fixed size string. Define a variable whose name has also the
 # same size, so we can easily align text.
@@ -441,7 +450,7 @@ distclean: $(SUBDIRS_DIST_CLEAN) _clean
 	rm -f include/asterisk/autoconfig.h
 	rm -f include/asterisk/buildopts.h
 	rm -rf doc/api
-	rm -f doc/asterisk-ng-doxygen
+	rm -f doc/Doxyfile
 	rm -f build_tools/menuselect-deps
 
 datafiles: _all $(CORE_XMLDOC)
@@ -474,53 +483,24 @@ endif
 		$(INSTALL) -m 644 $$x "$(DESTDIR)$(ASTDATADIR)/rest-api" ; \
 	done
 
-ifeq ($(GREP),)
-else ifeq ($(GREP),:)
-else
-  XML_core_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
-endif
-
+DOC_MOD_SUBDIRS := $(filter-out third-party,$(MOD_SUBDIRS))
+XML_core_en_US := $(shell build_tools/make_xml_documentation --command=print_dependencies --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)")
+# core-en_US.xml is the normal documentation created with asterisk builds.
 doc/core-en_US.xml: makeopts .lastclean $(XML_core_en_US)
-	@printf "Building Documentation For: "
-	@echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
-	@echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
-	@echo "<?xml-stylesheet type=\"text/xsl\" href=\"appdocsxml.xslt\"?>" >> $@
-	@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
-	@for x in $(MOD_SUBDIRS); do \
-		printf "$$x " ; \
-		for i in `find $$x -name '*.c'`; do \
-			$(AWK) -f build_tools/get_documentation $$i >> $@ ; \
-		done ; \
-	done
-	@echo
-	@echo "</docs>" >> $@
+	@build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \
+		--with-moduleinfo --output-file=$@
 
-ifeq ($(GREP),)
-else ifeq ($(GREP),:)
-else
-  XML_full_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
-endif
-
-doc/full-en_US.xml: makeopts .lastclean $(XML_full_en_US)
+# The full-en_US.xml target is only called by the wiki documentation generation process
+# and does special post-processing in preparation for uploading to the wiki.
+# It creates full-en_US.xml but then re-creates core-en_US.xml as well.
+doc/full-en_US.xml: makeopts .lastclean $(XML_core_en_US)
 ifeq ($(PYTHON),:)
 	@echo "--------------------------------------------------------------------------"
 	@echo "---        Please install python to build full documentation           ---"
 	@echo "--------------------------------------------------------------------------"
 else
-	@printf "Building Documentation For: "
-	@echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
-	@echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
-	@echo "<?xml-stylesheet type=\"text/xsl\" href=\"appdocsxml.xslt\"?>" >> $@
-	@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
-	@for x in $(filter-out third-party,$(MOD_SUBDIRS)); do \
-		printf "$$x " ; \
-		for i in `find $$x -name '*.c'`; do \
-			$(PYTHON) build_tools/get_documentation.py < $$i >> $@ ; \
-		done ; \
-	done
-	@echo
-	@echo "</docs>" >> $@
-	@$(PYTHON) build_tools/post_process_documentation.py -i $@ -o "doc/core-en_US.xml"
+	@build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \
+		--for-wiki --output-file=$@ --core-output-file=./doc/core-en_US.xml
 endif
 
 validate-docs: doc/core-en_US.xml
@@ -555,7 +535,7 @@ update:
 
 NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
 OLDHEADERS=$(filter-out $(NEWHEADERS) $(notdir $(DESTDIR)$(ASTHEADERDIR)),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
-INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTETCDIR)" "$(ASTVARRUNDIR)" \
+INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTCACHEDIR)" "$(ASTETCDIR)" "$(ASTVARRUNDIR)" \
 	"$(ASTSPOOLDIR)" "$(ASTSPOOLDIR)/dictate" "$(ASTSPOOLDIR)/meetme" \
 	"$(ASTSPOOLDIR)/monitor" "$(ASTSPOOLDIR)/system" "$(ASTSPOOLDIR)/tmp" \
 	"$(ASTSPOOLDIR)/voicemail" "$(ASTSPOOLDIR)/recording" \
@@ -581,9 +561,9 @@ bininstall: _all installdirs $(SUBDIRS_INSTALL) main-bininstall
 	$(INSTALL) -m 755 contrib/scripts/astversion "$(DESTDIR)$(ASTSBINDIR)/"
 	$(INSTALL) -m 755 contrib/scripts/astgenkey "$(DESTDIR)$(ASTSBINDIR)/"
 	$(INSTALL) -m 755 contrib/scripts/autosupport "$(DESTDIR)$(ASTSBINDIR)/"
-	if [ ! -f /sbin/launchd ]; then \
-		./build_tools/install_subst contrib/scripts/safe_asterisk "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk"; \
-	fi
+ifneq ($(HAVE_SBIN_LAUNCHD),1)
+	./build_tools/install_subst contrib/scripts/safe_asterisk "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk";
+endif
 
 ifneq ($(DISABLE_XMLDOC),yes)
 	$(INSTALL) -m 644 doc/core-*.xml "$(DESTDIR)$(ASTDATADIR)/documentation"
@@ -716,7 +696,17 @@ ifneq ($(filter ~%,$(DESTDIR)),)
 	@exit 1
 endif
 
-install: badshell bininstall datafiles
+versioncheck:
+ifeq ($(ASTERISKVERSION),UNKNOWN__git_check_fail)
+	@echo "Asterisk Version is unknown due to a git error. If you are running make"
+	@echo "as a different user than the project owner, this can be resolved by"
+	@echo "running the following command as the user currently executing make: "$$USER
+	@echo "git config --global --add safe.directory "$(mkfile_dir:/=)
+	@exit 1
+endif
+
+
+install: badshell versioncheck bininstall datafiles
 	@if [ -x /usr/sbin/asterisk-post-install ]; then \
 		/usr/sbin/asterisk-post-install "$(DESTDIR)" . ; \
 	fi
@@ -779,7 +769,8 @@ define INSTALL_CONFIGS
 	done ; \
 	if [ "$(OVERWRITE)" = "y" ]; then \
 		echo "Updating asterisk.conf" ; \
-		sed -e 's|^astetcdir.*$$|astetcdir => $(ASTETCDIR)|' \
+		sed -e 's|^astcachedir.*$$|astcachedir => $(ASTCACHEDIR)|' \
+			-e 's|^astetcdir.*$$|astetcdir => $(ASTETCDIR)|' \
 			-e 's|^astmoddir.*$$|astmoddir => $(ASTMODDIR)|' \
 			-e 's|^astvarlibdir.*$$|astvarlibdir => $(ASTVARLIBDIR)|' \
 			-e 's|^astdbdir.*$$|astdbdir => $(ASTDBDIR)|' \
@@ -882,29 +873,36 @@ webvmail:
 	@echo " +-------------------------------------------+"
 
 progdocs:
-# Note, Makefile conditionals must not be tabbed out. Wasted hours with that.
-	@cp doc/asterisk-ng-doxygen.in doc/asterisk-ng-doxygen
 ifeq ($(DOXYGEN),:)
 	@echo "Doxygen is not installed.  Please install and re-run the configuration script."
 else
+	@cp doc/Doxyfile.in doc/Doxyfile
 ifeq ($(DOT),:)
 	@echo "DOT is not installed. Doxygen will not produce any diagrams. Please install and re-run the configuration script."
 else
-	# Enable DOT
-	@echo "HAVE_DOT = YES" >> doc/asterisk-ng-doxygen
+	@echo "HAVE_DOT = YES" >> doc/Doxyfile
 endif
-	# Set Doxygen PROJECT_NUMBER variable
-ifneq ($(ASTERISKVERSION),UNKNOWN__and_probably_unsupported)
-	@echo "PROJECT_NUMBER = $(ASTERISKVERSION)" >> doc/asterisk-ng-doxygen
+ifneq ($(NOISY_BUILD),yes)
+	@echo "EXTRACT_ALL = YES" >> doc/Doxyfile
+endif
+ifeq ($(AST_DEVMODE),yes)
+	@echo "INTERNAL_DOCS = YES" >> doc/Doxyfile
+	@echo "WARN_NO_PARAMDOC = YES" >> doc/Doxyfile
+endif
+ifeq ($(ASTERISKVERSION),UNKNOWN__and_probably_unsupported)
+	@echo "Asterisk Version is unknown, not configuring Doxygen PROJECT_NUMBER."
+else ifeq ($(ASTERISKVERSION),UNKNOWN__git_check_fail)
+	@echo "Asterisk Version is unknown due to a git error. If you are running make"
+	@echo "as a different user than the project owner, this can be resolved by"
+	@echo "running the following command as the user currently executing make: "$$USER
+	@echo "git config --global --add safe.directory "$(mkfile_dir:/=)
+	@echo "not configuring Doxygen PROJECT_NUMBER."
 else
-	echo "Asterisk Version is unknown, not configuring Doxygen PROJECT_NUMBER."
+	@echo "PROJECT_NUMBER = $(ASTERISKVERSION)" >> doc/Doxyfile
 endif
-	# Validate and auto-update local copy
-	@doxygen -u doc/asterisk-ng-doxygen
-	# Run Doxygen
-	@doxygen doc/asterisk-ng-doxygen
-	# Remove configuration backup file
-	@rm -f doc/asterisk-ng-doxygen.bak
+	@echo "Generating C-API documentation. This will take a while."
+	@doxygen doc/Doxyfile
+	@echo "Generation complete. Any warnings are in ./doxygen.log."
 endif
 
 install-logrotate:
@@ -989,6 +987,7 @@ sounds:
 	@$(MAKE) clean
 	@[ -f "$(DESTDIR)$(ASTDBDIR)/astdb.sqlite3" ] || [ ! -f "$(DESTDIR)$(ASTDBDIR)/astdb" ] || [ ! -f menuselect.makeopts ] || grep -q MENUSELECT_UTILS=.*astdb2sqlite3 menuselect.makeopts || (sed -i.orig -e's/MENUSELECT_UTILS=\(.*\)/MENUSELECT_UTILS=\1 astdb2sqlite3/' menuselect.makeopts && echo "Updating menuselect.makeopts to include astdb2sqlite3" && echo "Original version backed up to menuselect.makeopts.orig")
 
+
 $(SUBDIRS_UNINSTALL):
 	+@DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" ASTDATADIR="$(ASTDATADIR)" $(SUBMAKE) -C $(@:-uninstall=) uninstall
 
@@ -1144,6 +1143,7 @@ check-alembic: makeopts
 .PHONY: uninstall-headers
 .PHONY: badshell
 .PHONY: installdirs
+.PHONY: progdocs
 .PHONY: validate-docs
 .PHONY: _clean
 .PHONY: ari-stubs

Makefile.rules

@@ -204,4 +204,19 @@ endif
 	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
 	$(CMD_PREFIX) $(CXX) -o $@ $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $^ $(CXX_LIBS) $(ASTLDFLAGS)
 
+# These CC commands just create an object file with the input file embedded in it.
+# It can be access from code as follows:
+# If your input file is named abc_def.xml...
+#
+# extern const uint8_t _binary_abc_def_xml_start[];
+# extern const uint8_t _binary_abc_def_xml_end[];
+# extern const size_t _binary_abc_def_xml_size;
+%.o: %.xml
+	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
+	$(CMD_PREFIX) $(CC) -g -Wl,-znoexecstack -nostartfiles  -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
+
+%.o: %.xslt
+	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
+	$(CMD_PREFIX) $(CC) -g -Wl,-znoexecstack -nostartfiles  -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
+
 dist-clean:: clean

README.md

@@ -1,7 +1,7 @@
 # The Asterisk(R) Open Source PBX
 ```text
         By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
-        Copyright (C) 2001-2019 Digium, Inc. and other copyright holders.
+        Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.
 ```
 ## SECURITY
 
@@ -47,7 +47,7 @@ and the BSD variants.
 ANY special hardware, not even a sound card) to install and run Asterisk.
 
 Supported telephony hardware includes:
-* All Analog and Digital Interface cards from [Digium]
+* All Analog and Digital Interface cards from [Sangoma]
 * QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
 * any full duplex sound card supported by ALSA, OSS, or PortAudio
 * any ISDN card supported by mISDN on Linux
@@ -91,7 +91,10 @@ guides in the [configs] directory.
 2. Run `./configure`
 
   Execute the configure script to guess values for system-dependent
-variables used during compilation.
+variables used during compilation. If the script indicates that some required 
+components are missing, you can run `./contrib/scripts/install_prereq install`
+to install the necessary components. Note that this will install all dependencies for every functionality of Asterisk. After running the script, you will need
+to rerun `./configure`.
 
 3. Run `make menuselect` _\[optional]_
 
@@ -251,7 +254,7 @@ Welcome to the growing worldwide community of Asterisk users!
 
 ---
 
-Asterisk is a trademark of Digium, Inc.
+Asterisk is a trademark of Sangoma Technologies Corporation
 
 [home page]: https://www.asterisk.org
 [support]: https://www.asterisk.org/support
@@ -261,7 +264,7 @@ Asterisk is a trademark of Digium, Inc.
 [voip-info.org]: http://www.voip-info.org/wiki-Asterisk
 [asteriskdocs.org]: http://www.asteriskdocs.org
 [NTP]: http://www.ntp.org/
-[Digium]: https://www.digium.com/
+[Sangoma]: https://www.sangoma.com/
 [UPGRADE.txt]: UPGRADE.txt
 [CHANGES]: CHANGES
 [configs]: configs

UPGRADE.txt

@@ -18,6 +18,141 @@
 ===
 ===========================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.16.0 to Asterisk 18.17.0 ----------
+------------------------------------------------------------------------------
+
+app_playback
+------------------
+ * In Asterisk 11, if a channel was redirected away during Playback(),
+   the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
+   (specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
+   behavior was inadvertently changed and the same operation would result
+   in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
+   behavior has been restored.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.15.0 to Asterisk 18.16.0 ----------
+------------------------------------------------------------------------------
+
+AMI (Asterisk Manager Interface)
+------------------
+ * Previously, GetConfig and UpdateConfig were able to access files outside of
+   the Asterisk configuration directory. Now this access is put behind the
+   live_dangerously configuration option in asterisk.conf, which is disabled by
+   default. If access to configuration files outside of the Asterisk configuation
+   directory is required via AMI, then the live_dangerously configuration option
+   must be set to yes.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.14.0 to Asterisk 18.15.0 ----------
+------------------------------------------------------------------------------
+
+res_crypto
+------------------
+ * In addition to only paying attention to files ending with .key or .pub
+   in the keys directory, we now also ignore any files which aren't regular
+   files.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.11.3 to Asterisk 18.12.0 ----------
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * The 'async_operations' setting on transports is no longer
+   obeyed and instead is always set to 1. This is due to the
+   functionality not being applicable to Asterisk and causing
+   excess unnecessary memory usage. This setting will now be
+   ignored but can also be removed from the configuration file.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.10.0 to Asterisk 18.11.0 ----------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * The XML Manager Event Interface (amxml) now generates attribute names
+   that are compliant with the XML 1.1 specification. Previously, an
+   attribute name that started with a digit would be rendered as-is, even
+   though attribute names must not begin with a digit. We now prefix
+   attribute names that start with a digit with an underscore ('_') to
+   prevent XML validation failures.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.8.0 to Asterisk 18.9.0 ------------
+------------------------------------------------------------------------------
+
+STIR/SHAKEN
+------------------
+ * The STIR/SHAKEN configuration option has been split into
+   4 different choices: off, attest, verify, and on. Off and
+   on behave the same way as before. Attest will only perform
+   attestation on the endpoint, and verify will only perform
+   verification on the endpoint.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.7.0 to Asterisk 18.8.0 ------------
+------------------------------------------------------------------------------
+
+chan_iax2
+------------------
+ * Encryption is now supported for RSA authentication.
+
+   Currently, these auth configurations will cause a crash:
+   auth = md5,rsa
+   auth = plaintext,md5,rsa
+
+   With a patched peer, the following will cause a crash:
+   auth = rsa
+   auth = md5,rsa
+   auth = plaintext,md5,rsa
+
+   If both the peer and user are patches, no crash occurs.
+   Existing good configurations should continue to work.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.4.0 to Asterisk 18.5.0 ------------
+------------------------------------------------------------------------------
+
+STIR/SHAKEN
+------------------
+ * The configuration option public_key_url in stir_shaken.conf
+   has been renamed to public_cert_url to better fit what it
+   contains. Only the name has changed - functionality is the
+   same.
+
+ * STIR/SHAKEN originally needed an origid to be specified in
+   stir_shaken.conf under the certificate config object in
+   order to work. Now, one is automatically created by
+   generating a UUID, as recommended by RFC8588. Any origid
+   you have in your stir_shaken.conf will need to be removed
+   for the module to read in certificates.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.3.0 to Asterisk 18.4.0 ------------
+------------------------------------------------------------------------------
+
+menuselect
+------------------
+ * menuselect --enable, --disable, --enable-category and --disable-category will
+   now fail with a non-zero exit code instead of silently failing if an invalid
+   option or category is specified.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
+------------------------------------------------------------------------------
+
+res_srtp
+------------------
+ * SRTP replay protection has been added to res_srtp and
+   a new configuration option "srtpreplayprotection" has
+   been added to the rtp.conf config file.  For security
+   reasons, the default setting is "yes".  Buggy clients
+   may not handle this correctly which could result in
+   no, or one way, audio and Asterisk error messages like
+   "replay check failed".
+
 ------------------------------------------------------------------------------
 --- New functionality introduced in Asterisk 18.0.0 --------------------------
 ------------------------------------------------------------------------------
@@ -610,7 +745,7 @@ chan_unistim:
    values used inside Unistim protocol
 
  - Added 'dtmf_duration' option with changing default operation to disable
-   receivied dtmf playback on unistim phone
+   received dtmf playback on unistim phone
 
 Core:
 
@@ -1528,7 +1663,7 @@ UDPTL:
 From 10.4 to 10.5:
 
 * The complex processor detection and optimization has been removed from
-  the makefile in favor of using native optimization suppport when available.
+  the makefile in favor of using native optimization support when available.
   BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
 
 From 10.2 to 10.3:
@@ -1601,7 +1736,7 @@ From 1.8.13 to 1.8.14:
 
 From 1.8.12 to 1.8.13:
 * The complex processor detection and optimization has been removed from
-  the makefile in favor of using native optimization suppport when available.
+  the makefile in favor of using native optimization support when available.
   BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
 
 From 1.8.10 to 1.8.11:
@@ -2380,7 +2515,7 @@ Applications:
   the 'm' option now provides the functionality of "initially muted".
   In practice, most existing dialplans using the 'm' flag should not notice
   any difference, unless the keypad menu is enabled, allowing the user
-  to unmute themsleves.
+  to unmute themselves.
 
 * ast_play_and_record would attempt to cancel the recording if a DTMF
   '0' was received.  This behavior was not documented in most of the

addons/Makefile

@@ -64,6 +64,10 @@ chan_ooh323.so: _ASTCFLAGS+=$(H323CFLAGS)
 $(call MOD_ADD_C,chan_ooh323,$(H323SOURCE))
 
 ifneq ($(wildcard mp3/Makefile),)
+# At the current time, the fate of mp3 is in flux so it didn't make sense to
+# add configure/makeopts processing for array-bounds since this is the only
+# source file that needs that warning suppressed.
+mp3/layer3.o: _ASTCFLAGS+=-Wno-array-bounds
 $(call MOD_ADD_C,format_mp3,mp3/common.c mp3/dct64_i386.c mp3/decode_ntom.c mp3/layer3.c mp3/tabinit.c mp3/interface.c)
 
 .PHONY: check_mp3

addons/app_mysql.c

@@ -37,6 +37,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>func_odbc</replacement>
+	<deprecated_in>1.8</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

addons/cdr_mysql.c

@@ -38,6 +38,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>cdr_adaptive_odbc</replacement>
+	<deprecated_in>1.8</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

addons/chan_mobile.c

@@ -102,7 +102,7 @@ struct adapter_pvt {
 	char id[31];					/* the 'name' from mobile.conf */
 	bdaddr_t addr;					/* adddress of adapter */
 	unsigned int inuse:1;				/* are we in use ? */
-	unsigned int alignment_detection:1;		/* do alignment detection on this adpater? */
+	unsigned int alignment_detection:1;		/* do alignment detection on this adapter? */
 	struct io_context *io;				/*!< io context for audio connections */
 	struct io_context *accept_io;			/*!< io context for sco listener */
 	int *sco_id;					/*!< the io context id of the sco listener socket */
@@ -2006,7 +2006,7 @@ static int at_match_prefix(char *buf, char *prefix)
 }
 
 /*!
- * \brief Read an AT message and clasify it.
+ * \brief Read an AT message and classify it.
  * \param rsock an rfcomm socket
  * \param buf the buffer to store the result in
  * \param count the size of the buffer or the maximum number of characters to read
@@ -2156,10 +2156,12 @@ static inline const char *at_msg2str(at_message_t msg)
  * \param buf the buffer to parse (null terminated)
  * \return -1 on error (parse error) or a ECAM value on success
  *
- * Example string: *ECAV: <ccid>,<ccstatus>,<calltype>[,<processid>]
- * [,exitcause][,<number>,<type>]
+ * Example:
+ * \verbatim *ECAV: <ccid>,<ccstatus>,<calltype>[,<processid>]
+                    [,exitcause][,<number>,<type>] \endverbatim
  *
- * Example indicating busy: *ECAV: 1,7,1
+ * Example indicating busy:
+ * \verbatim *ECAV: 1,7,1 \endverbatim
  */
 static int hfp_parse_ecav(struct hfp_pvt *hfp, char *buf)
 {
@@ -2176,7 +2178,7 @@ static int hfp_parse_ecav(struct hfp_pvt *hfp, char *buf)
 }
 
 /*!
- * \brief Enable Sony Erricson extensions / indications.
+ * \brief Enable Sony Ericsson extensions / indications.
  * \param hfp an hfp_pvt struct
  */
 static int hfp_send_ecam(struct hfp_pvt *hfp)
@@ -3040,7 +3042,7 @@ static void msg_queue_free_and_pop(struct mbl_pvt *pvt)
 }
 
 /*!
- * \brief Remove all itmes from the queue and free them.
+ * \brief Remove all items from the queue and free them.
  * \param pvt a mbl_pvt structure
  */
 static void msg_queue_flush(struct mbl_pvt *pvt)
@@ -3347,7 +3349,7 @@ static int handle_response_ok(struct mbl_pvt *pvt, char *buf)
 			}
 			break;
 		case AT_CLIP:
-			ast_debug(1, "[%s] caling line indication enabled\n", pvt->id);
+			ast_debug(1, "[%s] calling line indication enabled\n", pvt->id);
 			if (hfp_send_ecam(pvt->hfp) || msg_queue_push(pvt, AT_OK, AT_ECAM)) {
 				ast_debug(1, "[%s] error enabling Sony Ericsson call monitoring extensions\n", pvt->id);
 				goto e_return;
@@ -3567,7 +3569,7 @@ static int handle_response_ciev(struct mbl_pvt *pvt, char *buf)
 			if (pvt->owner) {
 				ast_debug(1, "[%s] hanging up owner\n", pvt->id);
 				if (mbl_queue_hangup(pvt)) {
-					ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnectiong...\n", pvt->id);
+					ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnecting...\n", pvt->id);
 					return -1;
 				}
 			}
@@ -3601,7 +3603,7 @@ static int handle_response_ciev(struct mbl_pvt *pvt, char *buf)
 						handle_response_busy(pvt);
 					}
 					if (mbl_queue_hangup(pvt)) {
-						ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnectiong...\n", pvt->id);
+						ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnecting...\n", pvt->id);
 						return -1;
 					}
 				}
@@ -4431,7 +4433,7 @@ static struct adapter_pvt *mbl_load_adapter(struct ast_config *cfg, const char *
 
 	/* bind the sco listener socket */
 	if (sco_bind(adapter) < 0) {
-		ast_log(LOG_ERROR, "Skipping adapter %s. Error binding audio connection listerner socket.\n", adapter->id);
+		ast_log(LOG_ERROR, "Skipping adapter %s. Error binding audio connection listener socket.\n", adapter->id);
 		goto e_destroy_io;
 	}
 
@@ -4443,7 +4445,7 @@ static struct adapter_pvt *mbl_load_adapter(struct ast_config *cfg, const char *
 
 	/* start the sco listener for this adapter */
 	if (ast_pthread_create_background(&adapter->sco_listener_thread, NULL, do_sco_listen, adapter)) {
-		ast_log(LOG_ERROR, "Skipping adapter %s. Error creating audio connection listerner thread.\n", adapter->id);
+		ast_log(LOG_ERROR, "Skipping adapter %s. Error creating audio connection listener thread.\n", adapter->id);
 		goto e_remove_sco;
 	}
 
@@ -4499,7 +4501,7 @@ static struct mbl_pvt *mbl_load_device(struct ast_config *cfg, const char *cat)
 	}
 	AST_RWLIST_UNLOCK(&adapters);
 	if (!adapter) {
-		ast_log(LOG_ERROR, "Skiping device %s. Unknown adapter '%s' specified.\n", cat, adapter_str);
+		ast_log(LOG_ERROR, "Skipping device %s. Unknown adapter '%s' specified.\n", cat, adapter_str);
 		goto e_return;
 	}
 

addons/chan_ooh323.c

@@ -2055,7 +2055,7 @@ int onOutgoingCall(ooCallData *call)
 			}
 			ooCallAddAliasDialedDigits(call, p->caller_dialedDigits);
 		} else if (!ast_strlen_zero(p->callerid_num)) {
-			if (ooIsDailedDigit(p->callerid_num)) {
+			if (ooIsDialedDigit(p->callerid_num)) {
 				if (gH323Debug) {
 					ast_verb(0, "setting callid number %s\n", p->callerid_num);
 				}
@@ -2136,7 +2136,7 @@ int onNewCallCreated(ooCallData *call)
 			}
 			ooCallAddAliasDialedDigits(call, p->caller_dialedDigits);
 		} else if (!ast_strlen_zero(p->callerid_num)) {
-			if (ooIsDailedDigit(p->callerid_num)) {
+			if (ooIsDialedDigit(p->callerid_num)) {
 				if (gH323Debug) {
 					ast_verb(0, "setting callid number %s\n", p->callerid_num);
 				}
@@ -2148,7 +2148,7 @@ int onNewCallCreated(ooCallData *call)
 
 
 		if (!ast_strlen_zero(p->exten))  {
-			if (ooIsDailedDigit(p->exten)) {
+			if (ooIsDialedDigit(p->exten)) {
 				ooCallSetCalledPartyNumber(call, p->exten);
 				ooCallAddRemoteAliasDialedDigits(call, p->exten);
 			} else {

addons/ooh323c/README

@@ -75,7 +75,7 @@ To run the stack test application chansetup
 3. For running calling instance
 
         ./h323peer [--use-ip ip] -n <number of calls> -duration <call duration>
-                   -interval <inetrval between successive calls> destination
+                   -interval <interval between successive calls> destination
 
    where all times are in seconds. Interval of 0 means next call will be placed
    after current call finishes. "destination" is the dotted ip address of the

addons/ooh323c/src/decode.c

@@ -417,7 +417,7 @@ int decodeDynBitString (OOCTXT* pctxt, ASN1DynBitStr* pBitStr)
    ASN1OCTET* ptmp;
    int nbits, stat = ASN_OK;
 
-   /* If "fast copy" option is not set (ASN1FATSCOPY) or if constructed,
+   /* If "fast copy" option is not set (ASN1FASTCOPY) or if constructed,
     * copy the bit string value into a dynamic memory buffer;
     * otherwise, store the pointer to the value in the decode
     * buffer in the data pointer argument. */

addons/ooh323c/src/encode.c

@@ -1007,7 +1007,7 @@ int encode2sCompBinInt (OOCTXT* pctxt, ASN1INT value)
 {
    /* 10.4.6  A minimum octet 2's-complement-binary-integer encoding    */
    /* of the whole number has a field width that is a multiple of 8     */
-   /* bits and also satisifies the condition that the leading 9 bits    */
+   /* bits and also satisfies the condition that the leading 9 bits     */
    /* field shall not be all zeros and shall not be all ones.           */
 
    /* first encode integer value into a local buffer */
@@ -1048,7 +1048,7 @@ static int encodeNonNegBinInt (OOCTXT* pctxt, ASN1UINT value)
    /* 10.3.6  A minimum octet non-negative binary integer encoding of   */
    /* the whole number (which does not predetermine the number of       */
    /* octets to be used for the encoding) has a field which is a        */
-   /* multiple of 8 bits and also satisifies the condition that the     */
+   /* multiple of 8 bits and also satisfies the condition that the      */
    /* leading eight bits of the field shall not be zero unless the      */
    /* field is precisely 8 bits long.                                   */
 

addons/ooh323c/src/eventHandler.h

@@ -49,7 +49,7 @@ extern "C" {
  * SEQUENCE OF, SET OF, or CHOICE construct is parsed.
  *
  * @param name         For SEQUENCE, SET, or CHOICE, this is the name of the
- *                       element as defined in the ASN.1 defination. For
+ *                       element as defined in the ASN.1 definition. For
  *                       SEQUENCE OF or SET OF, this is set to the name
  *                       "element".
  * @param index        For SEQUENCE, SET, or CHOICE, this is not used and is
@@ -68,7 +68,7 @@ typedef void (*StartElement) (const char* name, int index) ;
  * SEQUENCE, SET, SEQUENCE OF, SET OF, or CHOICE construct.
  *
  * @param name         For SEQUENCE, SET, or CHOICE, this is the name of the
- *                       element as defined in the ASN.1 defination. For
+ *                       element as defined in the ASN.1 definition. For
  *                       SEQUENCE OF or SET OF, this is set to the name
  *                       "element".
  * @param index        For SEQUENCE, SET, or CHOICE, this is not used and is
@@ -194,7 +194,7 @@ typedef void (*EnumValue) (ASN1UINT value) ;
  * within a decode function when an ASN.1 open type is parsed.
  *
  * @param numocts      Number of octets in the parsed value.
- * @param data         Pointer to byet array contain in tencoded ASN.1
+ * @param data         Pointer to byte array contain in tencoded ASN.1
  *                       value.
  * @return             - none
  */

addons/ooh323c/src/h323/H323-MESSAGES.h

@@ -1926,7 +1926,7 @@ typedef struct EXTERN H225SecurityCapabilities {
    } m;
    H225NonStandardParameter nonStandard;
    H225SecurityServiceMode encryption;
-   H225SecurityServiceMode authenticaton;
+   H225SecurityServiceMode authentication;
    H225SecurityServiceMode integrity;
 } H225SecurityCapabilities;
 

addons/ooh323c/src/h323/H323-MESSAGESDec.c

@@ -6357,14 +6357,14 @@ EXTERN int asn1PD_H225SecurityCapabilities (OOCTXT* pctxt, H225SecurityCapabilit
 
    invokeEndElement (pctxt, "encryption", -1);
 
-   /* decode authenticaton */
+   /* decode authentication */
 
-   invokeStartElement (pctxt, "authenticaton", -1);
+   invokeStartElement (pctxt, "authentication", -1);
 
-   stat = asn1PD_H225SecurityServiceMode (pctxt, &pvalue->authenticaton);
+   stat = asn1PD_H225SecurityServiceMode (pctxt, &pvalue->authentication);
    if (stat != ASN_OK) return stat;
 
-   invokeEndElement (pctxt, "authenticaton", -1);
+   invokeEndElement (pctxt, "authentication", -1);
 
    /* decode integrity */
 

addons/ooh323c/src/h323/H323-MESSAGESEnc.c

@@ -4408,9 +4408,9 @@ EXTERN int asn1PE_H225SecurityCapabilities (OOCTXT* pctxt, H225SecurityCapabilit
    stat = asn1PE_H225SecurityServiceMode (pctxt, &pvalue->encryption);
    if (stat != ASN_OK) return stat;
 
-   /* encode authenticaton */
+   /* encode authentication */
 
-   stat = asn1PE_H225SecurityServiceMode (pctxt, &pvalue->authenticaton);
+   stat = asn1PE_H225SecurityServiceMode (pctxt, &pvalue->authentication);
    if (stat != ASN_OK) return stat;
 
    /* encode integrity */

addons/ooh323c/src/memheap.c

@@ -265,7 +265,7 @@ void* memHeapAlloc (void** ppvMemHeap, int nbytes)
                   if (nunits <= (ASN1UINT)pElem_nunits (pElem)) {
                      RTMEMDIAG3
                         ("memHeapAlloc: "
-                         "found an exisiting free element 0x%x, size %d\n",
+                         "found an existing free element 0x%x, size %d\n",
                         pElem, (pElem_nunits (pElem) * 8u));
 
                      if (pMemBlk->freeElemOff ==
@@ -373,7 +373,7 @@ void* memHeapAlloc (void** ppvMemHeap, int nbytes)
          CHECKMEMBLOCK (pMemHeap, pMemBlk);
       }
       else  {
-	 ast_mutex_unlock(&pMemHeap->pLock);
+	      ast_mutex_unlock(&pMemHeap->pLock);
          return NULL;
       }
    }
@@ -753,9 +753,10 @@ void* memHeapRealloc (void** ppvMemHeap, void* mem_p, int nbytes_)
              if (newMemBlk == 0)
                 return 0;
              pMemLink->pMemBlk = newMemBlk;
-	 }
-         else
+	      }
+         else {
             return 0;
+         }
          *(int*)(((char*)pMemLink) + sizeof (OSMemLink)) = nbytes_;
          return pMemLink->pMemBlk;
       }
@@ -1132,7 +1133,7 @@ void* memHeapMarkSaved (void** ppvMemHeap, const void* mem_p,
          RTMEMDIAG2 ("memHeapMarkSaved: the element 0x%x is "
                          "already free!\n", pElem);
 
-	 ast_mutex_unlock(&pMemHeap->pLock);
+	      ast_mutex_unlock(&pMemHeap->pLock);
          return 0;
       }
 
@@ -1151,9 +1152,10 @@ void* memHeapMarkSaved (void** ppvMemHeap, const void* mem_p,
             CLEAR_SAVED (pMemBlk, pElem);
          nsaved = pMemBlk->nsaved;
       }
-      else
-	 ast_mutex_unlock(&pMemHeap->pLock);
-         return 0;
+      else {
+	      ast_mutex_unlock(&pMemHeap->pLock);
+      }
+      return 0;
    }
    if (saved && nsaved > 0)
       pMemLink->blockType |= RTMEMSAVED;
@@ -1212,7 +1214,7 @@ static OSMemLink* memHeapAddBlock (OSMemLink** ppMemLink,
 
    /* if pMemBlk has RTMEMLINK flags it means that it is allocated
     * cooperatively with OSMemLink, and we don't need to do additional
-    * allocations for it. Just use pointer's arithemtic. */
+    * allocations for it. Just use pointer's arithmetic. */
 
    if (blockType & RTMEMLINK)
       pMemLink = (OSMemLink*) (((ASN1OCTET*)pMemBlk) - sizeof (OSMemLink));

addons/ooh323c/src/ooCalls.c

@@ -783,7 +783,7 @@ OOH323CallData* ooFindCallByToken(const char *callToken)
 
 
 
-/* Checks whether session with suplied ID and direction is already active*/
+/* Checks whether session with supplied ID and direction is already active*/
 ASN1BOOL ooIsSessionEstablished(OOH323CallData *call, int sessionID, char* dir)
 {
    OOLogicalChannel * temp = NULL;

addons/ooh323c/src/ooCapability.h

@@ -294,7 +294,7 @@ EXTERN int ooCapabilityDisableDTMFQ931Keypad(struct OOH323CallData *call);
  * and txframes parameters to the endpoint or call.(ex. G711, G728, G723.1,
  * G729)
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param cap                  Type of G711 capability to be added.
@@ -326,7 +326,7 @@ EXTERN int ooCapabilityAddSimpleCapability
  * to local endpoints capability list or to remote endpoints capability list or
  * to a call's capability list.
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param cap                  Type of GSM capability to be added.
@@ -357,7 +357,7 @@ int ooCapabilityAddGSMCapability(struct OOH323CallData *call, int cap,
  * capability list or to remote endpoints capability list or to a call's
  * capability list.
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param sqcifMPI             Minimum picture interval for encoding/decoding
@@ -396,7 +396,7 @@ EXTERN int ooCapabilityAddH263VideoCapability(struct OOH323CallData *call,
 /**
  * This function is an helper function to ooCapabilityAddH263VideoCapability.
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param sqcifMPI             Minimum picture interval for encoding/decoding
@@ -717,11 +717,11 @@ EXTERN int ooAppendCapToCapPrefs(struct OOH323CallData *call, int cap);
 EXTERN int ooChangeCapPrefOrder(struct OOH323CallData *call, int cap, int pos);
 
 /**
- * This function is used to preppend a particular capability to preference
+ * This function is used to prepend a particular capability to preference
  * list.
  * @param call     Handle to call, if call's preference list has to be modified
  *                 else NULL, to modify endpoint's preference list.
- * @param cap      Capability to be preppended.
+ * @param cap      Capability to be prepended.
  *
  * @return         OO_OK, on success. OO_FAILED, otherwise.
  */

addons/ooh323c/src/ooCmdChannel.c

@@ -186,7 +186,7 @@ int ooReadAndProcessStackCommand()
                 if(!pCall) {
                    OOTRACEINFO2("Call \"%s\" does not exist\n",
                                 (char*)cmd.param1);
-                   OOTRACEINFO1("Call migth be cleared/closed\n");
+                   OOTRACEINFO1("Call might be cleared/closed\n");
                 }
                 else {
                      ooSendProgress(ooFindCallByToken((char*)cmd.param1));
@@ -200,7 +200,7 @@ int ooReadAndProcessStackCommand()
                   if(!pCall) {
                      OOTRACEINFO2("Call \"%s\" does not exist\n",
                                   (char*)cmd.param1);
-                     OOTRACEINFO1("Call migth be cleared/closed\n");
+                     OOTRACEINFO1("Call might be cleared/closed\n");
                   }
                   else {
                      ooSendAlerting(ooFindCallByToken((char*)cmd.param1));

addons/ooh323c/src/ooGkClient.c

@@ -1231,7 +1231,7 @@ int ooGkClientHandleRegistrationConfirm
                                 (&pGkClient->ctxt, sizeof(ooGkClientTimerCb));
       if(!cbData)
       {
-         OOTRACEERR1("Error:Failed to allocate memory for Regisration timer."
+         OOTRACEERR1("Error:Failed to allocate memory for Registration timer."
                      "\n");
          pGkClient->state = GkClientFailed;
          return OO_FAILED;
@@ -1927,7 +1927,7 @@ int ooGkClientSendAdmissionRequest
                                (&pGkClient->ctxt, sizeof(ooGkClientTimerCb));
    if(!cbData)
    {
-      OOTRACEERR1("Error:Failed to allocate memory for Regisration timer."
+      OOTRACEERR1("Error:Failed to allocate memory for Registration timer."
                   "\n");
       pGkClient->state = GkClientFailed;
       ast_mutex_unlock(&pGkClient->Lock);

addons/ooh323c/src/ooGkClient.h

@@ -318,7 +318,7 @@ EXTERN int ooGkClientHandleRASMessage
 
 
 /**
- * This function is used to send a message on Gatekeeper clien't RAS channel.
+ * This function is used to send a message on Gatekeeper client RAS channel.
  * @param pGkClient   Handle to the gatekeeper client.
  * @param pRasMsg     Handle to Ras message to be sent.
  *

addons/ooh323c/src/ooLogChan.c

@@ -78,7 +78,7 @@ OOLogicalChannel* ooAddNewLogicalChannel(OOH323CallData *call, int channelNo,
       OOTRACEDBGC3("Using configured media info (%s, %s)\n", call->callType,
                    call->callToken);
       pNewChannel->localRtpPort = pMediaInfo->lMediaRedirPort ? pMediaInfo->lMediaRedirPort : pMediaInfo->lMediaPort;
-      /* check MediaRedirPort here because RedirCPort is ReditPort + 1 and can't be 0 ;) */
+      /* check MediaRedirPort here because RedirCPort is RedirPort + 1 and can't be 0 ;) */
       pNewChannel->localRtcpPort = pMediaInfo->lMediaRedirPort ? pMediaInfo->lMediaRedirCPort : pMediaInfo->lMediaCntrlPort;
       /* If user application has not specified a specific ip and is using
          multihomed mode, substitute appropriate ip.

addons/ooh323c/src/ooSocket.c

@@ -537,8 +537,6 @@ int ooPDWrite(struct pollfd *pfds, int nfds, int fd)
 int ooGetLocalIPAddress(char * pIPAddrs)
 {
    int ret;
-   struct hostent *hp;
-   struct ast_hostent phost;
    char hostname[100];
 
    if(pIPAddrs == NULL)
@@ -546,20 +544,11 @@ int ooGetLocalIPAddress(char * pIPAddrs)
    ret = gethostname(hostname, 100);
    if(ret == 0)
    {
-      if ((hp = ast_gethostbyname(hostname, &phost))) {
-			if (hp->h_addrtype == AF_INET6) {
-				struct in6_addr i;
-				memcpy(&i, hp->h_addr, sizeof(i));
-				strcpy(pIPAddrs, (inet_ntop(AF_INET6, &i,
-				hostname, sizeof(hostname))) == NULL ? "::1" :
-				inet_ntop(AF_INET6, &i, hostname, sizeof(hostname)));
-			} else {
-	  			struct in_addr i;
-				memcpy(&i, hp->h_addr, sizeof(i));
-			  	strcpy(pIPAddrs, (ast_inet_ntoa(i) == NULL) ? "127.0.0.1" : ast_inet_ntoa(i));
-			}
-      } else {
+      struct ast_sockaddr addr = { {0,} };
+      if (ast_sockaddr_resolve_first_af(&addr, hostname, PARSE_PORT_FORBID, AF_UNSPEC)) {
          return -1;
+      } else {
+         strcpy(pIPAddrs, ast_sockaddr_stringify_addr(&addr));
       }
    }
    else{

addons/ooh323c/src/ooSocket.h

@@ -318,7 +318,7 @@ EXTERN int ooSocketSendTo(OOSOCKET socket, const ASN1OCTET* pdata,
 
 /**
  * This function is used for synchronous monitoring of multiple sockets.
- * For more information refer to documnetation of "select" system call.
+ * For more information refer to documentation of "select" system call.
  *
  * @param nfds         The highest numbered descriptor to be monitored
  *                     plus one.

addons/ooh323c/src/ooUtils.c

@@ -30,7 +30,7 @@ OOBOOL ooUtilsIsStrEmpty (const char* str)
 }
 
 
-OOBOOL ooIsDailedDigit(const char* str)
+OOBOOL ooIsDialedDigit(const char* str)
 {
    if(str == NULL || *str =='\0') { return FALSE; }
    while(*str != '\0')

addons/ooh323c/src/ooUtils.h

@@ -48,6 +48,6 @@ EXTERN OOBOOL ooUtilsIsStrEmpty (const char * str);
  * @param str    String to test
  * @return       TRUE if string contains all digits; FALSE otherwise
  */
-EXTERN OOBOOL ooIsDailedDigit(const char* str);
+EXTERN OOBOOL ooIsDialedDigit(const char* str);
 
 #endif

addons/ooh323c/src/ooasn1.h

@@ -607,7 +607,7 @@ EXTERN int initContextBuffer
 /**
  * This function initializes a context block. It makes sure that if the block
  * was not previosly initialized, that all key working parameters are set to
- * thier correct initial state values (i.e. declared within a function as a
+ * their correct initial state values (i.e. declared within a function as a
  * normal working variable), it is required that they invoke this function
  * before using it.
  *

addons/ooh323c/src/oochannels.c

@@ -297,7 +297,7 @@ int ooCreateH225Connection(OOH323CallData *call)
       {
          call->pH225Channel->sock = channelSocket;
 
-         OOTRACEINFO3("H2250 transmiter channel creation - successful "
+         OOTRACEINFO3("H2250 transmitter channel creation - successful "
                       "(%s, %s)\n", call->callType, call->callToken);
 
          /* If multihomed, get ip from socket */

addons/ooh323c/src/ooh245.c

@@ -2559,7 +2559,7 @@ int ooOnReceivedRequestChannelClose(OOH323CallData *call,
    if (lChannel->state == OO_LOGICALCHAN_ESTABLISHED) {
      ret = ooSendCloseLogicalChannel(call, lChannel);
      if (ret != OO_OK) {
-      OOTRACEERR3("ERROR:Failed to build CloseLgicalChannel message(%s, %s)\n",
+      OOTRACEERR3("ERROR:Failed to build CloseLogicalChannel message(%s, %s)\n",
                    call->callType, call->callToken);
       return OO_FAILED;
      }

addons/ooh323c/src/ooh245.h

@@ -174,7 +174,7 @@ EXTERN int ooSendMasterSlaveDeterminationReject (struct OOH323CallData* call);
  * MasterSlaveDetermination procedure.
  * @param call        Handle to the call for which MasterSlaveReject is
  *                    received.
- * @param reject      Poinetr to the received reject message.
+ * @param reject      Pointer to the received reject message.
  *
  * @return            OO_OK, on success. OO_FAILED, on failure.
  */
@@ -271,7 +271,7 @@ EXTERN int ooOnReceivedUserInputIndication
    (OOH323CallData *call, H245UserInputIndication *indication);
 
 /**
- * This function is called on receiving a TreminalCapabilitySetAck message.
+ * This function is called on receiving a TerminalCapabilitySetAck message.
  * If the MasterSlaveDetermination process is also over, this function
  * initiates the process of opening logical channels.
  * @param call          Pointer to call for which TCSAck is received.

addons/ooh323c/src/ooh323.c

@@ -616,7 +616,7 @@ int ooOnReceivedSetup(OOH323CallData *call, Q931Message *q931Msg)
               removeEventHandler(call->pctxt);
               return OO_FAILED;
          }
-         /* For now, just add decoded fast start elemts to list. This list
+         /* For now, just add decoded fast start elements to list. This list
             will be processed at the time of sending CONNECT message. */
          dListAppend(call->pctxt, &call->remoteFastStartOLCs, olc);
       }
@@ -1062,16 +1062,17 @@ int ooOnReceivedAlerting(OOH323CallData *call, Q931Message *q931Msg)
    if(q931Msg->userInfo->h323_uu_pdu.m.h245TunnelingPresent &&
       q931Msg->userInfo->h323_uu_pdu.h245Tunneling &&
 	 OO_TESTFLAG (call->flags, OO_M_TUNNELING)) {
-      if (alerting->m.h245AddressPresent)
+      if (alerting->m.h245AddressPresent) {
       	OOTRACEINFO3("Tunneling and h245address provided."
                      "Giving preference to Tunneling (%s, %s)\n",
                    	call->callType, call->callToken);
-	if (call->h225version >= 4) {
-		ret =ooSendTCSandMSD(call);
-	}
-	if (ret != OO_OK)
-		return ret;
-
+      }
+	   if (call->h225version >= 4) {
+		   ret =ooSendTCSandMSD(call);
+	   }
+	   if (ret != OO_OK) {
+		   return ret;
+      }
    } else if(alerting->m.h245AddressPresent) {
       if (OO_TESTFLAG (call->flags, OO_M_TUNNELING))
       {
@@ -1314,15 +1315,17 @@ int ooOnReceivedProgress(OOH323CallData *call, Q931Message *q931Msg)
    if(q931Msg->userInfo->h323_uu_pdu.m.h245TunnelingPresent &&
       q931Msg->userInfo->h323_uu_pdu.h245Tunneling &&
       OO_TESTFLAG (call->flags, OO_M_TUNNELING)) {
-      if (progress->m.h245AddressPresent)
+      if (progress->m.h245AddressPresent) {
       	OOTRACEINFO3("Tunneling and h245address provided."
                      "Giving preference to Tunneling (%s, %s)\n",
                      call->callType, call->callToken);
-	if (call->h225version >= 4) {
-		ret =ooSendTCSandMSD(call);
-	}
-	if (ret != OO_OK)
+      }
+	   if (call->h225version >= 4) {
+		   ret =ooSendTCSandMSD(call);
+	   }
+	   if (ret != OO_OK) {
 		return ret;
+      }
    } else if(progress->m.h245AddressPresent) {
       if (OO_TESTFLAG (call->flags, OO_M_TUNNELING))
       {

addons/ooh323c/src/ooh323ep.c

@@ -65,7 +65,7 @@ int ooH323EpInitialize
    }
 
    /* Initialize default port ranges that will be used by stack.
-      Apps can override these by explicitely setting port ranges
+      Apps can override these by explicitly setting port ranges
    */
 
    gH323ep.tcpPorts.start = TCPPORTSSTART;

addons/ooh323c/src/ooq931.c

@@ -192,11 +192,13 @@ EXTERN int ooQ931Decode
          screening indicators ;-) */
       if(ie->discriminator == Q931CallingPartyNumberIE)
       {
+         int numoffset=1;
          OOTRACEDBGB1("   CallingPartyNumber IE = {\n");
-         if(ie->length < OO_MAX_NUMBER_LENGTH)
+         if(!(0x80 & ie->data[0])) numoffset = 2;
+
+         if( (ie->length >= numoffset) &&
+             (ie->length < OO_MAX_NUMBER_LENGTH) )
          {
-            int numoffset=1;
-            if(!(0x80 & ie->data[0])) numoffset = 2;
             memcpy(number, ie->data+numoffset,ie->length-numoffset);
             number[ie->length-numoffset]='\0';
             OOTRACEDBGB2("      %s\n", number);
@@ -204,7 +206,7 @@ EXTERN int ooQ931Decode
                ooCallSetCallingPartyNumber(call, number);
          }
          else{
-            OOTRACEERR3("Error:Calling party number too long. (%s, %s)\n",
+            OOTRACEERR3("Error:Calling party number outside range. (%s, %s)\n",
                            call->callType, call->callToken);
          }
          OOTRACEDBGB1("   }\n");
@@ -214,7 +216,8 @@ EXTERN int ooQ931Decode
       if(ie->discriminator == Q931CalledPartyNumberIE)
       {
          OOTRACEDBGB1("   CalledPartyNumber IE = {\n");
-         if(ie->length < OO_MAX_NUMBER_LENGTH)
+         if( (ie->length >= 1) &&
+             (ie->length < OO_MAX_NUMBER_LENGTH) )
          {
             memcpy(number, ie->data+1,ie->length-1);
             number[ie->length-1]='\0';
@@ -223,7 +226,7 @@ EXTERN int ooQ931Decode
                ooCallSetCalledPartyNumber(call, number);
          }
          else{
-            OOTRACEERR3("Error:Calling party number too long. (%s, %s)\n",
+            OOTRACEERR3("Error:Calling party number outside range. (%s, %s)\n",
                            call->callType, call->callToken);
          }
          OOTRACEDBGB1("   }\n");
@@ -585,7 +588,7 @@ int ooDecodeUUIE(OOCTXT* pctxt, Q931Message *q931Msg)
    unsigned int i;
    ASN1BOOL aligned=TRUE;
    int stat;
-   Q931InformationElement *ie;
+   Q931InformationElement *ie=NULL;
    /* OOCTXT *pctxt = &gH323ep.msgctxt; */
    if(q931Msg ==NULL)
    {
@@ -1210,7 +1213,7 @@ int ooSetFastStartResponse(OOH323CallData *pCall, Q931Message *pQ931msg,
          pChannel = ooFindLogicalChannelByLogicalChannelNo
                       (pCall, olc->forwardLogicalChannelNumber);
 
-         /* start receive and tramsmit channel listening */
+         /* start receive and transmit channel listening */
          if(dir & OORX)
          {
             strcpy(pChannel->remoteIP, remoteMediaControlIP);
@@ -2025,7 +2028,7 @@ int ooSendStatusInquiry(OOH323CallData *call)
    /* OOCTXT *pctxt = &gH323ep.msgctxt; */
    OOCTXT *pctxt = call->msgctxt;
 
-   OOTRACEDBGC3("Building StatusInquryMsg (%s, %s)\n", call->callType,
+   OOTRACEDBGC3("Building StatusInquiryMsg (%s, %s)\n", call->callType,
                  call->callToken);
    ret = ooCreateQ931Message(pctxt, &q931msg, Q931StatusEnquiryMsg);
    if(ret != OO_OK)
@@ -2987,7 +2990,7 @@ int ooH323MakeCall_helper(OOH323CallData *call)
          }
          if(!epCap)
          {
-            OOTRACEWARN4("Warn:Preferred capability %s is abscent in "
+            OOTRACEWARN4("Warn:Preferred capability %s is absent in "
                          "capability list. (%s, %s)\n",
                          ooGetCapTypeText(call->capPrefs.order[k]),
                          call->callType, call->callToken);

addons/ooh323c/src/ooq931.h

@@ -582,7 +582,7 @@ EXTERN int ooH323MakeCall_helper(struct OOH323CallData *call);
  * @param dest      Destination string to be parsed.
  * @param parsedIP  Pointer to buffer in which parsed ip:port will be returned.
  * @param len       Length of the buffer passed.
- * @param aliasList Aliase List in which new aliases will be added.
+ * @param aliasList Aliases List in which new aliases will be added.
  *
  * @return          OO_OK, on success. OO_FAILED, on failure.
  */

agi/eagi-test.c

@@ -93,7 +93,7 @@ static char *wait_result(void)
 				fprintf(stderr, "Got %d/%d bytes of audio\n", res, bytes);
 #endif
 				bytes += res;
-				/* Prentend we detected some audio after 3 seconds */
+				/* Pretend we detected some audio after 3 seconds */
 				if (bytes > 16000 * 3) {
 					return "Sample Message";
 					bytes = 0;

agi/jukebox.agi

@@ -348,7 +348,7 @@ sub music_dir_cache {
 }
 
 # this function will fill the @masterCacheList of all the files that
-# need to have text2speeced ulaw files of their names generated
+# need to have text2speech ulaw files of their names generated
 sub music_dir_cache_genlist {
 	my $dir = shift;
 	if (!opendir(THEDIR, rmts($MUSIC.$dir))) {

apps/app_agent_pool.c

@@ -24,8 +24,14 @@
  *
  * See Also:
  * \arg \ref AstCREDITS
- * \arg \ref Config_agent
+ * \arg \ref agents.conf "Config_agent"
  */
+
+/*!
+ * \page agents.conf agents.conf
+ * \verbinclude agents.conf.sample
+ */
+
 /*** MODULEINFO
 	<support_level>core</support_level>
  ***/
@@ -456,17 +462,11 @@ struct agents_cfg {
 	struct ao2_container *agents;
 };
 
-static const char *agent_type_blacklist[] = {
-	"general",
-	"agents",
-	NULL,
-};
-
 static struct aco_type agent_type = {
 	.type = ACO_ITEM,
 	.name = "agent-id",
-	.category_match = ACO_BLACKLIST_ARRAY,
-	.category = (const char *)agent_type_blacklist,
+	.category_match = ACO_BLACKLIST_EXACT,
+	.category = "general",
 	.item_alloc = agent_cfg_alloc,
 	.item_find = agent_cfg_find,
 	.item_offset = offsetof(struct agents_cfg, agents),
@@ -652,8 +652,6 @@ static struct ao2_container *agents;
  * \brief Lock the agent.
  *
  * \param agent Agent to lock
- *
- * \return Nothing
  */
 #define agent_lock(agent)	_agent_lock(agent, __FILE__, __PRETTY_FUNCTION__, __LINE__, #agent)
 static inline void _agent_lock(struct agent_pvt *agent, const char *file, const char *function, int line, const char *var)
@@ -665,8 +663,6 @@ static inline void _agent_lock(struct agent_pvt *agent, const char *file, const
  * \brief Unlock the agent.
  *
  * \param agent Agent to unlock
- *
- * \return Nothing
  */
 #define agent_unlock(agent)	_agent_unlock(agent, __FILE__, __PRETTY_FUNCTION__, __LINE__, #agent)
 static inline void _agent_unlock(struct agent_pvt *agent, const char *file, const char *function, int line, const char *var)
@@ -684,8 +680,6 @@ static inline void _agent_unlock(struct agent_pvt *agent, const char *file, cons
  * \note Assumes the agent lock is already obtained.
  *
  * \note Defined locking order is channel lock then agent lock.
- *
- * \return Nothing
  */
 static struct ast_channel *agent_lock_logged(struct agent_pvt *agent)
 {
@@ -750,8 +744,6 @@ static enum ast_device_state agent_pvt_devstate_get(const char *agent_id)
  * \since 12.0.0
  *
  * \param agent_id Which agent needs the device state updated.
- *
- * \return Nothing
  */
 static void agent_devstate_changed(const char *agent_id)
 {
@@ -1011,8 +1003,6 @@ AST_MUTEX_DEFINE_STATIC(agent_holding_lock);
  * \param payload_size Size of the payload if payload is non-NULL.  A number otherwise.
  *
  * \note The payload MUST NOT have any resources that need to be freed.
- *
- * \return Nothing
  */
 static void clear_agent_status(struct ast_bridge_channel *bridge_channel, const void *payload, size_t payload_size)
 {
@@ -1028,8 +1018,6 @@ static void clear_agent_status(struct ast_bridge_channel *bridge_channel, const
  * \param agent Which agent is connecting to the caller.
  *
  * \note The agent is locked on entry and not locked on exit.
- *
- * \return Nothing
  */
 static void agent_connect_caller(struct ast_bridge_channel *bridge_channel, struct agent_pvt *agent)
 {
@@ -1363,8 +1351,6 @@ static int bridge_agent_hold_push(struct ast_bridge *self, struct ast_bridge_cha
  * bridge_channel->bridge_pvt.
  *
  * \note On entry, self is already locked.
- *
- * \return Nothing
  */
 static void bridge_agent_hold_pull(struct ast_bridge *self, struct ast_bridge_channel *bridge_channel)
 {
@@ -1382,8 +1368,6 @@ static void bridge_agent_hold_pull(struct ast_bridge *self, struct ast_bridge_ch
  * references to the bridge so it can be destroyed.
  *
  * \note On entry, self must NOT be locked.
- *
- * \return Nothing
  */
 static void bridge_agent_hold_dissolving(struct ast_bridge *self)
 {
@@ -1475,8 +1459,6 @@ static void send_agent_logoff(struct ast_channel *chan, const char *agent, long
  * \param agent Which agent logging out.
  *
  * \note On entry agent is already locked.  On exit it is no longer locked.
- *
- * \return Nothing
  */
 static void agent_logout(struct agent_pvt *agent)
 {
@@ -1514,8 +1496,6 @@ static void agent_logout(struct agent_pvt *agent)
  *
  * \param agent Which agent.
  * \param logged The logged in channel.
- *
- * \return Nothing
  */
 static void agent_run(struct agent_pvt *agent, struct ast_channel *logged)
 {
@@ -2034,8 +2014,6 @@ static int agent_request_exec(struct ast_channel *chan, const char *data)
  *
  * \param agent What to setup channel config values on.
  * \param chan Channel logging in as an agent.
- *
- * \return Nothing
  */
 static void agent_login_channel_config(struct agent_pvt *agent, struct ast_channel *chan)
 {

apps/app_alarmreceiver.c

@@ -158,7 +158,7 @@ static const char app[] = "AlarmReceiver";
 			events to the standard input of the application.
 			The configuration file also contains settings for DTMF timing, and for the loudness of the
 			acknowledgement tones.</para>
-			<note><para>Few Ademco DTMF signalling formats are detected automaticaly: Contact ID, Express 4+1,
+			<note><para>Few Ademco DTMF signalling formats are detected automatically: Contact ID, Express 4+1,
 			Express 4+2, High Speed and Super Fast.</para></note>
 			<para>The application is affected by the following variables:</para>
 			<variablelist>
@@ -203,7 +203,6 @@ static char event_file[14] = "/event-XXXXXX";
  * family, then create it and set its value to 1.
  *
  * \param key A database key to increment
- * \return Nothing
  */
 static void database_increment(char *key)
 {

apps/app_amd.c

@@ -92,6 +92,12 @@
 				<para>Is the maximum duration of a word to accept.</para>
 				<para>If exceeded, then the result is detection as a MACHINE</para>
 			</parameter>
+			<parameter name="audioFile" required="false">
+				<para>Is an audio file to play to the caller while AMD is in progress.</para>
+				<para>By default, no audio file is played.</para>
+				<para>If an audio file is configured in amd.conf, then that file will be used
+				if one is not specified here. That file may be overridden by this argument.</para>
+			</parameter>
 		</syntax>
 		<description>
 			<para>This application attempts to detect answering machines at the beginning
@@ -155,6 +161,9 @@ static int dfltBetweenWordsSilence  = 50;
 static int dfltMaximumNumberOfWords = 2;
 static int dfltSilenceThreshold     = 256;
 static int dfltMaximumWordLength    = 5000; /* Setting this to a large default so it is not used unless specify it in the configs or command line */
+static char *dfltAudioFile = NULL;
+
+static ast_mutex_t config_lock;
 
 /* Set to the lowest ms value provided in amd.conf or application parameters */
 static int dfltMaxWaitTimeForFrame  = 50;
@@ -179,7 +188,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	char amdCause[256] = "", amdStatus[256] = "";
 	char *parse = ast_strdupa(data);
 
-	/* Lets set the initial values of the variables that will control the algorithm.
+	/* Let's set the initial values of the variables that will control the algorithm.
 	   The initial values are the default ones. If they are passed as arguments
 	   when invoking the application, then the default values will be overwritten
 	   by the ones passed as parameters. */
@@ -193,6 +202,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	int silenceThreshold     = dfltSilenceThreshold;
 	int maximumWordLength    = dfltMaximumWordLength;
 	int maxWaitTimeForFrame  = dfltMaxWaitTimeForFrame;
+	const char *audioFile = NULL;
 
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(argInitialSilence);
@@ -204,8 +214,15 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 		AST_APP_ARG(argMaximumNumberOfWords);
 		AST_APP_ARG(argSilenceThreshold);
 		AST_APP_ARG(argMaximumWordLength);
+		AST_APP_ARG(audioFile);
 	);
 
+	ast_mutex_lock(&config_lock);
+	if (!ast_strlen_zero(dfltAudioFile)) {
+		audioFile = ast_strdupa(dfltAudioFile);
+	}
+	ast_mutex_unlock(&config_lock);
+
 	ast_verb(3, "AMD: %s %s %s (Fmt: %s)\n", ast_channel_name(chan),
 		S_COR(ast_channel_caller(chan)->ani.number.valid, ast_channel_caller(chan)->ani.number.str, "(N/A)"),
 		S_COR(ast_channel_redirecting(chan)->from.number.valid, ast_channel_redirecting(chan)->from.number.str, "(N/A)"),
@@ -233,6 +250,9 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 			silenceThreshold = atoi(args.argSilenceThreshold);
 		if (!ast_strlen_zero(args.argMaximumWordLength))
 			maximumWordLength = atoi(args.argMaximumWordLength);
+		if (!ast_strlen_zero(args.audioFile)) {
+			audioFile = args.audioFile;
+		}
 	} else {
 		ast_debug(1, "AMD using the default parameters.\n");
 	}
@@ -280,6 +300,11 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	/* Set our start time so we can tie the loop to real world time and not RTP updates */
 	amd_tvstart = ast_tvnow();
 
+	/* Optional audio file to play to caller while AMD is doing its thing. */
+	if (!ast_strlen_zero(audioFile)) {
+		ast_streamfile(chan, audioFile, ast_channel_language(chan));
+	}
+
 	/* Now we go into a loop waiting for frames from the channel */
 	while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
 		int ms = 0;
@@ -462,10 +487,14 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	/* Free the DSP used to detect silence */
 	ast_dsp_free(silenceDetector);
 
+	/* If we were playing something to pass the time, stop it now. */
+	if (!ast_strlen_zero(audioFile)) {
+		ast_stopstream(chan);
+	}
+
 	return;
 }
 
-
 static int amd_exec(struct ast_channel *chan, const char *data)
 {
 	isAnsweringMachine(chan, data);
@@ -516,7 +545,16 @@ static int load_config(int reload)
 					dfltMaximumNumberOfWords = atoi(var->value);
 				} else if (!strcasecmp(var->name, "maximum_word_length")) {
 					dfltMaximumWordLength = atoi(var->value);
-
+				} else if (!strcasecmp(var->name, "playback_file")) {
+					ast_mutex_lock(&config_lock);
+					if (dfltAudioFile) {
+						ast_free(dfltAudioFile);
+						dfltAudioFile = NULL;
+					}
+					if (!ast_strlen_zero(var->value)) {
+						dfltAudioFile = ast_strdup(var->value);
+					}
+					ast_mutex_unlock(&config_lock);
 				} else {
 					ast_log(LOG_WARNING, "%s: Cat:%s. Unknown keyword %s at line %d of amd.conf\n",
 						app, cat, var->name, var->lineno);
@@ -539,6 +577,12 @@ static int load_config(int reload)
 
 static int unload_module(void)
 {
+	ast_mutex_lock(&config_lock);
+	if (dfltAudioFile) {
+		ast_free(dfltAudioFile);
+	}
+	ast_mutex_unlock(&config_lock);
+	ast_mutex_destroy(&config_lock);
 	return ast_unregister_application(app);
 }
 
@@ -554,6 +598,7 @@ static int unload_module(void)
  */
 static int load_module(void)
 {
+	ast_mutex_init(&config_lock);
 	if (load_config(0) || ast_register_application_xml(app, amd_exec)) {
 		return AST_MODULE_LOAD_DECLINE;
 	}

apps/app_attended_transfer.c

@@ -140,4 +140,4 @@ static int load_module(void)
 	return ast_register_application_xml(app, attended_transfer_exec);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Attended transfer to the given extension");
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Attended transfer to the given extension");

apps/app_blind_transfer.c

@@ -134,4 +134,4 @@ static int load_module(void)
 	return ast_register_application_xml(app, blind_transfer_exec);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Blind transfer channel to the given destination");
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Blind transfer channel to the given destination");

apps/app_bridgewait.c

@@ -100,6 +100,9 @@
 						<para>Automatically exit the bridge and return to the PBX after
 						<emphasis>duration</emphasis> seconds.</para>
 					</option>
+					<option name="n">
+						<para>Do not automatically answer the channel.</para>
+					</option>
 				</optionlist>
 			</parameter>
 		</syntax>
@@ -108,7 +111,7 @@
 			The channel will then wait in the holding bridge until some event occurs
 			which removes it from the holding bridge.</para>
 			<note><para>This application will answer calls which haven't already
-			been answered.</para></note>
+			been answered, unless the n option is provided.</para></note>
 		</description>
 	</application>
  ***/
@@ -186,6 +189,7 @@ enum bridgewait_flags {
 	MUXFLAG_MOHCLASS = (1 << 0),
 	MUXFLAG_ENTERTAINMENT = (1 << 1),
 	MUXFLAG_TIMEOUT = (1 << 2),
+	MUXFLAG_NOANSWER = (1 << 3),
 };
 
 enum bridgewait_args {
@@ -199,6 +203,7 @@ AST_APP_OPTIONS(bridgewait_opts, {
 	AST_APP_OPTION_ARG('e', MUXFLAG_ENTERTAINMENT, OPT_ARG_ENTERTAINMENT),
 	AST_APP_OPTION_ARG('m', MUXFLAG_MOHCLASS, OPT_ARG_MOHCLASS),
 	AST_APP_OPTION_ARG('S', MUXFLAG_TIMEOUT, OPT_ARG_TIMEOUT),
+	AST_APP_OPTION('n', MUXFLAG_NOANSWER),
 });
 
 static int bridgewait_timeout_callback(struct ast_bridge_channel *bridge_channel, void *hook_pvt)
@@ -391,17 +396,6 @@ static void wait_wrapper_removal(struct wait_bridge_wrapper *wrapper)
 	ao2_cleanup(wrapper);
 }
 
-/*!
- * \internal
- * \since 12.0.0
- * \brief Application callback for the bridgewait application
- *
- * \param chan channel running the application
- * \param data Arguments to the application
- *
- * \retval 0 Ran successfully and the call didn't hang up
- * \retval -1 Failed or the call was hung up by the time the channel exited the holding bridge
- */
 static enum wait_bridge_roles validate_role(const char *role)
 {
 	if (!strcmp(role, "participant")) {
@@ -413,6 +407,17 @@ static enum wait_bridge_roles validate_role(const char *role)
 	}
 }
 
+/*!
+ * \internal
+ * \since 12.0.0
+ * \brief Application callback for the bridgewait application
+ *
+ * \param chan channel running the application
+ * \param data Arguments to the application
+ *
+ * \retval 0 Ran successfully and the call didn't hang up
+ * \retval -1 Failed or the call was hung up by the time the channel exited the holding bridge
+ */
 static int bridgewait_exec(struct ast_channel *chan, const char *data)
 {
 	char *bridge_name = DEFAULT_BRIDGE_NAME;
@@ -458,7 +463,7 @@ static int bridgewait_exec(struct ast_channel *chan, const char *data)
 	}
 
 	/* Answer the channel if needed */
-	if (ast_channel_state(chan) != AST_STATE_UP) {
+	if (ast_channel_state(chan) != AST_STATE_UP && !ast_test_flag(&flags, MUXFLAG_NOANSWER)) {
 		ast_answer(chan);
 	}
 

apps/app_broadcast.c

@@ -0,0 +1,619 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2022, Naveen Albert
+ *
+ * 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 Channel audio broadcasting
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <ctype.h>
+#include <errno.h>
+
+#include "asterisk/channel.h"
+#include "asterisk/audiohook.h"
+#include "asterisk/app.h"
+#include "asterisk/utils.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/options.h"
+#include "asterisk/autochan.h"
+#include "asterisk/format_cache.h"
+#include "asterisk/cli.h" /* use ESS macro */
+
+/*** DOCUMENTATION
+	<application name="Broadcast" language="en_US">
+		<synopsis>
+			Transmit or receive audio to or from multiple channels simultaneously
+		</synopsis>
+		<syntax>
+			<parameter name="options">
+				<optionlist>
+					<option name="b">
+						<para>In addition to broadcasting to target channels, also
+						broadcast to any channels to which target channels are bridged.</para>
+					</option>
+					<option name="l">
+						<para>Allow usage of a long queue to store audio frames.</para>
+						<note><para>This may introduce some delay in the received audio feed, but will improve the audio quality.</para></note>
+					</option>
+					<option name="o">
+						<para>Do not mix streams when combining audio from target channels (only applies with s option).</para>
+					</option>
+					<option name="r">
+						<para>Feed frames to barge channels in "reverse" by injecting them into the primary channel's read queue instead.</para>
+						<para>This option is required for barge to work in a n-party bridge (but not for 2-party bridges). Alternately, you
+						can add an intermediate channel by using a non-optimized Local channel, so that the target channel is bridged with
+						a single channel that is connected to the bridge, but it is recommended this option be used instead.</para>
+						<para>Note that this option will always feed injected audio to the other party, regardless of whether the target
+						channel is bridged or not.</para>
+					</option>
+					<option name="s">
+						<para>Rather than broadcast audio to a bunch of channels, receive the combined audio from the target channels.</para>
+					</option>
+					<option name="w">
+						<para>Broadcast audio received on this channel to other channels.</para>
+					</option>
+				</optionlist>
+			</parameter>
+			<parameter name="channels" required="true" argsep=",">
+				<para>List of channels for broadcast targets.</para>
+				<para>Channel names must be the full channel names, not merely device names.</para>
+				<para>Broadcasting will continue until the broadcasting channel hangs up or all target channels have hung up.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>This application can be used to broadcast audio to multiple channels at once.
+			Any audio received on this channel will be transmitted to all of the specified channels and, optionally, their bridged peers.</para>
+			<para>It can also be used to aggregate audio from multiple channels at once.
+			Any audio on any of the specified channels, and optionally their bridged peers, will be transmitted to this channel.</para>
+			<para>Execution of the application continues until either the broadcasting channel hangs up
+			or all specified channels have hung up.</para>
+			<para>This application is used for one-to-many and many-to-one audio applications where
+			bridge mixing cannot be done synchronously on all the involved channels.
+			This is primarily useful for injecting the same audio stream into multiple channels at once,
+			or doing the reverse, combining the audio from multiple channels into a single stream.
+			This contrasts with using a separate injection channel for each target channel and/or
+			using a conference bridge.</para>
+			<para>The channel running the Broadcast application must do so synchronously. The specified channels,
+			however, may be doing other things.</para>
+			<example title="Broadcast received audio to three channels and their bridged peers">
+			same => n,Broadcast(wb,DAHDI/1,DAHDI/3,PJSIP/doorphone)
+			</example>
+			<example title="Broadcast received audio to three channels, only">
+			same => n,Broadcast(w,DAHDI/1,DAHDI/3,PJSIP/doorphone)
+			</example>
+			<example title="Combine audio from three channels and their bridged peers to us">
+			same => n,Broadcast(s,DAHDI/1,DAHDI/3,PJSIP/doorphone)
+			</example>
+			<example title="Combine audio from three channels to us">
+			same => n,Broadcast(so,DAHDI/1,DAHDI/3,PJSIP/doorphone)
+			</example>
+			<example title="Two-way audio with a bunch of channels">
+			same => n,Broadcast(wbso,DAHDI/1,DAHDI/3,PJSIP/doorphone)
+			</example>
+			<para>Note that in the last example above, this is NOT the same as a conference bridge.
+			The specified channels are not audible to each other, only to the channel running the
+			Broadcast application. The two-way audio is only between the broadcasting channel and
+			each of the specified channels, individually.</para>
+		</description>
+		<see-also>
+			<ref type="application">ChanSpy</ref>
+		</see-also>
+	</application>
+ ***/
+
+static const char app_broadcast[] = "Broadcast";
+
+enum {
+	OPTION_READONLY          = (1 << 0),    /* Don't mix the two channels */
+	OPTION_BARGE             = (1 << 1),    /* Barge mode (whisper to both channels) */
+	OPTION_LONG_QUEUE        = (1 << 2),	/* Allow usage of a long queue to store audio frames. */
+	OPTION_WHISPER           = (1 << 3),
+	OPTION_SPY               = (1 << 4),
+	OPTION_REVERSE_FEED      = (1 << 5),
+	OPTION_ANSWER_WARN       = (1 << 6),	/* Internal flag, not set by user */
+};
+
+AST_APP_OPTIONS(spy_opts, {
+	AST_APP_OPTION('b', OPTION_BARGE),
+	AST_APP_OPTION('l', OPTION_LONG_QUEUE),
+	AST_APP_OPTION('o', OPTION_READONLY),
+	AST_APP_OPTION('r', OPTION_REVERSE_FEED),
+	AST_APP_OPTION('s', OPTION_SPY),
+	AST_APP_OPTION('w', OPTION_WHISPER),
+});
+
+struct multi_autochan {
+	char *name;
+	struct ast_autochan *autochan;
+	struct ast_autochan *bridge_autochan;
+	struct ast_audiohook whisper_audiohook;
+	struct ast_audiohook bridge_whisper_audiohook;
+	struct ast_audiohook spy_audiohook;
+	unsigned int connected:1;
+	unsigned int bridge_connected:1;
+	unsigned int spying:1;
+	AST_LIST_ENTRY(multi_autochan) entry;	/*!< Next record */
+};
+
+AST_RWLIST_HEAD(multi_autochan_list, multi_autochan);
+
+struct multi_spy {
+	struct multi_autochan_list *chanlist;
+	unsigned int readonly:1;
+};
+
+static void *spy_alloc(struct ast_channel *chan, void *data)
+{
+	return data; /* just store the data pointer in the channel structure */
+}
+
+static void spy_release(struct ast_channel *chan, void *data)
+{
+	return; /* nothing to do */
+}
+
+static int spy_generate(struct ast_channel *chan, void *data, int len, int samples)
+{
+	struct multi_spy *multispy = data;
+	struct multi_autochan_list *chanlist = multispy->chanlist;
+	struct multi_autochan *mac;
+	struct ast_frame *f;
+	short *data1, *data2;
+	int res, i;
+
+	/* All the frames we get are slin, so they will all have the same number of samples. */
+	static const int num_samples = 160;
+	short combine_buf[num_samples];
+	struct ast_frame wf = {
+		.frametype = AST_FRAME_VOICE,
+		.offset = 0,
+		.subclass.format = ast_format_slin,
+		.datalen = num_samples * 2,
+		.samples = num_samples,
+		.src = __FUNCTION__,
+	};
+
+	memset(&combine_buf, 0, sizeof(combine_buf));
+	wf.data.ptr = combine_buf;
+
+	AST_RWLIST_WRLOCK(chanlist);
+	AST_RWLIST_TRAVERSE_SAFE_BEGIN(chanlist, mac, entry) {
+		ast_audiohook_lock(&mac->spy_audiohook);
+		if (mac->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
+			ast_audiohook_unlock(&mac->spy_audiohook); /* Channel is already gone more than likely, the broadcasting channel will clean this up. */
+			continue;
+		}
+
+		if (multispy->readonly) { /* Option 'o' was set, so don't mix channel audio */
+			f = ast_audiohook_read_frame(&mac->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_READ, ast_format_slin);
+		} else {
+			f = ast_audiohook_read_frame(&mac->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, ast_format_slin);
+		}
+		ast_audiohook_unlock(&mac->spy_audiohook);
+
+		if (!f) {
+			continue; /* No frame? No problem. */
+		}
+
+		/* Mix the samples. */
+		for (i = 0, data1 = combine_buf, data2 = f->data.ptr; i < num_samples; i++, data1++, data2++) {
+			ast_slinear_saturated_add(data1, data2);
+		}
+		ast_frfree(f);
+	}
+	AST_RWLIST_TRAVERSE_SAFE_END;
+	AST_RWLIST_UNLOCK(chanlist);
+
+	res = ast_write(chan, &wf);
+	ast_frfree(&wf);
+
+	return res;
+}
+
+static struct ast_generator spygen = {
+	.alloc = spy_alloc,
+	.release = spy_release,
+	.generate = spy_generate,
+};
+
+static int start_spying(struct ast_autochan *autochan, const char *spychan_name, struct ast_audiohook *audiohook, struct ast_flags *flags)
+{
+	int res;
+
+	ast_autochan_channel_lock(autochan);
+	ast_debug(1, "Attaching spy channel %s to %s\n", spychan_name, ast_channel_name(autochan->chan));
+
+	if (ast_test_flag(flags, OPTION_READONLY)) {
+		ast_set_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE);
+	} else {
+		ast_set_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
+	}
+	if (ast_test_flag(flags, OPTION_LONG_QUEUE)) {
+		ast_debug(2, "Using a long queue to store audio frames in spy audiohook\n");
+	} else {
+		ast_set_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE);
+	}
+	res = ast_audiohook_attach(autochan->chan, audiohook);
+	ast_autochan_channel_unlock(autochan);
+	return res;
+}
+
+static int attach_barge(struct ast_autochan *spyee_autochan, struct ast_autochan **spyee_bridge_autochan,
+	struct ast_audiohook *bridge_whisper_audiohook, const char *spyer_name, const char *name, struct ast_flags *flags)
+{
+	int retval = 0;
+	struct ast_autochan *internal_bridge_autochan;
+	struct ast_channel *spyee_chan;
+	RAII_VAR(struct ast_channel *, bridged, NULL, ast_channel_cleanup);
+
+	ast_autochan_channel_lock(spyee_autochan);
+	spyee_chan = ast_channel_ref(spyee_autochan->chan);
+	ast_autochan_channel_unlock(spyee_autochan);
+
+	/* Note that ast_channel_bridge_peer only returns non-NULL for 2-party bridges, not n-party bridges (e.g. ConfBridge) */
+	bridged = ast_channel_bridge_peer(spyee_chan);
+	ast_channel_unref(spyee_chan);
+	if (!bridged) {
+		ast_debug(9, "Channel %s is not yet bridged, unable to setup barge\n", ast_channel_name(spyee_chan));
+		/* If we're bridged, but it's not a 2-party bridge, then we probably should have used OPTION_REVERSE_FEED. */
+		if (ast_test_flag(flags, OPTION_ANSWER_WARN) && ast_channel_is_bridged(spyee_chan)) {
+			ast_clear_flag(flags, OPTION_ANSWER_WARN); /* Don't warn more than once. */
+			ast_log(LOG_WARNING, "Barge failed: channel is bridged, but not to a 2-party bridge. Use the 'r' option.\n");
+		}
+		return -1;
+	}
+
+	ast_audiohook_init(bridge_whisper_audiohook, AST_AUDIOHOOK_TYPE_WHISPER, "Broadcast", 0);
+	internal_bridge_autochan = ast_autochan_setup(bridged);
+	if (!internal_bridge_autochan) {
+		return -1;
+	}
+
+	if (start_spying(internal_bridge_autochan, spyer_name, bridge_whisper_audiohook, flags)) {
+		ast_log(LOG_WARNING, "Unable to attach barge audiohook on spyee '%s'. Barge mode disabled.\n", name);
+		retval = -1;
+	}
+
+	*spyee_bridge_autochan = internal_bridge_autochan;
+	return retval;
+}
+
+static void multi_autochan_free(struct multi_autochan *mac)
+{
+	if (mac->connected) {
+		if (mac->whisper_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
+			ast_debug(2, "Whisper audiohook no longer running\n");
+		}
+		ast_audiohook_lock(&mac->whisper_audiohook);
+		ast_audiohook_detach(&mac->whisper_audiohook);
+		ast_audiohook_unlock(&mac->whisper_audiohook);
+		ast_audiohook_destroy(&mac->whisper_audiohook);
+	}
+	if (mac->bridge_connected) {
+		if (mac->bridge_whisper_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
+			ast_debug(2, "Whisper (bridged) audiohook no longer running\n");
+		}
+		ast_audiohook_lock(&mac->bridge_whisper_audiohook);
+		ast_audiohook_detach(&mac->bridge_whisper_audiohook);
+		ast_audiohook_unlock(&mac->bridge_whisper_audiohook);
+		ast_audiohook_destroy(&mac->bridge_whisper_audiohook);
+	}
+	if (mac->spying) {
+		if (mac->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
+			ast_debug(2, "Spy audiohook no longer running\n");
+		}
+		ast_audiohook_lock(&mac->spy_audiohook);
+		ast_audiohook_detach(&mac->spy_audiohook);
+		ast_audiohook_unlock(&mac->spy_audiohook);
+		ast_audiohook_destroy(&mac->spy_audiohook);
+	}
+	if (mac->name) {
+		int total = mac->connected + mac->bridge_connected + mac->spying;
+		ast_debug(1, "Removing channel %s from target list (%d hook%s)\n", mac->name, total, ESS(total));
+		ast_free(mac->name);
+	}
+	if (mac->autochan) {
+		ast_autochan_destroy(mac->autochan);
+	}
+	if (mac->bridge_autochan) {
+		ast_autochan_destroy(mac->bridge_autochan);
+	}
+	ast_free(mac);
+}
+
+static int do_broadcast(struct ast_channel *chan, struct ast_flags *flags, const char *channels)
+{
+	int res = 0;
+	struct ast_frame *f;
+	struct ast_silence_generator *silgen = NULL;
+	struct multi_spy multispy;
+	struct multi_autochan_list chanlist;
+	struct multi_autochan *mac;
+	int numchans = 0;
+	int readonly = ast_test_flag(flags, OPTION_READONLY) ? 1 : 0;
+	char *next, *chansdup = ast_strdupa(channels);
+
+	AST_RWLIST_HEAD_INIT(&chanlist);
+	ast_channel_set_flag(chan, AST_FLAG_SPYING);
+
+	ast_set_flag(flags, OPTION_ANSWER_WARN); /* Initialize answer warn to 1 */
+
+	/* Hey, look ma, no list lock needed! Sometimes, it's nice to not have to share... */
+
+	/* Build a list of targets */
+	while ((next = strsep(&chansdup, ","))) {
+		struct ast_channel *ochan;
+		if (ast_strlen_zero(next)) {
+			continue;
+		}
+		if (!strcmp(next, ast_channel_name(chan))) {
+			ast_log(LOG_WARNING, "Refusing to broadcast to ourself: %s\n", next);
+			continue;
+		}
+		ochan = ast_channel_get_by_name(next);
+		if (!ochan) {
+			ast_log(LOG_WARNING, "No such channel: %s\n", next);
+			continue;
+		}
+		/* Append to end of list. */
+		if (!(mac = ast_calloc(1, sizeof(*mac)))) {
+			ast_log(LOG_WARNING, "Multi autochan allocation failure\n");
+			continue;
+		}
+		mac->name = ast_strdup(next);
+		mac->autochan = ast_autochan_setup(ochan);
+		if (!mac->name || !mac->autochan) {
+			multi_autochan_free(mac);
+			continue;
+		}
+		if (ast_test_flag(flags, OPTION_WHISPER)) {
+			mac->connected = 1;
+			ast_audiohook_init(&mac->whisper_audiohook, AST_AUDIOHOOK_TYPE_WHISPER, "Broadcast", 0);
+			/* Inject audio from our channel to this target. */
+			if (start_spying(mac->autochan, next, &mac->whisper_audiohook, flags)) {
+				ast_log(LOG_WARNING, "Unable to attach whisper audiohook to %s\n", next);
+				multi_autochan_free(mac);
+				continue;
+			}
+		}
+		if (ast_test_flag(flags, OPTION_SPY)) {
+			mac->spying = 1;
+			ast_audiohook_init(&mac->spy_audiohook, AST_AUDIOHOOK_TYPE_SPY, "Broadcast", 0);
+			if (start_spying(mac->autochan, next, &mac->spy_audiohook, flags)) {
+				ast_log(LOG_WARNING, "Unable to attach spy audiohook to %s\n", next);
+				multi_autochan_free(mac);
+				continue;
+			}
+		}
+		AST_RWLIST_INSERT_TAIL(&chanlist, mac, entry);
+		numchans++;
+		ochan = ast_channel_unref(ochan);
+	}
+
+	ast_verb(4, "Broadcasting to %d channel%s on %s\n", numchans, ESS(numchans), ast_channel_name(chan));
+	ast_debug(1, "Broadcasting: (TX->1) whisper=%d, (TX->2) barge=%d, (RX<-%d) spy=%d (%s)\n",
+		ast_test_flag(flags, OPTION_WHISPER) ? 1 : 0,
+		ast_test_flag(flags, OPTION_BARGE) ? 1 : 0,
+		readonly ? 1 : 2,
+		ast_test_flag(flags, OPTION_SPY) ? 1 : 0,
+		readonly ? "single" : "both");
+
+	if (ast_test_flag(flags, OPTION_SPY)) {
+		multispy.chanlist = &chanlist;
+		multispy.readonly = readonly;
+		ast_activate_generator(chan, &spygen, &multispy);
+	} else {
+		/* We're not expecting to read any audio, just broadcast audio to a bunch of other channels. */
+		silgen = ast_channel_start_silence_generator(chan);
+	}
+
+	while (numchans && ast_waitfor(chan, -1) > 0) {
+		int fres = 0;
+		f = ast_read(chan);
+		if (!f) {
+			ast_debug(1, "Channel %s must have hung up\n", ast_channel_name(chan));
+			res = -1;
+			break;
+		}
+		if (f->frametype != AST_FRAME_VOICE) { /* Ignore any non-voice frames */
+			ast_frfree(f);
+			continue;
+		}
+		/* Write the frame to all our targets. */
+		AST_RWLIST_WRLOCK(&chanlist);
+		AST_RWLIST_TRAVERSE_SAFE_BEGIN(&chanlist, mac, entry) {
+			/* Note that if no media is received, execution is suspended, but assuming continuous or
+			 * or frequent audio on the broadcasting channel, we'll quickly enough detect hung up targets.
+			 * This isn't really an issue, just something that might be confusing at first, but this is
+			 * due to the limitation with audiohooks of using the channel for timing. */
+			if ((ast_test_flag(flags, OPTION_WHISPER) && mac->whisper_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING)
+				|| (ast_test_flag(flags, OPTION_SPY) && mac->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING)
+				|| (mac->bridge_connected && ast_test_flag(flags, OPTION_BARGE) && mac->bridge_whisper_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING)) {
+				/* Even if we're spying only and not actually broadcasting audio, we need to detect channel hangup. */
+				AST_RWLIST_REMOVE_CURRENT(entry);
+				ast_debug(2, "Looks like %s has hung up\n", mac->name);
+				multi_autochan_free(mac);
+				numchans--;
+				ast_debug(2, "%d channel%s remaining in broadcast on %s\n", numchans, ESS(numchans), ast_channel_name(chan));
+				continue;
+			}
+
+			if (ast_test_flag(flags, OPTION_WHISPER)) {
+				ast_audiohook_lock(&mac->whisper_audiohook);
+				fres |= ast_audiohook_write_frame(&mac->whisper_audiohook, AST_AUDIOHOOK_DIRECTION_WRITE, f);
+				ast_audiohook_unlock(&mac->whisper_audiohook);
+			}
+
+			if (ast_test_flag(flags, OPTION_BARGE)) {
+				/* This hook lets us inject audio into the channel that the spyee is currently
+				 * bridged with. If the spyee isn't bridged with anything yet, nothing will
+				 * be attached and we'll need to continue attempting to attach the barge
+				 * audio hook.
+				 * The exception to this is if we are emulating barge by doing it "directly",
+				 * that is injecting the frames onto this channel's read queue, rather than
+				 * its bridged peer's write queue, then skip this. We only do one or the other. */
+				if (!ast_test_flag(flags, OPTION_REVERSE_FEED) && !mac->bridge_connected && !attach_barge(mac->autochan, &mac->bridge_autochan,
+						&mac->bridge_whisper_audiohook, ast_channel_name(chan), mac->name, flags)) {
+					ast_debug(2, "Attached barge channel for %s\n", mac->name);
+					mac->bridge_connected = 1;
+				}
+
+				if (mac->bridge_connected) {
+					ast_audiohook_lock(&mac->bridge_whisper_audiohook);
+					fres |= ast_audiohook_write_frame(&mac->bridge_whisper_audiohook, AST_AUDIOHOOK_DIRECTION_WRITE, f);
+					ast_audiohook_unlock(&mac->bridge_whisper_audiohook);
+				} else if (ast_test_flag(flags, OPTION_REVERSE_FEED)) {
+					/* So, this is really clever...
+					 * If we're connected to an n-party bridge instead of a 2-party bridge,
+					 * attach_barge will ALWAYS fail because we're connected to a bridge, not
+					 * a single peer channel.
+					 * Recall that the objective is for injected audio to be audible to both
+					 * sides of the channel. So really, the typical way of doing this by
+					 * directly injecting frames separately onto both channels is kind of
+					 * bizarre to begin with, when you think about it.
+					 *
+					 * In other words, this is how ChanSpy and this module by default work:
+					 * We have audio F to inject onto channels A and B, which are <= bridged =>:
+					 * READ <- A -> WRITE <==> READ <- B -> WRITE
+					 *            F --^                  F --^
+					 *
+					 * So that makes the same audio audible to both channels A and B, but
+					 * in kind of a roundabout way. What if the bridged peer changes at
+					 * some point, for example?
+					 *
+					 * While that method works for 2-party bridges, it doesn't work at all
+					 * for an n-party bridge, so we do the thing that seems obvious to begin with:
+					 * dump the frames onto THIS channel's read queue, and the channels will
+					 * make their way into the bridge like any other audio from this channel,
+					 * and everything just works perfectly, no matter what kind of bridging
+					 * scenario is being used. At that point, we don't even care if we're
+					 * bridged or not, and really, why should we?
+					 *
+					 * In other words, we do this:
+					 * READ <- A -> WRITE <==> READ <- B -> WRITE
+					 *                       F --^       F --^
+					 */
+					ast_audiohook_lock(&mac->whisper_audiohook);
+					fres |= ast_audiohook_write_frame(&mac->whisper_audiohook, AST_AUDIOHOOK_DIRECTION_READ, f);
+					ast_audiohook_unlock(&mac->whisper_audiohook);
+				}
+			}
+			if (fres) {
+				ast_log(LOG_WARNING, "Failed to write to audiohook for %s\n", mac->name);
+				fres = 0;
+			}
+		}
+		AST_RWLIST_TRAVERSE_SAFE_END;
+		AST_RWLIST_UNLOCK(&chanlist);
+		ast_frfree(f);
+	}
+
+	if (!numchans) {
+		ast_debug(1, "Exiting due to all target channels having left the broadcast\n");
+	}
+
+	if (ast_test_flag(flags, OPTION_SPY)) {
+		ast_deactivate_generator(chan);
+	} else {
+		ast_channel_stop_silence_generator(chan, silgen);
+	}
+
+	/* Cleanup any remaining targets */
+	AST_RWLIST_TRAVERSE_SAFE_BEGIN(&chanlist, mac, entry) {
+		AST_RWLIST_REMOVE_CURRENT(entry);
+		multi_autochan_free(mac);
+	}
+	AST_RWLIST_TRAVERSE_SAFE_END;
+
+	ast_channel_clear_flag(chan, AST_FLAG_SPYING);
+	return res;
+}
+
+static int broadcast_exec(struct ast_channel *chan, const char *data)
+{
+	struct ast_flags flags;
+	struct ast_format *write_format;
+	int res = -1;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(options);
+		AST_APP_ARG(channels); /* Channel list last, so we can have multiple */
+	);
+	char *parse = NULL;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Broadcast requires at least one channel\n");
+		return -1;
+	}
+
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (ast_strlen_zero(args.channels)) {
+		ast_log(LOG_WARNING, "Must specify at least one channel for broadcast\n");
+		return -1;
+	}
+	if (args.options) {
+		ast_app_parse_options(spy_opts, &flags, NULL, args.options);
+	} else {
+		ast_clear_flag(&flags, AST_FLAGS_ALL);
+	}
+
+	if (!ast_test_flag(&flags, OPTION_BARGE) && !ast_test_flag(&flags, OPTION_SPY) && !ast_test_flag(&flags, OPTION_WHISPER)) {
+		ast_log(LOG_WARNING, "At least one of the b, s, or w option must be specified (provided options have no effect)\n");
+		return -1;
+	}
+
+	write_format = ao2_bump(ast_channel_writeformat(chan));
+	if (ast_set_write_format(chan, ast_format_slin) < 0) {
+		ast_log(LOG_ERROR, "Failed to set write format to slin.\n");
+		goto cleanup;
+	}
+
+	res = do_broadcast(chan, &flags, args.channels);
+
+	/* Restore previous write format */
+	if (ast_set_write_format(chan, write_format)) {
+		ast_log(LOG_ERROR, "Failed to restore write format for channel %s\n", ast_channel_name(chan));
+	}
+
+cleanup:
+	ao2_ref(write_format, -1);
+	return res;
+}
+
+static int unload_module(void)
+{
+	return ast_unregister_application(app_broadcast);
+}
+
+static int load_module(void)
+{
+	return ast_register_application_xml(app_broadcast, broadcast_exec);
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Channel Audio Broadcasting");

apps/app_chanspy.c

@@ -559,7 +559,7 @@ static int pack_channel_into_message(struct ast_channel *chan, const char *role,
  * \brief Publish the chanspy message over Stasis-Core
  * \param spyer The channel doing the spying
  * \param spyee Who is being spied upon
- * \start start If non-zero, the spying is starting. Otherwise, the spyer is
+ * \param start If non-zero, the spying is starting. Otherwise, the spyer is
  * finishing
  */
 static void publish_chanspy_message(struct ast_channel *spyer,
@@ -845,13 +845,13 @@ static int channel_spy(struct ast_channel *chan, struct ast_autochan *spyee_auto
 	ast_audiohook_unlock(&csth.spy_audiohook);
 	ast_audiohook_destroy(&csth.spy_audiohook);
 
+	ast_verb(2, "Done Spying on channel %s\n", name);
+	publish_chanspy_message(chan, spyee_autochan->chan, 0);
+
 	if (spyee_bridge_autochan) {
 		ast_autochan_destroy(spyee_bridge_autochan);
 	}
 
-	ast_verb(2, "Done Spying on channel %s\n", name);
-	publish_chanspy_message(chan, NULL, 0);
-
 	return running;
 }
 
@@ -899,7 +899,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 	signed char zero_volume = 0;
 	int waitms;
 	int res;
-	int num_spyed_upon = 1;
+	int num_spied_upon = 1;
 	struct ast_channel_iterator *iter = NULL;
 
 	if (ast_test_flag(flags, OPTION_EXIT)) {
@@ -926,7 +926,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 		struct ast_autochan *autochan = NULL, *next_autochan = NULL;
 		struct ast_channel *prev = NULL;
 
-		if (!ast_test_flag(flags, OPTION_QUIET) && num_spyed_upon) {
+		if (!ast_test_flag(flags, OPTION_QUIET) && num_spied_upon) {
 			res = ast_streamfile(chan, "beep", ast_channel_language(chan));
 			if (!res)
 				res = ast_waitstream(chan, "");
@@ -991,7 +991,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 
 		/* reset for the next loop around, unless overridden later */
 		waitms = 100;
-		num_spyed_upon = 0;
+		num_spied_upon = 0;
 
 		for (autochan = next_channel(iter, chan);
 			autochan;
@@ -1146,7 +1146,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 			}
 
 			res = channel_spy(chan, autochan, &volfactor, fd, user_options, flags, exitcontext);
-			num_spyed_upon++;
+			num_spied_upon++;
 
 			if (res == -1) {
 				ast_autochan_destroy(autochan);

apps/app_confbridge.c

@@ -124,10 +124,48 @@
 				</variable>
 			</variablelist>
 		</description>
+		<see-also>
+			<ref type="application">ConfKick</ref>
+			<ref type="function">CONFBRIDGE</ref>
+			<ref type="function">CONFBRIDGE_INFO</ref>
+			<ref type="function">CONFBRIDGE_CHANNELS</ref>
+		</see-also>
+	</application>
+	<application name="ConfKick" language="en_US">
+		<since>
+			<version>16.19.0</version>
+			<version>18.5.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Kicks channel(s) from the requested ConfBridge.
+		</synopsis>
+		<syntax>
+			<parameter name="conference" required="true" />
+			<parameter name="channel">
+				<para>The channel to kick, <literal>all</literal>
+				to kick all users, or <literal>participants</literal>
+				to kick all non-admin participants. Default is all.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Kicks the requested channel(s) from a conference bridge.</para>
+			<variablelist>
+				<variable name="CONFKICKSTATUS">
+					<value name="FAILURE">
+						Could not kick any users with the provided arguments.
+					</value>
+					<value name="SUCCESS">
+						Successfully kicked users from specified conference bridge.
+					</value>
+				</variable>
+			</variablelist>
+		</description>
 		<see-also>
 			<ref type="application">ConfBridge</ref>
 			<ref type="function">CONFBRIDGE</ref>
 			<ref type="function">CONFBRIDGE_INFO</ref>
+			<ref type="function">CONFBRIDGE_CHANNELS</ref>
 		</see-also>
 	</application>
 	<function name="CONFBRIDGE" language="en_US">
@@ -160,23 +198,29 @@
 			<para>---- Example 1 ----</para>
 			<para>In this example the custom user profile set on the channel will
 			automatically be used by the ConfBridge application.</para>
-			<para>exten => 1,1,Answer()</para>
+			<example title="Example 1">
+			exten => 1,1,Answer()
+			</example>
 			<para>; In this example the effect of the following line is</para>
 			<para>; implied:</para>
-			<para>; same => n,Set(CONFBRIDGE(user,template)=default_user)</para>
-			<para>same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes)</para>
-			<para>same => n,Set(CONFBRIDGE(user,startmuted)=yes)</para>
-			<para>same => n,ConfBridge(1) </para>
+			<example title="Example 1b">
+			same => n,Set(CONFBRIDGE(user,template)=default_user)
+			same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes)
+			same => n,Set(CONFBRIDGE(user,startmuted)=yes)
+			same => n,ConfBridge(1)
+			</example>
 			<para>---- Example 2 ----</para>
 			<para>This example shows how to use a predefined user profile in
 			<filename>confbridge.conf</filename> as a template for a dynamic profile.
 			Here we make an admin/marked user out of the <literal>my_user</literal>
 			profile that you define in <filename>confbridge.conf</filename>.</para>
-			<para>exten => 1,1,Answer()</para>
-			<para>same => n,Set(CONFBRIDGE(user,template)=my_user)</para>
-			<para>same => n,Set(CONFBRIDGE(user,admin)=yes)</para>
-			<para>same => n,Set(CONFBRIDGE(user,marked)=yes)</para>
-			<para>same => n,ConfBridge(1)</para>
+			<example title="Example 2">
+			exten => 1,1,Answer()
+			same => n,Set(CONFBRIDGE(user,template)=my_user)
+			same => n,Set(CONFBRIDGE(user,admin)=yes)
+			same => n,Set(CONFBRIDGE(user,marked)=yes)
+			same => n,ConfBridge(1)
+			</example>
 		</description>
 	</function>
 	<function name="CONFBRIDGE_INFO" language="en_US">
@@ -212,6 +256,50 @@
 			<para>This function returns a non-negative integer for valid conference
 			names and an empty string for invalid conference names.</para>
 		</description>
+		<see-also>
+			<ref type="function">CONFBRIDGE_CHANNELS</ref>
+		</see-also>
+	</function>
+	<function name="CONFBRIDGE_CHANNELS" language="en_US">
+		<since>
+			<version>16.26.0</version>
+			<version>18.12.0</version>
+			<version>19.4.0</version>
+		</since>
+		<synopsis>
+			Get a list of channels in a ConfBridge conference.
+		</synopsis>
+		<syntax>
+			<parameter name="type" required="true">
+				<para>What conference information is requested.</para>
+				<enumlist>
+					<enum name="admins">
+						<para>Get the number of admin users in the conference.</para>
+					</enum>
+					<enum name="marked">
+						<para>Get the number of marked users in the conference.</para>
+					</enum>
+					<enum name="parties">
+						<para>Get the number of total users in the conference.</para>
+					</enum>
+					<enum name="active">
+						<para>Get the number of active users in the conference.</para>
+					</enum>
+					<enum name="waiting">
+						<para>Get the number of waiting users in the conference.</para>
+					</enum>
+				</enumlist>
+			</parameter>
+			<parameter name="conf" required="true">
+				<para>The name of the conference being referenced.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>This function returns a comma-separated list of channels in a ConfBridge conference, optionally filtered by a type of participant.</para>
+		</description>
+		<see-also>
+			<ref type="function">CONFBRIDGE_INFO</ref>
+		</see-also>
 	</function>
 	<manager name="ConfbridgeList" language="en_US">
 		<synopsis>
@@ -305,6 +393,37 @@
 				ConfbridgeListRoomsComplete.</para>
 		</description>
 	</manager>
+	<managerEvent language="en_US" name="ConfbridgeListRooms">
+		<managerEventInstance class="EVENT_FLAG_REPORTING">
+			<synopsis>Raised as part of the ConfbridgeListRooms action response list.</synopsis>
+			<syntax>
+				<parameter name="Conference">
+					<para>The name of the Confbridge conference.</para>
+				</parameter>
+				<parameter name="Parties">
+					<para>Number of users in the conference.</para>
+					<para>This includes both active and waiting users.</para>
+				</parameter>
+				<parameter name="Marked">
+					<para>Number of marked users in the conference.</para>
+				</parameter>
+				<parameter name="Locked">
+					<para>Is the conference locked?</para>
+					<enumlist>
+						<enum name="Yes"/>
+						<enum name="No"/>
+					</enumlist>
+				</parameter>
+				<parameter name="Muted">
+					<para>Is the conference muted?</para>
+					<enumlist>
+						<enum name="Yes"/>
+						<enum name="No"/>
+					</enumlist>
+				</parameter>
+			</syntax>
+		</managerEventInstance>
+	</managerEvent>
 	<manager name="ConfbridgeMute" language="en_US">
 		<synopsis>
 			Mute a Confbridge user.
@@ -426,6 +545,7 @@
  */
 
 static const char app[] = "ConfBridge";
+static const char app2[] = "ConfKick";
 
 /*! Number of buckets our conference bridges container can have */
 #define CONFERENCE_BRIDGE_BUCKETS 53
@@ -765,8 +885,8 @@ struct confbridge_conference *conf_find_bridge(const char *conference_name)
  *
  * \note Must be called with the conference locked
  *
- * \retval 1, conference is recording.
- * \retval 0, conference is NOT recording.
+ * \retval 1 conference is recording.
+ * \retval 0 conference is NOT recording.
  */
 static int conf_is_recording(struct confbridge_conference *conference)
 {
@@ -877,7 +997,7 @@ static int conf_start_record(struct confbridge_conference *conference)
 	return 0;
 }
 
-/* \brief Playback the given filename and monitor for any dtmf interrupts.
+/*! \brief Playback the given filename and monitor for any dtmf interrupts.
  *
  * This function is used to playback sound files on a given channel and optionally
  * allow dtmf interrupts to occur.
@@ -891,7 +1011,9 @@ static int conf_start_record(struct confbridge_conference *conference)
  * \param channel Optional channel to play file on if bridge_channel not given
  * \param filename The file name to playback
  *
- * \retval -1 failure during playback, 0 on file was fully played, 1 on dtmf interrupt.
+ * \retval -1 failure during playback.
+ * \retval 0 on file was fully played.
+ * \retval 1 on dtmf interrupt.
  */
 static int play_file(struct ast_bridge_channel *bridge_channel, struct ast_channel *channel,
 		     const char *filename)
@@ -948,7 +1070,8 @@ static int sound_file_exists(const char *filename)
  * \param bridge_channel The bridged channel involved
  *
  * \note if caller is NULL, the announcment will be sent to all participants in the conference.
- * \return Returns 0 on success, -1 if the user hung up
+ * \retval 0 on success.
+ * \retval -1 if the user hung up.
  */
 static int announce_user_count(struct confbridge_conference *conference, struct confbridge_user *user,
 			       struct ast_bridge_channel *bridge_channel)
@@ -998,7 +1121,9 @@ static int announce_user_count(struct confbridge_conference *conference, struct
  * \param user User to play audio prompt to
  * \param filename Prompt to play
  *
- * \return Returns 0 on success, -1 if the user hung up
+ * \retval 0 on success.
+ * \retval -1 if the user hung up.
+ *
  * \note Generally this should be called when the conference is unlocked to avoid blocking
  * the entire conference while the sound is played. But don't unlock the conference bridge
  * in the middle of a state transition.
@@ -1135,8 +1260,6 @@ static void hangup_data_destroy(struct hangup_data *hangup)
  * \brief Destroy a conference bridge
  *
  * \param obj The conference bridge object
- *
- * \return Returns nothing
  */
 static void destroy_conference_bridge(void *obj)
 {
@@ -1273,7 +1396,7 @@ void conf_update_user_mute(struct confbridge_user *user)
 		ast_channel_name(user->chan));
 }
 
-/*
+/*!
  * \internal
  * \brief Mute/unmute a single user.
  */
@@ -1357,8 +1480,6 @@ void conf_moh_start(struct confbridge_user *user)
  * \brief Unsuspend MOH for the conference user.
  *
  * \param user Conference user to unsuspend MOH on.
- *
- * \return Nothing
  */
 static void conf_moh_unsuspend(struct confbridge_user *user)
 {
@@ -1374,8 +1495,6 @@ static void conf_moh_unsuspend(struct confbridge_user *user)
  * \brief Suspend MOH for the conference user.
  *
  * \param user Conference user to suspend MOH on.
- *
- * \return Nothing
  */
 static void conf_moh_suspend(struct confbridge_user *user)
 {
@@ -1650,7 +1769,7 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
 	struct post_join_action *action;
 	int max_members_reached = 0;
 
-	/* We explictly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same */
+	/* We explicitly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same time */
 	ao2_lock(conference_bridges);
 
 	ast_debug(1, "Trying to find conference bridge '%s'\n", conference_name);
@@ -1716,7 +1835,6 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
 			ast_bridge_set_talker_src_video_mode(conference->bridge);
 		} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_VIDEO_SRC_SFU)) {
 			ast_bridge_set_sfu_video_mode(conference->bridge);
-			ast_bridge_set_video_update_discard(conference->bridge, conference->b_profile.video_update_discard);
 			ast_bridge_set_remb_send_interval(conference->bridge, conference->b_profile.remb_send_interval);
 			if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_AVERAGE)) {
 				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_AVERAGE);
@@ -1730,9 +1848,15 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
 				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_LOWEST_ALL);
 			} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST_ALL)) {
 				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_HIGHEST_ALL);
+			} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_FORCE)) {
+				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_FORCE);
+				ast_bridge_set_remb_estimated_bitrate(conference->bridge, conference->b_profile.remb_estimated_bitrate);
 			}
 		}
 
+		/* Always set the minimum interval between video updates, to avoid infinite video updates. */
+		ast_bridge_set_video_update_discard(conference->bridge, conference->b_profile.video_update_discard);
+
 		if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_ENABLE_EVENTS)) {
 			ast_bridge_set_send_sdp_label(conference->bridge, 1);
 		}
@@ -2532,10 +2656,6 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		AST_APP_ARG(menu_profile_name);
 	);
 
-	if (ast_channel_state(chan) != AST_STATE_UP) {
-		ast_answer(chan);
-	}
-
 	if (ast_bridge_features_init(&user.features)) {
 		pbx_builtin_setvar_helper(chan, "CONFBRIDGE_RESULT", "FAILED");
 		res = -1;
@@ -2585,6 +2705,11 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		goto confbridge_cleanup;
 	}
 
+	/* If channel hasn't been answered already, answer it, unless we're explicitly not supposed to */
+	if ((ast_channel_state(chan) != AST_STATE_UP) && (ast_test_flag(&user.u_profile, USER_OPT_ANSWER_CHANNEL))) {
+		ast_answer(chan);
+	}
+
 	quiet = ast_test_flag(&user.u_profile, USER_OPT_QUIET);
 
 	/* ask for a PIN immediately after finding user profile.  This has to be
@@ -2685,17 +2810,24 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		ast_autoservice_stop(chan);
 	}
 
-	/* Play the Join sound to both the conference and the user entering. */
 	if (!quiet) {
 		const char *join_sound = conf_get_sound(CONF_SOUND_JOIN, conference->b_profile.sounds);
 
-		if (strcmp(conference->b_profile.language, ast_channel_language(chan))) {
-			ast_stream_and_wait(chan, join_sound, "");
+		/* if hear_own_join_sound is enabled play the Join sound to everyone */
+		if (ast_test_flag(&user.u_profile, USER_OPT_HEAR_OWN_JOIN_SOUND) ) {
+			if (strcmp(conference->b_profile.language, ast_channel_language(chan))) {
+				ast_stream_and_wait(chan, join_sound, "");
+				ast_autoservice_start(chan);
+				play_sound_file(conference, join_sound);
+				ast_autoservice_stop(chan);
+			} else {
+				async_play_sound_file(conference, join_sound, chan);
+			}
+		/* if hear_own_join_sound is disabled only play the Join sound to just the conference */
+		} else {
 			ast_autoservice_start(chan);
 			play_sound_file(conference, join_sound);
 			ast_autoservice_stop(chan);
-		} else {
-			async_play_sound_file(conference, join_sound, chan);
 		}
 	}
 
@@ -3458,7 +3590,7 @@ static char *handle_cli_confbridge_list(struct ast_cli_entry *e, int cmd, struct
 	return CLI_SHOWUSAGE;
 }
 
-/* \internal
+/*! \internal
  * \brief finds a conference by name and locks/unlocks.
  *
  * \retval 0 success
@@ -3482,7 +3614,7 @@ static int generic_lock_unlock_helper(int lock, const char *conference_name)
 	return res;
 }
 
-/* \internal
+/*! \internal
  * \brief finds a conference user by channel name and mutes/unmutes them.
  *
  * \retval 0 success
@@ -3782,6 +3914,90 @@ static struct ast_custom_function confbridge_info_function = {
 	.read = func_confbridge_info,
 };
 
+static int func_confbridge_channels(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+	char *parse, *outbuf;
+	struct confbridge_conference *conference;
+	struct confbridge_user *user;
+	int bytes, count = 0;
+	size_t outlen;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(type);
+		AST_APP_ARG(confno);
+	);
+
+	/* parse all the required arguments and make sure they exist. */
+	if (ast_strlen_zero(data)) {
+		return -1;
+	}
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+	if (ast_strlen_zero(args.confno) || ast_strlen_zero(args.type)) {
+		ast_log(LOG_WARNING, "Usage: %s(category,confno)", cmd);
+		return -1;
+	}
+	conference = ao2_find(conference_bridges, args.confno, OBJ_KEY);
+	if (!conference) {
+		ast_debug(1, "No such conference: %s\n", args.confno);
+		return -1;
+	}
+
+	outbuf = buf;
+	outlen = len;
+
+	ao2_lock(conference);
+	if (!strcasecmp(args.type, "parties")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+		AST_LIST_TRAVERSE(&conference->waiting_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+	} else if (!strcasecmp(args.type, "active")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+	} else if (!strcasecmp(args.type, "waiting")) {
+		AST_LIST_TRAVERSE(&conference->waiting_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+	} else if (!strcasecmp(args.type, "admins")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			if (ast_test_flag(&user->u_profile, USER_OPT_ADMIN)) {
+				bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+				outbuf += bytes;
+				outlen -= bytes;
+			}
+		}
+	} else if (!strcasecmp(args.type, "marked")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			if (ast_test_flag(&user->u_profile, USER_OPT_MARKEDUSER)) {
+				bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+				outbuf += bytes;
+				outlen -= bytes;
+			}
+		}
+	} else {
+		ast_log(LOG_ERROR, "Invalid keyword '%s' passed to %s.\n", args.type, cmd);
+	}
+	ao2_unlock(conference);
+	ao2_ref(conference, -1);
+	return 0;
+}
+
+static struct ast_custom_function confbridge_channels_function = {
+	.name = "CONFBRIDGE_CHANNELS",
+	.read = func_confbridge_channels,
+};
+
 static int action_confbridgelist_item(struct mansession *s, const char *id_text, struct confbridge_conference *conference, struct confbridge_user *user, int waiting)
 {
 	struct ast_channel_snapshot *snapshot;
@@ -3806,6 +4022,7 @@ static int action_confbridgelist_item(struct mansession *s, const char *id_text,
 		"MarkedUser: %s\r\n"
 		"WaitMarked: %s\r\n"
 		"EndMarked: %s\r\n"
+		"EndMarkedAny: %s\r\n"
 		"Waiting: %s\r\n"
 		"Muted: %s\r\n"
 		"Talking: %s\r\n"
@@ -3818,6 +4035,7 @@ static int action_confbridgelist_item(struct mansession *s, const char *id_text,
 		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_MARKEDUSER)),
 		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_WAITMARKED)),
 		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_ENDMARKED)),
+		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_ENDMARKEDANY)),
 		AST_YESNO(waiting),
 		AST_YESNO(user->muted),
 		AST_YESNO(user->talking),
@@ -4217,6 +4435,53 @@ static int func_confbridge_info(struct ast_channel *chan, const char *cmd, char
 	return 0;
 }
 
+static int confkick_exec(struct ast_channel *chan, const char *data)
+{
+	char *parse;
+	struct confbridge_conference *conference;
+	int not_found;
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(confbridge);
+		AST_APP_ARG(channel);
+	);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "No conference bridge specified.\n");
+		pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "FAILURE");
+		return 0;
+	}
+
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	conference = ao2_find(conference_bridges, args.confbridge, OBJ_KEY);
+	if (!conference) {
+		ast_log(LOG_WARNING, "No conference bridge named '%s' found!\n", args.confbridge);
+		pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "FAILURE");
+		return 0;
+	}
+	if (ast_strlen_zero(args.channel)) {
+		not_found = kick_conference_participant(conference, "all");
+	} else {
+		not_found = kick_conference_participant(conference, args.channel);
+	}
+
+	ao2_ref(conference, -1);
+	if (not_found) {
+		if (ast_strlen_zero(args.channel) || !strcasecmp("all", args.channel) || !strcasecmp("participants", args.channel)) {
+			ast_log(LOG_WARNING, "No participants found in conference bridge '%s'!\n", args.confbridge);
+		} else {
+			ast_log(LOG_WARNING, "No participant named '%s' found in conference bridge '%s'!\n", args.channel, args.confbridge);
+		}
+		pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "FAILURE");
+		return 0;
+	}
+	ast_debug(1, "Kicked '%s' out of conference '%s'\n", args.channel, args.confbridge);
+	pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "SUCCESS");
+	return 0;
+}
+
 void conf_add_user_active(struct confbridge_conference *conference, struct confbridge_user *user)
 {
 	AST_LIST_INSERT_TAIL(&conference->active_list, user, list);
@@ -4272,8 +4537,6 @@ void conf_remove_user_waiting(struct confbridge_conference *conference, struct c
  * \since 12.0.0
  *
  * \param tech What to unregister.
- *
- * \return Nothing
  */
 static void unregister_channel_tech(struct ast_channel_tech *tech)
 {
@@ -4310,9 +4573,11 @@ static int register_channel_tech(struct ast_channel_tech *tech)
 static int unload_module(void)
 {
 	ast_unregister_application(app);
+	ast_unregister_application(app2);
 
 	ast_custom_function_unregister(&confbridge_function);
 	ast_custom_function_unregister(&confbridge_info_function);
+	ast_custom_function_unregister(&confbridge_channels_function);
 
 	ast_cli_unregister_multiple(cli_confbridge, ARRAY_LEN(cli_confbridge));
 
@@ -4380,9 +4645,11 @@ static int load_module(void)
 	res |= manager_confbridge_init();
 
 	res |= ast_register_application_xml(app, confbridge_exec);
+	res |= ast_register_application_xml(app2, confkick_exec);
 
 	res |= ast_custom_function_register_escalating(&confbridge_function, AST_CFE_WRITE);
 	res |= ast_custom_function_register(&confbridge_info_function);
+	res |= ast_custom_function_register(&confbridge_channels_function);
 
 	res |= ast_cli_register_multiple(cli_confbridge, ARRAY_LEN(cli_confbridge));
 

apps/app_dahdiras.c

@@ -28,6 +28,8 @@
 /*** MODULEINFO
 	<depend>dahdi</depend>
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_dial.c

@@ -86,6 +86,7 @@
 					<para>If you need more than one enter them as
 					Technology2/Resource2&amp;Technology3/Resource3&amp;.....</para>
 				</argument>
+				<xi:include xpointer="xpointer(/docs/info[@name='Dial_Resource'])" />
 			</parameter>
 			<parameter name="timeout" required="false">
 				<para>Specifies the number of seconds we attempt to dial the specified devices.</para>
@@ -93,11 +94,17 @@
 			</parameter>
 			<parameter name="options" required="false">
 				<optionlist>
-				<option name="A">
-					<argument name="x" required="true">
+				<option name="A" argsep=":">
+					<argument name="x">
 						<para>The file to play to the called party</para>
 					</argument>
-					<para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
+					<argument name="y">
+						<para>The file to play to the calling party</para>
+					</argument>
+					<para>Play an announcement to the called and/or calling parties, where <replaceable>x</replaceable>
+					is the prompt to be played to the called party and <replaceable>y</replaceable> is the prompt
+					to be played to the caller. The files may be different and will be played to each party
+					simultaneously.</para>
 				</option>
 				<option name="a">
 					<para>Immediately answer the calling channel when the called channel answers in
@@ -150,6 +157,10 @@
 					<argument name="called" />
 					<argument name="calling" />
 					<argument name="progress" />
+					<argument name="mfprogress" />
+					<argument name="mfwink" />
+					<argument name="sfprogress" />
+					<argument name="sfwink" />
 					<para>Send the specified DTMF strings <emphasis>after</emphasis> the called
 					party has answered, but before the call gets bridged.  The
 					<replaceable>called</replaceable> DTMF string is sent to the called party, and the
@@ -157,6 +168,20 @@
 					can be used alone.  If <replaceable>progress</replaceable> is specified, its DTMF is sent
 					to the called party immediately after receiving a <literal>PROGRESS</literal> message.</para>
 					<para>See <literal>SendDTMF</literal> for valid digits.</para>
+					<para>If <replaceable>mfprogress</replaceable> is specified, its MF is sent
+					to the called party immediately after receiving a <literal>PROGRESS</literal> message.
+					If <replaceable>mfwink</replaceable> is specified, its MF is sent
+					to the called party immediately after receiving a <literal>WINK</literal> message.</para>
+					<para>See <literal>SendMF</literal> for valid digits.</para>
+					<para>If <replaceable>sfprogress</replaceable> is specified, its SF is sent
+					to the called party immediately after receiving a <literal>PROGRESS</literal> message.
+					If <replaceable>sfwink</replaceable> is specified, its SF is sent
+					to the called party immediately after receiving a <literal>WINK</literal> message.</para>
+					<para>See <literal>SendSF</literal> for valid digits.</para>
+				</option>
+				<option name="E">
+					<para>Enable echoing of sent MF or SF digits back to caller (e.g. "hearpulsing").
+					Used in conjunction with the D option.</para>
 				</option>
 				<option name="e">
 					<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
@@ -347,7 +372,7 @@
 					</argument>
 					<para>Enables <emphasis>operator services</emphasis> mode.  This option only
 					works when bridging a DAHDI channel to another DAHDI channel
-					only. if specified on non-DAHDI interfaces, it will be ignored.
+					only. If specified on non-DAHDI interfaces, it will be ignored.
 					When the destination answers (presumably an operator services
 					station), the originator no longer has control of their line.
 					They may hang up, but the switch will not release their line
@@ -493,7 +518,6 @@
 			answered, if it has not already been answered. These two channels will then
 			be active in a bridged call. All other channels that were requested will then
 			be hung up.</para>
-
 			<para>Unless there is a timeout specified, the Dial application will wait
 			indefinitely until one of the called channels answers, the user hangs up, or
 			if all of the called channels are busy or unavailable. Dialplan execution will
@@ -506,7 +530,6 @@
 			If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
 			application will be put into that group (as in <literal>Set(GROUP()=...</literal>). Unlike <variable>OUTBOUND_GROUP</variable>,
 			however, the variable will be unset after use.</para>
-
 			<example title="Dial with 30 second timeout">
 			 same => n,Dial(PJSIP/alice,30)
 			</example>
@@ -528,28 +551,22 @@
 			</example>
 			<example title="Dial with pre-dial subroutines">
 			[default]
-
 			exten => callee_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 			 same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
 			 same => n,Return()
-
 			exten => called_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 			 same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
 			 same => n,Return()
-
 			exten => _X.,1,NoOp()
 			 same => n,Dial(PJSIP/alice,,b(default^called_channel^1(my_gosub_arg1^my_gosub_arg2))B(default^callee_channel^1(my_gosub_arg1^my_gosub_arg2)))
 			 same => n,Hangup()
 			</example>
 			<example title="Dial with post-answer subroutine executed on outbound channel">
 			[my_gosub_routine]
-
 			exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 			 same => n,Playback(hello)
 			 same => n,Return()
-
 			[default]
-
 			exten => _X.,1,NoOp()
 			 same => n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
 			 same => n,Hangup()
@@ -597,12 +614,31 @@
 				</variable>
 				<variable name="DIALSTATUS">
 					<para>This is the status of the call</para>
-					<value name="CHANUNAVAIL" />
-					<value name="CONGESTION" />
-					<value name="NOANSWER" />
-					<value name="BUSY" />
-					<value name="ANSWER" />
-					<value name="CANCEL" />
+					<value name="CHANUNAVAIL">
+						Either the dialed peer exists but is not currently reachable, e.g.
+						endpoint is not registered, or an attempt was made to call a
+						nonexistent location, e.g. nonexistent DNS hostname.
+					</value>
+					<value name="CONGESTION">
+						Channel or switching congestion occured when routing the call.
+						This can occur if there is a slow or no response from the remote end.
+					</value>
+					<value name="NOANSWER">
+						Called party did not answer.
+					</value>
+					<value name="BUSY">
+						The called party was busy or indicated a busy status.
+						Note that some SIP devices will respond with 486 Busy if their Do Not Disturb
+						modes are active. In this case, you can use DEVICE_STATUS to check if the
+						endpoint is actually in use, if needed.
+					</value>
+					<value name="ANSWER">
+						The call was answered.
+						Any other result implicitly indicates the call was not answered.
+					</value>
+					<value name="CANCEL">
+						Dial was cancelled before call was answered or reached some other terminating event.
+					</value>
 					<value name="DONTCALL">
 						For the Privacy and Screening Modes.
 						Will be set if the called party chooses to send the calling party to the 'Go Away' script.
@@ -611,7 +647,9 @@
 						For the Privacy and Screening Modes.
 						Will be set if the called party chooses to send the calling party to the 'torture' script.
 					</value>
-					<value name="INVALIDARGS" />
+					<value name="INVALIDARGS">
+						Dial failed due to invalid syntax.
+					</value>
 				</variable>
 			</variablelist>
 		</description>
@@ -711,6 +749,7 @@ enum {
 #define OPT_PREDIAL_CALLER   (1LLU << 42)
 #define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
 #define OPT_HANGUPCAUSE      (1LLU << 44)
+#define OPT_HEARPULSING      (1LLU << 45)
 
 enum {
 	OPT_ARG_ANNOUNCE = 0,
@@ -746,6 +785,7 @@ AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
 	AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
 	AST_APP_OPTION('d', OPT_DTMF_EXIT),
 	AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
+	AST_APP_OPTION('E', OPT_HEARPULSING),
 	AST_APP_OPTION('e', OPT_PEER_H),
 	AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
 	AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
@@ -924,7 +964,7 @@ static const char *get_cid_name(char *name, int namelen, struct ast_channel *cha
  * XXX this code is highly suspicious, as it essentially overwrites
  * the outgoing channel without properly deleting it.
  *
- * \todo eventually this function should be intergrated into and replaced by ast_call_forward()
+ * \todo eventually this function should be integrated into and replaced by ast_call_forward()
  */
 static void do_forward(struct chanlist *o, struct cause_args *num,
 	struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
@@ -1140,6 +1180,7 @@ struct privacy_args {
 	char privcid[256];
 	char privintro[1024];
 	char status[256];
+	int canceled;
 };
 
 static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
@@ -1161,8 +1202,6 @@ static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out
  * \param chan Channel to get connected line updated.
  * \param peer Channel providing connected line information.
  * \param is_caller Non-zero if chan is the calling channel.
- *
- * \return Nothing
  */
 static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
 {
@@ -1203,8 +1242,12 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 	char *opt_args[],
 	struct privacy_args *pa,
 	const struct cause_args *num_in, int *result, char *dtmf_progress,
+	char *mf_progress, char *mf_wink,
+	char *sf_progress, char *sf_wink,
+	const int hearpulsing,
 	const int ignore_cc,
-	struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
+	struct ast_party_id *forced_clid, struct ast_party_id *stored_clid,
+	struct ast_bridge_config *config)
 {
 	struct cause_args num = *num_in;
 	int prestart = num.busy + num.congestion + num.nochan;
@@ -1221,9 +1264,9 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 	int cc_frame_received = 0;
 	int num_ringing = 0;
 	int sent_ring = 0;
-	int sent_progress = 0;
+	int sent_progress = 0, sent_wink = 0;
 	struct timeval start = ast_tvnow();
-	SCOPE_TRACE(1, "%s\n", ast_channel_name(in));
+	SCOPE_ENTER(3, "%s\n", ast_channel_name(in));
 
 	if (single) {
 		/* Turn off hold music, etc */
@@ -1239,7 +1282,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				*to = -1;
 				strcpy(pa->status, "CONGESTION");
 				ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
-				return NULL;
+				SCOPE_EXIT_RTN_VALUE(NULL, "%s: can't be made compat with %s\n",
+					ast_channel_name(in), ast_channel_name(outgoing->chan));
 			}
 		}
 
@@ -1281,7 +1325,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 			if (is_cc_recall) {
 				ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
 			}
-			return NULL;
+			SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));
 		}
 		winner = ast_waitfor_n(watchers, pos, to);
 		AST_LIST_TRAVERSE(out_chans, o, node) {
@@ -1390,7 +1434,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				case AST_CONTROL_ANSWER:
 					/* This is our guy if someone answered. */
 					if (!peer) {
-						ast_trace(1, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
+						ast_trace(-1, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
 						ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
 						if (o->orig_chan_name
 							&& strcmp(o->orig_chan_name, ast_channel_name(c))) {
@@ -1418,6 +1462,18 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 							}
 						}
 						peer = c;
+						/* Answer can optionally include a topology */
+						if (f->subclass.topology) {
+							/*
+							 * We need to bump the refcount on the topology to prevent it
+							 * from being cleaned up when the frame is cleaned up.
+							 */
+							config->answer_topology = ao2_bump(f->subclass.topology);
+							ast_trace(-1, "%s Found topology in frame: %p %p %s\n",
+								ast_channel_name(peer), f, config->answer_topology,
+								ast_str_tmp(256, ast_stream_topology_to_str(config->answer_topology, &STR_TMP)));
+						}
+
 						/* Inform everyone else that they've been canceled.
 						 * The dial end event for the peer will be sent out after
 						 * other Dial options have been handled.
@@ -1545,15 +1601,56 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 						ast_channel_stage_snapshot_done(in);
 						ast_channel_unlock(in);
 						sent_progress = 1;
-					}
-					if (!ast_strlen_zero(dtmf_progress)) {
-						ast_verb(3,
-							"Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n",
-							dtmf_progress);
-						ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
+
+						if (!ast_strlen_zero(mf_progress)) {
+							ast_verb(3,
+								"Sending MF '%s' to %s as result of "
+								"receiving a PROGRESS message.\n",
+								mf_progress, hearpulsing ? "parties" : "called party");
+							ast_mf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
+						}
+						if (!ast_strlen_zero(sf_progress)) {
+							ast_verb(3,
+								"Sending SF '%s' to %s as result of "
+								"receiving a PROGRESS message.\n",
+								sf_progress, (hearpulsing ? "parties" : "called party"));
+							ast_sf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), sf_progress, 0, 0);
+						}
+						if (!ast_strlen_zero(dtmf_progress)) {
+							ast_verb(3,
+								"Sending DTMF '%s' to the called party as result of "
+								"receiving a PROGRESS message.\n",
+								dtmf_progress);
+							ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
+						}
 					}
 					ast_channel_publish_dial(in, c, NULL, "PROGRESS");
 					break;
+				case AST_CONTROL_WINK:
+					ast_verb(3, "%s winked, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
+					if (!sent_wink) {
+						sent_wink = 1;
+						if (!ast_strlen_zero(mf_wink)) {
+							ast_verb(3,
+								"Sending MF '%s' to %s as result of "
+								"receiving a WINK message.\n",
+								mf_wink, (hearpulsing ? "parties" : "called party"));
+							ast_mf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
+						}
+						if (!ast_strlen_zero(sf_wink)) {
+							ast_verb(3,
+								"Sending SF '%s' to %s as result of "
+								"receiving a WINK message.\n",
+								sf_wink, (hearpulsing ? "parties" : "called party"));
+							ast_sf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), sf_wink, 0, 0);
+						}
+					}
+					ast_indicate(in, AST_CONTROL_WINK);
+					break;
 				case AST_CONTROL_VIDUPDATE:
 				case AST_CONTROL_SRCUPDATE:
 				case AST_CONTROL_SRCCHANGE:
@@ -1698,6 +1795,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				/* Got hung up */
 				*to = -1;
 				strcpy(pa->status, "CANCEL");
+				pa->canceled = 1;
 				publish_dial_end_event(in, out_chans, NULL, pa->status);
 				if (f) {
 					if (f->data.uint32) {
@@ -1708,7 +1806,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				if (is_cc_recall) {
 					ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
 				}
-				return NULL;
+				SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller hung up\n", ast_channel_name(in));
 			}
 
 			/* now f is guaranteed non-NULL */
@@ -1722,13 +1820,15 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 						*to = 0;
 						*result = f->subclass.integer;
 						strcpy(pa->status, "CANCEL");
+						pa->canceled = 1;
 						publish_dial_end_event(in, out_chans, NULL, pa->status);
 						ast_frfree(f);
 						ast_channel_unlock(in);
 						if (is_cc_recall) {
 							ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
 						}
-						return NULL;
+						SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller pressed %c to end call\n",
+							ast_channel_name(in), f->subclass.integer);
 					}
 					ast_channel_unlock(in);
 				}
@@ -1738,12 +1838,14 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 					ast_verb(3, "User requested call disconnect.\n");
 					*to = 0;
 					strcpy(pa->status, "CANCEL");
+					pa->canceled = 1;
 					publish_dial_end_event(in, out_chans, NULL, pa->status);
 					ast_frfree(f);
 					if (is_cc_recall) {
 						ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
 					}
-					return NULL;
+					SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller requested disconnect\n",
+						ast_channel_name(in));
 				}
 			}
 
@@ -1791,6 +1893,10 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 						ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
 						ast_indicate(o->chan, AST_CONTROL_UNHOLD);
 						break;
+					case AST_CONTROL_FLASH:
+						ast_verb(3, "Hook flash on %s\n", ast_channel_name(o->chan));
+						ast_indicate(o->chan, AST_CONTROL_FLASH);
+						break;
 					case AST_CONTROL_VIDUPDATE:
 					case AST_CONTROL_SRCUPDATE:
 					case AST_CONTROL_SRCCHANGE:
@@ -1849,7 +1955,8 @@ skip_frame:;
 	if (is_cc_recall) {
 		ast_cc_completed(in, "Recall completed!");
 	}
-	return peer;
+	SCOPE_EXIT_RTN_VALUE(peer, "%s: %s%s\n", ast_channel_name(in),
+		peer ? "Answered by " : "No answer", peer ? ast_channel_name(peer) : "");
 }
 
 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
@@ -2105,9 +2212,7 @@ static int setup_privacy_args(struct privacy_args *pa,
 			/* the file doesn't exist yet. Let the caller submit his
 			   vocal intro for posterity */
 			/* priv-recordintro script:
-
 			   "At the tone, please say your name:"
-
 			*/
 			int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
 			ast_answer(chan);
@@ -2185,8 +2290,6 @@ static int dial_handle_playtones(struct ast_channel *chan, const char *data)
  * \param peer Peer channel for bridge.
  * \param opts Dialing option flags.
  * \param opt_args Dialing option argument strings.
- *
- * \return Nothing
  */
 static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
 {
@@ -2217,18 +2320,21 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
 	struct chanlist *outgoing;
 	struct chanlist *tmp;
-	struct ast_channel *peer;
+	struct ast_channel *peer = NULL;
 	int to; /* timeout */
 	struct cause_args num = { chan, 0, 0, 0 };
-	int cause;
+	int cause, hanguptreecause = -1;
 
 	struct ast_bridge_config config = { { 0, } };
 	struct timeval calldurationlimit = { 0, };
-	char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
+	char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress = NULL;
+	char *mf_progress = NULL, *mf_wink = NULL;
+	char *sf_progress = NULL, *sf_wink = NULL;
 	struct privacy_args pa = {
 		.sentringing = 0,
 		.privdb_val = 0,
 		.status = "INVALIDARGS",
+		.canceled = 0,
 	};
 	int sentringing = 0, moh = 0;
 	const char *outbound_group = NULL;
@@ -2271,7 +2377,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	 */
 	struct ast_party_caller caller;
 	int max_forwards;
-	SCOPE_TRACE(1, "%s Data: %s\n", ast_channel_name(chan), data);
+	SCOPE_ENTER(1, "%s: Data: %s\n", ast_channel_name(chan), data);
 
 	/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
 	ast_channel_lock(chan);
@@ -2295,7 +2401,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
 				ast_channel_name(chan));
 		pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
-		return -1;
+		SCOPE_EXIT_RTN_VALUE(-1, "%s: Max forwards exceeded\n", ast_channel_name(chan));
 	}
 
 	if (ast_check_hangup_locked(chan)) {
@@ -2313,7 +2419,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		 */
 		ast_verb(3, "Caller hung up before dial.\n");
 		pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
-		return -1;
+		SCOPE_EXIT_RTN_VALUE(-1, "%s: Caller hung up before dial\n", ast_channel_name(chan));
 	}
 
 	parse = ast_strdupa(data ?: "");
@@ -2359,9 +2465,13 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	}
 
 	if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
-		dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
-		dtmfcalled = strsep(&dtmf_progress, ":");
-		dtmfcalling = strsep(&dtmf_progress, ":");
+		sf_wink = opt_args[OPT_ARG_SENDDTMF];
+		dtmfcalled = strsep(&sf_wink, ":");
+		dtmfcalling = strsep(&sf_wink, ":");
+		dtmf_progress = strsep(&sf_wink, ":");
+		mf_progress = strsep(&sf_wink, ":");
+		mf_wink = strsep(&sf_wink, ":");
+		sf_progress = strsep(&sf_wink, ":");
 	}
 
 	if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
@@ -2502,9 +2612,11 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		struct ast_channel *tc; /* channel for this destination */
 		char *number;
 		char *tech;
+		int i;
 		size_t tech_len;
 		size_t number_len;
 		struct ast_stream_topology *topology;
+		struct ast_stream *stream;
 
 		cur = ast_strip(cur);
 		if (ast_strlen_zero(cur)) {
@@ -2570,13 +2682,29 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 
 		ast_channel_unlock(chan);
 
+		for (i = 0; i < ast_stream_topology_get_count(topology); ++i) {
+			stream = ast_stream_topology_get_stream(topology, i);
+			/* For both recvonly and sendonly the stream state reflects our state, that is we
+			 * are receiving only and we are sending only. Since we are requesting a
+			 * channel for the peer, we need to swap this to reflect what we will be doing.
+			 * That is, if we are receiving from Alice then we want to be sending to Bob,
+			 * so swap recvonly to sendonly and vice versa.
+			 */
+			if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
+				ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
+			} else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
+				ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
+			}
+		}
+
 		tc = ast_request_with_stream_topology(tmp->tech, topology, NULL, chan, tmp->number, &cause);
 
 		ast_stream_topology_free(topology);
 
 		if (!tc) {
 			/* If we can't, just go on to the next call */
-			ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
+			/* Failure doesn't necessarily mean user error. DAHDI channels could be busy. */
+			ast_log(LOG_NOTICE, "Unable to create channel of type '%s' (cause %d - %s)\n",
 				tmp->tech, cause, ast_cause2str(cause));
 			handle_cause(cause, &num);
 			if (!rest) {
@@ -2714,7 +2842,9 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
 	}
 
-	if (AST_LIST_EMPTY(&out_chans)) {
+	/* As long as we attempted to dial valid peers, don't throw a warning. */
+	/* If a DAHDI peer is busy, out_chans will be empty so checking list size is misleading. */
+	if (!num_dialed) {
 		ast_verb(3, "No devices or endpoints to dial (technology/resource)\n");
 		if (continue_exec) {
 			/* There is no point in having RetryDial try again */
@@ -2838,7 +2968,9 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	}
 
 	peer = wait_for_answer(chan, &out_chans, &to, peerflags, opt_args, &pa, &num, &result,
-		dtmf_progress, ignore_cc, &forced_clid, &stored_clid);
+		dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink,
+		(ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
+		ignore_cc, &forced_clid, &stored_clid, &config);
 
 	if (!peer) {
 		if (result) {
@@ -2922,33 +3054,71 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 			int digit = 0;
 			struct ast_channel *chans[2];
 			struct ast_channel *active_chan;
+			char *calledfile = NULL, *callerfile = NULL;
+			int calledstream = 0, callerstream = 0;
 
 			chans[0] = chan;
 			chans[1] = peer;
 
-			/* we need to stream the announcement to the called party when the OPT_ARG_ANNOUNCE (-A) is setted */
+			/* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
+			callerfile = opt_args[OPT_ARG_ANNOUNCE];
+			calledfile = strsep(&callerfile, ":");
 
-			/* stream the file */
-			res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], ast_channel_language(peer));
-			if (res) {
-				res = 0;
-				ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
+			/* stream the file(s) */
+			if (!ast_strlen_zero(calledfile)) {
+				res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
+				if (res) {
+					res = 0;
+					ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
+				} else {
+					calledstream = 1;
+				}
+			}
+			if (!ast_strlen_zero(callerfile)) {
+				res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
+				if (res) {
+					res = 0;
+					ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
+				} else {
+					callerstream = 1;
+				}
 			}
 
+			/* can't use ast_waitstream, because we're streaming two files at once, and can't block
+				We'll need to handle both channels at once. */
+
 			ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
-			while (ast_channel_stream(peer)) {
-				int ms;
+			while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
+				int mspeer, mschan;
 
-				ms = ast_sched_wait(ast_channel_sched(peer));
+				mspeer = ast_sched_wait(ast_channel_sched(peer));
+				mschan = ast_sched_wait(ast_channel_sched(chan));
 
-				if (ms < 0 && !ast_channel_timingfunc(peer)) {
-					ast_stopstream(peer);
+				if (calledstream) {
+					if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
+						ast_stopstream(peer);
+						calledstream = 0;
+					}
+				}
+				if (callerstream) {
+					if (mschan < 0 && !ast_channel_timingfunc(chan)) {
+						ast_stopstream(chan);
+						callerstream = 0;
+					}
+				}
+
+				if (!calledstream && !callerstream) {
 					break;
 				}
-				if (ms < 0)
-					ms = 1000;
 
-				active_chan = ast_waitfor_n(chans, 2, &ms);
+				if (mspeer < 0)
+					mspeer = 1000;
+
+				if (mschan < 0)
+					mschan = 1000;
+
+				/* wait for the lowest maximum of the two */
+				active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
 				if (active_chan) {
 					struct ast_channel *other_chan;
 					struct ast_frame *fr = ast_read(active_chan);
@@ -2998,6 +3168,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 					ast_frfree(fr);
 				}
 				ast_sched_runq(ast_channel_sched(peer));
+				ast_sched_runq(ast_channel_sched(chan));
 			}
 			ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
 		}
@@ -3267,6 +3438,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 				ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
 			}
 			setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
+
 			res = ast_bridge_call(chan, peer, &config);
 		}
 	}
@@ -3289,11 +3461,16 @@ out:
 	}
 
 	ast_channel_early_bridge(chan, NULL);
-	 /* forward 'answered elsewhere' if we received it */
-	hanguptree(&out_chans, NULL,
-		ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE
-		|| ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)
-		? AST_CAUSE_ANSWERED_ELSEWHERE : -1);
+	/* forward 'answered elsewhere' if we received it */
+	if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)) {
+		hanguptreecause = AST_CAUSE_ANSWERED_ELSEWHERE;
+	} else if (pa.canceled) { /* Caller canceled */
+		if (ast_channel_hangupcause(chan))
+			hanguptreecause = ast_channel_hangupcause(chan);
+		else
+			hanguptreecause = AST_CAUSE_NORMAL_CLEARING;
+	}
+	hanguptree(&out_chans, NULL, hanguptreecause);
 	pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
 	ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
 
@@ -3304,6 +3481,18 @@ out:
 	}
 
 done:
+	if (config.answer_topology) {
+		ast_trace(2, "%s Cleaning up topology: %p %s\n",
+			peer ? ast_channel_name(peer) : "<no channel>", &config.answer_topology,
+			ast_str_tmp(256, ast_stream_topology_to_str(config.answer_topology, &STR_TMP)));
+
+		/*
+		 * At this point, the channel driver that answered should have bumped the
+		 * topology refcount for itself.  Here we're cleaning up the reference we added
+		 * in wait_for_answer().
+		 */
+		ast_stream_topology_free(config.answer_topology);
+	}
 	if (config.warning_sound) {
 		ast_free((char *)config.warning_sound);
 	}
@@ -3314,7 +3503,7 @@ done:
 		ast_free((char *)config.start_sound);
 	}
 	ast_ignore_cc(chan);
-	return res;
+	SCOPE_EXIT_RTN_VALUE(res, "%s: Done\n", ast_channel_name(chan));
 }
 
 static int dial_exec(struct ast_channel *chan, const char *data)

apps/app_directory.c

@@ -103,6 +103,14 @@
 						receiver to their ear while entering DTMF.</para>
 						<argument name="n" required="true" />
 					</option>
+					<option name="c">
+						<para>Load the specified config file instead of voicemail.conf</para>
+						<argument name="filename" required="true" />
+					</option>
+					<option name="s">
+						<para>Skip calling the extension, instead set it in the <variable>DIRECTORY_EXTEN</variable>
+						channel variable.</para>
+					</option>
 				</optionlist>
 				<note><para>Only one of the <replaceable>f</replaceable>, <replaceable>l</replaceable>, or <replaceable>b</replaceable>
 				options may be specified. <emphasis>If more than one is specified</emphasis>, then Directory will act as
@@ -114,12 +122,12 @@
 		<description>
 			<para>This application will present the calling channel with a directory of extensions from which they can search
 			by name. The list of names and corresponding extensions is retrieved from the
-			voicemail configuration file, <filename>voicemail.conf</filename>.</para>
+			voicemail configuration file, <filename>voicemail.conf</filename>, or from the specified filename.</para>
 			<para>This application will immediately exit if one of the following DTMF digits are
 			received and the extension to jump to exists:</para>
 			<para><literal>0</literal> - Jump to the 'o' extension, if it exists.</para>
 			<para><literal>*</literal> - Jump to the 'a' extension, if it exists.</para>
-			<para>This application will set the following channel variable before completion:</para>
+			<para>This application will set the following channel variables before completion:</para>
 			<variablelist>
 				<variable name="DIRECTORY_RESULT">
 					<para>Reason Directory application exited.</para>
@@ -131,6 +139,10 @@
 					<value name="USEREXIT">User exited with '#' during selection</value>
 					<value name="FAILED">The application failed</value>
 				</variable>
+				<variable name="DIRECTORY_EXTEN">
+					<para>If the skip calling option is set this will be set to the selected extension
+					provided one is selected.</para>
+				</variable>
 			</variablelist>
 		</description>
 	</application>
@@ -153,6 +165,8 @@ enum {
 	OPT_PAUSE =           (1 << 5),
 	OPT_NOANSWER =        (1 << 6),
 	OPT_ALIAS =           (1 << 7),
+	OPT_CONFIG_FILE =     (1 << 8),
+	OPT_SKIP =            (1 << 9),
 };
 
 enum {
@@ -160,8 +174,9 @@ enum {
 	OPT_ARG_LASTNAME =    1,
 	OPT_ARG_EITHER =      2,
 	OPT_ARG_PAUSE =       3,
+	OPT_ARG_FILENAME =    4,
 	/* This *must* be the last value in this enum! */
-	OPT_ARG_ARRAY_SIZE =  4,
+	OPT_ARG_ARRAY_SIZE =  5,
 };
 
 struct directory_item {
@@ -183,6 +198,8 @@ AST_APP_OPTIONS(directory_app_options, {
 	AST_APP_OPTION('m', OPT_SELECTFROMMENU),
 	AST_APP_OPTION('n', OPT_NOANSWER),
 	AST_APP_OPTION('a', OPT_ALIAS),
+	AST_APP_OPTION_ARG('c', OPT_CONFIG_FILE, OPT_ARG_FILENAME),
+	AST_APP_OPTION('s', OPT_SKIP),
 });
 
 static int compare(const char *text, const char *template)
@@ -318,6 +335,9 @@ static int select_entry(struct ast_channel *chan, const char *dialcontext, const
 	if (ast_test_flag(flags, OPT_FROMVOICEMAIL)) {
 		/* We still want to set the exten though */
 		ast_channel_exten_set(chan, item->exten);
+	} else if (ast_test_flag(flags, OPT_SKIP)) {
+		/* Skip calling the extension, only set it in the channel variable. */
+		pbx_builtin_setvar_helper(chan, "DIRECTORY_EXTEN", item->exten);
 	} else if (ast_goto_if_exists(chan, S_OR(dialcontext, item->context), item->exten, 1)) {
 		ast_log(LOG_WARNING,
 			"Can't find extension '%s' in context '%s'.  "
@@ -396,7 +416,7 @@ static int select_item_menu(struct ast_channel *chan, struct directory_item **it
 {
 	struct directory_item **block, *item;
 	int i, limit, res = 0;
-	char buf[9];
+	char buf[7+12]; /* INT_MIN has a length of 12 chars */
 
 	/* option p(n): cellphone pause option */
 	select_item_pause(chan, flags, opts);
@@ -458,7 +478,7 @@ static int select_item_menu(struct ast_channel *chan, struct directory_item **it
 
 AST_THREADSTORAGE(commonbuf);
 
-static struct ast_config *realtime_directory(char *context)
+static struct ast_config *realtime_directory(char *context, const char *filename)
 {
 	struct ast_config *cfg;
 	struct ast_config *rtdata = NULL;
@@ -475,14 +495,14 @@ static struct ast_config *realtime_directory(char *context)
 	}
 
 	/* Load flat file config. */
-	cfg = ast_config_load(VOICEMAIL_CONFIG, config_flags);
+	cfg = ast_config_load(filename, config_flags);
 
 	if (!cfg) {
 		/* Loading config failed. */
 		ast_log(LOG_WARNING, "Loading config failed.\n");
 		return NULL;
 	} else if (cfg == CONFIG_STATUS_FILEINVALID) {
-		ast_log(LOG_ERROR, "Config file %s is in an invalid format.  Aborting.\n", VOICEMAIL_CONFIG);
+		ast_log(LOG_ERROR, "Config file %s is in an invalid format.  Aborting.\n", filename);
 		return NULL;
 	}
 
@@ -867,7 +887,9 @@ static int directory_exec(struct ast_channel *chan, const char *data)
 	if (args.options && ast_app_parse_options(directory_app_options, &flags, opts, args.options))
 		return -1;
 
-	if (!(cfg = realtime_directory(args.vmcontext))) {
+	cfg = realtime_directory(args.vmcontext, S_OR(opts[OPT_ARG_FILENAME], VOICEMAIL_CONFIG));
+
+	if (!cfg) {
 		ast_log(LOG_ERROR, "Unable to read the configuration data!\n");
 		return -1;
 	}

apps/app_disa.c

@@ -361,7 +361,6 @@ static int disa_exec(struct ast_channel *chan, const char *data)
 
 	if (k == 3) {
 		int recheck = 0;
-		struct ast_app *app_reset_cdr;
 
 		if (!ast_exists_extension(chan, args.context, exten, 1,
 			S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
@@ -386,10 +385,7 @@ static int disa_exec(struct ast_channel *chan, const char *data)
 				ast_channel_unlock(chan);
 			}
 
-			app_reset_cdr = pbx_findapp("ResetCDR");
-			if (app_reset_cdr) {
-				pbx_exec(chan, app_reset_cdr, special_noanswer ? "" : "e");
-			} else {
+			if (ast_pbx_exec_application(chan, "ResetCDR", special_noanswer ? "" : "e")) {
 				ast_log(AST_LOG_NOTICE, "ResetCDR application not found; CDR will not be reset\n");
 			}
 			ast_explicit_goto(chan, args.context, exten, 1);

apps/app_dtmfstore.c

@@ -0,0 +1,297 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * 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 Technology independent asynchronous DTMF collection
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup functions
+ *
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/framehook.h"
+#include "asterisk/app.h"
+#include "asterisk/conversions.h"
+
+/*** DOCUMENTATION
+	<application name="StoreDTMF" language="en_US">
+		<since>
+			<version>16.20.0</version>
+			<version>18.6.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Stores DTMF digits transmitted or received on a channel.
+		</synopsis>
+		<syntax>
+			<parameter name="direction" required="true">
+				<para>Must be <literal>TX</literal> or <literal>RX</literal> to
+				store digits, or <literal>remove</literal> to disable.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>The StoreDTMF function can be used to obtain digits sent in the
+			<literal>TX</literal> or <literal>RX</literal> direction of any channel.</para>
+			<para>The arguments are:</para>
+			<para><replaceable>var_name</replaceable>: Name of variable to which to append
+			digits.</para>
+			<para><replaceable>max_digits</replaceable>: The maximum number of digits to
+			store in the variable. Defaults to 0 (no maximum). After reading <literal>
+			maximum</literal> digits, no more digits will be stored.</para>
+			<example title="Store digits in CDR variable">
+			same => n,StoreDTMF(TX,CDR(digits))
+			</example>
+			<example title="Store up to 24 digits">
+			same => n,StoreDTMF(RX,testvar,24)
+			</example>
+			<example title="Disable digit collection">
+			same => n,StoreDTMF(remove)
+			</example>
+		</description>
+	</application>
+ ***/
+
+static char *app = "StoreDTMF";
+
+/*! \brief Private data structure used with the function's datastore */
+struct dtmf_store_data {
+	int framehook_id;
+	char *rx_var;
+	char *tx_var;
+	int maxdigits;
+};
+
+static void datastore_destroy_cb(void *data) {
+	struct dtmf_store_data *d;
+	d = data;
+	if (d) {
+		if (d->rx_var) {
+			ast_free(d->rx_var);
+		}
+		if (d->tx_var) {
+			ast_free(d->tx_var);
+		}
+		ast_free(data);
+	}
+}
+
+/*! \brief The channel datastore the function uses to store state */
+static const struct ast_datastore_info dtmf_store_datastore = {
+	.type = "dtmf_store",
+	.destroy = datastore_destroy_cb
+};
+
+/*! \internal \brief Store digits tx/rx on the channel */
+static int remove_dtmf_store(struct ast_channel *chan)
+{
+	struct ast_datastore *datastore = NULL;
+	struct dtmf_store_data *data;
+	SCOPED_CHANNELLOCK(chan_lock, chan);
+
+	datastore = ast_channel_datastore_find(chan, &dtmf_store_datastore, NULL);
+	if (!datastore) {
+		ast_log(AST_LOG_WARNING, "Cannot remove StoreDTMF from %s: StoreDTMF not currently enabled\n",
+		        ast_channel_name(chan));
+		return -1;
+	}
+	data = datastore->data;
+
+	if (ast_framehook_detach(chan, data->framehook_id)) {
+		ast_log(AST_LOG_WARNING, "Failed to remove StoreDTMF framehook from channel %s\n",
+		        ast_channel_name(chan));
+		return -1;
+	}
+
+	if (ast_channel_datastore_remove(chan, datastore)) {
+		ast_log(AST_LOG_WARNING, "Failed to remove StoreDTMF datastore from channel %s\n",
+		        ast_channel_name(chan));
+		return -1;
+	}
+	ast_datastore_free(datastore);
+
+	return 0;
+}
+
+/*! \brief Frame hook that is called to intercept digit/undigit */
+static struct ast_frame *dtmf_store_framehook(struct ast_channel *chan,
+	struct ast_frame *f, enum ast_framehook_event event, void *data)
+{
+	char currentdata[512];
+	char varnamesub[64];
+	char *varname = NULL;
+	struct dtmf_store_data *framedata = data;
+	int len;
+
+	if (!f || !framedata) {
+		return f;
+	}
+
+	if ((event != AST_FRAMEHOOK_EVENT_WRITE) && (event != AST_FRAMEHOOK_EVENT_READ)) {
+		return f;
+	}
+
+	if (f->frametype != AST_FRAME_DTMF_END) {
+		return f;
+	}
+
+	/* If this is DTMF then store the digits */
+	if (event == AST_FRAMEHOOK_EVENT_READ && framedata->rx_var) { /* coming from source */
+		varname = framedata->rx_var;
+	} else if (event == AST_FRAMEHOOK_EVENT_WRITE && framedata->tx_var) { /* going to source */
+		varname = framedata->tx_var;
+	}
+
+	if (!varname) {
+		return f;
+	}
+
+	sprintf(varnamesub, "${%s}", varname);
+	pbx_substitute_variables_helper(chan, varnamesub, currentdata, 511);
+	/* pbx_builtin_getvar_helper works for regular vars but not CDR vars */
+	if (ast_strlen_zero(currentdata)) { /* var doesn't exist yet */
+		ast_debug(3, "Creating new digit store: %s\n", varname);
+	}
+	len = strlen(currentdata);
+	if (framedata->maxdigits > 0 && len >= framedata->maxdigits) {
+		ast_debug(3, "Reached digit limit: %d\n", framedata->maxdigits);
+		remove_dtmf_store(chan); /* reached max digit count, stop now */
+		return f;
+	} else {
+		char newdata[len + 2]; /* one more char + terminator */
+		if (len > 0) {
+			ast_copy_string(newdata, currentdata, len + 2);
+		}
+		newdata[len] = (unsigned) f->subclass.integer;
+		newdata[len + 1] = '\0';
+		ast_debug(3, "Appending to digit store: now %s\n", newdata);
+		pbx_builtin_setvar_helper(chan, varname, newdata);
+	}
+	return f;
+}
+
+/*! \internal \brief Enable digit interception on the channel */
+static int dtmfstore_exec(struct ast_channel *chan, const char *appdata)
+{
+	struct ast_datastore *datastore;
+	struct dtmf_store_data *data;
+	static struct ast_framehook_interface digit_framehook_interface = {
+		.version = AST_FRAMEHOOK_INTERFACE_VERSION,
+		.event_cb = dtmf_store_framehook,
+		.disable_inheritance = 1,
+	};
+	char *parse = ast_strdupa(appdata);
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(direction);
+		AST_APP_ARG(varname);
+		AST_APP_ARG(maxdigits);
+	);
+	SCOPED_CHANNELLOCK(chan_lock, chan);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (ast_strlen_zero(appdata)) {
+		ast_log(AST_LOG_WARNING, "StoreDTMF requires an argument\n");
+		return -1;
+	}
+
+	if (!strcasecmp(args.direction, "remove")) {
+		return remove_dtmf_store(chan);
+	}
+
+	datastore = ast_channel_datastore_find(chan, &dtmf_store_datastore, NULL);
+	if (datastore) {
+		ast_log(AST_LOG_WARNING, "StoreDTMF already set on '%s'\n",
+		        ast_channel_name(chan));
+		return 0;
+	}
+
+	datastore = ast_datastore_alloc(&dtmf_store_datastore, NULL);
+	if (!datastore) {
+		return -1;
+	}
+
+	data = ast_calloc(1, sizeof(*data));
+	if (!data) {
+		ast_datastore_free(datastore);
+		return -1;
+	}
+
+	digit_framehook_interface.data = data;
+
+	data->rx_var = NULL;
+	data->tx_var = NULL;
+	data->maxdigits = 0;
+
+	if (!strcasecmp(args.direction, "tx")) {
+		data->tx_var = ast_strdup(args.varname);
+	} else if (!strcasecmp(args.direction, "rx")) {
+		data->rx_var = ast_strdup(args.varname);
+	} else {
+		ast_log(LOG_ERROR, "Direction must be either RX or TX\n");
+		return -1;
+	}
+
+	if (!ast_strlen_zero(args.maxdigits)) {
+		if (ast_str_to_int(args.maxdigits,&(data->maxdigits))) {
+			ast_log(LOG_ERROR, "Invalid integer: %s\n", args.maxdigits);
+			return -1;
+		}
+		if (data->maxdigits < 0) {
+			ast_log(LOG_ERROR, "Invalid natural number: %d\n", data->maxdigits);
+			return -1;
+		} else if (data->maxdigits == 0) {
+			ast_log(LOG_WARNING, "No maximum digit count set\n");
+		}
+	}
+
+	data->framehook_id = ast_framehook_attach(chan, &digit_framehook_interface);
+	if (data->framehook_id < 0) {
+		ast_log(AST_LOG_WARNING, "Failed to attach StoreDTMF framehook to '%s'\n",
+		        ast_channel_name(chan));
+		ast_datastore_free(datastore);
+		ast_free(data);
+		return -1;
+	}
+	datastore->data = data;
+
+	ast_channel_datastore_add(chan, datastore);
+
+	return 0;
+}
+
+static int unload_module(void)
+{
+	return ast_unregister_application(app);
+}
+
+static int load_module(void)
+{
+	return ast_register_application_xml(app, dtmfstore_exec);
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Technology independent async DTMF storage");

apps/app_externalivr.c

@@ -254,9 +254,9 @@ static struct ast_frame *gen_readframe(struct gen_state *state)
 	if (!(state->stream && (f = ast_readframe(state->stream)))) {
 		if (state->current) {
 			/* remove finished file from playlist */
-                        AST_LIST_LOCK(&u->playlist);
-                        AST_LIST_REMOVE_HEAD(&u->playlist, list);
-                        AST_LIST_UNLOCK(&u->playlist);
+			AST_LIST_LOCK(&u->playlist);
+			AST_LIST_REMOVE_HEAD(&u->playlist, list);
+			AST_LIST_UNLOCK(&u->playlist);
 			/* add finished file to finishlist */
 			AST_LIST_LOCK(&u->finishlist);
 			AST_LIST_INSERT_TAIL(&u->finishlist, state->current, list);
@@ -581,7 +581,7 @@ static int app_exec(struct ast_channel *chan, const char *data)
 		}
 	}
 
-	exit:
+  exit:
 	if (u->gen_active) {
 		ast_deactivate_generator(chan);
 	}
@@ -622,21 +622,21 @@ static int app_exec(struct ast_channel *chan, const char *data)
 }
 
 static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
-				struct ast_iostream *eivr_events,
-				struct ast_iostream *eivr_commands,
-				struct ast_iostream *eivr_errors,
-				const struct ast_str *args, const struct ast_flags flags)
+	struct ast_iostream *eivr_events,
+	struct ast_iostream *eivr_commands,
+	struct ast_iostream *eivr_errors,
+	const struct ast_str *args, const struct ast_flags flags)
 {
 	char input[1024];
 	struct playlist_entry *entry;
 	struct ast_frame *f;
 	int ms;
- 	int exception;
- 	int ready_fd;
+	int exception;
+	int ready_fd;
 	int waitfds[2];
 	int r;
- 	struct ast_channel *rchan;
- 	int res = -1;
+	struct ast_channel *rchan;
+	int res = -1;
 	int hangup_info_sent = 0;
 
 	waitfds[0] = ast_iostream_get_fd(eivr_commands);
@@ -645,78 +645,78 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
 	while (1) {
 		if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_ZOMBIE)) {
 			ast_chan_log(LOG_ERROR, chan, "Is a zombie\n");
- 			break;
- 		}
- 		if (!hangup_info_sent && !(ast_test_flag(&flags, run_dead)) && ast_check_hangup(chan)) {
+			break;
+		}
+		if (!hangup_info_sent && !(ast_test_flag(&flags, run_dead)) && ast_check_hangup(chan)) {
 			if (ast_test_flag(&flags, ignore_hangup)) {
 				ast_verb(3, "Got check_hangup, but ignore_hangup set so sending 'I' command\n");
 				send_eivr_event(eivr_events, 'I', "HANGUP", chan);
 				hangup_info_sent = 1;
 			} else {
- 				ast_verb(3, "Got check_hangup\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
-	 			break;
+				ast_verb(3, "Got check_hangup\n");
+				send_eivr_event(eivr_events, 'H', NULL, chan);
+				break;
 			}
- 		}
+		}
 
- 		ready_fd = 0;
- 		ms = 100;
- 		errno = 0;
- 		exception = 0;
+		ready_fd = 0;
+		ms = 100;
+		errno = 0;
+		exception = 0;
 
 		rchan = ast_waitfor_nandfds(&chan, 1, waitfds, (eivr_errors) ? 2 : 1, &exception, &ready_fd, &ms);
 
- 		if (ast_channel_state(chan) == AST_STATE_UP && !AST_LIST_EMPTY(&u->finishlist)) {
- 			AST_LIST_LOCK(&u->finishlist);
- 			while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
- 				send_eivr_event(eivr_events, 'F', entry->filename, chan);
- 				ast_free(entry);
- 			}
- 			AST_LIST_UNLOCK(&u->finishlist);
- 		}
+		if (ast_channel_state(chan) == AST_STATE_UP && !AST_LIST_EMPTY(&u->finishlist)) {
+			AST_LIST_LOCK(&u->finishlist);
+			while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
+				send_eivr_event(eivr_events, 'F', entry->filename, chan);
+				ast_free(entry);
+			}
+			AST_LIST_UNLOCK(&u->finishlist);
+		}
 
- 		if (ast_channel_state(chan) == AST_STATE_UP && !(ast_check_hangup(chan)) && rchan) {
- 			/* the channel has something */
- 			f = ast_read(chan);
- 			if (!f) {
- 				ast_verb(3, "Returned no frame\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				break;
- 			}
- 			if (f->frametype == AST_FRAME_DTMF) {
- 				send_eivr_event(eivr_events, f->subclass.integer, NULL, chan);
- 				if (u->option_autoclear) {
-  					AST_LIST_LOCK(&u->playlist);
-  					if (!u->abort_current_sound && !u->playing_silence) {
+		if (ast_channel_state(chan) == AST_STATE_UP && !(ast_check_hangup(chan)) && rchan) {
+			/* the channel has something */
+			f = ast_read(chan);
+			if (!f) {
+				ast_verb(3, "Returned no frame\n");
+				send_eivr_event(eivr_events, 'H', NULL, chan);
+				break;
+			}
+			if (f->frametype == AST_FRAME_DTMF) {
+				send_eivr_event(eivr_events, f->subclass.integer, NULL, chan);
+				if (u->option_autoclear) {
+					AST_LIST_LOCK(&u->playlist);
+					if (!u->abort_current_sound && !u->playing_silence) {
 						/* send interrupted file as T data */
- 						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
-	 						send_eivr_event(eivr_events, 'T', entry->filename, chan);
+						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+							send_eivr_event(eivr_events, 'T', entry->filename, chan);
 							ast_free(entry);
 						}
 					}
-  					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 						send_eivr_event(eivr_events, 'D', entry->filename, chan);
-  						ast_free(entry);
-  					}
-  					if (!u->playing_silence)
-  						u->abort_current_sound = 1;
-  					AST_LIST_UNLOCK(&u->playlist);
-  				}
- 			} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP)) {
- 				ast_verb(3, "Got AST_CONTROL_HANGUP\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
+					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+						send_eivr_event(eivr_events, 'D', entry->filename, chan);
+						ast_free(entry);
+					}
+					if (!u->playing_silence)
+						u->abort_current_sound = 1;
+					AST_LIST_UNLOCK(&u->playlist);
+				}
+			} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP)) {
+				ast_verb(3, "Got AST_CONTROL_HANGUP\n");
+				send_eivr_event(eivr_events, 'H', NULL, chan);
 				if (f->data.uint32) {
 					ast_channel_hangupcause_set(chan, f->data.uint32);
 				}
- 				ast_frfree(f);
- 				break;
- 			}
- 			ast_frfree(f);
- 		} else if (ready_fd == waitfds[0]) {
- 			if (exception) {
- 				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
-  				break;
-  			}
+				ast_frfree(f);
+				break;
+			}
+			ast_frfree(f);
+		} else if (ready_fd == waitfds[0]) {
+			if (exception) {
+				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
+				break;
+			}
 
 			r = ast_iostream_gets(eivr_commands, input, sizeof(input));
 			if (r <= 0) {
@@ -784,112 +784,112 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
 					u->abort_current_sound = 1;
 				}
 				AST_LIST_UNLOCK(&u->playlist);
- 			} else if (input[0] == EIVR_CMD_SQUE) {
+			} else if (input[0] == EIVR_CMD_SQUE) {
 				if (ast_channel_state(chan) != AST_STATE_UP || ast_check_hangup(chan)) {
 					ast_chan_log(LOG_WARNING, chan, "Queue re'S'et called on unanswered channel\n");
 					send_eivr_event(eivr_events, 'Z', NULL, chan);
 					continue;
 				}
- 				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', &input[2], chan);
- 				} else {
- 					AST_LIST_LOCK(&u->playlist);
-	 				if (!u->abort_current_sound && !u->playing_silence) {
+				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
+					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+					send_eivr_event(eivr_events, 'Z', &input[2], chan);
+				} else {
+					AST_LIST_LOCK(&u->playlist);
+					if (!u->abort_current_sound && !u->playing_silence) {
 						/* send interrupted file as T data */
- 						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
-	 						send_eivr_event(eivr_events, 'T', entry->filename, chan);
+						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+							send_eivr_event(eivr_events, 'T', entry->filename, chan);
 							ast_free(entry);
 						}
 					}
- 					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 						send_eivr_event(eivr_events, 'D', entry->filename, chan);
- 						ast_free(entry);
+					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+						send_eivr_event(eivr_events, 'D', entry->filename, chan);
+						ast_free(entry);
 					}
-	 				if (!u->playing_silence) {
- 						u->abort_current_sound = 1;
+					if (!u->playing_silence) {
+						u->abort_current_sound = 1;
 					}
- 					entry = make_entry(&input[2]);
- 					if (entry) {
- 						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+					entry = make_entry(&input[2]);
+					if (entry) {
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
 					}
-	 				AST_LIST_UNLOCK(&u->playlist);
+					AST_LIST_UNLOCK(&u->playlist);
 				}
- 			} else if (input[0] == EIVR_CMD_APND) {
+			} else if (input[0] == EIVR_CMD_APND) {
 				if (ast_channel_state(chan) != AST_STATE_UP || ast_check_hangup(chan)) {
 					ast_chan_log(LOG_WARNING, chan, "Queue 'A'ppend called on unanswered channel\n");
 					send_eivr_event(eivr_events, 'Z', NULL, chan);
 					continue;
 				}
- 				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', &input[2], chan);
- 				} else {
-	 				entry = make_entry(&input[2]);
- 					if (entry) {
- 						AST_LIST_LOCK(&u->playlist);
- 						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
- 						AST_LIST_UNLOCK(&u->playlist);
+				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
+					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+					send_eivr_event(eivr_events, 'Z', &input[2], chan);
+				} else {
+					entry = make_entry(&input[2]);
+					if (entry) {
+						AST_LIST_LOCK(&u->playlist);
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+						AST_LIST_UNLOCK(&u->playlist);
 					}
- 				}
- 			} else if (input[0] == EIVR_CMD_GET) {
- 				char response[2048];
+				}
+			} else if (input[0] == EIVR_CMD_GET) {
+				char response[2048];
 				ast_verb(4, "Retriving Variables from channel: %s\n", &input[2]);
- 				ast_eivr_getvariable(chan, &input[2], response, sizeof(response));
- 				send_eivr_event(eivr_events, 'G', response, chan);
- 			} else if (input[0] == EIVR_CMD_SVAR) {
+				ast_eivr_getvariable(chan, &input[2], response, sizeof(response));
+				send_eivr_event(eivr_events, 'G', response, chan);
+			} else if (input[0] == EIVR_CMD_SVAR) {
 				ast_verb(4, "Setting Variables in channel: %s\n", &input[2]);
- 				ast_eivr_setvariable(chan, &input[2]);
- 			} else if (input[0] == EIVR_CMD_LOG) {
- 				ast_chan_log(LOG_NOTICE, chan, "Log message from EIVR: %s\n", &input[2]);
- 			} else if (input[0] == EIVR_CMD_XIT) {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
+				ast_eivr_setvariable(chan, &input[2]);
+			} else if (input[0] == EIVR_CMD_LOG) {
+				ast_chan_log(LOG_NOTICE, chan, "Log message from EIVR: %s\n", &input[2]);
+			} else if (input[0] == EIVR_CMD_XIT) {
+				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
 				ast_chan_log(LOG_WARNING, chan, "e'X'it command is depricated, use 'E'xit instead\n");
- 				res = 0;
- 				break;
+				res = 0;
+				break;
 			} else if (input[0] == EIVR_CMD_EXIT) {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'E', NULL, chan);
- 				res = 0;
- 				break;
- 			} else if (input[0] == EIVR_CMD_HGUP) {
- 				ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				break;
- 			} else if (input[0] == EIVR_CMD_OPT) {
+				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
+				send_eivr_event(eivr_events, 'E', NULL, chan);
+				res = 0;
+				break;
+			} else if (input[0] == EIVR_CMD_HGUP) {
+				ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
+				send_eivr_event(eivr_events, 'H', NULL, chan);
+				break;
+			} else if (input[0] == EIVR_CMD_OPT) {
 				if (ast_channel_state(chan) != AST_STATE_UP || ast_check_hangup(chan)) {
 					ast_chan_log(LOG_WARNING, chan, "Option called on unanswered channel\n");
 					send_eivr_event(eivr_events, 'Z', NULL, chan);
 					continue;
 				}
- 				if (!strcasecmp(&input[2], "autoclear"))
- 					u->option_autoclear = 1;
- 				else if (!strcasecmp(&input[2], "noautoclear"))
- 					u->option_autoclear = 0;
- 				else
- 					ast_chan_log(LOG_WARNING, chan, "Unknown option requested: %s\n", &input[2]);
- 			}
- 		} else if (ready_fd == waitfds[1]) {
- 			if (exception) {
- 				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
- 				break;
- 			}
-
- 			r = ast_iostream_gets(eivr_errors, input, sizeof(input));
- 			if (r > 0) {
- 				ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", ast_strip(input));
- 			} else if (r == 0) {
- 				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
- 				break;
+				if (!strcasecmp(&input[2], "autoclear"))
+					u->option_autoclear = 1;
+				else if (!strcasecmp(&input[2], "noautoclear"))
+					u->option_autoclear = 0;
+				else
+					ast_chan_log(LOG_WARNING, chan, "Unknown option requested: %s\n", &input[2]);
+			}
+		} else if (ready_fd == waitfds[1]) {
+			if (exception) {
+				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
+				break;
 			}
- 		} else if ((ready_fd < 0) && ms) {
- 			if (errno == 0 || errno == EINTR)
- 				continue;
 
- 			ast_chan_log(LOG_ERROR, chan, "Wait failed (%s)\n", strerror(errno));
- 			break;
- 		}
- 	}
+			r = ast_iostream_gets(eivr_errors, input, sizeof(input));
+			if (r > 0) {
+				ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", ast_strip(input));
+			} else if (r == 0) {
+				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
+				break;
+			}
+		} else if ((ready_fd < 0) && ms) {
+			if (errno == 0 || errno == EINTR)
+				continue;
+
+			ast_chan_log(LOG_ERROR, chan, "Wait failed (%s)\n", strerror(errno));
+			break;
+		}
+	}
 
 	return res;
 }

apps/app_fax.c

@@ -13,11 +13,13 @@
  */
 
 /*** MODULEINFO
-	<defaultenabled>no</defaultenabled>
 	<depend>spandsp</depend>
 	<conflict>res_fax</conflict>
+	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>res_fax</replacement>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
 ***/
 
 /* Needed for spandsp headers */

apps/app_festival.c

@@ -281,8 +281,6 @@ static int festival_exec(struct ast_channel *chan, const char *vdata)
 	int usecache;
 	int res = 0;
 	struct sockaddr_in serv_addr;
-	struct hostent *serverhost;
-	struct ast_hostent ahp;
 	int fd;
 	FILE *fs;
 	const char *host;
@@ -398,15 +396,17 @@ static int festival_exec(struct ast_channel *chan, const char *vdata)
 
 	if ((serv_addr.sin_addr.s_addr = inet_addr(host)) == -1) {
 		/* its a name rather than an ipnum */
-		serverhost = ast_gethostbyname(host, &ahp);
+		struct ast_sockaddr addr = { {0,} };
 
-		if (serverhost == NULL) {
-			ast_log(LOG_WARNING, "festival_client: gethostbyname failed\n");
+		if (ast_sockaddr_resolve_first_af(&addr, host, PARSE_PORT_FORBID, AF_INET)) {
+			ast_log(LOG_WARNING, "festival_client: ast_sockaddr_resolve_first_af() failed\n");
 			ast_config_destroy(cfg);
 			close(fd);
 			return -1;
 		}
-		memmove(&serv_addr.sin_addr, serverhost->h_addr, serverhost->h_length);
+
+		/* We'll overwrite port and family in a sec */
+		ast_sockaddr_to_sin(&addr, &serv_addr);
 	}
 
 	serv_addr.sin_family = AF_INET;
@@ -433,7 +433,7 @@ static int festival_exec(struct ast_channel *chan, const char *vdata)
 	}
 	readcache = 0;
 	writecache = 0;
-	if (strlen(cachedir) + strlen(MD5Hex) + 1 <= MAXFESTLEN && (usecache == -1)) {
+	if (strlen(cachedir) + sizeof(MD5Hex) + 1 <= MAXFESTLEN && (usecache == -1)) {
 		snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5Hex);
 		fdesc = open(cachefile, O_RDWR);
 		if (fdesc == -1) {

apps/app_forkcdr.c

@@ -2,7 +2,7 @@
  * Asterisk -- An open source telephony toolkit.
  *
  * Copyright (C) 1999 - 2005, Anthony Minessale anthmct@yahoo.com
- * Development of this app Sponsered/Funded  by TAAN Softworks Corp
+ * Development of this app Sponsored/Funded  by TAAN Softworks Corp
  *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact

apps/app_ices.c

@@ -29,6 +29,8 @@
 
 /*** MODULEINFO
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_if.c

@@ -0,0 +1,383 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright 2022, Naveen Albert <asterisk@phreaknet.org>
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * 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 If Branch Implementation
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+
+/*** DOCUMENTATION
+	<application name="If" language="en_US">
+		<synopsis>
+			Start an if branch.
+		</synopsis>
+		<syntax>
+			<parameter name="expr" required="true" />
+		</syntax>
+		<description>
+			<para>Start an If branch.  Execution will continue inside the branch
+			if expr is true.</para>
+			<note><para>This application (and related applications) set variables
+			internally during execution.</para></note>
+		</description>
+		<see-also>
+			<ref type="application">ElseIf</ref>
+			<ref type="application">Else</ref>
+			<ref type="application">EndIf</ref>
+			<ref type="application">ExitIf</ref>
+		</see-also>
+	</application>
+	<application name="ElseIf" language="en_US">
+		<synopsis>
+			Start an else if branch.
+		</synopsis>
+		<syntax>
+			<parameter name="expr" required="true" />
+		</syntax>
+		<description>
+			<para>Start an optional ElseIf branch. Execution will continue inside the branch
+			if expr is true and if previous If and ElseIf branches evaluated to false.</para>
+			<para>Please note that execution inside a true If branch will fallthrough into
+			ElseIf unless the If segment is terminated with an ExitIf call. This is only
+			necessary with ElseIf but not with Else.</para>
+		</description>
+		<see-also>
+			<ref type="application">If</ref>
+			<ref type="application">Else</ref>
+			<ref type="application">EndIf</ref>
+			<ref type="application">ExitIf</ref>
+		</see-also>
+	</application>
+	<application name="Else" language="en_US">
+		<synopsis>
+			Define an optional else branch.
+		</synopsis>
+		<syntax>
+			<parameter name="expr" required="true" />
+		</syntax>
+		<description>
+			<para>Start an Else branch. Execution will jump here if all previous
+			If and ElseIf branches evaluated to false.</para>
+		</description>
+		<see-also>
+			<ref type="application">If</ref>
+			<ref type="application">ElseIf</ref>
+			<ref type="application">EndIf</ref>
+			<ref type="application">ExitIf</ref>
+		</see-also>
+	</application>
+	<application name="EndIf" language="en_US">
+		<synopsis>
+			End an if branch.
+		</synopsis>
+		<syntax />
+		<description>
+			<para>Ends the branch begun by the preceding <literal>If()</literal> application.</para>
+		</description>
+		<see-also>
+			<ref type="application">If</ref>
+			<ref type="application">ElseIf</ref>
+			<ref type="application">Else</ref>
+			<ref type="application">ExitIf</ref>
+		</see-also>
+	</application>
+	<application name="ExitIf" language="en_US">
+		<synopsis>
+			End an If branch.
+		</synopsis>
+		<syntax />
+		<description>
+			<para>Exits an <literal>If()</literal> branch, whether or not it has completed.</para>
+		</description>
+		<see-also>
+			<ref type="application">If</ref>
+			<ref type="application">ElseIf</ref>
+			<ref type="application">Else</ref>
+			<ref type="application">EndIf</ref>
+		</see-also>
+	</application>
+ ***/
+
+static char *if_app = "If";
+static char *elseif_app = "ElseIf";
+static char *else_app = "Else";
+static char *stop_app = "EndIf";
+static char *exit_app = "ExitIf";
+
+#define VAR_SIZE 64
+
+static const char *get_index(struct ast_channel *chan, const char *prefix, int idx)
+{
+	char varname[VAR_SIZE];
+
+	snprintf(varname, VAR_SIZE, "%s_%d", prefix, idx);
+	return pbx_builtin_getvar_helper(chan, varname);
+}
+
+static struct ast_exten *find_matching_priority(struct ast_context *c, const char *exten, int priority, const char *callerid)
+{
+	struct ast_exten *e;
+	struct ast_context *c2;
+	int idx;
+
+	for (e = ast_walk_context_extensions(c, NULL); e; e = ast_walk_context_extensions(c, e)) {
+		if (ast_extension_match(ast_get_extension_name(e), exten)) {
+			int needmatch = ast_get_extension_matchcid(e);
+			if ((needmatch && ast_extension_match(ast_get_extension_cidmatch(e), callerid)) ||
+				(!needmatch)) {
+				/* This is the matching extension we want */
+				struct ast_exten *p;
+				for (p = ast_walk_extension_priorities(e, NULL); p; p = ast_walk_extension_priorities(e, p)) {
+					if (priority != ast_get_extension_priority(p))
+						continue;
+					return p;
+				}
+			}
+		}
+	}
+
+	/* No match; run through includes */
+	for (idx = 0; idx < ast_context_includes_count(c); idx++) {
+		const struct ast_include *i = ast_context_includes_get(c, idx);
+
+		for (c2 = ast_walk_contexts(NULL); c2; c2 = ast_walk_contexts(c2)) {
+			if (!strcmp(ast_get_context_name(c2), ast_get_include_name(i))) {
+				e = find_matching_priority(c2, exten, priority, callerid);
+				if (e)
+					return e;
+			}
+		}
+	}
+	return NULL;
+}
+
+static int find_matching_endif(struct ast_channel *chan, const char *otherapp)
+{
+	struct ast_context *c;
+	int res = -1;
+
+	if (ast_rdlock_contexts()) {
+		ast_log(LOG_ERROR, "Failed to lock contexts list\n");
+		return -1;
+	}
+
+	for (c = ast_walk_contexts(NULL); c; c = ast_walk_contexts(c)) {
+		struct ast_exten *e;
+
+		if (!ast_rdlock_context(c)) {
+			if (!strcmp(ast_get_context_name(c), ast_channel_context(chan))) {
+				/* This is the matching context we want */
+				int cur_priority = ast_channel_priority(chan) + 1, level = 1;
+
+				for (e = find_matching_priority(c, ast_channel_exten(chan), cur_priority,
+					S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL));
+					e;
+					e = find_matching_priority(c, ast_channel_exten(chan), ++cur_priority,
+						S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
+					if (!strcasecmp(ast_get_extension_app(e), "IF")) {
+						level++;
+					} else if (!strcasecmp(ast_get_extension_app(e), "ENDIF")) {
+						level--;
+					}
+
+					if (!otherapp && level == 0) {
+						res = cur_priority;
+						break;
+					} else if (otherapp && level == 1 && !strcasecmp(ast_get_extension_app(e), otherapp)) {
+						res = cur_priority;
+						break;
+					}
+				}
+			}
+			ast_unlock_context(c);
+			if (res > 0) {
+				break;
+			}
+		}
+	}
+	ast_unlock_contexts();
+	return res;
+}
+
+static int if_helper(struct ast_channel *chan, const char *data, int end)
+{
+	int res = 0;
+	const char *if_pri = NULL;
+	char *my_name = NULL;
+	const char *label = NULL;
+	char varname[VAR_SIZE + 3]; /* + IF_ */
+	char end_varname[sizeof(varname) + 4]; /* + END_ + sizeof(varname) */
+	const char *prefix = "IF";
+	size_t size = 0;
+	int used_index_i = -1, x = 0;
+	char used_index[VAR_SIZE] = "0", new_index[VAR_SIZE] = "0";
+
+	if (!chan) {
+		return -1;
+	}
+
+	for (x = 0 ;; x++) {
+		if (get_index(chan, prefix, x)) {
+			used_index_i = x;
+		} else {
+			break;
+		}
+	}
+
+	snprintf(used_index, sizeof(used_index), "%d", used_index_i);
+	snprintf(new_index, sizeof(new_index), "%d", used_index_i + 1);
+
+	size = strlen(ast_channel_context(chan)) + strlen(ast_channel_exten(chan)) + 32;
+	my_name = ast_alloca(size);
+	memset(my_name, 0, size);
+	snprintf(my_name, size, "%s_%s_%d", ast_channel_context(chan), ast_channel_exten(chan), ast_channel_priority(chan));
+
+	ast_channel_lock(chan);
+	if (end > 1) {
+		label = used_index;
+	} else if (!(label = pbx_builtin_getvar_helper(chan, my_name))) {
+		label = new_index;
+		pbx_builtin_setvar_helper(chan, my_name, label);
+	}
+	snprintf(varname, sizeof(varname), "%s_%s", prefix, label);
+	if ((if_pri = pbx_builtin_getvar_helper(chan, varname)) && !end) {
+		if_pri = ast_strdupa(if_pri);
+		snprintf(end_varname,sizeof(end_varname),"END_%s",varname);
+	}
+	ast_channel_unlock(chan);
+
+	if ((end <= 1 && !pbx_checkcondition(ast_strdupa(data))) || (end > 1)) {
+		/* Condition Met (clean up helper vars) */
+		const char *goto_str;
+		int pri, endifpri;
+		pbx_builtin_setvar_helper(chan, varname, NULL);
+		pbx_builtin_setvar_helper(chan, my_name, NULL);
+		snprintf(end_varname,sizeof(end_varname),"END_%s",varname);
+		ast_channel_lock(chan);
+		endifpri = find_matching_endif(chan, NULL);
+		if ((goto_str = pbx_builtin_getvar_helper(chan, end_varname))) {
+			ast_parseable_goto(chan, goto_str);
+			pbx_builtin_setvar_helper(chan, end_varname, NULL);
+		} else if (end <= 1 && (pri = find_matching_endif(chan, "ElseIf")) > 0 && pri < endifpri) {
+			pri--; /* back up a priority, since it returned the priority after the ElseIf */
+			/* If is false, and ElseIf exists, so jump to ElseIf */
+			ast_verb(3, "Taking conditional false branch, jumping to priority %d\n", pri);
+			ast_channel_priority_set(chan, pri);
+		} else if (end <= 1 && (pri = find_matching_endif(chan, "Else")) > 0 && pri < endifpri) {
+			/* don't need to back up a priority, because we don't actually need to execute Else, just jump to the priority after. Directly executing Else will exit the conditional. */
+			/* If is false, and Else exists, so jump to Else */
+			ast_verb(3, "Taking absolute false branch, jumping to priority %d\n", pri);
+			ast_channel_priority_set(chan, pri);
+		} else {
+			pri = endifpri;
+			if (pri > 0) {
+				ast_verb(3, "Exiting conditional, jumping to priority %d\n", pri);
+				ast_channel_priority_set(chan, pri);
+			} else if (end == 4) { /* Condition added because of end > 0 instead of end == 4 */
+				ast_log(LOG_WARNING, "Couldn't find matching EndIf? (If at %s@%s priority %d)\n", ast_channel_context(chan), ast_channel_exten(chan), ast_channel_priority(chan));
+			}
+		}
+		ast_channel_unlock(chan);
+		return res;
+	}
+
+	if (end <= 1 && !if_pri) {
+		char *goto_str;
+		size = strlen(ast_channel_context(chan)) + strlen(ast_channel_exten(chan)) + 32;
+		goto_str = ast_alloca(size);
+		memset(goto_str, 0, size);
+		snprintf(goto_str, size, "%s,%s,%d", ast_channel_context(chan), ast_channel_exten(chan), ast_channel_priority(chan));
+		pbx_builtin_setvar_helper(chan, varname, goto_str);
+	} else if (end > 1 && if_pri) {
+		/* END of branch */
+		snprintf(end_varname, sizeof(end_varname), "END_%s", varname);
+		if (!pbx_builtin_getvar_helper(chan, end_varname)) {
+			char *goto_str;
+			size = strlen(ast_channel_context(chan)) + strlen(ast_channel_exten(chan)) + 32;
+			goto_str = ast_alloca(size);
+			memset(goto_str, 0, size);
+			snprintf(goto_str, size, "%s,%s,%d", ast_channel_context(chan), ast_channel_exten(chan), ast_channel_priority(chan)+1);
+			pbx_builtin_setvar_helper(chan, end_varname, goto_str);
+		}
+		ast_parseable_goto(chan, if_pri);
+	}
+
+	return res;
+}
+
+static int if_exec(struct ast_channel *chan, const char *data) {
+	return if_helper(chan, data, 0);
+}
+
+static int elseif_exec(struct ast_channel *chan, const char *data) {
+	return if_helper(chan, data, 1);
+}
+
+static int end_exec(struct ast_channel *chan, const char *data) {
+	return if_helper(chan, data, 2);
+}
+
+static int else_exec(struct ast_channel *chan, const char *data) {
+	return if_helper(chan, data, 3);
+}
+
+static int exit_exec(struct ast_channel *chan, const char *data) {
+	return if_helper(chan, data, 4);
+}
+
+static int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(if_app);
+	res |= ast_unregister_application(elseif_app);
+	res |= ast_unregister_application(stop_app);
+	res |= ast_unregister_application(else_app);
+	res |= ast_unregister_application(exit_app);
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	res = ast_register_application_xml(if_app, if_exec);
+	res |= ast_register_application_xml(elseif_app, elseif_exec);
+	res |= ast_register_application_xml(stop_app, end_exec);
+	res |= ast_register_application_xml(else_app, else_exec);
+	res |= ast_register_application_xml(exit_app, exit_exec);
+
+	return res;
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "If Branch and Conditional Execution");

apps/app_image.c

@@ -27,6 +27,8 @@
 
 /*** MODULEINFO
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_jack.c

@@ -615,8 +615,6 @@ static int queue_voice_frame(struct jack_data *jack_data, struct ast_frame *f)
  * Read data from the input ringbuffer, which is the properly resampled audio
  * that was read from the jack input port.  Write it to the channel in 20 ms frames,
  * or fill up an output frame instead if one is provided.
- *
- * \return Nothing.
  */
 static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_data,
 	struct ast_frame *out_frame)

apps/app_macro.c

@@ -29,6 +29,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>app_stack (GoSub)</replacement>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>21</removed_in>
  ***/
 
 #include "asterisk.h"
@@ -73,7 +75,7 @@
 			of nesting (macro calling macro calling macro, etc.); It may be possible that stack-intensive
 			applications in deeply nested macros could cause asterisk to crash earlier than this limit.
 			It is advised that if you need to deeply nest macro calls, that you use the Gosub application
-			(now allows arguments like a Macro) with explict Return() calls instead.</para></warning>
+			(now allows arguments like a Macro) with explicit Return() calls instead.</para></warning>
 			<warning><para>Use of the application <literal>WaitExten</literal> within a macro will not function
 			as expected. Please use the <literal>Read</literal> application in order to read DTMF from a channel
 			currently executing a macro.</para></warning>

apps/app_meetme.c

@@ -43,6 +43,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>extended</support_level>
 	<replacement>app_confbridge</replacement>
+	<deprecated_in>19</deprecated_in>
+	<removed_in>21</removed_in>
  ***/
 
 #include "asterisk.h"
@@ -637,6 +639,82 @@
 			</syntax>
 		</managerEventInstance>
 	</managerEvent>
+	<managerEvent language="en_US" name="MeetmeList">
+		<managerEventInstance class="EVENT_FLAG_CALL">
+			<synopsis>Raised in response to a MeetmeList command.</synopsis>
+			<syntax>
+				<parameter name="Conference">
+					<para>Conference ID.</para>
+				</parameter>
+				<parameter name="UserNumber">
+					<para>User ID.</para>
+				</parameter>
+				<parameter name="CallerIDNum">
+					<para>Caller ID number.</para>
+				</parameter>
+				<parameter name="CallerIDName">
+					<para>Caller ID name.</para>
+				</parameter>
+				<parameter name="ConnectedLineNum">
+					<para>Connected Line number.</para>
+				</parameter>
+				<parameter name="ConnectedLineName">
+					<para>Connected Line name.</para>
+				</parameter>
+				<parameter name="Channel">
+					<para>Channel name</para>
+				</parameter>
+				<parameter name="Admin">
+					<para>Whether or not the user is an admin.</para>
+				</parameter>
+				<parameter name="Role">
+					<para>User role. Can be "Listen only", "Talk only", or "Talk and listen".</para>
+				</parameter>
+				<parameter name="MarkedUser">
+					<para>Whether or not the user is a marked user.</para>
+				</parameter>
+				<parameter name="Muted">
+					<para>Whether or not the user is currently muted.</para>
+				</parameter>
+				<parameter name="Talking">
+					<para>Whether or not the user is currently talking.</para>
+				</parameter>
+			</syntax>
+			<see-also>
+				<ref type="manager">MeetmeList</ref>
+				<ref type="application">MeetMe</ref>
+			</see-also>
+		</managerEventInstance>
+	</managerEvent>
+	<managerEvent language="en_US" name="MeetmeListRooms">
+		<managerEventInstance class="EVENT_FLAG_CALL">
+			<synopsis>Raised in response to a MeetmeListRooms command.</synopsis>
+			<syntax>
+				<parameter name="Conference">
+					<para>Conference ID.</para>
+				</parameter>
+				<parameter name="Parties">
+					<para>Number of parties in the conference.</para>
+				</parameter>
+				<parameter name="Marked">
+					<para>Number of marked users in the conference.</para>
+				</parameter>
+				<parameter name="Activity">
+					<para>Total duration of conference in HH:MM:SS format.</para>
+				</parameter>
+				<parameter name="Creation">
+					<para>How the conference was created: "Dyanmic" or "Static".</para>
+				</parameter>
+				<parameter name="Locked">
+					<para>Whether or not the conference is locked.</para>
+				</parameter>
+			</syntax>
+			<see-also>
+				<ref type="manager">MeetmeListRooms</ref>
+				<ref type="application">MeetMe</ref>
+			</see-also>
+		</managerEventInstance>
+	</managerEvent>
  ***/
 
 #define CONFIG_FILE_NAME	"meetme.conf"
@@ -1334,7 +1412,7 @@ static struct ast_json *status_to_json(int on)
  * \brief Generate a stasis message associated with a meetme event
  * \since 12.0.0
  *
- * \param meetme_confere The conference responsible for generating this message
+ * \param meetme_conference The conference responsible for generating this message
  * \param chan The channel involved in the message (NULL allowed)
  * \param user The conference user involved in the message (NULL allowed)
  * \param message_type the type the stasis message being generated
@@ -2834,7 +2912,7 @@ static void meetme_menu_normal(enum menu_modes *menu_mode, int *dtmf, struct ast
 }
 
 /*! \internal
- * \brief Processes menu options for the adminstrator menu (accessible through the 's' option for app_meetme)
+ * \brief Processes menu options for the administrator menu (accessible through the 's' option for app_meetme)
  *
  * \param menu_mode a pointer to the currently active menu_mode.
  * \param dtmf a pointer to the dtmf value currently being processed against the menu.
@@ -2961,7 +3039,8 @@ static void meetme_menu_admin(enum menu_modes *menu_mode, int *dtmf, struct ast_
  * \param confflags flags used by conf for various options
  * \param chan ast_channel belonging to the user who called the menu
  * \param user which meetme conference user invoked the menu
- * \param recordingtmp character buffer which may hold the name of the conference recording file
+ * \param recordingtmp, recordingtmp_size character buffer which may hold the name of the conference recording file
+ * \param cap_slin
  */
 static void meetme_menu_admin_extended(enum menu_modes *menu_mode, int *dtmf,
 	struct ast_conference *conf, struct ast_flags64 *confflags, struct ast_channel *chan,
@@ -3148,7 +3227,8 @@ static void meetme_menu_admin_extended(enum menu_modes *menu_mode, int *dtmf,
  * \param confflags flags used by conf for various options
  * \param chan ast_channel belonging to the user who called the menu
  * \param user which meetme conference user invoked the menu
- * \param recordingtmp character buffer which may hold the name of the conference recording file
+ * \param recordingtmp,recordingtmp_size character buffer which may hold the name of the conference recording file
+ * \param cap_slin
  */
 static void meetme_menu(enum menu_modes *menu_mode, int *dtmf,
 	struct ast_conference *conf, struct ast_flags64 *confflags, struct ast_channel *chan,
@@ -4757,7 +4837,7 @@ static struct ast_conference *find_conf(struct ast_channel *chan, char *confno,
 				}
 			}
 			if (!var) {
-				ast_debug(1, "%s isn't a valid conference\n", confno);
+				ast_log(LOG_WARNING, "%s isn't a valid conference\n", confno);
 			}
 			ast_config_destroy(cfg);
 		}
@@ -5219,7 +5299,7 @@ static int user_chan_cb(void *obj, void *args, int flags)
 	return 0;
 }
 
-/*! \brief The MeetMeadmin application
+/*! \brief The MeetMeAdmin application
 
   MeetMeAdmin(confno, command, caller) */
 static int admin_exec(struct ast_channel *chan, const char *data) {
@@ -5235,7 +5315,9 @@ static int admin_exec(struct ast_channel *chan, const char *data) {
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "MeetMeAdmin requires an argument!\n");
-		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		if (chan) {
+			pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		}
 		return -1;
 	}
 
@@ -5244,7 +5326,9 @@ static int admin_exec(struct ast_channel *chan, const char *data) {
 
 	if (!args.command) {
 		ast_log(LOG_WARNING, "MeetmeAdmin requires a command!\n");
-		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		if (chan) {
+			pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		}
 		return -1;
 	}
 
@@ -5257,7 +5341,9 @@ static int admin_exec(struct ast_channel *chan, const char *data) {
 	if (!cnf) {
 		ast_log(LOG_WARNING, "Conference number '%s' not found!\n", args.confno);
 		AST_LIST_UNLOCK(&confs);
-		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOTFOUND");
+		if (chan) {
+			pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOTFOUND");
+		}
 		return 0;
 	}
 
@@ -5380,7 +5466,9 @@ usernotfound:
 	AST_LIST_UNLOCK(&confs);
 
 	dispose_conf(cnf);
-	pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", res == -2 ? "NOTFOUND" : res ? "FAILED" : "OK");
+	if (chan) {
+		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", res == -2 ? "NOTFOUND" : res ? "FAILED" : "OK");
+	}
 
 	return 0;
 }
@@ -6061,7 +6149,7 @@ struct run_station_args {
 
 static void answer_trunk_chan(struct ast_channel *chan)
 {
-	ast_answer(chan);
+	ast_raw_answer(chan);
 	ast_indicate(chan, -1);
 }
 
@@ -6906,8 +6994,18 @@ static void *dial_trunk(void *data)
 		return NULL;
 	}
 
-	for (;;) {
+	/* Wait for dial to end, while servicing the channel */
+	while (ast_waitfor(trunk_ref->chan, 100)) {
 		unsigned int done = 0;
+		struct ast_frame *fr = ast_read(trunk_ref->chan);
+
+		if (!fr) {
+			ast_debug(1, "Channel %s did not return a frame, must have hung up\n", ast_channel_name(trunk_ref->chan));
+			done = 1;
+			break;
+		}
+		ast_frfree(fr); /* Ignore while dialing */
+
 		switch ((dial_res = ast_dial_state(dial))) {
 		case AST_DIAL_RESULT_ANSWERED:
 			trunk_ref->trunk->chan = ast_dial_answered(dial);
@@ -6944,8 +7042,6 @@ static void *dial_trunk(void *data)
 			last_state = current_state;
 		}
 
-		/* avoid tight loop... sleep for 1/10th second */
-		ast_safe_sleep(trunk_ref->chan, 100);
 	}
 
 	if (!trunk_ref->trunk->chan) {
@@ -7104,8 +7200,10 @@ static int sla_station_exec(struct ast_channel *chan, const char *data)
 		sla_change_trunk_state(trunk_ref->trunk, SLA_TRUNK_STATE_UP, ALL_TRUNK_REFS, NULL);
 		/* Create a thread to dial the trunk and dump it into the conference.
 		 * However, we want to wait until the trunk has been dialed and the
-		 * conference is created before continuing on here. */
-		ast_autoservice_start(chan);
+		 * conference is created before continuing on here.
+		 * Don't autoservice the channel or we'll have multiple threads
+		 * handling it. dial_trunk services the channel.
+		 */
 		ast_mutex_init(&cond_lock);
 		ast_cond_init(&cond, NULL);
 		ast_mutex_lock(&cond_lock);
@@ -7114,7 +7212,7 @@ static int sla_station_exec(struct ast_channel *chan, const char *data)
 		ast_mutex_unlock(&cond_lock);
 		ast_mutex_destroy(&cond_lock);
 		ast_cond_destroy(&cond);
-		ast_autoservice_stop(chan);
+
 		if (!trunk_ref->trunk->chan) {
 			ast_debug(1, "Trunk didn't get created. chan: %lx\n", (unsigned long) trunk_ref->trunk->chan);
 			pbx_builtin_setvar_helper(chan, "SLASTATION_STATUS", "CONGESTION");

apps/app_mf.c

@@ -0,0 +1,547 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * 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 MF sender and receiver applications
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/file.h"
+#include "asterisk/pbx.h"
+#include "asterisk/channel.h"
+#include "asterisk/dsp.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+#include "asterisk/indications.h"
+#include "asterisk/conversions.h"
+
+/*** DOCUMENTATION
+	<application name="ReceiveMF" language="en_US">
+		<since>
+			<version>16.21.0</version>
+			<version>18.7.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Detects MF digits on a channel and saves them to a variable.
+		</synopsis>
+		<syntax>
+			<parameter name="variable" required="true">
+				<para>The input digits will be stored in the given
+				<replaceable>variable</replaceable> name.</para>
+			</parameter>
+			<parameter name="timeout">
+				<para>The number of seconds to wait for all digits, if greater
+				than <literal>0</literal>. Can be floating point. Default
+				is no timeout.</para>
+			</parameter>
+			<parameter name="options">
+				<optionlist>
+					<option name="d">
+						<para>Delay audio by a frame to try to extra quelch.</para>
+					</option>
+					<option name="l">
+						<para>Receive digits even if a key pulse (KP) has not yet
+						been received. By default, this application will ignore
+						all other digits until a KP has been received.</para>
+					</option>
+					<option name="k">
+						<para>Do not return a character for the KP digit.</para>
+					</option>
+					<option name="m">
+						<para>Mute conference.</para>
+					</option>
+					<option name="n">
+						<para>Maximum number of digits, regardless of the sequence.</para>
+					</option>
+					<option name="o">
+						<para>Enable override. Repeated KPs will clear all previous digits.</para>
+					</option>
+					<option name="q">
+						<para>Quelch MF from in-band.</para>
+					</option>
+					<option name="r">
+						<para>"Radio" mode (relaxed MF).</para>
+					</option>
+					<option name="s">
+						<para>Do not return a character for ST digits.</para>
+					</option>
+				</optionlist>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Reads a ST, STP, ST2P, or ST3P-terminated string of MF digits from
+			the user in to the given <replaceable>variable</replaceable>.</para>
+			<para>This application does not automatically answer the channel and
+			should be preceded with <literal>Answer</literal> or
+			<literal>Progress</literal> as needed.</para>
+			<variablelist>
+				<variable name="RECEIVEMFSTATUS">
+					<para>This is the status of the read operation.</para>
+					<value name="START" />
+					<value name="ERROR" />
+					<value name="HANGUP" />
+					<value name="MAXDIGITS" />
+					<value name="TIMEOUT" />
+				</variable>
+			</variablelist>
+		</description>
+		<see-also>
+			<ref type="application">Read</ref>
+			<ref type="application">SendMF</ref>
+			<ref type="application">ReceiveSF</ref>
+		</see-also>
+	</application>
+	<application name="SendMF" language="en_US">
+		<since>
+			<version>16.21.0</version>
+			<version>18.7.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Sends arbitrary MF digits on the current or specified channel.
+		</synopsis>
+		<syntax>
+			<parameter name="digits" required="true">
+				<para>List of digits 0-9,*#ABC to send; w for a half-second pause,
+				also f or F for a flash-hook if the channel supports flash-hook,
+				h or H for 250 ms of 2600 Hz,
+				and W for a wink if the channel supports wink.</para>
+				<para>Key pulse and start digits are not included automatically.
+				* is used for KP, # for ST, A for STP, B for ST2P, and C for ST3P.</para>
+			</parameter>
+			<parameter name="timeout_ms" required="false">
+				<para>Amount of time to wait in ms between tones. (defaults to 50ms).</para>
+			</parameter>
+			<parameter name="duration_ms" required="false">
+				<para>Duration of each numeric digit (defaults to 55ms).</para>
+			</parameter>
+			<parameter name="duration_ms_kp" required="false">
+				<para>Duration of KP digits (defaults to 120ms).</para>
+			</parameter>
+			<parameter name="duration_ms_st" required="false">
+				<para>Duration of ST, STP, ST2P, and ST3P digits (defaults to 65ms).</para>
+			</parameter>
+			<parameter name="channel" required="false">
+				<para>Channel where digits will be played</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>It will send all digits or terminate if it encounters an error.</para>
+		</description>
+		<see-also>
+			<ref type="application">ReceiveMF</ref>
+			<ref type="application">SendSF</ref>
+			<ref type="application">SendDTMF</ref>
+		</see-also>
+	</application>
+	<manager name="PlayMF" language="en_US">
+		<since>
+			<version>16.21.0</version>
+			<version>18.7.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Play MF digit on a specific channel.
+		</synopsis>
+		<syntax>
+			<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+			<parameter name="Channel" required="true">
+				<para>Channel name to send digit to.</para>
+			</parameter>
+			<parameter name="Digit" required="true">
+				<para>The MF digit to play.</para>
+			</parameter>
+			<parameter name="Duration" required="false">
+				<para>The duration, in milliseconds, of the digit to be played.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Plays an MF digit on the specified channel.</para>
+		</description>
+	</manager>
+ ***/
+
+enum read_option_flags {
+	OPT_DELAY = (1 << 0),
+	OPT_MUTE = (1 << 1),
+	OPT_QUELCH = (1 << 2),
+	OPT_RELAXED = (1 << 3),
+	OPT_LAX_KP = (1 << 4),
+	OPT_PROCESS = (1 << 5),
+	OPT_NO_KP = (1 << 6),
+	OPT_NO_ST = (1 << 7),
+	OPT_KP_OVERRIDE = (1 << 8),
+	OPT_MAXDIGITS = (1 << 9),
+};
+
+enum {
+	OPT_ARG_MAXDIGITS,
+	/* Must be the last element */
+	OPT_ARG_ARRAY_SIZE,
+};
+
+AST_APP_OPTIONS(read_app_options, {
+	AST_APP_OPTION('d', OPT_DELAY),
+	AST_APP_OPTION('l', OPT_LAX_KP),
+	AST_APP_OPTION('k', OPT_NO_KP),
+	AST_APP_OPTION('m', OPT_MUTE),
+	AST_APP_OPTION_ARG('n', OPT_MAXDIGITS, OPT_ARG_MAXDIGITS),
+	AST_APP_OPTION('o', OPT_KP_OVERRIDE),
+	AST_APP_OPTION('p', OPT_PROCESS),
+	AST_APP_OPTION('q', OPT_QUELCH),
+	AST_APP_OPTION('r', OPT_RELAXED),
+	AST_APP_OPTION('s', OPT_NO_ST),
+});
+
+static const char *readmf_name = "ReceiveMF";
+static const char sendmf_name[] = "SendMF";
+
+#define MF_BETWEEN_MS 50
+#define MF_DURATION 55
+#define MF_KP_DURATION 120
+#define MF_ST_DURATION 65
+
+/*!
+ * \brief Detects MF digits on channel using DSP, terminated by ST, STP, ST2P, or ST3P
+ *
+ * \param chan channel on which to read digits
+ * \param buf Buffer in which to store digits
+ * \param buflen Size of buffer
+ * \param timeout ms to wait for all digits before giving up
+ * \param features Any additional DSP features to use
+ * \param laxkp Receive digits even if KP not received
+ * \param override Start over if we receive additional KPs
+ * \param no_kp Don't include KP in the output
+ * \param no_st Don't include start digits in the output
+ * \param maxdigits If greater than 0, only read this many digits no matter what
+ *
+ * \retval 0 if successful
+ * \retval -1 if unsuccessful (including hangup).
+ */
+static int read_mf_digits(struct ast_channel *chan, char *buf, int buflen, int timeout, int features, int laxkp, int override, int no_kp, int no_st, int maxdigits) {
+	struct ast_dsp *dsp;
+	struct ast_frame *frame = NULL;
+	struct timeval start;
+	int remaining_time = timeout;
+	int digits_read = 0;
+	int is_start_digit = 0;
+	char *str = buf;
+	int res = 0;
+
+	if (!(dsp = ast_dsp_new())) {
+		ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+		pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "ERROR");
+		return -1;
+	}
+	ast_dsp_set_features(dsp, DSP_FEATURE_DIGIT_DETECT);
+	ast_dsp_set_digitmode(dsp, DSP_DIGITMODE_MF | features);
+
+	start = ast_tvnow();
+	*str = 0; /* start with empty output buffer */
+
+	/* based on app_read and generic_fax_exec from res_fax */
+	while (timeout == 0 || remaining_time > 0) {
+		if (timeout > 0) {
+			remaining_time = ast_remaining_ms(start, timeout);
+			if (remaining_time <= 0) {
+				pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "TIMEOUT");
+				break;
+			}
+		}
+		if ((maxdigits && digits_read >= maxdigits) || digits_read >= (buflen - 1)) { /* we don't have room to store any more digits (very unlikely to happen for a legitimate reason) */
+			/* This result will probably not be usable, so status should not be START */
+			pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "MAXDIGITS");
+			break;
+		}
+		/* ast_waitfordigit only waits for DTMF frames, we need to do DSP on voice frames */
+		if (ast_waitfor(chan, 1000) > 0) {
+			frame = ast_read(chan);
+			if (!frame) {
+				ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
+				pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "HANGUP");
+				break;
+			} else if (frame->frametype == AST_FRAME_VOICE) {
+				frame = ast_dsp_process(chan, dsp, frame);
+				/* AST_FRAME_DTMF is used all over the DSP code for DTMF, MF, fax, etc.
+					It's used because we can use the frame to store the digit detected.
+					All this means is that we received something we care about. */
+				if (frame->frametype == AST_FRAME_DTMF) {
+					char result = frame->subclass.integer;
+					if (digits_read == 0 && !laxkp && result != '*') {
+						ast_debug(1, "Received MF digit, but no KP yet, ignoring: %c\n", result);
+						ast_frfree(frame);
+						continue;
+					}
+					ast_debug(1, "Received MF digit: %c\n", result);
+					if (result == '*') {
+						/* We received an additional KP, start over? */
+						if (override && digits_read > 0) {
+							ast_debug(1, "Received another KP, starting over\n");
+							str = buf;
+							*str = 0;
+							digits_read = 1; /* we just detected a KP */
+						} else {
+							digits_read++;
+						}
+						/* if we were told not to include the KP digit in the output string, then skip it */
+						if (no_kp) {
+							ast_frfree(frame);
+							continue;
+						}
+					} else {
+						digits_read++;
+					}
+					is_start_digit = (strchr("#", result) || strchr("A", result) || strchr("B", result) || strchr("C", result));
+					/* if we were told not to include the ST digit in the output string, then skip it */
+					if (!no_st || !is_start_digit) {
+						*str++ = result; /* won't write past allotted memory, because of buffer check at top of loop */
+						*str = 0;
+					}
+					/* we received a ST digit (ST, STP, ST2P, or ST3P), so we're done */
+					if (is_start_digit) {
+						pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "START");
+						ast_frfree(frame);
+						break;
+					}
+					/* only free frame if it was a DSP match. The MF itself should not be muted. */
+					ast_frfree(frame);
+				}
+			}
+		} else {
+			pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "HANGUP");
+			res = -1;
+		}
+	}
+	ast_dsp_free(dsp);
+	ast_debug(3, "channel '%s' - event loop stopped { timeout: %d, remaining_time: %d }\n", ast_channel_name(chan), timeout, remaining_time);
+	return res;
+}
+
+static int read_mf_exec(struct ast_channel *chan, const char *data)
+{
+#define BUFFER_SIZE 256
+	char tmp[BUFFER_SIZE] = "";
+	int to = 0;
+	double tosec;
+	struct ast_flags flags = {0};
+	char *optargs[OPT_ARG_ARRAY_SIZE];
+	char *argcopy = NULL;
+	int res, features = 0, maxdigits = 0;
+
+	AST_DECLARE_APP_ARGS(arglist,
+		AST_APP_ARG(variable);
+		AST_APP_ARG(timeout);
+		AST_APP_ARG(options);
+	);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "ReceiveMF requires an argument (variable)\n");
+		return -1;
+	}
+
+	argcopy = ast_strdupa(data);
+
+	AST_STANDARD_APP_ARGS(arglist, argcopy);
+
+	if (!ast_strlen_zero(arglist.options)) {
+		ast_app_parse_options(read_app_options, &flags, optargs, arglist.options);
+	}
+
+	if (!ast_strlen_zero(arglist.timeout)) {
+		tosec = atof(arglist.timeout);
+		if (tosec <= 0) {
+			to = 0;
+		} else {
+			to = tosec * 1000.0;
+		}
+	}
+
+	if (ast_strlen_zero(arglist.variable)) {
+		ast_log(LOG_WARNING, "Invalid! Usage: ReceiveMF(variable[,timeout][,option])\n");
+		return -1;
+	}
+	if (ast_test_flag(&flags, OPT_MAXDIGITS) && !ast_strlen_zero(optargs[OPT_ARG_MAXDIGITS])) {
+		maxdigits = atoi(optargs[OPT_ARG_MAXDIGITS]);
+		if (maxdigits <= 0) {
+			ast_log(LOG_WARNING, "Invalid maximum number of digits, ignoring: '%s'\n", optargs[OPT_ARG_MAXDIGITS]);
+			maxdigits = 0;
+		}
+	}
+
+	if (ast_test_flag(&flags, OPT_DELAY)) {
+		features |= DSP_DIGITMODE_MUTEMAX;
+	}
+
+	if (ast_test_flag(&flags, OPT_MUTE)) {
+		features |= DSP_DIGITMODE_MUTECONF;
+	}
+
+	if (!ast_test_flag(&flags, OPT_QUELCH)) {
+		features |= DSP_DIGITMODE_NOQUELCH;
+	}
+
+	if (ast_test_flag(&flags, OPT_RELAXED)) {
+		features |= DSP_DIGITMODE_RELAXDTMF;
+	}
+
+	res = read_mf_digits(chan, tmp, BUFFER_SIZE, to, features, (ast_test_flag(&flags, OPT_LAX_KP)),
+		(ast_test_flag(&flags, OPT_KP_OVERRIDE)), (ast_test_flag(&flags, OPT_NO_KP)), (ast_test_flag(&flags, OPT_NO_ST)), maxdigits);
+	pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
+	if (!ast_strlen_zero(tmp)) {
+		ast_verb(3, "MF digits received: '%s'\n", tmp);
+	} else if (!res) { /* if channel hung up, don't print anything out */
+		ast_verb(3, "No MF digits received.\n");
+	}
+	return res;
+}
+
+static int sendmf_exec(struct ast_channel *chan, const char *vdata)
+{
+	int res;
+	char *data;
+	int dinterval = 0, duration = 0, durationkp = 0, durationst = 0;
+	struct ast_channel *chan_found = NULL;
+	struct ast_channel *chan_dest = chan;
+	struct ast_channel *chan_autoservice = NULL;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(digits);
+		AST_APP_ARG(dinterval);
+		AST_APP_ARG(duration);
+		AST_APP_ARG(durationkp);
+		AST_APP_ARG(durationst);
+		AST_APP_ARG(channel);
+	);
+
+	if (ast_strlen_zero(vdata)) {
+		ast_log(LOG_WARNING, "SendMF requires an argument\n");
+		return 0;
+	}
+
+	data = ast_strdupa(vdata);
+	AST_STANDARD_APP_ARGS(args, data);
+
+	if (ast_strlen_zero(args.digits)) {
+		ast_log(LOG_WARNING, "The digits argument is required (0-9,*#ABC,wf)\n");
+		return 0;
+	}
+	if (!ast_strlen_zero(args.dinterval)) {
+		ast_app_parse_timelen(args.dinterval, &dinterval, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.duration)) {
+		ast_app_parse_timelen(args.duration, &duration, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.durationkp)) {
+		ast_app_parse_timelen(args.durationkp, &durationkp, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.durationst)) {
+		ast_app_parse_timelen(args.durationst, &durationst, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.channel)) {
+		chan_found = ast_channel_get_by_name(args.channel);
+		if (!chan_found) {
+			ast_log(LOG_WARNING, "No such channel: %s\n", args.channel);
+			return 0;
+		}
+		chan_dest = chan_found;
+		if (chan_found != chan) {
+			chan_autoservice = chan;
+		}
+	}
+	res = ast_mf_stream(chan_dest, chan_autoservice, NULL, args.digits, dinterval <= 0 ? MF_BETWEEN_MS : dinterval,
+		duration <= 0 ? MF_DURATION : duration, durationkp <= 0 ? MF_KP_DURATION : durationkp,
+		durationst <= 0 ? MF_ST_DURATION : durationst, 0);
+	ast_channel_cleanup(chan_found);
+
+	return chan_autoservice ? 0 : res;
+}
+
+static int manager_play_mf(struct mansession *s, const struct message *m)
+{
+	const char *channel = astman_get_header(m, "Channel");
+	const char *digit = astman_get_header(m, "Digit");
+	const char *duration = astman_get_header(m, "Duration");
+	struct ast_channel *chan;
+	unsigned int duration_ms = MF_DURATION;
+
+	if (!(chan = ast_channel_get_by_name(channel))) {
+		astman_send_error(s, m, "Channel not found");
+		return 0;
+	}
+
+	if (ast_strlen_zero(digit)) {
+		astman_send_error(s, m, "No digit specified");
+		chan = ast_channel_unref(chan);
+		return 0;
+	}
+
+	/* Override default duration with KP or ST-specific default durations */
+	if (!strcmp(digit, "*"))
+		duration_ms = MF_KP_DURATION;
+
+	if (!strcmp(digit, "#") || !strcmp(digit, "A") || !strcmp(digit, "B") || !strcmp(digit, "C"))
+		duration_ms = MF_ST_DURATION;
+
+	if (!ast_strlen_zero(duration) && (sscanf(duration, "%30u", &duration_ms) != 1)) {
+		astman_send_error(s, m, "Could not convert Duration parameter");
+		chan = ast_channel_unref(chan);
+		return 0;
+	}
+
+	ast_mf_stream(chan, NULL, NULL, digit, 0, duration_ms, duration_ms, duration_ms, 1);
+
+	chan = ast_channel_unref(chan);
+
+	astman_send_ack(s, m, "MF successfully queued");
+
+	return 0;
+}
+
+static int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(readmf_name);
+	res |= ast_unregister_application(sendmf_name);
+	res |= ast_manager_unregister("PlayMF");
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	res = ast_register_application_xml(readmf_name, read_mf_exec);
+	res |= ast_register_application_xml(sendmf_name, sendmf_exec);
+	res |= ast_manager_register_xml("PlayMF", EVENT_FLAG_CALL, manager_play_mf);
+
+	return res;
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "MF Sender and Receiver Applications");

apps/app_milliwatt.c

@@ -40,19 +40,29 @@
 /*** DOCUMENTATION
 	<application name="Milliwatt" language="en_US">
 		<synopsis>
-			Generate a Constant 1004Hz tone at 0dbm (mu-law).
+			Generates a 1004 Hz test tone at 0dbm (mu-law).
 		</synopsis>
 		<syntax>
 			<parameter name="options">
 				<optionlist>
+					<option name="m">
+						<para>Generate a 1004 Hz Milliwatt test tone at 0dbm, with a
+						1 second silent interval. This option must be specified
+						if you are using this for a milliwatt test line.</para>
+					</option>
 					<option name="o">
-						<para>Generate the tone at 1000Hz like previous version.</para>
+						<para>Generate a constant tone at 1000 Hz like previous version.</para>
 					</option>
 				</optionlist>
 			</parameter>
 		</syntax>
 		<description>
-			<para>Previous versions of this application generated the tone at 1000Hz.  If for
+			<para>Generates a 1004 Hz test tone.</para>
+			<para>By default, this application does not provide a Milliwatt test tone. It simply
+			plays a 1004 Hz tone, which is not suitable for performing a milliwatt test.
+			The <literal>m</literal> option should be used so that a real Milliwatt test tone
+			is provided. This will include a 1 second silent interval every 10 seconds.</para>
+			<para>Previous versions of this application generated a constant tone at 1000 Hz. If for
 			some reason you would prefer that behavior, supply the <literal>o</literal> option to get the
 			old behavior.</para>
 		</description>
@@ -155,8 +165,11 @@ static int milliwatt_exec(struct ast_channel *chan, const char *data)
 	if (!ast_strlen_zero(options) && strchr(options, 'o')) {
 		return old_milliwatt_exec(chan);
 	}
-
-	res = ast_playtones_start(chan, 23255, "1004/1000", 0);
+	if (!ast_strlen_zero(options) && strchr(options, 'm')) {
+		res = ast_playtones_start(chan, 23255, "1004/9000,0/1000", 0);
+	} else {
+		res = ast_playtones_start(chan, 23255, "1004/1000", 0);
+	}
 
 	while (!res) {
 		res = ast_safe_sleep(chan, 10000);

apps/app_minivm.c

@@ -26,7 +26,7 @@
  * based on the Comedian Mail voicemail system (app_voicemail.c).
  *
  * \par See also
- * \arg \ref Config_minivm_examples
+ * \arg \ref minivm.conf "Config_minivm"
  * \arg \ref App_minivm
  *
  * \ingroup applications
@@ -78,16 +78,16 @@
  *		- English, GB		en_gb
  *
  * \par See also
- * \arg \ref Config_minivm
+ * \arg \ref minivm.conf "Config_minivm"
  * \arg \ref Config_minivm_examples
  * \arg \ref Minivm_directories
  * \arg \ref app_minivm.c
  * \arg Comedian mail: app_voicemail.c
- * \arg \ref descrip_minivm_accmess
- * \arg \ref descrip_minivm_greet
- * \arg \ref descrip_minivm_record
- * \arg \ref descrip_minivm_delete
- * \arg \ref descrip_minivm_notify
+ * \arg \ref minivm_accmess_exec
+ * \arg \ref minivm_greet_exec
+ * \arg \ref minivm_record_exec
+ * \arg \ref minivm_delete_exec
+ * \arg \ref minivm_notify_exec
  *
  * \arg \ref App_minivm_todo
  */
@@ -113,6 +113,13 @@
  * Back: \ref App_minivm
  */
 
+/*!
+ * \page minivm.conf minivm.conf
+ * \verbinclude minivm.conf.sample
+ *
+ * Back: \ref App_minivm
+ */
+
 /*! \page Config_minivm_examples Example dialplan for Mini-Voicemail
  * \section Example dialplan scripts for Mini-Voicemail
  *  \verbinclude extensions_minivm.conf.sample
@@ -154,7 +161,6 @@
 #include <dirent.h>
 #include <locale.h>
 
-
 #include "asterisk/paths.h"	/* use various paths */
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
@@ -535,8 +541,6 @@
 #define SENDMAIL "/usr/sbin/sendmail -t"
 
 #define SOUND_INTRO		"vm-intro"
-#define B64_BASEMAXINLINE 	256	/*!< Buffer size for Base 64 attachment encoding */
-#define B64_BASELINELEN 	72	/*!< Line length for Base 64 endoded messages */
 #define EOL			"\r\n"
 
 #define MAX_DATETIME_FORMAT	512
@@ -652,15 +656,6 @@ struct leave_vm_options {
 	signed char record_gain;
 };
 
-/*! \brief Structure for base64 encoding */
-struct b64_baseio {
-	int iocp;
-	int iolen;
-	int linelength;
-	int ateof;
-	unsigned char iobuf[B64_BASEMAXINLINE];
-};
-
 /*! \brief Voicemail time zones */
 struct minivm_zone {
 	char name[80];				/*!< Name of this time zone */
@@ -846,134 +841,6 @@ static void message_destroy_list(void)
 	AST_LIST_UNLOCK(&message_templates);
 }
 
-/*!\internal
- * \brief read buffer from file (base64 conversion) */
-static int b64_inbuf(struct b64_baseio *bio, FILE *fi)
-{
-	int l;
-
-	if (bio->ateof)
-		return 0;
-
-	if ((l = fread(bio->iobuf, 1, B64_BASEMAXINLINE, fi)) != B64_BASEMAXINLINE) {
-		bio->ateof = 1;
-		if (l == 0) {
-			/* Assume EOF */
-			return 0;
-		}
-	}
-
-	bio->iolen = l;
-	bio->iocp = 0;
-
-	return 1;
-}
-
-/*!\internal
- * \brief read character from file to buffer (base64 conversion) */
-static int b64_inchar(struct b64_baseio *bio, FILE *fi)
-{
-	if (bio->iocp >= bio->iolen) {
-		if (!b64_inbuf(bio, fi))
-			return EOF;
-	}
-
-	return bio->iobuf[bio->iocp++];
-}
-
-/*!\internal
- * \brief write buffer to file (base64 conversion) */
-static int b64_ochar(struct b64_baseio *bio, int c, FILE *so)
-{
-	if (bio->linelength >= B64_BASELINELEN) {
-		if (fputs(EOL,so) == EOF)
-			return -1;
-
-		bio->linelength= 0;
-	}
-
-	if (putc(((unsigned char) c), so) == EOF)
-		return -1;
-
-	bio->linelength++;
-
-	return 1;
-}
-
-/*!\internal
- * \brief Encode file to base64 encoding for email attachment (base64 conversion) */
-static int base_encode(char *filename, FILE *so)
-{
-	unsigned char dtable[B64_BASEMAXINLINE];
-	int i,hiteof= 0;
-	FILE *fi;
-	struct b64_baseio bio;
-
-	memset(&bio, 0, sizeof(bio));
-	bio.iocp = B64_BASEMAXINLINE;
-
-	if (!(fi = fopen(filename, "rb"))) {
-		ast_log(LOG_WARNING, "Failed to open file: %s: %s\n", filename, strerror(errno));
-		return -1;
-	}
-
-	for (i= 0; i<9; i++) {
-		dtable[i]= 'A'+i;
-		dtable[i+9]= 'J'+i;
-		dtable[26+i]= 'a'+i;
-		dtable[26+i+9]= 'j'+i;
-	}
-	for (i= 0; i < 8; i++) {
-		dtable[i+18]= 'S'+i;
-		dtable[26+i+18]= 's'+i;
-	}
-	for (i= 0; i < 10; i++) {
-		dtable[52+i]= '0'+i;
-	}
-	dtable[62]= '+';
-	dtable[63]= '/';
-
-	while (!hiteof){
-		unsigned char igroup[3], ogroup[4];
-		int c,n;
-
-		igroup[0]= igroup[1]= igroup[2]= 0;
-
-		for (n= 0; n < 3; n++) {
-			if ((c = b64_inchar(&bio, fi)) == EOF) {
-				hiteof= 1;
-				break;
-			}
-			igroup[n]= (unsigned char)c;
-		}
-
-		if (n> 0) {
-			ogroup[0]= dtable[igroup[0]>>2];
-			ogroup[1]= dtable[((igroup[0]&3)<<4) | (igroup[1]>>4)];
-			ogroup[2]= dtable[((igroup[1]&0xF)<<2) | (igroup[2]>>6)];
-			ogroup[3]= dtable[igroup[2]&0x3F];
-
-			if (n<3) {
-				ogroup[3]= '=';
-
-				if (n<2)
-					ogroup[2]= '=';
-			}
-
-			for (i= 0;i<4;i++)
-				b64_ochar(&bio, ogroup[i], so);
-		}
-	}
-
-	/* Put end of line - line feed */
-	if (fputs(EOL, so) == EOF)
-		return 0;
-
-	fclose(fi);
-
-	return 1;
-}
-
 static int get_date(char *s, int len)
 {
 	struct ast_tm tm;
@@ -1474,7 +1341,7 @@ static int sendmail(struct minivm_template *template, struct minivm_account *vmu
 		fprintf(p, "Content-Description: Voicemail sound attachment.\n");
 		fprintf(p, "Content-Disposition: attachment; filename=\"voicemail%s.%s\"\n\n", counter ? counter : "", format);
 
-		base_encode(fname, p);
+		ast_base64_encode_file_path(fname, p, EOL);
 		fprintf(p, "\n\n--%s--\n.\n", bound);
 	}
 	fclose(p);
@@ -1737,7 +1604,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
 			}
 			return cmd;
  		default:
-			/* If the caller is an ouside caller, and the review option is enabled,
+			/* If the caller is an outside caller, and the review option is enabled,
 			   allow them to review the message, but let the owner of the box review
 			   their OGM's */
 			if (outsidecaller && !ast_test_flag(vmu, MVM_REVIEW))
@@ -2827,7 +2694,7 @@ static int apply_general_options(struct ast_variable *var)
 		} else if (!strcmp(var->name, "externnotify")) {
 			/* External voicemail notify application */
 			ast_copy_string(global_externnotify, var->value, sizeof(global_externnotify));
-		} else if (!strcmp(var->name, "silencetreshold")) {
+		} else if (!strcmp(var->name, "silencethreshold") || !strcmp(var->name, "silencetreshold")) {
 			/* Silence treshold */
 			global_silencethreshold = atoi(var->value);
 		} else if (!strcmp(var->name, "maxmessage")) {
@@ -2956,6 +2823,8 @@ static int load_config(int reload)
 		ast_copy_string(template->fromaddress, chanvar, sizeof(template->fromaddress));
 	if ((chanvar = ast_variable_retrieve(cfg, "general", "emailaaddress")))
 		ast_copy_string(template->serveremail, chanvar, sizeof(template->serveremail));
+	if ((chanvar = ast_variable_retrieve(cfg, "general", "emailaddress")))
+		ast_copy_string(template->serveremail, chanvar, sizeof(template->serveremail));
 	if ((chanvar = ast_variable_retrieve(cfg, "general", "emailcharset")))
 		ast_copy_string(template->charset, chanvar, sizeof(template->charset));
 	if ((chanvar = ast_variable_retrieve(cfg, "general", "emailsubject")))

apps/app_mixmonitor.c

@@ -51,6 +51,8 @@
 #include "asterisk/channel.h"
 #include "asterisk/autochan.h"
 #include "asterisk/manager.h"
+#include "asterisk/stasis.h"
+#include "asterisk/stasis_channels.h"
 #include "asterisk/callerid.h"
 #include "asterisk/mod_format.h"
 #include "asterisk/linkedlists.h"
@@ -88,6 +90,16 @@
 						<para>Play a periodic beep while this call is being recorded.</para>
 						<argument name="interval"><para>Interval, in seconds. Default is 15.</para></argument>
 					</option>
+					<option name="c">
+						<para>Use the real Caller ID from the channel for the voicemail Caller ID.</para>
+						<para>By default, the Connected Line is used. If you want the channel caller's
+						real number, you may need to specify this option.</para>
+					</option>
+					<option name="d">
+						<para>Delete the recording file as soon as MixMonitor is done with it.</para>
+						<para>For example, if you use the m option to dispatch the recording to a voicemail box,
+						you can specify this option to delete the original copy of it afterwards.</para>
+					</option>
 					<option name="v">
 						<para>Adjust the <emphasis>heard</emphasis> volume by a factor of <replaceable>x</replaceable>
 						(range <literal>-4</literal> to <literal>4</literal>)</para>
@@ -295,6 +307,51 @@
 			</parameter>
 		</syntax>
 	</function>
+	<managerEvent language="en_US" name="MixMonitorStart">
+		<managerEventInstance class="EVENT_FLAG_CALL">
+			<synopsis>Raised when monitoring has started on a channel.</synopsis>
+			<syntax>
+				<channel_snapshot/>
+			</syntax>
+			<see-also>
+				<ref type="managerEvent">MixMonitorStop</ref>
+				<ref type="application">MixMonitor</ref>
+				<ref type="manager">MixMonitor</ref>
+			</see-also>
+		</managerEventInstance>
+	</managerEvent>
+	<managerEvent language="en_US" name="MixMonitorStop">
+		<managerEventInstance class="EVENT_FLAG_CALL">
+		<synopsis>Raised when monitoring has stopped on a channel.</synopsis>
+		<syntax>
+			<channel_snapshot/>
+		</syntax>
+		<see-also>
+			<ref type="managerEvent">MixMonitorStart</ref>
+			<ref type="application">StopMixMonitor</ref>
+			<ref type="manager">StopMixMonitor</ref>
+		</see-also>
+		</managerEventInstance>
+	</managerEvent>
+	<managerEvent language="en_US" name="MixMonitorMute">
+		<managerEventInstance class="EVENT_FLAG_CALL">
+		<synopsis>Raised when monitoring is muted or unmuted on a channel.</synopsis>
+		<syntax>
+			<channel_snapshot/>
+			<parameter name="Direction">
+				<para>Which part of the recording was muted or unmuted: read, write or both
+				(from channel, to channel or both directions).</para>
+			</parameter>
+			<parameter name="State">
+				<para>If the monitoring was muted or unmuted: 1 when muted, 0 when unmuted.</para>
+			</parameter>
+		</syntax>
+		<see-also>
+			<ref type="manager">MixMonitorMute</ref>
+		</see-also>
+		</managerEventInstance>
+	</managerEvent>
+
 
  ***/
 
@@ -360,6 +417,8 @@ enum mixmonitor_flags {
 	MUXFLAG_BEEP_STOP = (1 << 13),
 	MUXFLAG_DEPRECATED_RWSYNC = (1 << 14),
 	MUXFLAG_NO_RWSYNC = (1 << 15),
+	MUXFLAG_AUTO_DELETE = (1 << 16),
+	MUXFLAG_REAL_CALLERID = (1 << 17),
 };
 
 enum mixmonitor_args {
@@ -380,6 +439,8 @@ AST_APP_OPTIONS(mixmonitor_opts, {
 	AST_APP_OPTION('a', MUXFLAG_APPEND),
 	AST_APP_OPTION('b', MUXFLAG_BRIDGED),
 	AST_APP_OPTION_ARG('B', MUXFLAG_BEEP, OPT_ARG_BEEP_INTERVAL),
+	AST_APP_OPTION('c', MUXFLAG_REAL_CALLERID),
+	AST_APP_OPTION('d', MUXFLAG_AUTO_DELETE),
 	AST_APP_OPTION('p', MUXFLAG_BEEP_START),
 	AST_APP_OPTION('P', MUXFLAG_BEEP_STOP),
 	AST_APP_OPTION_ARG('v', MUXFLAG_READVOLUME, OPT_ARG_READVOLUME),
@@ -584,6 +645,7 @@ static void mixmonitor_free(struct mixmonitor *mixmonitor)
  * \brief Copies the mixmonitor to all voicemail recipients
  * \param mixmonitor The mixmonitor that needs to forward its file to recipients
  * \param ext Format of the file that was saved
+ * \param filename
  */
 static void copy_to_voicemail(struct mixmonitor *mixmonitor, const char *ext, const char *filename)
 {
@@ -812,6 +874,19 @@ static void *mixmonitor_thread(void *obj)
 		ast_debug(3, "No recipients to forward monitor to, moving on.\n");
 	}
 
+	if (ast_test_flag(mixmonitor, MUXFLAG_AUTO_DELETE)) {
+		ast_debug(3, "Deleting our copies of recording files\n");
+		if (!ast_strlen_zero(fs_ext)) {
+			ast_filedelete(mixmonitor->filename, fs_ext);
+		}
+		if (!ast_strlen_zero(fs_read_ext)) {
+			ast_filedelete(mixmonitor->filename_read, fs_ext);
+		}
+		if (!ast_strlen_zero(fs_write_ext)) {
+			ast_filedelete(mixmonitor->filename_write, fs_ext);
+		}
+	}
+
 	mixmonitor_free(mixmonitor);
 
 	ast_module_unref(ast_module_info->self);
@@ -865,6 +940,24 @@ static int setup_mixmonitor_ds(struct mixmonitor *mixmonitor, struct ast_channel
 	return 0;
 }
 
+static void mixmonitor_ds_remove_and_free(struct ast_channel *chan, const char *datastore_id)
+{
+	struct ast_datastore *datastore;
+
+	ast_channel_lock(chan);
+
+	datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, datastore_id);
+
+	/*
+	 * Currently the one place this function is called from guarantees a
+	 * datastore is present, thus return checks can be avoided here.
+	 */
+	ast_channel_datastore_remove(chan, datastore);
+	ast_datastore_free(datastore);
+
+	ast_channel_unlock(chan);
+}
+
 static int launch_monitor_thread(struct ast_channel *chan, const char *filename,
 				  unsigned int flags, int readvol, int writevol,
 				  const char *post_process, const char *filename_write,
@@ -940,7 +1033,6 @@ static int launch_monitor_thread(struct ast_channel *chan, const char *filename,
 			pbx_builtin_setvar_helper(chan, uid_channel_var, datastore_id);
 		}
 	}
-	ast_free(datastore_id);
 
 	mixmonitor->name = ast_strdup(ast_channel_name(chan));
 
@@ -950,20 +1042,37 @@ static int launch_monitor_thread(struct ast_channel *chan, const char *filename,
 
 	if (!ast_strlen_zero(recipients)) {
 		char callerid[256];
-		struct ast_party_connected_line *connected;
 
 		ast_channel_lock(chan);
 
-		/* We use the connected line of the invoking channel for caller ID. */
+		/* We use the connected line of the invoking channel for caller ID,
+		 * unless we've been told to use the Caller ID.
+		 * The initial use for this relied on Connected Line to get the
+		 * actual number for recording with Digium phones,
+		 * but in generic use the Caller ID is likely what people want.
+		 */
 
-		connected = ast_channel_connected(chan);
-		ast_debug(3, "Connected Line CID = %d - %s : %d - %s\n", connected->id.name.valid,
-			connected->id.name.str, connected->id.number.valid,
-			connected->id.number.str);
-		ast_callerid_merge(callerid, sizeof(callerid),
-			S_COR(connected->id.name.valid, connected->id.name.str, NULL),
-			S_COR(connected->id.number.valid, connected->id.number.str, NULL),
-			"Unknown");
+		if (ast_test_flag(mixmonitor, MUXFLAG_REAL_CALLERID)) {
+			struct ast_party_caller *caller;
+			caller = ast_channel_caller(chan);
+			ast_debug(3, "Caller ID = %d - %s : %d - %s\n", caller->id.name.valid,
+				caller->id.name.str, caller->id.number.valid,
+				caller->id.number.str);
+			ast_callerid_merge(callerid, sizeof(callerid),
+				S_COR(caller->id.name.valid, caller->id.name.str, NULL),
+				S_COR(caller->id.number.valid, caller->id.number.str, NULL),
+				"Unknown");
+		} else {
+			struct ast_party_connected_line *connected;
+			connected = ast_channel_connected(chan);
+			ast_debug(3, "Connected Line CID = %d - %s : %d - %s\n", connected->id.name.valid,
+				connected->id.name.str, connected->id.number.valid,
+				connected->id.number.str);
+			ast_callerid_merge(callerid, sizeof(callerid),
+				S_COR(connected->id.name.valid, connected->id.name.str, NULL),
+				S_COR(connected->id.number.valid, connected->id.number.str, NULL),
+				"Unknown");
+		}
 
 		ast_string_field_set(mixmonitor, call_context, ast_channel_context(chan));
 		ast_string_field_set(mixmonitor, call_macrocontext, ast_channel_macrocontext(chan));
@@ -990,12 +1099,16 @@ static int launch_monitor_thread(struct ast_channel *chan, const char *filename,
 	if (startmon(chan, &mixmonitor->audiohook)) {
 		ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
 			mixmonitor_spy_type, ast_channel_name(chan));
+		mixmonitor_ds_remove_and_free(chan, datastore_id);
+		ast_free(datastore_id);
 		ast_autochan_destroy(mixmonitor->autochan);
 		ast_audiohook_destroy(&mixmonitor->audiohook);
 		mixmonitor_free(mixmonitor);
 		return -1;
 	}
 
+	ast_free(datastore_id);
+
 	/* reference be released at mixmonitor destruction */
 	mixmonitor->callid = ast_read_threadstorage_callid();
 
@@ -1056,6 +1169,7 @@ static int mixmonitor_exec(struct ast_channel *chan, const char *data)
 	struct ast_flags flags = { 0 };
 	char *recipients = NULL;
 	char *parse;
+	RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(filename);
 		AST_APP_ARG(options);
@@ -1176,6 +1290,12 @@ static int mixmonitor_exec(struct ast_channel *chan, const char *data)
 		ast_module_unref(ast_module_info->self);
 	}
 
+	message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan),
+		ast_channel_mixmonitor_start_type(), NULL);
+	if (message) {
+		stasis_publish(ast_channel_topic(chan), message);
+	}
+
 	return 0;
 }
 
@@ -1185,6 +1305,7 @@ static int stop_mixmonitor_full(struct ast_channel *chan, const char *data)
 	char *parse = "";
 	struct mixmonitor_ds *mixmonitor_ds;
 	const char *beep_id = NULL;
+	RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
 
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(mixmonid);
@@ -1242,6 +1363,13 @@ static int stop_mixmonitor_full(struct ast_channel *chan, const char *data)
 		ast_beep_stop(chan, beep_id);
 	}
 
+	message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan),
+	                                             ast_channel_mixmonitor_stop_type(),
+	                                             NULL);
+	if (message) {
+		stasis_publish(ast_channel_topic(chan), message);
+	}
+
 	return 0;
 }
 
@@ -1329,6 +1457,8 @@ static int manager_mute_mixmonitor(struct mansession *s, const struct message *m
 	const char *direction = astman_get_header(m,"Direction");
 	int clearmute = 1;
 	enum ast_audiohook_flags flag;
+	RAII_VAR(struct stasis_message *, stasis_message, NULL, ao2_cleanup);
+	RAII_VAR(struct ast_json *, stasis_message_blob, NULL, ast_json_unref);
 
 	if (ast_strlen_zero(direction)) {
 		astman_send_error(s, m, "No direction specified. Must be read, write or both");
@@ -1370,6 +1500,17 @@ static int manager_mute_mixmonitor(struct mansession *s, const struct message *m
 		return AMI_SUCCESS;
 	}
 
+	stasis_message_blob = ast_json_pack("{s: s, s: b}",
+		"direction", direction,
+		"state", ast_true(state));
+
+	stasis_message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(c),
+		ast_channel_mixmonitor_mute_type(), stasis_message_blob);
+
+	if (stasis_message) {
+		stasis_publish(ast_channel_topic(c), stasis_message);
+	}
+
 	astman_append(s, "Response: Success\r\n");
 
 	if (!ast_strlen_zero(id)) {

apps/app_morsecode.c

@@ -3,6 +3,8 @@
  *
  * Copyright (c) 2006, Tilghman Lesher.  All rights reserved.
  *
+ * Updated by Naveen Albert <asterisk@phreaknet.org>
+ *
  * Tilghman Lesher <app_morsecode__v001@the-tilghman.com>
  *
  * This code is released by the author with no restrictions on usage.
@@ -20,6 +22,7 @@
  * \brief Morsecode application
  *
  * \author Tilghman Lesher <app_morsecode__v001@the-tilghman.com>
+ * \author Naveen Albert <asterisk@phreaknet.org>
  *
  * \ingroup applications
  */
@@ -58,6 +61,14 @@
 				<variable name="MORSETONE">
 					<para>The pitch of the tone in (Hz), default is 800</para>
 				</variable>
+				<variable name="MORSESPACETONE">
+					<para>The pitch of the spaces in (Hz), default is 0</para>
+				</variable>
+				<variable name="MORSETYPE">
+					<para>The code type to use (AMERICAN for standard American Morse
+					or INTERNATIONAL for international code.
+					Default is INTERNATIONAL).</para>
+				</variable>
 			</variablelist>
 		</description>
 		<see-also>
@@ -68,7 +79,7 @@
  ***/
 static const char app_morsecode[] = "Morsecode";
 
-static const char * const morsecode[] = {
+static const char * const internationalcode[] = {
 	"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /*  0-15 */
 	"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /* 16-31 */
 	" ",      /* 32 - <space> */
@@ -95,16 +106,16 @@ static const char * const morsecode[] = {
 	"",       /* 62 - > */
 	"..--..", /* 63 - ? */
 	".--.-.", /* 64 - @ */
-	".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--",
-	"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..",
+	".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--", /* A-M */
+	"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..", /* N-Z */
 	"-.--.-", /* 91 - [ (really '(') */
 	"-..-.",  /* 92 - \ (really '/') */
 	"-.--.-", /* 93 - ] (really ')') */
 	"",       /* 94 - ^ */
 	"..--.-", /* 95 - _ */
 	".----.", /* 96 - ` */
-	".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--",
-	"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..",
+	".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--", /* a-m */
+	"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..", /* n-z */
 	"-.--.-", /* 123 - { (really '(') */
 	"",       /* 124 - | */
 	"-.--.-", /* 125 - } (really ')') */
@@ -112,63 +123,159 @@ static const char * const morsecode[] = {
 	". . .",  /* 127 - <del> (error) */
 };
 
-static void playtone(struct ast_channel *chan, int tone, int len)
+static const char * const americanmorsecode[] = {
+	"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /*  0-15 */
+	"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /* 16-31 */
+	"  ",    /* 32 - <space> */
+	"---.",   /* 33 - ! */
+	"..-. -.",/* 34 - " (QN)*/
+	"",       /* 35 - # */
+	"... .-..",/* 36 - $ (SX) */
+	"",       /* 37 - % */
+	". ...",  /* 38 - & (ES) */
+	"..-. .-..",/* 39 - ' (QX) */
+	"..... -.", /* 40 - ( (PN) */
+	"..... .. ..", /* 41 - ) (PY) */
+	"",       /* 42 - * */
+	"",       /* 43 - + */
+	".-.-",   /* 44 - , */
+	".... .-..",/* 45 - (HX) */
+	"..--..", /* 46 - . */
+	"..- -",  /* 47 - / (UT) */
+	".--.", "..-..", "...-.", "....-", "---", "......", "--..", "-....", "-..-", "0", /* 48-57 - 0-9 */
+	"-.- . .",/* 58 - : (KO) */
+	"... ..", /* 59 - ; */
+	"",       /* 60 - < */
+	"-...-",  /* 61 - = (paragraph mark) */
+	"",       /* 62 - > */
+	"-..-.",  /* 63 - ? */
+	".--.-.", /* 64 - @ */
+	".-", "-...", ".. .", "-..", ".", ".-.", "--.", "....", "..", ".-.-", "-.-", "L", "--", /* A-M */
+	"-.", ". .", ".....", "..-.", ". ..", "...", "-", "..-", "...-", ".--", ".-..", ".. ..", "... .", /* N-Z */
+	"..... -.", /* 91 - [ (really '(') */
+	"..- -",  /* 92 - \ (really '/') */
+	"..... .. ..", /* 93 - ] (really ')') */
+	"",       /* 94 - ^ */
+	"..--.-", /* 95 - _ */
+	".----.", /* 96 - ` */
+	".-", "-...", ".. .", "-..", ".", ".-.", "--.", "....", "..", ".-.-", "-.-", "L", "--", /* a-m */
+	"-.", ". .", ".....", "..-.", ". ..", "...", "-", "..-", "...-", ".--", ".-..", ".. ..", "... .", /* n-z */
+	"..... -.", /* 123 - { (really '(') */
+	"",       /* 124 - | */
+	"..... .. ..", /* 125 - } (really ')') */
+	"..- -",  /* 126 - ~ (really bar) */
+	". . .",  /* 127 - <del> (error) */
+};
+
+static int playtone(struct ast_channel *chan, int tone, int len)
 {
+	int res;
 	char dtmf[20];
 	snprintf(dtmf, sizeof(dtmf), "%d/%d", tone, len);
 	ast_playtones_start(chan, 0, dtmf, 0);
-	ast_safe_sleep(chan, len);
+	res = ast_safe_sleep(chan, len);
 	ast_playtones_stop(chan);
+	return res;
 }
 
 static int morsecode_exec(struct ast_channel *chan, const char *data)
 {
-	int res=0, ditlen, tone;
+	int res = 0, ditlen, tone, toneoff, digit2;
 	const char *digit;
-	const char *ditlenc, *tonec;
+	const char *ditlenc, *tonec, *toneb, *codetype;
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Syntax: Morsecode(<string>) - no argument found\n");
 		return 0;
 	}
 
-	/* Use variable MORESEDITLEN, if set (else 80) */
 	ast_channel_lock(chan);
+	/* Use variable MORESEDITLEN, if set (else 80) */
 	ditlenc = pbx_builtin_getvar_helper(chan, "MORSEDITLEN");
 	if (ast_strlen_zero(ditlenc) || (sscanf(ditlenc, "%30d", &ditlen) != 1)) {
 		ditlen = 80;
 	}
-	ast_channel_unlock(chan);
 
 	/* Use variable MORSETONE, if set (else 800) */
-	ast_channel_lock(chan);
 	tonec = pbx_builtin_getvar_helper(chan, "MORSETONE");
 	if (ast_strlen_zero(tonec) || (sscanf(tonec, "%30d", &tone) != 1)) {
 		tone = 800;
 	}
+
+	/* Use variable MORSESPACETONE, if set (else 0) */
+	toneb = pbx_builtin_getvar_helper(chan, "MORSESPACETONE");
+	if (ast_strlen_zero(toneb) || (sscanf(toneb, "%30d", &toneoff) != 1)) {
+		toneoff = 0;
+	}
+
+	/* Use variable MORSETYPE, if set (else INTERNATIONAL) */
+	codetype = pbx_builtin_getvar_helper(chan, "MORSETYPE");
+	if (!codetype || strcmp(codetype, "AMERICAN")) {
+		codetype = "INTERNATIONAL";
+	}
+
 	ast_channel_unlock(chan);
-
-	for (digit = data; *digit; digit++) {
-		int digit2 = *digit;
-		const char *dahdit;
-		if (digit2 < 0) {
-			continue;
-		}
-		for (dahdit = morsecode[digit2]; *dahdit; dahdit++) {
-			if (*dahdit == '-') {
-				playtone(chan, tone, 3 * ditlen);
-			} else if (*dahdit == '.') {
-				playtone(chan, tone, 1 * ditlen);
-			} else {
-				/* Account for ditlen of silence immediately following */
-				playtone(chan, 0, 2 * ditlen);
+	if (!strcmp(codetype, "AMERICAN")) {
+		for (digit = data; *digit; digit++) {
+			const char *dahdit;
+			digit2 = *digit;
+			if (digit2 < 0 || digit2 > 127) {
+				continue;
 			}
+			for (dahdit = americanmorsecode[digit2]; *dahdit; dahdit++) {
+				if (*dahdit == '-') {
+					res = playtone(chan, tone, 3 * ditlen);
+				} else if (*dahdit == '.') {
+					res = playtone(chan, tone, 1 * ditlen);
+				} else if (*dahdit == 'L' || *dahdit == 'l') {
+					res = playtone(chan, tone, 6 * ditlen); /* long dash */
+				} else if (*dahdit == '0') {
+					res = playtone(chan, tone, 9 * ditlen); /* extra long dash */
+				} else if (*dahdit == ' ') { /* space char (x20) = 6 dot lengths */
+					/* Intra-char pauses, specific to American Morse */
+					res = playtone(chan, toneoff, 3 * ditlen);
+				} else {
+					/* Account for ditlen of silence immediately following */
+					res = playtone(chan, toneoff, 2 * ditlen);
+				}
 
-			/* Pause slightly between each dit and dah */
-			playtone(chan, 0, 1 * ditlen);
+				/* Pause slightly between each dit and dah */
+				res = playtone(chan, toneoff, 1 * ditlen);
+				if (res)
+					break;
+			}
+			/* Pause between characters */
+			res = playtone(chan, toneoff, 3 * ditlen);
+			if (res)
+				break;
+		}
+	} else { /* International */
+		for (digit = data; *digit; digit++) {
+			const char *dahdit;
+			digit2 = *digit;
+			if (digit2 < 0 || digit2 > 127) {
+				continue;
+			}
+			for (dahdit = internationalcode[digit2]; *dahdit; dahdit++) {
+				if (*dahdit == '-') {
+					res = playtone(chan, tone, 3 * ditlen);
+				} else if (*dahdit == '.') {
+					res = playtone(chan, tone, 1 * ditlen);
+				} else {
+					/* Account for ditlen of silence immediately following */
+					res = playtone(chan, toneoff, 2 * ditlen);
+				}
+
+				/* Pause slightly between each dit and dah */
+				res = playtone(chan, toneoff, 1 * ditlen);
+				if (res)
+					break;
+			}
+			/* Pause between characters */
+			res = playtone(chan, toneoff, 2 * ditlen);
+			if (res)
+				break;
 		}
-		/* Pause between characters */
-		playtone(chan, 0, 2 * ditlen);
 	}
 
 	return res;

apps/app_mp3.c

@@ -64,10 +64,13 @@
 		</syntax>
 		<description>
 			<para>Executes mpg123 to play the given location, which typically would be a mp3 filename
-			or m3u playlist filename or a URL. Please read http://en.wikipedia.org/wiki/M3U
-			to see how M3U playlist file format is like, Example usage would be
+			or m3u playlist filename or a URL. Please read https://en.wikipedia.org/wiki/M3U
+			to see what the M3U playlist file format is like.</para>
+			<para>Note that mpg123 does not support HTTPS, so use HTTP for web streams.</para>
+			<para>User can exit by pressing any key on the dialpad, or by hanging up.</para>
+			<example title="Play an MP3 playlist">
 			exten => 1234,1,MP3Player(/var/lib/asterisk/playlist.m3u)
-			User can exit by pressing any key on the dialpad, or by hanging up.</para>
+			</example>
 			<para>This application does not automatically answer and should be preceeded by an
 			application such as Answer() or Progress().</para>
 		</description>
@@ -98,39 +101,39 @@ static int mp3play(const char *filename, unsigned int sampling_rate, int fd)
 	/* Execute mpg123, but buffer if it's a net connection */
 	if (!strncasecmp(filename, "http://", 7) && strstr(filename, ".m3u")) {
 	    char buffer_size_str[8];
-	    snprintf(buffer_size_str, 8, "%u", (int) 0.5*2*sampling_rate/1000); // 0.5 seconds for a live stream
+	    snprintf(buffer_size_str, 8, "%u", (int) 0.5*2*sampling_rate/1000); /* 0.5 seconds for a live stream */
 		/* Most commonly installed in /usr/local/bin */
-	    execl(LOCAL_MPG_123, "mpg123", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
+	    execl(LOCAL_MPG_123, "mpg123", "-e", "s16", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
 		/* But many places has it in /usr/bin */
-	    execl(MPG_123, "mpg123", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
+	    execl(MPG_123, "mpg123", "-e", "s16", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
 		/* As a last-ditch effort, try to use PATH */
-	    execlp("mpg123", "mpg123", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
+	    execlp("mpg123", "mpg123", "-e", "s16", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
 	}
 	else if (!strncasecmp(filename, "http://", 7)) {
 	    char buffer_size_str[8];
-	    snprintf(buffer_size_str, 8, "%u", 6*2*sampling_rate/1000); // 6 seconds for a remote MP3 file
+	    snprintf(buffer_size_str, 8, "%u", 6*2*sampling_rate/1000); /* 6 seconds for a remote MP3 file */
 		/* Most commonly installed in /usr/local/bin */
-	    execl(LOCAL_MPG_123, "mpg123", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
+	    execl(LOCAL_MPG_123, "mpg123", "-e", "s16", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
 		/* But many places has it in /usr/bin */
-	    execl(MPG_123, "mpg123", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
+	    execl(MPG_123, "mpg123", "-e", "s16", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
 		/* As a last-ditch effort, try to use PATH */
-	    execlp("mpg123", "mpg123", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
+	    execlp("mpg123", "mpg123", "-e", "s16", "-q", "-s", "-b", buffer_size_str, "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
 	}
 	else if (strstr(filename, ".m3u")) {
 		/* Most commonly installed in /usr/local/bin */
-	    execl(LOCAL_MPG_123, "mpg123", "-q", "-z", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
+	    execl(LOCAL_MPG_123, "mpg123", "-e", "s16", "-q", "-z", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
 		/* But many places has it in /usr/bin */
-	    execl(MPG_123, "mpg123", "-q", "-z", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
+	    execl(MPG_123, "mpg123", "-e", "s16", "-q", "-z", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
 		/* As a last-ditch effort, try to use PATH */
-	    execlp("mpg123", "mpg123", "-q", "-z", "-s",  "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
+	    execlp("mpg123", "mpg123", "-e", "s16", "-q", "-z", "-s",  "-f", "8192", "--mono", "-r", sampling_rate_str, "-@", filename, (char *)NULL);
 	}
 	else {
 		/* Most commonly installed in /usr/local/bin */
-	    execl(MPG_123, "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
+	    execl(MPG_123, "mpg123", "-e", "s16", "-q", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
 		/* But many places has it in /usr/bin */
-	    execl(LOCAL_MPG_123, "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
+	    execl(LOCAL_MPG_123, "mpg123", "-e", "s16", "-q", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
 		/* As a last-ditch effort, try to use PATH */
-	    execlp("mpg123", "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
+	    execlp("mpg123", "mpg123", "-e", "s16", "-q", "-s", "-f", "8192", "--mono", "-r", sampling_rate_str, filename, (char *)NULL);
 	}
 	/* Can't use ast_log since FD's are closed */
 	fprintf(stderr, "Execute of mpg123 failed\n");
@@ -177,6 +180,7 @@ static int mp3_exec(struct ast_channel *chan, const char *data)
 	int pid = -1;
 	RAII_VAR(struct ast_format *, owriteformat, NULL, ao2_cleanup);
 	int timeout = 2;
+	int startedmp3 = 0;
 	struct timeval next;
 	struct ast_frame *f;
 	struct myframe {
@@ -240,12 +244,19 @@ static int mp3_exec(struct ast_channel *chan, const char *data)
 				if (res > 0) {
 					myf.f.datalen = res;
 					myf.f.samples = res / 2;
+					startedmp3 = 1;
 					if (ast_write(chan, &myf.f) < 0) {
 						res = -1;
 						break;
 					}
 				} else {
 					ast_debug(1, "No more mp3\n");
+					if (!startedmp3) { /* we couldn't do anything, which means this stream doesn't work */
+						if (!strncasecmp(data, "https://", 8)) {
+							ast_log(LOG_WARNING, "%s() does not support HTTPS streams. Use HTTP instead.\n", app);
+						}
+						ast_log(LOG_WARNING, "MP3 stream '%s' is broken or nonexistent\n", data);
+					}
 					res = 0;
 					break;
 				}

apps/app_nbscat.c

@@ -27,6 +27,8 @@
 
 /*** MODULEINFO
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_originate.c

@@ -27,9 +27,6 @@
  *
  * \ingroup applications
  *
- * \todo Make a way to be able to set variables (and functions) on the outbound
- *       channel, similar to the Variable headers for the AMI Originate, and the
- *       Set options for call files.
  */
 
 /*** MODULEINFO
@@ -98,12 +95,30 @@ static const char app_originate[] = "Originate";
 						<argument name="argN" />
 					</argument>
 				</option>
+				<option name="C">
+					<para>Comma-separated list of codecs to use for this call.
+					Default is <literal>slin</literal>.</para>
+				</option>
+				<option name="c">
+					<para>The caller ID number to use for the called channel. Default is
+					the current channel's Caller ID number.</para>
+				</option>
+				<option name="n">
+					<para>The caller ID name to use for the called channel. Default is
+					the current channel's Caller ID name.</para>
+				</option>
+				<option name="v" argsep="^">
+					<para>A series of channel variables to set on the destination channel.</para>
+					<argument name="var1" multiple="true" argsep="=">
+						<argument name="name" required="true" />
+						<argument name="value" required="true" />
+					</argument>
+				</option>
 				</optionlist>
 			</parameter>
 		</syntax>
 		<description>
-		<para>This application originates an outbound call and connects it to a specified extension or application.  This application will block until the outgoing call fails or gets answered.  At that point, this application will exit with the status variable set and dialplan processing will continue.</para>
-
+		<para>This application originates an outbound call and connects it to a specified extension or application.  This application will block until the outgoing call fails or gets answered, unless the async option is used.  At that point, this application will exit with the status variable set and dialplan processing will continue.</para>
 		<para>This application sets the following channel variable before exiting:</para>
 		<variablelist>
 			<variable name="ORIGINATE_STATUS">
@@ -128,11 +143,19 @@ enum {
 	OPT_PREDIAL_CALLEE =    (1 << 0),
 	OPT_PREDIAL_CALLER =    (1 << 1),
 	OPT_ASYNC =             (1 << 2),
+	OPT_CALLER_NUM =        (1 << 3),
+	OPT_CALLER_NAME =       (1 << 4),
+	OPT_CODECS =            (1 << 5),
+	OPT_VARIABLES =         (1 << 6),
 };
 
 enum {
 	OPT_ARG_PREDIAL_CALLEE,
 	OPT_ARG_PREDIAL_CALLER,
+	OPT_ARG_CALLER_NUM,
+	OPT_ARG_CALLER_NAME,
+	OPT_ARG_CODECS,
+	OPT_ARG_VARIABLES,
 	/* note: this entry _MUST_ be the last one in the enum */
 	OPT_ARG_ARRAY_SIZE,
 };
@@ -141,9 +164,12 @@ AST_APP_OPTIONS(originate_exec_options, BEGIN_OPTIONS
 	AST_APP_OPTION('a', OPT_ASYNC),
 	AST_APP_OPTION_ARG('b', OPT_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
 	AST_APP_OPTION_ARG('B', OPT_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
+	AST_APP_OPTION_ARG('C', OPT_CODECS, OPT_ARG_CODECS),
+	AST_APP_OPTION_ARG('c', OPT_CALLER_NUM, OPT_ARG_CALLER_NUM),
+	AST_APP_OPTION_ARG('n', OPT_CALLER_NAME, OPT_ARG_CALLER_NAME),
+	AST_APP_OPTION_ARG('v', OPT_VARIABLES, OPT_ARG_VARIABLES),
 END_OPTIONS );
 
-
 static int originate_exec(struct ast_channel *chan, const char *data)
 {
 	AST_DECLARE_APP_ARGS(args,
@@ -158,29 +184,24 @@ static int originate_exec(struct ast_channel *chan, const char *data)
 	struct ast_flags64 opts = { 0, };
 	char *opt_args[OPT_ARG_ARRAY_SIZE];
 	char *predial_callee = NULL;
-	char *parse;
+	char *parse, *cnum = NULL, *cname = NULL;
+
+	struct ast_variable *vars = NULL;
 	char *chantech, *chandata;
 	int res = -1;
 	int continue_in_dialplan = 0;
 	int outgoing_status = 0;
 	unsigned int timeout = 30;
 	static const char default_exten[] = "s";
-	struct ast_format_cap *cap_slin = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+	struct ast_format_cap *capabilities;
+	capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
 
 	ast_autoservice_start(chan);
-	if (!cap_slin) {
+	if (!capabilities) {
 		goto return_cleanup;
 	}
 
-	ast_format_cap_append(cap_slin, ast_format_slin, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin12, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin16, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin24, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin32, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin44, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin48, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin96, 0);
-	ast_format_cap_append(cap_slin, ast_format_slin192, 0);
+	ast_format_cap_append(capabilities, ast_format_slin, 0);
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_ERROR, "Originate() requires arguments\n");
@@ -236,7 +257,57 @@ static int originate_exec(struct ast_channel *chan, const char *data)
 		goto return_cleanup;
 	}
 
+	if (ast_test_flag64(&opts, OPT_CODECS)) {
+		if (!ast_strlen_zero(opt_args[OPT_ARG_CODECS])) {
+			ast_format_cap_remove_by_type(capabilities, AST_MEDIA_TYPE_UNKNOWN);
+			ast_format_cap_update_by_allow_disallow(capabilities, opt_args[OPT_ARG_CODECS], 1);
+		}
+	}
+
+	if (ast_test_flag64(&opts, OPT_CALLER_NUM)) {
+		if (!ast_strlen_zero(opt_args[OPT_ARG_CALLER_NUM])) {
+			cnum = opt_args[OPT_ARG_CALLER_NUM];
+		} else if (ast_channel_caller(chan)->id.number.str) {
+			cnum = ast_channel_caller(chan)->id.number.str;
+		}
+	}
+
+	if (ast_test_flag64(&opts, OPT_CALLER_NAME)) {
+		if (!ast_strlen_zero(opt_args[OPT_ARG_CALLER_NAME])) {
+			cname = opt_args[OPT_ARG_CALLER_NAME];
+		} else if (ast_channel_caller(chan)->id.name.str) {
+			cname = ast_channel_caller(chan)->id.name.str;
+		}
+	}
+
+	/* Assign variables */
+	if (ast_test_flag64(&opts, OPT_VARIABLES)
+		&& !ast_strlen_zero(opt_args[OPT_ARG_VARIABLES])) {
+		char *vartext;
+		char *text = opt_args[OPT_ARG_VARIABLES];
+		while ((vartext = ast_strsep(&text, '^', 0))) {
+			struct ast_variable *var;
+			char *varname, *varvalue;
+			if (!(varname = ast_strsep(&vartext, '=', 0))) {
+				ast_log(LOG_ERROR, "Variable syntax error: %s\n", vartext);
+				goto return_cleanup;
+			}
+			if (!(varvalue = ast_strsep(&vartext, '=', 0))) {
+				varvalue = ""; /* empty values are allowed */
+			}
+			var = ast_variable_new(varname, varvalue, "");
+			if (!var) {
+				ast_log(LOG_ERROR, "Failed to allocate variable: %s\n", varname);
+				goto return_cleanup;
+			}
+			ast_debug(1, "Appending variable '%s' with value '%s'", varname, varvalue);
+			ast_variable_list_append(&vars, var);
+		}
+	}
+
 	if (!strcasecmp(args.type, "exten")) {
+		const char *cid_num = cnum;
+		const char *cid_name = cname;
 		int priority = 1; /* Initialized in case priority not specified */
 		const char *exten = args.arg2;
 
@@ -254,19 +325,21 @@ static int originate_exec(struct ast_channel *chan, const char *data)
 		ast_debug(1, "Originating call to '%s/%s' and connecting them to extension %s,%s,%d\n",
 				chantech, chandata, args.arg1, exten, priority);
 
-		res = ast_pbx_outgoing_exten_predial(chantech, cap_slin, chandata,
+		res = ast_pbx_outgoing_exten_predial(chantech, capabilities, chandata,
 				timeout * 1000, args.arg1, exten, priority, &outgoing_status,
 				ast_test_flag64(&opts, OPT_ASYNC) ? AST_OUTGOING_NO_WAIT : AST_OUTGOING_WAIT,
-				NULL, NULL, NULL, NULL, NULL, 0, NULL,
+				cid_num, cid_name, vars, NULL, NULL, 0, NULL,
 				predial_callee);
 	} else {
+		const char *cid_num = cnum;
+		const char *cid_name = cname;
 		ast_debug(1, "Originating call to '%s/%s' and connecting them to %s(%s)\n",
 				chantech, chandata, args.arg1, S_OR(args.arg2, ""));
 
-		res = ast_pbx_outgoing_app_predial(chantech, cap_slin, chandata,
+		res = ast_pbx_outgoing_app_predial(chantech, capabilities, chandata,
 				timeout * 1000, args.arg1, args.arg2, &outgoing_status,
 				ast_test_flag64(&opts, OPT_ASYNC) ? AST_OUTGOING_NO_WAIT : AST_OUTGOING_WAIT,
-				NULL, NULL, NULL, NULL, NULL, NULL,
+				cid_num, cid_name, vars, NULL, NULL, NULL,
 				predial_callee);
 	}
 
@@ -311,7 +384,10 @@ return_cleanup:
 			break;
 		}
 	}
-	ao2_cleanup(cap_slin);
+	if (vars) {
+		ast_variables_destroy(vars);
+	}
+	ao2_cleanup(capabilities);
 	ast_autoservice_stop(chan);
 
 	return continue_in_dialplan ? 0 : -1;

apps/app_osplookup.c

@@ -32,6 +32,8 @@
 	<depend>osptk</depend>
 	<depend>openssl</depend>
 	<support_level>extended</support_level>
+	<deprecated_in>19</deprecated_in>
+	<removed_in>21</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_page.c

@@ -160,6 +160,8 @@ AST_APP_OPTIONS(page_opts, {
 	AST_APP_OPTION('n', PAGE_NOCALLERANNOUNCE),
 });
 
+#define PAGE_BEEP "beep"
+
 /* We use this structure as a way to pass this to all dialed channels */
 struct page_options {
 	char *opts[OPT_ARG_ARRAY_SIZE];
@@ -172,8 +174,6 @@ struct page_options {
  *
  * \param chan Setup bridge profile on this channel.
  * \param options Options to setup bridge profile.
- *
- * \return Nothing
  */
 static void setup_profile_bridge(struct ast_channel *chan, struct page_options *options)
 {
@@ -190,8 +190,6 @@ static void setup_profile_bridge(struct ast_channel *chan, struct page_options *
  *
  * \param chan Setup user profile on this channel.
  * \param options Options to setup paged user profile.
- *
- * \return Nothing
  */
 static void setup_profile_paged(struct ast_channel *chan, struct page_options *options)
 {
@@ -214,8 +212,6 @@ static void setup_profile_paged(struct ast_channel *chan, struct page_options *o
  *
  * \param chan Setup user profile on this channel.
  * \param options Options to setup caller user profile.
- *
- * \return Nothing
  */
 static void setup_profile_caller(struct ast_channel *chan, struct page_options *options)
 {
@@ -402,9 +398,14 @@ static int page_exec(struct ast_channel *chan, const char *data)
 	ast_free(predial_callee);
 
 	if (!ast_test_flag(&options.flags, PAGE_QUIET)) {
-		res = ast_streamfile(chan, "beep", ast_channel_language(chan));
-		if (!res)
-			res = ast_waitstream(chan, "");
+		if (!ast_fileexists(PAGE_BEEP, NULL, NULL)) {
+			ast_log(LOG_WARNING, "Missing required sound file: '" PAGE_BEEP "'\n");
+		} else {
+			res = ast_streamfile(chan, PAGE_BEEP, ast_channel_language(chan));
+			if (!res) {
+				res = ast_waitstream(chan, "");
+			}
+		}
 	}
 
 	if (!res) {

apps/app_playback.c

@@ -35,7 +35,7 @@
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/app.h"
-/* This file provides config-file based 'say' functions, and implenents
+/* This file provides config-file based 'say' functions, and implements
  * some CLI commands.
  */
 #include "asterisk/say.h"	/*!< provides config-file based 'say' functions */
@@ -62,13 +62,19 @@
 						be answered before the sound is played.</para>
 						<note><para>Not all channel types support playing messages while still on hook.</para></note>
 					</option>
+					<option name="say">
+						<para>Play using the say.conf file.</para>
+					</option>
+					<option name="mix">
+						<para>Play using a mix of filename and the say.conf file.</para>
+					</option>
 				</optionlist>
 			</parameter>
 		</syntax>
 		<description>
 			<para>Plays back given filenames (do not put extension of wav/alaw etc).
-			The playback command answer the channel if no options are specified.
-			If the file is non-existant it will fail</para>
+			The Playback application answers the channel if no options are specified.
+			If the file is non-existent it will fail.</para>
 			<para>This application sets the following channel variable upon completion:</para>
 			<variablelist>
 				<variable name="PLAYBACKSTATUS">
@@ -446,6 +452,7 @@ static int playback_exec(struct ast_channel *chan, const char *data)
 	char *tmp;
 	int option_skip=0;
 	int option_say=0;
+	int option_mix=0;
 	int option_noanswer = 0;
 
 	AST_DECLARE_APP_ARGS(args,
@@ -466,6 +473,8 @@ static int playback_exec(struct ast_channel *chan, const char *data)
 			option_skip = 1;
 		if (strcasestr(args.options, "say"))
 			option_say = 1;
+		if (strcasestr(args.options, "mix"))
+			option_mix = 1;
 		if (strcasestr(args.options, "noanswer"))
 			option_noanswer = 1;
 	}
@@ -486,13 +495,19 @@ static int playback_exec(struct ast_channel *chan, const char *data)
 		while (!res && (front = strsep(&back, "&"))) {
 			if (option_say)
 				res = say_full(chan, front, "", ast_channel_language(chan), NULL, -1, -1);
+			else if (option_mix){
+				/* Check if it is in say format but not remote audio file */
+				if (strcasestr(front, ":") && !strcasestr(front, "://"))
+					res = say_full(chan, front, "", ast_channel_language(chan), NULL, -1, -1);
+				else
+					res = ast_streamfile(chan, front, ast_channel_language(chan));
+			}
 			else
 				res = ast_streamfile(chan, front, ast_channel_language(chan));
 			if (!res) {
 				res = ast_waitstream(chan, "");
 				ast_stopstream(chan);
-			}
-			if (res) {
+			} else {
 				if (!ast_check_hangup(chan)) {
 					ast_log(LOG_WARNING, "Playback failed on %s for %s\n", ast_channel_name(chan), (char *)data);
 				}

apps/app_read.c

@@ -75,6 +75,23 @@
 					<option name="n">
 						<para>to read digits even if the line is not up.</para>
 					</option>
+					<option name="t">
+						<para>Terminator digit(s) to use for ending input.
+						Default is <literal>#</literal>. If you need to read
+						the digit <literal>#</literal> literally, you should
+						remove or change the terminator character. Multiple
+						terminator characters may be specified. If no terminator
+						digit is present, input cannot be ended using digits
+						and you will need to rely on duration and max digits
+						for ending input.</para>
+					</option>
+					<option name="e">
+						<para>to read the terminator as the digit string if the
+						only digit read is the terminator. This is for cases
+						where the terminator is a valid digit, but only by itself.
+						ie; <literal>1234</literal> and <literal>#</literal> are
+						valid, but <literal>1234#</literal> is not.</para>
+					</option>
 				</optionlist>
 			</parameter>
 			<parameter name="attempts">
@@ -114,12 +131,22 @@ enum read_option_flags {
 	OPT_SKIP = (1 << 0),
 	OPT_INDICATION = (1 << 1),
 	OPT_NOANSWER = (1 << 2),
+	OPT_TERMINATOR = (1 << 3),
+	OPT_KEEP_TERMINATOR = (1 << 4),
+};
+
+enum {
+	OPT_ARG_TERMINATOR,
+	/* note: this entry _MUST_ be the last one in the enum */
+	OPT_ARG_ARRAY_SIZE,
 };
 
 AST_APP_OPTIONS(read_app_options, {
 	AST_APP_OPTION('s', OPT_SKIP),
 	AST_APP_OPTION('i', OPT_INDICATION),
 	AST_APP_OPTION('n', OPT_NOANSWER),
+	AST_APP_OPTION_ARG('t', OPT_TERMINATOR, OPT_ARG_TERMINATOR),
+	AST_APP_OPTION('e', OPT_KEEP_TERMINATOR),
 });
 
 static char *app = "Read";
@@ -132,9 +159,11 @@ static int read_exec(struct ast_channel *chan, const char *data)
 	int tries = 1, to = 0, x = 0;
 	double tosec;
 	char *argcopy = NULL;
+	char *opt_args[OPT_ARG_ARRAY_SIZE];
 	struct ast_tone_zone_sound *ts = NULL;
 	struct ast_flags flags = {0};
 	const char *status = "ERROR";
+	char *terminator = "#"; /* use default terminator # by default */
 
 	AST_DECLARE_APP_ARGS(arglist,
 		AST_APP_ARG(variable);
@@ -156,7 +185,7 @@ static int read_exec(struct ast_channel *chan, const char *data)
 	AST_STANDARD_APP_ARGS(arglist, argcopy);
 
 	if (!ast_strlen_zero(arglist.options)) {
-		ast_app_parse_options(read_app_options, &flags, NULL, arglist.options);
+		ast_app_parse_options(read_app_options, &flags, opt_args, arglist.options);
 	}
 
 	if (!ast_strlen_zero(arglist.attempts)) {
@@ -192,9 +221,19 @@ static int read_exec(struct ast_channel *chan, const char *data)
 			ts = ast_get_indication_tone(ast_channel_zone(chan), arglist.filename);
 		}
 	}
+	if (ast_test_flag(&flags, OPT_TERMINATOR)) {
+		if (!ast_strlen_zero(opt_args[OPT_ARG_TERMINATOR])) {
+			terminator = opt_args[OPT_ARG_TERMINATOR];
+		} else {
+			terminator = ""; /* no digit inherently will terminate input */
+		}
+	}
 	if (ast_channel_state(chan) != AST_STATE_UP) {
 		if (ast_test_flag(&flags, OPT_SKIP)) {
 			/* At the user's option, skip if the line is not up */
+			if (ts) {
+				ts = ast_tone_zone_sound_unref(ts);
+			}
 			pbx_builtin_setvar_helper(chan, arglist.variable, "");
 			pbx_builtin_setvar_helper(chan, "READSTATUS", "SKIPPED");
 			return 0;
@@ -220,7 +259,7 @@ static int read_exec(struct ast_channel *chan, const char *data)
 						break;
 					}
 					tmp[x++] = res;
-					if (tmp[x-1] == '#') {
+					if (terminator && strchr(terminator, tmp[x-1])) {
 						tmp[x-1] = '\0';
 						status = "OK";
 						break;
@@ -230,13 +269,21 @@ static int read_exec(struct ast_channel *chan, const char *data)
 					}
 				}
 			} else {
-				res = ast_app_getdata(chan, arglist.filename, tmp, maxdigits, to);
-				if (res == AST_GETDATA_COMPLETE || res == AST_GETDATA_EMPTY_END_TERMINATED)
+				res = ast_app_getdata_terminator(chan, arglist.filename, tmp, maxdigits, to, terminator);
+				if (res == AST_GETDATA_COMPLETE) {
 					status = "OK";
-				else if (res == AST_GETDATA_TIMEOUT)
+				} else if (res == AST_GETDATA_EMPTY_END_TERMINATED) {
+					if (ast_test_flag(&flags, OPT_KEEP_TERMINATOR)) {
+						/* if the option is set to do so, read the
+						   returned string as the terminator string */
+						ast_copy_string(tmp, terminator, sizeof(tmp));
+					}
+					status = "OK";
+				} else if (res == AST_GETDATA_TIMEOUT) {
 					status = "TIMEOUT";
-				else if (res == AST_GETDATA_INTERRUPTED)
+				} else if (res == AST_GETDATA_INTERRUPTED) {
 					status = "INTERRUPTED";
+				}
 			}
 			if (res > -1) {
 				pbx_builtin_setvar_helper(chan, arglist.variable, tmp);

apps/app_reload.c

@@ -0,0 +1,115 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * 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 Reload Asterisk modules
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+
+/*** DOCUMENTATION
+	<application name="Reload" language="en_US">
+		<since>
+			<version>16.20.0</version>
+			<version>18.6.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Reloads an Asterisk module, blocking the channel until the reload has completed.
+		</synopsis>
+		<syntax>
+			<parameter name="module" required="false">
+				<para>The full name(s) of the target module(s) or resource(s) to reload.
+				If omitted, everything will be reloaded.</para>
+				<para>The full names MUST be specified (e.g. <literal>chan_iax2</literal>
+				to reload IAX2 or <literal>pbx_config</literal> to reload the dialplan.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Reloads the specified (or all) Asterisk modules and reports success or failure.
+			Success is determined by each individual module, and if all reloads are successful,
+			that is considered an aggregate success. If multiple modules are specified and any
+			module fails, then FAILURE will be returned. It is still possible that other modules
+			did successfully reload, however.</para>
+			<para>Sets <variable>RELOADSTATUS</variable> to one of the following values:</para>
+			<variablelist>
+				<variable name="RELOADSTATUS">
+					<value name="SUCCESS">
+						Specified module(s) reloaded successfully.
+					</value>
+					<value name="FAILURE">
+						Some or all of the specified modules failed to reload.
+					</value>
+				</variable>
+			</variablelist>
+		</description>
+	</application>
+ ***/
+
+static char *app = "Reload";
+
+static int reload_exec(struct ast_channel *chan, const char *data)
+{
+	char *targets, *target = NULL;
+	enum ast_module_reload_result res = AST_MODULE_RELOAD_SUCCESS;
+
+	targets = ast_strdupa(data);
+	ast_autoservice_start(chan);
+	if (ast_strlen_zero(targets)) { /* Reload everything */
+		res = ast_module_reload(targets);
+	} else {
+		while((target = ast_strsep(&targets, ',', AST_STRSEP_ALL))) {
+			res |= ast_module_reload(target);
+		}
+	}
+	ast_autoservice_stop(chan);
+
+	if (res == AST_MODULE_RELOAD_SUCCESS) {
+		pbx_builtin_setvar_helper(chan, "RELOADSTATUS", "SUCCESS");
+	} else {
+		pbx_builtin_setvar_helper(chan, "RELOADSTATUS", "FAILURE");
+	}
+	return 0;
+}
+
+static int unload_module(void)
+{
+	return ast_unregister_application(app);
+}
+
+static int load_module(void)
+{
+	return ast_register_application_xml(app, reload_exec);
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Reload module(s)");

apps/app_senddtmf.c

@@ -57,6 +57,15 @@
 			<parameter name="channel" required="false">
 				<para>Channel where digits will be played</para>
 			</parameter>
+			<parameter name="options">
+				<optionlist>
+					<option name="a">
+						<para>Answer the channel specified by the <literal>channel</literal>
+						parameter if it is not already up. If no <literal>channel</literal>
+						parameter is provided, the current channel will be answered.</para>
+					</option>
+				</optionlist>
+			</parameter>
 		</syntax>
 		<description>
 			<para>It will send all digits or terminate if it encounters an error.</para>
@@ -90,6 +99,19 @@
 	</manager>
  ***/
 
+enum read_option_flags {
+	OPT_ANSWER = (1 << 0),
+};
+
+AST_APP_OPTIONS(senddtmf_app_options, {
+	AST_APP_OPTION('a', OPT_ANSWER),
+});
+
+enum {
+	/* note: this entry _MUST_ be the last one in the enum */
+	OPT_ARG_ARRAY_SIZE,
+};
+
 static const char senddtmf_name[] = "SendDTMF";
 
 static int senddtmf_exec(struct ast_channel *chan, const char *vdata)
@@ -100,11 +122,14 @@ static int senddtmf_exec(struct ast_channel *chan, const char *vdata)
 	struct ast_channel *chan_found = NULL;
 	struct ast_channel *chan_dest = chan;
 	struct ast_channel *chan_autoservice = NULL;
+	char *opt_args[OPT_ARG_ARRAY_SIZE];
+	struct ast_flags flags = {0};
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(digits);
 		AST_APP_ARG(dinterval);
 		AST_APP_ARG(duration);
 		AST_APP_ARG(channel);
+		AST_APP_ARG(options);
 	);
 
 	if (ast_strlen_zero(vdata)) {
@@ -136,6 +161,12 @@ static int senddtmf_exec(struct ast_channel *chan, const char *vdata)
 			chan_autoservice = chan;
 		}
 	}
+	if (!ast_strlen_zero(args.options)) {
+		ast_app_parse_options(senddtmf_app_options, &flags, opt_args, args.options);
+	}
+	if (ast_test_flag(&flags, OPT_ANSWER)) {
+		ast_auto_answer(chan_dest);
+	}
 	res = ast_dtmf_stream(chan_dest, chan_autoservice, args.digits,
 		dinterval <= 0 ? 250 : dinterval, duration);
 	if (chan_found) {

apps/app_sendtext.c

@@ -21,6 +21,7 @@
  * \brief App to transmit a text message
  *
  * \author Mark Spencer <markster@digium.com>
+ * \author Naveen Albert <asterisk@phreaknet.org>
  *
  * \note Requires support of sending text messages from channel driver
  *
@@ -140,11 +141,55 @@
 		<see-also>
 			<ref type="application">SendImage</ref>
 			<ref type="application">SendURL</ref>
+			<ref type="application">ReceiveText</ref>
+		</see-also>
+	</application>
+	<application name="ReceiveText" language="en_US">
+		<since>
+			<version>16.24.0</version>
+			<version>18.10.0</version>
+			<version>19.2.0</version>
+		</since>
+		<synopsis>
+			Receive a Text Message on a channel.
+		</synopsis>
+		<syntax>
+			<parameter name="timeout" required="false">
+				<para>Time in seconds to wait for text. Default is 0 (forever).</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Waits for <replaceable>timeout</replaceable> seconds on the current channel
+			to receive text.</para>
+			<para>Result of transmission will be stored in the following variables:</para>
+			<variablelist>
+				<variable name="RECEIVETEXTMESSAGE">
+					<para>The received text message.</para>
+				</variable>
+				<variable name="RECEIVETEXTSTATUS">
+					<value name="SUCCESS">
+						Transmission succeeded.
+					</value>
+					<value name="FAILURE">
+						Transmission failed or timed out.
+					</value>
+				</variable>
+			</variablelist>
+			<example title="Receive message on channel">
+			 same => n,ReceiveText()
+			 same => n,NoOp(${RECEIVETEXTMESSAGE})
+			</example>
+		</description>
+		<see-also>
+			<ref type="application">SendText</ref>
+			<ref type="application">SendImage</ref>
+			<ref type="application">SendURL</ref>
 		</see-also>
 	</application>
  ***/
 
 static const char * const app = "SendText";
+static const char * const app2 = "ReceiveText";
 
 static int sendtext_exec(struct ast_channel *chan, const char *data)
 {
@@ -237,14 +282,55 @@ cleanup:
 	return rc;
 }
 
+static int recvtext_exec(struct ast_channel *chan, const char *data)
+{
+	double timeout = 0, timeout_ms = 0;
+	char *parse, *buf;
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(timeout);
+	);
+
+	parse = ast_strdupa(data);
+
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (!ast_strlen_zero(args.timeout)) {
+		if (sscanf(args.timeout, "%30lg", &timeout) != 1) {
+			ast_log(LOG_WARNING, "Invalid timeout provided: %s. No timeout set.\n", args.timeout);
+			return -1;
+		}
+		timeout_ms = timeout * 1000.0;
+	}
+
+	buf = ast_recvtext(chan, timeout_ms);
+	pbx_builtin_setvar_helper(chan, "RECEIVETEXTSTATUS", buf ? "SUCCESS" : "FAILURE");
+	if (buf) {
+		pbx_builtin_setvar_helper(chan, "RECEIVETEXTMESSAGE", buf);
+		ast_free(buf);
+	}
+
+	return 0;
+}
+
 static int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+	
+	res = ast_unregister_application(app);
+	res |= ast_unregister_application(app2);
+
+	return res;
 }
 
 static int load_module(void)
 {
-	return ast_register_application_xml(app, sendtext_exec);
+	int res;
+
+	res = ast_register_application_xml(app, sendtext_exec);
+	res |= ast_register_application_xml(app2, recvtext_exec);
+
+	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send Text Applications");
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send and Receive Text Applications");

apps/app_sf.c

@@ -0,0 +1,467 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * 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 SF sender and receiver applications
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/file.h"
+#include "asterisk/pbx.h"
+#include "asterisk/channel.h"
+#include "asterisk/dsp.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+#include "asterisk/indications.h"
+#include "asterisk/conversions.h"
+
+/*** DOCUMENTATION
+	<application name="ReceiveSF" language="en_US">
+		<since>
+			<version>16.24.0</version>
+			<version>18.10.0</version>
+			<version>19.2.0</version>
+		</since>
+		<synopsis>
+			Detects SF digits on a channel and saves them to a variable.
+		</synopsis>
+		<syntax>
+			<parameter name="variable" required="true">
+				<para>The input digits will be stored in the given
+				<replaceable>variable</replaceable> name.</para>
+			</parameter>
+			<parameter name="digits" required="false">
+				<para>Maximum number of digits to read. Default is unlimited.</para>
+			</parameter>
+			<parameter name="timeout">
+				<para>The number of seconds to wait for all digits, if greater
+				than <literal>0</literal>. Can be floating point. Default
+				is no timeout.</para>
+			</parameter>
+			<parameter name="frequency">
+				<para>The frequency for which to detect pulsed digits.
+				Default is 2600 Hz.</para>
+			</parameter>
+			<parameter name="options">
+				<optionlist>
+					<option name="d">
+						<para>Delay audio by a frame to try to extra quelch.</para>
+					</option>
+					<option name="e">
+						<para>Allow receiving extra pulses 11 through 16.</para>
+					</option>
+					<option name="m">
+						<para>Mute conference.</para>
+					</option>
+					<option name="q">
+						<para>Quelch SF from in-band.</para>
+					</option>
+					<option name="r">
+						<para>"Radio" mode (relaxed SF).</para>
+					</option>
+				</optionlist>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Reads SF digits from the user in to the given
+			<replaceable>variable</replaceable>.</para>
+			<para>This application does not automatically answer the channel and
+			should be preceded with <literal>Answer</literal> or
+			<literal>Progress</literal> as needed.</para>
+			<variablelist>
+				<variable name="RECEIVESFSTATUS">
+					<para>This is the status of the read operation.</para>
+					<value name="START" />
+					<value name="ERROR" />
+					<value name="HANGUP" />
+					<value name="MAXDIGITS" />
+					<value name="TIMEOUT" />
+				</variable>
+			</variablelist>
+		</description>
+		<see-also>
+			<ref type="application">ReceiveMF</ref>
+			<ref type="application">SendMF</ref>
+			<ref type="application">SendSF</ref>
+			<ref type="application">Read</ref>
+		</see-also>
+	</application>
+	<application name="SendSF" language="en_US">
+		<since>
+			<version>16.24.0</version>
+			<version>18.10.0</version>
+			<version>19.2.0</version>
+		</since>
+		<synopsis>
+			Sends arbitrary SF digits on the current or specified channel.
+		</synopsis>
+		<syntax>
+			<parameter name="digits" required="true">
+				<para>List of digits 0-9 to send; w for a half-second pause,
+				also f or F for a flash-hook if the channel supports flash-hook,
+				h or H for 250 ms of 2600 Hz, and W for a wink if the channel
+				supports wink.</para>
+			</parameter>
+			<parameter name="frequency" required="false">
+				<para>Frequency to use. (defaults to 2600 Hz).</para>
+			</parameter>
+			<parameter name="channel" required="false">
+				<para>Channel where digits will be played</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>It will send all digits or terminate if it encounters an error.</para>
+		</description>
+		<see-also>
+			<ref type="application">SendDTMF</ref>
+			<ref type="application">SendMF</ref>
+			<ref type="application">ReceiveMF</ref>
+			<ref type="application">ReceiveSF</ref>
+		</see-also>
+	</application>
+ ***/
+
+enum read_option_flags {
+	OPT_DELAY = (1 << 0),
+	OPT_MUTE = (1 << 1),
+	OPT_QUELCH = (1 << 2),
+	OPT_RELAXED = (1 << 3),
+	OPT_EXTRAPULSES = (1 << 4),
+};
+
+AST_APP_OPTIONS(read_app_options, {
+	AST_APP_OPTION('d', OPT_DELAY),
+	AST_APP_OPTION('e', OPT_EXTRAPULSES),
+	AST_APP_OPTION('m', OPT_MUTE),
+	AST_APP_OPTION('q', OPT_QUELCH),
+	AST_APP_OPTION('r', OPT_RELAXED),
+});
+
+static const char *readsf_name = "ReceiveSF";
+static const char sendsf_name[] = "SendSF";
+
+/*!
+ * \brief Detects SF digits on channel using DSP
+ *
+ * \param chan channel on which to read digits
+ * \param buf Buffer in which to store digits
+ * \param buflen Size of buffer
+ * \param timeout ms to wait for all digits before giving up
+ * \param maxdigits Maximum number of digits
+ * \param freq Frequency to use
+ * \param features DSP features
+ * \param extrapulses Whether to recognize extra pulses
+ *
+ * \retval 0 if successful
+ * \retval -1 if unsuccessful (including hangup).
+ */
+static int read_sf_digits(struct ast_channel *chan, char *buf, int buflen, int timeout, int maxdigits, int freq, int features, int extrapulses) {
+	/* Bell System Technical Journal 39 (Nov. 1960) */
+	#define SF_MIN_OFF 25
+	#define SF_ON 67
+	#define SF_BETWEEN 600
+	#define SF_MIN_DETECT 50
+
+	struct ast_dsp *dsp = NULL;
+	struct ast_frame *frame = NULL;
+	struct timeval start, pulsetimer, digittimer;
+	int remaining_time = timeout;
+	char *str = buf;
+	int hits = 0, digits_read = 0;
+	unsigned short int sf_on = 0;
+	int res = 0;
+
+	if (!(dsp = ast_dsp_new())) {
+		ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+		pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "ERROR");
+		return -1;
+	}
+	ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
+	/* tolerance is 46 to 76% make break at 8 to 12 pps */
+	ast_dsp_set_freqmode(dsp, freq, SF_MIN_DETECT, 16, 0);
+
+	start = ast_tvnow();
+	*str = 0; /* start with empty output buffer */
+
+	while (timeout == 0 || remaining_time > 0) {
+		if (timeout > 0) {
+			remaining_time = ast_remaining_ms(start, timeout);
+			if (remaining_time <= 0) {
+				pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "TIMEOUT");
+				break;
+			}
+		}
+		if (digits_read >= (buflen - 1)) { /* we don't have room to store any more digits (very unlikely to happen for a legitimate reason) */
+			/* This result will probably not be usable, so status should not be START */
+			pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "MAXDIGITS");
+			break;
+		}
+		if (ast_waitfor(chan, 1000) > 0) {
+			frame = ast_read(chan);
+			if (!frame) {
+				ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
+				pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "HANGUP");
+				break;
+			} else if (frame->frametype == AST_FRAME_VOICE) {
+				frame = ast_dsp_process(chan, dsp, frame);
+				if (frame->frametype == AST_FRAME_DTMF) {
+					char result = frame->subclass.integer;
+					if (result == 'q') {
+						sf_on = 1;
+						pulsetimer = ast_tvnow(); /* reset the pulse timer */
+						/* now, we need at least a 33ms pause to register the pulse */
+					}
+				} else {
+					if (sf_on) {
+						int timeleft = ast_remaining_ms(pulsetimer, SF_MIN_OFF);
+						if (timeleft <= 0) {
+							sf_on = 0;
+							/* The pulse needs to end no more than 30ms after we detected it */
+							if (timeleft > -30) {
+								hits++;
+								digittimer = ast_tvnow(); /* reset the digit timer */
+								ast_debug(5, "Detected SF pulse (pulse #%d)\n", hits);
+								ast_dsp_free(dsp);
+								if (!(dsp = ast_dsp_new())) {
+									ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+									pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "ERROR");
+									ast_frfree(frame);
+									return -1;
+								}
+								ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
+								ast_dsp_set_freqmode(dsp, freq, SF_MIN_DETECT, 16, 0);
+							} else {
+								ast_debug(5, "SF noise, ignoring, time elapsed was %d ms\n", timeleft);
+							}
+						}
+					} else if (hits > 0 && ast_remaining_ms(digittimer, SF_BETWEEN) <= 0) {
+						/* has the digit finished? */
+						ast_debug(2, "Received SF digit: %d\n", hits);
+						digits_read++;
+						if (hits > 10) {
+							if (extrapulses) {
+								/* dahdi-base.c translates 11 to * and 12 to # */
+								if (hits == 11) {
+									hits = '*';
+								} else if (hits == 12) {
+									hits = '#';
+								} else if (hits == 13) {
+									hits = 'D';
+								} else if (hits == 14) {
+									hits = 'C';
+								} else if (hits == 15) {
+									hits = 'B';
+								} else if (hits == 16) {
+									hits = 'A';
+								} else {
+									ast_debug(3, "Got %d SF pulses, is someone playing with the phone?\n", hits);
+									hits = 'A';
+								}
+								*str++ = hits;
+							} else {
+								ast_debug(2, "Got more than 10 pulses, truncating to 10\n");
+								hits = 0; /* 10 dial pulses = digit 0 */
+								*str++ = hits + '0';
+							}
+						} else {
+							if (hits == 10) {
+								hits = 0; /* 10 dial pulses = digit 0 */
+							}
+							*str++ = hits + '0';
+						}
+						*str = 0;
+						hits = 0;
+						if (maxdigits > 0 && digits_read >= maxdigits) {
+							pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "START");
+							ast_frfree(frame);
+							break;
+						}
+					}
+				}
+			}
+			ast_frfree(frame);
+		} else {
+			pbx_builtin_setvar_helper(chan, "RECEIVESFSTATUS", "HANGUP");
+			res = -1;
+		}
+	}
+	if (dsp) {
+		ast_dsp_free(dsp);
+	}
+	ast_debug(3, "channel '%s' - event loop stopped { timeout: %d, remaining_time: %d }\n", ast_channel_name(chan), timeout, remaining_time);
+	return res;
+}
+
+static int read_sf_exec(struct ast_channel *chan, const char *data)
+{
+#define BUFFER_SIZE 256
+	char tmp[BUFFER_SIZE] = "";
+	double tosec;
+	struct ast_flags flags = {0};
+	char *argcopy = NULL;
+	int res, features = 0, digits = 0, to = 0, freq = 2600;
+
+	AST_DECLARE_APP_ARGS(arglist,
+		AST_APP_ARG(variable);
+		AST_APP_ARG(digits);
+		AST_APP_ARG(timeout);
+		AST_APP_ARG(freq);
+		AST_APP_ARG(options);
+	);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "ReceiveSF requires an argument (variable)\n");
+		return -1;
+	}
+
+	argcopy = ast_strdupa(data);
+
+	AST_STANDARD_APP_ARGS(arglist, argcopy);
+
+	if (!ast_strlen_zero(arglist.options)) {
+		ast_app_parse_options(read_app_options, &flags, NULL, arglist.options);
+	}
+
+	if (!ast_strlen_zero(arglist.timeout)) {
+		tosec = atof(arglist.timeout);
+		if (tosec <= 0) {
+			to = 0;
+		} else {
+			to = tosec * 1000.0;
+		}
+	}
+
+	if (!ast_strlen_zero(arglist.digits) && (ast_str_to_int(arglist.digits, &digits) || digits <= 0)) {
+		ast_log(LOG_WARNING, "Invalid number of digits: %s\n", arglist.digits);
+		return -1;
+	}
+
+	if (!ast_strlen_zero(arglist.freq) && (ast_str_to_int(arglist.freq, &freq) || freq <= 0)) {
+		ast_log(LOG_WARNING, "Invalid freq: %s\n", arglist.freq);
+		return -1;
+	}
+
+	if (ast_strlen_zero(arglist.variable)) {
+		ast_log(LOG_WARNING, "Invalid! Usage: ReceiveSF(variable[,timeout][,option])\n");
+		return -1;
+	}
+
+	if (ast_test_flag(&flags, OPT_DELAY)) {
+		features |= DSP_DIGITMODE_MUTEMAX;
+	}
+
+	if (ast_test_flag(&flags, OPT_MUTE)) {
+		features |= DSP_DIGITMODE_MUTECONF;
+	}
+
+	if (!ast_test_flag(&flags, OPT_QUELCH)) {
+		features |= DSP_DIGITMODE_NOQUELCH;
+	}
+
+	if (ast_test_flag(&flags, OPT_RELAXED)) {
+		features |= DSP_DIGITMODE_RELAXDTMF;
+	}
+
+	res = read_sf_digits(chan, tmp, BUFFER_SIZE, to, digits, freq, features, ast_test_flag(&flags, OPT_EXTRAPULSES));
+	pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
+	if (!ast_strlen_zero(tmp)) {
+		ast_verb(3, "SF digits received: '%s'\n", tmp);
+	} else if (!res) { /* if channel hung up, don't print anything out */
+		ast_verb(3, "No SF digits received.\n");
+	}
+	return res;
+}
+
+static int sendsf_exec(struct ast_channel *chan, const char *vdata)
+{
+	int res;
+	char *data;
+	int frequency = 2600;
+	struct ast_channel *chan_found = NULL;
+	struct ast_channel *chan_dest = chan;
+	struct ast_channel *chan_autoservice = NULL;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(digits);
+		AST_APP_ARG(frequency);
+		AST_APP_ARG(channel);
+	);
+
+	if (ast_strlen_zero(vdata)) {
+		ast_log(LOG_WARNING, "SendSF requires an argument\n");
+		return 0;
+	}
+
+	data = ast_strdupa(vdata);
+	AST_STANDARD_APP_ARGS(args, data);
+
+	if (ast_strlen_zero(args.digits)) {
+		ast_log(LOG_WARNING, "The digits argument is required (0-9,wf)\n");
+		return 0;
+	}
+	if (!ast_strlen_zero(args.frequency) && (ast_str_to_int(args.frequency, &frequency) || frequency < 1)) {
+		ast_log(LOG_WARNING, "Invalid duration: %s\n", args.frequency);
+		return -1;
+	}
+	if (!ast_strlen_zero(args.channel)) {
+		chan_found = ast_channel_get_by_name(args.channel);
+		if (!chan_found) {
+			ast_log(LOG_WARNING, "No such channel: %s\n", args.channel);
+			return 0;
+		}
+		chan_dest = chan_found;
+		if (chan_found != chan) {
+			chan_autoservice = chan;
+		}
+	}
+	res = ast_sf_stream(chan_dest, chan_autoservice, NULL, args.digits, frequency, 0);
+	ast_channel_cleanup(chan_found);
+
+	return chan_autoservice ? 0 : res;
+}
+
+static int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(readsf_name);
+	res |= ast_unregister_application(sendsf_name);
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	res = ast_register_application_xml(readsf_name, read_sf_exec);
+	res |= ast_register_application_xml(sendsf_name, sendsf_exec);
+
+	return res;
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "SF Sender and Receiver Applications");

apps/app_signal.c

@@ -0,0 +1,471 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2022, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * 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 Channel signaling applications
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/file.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+
+/*** DOCUMENTATION
+	<application name="Signal" language="en_US">
+		<synopsis>
+			Sends a signal to any waiting channels.
+		</synopsis>
+		<syntax>
+			<parameter name="signalname" required="true">
+				<para>Name of signal to send.</para>
+			</parameter>
+			<parameter name="payload" required="false">
+				<para>Payload data to deliver.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Sends a named signal to any channels that may be
+			waiting for one. Acts as a producer in a simple
+			message queue.</para>
+			<variablelist>
+				<variable name="SIGNALSTATUS">
+					<value name="SUCCESS">
+						Signal was successfully sent to at least
+						one listener for processing.
+					</value>
+					<value name="FAILURE">
+						Signal could not be sent or nobody
+						was listening for this signal.
+					</value>
+				</variable>
+			</variablelist>
+			<example title="Send a signal named workdone">
+			same => n,Signal(workdone,Work has completed)
+			</example>
+		</description>
+		<see-also>
+			<ref type="application">WaitForSignal</ref>
+		</see-also>
+	</application>
+	<application name="WaitForSignal" language="en_US">
+		<synopsis>
+			Waits for a named signal on a channel.
+		</synopsis>
+		<syntax>
+			<parameter name="signalname" required="true">
+				<para>Name of signal to send.</para>
+			</parameter>
+			<parameter name="signaltimeout" required="false">
+				<para>Maximum time, in seconds, to wait for signal.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Waits for <replaceable>signaltimeout</replaceable> seconds on the current
+			channel to receive a signal with name <replaceable>signalname</replaceable>.
+			Acts as a consumer in a simple message queue.</para>
+			<para>Result of signal wait will be stored in the following variables:</para>
+			<variablelist>
+				<variable name="WAITFORSIGNALSTATUS">
+					<value name="SIGNALED">
+						Signal was received.
+					</value>
+					<value name="TIMEOUT">
+						Timed out waiting for signal.
+					</value>
+					<value name="HANGUP">
+						Channel hung up before signal was received.
+					</value>
+				</variable>
+				<variable name="WAITFORSIGNALPAYLOAD">
+					<para>Data payload attached to signal, if it exists</para>
+				</variable>
+			</variablelist>
+			<example title="Wait for the workdone signal, indefinitely, and print out payload">
+			same => n,WaitForSignal(workdone)
+			same => n,NoOp(Received: ${WAITFORSIGNALPAYLOAD})
+			</example>
+		</description>
+		<see-also>
+			<ref type="application">Signal</ref>
+		</see-also>
+	</application>
+ ***/
+
+static const char * const app = "Signal";
+static const char * const app2 = "WaitForSignal";
+
+struct signalitem {
+	ast_mutex_t lock;
+	char name[AST_MAX_CONTEXT];
+	int sig_alert_pipe[2];
+	int watchers;
+	unsigned int signaled:1;
+	char *payload;
+	AST_LIST_ENTRY(signalitem) entry;		/*!< Next Signal item */
+};
+
+static AST_RWLIST_HEAD_STATIC(signals, signalitem);
+
+static struct signalitem *alloc_signal(const char *sname)
+{
+	struct signalitem *s;
+
+	if (!(s = ast_calloc(1, sizeof(*s)))) {
+		return NULL;
+	}
+
+	ast_mutex_init(&s->lock);
+	ast_copy_string(s->name, sname, sizeof(s->name));
+
+	s->sig_alert_pipe[0] = -1;
+	s->sig_alert_pipe[1] = -1;
+	s->watchers = 0;
+	s->payload = NULL;
+	ast_alertpipe_init(s->sig_alert_pipe);
+
+	return s;
+}
+
+static int dealloc_signal(struct signalitem *s)
+{
+	if (s->watchers) { /* somebody is still using us... refuse to go away */
+		ast_debug(1, "Signal '%s' is still being used by %d listener(s)\n", s->name, s->watchers);
+		return -1;
+	}
+	ast_alertpipe_close(s->sig_alert_pipe);
+	ast_mutex_destroy(&s->lock);
+	if (s->payload) {
+		ast_free(s->payload);
+		s->payload = NULL;
+	}
+	ast_free(s);
+	s = NULL;
+	return 0;
+}
+
+static int remove_signal(char *sname)
+{
+	int res = -1;
+	struct signalitem *s;
+
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&signals, s, entry) {
+		if (!strcmp(s->name, sname)) {
+			AST_LIST_REMOVE_CURRENT(entry);
+			res = dealloc_signal(s);
+			ast_debug(1, "Removed signal '%s'\n", sname);
+		}
+	}
+	AST_LIST_TRAVERSE_SAFE_END;
+
+	return res;
+}
+
+static struct signalitem *get_signal(char *sname, int addnew)
+{
+	struct signalitem *s = NULL;
+	AST_RWLIST_WRLOCK(&signals);
+	AST_LIST_TRAVERSE(&signals, s, entry) {
+		if (!strcasecmp(s->name, sname)) {
+			ast_debug(1, "Using existing signal item '%s'\n", sname);
+			break;
+		}
+	}
+	if (!s) {
+		if (addnew) { /* signal doesn't exist, so create it */
+			s = alloc_signal(sname);
+			/* Totally fail if we fail to find/create an entry */
+			if (s) {
+				ast_debug(1, "Created new signal item '%s'\n", sname);
+				AST_RWLIST_INSERT_HEAD(&signals, s, entry);
+			} else {
+				ast_log(LOG_WARNING, "Failed to create signal item for '%s'\n", sname);
+			}
+		} else {
+			ast_debug(1, "Signal '%s' doesn't exist, and not creating it\n", sname);
+		}
+	}
+	AST_RWLIST_UNLOCK(&signals);
+	return s;
+}
+
+static int wait_for_signal_or_hangup(struct ast_channel *chan, char *signame, int timeout)
+{
+	struct signalitem *s = NULL;
+	int ms, remaining_time, res = 1, goaway = 0;
+	struct timeval start;
+	struct ast_frame *frame = NULL;
+
+	remaining_time = timeout;
+	start = ast_tvnow();
+
+	s = get_signal(signame, 1);
+
+	ast_mutex_lock(&s->lock);
+	s->watchers = s->watchers + 1; /* we unlock, because a) other people need to use this and */
+	ast_mutex_unlock(&s->lock); /* b) the signal will be available to us as long as watchers > 0 */
+
+	while (timeout == 0 || remaining_time > 0) {
+		int ofd, exception;
+
+		ms = 1000;
+		errno = 0;
+		if (ast_waitfor_nandfds(&chan, 1, &s->sig_alert_pipe[0], 1, &exception, &ofd, &ms)) { /* channel won */
+			if (!(frame = ast_read(chan))) { /* channel hung up */
+				ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
+				res = -1;
+				break;
+			} else {
+				ast_frfree(frame); /* handle frames */
+			}
+		} else if (ofd == s->sig_alert_pipe[0]) { /* fd won */
+			if (ast_alertpipe_read(s->sig_alert_pipe) == AST_ALERT_READ_SUCCESS) {
+				ast_debug(1, "Alert pipe has data for us\n");
+				res = 0;
+				break;
+			} else {
+				ast_debug(1, "Alert pipe does not have data for us\n");
+			}
+		} else { /* nobody won */
+			if (ms && (ofd < 0)) {
+				if (!((errno == 0) || (errno == EINTR))) {
+					ast_log(LOG_WARNING, "Something bad happened while channel '%s' was polling.\n", ast_channel_name(chan));
+					break;
+				}
+			} /* else, nothing happened */
+		}
+		if (timeout) {
+			remaining_time = ast_remaining_ms(start, timeout);
+		}
+	}
+
+	/* WRLOCK the list so that if we're going to destroy the signal now, nobody else can grab it before that happens. */
+	AST_RWLIST_WRLOCK(&signals);
+	ast_mutex_lock(&s->lock);
+	if (s->payload) {
+		pbx_builtin_setvar_helper(chan, "WAITFORSIGNALPAYLOAD", s->payload);
+	}
+	s->watchers = s->watchers - 1;
+	if (s->watchers) { /* folks are still waiting for this, pass it on... */
+		int save_errno = errno;
+		if (ast_alertpipe_write(s->sig_alert_pipe)) {
+			ast_log(LOG_WARNING, "%s: write() failed: %s\n", __FUNCTION__, strerror(errno));
+		}
+		errno = save_errno;
+	} else { /* nobody else is waiting for this */
+		goaway = 1; /* we were the last guy using this, so mark signal item for destruction */
+	}
+	ast_mutex_unlock(&s->lock);
+
+	if (goaway) {
+		/* remove_signal calls ast_mutex_destroy, so don't call it with the mutex itself locked. */
+		remove_signal(signame);
+	}
+	AST_RWLIST_UNLOCK(&signals);
+
+	return res;
+}
+
+static int send_signal(char *signame, char *payload)
+{
+	struct signalitem *s;
+	int save_errno = errno;
+	int res = 0;
+
+	s = get_signal(signame, 0); /* if signal doesn't exist already, no point in creating it, because nobody could be waiting for it! */
+
+	if (!s) {
+		return -1; /* this signal didn't exist, so we can't send a signal for it */
+	}
+
+	/* at this point, we know someone is listening, since signals are destroyed when watchers gets down to 0 */
+	ast_mutex_lock(&s->lock);
+	s->signaled = 1;
+	if (payload && *payload) {
+		int len = strlen(payload);
+		if (s->payload) {
+			ast_free(s->payload); /* if there was already a payload, replace it */
+			s->payload = NULL;
+		}
+		s->payload = ast_malloc(len + 1);
+		if (!s->payload) {
+			ast_log(LOG_WARNING, "Failed to allocate signal payload '%s'\n", payload);
+		} else {
+			ast_copy_string(s->payload, payload, len + 1);
+		}
+	}
+	if (ast_alertpipe_write(s->sig_alert_pipe)) {
+		ast_log(LOG_WARNING, "%s: write() failed: %s\n", __FUNCTION__, strerror(errno));
+		s->signaled = 0; /* okay, so we didn't send a signal after all... */
+		res = -1;
+	}
+	errno = save_errno;
+	ast_debug(1, "Sent '%s' signal to %d listeners\n", signame, s->watchers);
+	ast_mutex_unlock(&s->lock);
+
+	return res;
+}
+
+static int waitsignal_exec(struct ast_channel *chan, const char *data)
+{
+	char *argcopy;
+	int r = 0, timeoutms = 0;
+	double timeout = 0;
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(signame);
+		AST_APP_ARG(sigtimeout);
+	);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Signal() requires arguments\n");
+		return -1;
+	}
+
+	argcopy = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, argcopy);
+
+	if (ast_strlen_zero(args.signame)) {
+		ast_log(LOG_WARNING, "Missing signal name\n");
+		return -1;
+	}
+	if (strlen(args.signame) >= AST_MAX_CONTEXT) {
+		ast_log(LOG_WARNING, "Signal name '%s' is too long\n", args.signame);
+		return -1;
+	}
+	if (!ast_strlen_zero(args.sigtimeout)) {
+		if (sscanf(args.sigtimeout, "%30lg", &timeout) != 1 || timeout < 0) {
+			ast_log(LOG_WARNING, "Invalid timeout provided: %s. Defaulting to no timeout.\n", args.sigtimeout);
+		} else {
+			timeoutms = timeout * 1000; /* sec to msec */
+		}
+	}
+
+	if (timeout > 0) {
+		ast_debug(1, "Waiting for signal '%s' for %d ms\n", args.signame, timeoutms);
+	} else {
+		ast_debug(1, "Waiting for signal '%s', indefinitely\n", args.signame);
+	}
+
+	r = wait_for_signal_or_hangup(chan, args.signame, timeoutms);
+
+	if (r == 1) {
+		ast_verb(3, "Channel '%s' timed out, waiting for signal '%s'\n", ast_channel_name(chan), args.signame);
+		pbx_builtin_setvar_helper(chan, "WAITFORSIGNALSTATUS", "TIMEOUT");
+	} else if (!r) {
+		ast_verb(3, "Received signal '%s' on channel '%s'\n", args.signame, ast_channel_name(chan));
+		pbx_builtin_setvar_helper(chan, "WAITFORSIGNALSTATUS", "SIGNALED");
+	} else {
+		pbx_builtin_setvar_helper(chan, "WAITFORSIGNALSTATUS", "HANGUP");
+		ast_verb(3, "Channel '%s' hung up\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	return 0;
+}
+
+static int signal_exec(struct ast_channel *chan, const char *data)
+{
+	char *argcopy;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(signame);
+		AST_APP_ARG(payload);
+	);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Signal() requires arguments\n");
+		return -1;
+	}
+
+	argcopy = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, argcopy);
+
+	if (ast_strlen_zero(args.signame)) {
+		ast_log(LOG_WARNING, "Missing signal name\n");
+		return -1;
+	}
+	if (strlen(args.signame) >= AST_MAX_CONTEXT) {
+		ast_log(LOG_WARNING, "Signal name '%s' is too long\n", args.signame);
+		return -1;
+	}
+
+	if (send_signal(args.signame, args.payload)) {
+		pbx_builtin_setvar_helper(chan, "SIGNALSTATUS", "FAILURE");
+	} else {
+		pbx_builtin_setvar_helper(chan, "SIGNALSTATUS", "SUCCESS");
+	}
+
+	return 0;
+}
+
+static int unload_module(void)
+{
+	struct signalitem *s;
+	int res = 0;
+
+	/* To avoid a locking nightmare, and for logistical reasons, this module
+	 * will refuse to unload if watchers > 0. That way we know a signal's
+	 * pipe won't disappear while it's being used. */
+
+	AST_RWLIST_WRLOCK(&signals);
+	/* Don't just use AST_RWLIST_REMOVE_HEAD, because if dealloc_signal fails, it should stay in the list. */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&signals, s, entry) {
+		int mres = dealloc_signal(s);
+		res |= mres;
+		if (!mres) {
+			AST_LIST_REMOVE_CURRENT(entry);
+		}
+	}
+	AST_LIST_TRAVERSE_SAFE_END;
+	AST_RWLIST_UNLOCK(&signals);
+
+	/* One or more signals still has watchers. */
+	if (res) {
+		ast_log(LOG_WARNING, "One or more signals is currently in use. Unload failed.\n");
+		return res;
+	}
+
+	res |= ast_unregister_application(app);
+	res |= ast_unregister_application(app2);
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	res = ast_register_application_xml(app, signal_exec);
+	res |= ast_register_application_xml(app2, waitsignal_exec);
+
+	return res;
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Channel Signaling Applications");

apps/app_speech_utils.c

@@ -26,8 +26,8 @@
  */
 
 /*** MODULEINFO
-	<support_level>core</support_level>
 	<depend>res_speech</depend>
+	<support_level>core</support_level>
  ***/
 
 #include "asterisk.h"

apps/app_stack.c

@@ -377,6 +377,26 @@ static int pop_exec(struct ast_channel *chan, const char *data)
 	return res;
 }
 
+static int frames_left(struct ast_channel *chan)
+{
+	struct ast_datastore *stack_store;
+	struct gosub_stack_list *oldlist;
+	int exists;
+
+	ast_channel_lock(chan);
+	stack_store = ast_channel_datastore_find(chan, &stack_info, NULL);
+	if (!stack_store) {
+		ast_channel_unlock(chan);
+		return -1;
+	}
+	oldlist = stack_store->data;
+	AST_LIST_LOCK(oldlist);
+	exists = oldlist->first ? 1 : 0;
+	AST_LIST_UNLOCK(oldlist);
+	ast_channel_unlock(chan);
+	return exists;
+}
+
 static int return_exec(struct ast_channel *chan, const char *data)
 {
 	struct ast_datastore *stack_store;
@@ -384,6 +404,7 @@ static int return_exec(struct ast_channel *chan, const char *data)
 	struct gosub_stack_list *oldlist;
 	const char *retval = data;
 	int res = 0;
+	int lastframe;
 
 	ast_channel_lock(chan);
 	if (!(stack_store = ast_channel_datastore_find(chan, &stack_info, NULL))) {
@@ -395,6 +416,7 @@ static int return_exec(struct ast_channel *chan, const char *data)
 	oldlist = stack_store->data;
 	AST_LIST_LOCK(oldlist);
 	oldframe = AST_LIST_REMOVE_HEAD(oldlist, entries);
+	lastframe = oldlist->first ? 0 : 1;
 	AST_LIST_UNLOCK(oldlist);
 
 	if (!oldframe) {
@@ -412,12 +434,19 @@ static int return_exec(struct ast_channel *chan, const char *data)
 	 * what was there before.  Channels that do not have a PBX may
 	 * not have the context or exten set.
 	 */
-	ast_channel_context_set(chan, oldframe->context);
-	ast_channel_exten_set(chan, oldframe->extension);
-	if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_IN_AUTOLOOP)) {
-		--oldframe->priority;
+	if (ast_channel_pbx(chan) || !lastframe) {
+		/* If there's no PBX, the "old location" is simply
+		 * the configured context for the device, such as
+		 * for pre-dial handlers, and restoring this location
+		 * is nonsensical. So if no PBX and there are no further
+		 * frames, leave the location as it is. */
+		ast_channel_context_set(chan, oldframe->context);
+		ast_channel_exten_set(chan, oldframe->extension);
+		if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_IN_AUTOLOOP)) {
+			--oldframe->priority;
+		}
+		ast_channel_priority_set(chan, oldframe->priority);
 	}
-	ast_channel_priority_set(chan, oldframe->priority);
 	ast_set2_flag(ast_channel_flags(chan), oldframe->in_subroutine, AST_FLAG_SUBROUTINE_EXEC);
 
 	gosub_release_frame(chan, oldframe);
@@ -590,8 +619,8 @@ static int gosub_exec(struct ast_channel *chan, const char *data)
 	ast_channel_unlock(chan);
 
 	if (!ast_exists_extension(chan, dest_context, dest_exten, dest_priority, caller_id)) {
-		ast_log(LOG_ERROR, "Attempt to reach a non-existent destination for %s: (Context:%s, Extension:%s, Priority:%d)\n",
-			app_gosub, dest_context, dest_exten, dest_priority);
+		ast_log(LOG_ERROR, "%s attempted to reach non-existent destination '%s,%s,%d' from '%s,%s,%d'",
+			app_gosub, dest_context, dest_exten, dest_priority, orig_context, orig_exten, orig_priority);
 		goto error_exit;
 	}
 
@@ -924,8 +953,6 @@ static struct ast_custom_function stackpeek_function = {
  * \param chan Channel to balance stack on.
  *
  * \note The channel is already locked when called.
- *
- * \return Nothing
  */
 static void balance_stack(struct ast_channel *chan)
 {
@@ -1070,15 +1097,18 @@ static int gosub_run(struct ast_channel *chan, const char *sub_args, int ignore_
 				ast_channel_priority(chan), ast_channel_name(chan));
 		}
 
-		/* Did the routine return? */
-		if (ast_channel_priority(chan) == saved_priority
+		/* Did the routine return?
+		 * For things like predial where there's no PBX on the channel yet,
+		 * the last return leaves the location alone so we can print it out correctly here.
+		 * So to ensure we finished properly, make sure there are no frames left in that case. */
+		if ((!ast_channel_pbx(chan) && !frames_left(chan)) || (ast_channel_priority(chan) == saved_priority
 			&& !strcmp(ast_channel_context(chan), saved_context)
-			&& !strcmp(ast_channel_exten(chan), saved_exten)) {
+			&& !strcmp(ast_channel_exten(chan), saved_exten))) {
 			ast_verb(3, "%s Internal %s(%s) complete GOSUB_RETVAL=%s\n",
 				ast_channel_name(chan), app_gosub, sub_args,
 				S_OR(pbx_builtin_getvar_helper(chan, "GOSUB_RETVAL"), ""));
 		} else {
-			ast_log(LOG_NOTICE, "%s Abnormal '%s(%s)' exit.  Popping routine return locations.\n",
+			ast_log(LOG_WARNING, "%s Abnormal '%s(%s)' exit.  Popping routine return locations.\n",
 				ast_channel_name(chan), app_gosub, sub_args);
 			balance_stack(chan);
 			pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", "");

apps/app_stasis.c

@@ -79,7 +79,7 @@
 /*! \brief Dialplan application name */
 static const char *stasis = "Stasis";
 
-/*! /brief Stasis dialplan application callback */
+/*! \brief Stasis dialplan application callback */
 static int app_exec(struct ast_channel *chan, const char *data)
 {
 	char *parse = NULL;

apps/app_statsd.c

@@ -78,7 +78,7 @@ static const char app[] = "StatsD";
  *
  * \param value The value of the statistic to be sent to StatsD.
  *
- * This function checks to see if the value given to the StatsD daialplan
+ * This function checks to see if the value given to the StatsD dailplan
  * application is within the allowed range of [-2^63, 2^63] as specified by StatsD.
  *
  * \retval zero on success.
@@ -100,7 +100,7 @@ static int value_in_range(const char *value) {
  *
  * \param value The value of the statistic to be sent to StatsD.
  *
- * This function checks to see if the value given to the StatsD daialplan
+ * This function checks to see if the value given to the StatsD dailplan
  * application is within the allowed range of [0, 2^64] as specified by StatsD.
  *
  * \retval zero on success.

apps/app_talkdetect.c

@@ -254,4 +254,4 @@ static int load_module(void)
 	return ast_register_application_xml(app, background_detect_exec);
 }
 
-AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Playback with Talk Detection");
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Playback with Talk Detection");

apps/app_test.c

@@ -352,7 +352,7 @@ static int testserver_exec(struct ast_channel *chan, const char *data)
 	if (!res)
 		res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
 	ast_debug(1, "read test identifier: %s\n", testid);
-	/* Check for sneakyness */
+	/* Check for sneakiness */
 	if (strchr(testid, '/'))
 		res = -1;
 	if ((res >=0) && (!ast_strlen_zero(testid))) {

apps/app_transfer.c

@@ -69,6 +69,14 @@
 						Transfer unsupported by channel driver.
 					</value>
 				</variable>
+				<variable name="TRANSFERSTATUSPROTOCOL">
+					<value name="0">
+						No error.
+					</value>
+					<value name="3xx-6xx">
+						SIP example - Error result code.
+					</value>
+				</variable>
 			</variablelist>
 		</description>
 	</application>
@@ -85,6 +93,8 @@ static int transfer_exec(struct ast_channel *chan, const char *data)
 	char *dest = NULL;
 	char *status;
 	char *parse;
+	int protocol = 0;
+	char status_protocol[20];
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(dest);
 	);
@@ -92,6 +102,8 @@ static int transfer_exec(struct ast_channel *chan, const char *data)
 	if (ast_strlen_zero((char *)data)) {
 		ast_log(LOG_WARNING, "Transfer requires an argument ([Tech/]destination)\n");
 		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
+		snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
+		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
 		return 0;
 	} else
 		parse = ast_strdupa(data);
@@ -106,6 +118,8 @@ static int transfer_exec(struct ast_channel *chan, const char *data)
 		/* Allow execution only if the Tech/destination agrees with the type of the channel */
 		if (strncasecmp(ast_channel_tech(chan)->type, tech, len)) {
 			pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
+			snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
+			pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
 			return 0;
 		}
 	}
@@ -113,10 +127,14 @@ static int transfer_exec(struct ast_channel *chan, const char *data)
 	/* Check if the channel supports transfer before we try it */
 	if (!ast_channel_tech(chan)->transfer) {
 		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "UNSUPPORTED");
+		snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
+		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
 		return 0;
 	}
 
-	res = ast_transfer(chan, dest);
+	/* New transfer API returns a protocol code
+	   SIP example, 0 = success, 3xx-6xx are sip error codes for the REFER */
+	res = ast_transfer_protocol(chan, dest, &protocol);
 
 	if (res < 0) {
 		status = "FAILURE";
@@ -126,7 +144,11 @@ static int transfer_exec(struct ast_channel *chan, const char *data)
 		res = 0;
 	}
 
+	snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
+	ast_debug(1, "ast_transfer channel %s TRANSFERSTATUS=%s, TRANSFERSTATUSPROTOCOL=%s\n",
+		  ast_channel_name(chan), status, status_protocol);
 	pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", status);
+	pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
 
 	return res;
 }

apps/app_url.c

@@ -27,6 +27,8 @@
 
 /*** MODULEINFO
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_verbose.c

@@ -33,6 +33,7 @@
 #include "asterisk/module.h"
 #include "asterisk/app.h"
 #include "asterisk/channel.h"
+#include "asterisk/logger.h"
 
 static char *app_verbose = "Verbose";
 static char *app_log = "Log";
@@ -61,7 +62,8 @@ static char *app_log = "Log";
 		<syntax>
 			<parameter name="level" required="true">
 				<para>Level must be one of <literal>ERROR</literal>, <literal>WARNING</literal>, <literal>NOTICE</literal>,
-				<literal>DEBUG</literal>, <literal>VERBOSE</literal> or <literal>DTMF</literal>.</para>
+				<literal>DEBUG</literal>, <literal>VERBOSE</literal>, <literal>DTMF</literal>, or
+				the name of a custom dynamic logging level.</para>
 			</parameter>
 			<parameter name="message" required="true">
 				<para>Output text message.</para>
@@ -135,7 +137,7 @@ static int log_exec(struct ast_channel *chan, const char *data)
 	} else if (!strcasecmp(args.level, "DTMF")) {
 		lnum = __LOG_DTMF;
 	} else {
-		ast_log(LOG_ERROR, "Unknown log level: '%s'\n", args.level);
+		lnum = ast_logger_get_dynamic_level(args.level);
 	}
 
 	if (lnum > -1) {
@@ -143,6 +145,9 @@ static int log_exec(struct ast_channel *chan, const char *data)
 		snprintf(extension, sizeof(extension), "Ext. %s", ast_channel_exten(chan));
 
 		ast_log(lnum, extension, ast_channel_priority(chan), context, "%s\n", args.msg);
+	} else {
+		ast_log(LOG_ERROR, "Unknown log level: '%s'\n", args.level);
+		return 0;
 	}
 
 	return 0;

apps/app_voicemail.c

@@ -26,7 +26,7 @@
  *         (http://www.washington.edu/imap/)
  *
  * \par See also
- * \arg \ref Config_vm
+ * \arg \ref voicemail.conf "Config_voicemail"
  * \note For information about voicemail IMAP storage, https://wiki.asterisk.org/wiki/display/AST/IMAP+Voicemail+Storage
  * \ingroup applications
  * \todo This module requires res_adsi to load. This needs to be optional
@@ -135,6 +135,10 @@
 						<para>Accept digits for a new extension in context <replaceable>c</replaceable>,
 						if played during the greeting. Context defaults to the current context.</para>
 					</option>
+					<option name="e">
+						<para>Play greetings as early media -- only answer the channel just
+						before accepting the voice message.</para>
+					</option>
 					<option name="g">
 						<argument name="#" required="true" />
 						<para>Use the specified amount of gain when recording the voicemail
@@ -145,6 +149,16 @@
 						<para>Skip the playback of instructions for leaving a message to the
 						calling party.</para>
 					</option>
+					<option name="S">
+						<para>Skip the playback of instructions for leaving a message to the
+						calling party, but only if a greeting has been recorded by the
+						mailbox user.</para>
+					</option>
+					<option name="t">
+						<argument name="x" required="false" />
+						<para>Play a custom beep tone to the caller instead of the default one.
+						If this option is used but no file is specified, the beep is suppressed.</para>
+					</option>
 					<option name="u">
 						<para>Play the <literal>unavailable</literal> greeting.</para>
 					</option>
@@ -205,6 +219,10 @@
 						<para>Use the specified amount of gain when recording a voicemail message.
 						The units are whole-number decibels (dB).</para>
 					</option>
+					<option name="r">
+						<para>"Read only". Prevent user from deleting any messages.</para>
+						<para>This applies only to specific executions of VoiceMailMain, NOT the mailbox itself.</para>
+					</option>
 					<option name="s">
 						<para>Skip checking the passcode for the mailbox.</para>
 					</option>
@@ -315,6 +333,9 @@
 		<description>
 			<para>This application will say the recorded name of the voicemail user specified as the
 			argument to this application. If no context is provided, <literal>default</literal> is assumed.</para>
+			<para>Similar to the Background() application, playback of the recorded
+			name can be interrupted by entering an extension, which will be searched
+			for in the current context.</para>
 		</description>
 	</application>
 	<function name="VM_INFO" language="en_US">
@@ -512,7 +533,6 @@ static AST_LIST_HEAD_STATIC(vmstates, vmstate);
 /* Default mail command to mail voicemail. Change it with the
  * mailcmd= command in voicemail.conf */
 #define SENDMAIL "/usr/sbin/sendmail -t"
-
 #define INTRO "vm-intro"
 
 #define MAX_MAIL_BODY_CONTENT_SIZE 134217728L // 128 Mbyte
@@ -522,8 +542,6 @@ static AST_LIST_HEAD_STATIC(vmstates, vmstate);
 
 #define MINPASSWORD 0 /*!< Default minimum mailbox password length */
 
-#define BASELINELEN 72
-#define BASEMAXINLINE 256
 #ifdef IMAP_STORAGE
 #define ENDL "\r\n"
 #else
@@ -543,7 +561,7 @@ static AST_LIST_HEAD_STATIC(vmstates, vmstate);
 #define VM_FORCENAME     (1 << 7)   /*!< Have new users record their name */
 #define VM_FORCEGREET    (1 << 8)   /*!< Have new users record their greetings */
 #define VM_PBXSKIP       (1 << 9)   /*!< Skip the [PBX] preamble in the Subject line of emails */
-#define VM_DIRECFORWARD  (1 << 10)  /*!< Permit caller to use the Directory app for selecting to which mailbox to forward a VM */
+#define VM_DIRECTFORWARD (1 << 10)  /*!< Permit caller to use the Directory app for selecting to which mailbox to forward a VM */
 #define VM_ATTACH        (1 << 11)  /*!< Attach message to voicemail notifications? */
 #define VM_DELETE        (1 << 12)  /*!< Delete message after sending notification */
 #define VM_ALLOCED       (1 << 13)  /*!< Structure was malloc'ed, instead of placed in a return (usually static) buffer */
@@ -552,6 +570,7 @@ static AST_LIST_HEAD_STATIC(vmstates, vmstate);
 #define VM_MOVEHEARD     (1 << 16)  /*!< Move a "heard" message to Old after listening to it */
 #define VM_MESSAGEWRAP   (1 << 17)  /*!< Wrap around from the last message to the first, and vice-versa */
 #define VM_FWDURGAUTO    (1 << 18)  /*!< Autoset of Urgent flag on forwarded Urgent messages set globally */
+#define VM_EMAIL_EXT_RECS (1 << 19)  /*!< Send voicemail emails when an external recording is added to a mailbox */
 #define ERROR_LOCK_PATH  -100
 #define ERROR_MAX_MSGS   -101
 #define OPERATOR_EXIT     300
@@ -574,15 +593,20 @@ enum vm_option_flags {
 	OPT_AUTOPLAY =         (1 << 6),
 	OPT_DTMFEXIT =         (1 << 7),
 	OPT_MESSAGE_Urgent =   (1 << 8),
-	OPT_MESSAGE_PRIORITY = (1 << 9)
+	OPT_MESSAGE_PRIORITY = (1 << 9),
+	OPT_EARLYM_GREETING =  (1 << 10),
+	OPT_BEEP =             (1 << 11),
+	OPT_SILENT_IF_GREET =  (1 << 12),
+	OPT_READONLY =         (1 << 13),
 };
 
 enum vm_option_args {
 	OPT_ARG_RECORDGAIN = 0,
 	OPT_ARG_PLAYFOLDER = 1,
 	OPT_ARG_DTMFEXIT   = 2,
+	OPT_ARG_BEEP_TONE  = 3,
 	/* This *must* be the last value in this enum! */
-	OPT_ARG_ARRAY_SIZE = 3,
+	OPT_ARG_ARRAY_SIZE = 4,
 };
 
 enum vm_passwordlocation {
@@ -593,6 +617,7 @@ enum vm_passwordlocation {
 
 AST_APP_OPTIONS(vm_app_options, {
 	AST_APP_OPTION('s', OPT_SILENT),
+	AST_APP_OPTION('S', OPT_SILENT_IF_GREET),
 	AST_APP_OPTION('b', OPT_BUSY_GREETING),
 	AST_APP_OPTION('u', OPT_UNAVAIL_GREETING),
 	AST_APP_OPTION_ARG('g', OPT_RECORDGAIN, OPT_ARG_RECORDGAIN),
@@ -600,7 +625,10 @@ AST_APP_OPTIONS(vm_app_options, {
 	AST_APP_OPTION('p', OPT_PREPEND_MAILBOX),
 	AST_APP_OPTION_ARG('a', OPT_AUTOPLAY, OPT_ARG_PLAYFOLDER),
 	AST_APP_OPTION('U', OPT_MESSAGE_Urgent),
-	AST_APP_OPTION('P', OPT_MESSAGE_PRIORITY)
+	AST_APP_OPTION('P', OPT_MESSAGE_PRIORITY),
+	AST_APP_OPTION('e', OPT_EARLYM_GREETING),
+	AST_APP_OPTION_ARG('t', OPT_BEEP, OPT_ARG_BEEP_TONE),
+	AST_APP_OPTION('r', OPT_READONLY),
 });
 
 static const char * const mailbox_folders[] = {
@@ -720,14 +748,6 @@ and vm-Old are spelled plural, to make them sound more as folder name than an ad
 
 */
 
-struct baseio {
-	int iocp;
-	int iolen;
-	int linelength;
-	int ateof;
-	unsigned char iobuf[BASEMAXINLINE];
-};
-
 #define MAX_VM_MBOX_ID_LEN (AST_MAX_EXTENSION)
 #define MAX_VM_CONTEXT_LEN (AST_MAX_CONTEXT)
 /* MAX_VM_MAILBOX_LEN allows enough room for the '@' and NULL terminator */
@@ -1067,7 +1087,7 @@ static int vm_test_create_user(const char *context, const char *mailbox);
  * \brief Parse the given mailbox_id into mailbox and context.
  * \since 12.0.0
  *
- * \param mailbox_id The mailbox@context string to separate.
+ * \param mailbox_id The mailbox\@context string to separate.
  * \param mailbox Where the mailbox part will start.
  * \param context Where the context part will start.  ("default" if not present)
  *
@@ -1239,6 +1259,8 @@ static void apply_option(struct ast_vm_user *vmu, const char *var, const char *v
 		ast_set2_flag(vmu, ast_true(value), VM_ATTACH);
 	} else if (!strcasecmp(var, "attachfmt")) {
 		ast_copy_string(vmu->attachfmt, value, sizeof(vmu->attachfmt));
+	} else if (!strcasecmp(var, "attachextrecs")) {
+		ast_set2_flag(vmu, ast_true(value), VM_EMAIL_EXT_RECS);
 	} else if (!strcasecmp(var, "serveremail")) {
 		ast_copy_string(vmu->serveremail, value, sizeof(vmu->serveremail));
 	} else if (!strcasecmp(var, "fromstring")) {
@@ -1904,22 +1926,6 @@ static int make_file(char *dest, const int len, const char *dir, const int num)
 	return snprintf(dest, len, "%s/msg%04d", dir, num);
 }
 
-/* same as mkstemp, but return a FILE * */
-static FILE *vm_mkftemp(char *template)
-{
-	FILE *p = NULL;
-	int pfd = mkstemp(template);
-	chmod(template, VOICEMAIL_FILE_MODE & ~my_umask);
-	if (pfd > -1) {
-		p = fdopen(pfd, "w+");
-		if (!p) {
-			close(pfd);
-			pfd = -1;
-		}
-	}
-	return p;
-}
-
 /*! \brief basically mkdir -p $dest/$context/$ext/$folder
  * \param dest    String. base directory.
  * \param len     Length of dest.
@@ -2673,7 +2679,7 @@ static int imap_store_file(const char *dir, const char *mailboxuser, const char
 
 	/* Make a temporary file instead of piping directly to sendmail, in case the mail
 	   command hangs. */
-	if (!(p = vm_mkftemp(tmp))) {
+	if (!(p = ast_file_mkftemp(tmp, VOICEMAIL_FILE_MODE & ~my_umask))) {
 		ast_log(AST_LOG_WARNING, "Unable to store '%s' (can't create temporary file)\n", fn);
 		if (tempcopy) {
 			ast_free(vmu->email);
@@ -2826,9 +2832,9 @@ static int inboxcount2(const char *mailbox_context, int *urgentmsgs, int *newmsg
 	return 0;
 }
 
-/**
+/*!
  * \brief Determines if the given folder has messages.
- * \param mailbox The @ delimited string for user@context. If no context is found, uses 'default' for the context.
+ * \param mailbox The \@ delimited string for user\@context. If no context is found, uses 'default' for the context.
  * \param folder the folder to look in
  *
  * This function is used when the mailbox is stored in an IMAP back end.
@@ -2868,6 +2874,7 @@ static int has_voicemail(const char *mailbox, const char *folder)
  * \param recip
  * \param fmt
  * \param dir
+ * \param flag, dest_folder
  *
  * This works with IMAP storage based mailboxes.
  *
@@ -3000,7 +3007,7 @@ static int init_mailstream(struct vm_state *vms, int box)
 	ast_mutex_lock(&vms->lock);
 	ast_mutex_lock(&mail_open_lock);
 	vms->mailstream = mail_open (stream, tmp, debug ? OP_DEBUG : NIL);
-	/* Create the folder if it dosn't exist */
+	/* Create the folder if it doesn't exist */
 	if (vms->mailstream && !mail_status(vms->mailstream, tmp, SA_UIDNEXT)) {
 		mail_create(vms->mailstream, tmp);
 	}
@@ -3825,6 +3832,7 @@ static int retrieve_file(char *dir, int msgnum)
 	char fn[PATH_MAX];
 	char full_fn[PATH_MAX];
 	char msgnums[80];
+	char msg_id[MSG_ID_LEN] = "";
 	char *argv[] = { dir, msgnums };
 	struct generic_prepare_struct gps = { .sql = sql, .argc = 2, .argv = argv };
 	struct odbc_obj *obj;
@@ -3929,10 +3937,10 @@ static int retrieve_file(char *dir, int msgnum)
 		} else {
 			res = SQLGetData(stmt, x + 1, SQL_CHAR, rowdata, sizeof(rowdata), NULL);
 			if (res == SQL_NULL_DATA && !strcasecmp(coltitle, "msg_id")) {
-				char msg_id[MSG_ID_LEN];
+				/* Generate msg_id now, but don't store it until we're done with this
+				   connection */
 				generate_msg_id(msg_id);
 				snprintf(rowdata, sizeof(rowdata), "%s", msg_id);
-				odbc_update_msg_id(dir, msgnum, msg_id);
 			} else if (res == SQL_NULL_DATA && !strcasecmp(coltitle, "category")) {
 				/* Ignore null column value for category */
 				ast_debug(3, "Ignoring null category column in ODBC voicemail retrieve_file.\n");
@@ -3958,6 +3966,13 @@ bail:
 
 	ast_odbc_release_obj(obj);
 
+	/* If res_odbc is configured to only allow a single database connection, we
+	   will deadlock if we try to do this before releasing the connection we
+	   were just using. */
+	if (!ast_strlen_zero(msg_id)) {
+		odbc_update_msg_id(dir, msgnum, msg_id);
+	}
+
 	return x - 1;
 }
 
@@ -4149,8 +4164,6 @@ bail:
  *
  * This method is used when mailboxes are stored in an ODBC back end.
  * The specified message is directly deleted from the database 'voicemessages' table.
- *
- * \return the value greater than zero on success to indicate the number of messages, less than zero on error.
  */
 static void delete_file(const char *sdir, int smsg)
 {
@@ -4471,15 +4484,16 @@ static void rename_file(char *sdir, int smsg, char *mailboxuser, char *mailboxco
  */
 static int remove_file(char *dir, int msgnum)
 {
-	char fn[PATH_MAX];
-	char full_fn[PATH_MAX];
+	char fn[PATH_MAX] = "";
+	char full_fn[PATH_MAX + 4]; /* Plus .txt */
 	char msgnums[80];
 
 	if (msgnum > -1) {
 		snprintf(msgnums, sizeof(msgnums), "%d", msgnum);
 		make_file(fn, sizeof(fn), dir, msgnum);
-	} else
+	} else {
 		ast_copy_string(fn, dir, sizeof(fn));
+	}
 	ast_filedelete(fn, NULL);
 	snprintf(full_fn, sizeof(full_fn), "%s.txt", fn);
 	unlink(full_fn);
@@ -4739,134 +4753,6 @@ static int vm_delete(char *file)
 	return ast_filedelete(file, NULL);
 }
 
-/*!
- * \brief utility used by inchar(), for base_encode()
- */
-static int inbuf(struct baseio *bio, FILE *fi)
-{
-	int l;
-
-	if (bio->ateof)
-		return 0;
-
-	if ((l = fread(bio->iobuf, 1, BASEMAXINLINE, fi)) != BASEMAXINLINE) {
-		bio->ateof = 1;
-		if (l == 0) {
-			/* Assume EOF */
-			return 0;
-		}
-	}
-
-	bio->iolen = l;
-	bio->iocp = 0;
-
-	return 1;
-}
-
-/*!
- * \brief utility used by base_encode()
- */
-static int inchar(struct baseio *bio, FILE *fi)
-{
-	if (bio->iocp>=bio->iolen) {
-		if (!inbuf(bio, fi))
-			return EOF;
-	}
-
-	return bio->iobuf[bio->iocp++];
-}
-
-/*!
- * \brief utility used by base_encode()
- */
-static int ochar(struct baseio *bio, int c, FILE *so)
-{
-	if (bio->linelength >= BASELINELEN) {
-		if (fputs(ENDL, so) == EOF) {
-			return -1;
-		}
-
-		bio->linelength = 0;
-	}
-
-	if (putc(((unsigned char) c), so) == EOF) {
-		return -1;
-	}
-
-	bio->linelength++;
-
-	return 1;
-}
-
-/*!
- * \brief Performs a base 64 encode algorithm on the contents of a File
- * \param filename The path to the file to be encoded. Must be readable, file is opened in read mode.
- * \param so A FILE handle to the output file to receive the base 64 encoded contents of the input file, identified by filename.
- *
- * TODO: shouldn't this (and the above 3 support functions) be put into some kind of external utility location, such as funcs/func_base64.c ?
- *
- * \return zero on success, -1 on error.
- */
-static int base_encode(char *filename, FILE *so)
-{
-	static const unsigned char dtable[] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K',
-		'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z', 'a', 'b', 'c', 'd', 'e', 'f',
-		'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z', '0',
-		'1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'};
-	int i, hiteof = 0;
-	FILE *fi;
-	struct baseio bio;
-
-	memset(&bio, 0, sizeof(bio));
-	bio.iocp = BASEMAXINLINE;
-
-	if (!(fi = fopen(filename, "rb"))) {
-		ast_log(AST_LOG_WARNING, "Failed to open file: %s: %s\n", filename, strerror(errno));
-		return -1;
-	}
-
-	while (!hiteof){
-		unsigned char igroup[3], ogroup[4];
-		int c, n;
-
-		memset(igroup, 0, sizeof(igroup));
-
-		for (n = 0; n < 3; n++) {
-			if ((c = inchar(&bio, fi)) == EOF) {
-				hiteof = 1;
-				break;
-			}
-
-			igroup[n] = (unsigned char) c;
-		}
-
-		if (n > 0) {
-			ogroup[0]= dtable[igroup[0] >> 2];
-			ogroup[1]= dtable[((igroup[0] & 3) << 4) | (igroup[1] >> 4)];
-			ogroup[2]= dtable[((igroup[1] & 0xF) << 2) | (igroup[2] >> 6)];
-			ogroup[3]= dtable[igroup[2] & 0x3F];
-
-			if (n < 3) {
-				ogroup[3] = '=';
-
-				if (n < 2)
-					ogroup[2] = '=';
-			}
-
-			for (i = 0; i < 4; i++)
-				ochar(&bio, ogroup[i], so);
-		}
-	}
-
-	fclose(fi);
-
-	if (fputs(ENDL, so) == EOF) {
-		return 0;
-	}
-
-	return 1;
-}
-
 static void prep_email_sub_vars(struct ast_channel *ast, struct ast_vm_user *vmu, int msgnum, char *context, char *mailbox, const char *fromfolder, char *cidnum, char *cidname, char *dur, char *date, const char *category, const char *flag)
 {
 	char callerid[256];
@@ -5055,7 +4941,7 @@ static const char *ast_str_encode_mime(struct ast_str **end, ssize_t maxlen, con
  * \param imap if == 1, indicates the target folder for the email notification to be sent to will be an IMAP mailstore. This causes additional mailbox headers to be set, which would facilitate searching for the email in the destination IMAP folder.
  * \param flag, msg_id
  *
- * The email body, and base 64 encoded attachement (if any) are stored to the file identified by *p. This method does not actually send the email.  That is done by invoking the configure 'mailcmd' and piping this generated file into it, or with the sendemail() function.
+ * The email body, and base 64 encoded attachment (if any) are stored to the file identified by *p. This method does not actually send the email.  That is done by invoking the configure 'mailcmd' and piping this generated file into it, or with the sendemail() function.
  */
 static void make_email_file(FILE *p,
 		char *srcemail,
@@ -5477,7 +5363,7 @@ static int add_email_attachment(FILE *p, struct ast_vm_user *vmu, char *format,
 		fprintf(p, "Content-Disposition: attachment; filename=\"%s\"" ENDL ENDL, filename);
 	else
 		fprintf(p, "Content-Disposition: attachment; filename=\"%s.%s\"" ENDL ENDL, greeting_attachment, format);
-	base_encode(fname, p);
+	ast_base64_encode_file_path(fname, p, ENDL);
 	if (last)
 		fprintf(p, ENDL ENDL "--%s--" ENDL "." ENDL, bound);
 
@@ -5530,7 +5416,7 @@ static int sendmail(char *srcemail,
 	ast_debug(3, "Attaching file '%s', format '%s', uservm is '%d', global is %u\n", attach, format, attach_user_voicemail, ast_test_flag((&globalflags), VM_ATTACH));
 	/* Make a temporary file instead of piping directly to sendmail, in case the mail
 	   command hangs */
-	if ((p = vm_mkftemp(tmp)) == NULL) {
+	if ((p = ast_file_mkftemp(tmp, VOICEMAIL_FILE_MODE & ~my_umask)) == NULL) {
 		ast_log(AST_LOG_WARNING, "Unable to launch '%s' (can't create temporary file)\n", mailcmd);
 		return -1;
 	} else {
@@ -5569,7 +5455,7 @@ static int sendpage(char *srcemail, char *pager, int msgnum, char *context, char
 		strip_control_and_high(cidname, enc_cidname, sizeof(enc_cidname));
 	}
 
-	if ((p = vm_mkftemp(tmp)) == NULL) {
+	if ((p = ast_file_mkftemp(tmp, VOICEMAIL_FILE_MODE & ~my_umask)) == NULL) {
 		ast_log(AST_LOG_WARNING, "Unable to launch '%s' (can't create temporary file)\n", mailcmd);
 		ast_free(str1);
 		ast_free(str2);
@@ -5915,9 +5801,9 @@ bail:
 	return nummsgs;
 }
 
-/**
+/*!
  * \brief Determines if the given folder has messages.
- * \param mailbox The @ delimited string for user@context. If no context is found, uses 'default' for the context.
+ * \param mailbox The \@ delimited string for user\@context. If no context is found, uses 'default' for the context.
  *
  * This function is used when the mailbox is stored in an ODBC back end.
  * This invokes the messagecount(). Here we are interested in the presence of messages (> 0) only, not the actual count.
@@ -6087,7 +5973,7 @@ static int __has_voicemail(const char *context, const char *mailbox, const char
 	return ret;
 }
 
-/**
+/*!
  * \brief Determines if the given folder has messages.
  * \param mailbox The \@ delimited string for user\@context. If no context is found, uses 'default' for the context.
  * \param folder the folder to look in
@@ -6120,7 +6006,7 @@ static int has_voicemail(const char *mailbox, const char *folder)
 
 /*!
  * \brief Check the given mailbox's message count.
- * \param mailbox The @ delimited string for user@context. If no context is found, uses 'default' for the context.
+ * \param mailbox The \@ delimited string for user\@context. If no context is found, uses 'default' for the context.
  * \param urgentmsgs  urgent message count.
  * \param newmsgs new message count.
  * \param oldmsgs old message count pointer
@@ -6260,6 +6146,7 @@ struct leave_vm_options {
 	unsigned int flags;
 	signed char record_gain;
 	char *exitcontext;
+	char *beeptone;
 };
 
 static void generate_msg_id(char *dst)
@@ -6535,6 +6422,12 @@ static int msg_create_from_file(struct ast_vm_recording_data *recdata)
 	 * to do both with one line and is also safe to use with file storage mode. Also, if we are using ODBC, now is a good
 	 * time to create the voicemail database entry. */
 	if (ast_fileexists(destination, NULL, NULL) > 0) {
+		struct ast_channel *chan = NULL;
+		char fmt[80];
+		char clid[80];
+		char cidnum[80], cidname[80];
+		int send_email;
+
 		if (ast_check_realtime("voicemail_data")) {
 			get_date(date, sizeof(date));
 			ast_store_realtime("voicemail_data",
@@ -6554,7 +6447,27 @@ static int msg_create_from_file(struct ast_vm_recording_data *recdata)
 		}
 
 		STORE(dir, recipient->mailbox, recipient->context, msgnum, NULL, recipient, fmt, 0, vms, "", msg_id);
-		notify_new_state(recipient);
+
+		send_email = ast_test_flag(recipient, VM_EMAIL_EXT_RECS);
+
+		if (send_email) {
+			/* Send an email if possible, fall back to just notifications if not. */
+			ast_copy_string(fmt, recdata->recording_ext, sizeof(fmt));
+			ast_copy_string(clid, recdata->call_callerid, sizeof(clid));
+			ast_callerid_split(clid, cidname, sizeof(cidname), cidnum, sizeof(cidnum));
+
+			/* recdata->call_callerchan itself no longer exists, so we can't use the real channel. Use a dummy one. */
+			chan = ast_dummy_channel_alloc();
+		}
+		if (chan) {
+			notify_new_message(chan, recipient, NULL, msgnum, duration, fmt, cidnum, cidname, "");
+			ast_channel_unref(chan);
+		} else {
+			if (send_email) { /* We tried and failed. */
+				ast_log(LOG_WARNING, "Failed to allocate dummy channel, email will not be sent\n");
+			}
+			notify_new_state(recipient);
+		}
 	}
 
 	free_user(recipient);
@@ -6576,8 +6489,6 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 {
 #ifdef IMAP_STORAGE
 	int newmsgs, oldmsgs;
-#else
-	char urgdir[PATH_MAX];
 #endif
 	char txtfile[PATH_MAX];
 	char tmptxtfile[PATH_MAX];
@@ -6757,8 +6668,14 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 #endif
 			RETRIEVE(prefile, -1, ext, context);
 		if (ast_fileexists(prefile, NULL, NULL) > 0) {
-			if (ast_streamfile(chan, prefile, ast_channel_language(chan)) > -1)
+			if (ast_streamfile(chan, prefile, ast_channel_language(chan)) > -1) {
+				/* We know we have a greeting at this point, so squelch the instructions
+				 * if that is what is being asked of us */
+				if (ast_test_flag(options, OPT_SILENT_IF_GREET)) {
+					ast_set_flag(options, OPT_SILENT);
+				}
 				res = ast_waitstream(chan, ecodes);
+			}
 #ifdef ODBC_STORAGE
 			if (success == -1) {
 				/* We couldn't retrieve the file from the database, but we found it on the file system. Let's put it in the database. */
@@ -6853,6 +6770,9 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 		return -1;
 	}
 	/* The meat of recording the message...  All the announcements and beeps have been played*/
+	if (ast_channel_state(chan) != AST_STATE_UP) {
+		ast_answer(chan);
+	}
 	ast_copy_string(fmt, vmfmts, sizeof(fmt));
 	if (!ast_strlen_zero(fmt)) {
 		char msg_id[MSG_ID_LEN] = "";
@@ -6920,7 +6840,10 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 		/* Now play the beep once we have the message number for our next message. */
 		if (res >= 0) {
 			/* Unless we're *really* silent, try to send the beep */
-			res = ast_stream_and_wait(chan, "beep", "");
+			/* Play default or custom beep, unless no beep desired */
+			if (!ast_strlen_zero(options->beeptone)) {
+				res = ast_stream_and_wait(chan, options->beeptone, "");
+			}
 		}
 
 		/* Store information in real-time storage */
@@ -6996,6 +6919,14 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 		}
 		res = play_record_review(chan, NULL, tmptxtfile, vmu->maxsecs, fmt, 1, vmu, &duration, &sound_duration, NULL, options->record_gain, vms, flag, msg_id, 0);
 
+		/* At this point, either we were instructed to make the message Urgent
+		   by arguments to VoiceMail or during the review process by the person
+		   leaving the message. So we update the directory where we want this
+		   message to go. */
+		if (!strcmp(flag, "Urgent")) {
+			create_dirpath(dir, sizeof(dir), vmu->context, ext, "Urgent");
+		}
+
 		if (txt) {
 			fprintf(txt, "flag=%s\n", flag);
 			if (sound_duration < vmu->minsecs) {
@@ -7076,25 +7007,7 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 							free_user(recip);
 						}
 					}
-#ifndef IMAP_STORAGE
-					if (!ast_strlen_zero(flag) && !strcmp(flag, "Urgent")) { /* If this is an Urgent message */
-						/* Move the message from INBOX to Urgent folder if this is urgent! */
-						char sfn[PATH_MAX];
-						char dfn[PATH_MAX];
-						int x;
-						/* It's easier just to try to make it than to check for its existence */
-						create_dirpath(urgdir, sizeof(urgdir), vmu->context, ext, "Urgent");
-						x = last_message_index(vmu, urgdir) + 1;
-						make_file(sfn, sizeof(sfn), dir, msgnum);
-						make_file(dfn, sizeof(dfn), urgdir, x);
-						ast_debug(5, "Created an Urgent message, moving file from %s to %s.\n", sfn, dfn);
-						RENAME(dir, msgnum, vmu->mailbox, vmu->context, urgdir, x, sfn, dfn);
-						/* Notification must happen for this new message in Urgent folder, not INBOX */
-						ast_copy_string(fn, dfn, sizeof(fn));
-						pbx_builtin_setvar_helper(chan, "VM_MESSAGEFILE", fn);
-						msgnum = x;
-					}
-#endif
+
 					/* Notification needs to happen after the copy, though. */
 					if (ast_fileexists(fn, NULL, NULL)) {
 #ifdef IMAP_STORAGE
@@ -7220,7 +7133,7 @@ static int save_to_folder(struct ast_vm_user *vmu, struct vm_state *vms, int msg
 	/* get the current mailbox so that we can point the mailstream back to it later */
 	curr_mbox = get_folder_by_name(vms->curbox);
 
-	/* Create the folder if it dosn't exist */
+	/* Create the folder if it doesn't exist */
 	imap_mailbox_name(mailbox, sizeof(mailbox), vms, box, 1); /* Get the full mailbox name */
 	if (vms->mailstream && !mail_status(vms->mailstream, mailbox, SA_UIDNEXT)) {
     		if (mail_create(vms->mailstream, mailbox) != NIL) {
@@ -7339,7 +7252,7 @@ static int adsi_load_vmail(struct ast_channel *chan, int *useadsi)
 	bytes = 0;
 	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 0, "Listen", "Listen", "1", 1);
 	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 1, "Folder", "Folder", "2", 1);
-	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 2, "Advanced", "Advnced", "3", 1);
+	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 2, "Advanced", "Advanced", "3", 1);
 	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 3, "Options", "Options", "0", 1);
 	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 4, "Help", "Help", "*", 1);
 	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 5, "Exit", "Exit", "#", 1);
@@ -8281,7 +8194,7 @@ static int forward_message(struct ast_channel *chan, char *context, struct vm_st
 	ast_test_suite_event_notify("FORWARD", "Message: entering forward message menu");
 	while (!res && !valid_extensions) {
 		int use_directory = 0;
-		if (ast_test_flag((&globalflags), VM_DIRECFORWARD)) {
+		if (ast_test_flag((&globalflags), VM_DIRECTFORWARD)) {
 			int done = 0;
 			int retries = 0;
 			cmd = 0;
@@ -8350,7 +8263,7 @@ static int forward_message(struct ast_channel *chan, char *context, struct vm_st
 				ast_channel_priority_set(chan, old_priority);
 			} else {
 				ast_log(AST_LOG_WARNING, "Could not find the Directory application, disabling directory_forward\n");
-				ast_clear_flag((&globalflags), VM_DIRECFORWARD);
+				ast_clear_flag((&globalflags), VM_DIRECTFORWARD);
 			}
 		} else {
 			/* Ask for an extension */
@@ -8465,6 +8378,7 @@ static int forward_message(struct ast_channel *chan, char *context, struct vm_st
 		/* Send VoiceMail */
 		memset(&leave_options, 0, sizeof(leave_options));
 		leave_options.record_gain = record_gain;
+		leave_options.beeptone = "beep";
 		cmd = leave_voicemail(chan, mailbox, &leave_options);
 	} else {
 		/* Forward VoiceMail */
@@ -8673,7 +8587,7 @@ static int play_message_datetime(struct ast_channel *chan, struct ast_vm_user *v
 		res = ast_say_date_with_format(chan, t, AST_DIGIT_ANY, ast_channel_language(chan), "'vm-received' Q 'digits/at' HM", NULL);
 	} else if (!strncasecmp(ast_channel_language(chan), "pl", 2)) {     /* POLISH syntax */
 		res = ast_say_date_with_format(chan, t, AST_DIGIT_ANY, ast_channel_language(chan), "'vm-received' Q HM", NULL);
-	} else if (!strncasecmp(ast_channel_language(chan), "pt_BR", 5)) {  /* Brazillian PORTUGUESE syntax */
+	} else if (!strncasecmp(ast_channel_language(chan), "pt_BR", 5)) {  /* Brazilian PORTUGUESE syntax */
 		res = ast_say_date_with_format(chan, t, AST_DIGIT_ANY, ast_channel_language(chan), "'vm-received' Ad 'digits/pt-de' B 'digits/pt-de' Y 'digits/pt-as' HM ", NULL);
 	} else if (!strncasecmp(ast_channel_language(chan), "se", 2)) {     /* SWEDISH syntax */
 		res = ast_say_date_with_format(chan, t, AST_DIGIT_ANY, ast_channel_language(chan), "'vm-received' dB 'digits/at' k 'and' M", NULL);
@@ -9413,9 +9327,9 @@ static int vm_intro_gr(struct ast_channel *chan, struct vm_state *vms)
  *     on vm-message.
  *  3) Call ast_say_counted_adjective() to put the proper gender and number
  *     prefix on vm-new and vm-old (none for English).
- *  4) Pass the gender of the language's word for "message" as an agument to
+ *  4) Pass the gender of the language's word for "message" as an argument to
  *     this function which is can in turn pass on to the functions which
- *     say numbers and put endings on nounds and adjectives.
+ *     say numbers and put endings on nouns and adjectives.
  *
  * All languages require these messages:
  *  vm-youhave		"You have..."
@@ -9902,6 +9816,51 @@ static int vm_intro_no(struct ast_channel *chan, struct vm_state *vms)
 	return res;
 }
 
+/* Danish syntax */
+static int vm_intro_da(struct ast_channel *chan, struct vm_state *vms)
+{
+	/* Introduce messages they have */
+	int res;
+
+	res = ast_play_and_wait(chan, "vm-youhave");
+	if (res)
+		return res;
+
+	if (!vms->oldmessages && !vms->newmessages && !vms->urgentmessages) {
+		res = ast_play_and_wait(chan, "vm-no");
+		res = res ? res : ast_play_and_wait(chan, "vm-messages");
+		return res;
+	}
+
+	if (vms->newmessages) {
+		if ((vms->newmessages == 1)) {
+			res = ast_play_and_wait(chan, "digits/1");
+			res = res ? res : ast_play_and_wait(chan, "vm-INBOX");
+			res = res ? res : ast_play_and_wait(chan, "vm-message");
+		} else {
+			res = say_and_wait(chan, vms->newmessages, ast_channel_language(chan));
+			res = res ? res : ast_play_and_wait(chan, "vm-INBOXs");
+			res = res ? res : ast_play_and_wait(chan, "vm-messages");
+		}
+		if (!res && vms->oldmessages)
+			res = ast_play_and_wait(chan, "vm-and");
+	}
+	if (!res && vms->oldmessages) {
+		if (vms->oldmessages == 1) {
+			res = ast_play_and_wait(chan, "digits/1");
+			res = res ? res : ast_play_and_wait(chan, "vm-Old");
+			res = res ? res : ast_play_and_wait(chan, "vm-message");
+		} else {
+			res = say_and_wait(chan, vms->oldmessages, ast_channel_language(chan));
+			res = res ? res : ast_play_and_wait(chan, "vm-Olds");
+			res = res ? res : ast_play_and_wait(chan, "vm-messages");
+		}
+	}
+
+	return res;
+}
+
+
 /* GERMAN syntax */
 static int vm_intro_de(struct ast_channel *chan, struct vm_state *vms)
 {
@@ -10382,6 +10341,8 @@ static int vm_intro(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm
 		return vm_intro_nl(chan, vms);
 	} else if (!strncasecmp(ast_channel_language(chan), "no", 2)) {  /* NORWEGIAN syntax */
 		return vm_intro_no(chan, vms);
+	} else if (!strncasecmp(ast_channel_language(chan), "da", 2)) {  /* DANISH syntax */
+		return vm_intro_da(chan, vms);
 	} else if (!strncasecmp(ast_channel_language(chan), "pl", 2)) {  /* POLISH syntax */
 		return vm_intro_pl(chan, vms);
 	} else if (!strncasecmp(ast_channel_language(chan), "pt_BR", 5)) {  /* BRAZILIAN PORTUGUESE syntax */
@@ -10403,7 +10364,7 @@ static int vm_intro(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm
 	}
 }
 
-static int vm_instructions_en(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms, int skipadvanced, int in_urgent)
+static int vm_instructions_en(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms, int skipadvanced, int in_urgent, int nodelete)
 {
 	int res = 0;
 	/* Play instructions and wait for new command */
@@ -10463,10 +10424,12 @@ static int vm_instructions_en(struct ast_channel *chan, struct ast_vm_user *vmu,
 #ifdef IMAP_STORAGE
 				ast_mutex_unlock(&vms->lock);
 #endif
-				if (!curmsg_deleted) {
-					res = ast_play_and_wait(chan, "vm-delete");
-				} else {
-					res = ast_play_and_wait(chan, "vm-undelete");
+				if (!nodelete) {
+					if (!curmsg_deleted) {
+						res = ast_play_and_wait(chan, "vm-delete");
+					} else {
+						res = ast_play_and_wait(chan, "vm-undelete");
+					}
 				}
 				if (!res) {
 					res = ast_play_and_wait(chan, "vm-toforward");
@@ -10491,7 +10454,7 @@ static int vm_instructions_en(struct ast_channel *chan, struct ast_vm_user *vmu,
 	return res;
 }
 
-static int vm_instructions_ja(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms,  int skipadvanced, int in_urgent)
+static int vm_instructions_ja(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms,  int skipadvanced, int in_urgent, int nodelete)
 {
 	int res = 0;
 	/* Play instructions and wait for new command */
@@ -10587,7 +10550,7 @@ static int vm_instructions_ja(struct ast_channel *chan, struct ast_vm_user *vmu,
 	return res;
 }
 
-static int vm_instructions_zh(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms,  int skipadvanced, int in_urgent)
+static int vm_instructions_zh(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms,  int skipadvanced, int in_urgent, int nodelete)
 {
 	int res = 0;
 	/* Play instructions and wait for new command */
@@ -10605,20 +10568,20 @@ static int vm_instructions_zh(struct ast_channel *chan, struct ast_vm_user *vmu,
 			res = ast_play_and_wait(chan, "vm-opts");
 		if (!res) {
 			vms->starting = 0;
-			return vm_instructions_en(chan, vmu, vms, skipadvanced, in_urgent);
+			return vm_instructions_en(chan, vmu, vms, skipadvanced, in_urgent, nodelete);
 		}
 	}
 	return res;
 }
 
-static int vm_instructions(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms, int skipadvanced, int in_urgent)
+static int vm_instructions(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms, int skipadvanced, int in_urgent, int nodelete)
 {
 	if (!strncasecmp(ast_channel_language(chan), "ja", 2)) { /* Japanese syntax */
-		return vm_instructions_ja(chan, vmu, vms, skipadvanced, in_urgent);
+		return vm_instructions_ja(chan, vmu, vms, skipadvanced, in_urgent, nodelete);
 	} else if (vms->starting && !strncasecmp(ast_channel_language(chan), "zh", 2)) { /* CHINESE (Taiwan) syntax */
-		return vm_instructions_zh(chan, vmu, vms, skipadvanced, in_urgent);
+		return vm_instructions_zh(chan, vmu, vms, skipadvanced, in_urgent, nodelete);
 	} else {					/* Default to ENGLISH */
-		return vm_instructions_en(chan, vmu, vms, skipadvanced, in_urgent);
+		return vm_instructions_en(chan, vmu, vms, skipadvanced, in_urgent, nodelete);
 	}
 }
 
@@ -11207,7 +11170,7 @@ static int vm_authenticate(struct ast_channel *chan, char *mailbox, int mailbox_
 			int skipuser, int max_logins, int silent)
 {
 	int useadsi = 0, valid = 0, logretries = 0;
-	char password[AST_MAX_EXTENSION], *passptr;
+	char password[AST_MAX_EXTENSION], *passptr = NULL;
 	struct ast_vm_user vmus, *vmu = NULL;
 
 	/* If ADSI is supported, setup login screen */
@@ -11264,12 +11227,14 @@ static int vm_authenticate(struct ast_channel *chan, char *mailbox, int mailbox_
 			password[0] = '\0';
 		} else {
 			if (ast_streamfile(chan, vm_password, ast_channel_language(chan))) {
-				ast_log(AST_LOG_WARNING, "Unable to stream password file\n");
+				if (!ast_check_hangup(chan)) {
+					ast_log(AST_LOG_WARNING, "Unable to stream password file\n");
+				}
 				free_user(vmu);
 				return -1;
 			}
 			if (ast_readstring(chan, password, sizeof(password) - 1, 2000, 10000, "#") < 0) {
-				ast_log(AST_LOG_WARNING, "Unable to read password\n");
+				ast_log(AST_LOG_NOTICE, "Unable to read password\n");
 				free_user(vmu);
 				return -1;
 			} else if (password[0] == '*') {
@@ -11501,6 +11466,7 @@ static int vm_execmain(struct ast_channel *chan, const char *data)
 	int play_auto = 0;
 	int play_folder = 0;
 	int in_urgent = 0;
+	int nodelete = 0;
 #ifdef IMAP_STORAGE
 	int deleted = 0;
 #endif
@@ -11563,6 +11529,9 @@ static int vm_execmain(struct ast_channel *chan, const char *data)
 					play_folder = 0;
 				}
 			}
+			if (ast_test_flag(&flags, OPT_READONLY)) {
+				nodelete = 1;
+			}
 		} else {
 			/* old style options parsing */
 			while (*(args.argv0)) {
@@ -11587,10 +11556,16 @@ static int vm_execmain(struct ast_channel *chan, const char *data)
 		else
 			ast_copy_string(vms.username, args.argv0, sizeof(vms.username));
 
-		if (!ast_strlen_zero(vms.username) && (vmu = find_user(&vmus, context ,vms.username)))
-			skipuser++;
-		else
+		if (!ast_strlen_zero(vms.username)) {
+			if ((vmu = find_user(&vmus, context ,vms.username))) {
+				skipuser++;
+			} else {
+				ast_log(LOG_WARNING, "Mailbox '%s%s%s' doesn't exist\n", vms.username, context ? "@": "", context ? context : "");
+				valid = 0;
+			}
+		} else {
 			valid = 0;
+		}
 	}
 
 	if (!valid)
@@ -11970,7 +11945,7 @@ static int vm_execmain(struct ast_channel *chan, const char *data)
 			}
 			break;
 		case '7': /* Delete the current message */
-			if (vms.curmsg >= 0 && vms.curmsg <= vms.lastmsg) {
+			if (!nodelete && vms.curmsg >= 0 && vms.curmsg <= vms.lastmsg) {
 				vms.deleted[vms.curmsg] = !vms.deleted[vms.curmsg];
 				if (useadsi)
 					adsi_delete(chan, &vms);
@@ -12159,7 +12134,7 @@ static int vm_execmain(struct ast_channel *chan, const char *data)
 					if (!cmd)
 						cmd = ast_play_and_wait(chan, "vm-opts");
 					if (!cmd)
-						cmd = vm_instructions(chan, vmu, &vms, 1, in_urgent);
+						cmd = vm_instructions(chan, vmu, &vms, 1, in_urgent, nodelete);
 					break;
 				}
 				cmd = ast_play_and_wait(chan, "vm-onefor");
@@ -12171,7 +12146,7 @@ static int vm_execmain(struct ast_channel *chan, const char *data)
 				if (!cmd)
 					cmd = ast_play_and_wait(chan, "vm-opts");
 				if (!cmd)
-					cmd = vm_instructions(chan, vmu, &vms, 1, in_urgent);
+					cmd = vm_instructions(chan, vmu, &vms, 1, in_urgent, nodelete);
 			} else
 				cmd = 0;
 			break;
@@ -12188,7 +12163,7 @@ static int vm_execmain(struct ast_channel *chan, const char *data)
  			break;
 		default:	/* Nothing */
 			ast_test_suite_event_notify("PLAYBACK", "Message: instructions");
-			cmd = vm_instructions(chan, vmu, &vms, 0, in_urgent);
+			cmd = vm_instructions(chan, vmu, &vms, 0, in_urgent, nodelete);
 			break;
 		}
 	}
@@ -12268,16 +12243,13 @@ static int vm_exec(struct ast_channel *chan, const char *data)
 
 	memset(&leave_options, 0, sizeof(leave_options));
 
-	if (ast_channel_state(chan) != AST_STATE_UP)
-		ast_answer(chan);
-
 	if (!ast_strlen_zero(data)) {
 		tmp = ast_strdupa(data);
 		AST_STANDARD_APP_ARGS(args, tmp);
 		if (args.argc == 2) {
 			if (ast_app_parse_options(vm_app_options, &flags, opts, args.argv1))
 				return -1;
-			ast_copy_flags(&leave_options, &flags, OPT_SILENT | OPT_BUSY_GREETING | OPT_UNAVAIL_GREETING | OPT_MESSAGE_Urgent | OPT_MESSAGE_PRIORITY | OPT_DTMFEXIT);
+			ast_copy_flags(&leave_options, &flags, OPT_SILENT | OPT_SILENT_IF_GREET | OPT_BUSY_GREETING | OPT_UNAVAIL_GREETING | OPT_MESSAGE_Urgent | OPT_MESSAGE_PRIORITY | OPT_DTMFEXIT);
 			if (ast_test_flag(&flags, OPT_RECORDGAIN)) {
 				int gain;
 
@@ -12293,6 +12265,11 @@ static int vm_exec(struct ast_channel *chan, const char *data)
 					leave_options.exitcontext = opts[OPT_ARG_DTMFEXIT];
 			}
 		}
+		if (ast_test_flag(&flags, OPT_BEEP)) { /* Use custom beep (or none at all) */
+			leave_options.beeptone = opts[OPT_ARG_BEEP_TONE];
+		} else { /* Use default beep */
+			leave_options.beeptone = "beep";
+		}
 	} else {
 		char temp[256];
 		res = ast_app_getdata(chan, "vm-whichbox", temp, sizeof(temp) - 1, 0);
@@ -12303,6 +12280,14 @@ static int vm_exec(struct ast_channel *chan, const char *data)
 		args.argv0 = ast_strdupa(temp);
 	}
 
+	if (ast_channel_state(chan) != AST_STATE_UP) {
+		if (ast_test_flag(&flags, OPT_EARLYM_GREETING)) {
+			ast_indicate(chan, AST_CONTROL_PROGRESS);
+		} else {
+			ast_answer(chan);
+		}
+	}
+
 	res = leave_voicemail(chan, args.argv0, &leave_options);
 	if (res == 't') {
 		ast_play_and_wait(chan, "vm-goodbye");
@@ -12496,7 +12481,7 @@ AST_TEST_DEFINE(test_voicemail_vmuser)
 		res = 1;
 	}
 	if (strcasecmp(vmu->attachfmt, "wav49")) {
-		ast_test_status_update(test, "Parse failure for attachftm option\n");
+		ast_test_status_update(test, "Parse failure for attachfmt option\n");
 		res = 1;
 	}
 	if (strcasecmp(vmu->fromstring, "Voicemail System")) {
@@ -14168,7 +14153,7 @@ static int actual_load_config(int reload, struct ast_config *cfg, struct ast_con
 
 		if (!(val = ast_variable_retrieve(cfg, "general", "usedirectory")))
 			val = "no";
-		ast_set2_flag((&globalflags), ast_true(val), VM_DIRECFORWARD);
+		ast_set2_flag((&globalflags), ast_true(val), VM_DIRECTFORWARD);
 
 		if (!(val = ast_variable_retrieve(cfg, "general", "passwordlocation"))) {
 			val = "voicemail.conf";
@@ -15437,6 +15422,7 @@ static int advanced_options(struct ast_channel *chan, struct ast_vm_user *vmu, s
 
 				memset(&leave_options, 0, sizeof(leave_options));
 				leave_options.record_gain = record_gain;
+				leave_options.beeptone = "beep";
 				res = leave_voicemail(chan, mailbox, &leave_options);
 				if (!res)
 					res = 't';
@@ -15545,6 +15531,12 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
 				}
 				cmd = ast_play_and_wait(chan, "beep");
 			}
+			if (cmd == -1) {
+				/* User has hung up, no options to give */
+				ast_debug(1, "User hung up before message could be rerecorded\n");
+				ast_filedelete(tempfile, NULL);
+				return cmd;
+			}
 			recorded = 1;
 			/* After an attempt has been made to record message, we have to take care of INTRO and beep for incoming messages, but not for greetings */
 			if (record_gain)
@@ -15663,7 +15655,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
 			}
 			return cmd;
 		default:
-			/* If the caller is an ouside caller and the review option is enabled or it's forward intro
+			/* If the caller is an outside caller and the review option is enabled or it's forward intro
 			   allow them to review the message, but let the owner of the box review
 			   their OGM's */
 			if (outsidecaller && !ast_test_flag(vmu, VM_REVIEW) && !forwardintro)
@@ -16039,7 +16031,7 @@ static struct ast_vm_mailbox_snapshot *vm_mailbox_snapshot_destroy(struct ast_vm
  * \param vms The voicemail state corresponding to an open mailbox
  * \param msg_ids An array of message identifiers
  * \param num_msgs The number of identifiers in msg_ids
- * \param msg_nums [out] The message indexes corresponding to the given
+ * \param[out] msg_nums The message indexes corresponding to the given
  * \param vmu
  * message IDs
  * \pre vms must have open_mailbox() called on it prior to this function.

apps/app_waitforcond.c

@@ -0,0 +1,239 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * 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 Sleep until a condition is true
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+
+/*** DOCUMENTATION
+	<application name="WaitForCondition" language="en_US">
+		<since>
+			<version>16.20.0</version>
+			<version>18.6.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Wait (sleep) until the given condition is true.
+		</synopsis>
+		<syntax>
+			<parameter name="replacementchar" required="true">
+				<para>Specifies the character in the expression used to replace the <literal>$</literal>
+				character. This character should not be used anywhere in the expression itself.</para>
+			</parameter>
+			<parameter name="expression" required="true">
+				<para>A modified logical expression with the <literal>$</literal> characters replaced by
+				<replaceable>replacementchar</replaceable>. This is necessary to pass the expression itself
+				into the application, rather than its initial evaluation.</para>
+			</parameter>
+			<parameter name="timeout">
+				<para>The maximum amount of time, in seconds, this application should wait for a condition
+				to become true before dialplan execution continues automatically to the next priority.
+				By default, there is no timeout.</para>
+			</parameter>
+			<parameter name="interval">
+				<para>The frequency, in seconds, of polling the condition, which can be adjusted depending
+				on how time-sensitive execution needs to be. By default, this is 0.05.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Waits until <replaceable>expression</replaceable> evaluates to true, checking every
+			<replaceable>interval</replaceable> seconds for up to <replaceable>timeout</replaceable>. Default
+			is evaluate <replaceable>expression</replaceable> every 50 milliseconds with no timeout.</para>
+			<example title="Wait for condition dialplan variable/function to become 1 for up to 40 seconds, checking every 500ms">
+			 same => n,WaitForCondition(#,#["#{condition}"="1"],40,0.5)
+			</example>
+			<para>Sets <variable>WAITFORCONDITIONSTATUS</variable> to one of the following values:</para>
+			<variablelist>
+				<variable name="WAITFORCONDITIONSTATUS">
+					<value name="TRUE">
+						Condition evaluated to true before timeout expired.
+					</value>
+					<value name="FAILURE">
+						Invalid argument.
+					</value>
+					<value name="TIMEOUT">
+						Timeout elapsed without condition evaluating to true.
+					</value>
+					<value name="HANGUP">
+						Channel hung up before condition became true.
+					</value>
+				</variable>
+			</variablelist>
+		</description>
+	</application>
+ ***/
+
+static char *app = "WaitForCondition";
+
+static int waitforcond_exec(struct ast_channel *chan, const char *data)
+{
+	int ms, i;
+	double timeout = 0, poll = 0;
+	int timeout_ms = 0;
+	int poll_ms = 50; /* default is evaluate the condition every 50ms */
+	struct timeval start = ast_tvnow();
+	char dollarsignrep;
+	int brackets = 0;
+	char *pos, *open_bracket, *expression, *optargs = NULL;
+	char condition[512];
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(timeout);
+		AST_APP_ARG(interval);
+	);
+
+	pos = ast_strdupa(data);
+
+	if (ast_strlen_zero(pos)) {
+		ast_log(LOG_ERROR, "WaitForCondition requires a condition\n");
+		pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "FAILURE");
+		return 0;
+	}
+
+	/* is there at least a [ followed by a ] somewhere ? */
+	if (!(open_bracket = strchr(pos, '[')) || !strchr(open_bracket, ']')) {
+		ast_log(LOG_ERROR, "No expression detected. Did you forget to replace the $ signs?\n");
+		pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "FAILURE");
+		return 0;
+	}
+
+	dollarsignrep = pos[0];
+	if (dollarsignrep == '$' || dollarsignrep == '[' || dollarsignrep == ']'
+		|| dollarsignrep == '{' || dollarsignrep == '}') {
+		ast_log(LOG_ERROR, "Dollar sign replacement cannot be %c.\n", dollarsignrep);
+		pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "FAILURE");
+		return 0;
+	}
+	++pos;
+	if (pos[0] != ',') {
+		ast_log(LOG_ERROR, "Invalid separator: %c\n", pos[0]);
+		pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "FAILURE");
+		return 0;
+	}
+	++pos;
+	if (pos[0] != dollarsignrep) {
+		ast_log(LOG_ERROR, "Expression start does not match provided replacement: %c\n", pos[0]);
+		pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "FAILURE");
+		return 0;
+	}
+
+	expression = pos; /* we're at the start of the expression */
+
+	/* commas may appear within the expression, so go until we've encountered as many closing brackets as opening */
+	while (++pos) {
+		if (pos[0] == '\0') {
+			ast_log(LOG_ERROR, "Could not parse end of expression.\n");
+			pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "FAILURE");
+			return 0;
+		}
+		if (pos[0] == '[') {
+			brackets++;
+		} else if (pos[0] == ']') {
+			brackets--;
+		}
+		if (brackets == 0) { /* reached end of expression */
+			break;
+		}
+	}
+	++pos;
+	if (pos[0] != '\0') {
+		++pos; /* eat comma separator */
+		if (pos[0] != '\0') {
+			optargs = ast_strdupa(pos);
+			AST_STANDARD_APP_ARGS(args, optargs);
+			if (!ast_strlen_zero(args.timeout)) {
+				if (sscanf(args.timeout, "%30lg", &timeout) != 1) {
+					ast_log(LOG_WARNING, "Invalid timeout provided: %s. No timeout set.\n", args.timeout);
+					return -1;
+				}
+				timeout_ms = timeout * 1000.0;
+			}
+
+			if (!ast_strlen_zero(args.interval)) {
+				if (sscanf(args.interval, "%30lg", &poll) != 1) {
+					ast_log(LOG_WARNING, "Invalid polling interval provided: %s. Default unchanged.\n", args.interval);
+					return -1;
+				}
+				if (poll < 0.001) {
+					ast_log(LOG_WARNING, "Polling interval cannot be less than 1ms. Default unchanged.\n");
+					return -1;
+				}
+				poll_ms = poll * 1000.0;
+			}
+		}
+	}
+
+	for (i = 0; expression[i] != '\0'; i++) {
+		if (expression[i] == dollarsignrep) {
+			expression[i] = '$'; /* replace $s back into expression for variable parsing */
+		}
+	}
+
+	if (timeout_ms > 0) {
+		ast_debug(1, "Waiting for condition for %f seconds: %s (checking every %d ms)", timeout, expression, poll_ms);
+	} else {
+		ast_debug(1, "Waiting for condition, forever: %s (checking every %d ms)", expression, poll_ms);
+	}
+
+	while (1) {
+		/* Substitute variables now */
+		pbx_substitute_variables_helper(chan, expression, condition, sizeof(condition) - 1);
+		if (pbx_checkcondition(condition)) {
+			pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "TRUE");
+			return 0;
+		}
+		/* If a timeout was specified, check that it hasn't expired */
+		if ((timeout_ms > 0) && !(ms = ast_remaining_ms(start, timeout_ms))) {
+			pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "TIMEOUT");
+			return 0;
+		}
+		if (ast_safe_sleep(chan, poll_ms)) { /* don't waste CPU, we don't need a super tight loop */
+			pbx_builtin_setvar_helper(chan, "WAITFORCONDITIONSTATUS", "HANGUP");
+			return -1; /* channel hung up */
+		}
+	}
+}
+
+static int unload_module(void)
+{
+	return ast_unregister_application(app);
+}
+
+static int load_module(void)
+{
+	return ast_register_application_xml(app, waitforcond_exec);
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Wait until condition is true");

apps/app_waitforsilence.c

@@ -23,9 +23,6 @@
  *
  * \brief Wait for Silence
  *   - Waits for up to 'x' milliseconds of silence, 'y' times \n
- *   - WaitForSilence(500,2) will wait for 1/2 second of silence, twice \n
- *   - WaitForSilence(1000,1) will wait for 1 second of silence, once \n
- *   - WaitForSilence(300,3,10) will wait for 300ms of silence, 3 times, and return after 10sec \n
  *
  * \author David C. Troy <dave@popvox.com>
  *
@@ -78,11 +75,15 @@
 			playing a message.</para>
 			<para>Typically you will want to include two or more calls to WaitForSilence when dealing with an answering
 			machine; first waiting for the spiel to finish, then waiting for the beep, etc.</para>
-			<para>Examples:</para>
-			<para>WaitForSilence(500,2) will wait for 1/2 second of silence, twice</para>
-			<para>WaitForSilence(1000) will wait for 1 second of silence, once</para>
-			<para>WaitForSilence(300,3,10) will wait for 300ms silence, 3 times, and returns after 10 sec, even if silence
-			is not detected</para>
+			<example title="Wait for half a second of silence, twice">
+			same => n,WaitForSilence(500,2)
+			</example>
+			<example title="Wait for one second of silence, once">
+			same => n,WaitForSilence(1000)
+			</example>
+			<example title="Wait for 300 ms of silence, 3 times, and returns after 10 seconds, even if no silence detected">
+			same => n,WaitForSilence(300,3,10)
+			</example>
 			<para>Sets the channel variable <variable>WAITSTATUS</variable> to one of these values:</para>
 			<variablelist>
 				<variable name="WAITSTATUS">

apps/confbridge/conf_config_parser.c

@@ -97,6 +97,9 @@
 				<configOption name="quiet">
 					<synopsis>Silence enter/leave prompts and user intros for this user</synopsis>
 				</configOption>
+				<configOption name="hear_own_join_sound">
+					<synopsis>Determines if the user also hears the join sound when they enter a conference</synopsis>
+				</configOption>
 				<configOption name="announce_user_count">
 					<synopsis>Sets if the number of users should be announced to the user</synopsis>
 				</configOption>
@@ -117,6 +120,9 @@
 				<configOption name="end_marked">
 					<synopsis>Kick the user from the conference when the last marked user leaves</synopsis>
 				</configOption>
+				<configOption name="end_marked_any">
+					<synopsis>Kick the user from the conference when any marked user leaves</synopsis>
+				</configOption>
 				<configOption name="talk_detection_events">
 					<synopsis>Set whether or not notifications of when a user begins and ends talking should be sent out as events over AMI</synopsis>
 				</configOption>
@@ -247,6 +253,9 @@
 					participants in the conference bridge. If disabled then no text
 					messages are sent to the user.</para></description>
 				</configOption>
+				<configOption name="answer_channel" default="yes">
+					<synopsis>Sets if a user's channel should be answered if currently unanswered.</synopsis>
+				</configOption>
 			</configObject>
 			<configObject name="bridge_profile">
 				<synopsis>A named profile to apply to specific bridges.</synopsis>
@@ -381,7 +390,7 @@
 						specific conference bridge.
 						You should be aware that there are potential races between testing for the
 						existence of a bridge, and taking action upon that information, consider
-						for example two callers executing the check simultaniously, and then taking
+						for example two callers executing the check simultaneously, and then taking
 						special action as "first caller" into the bridge.  The same for exiting,
 						directly after the check the bridge can be destroyed before the new caller
 						enters (creating a new bridge), for example, and the "first member" actions
@@ -456,7 +465,7 @@
 							<enum name="sound_kicked"><para>The sound played to a user who has been kicked from the conference.</para></enum>
 							<enum name="sound_muted"><para>The sound played when the mute option it toggled on.</para></enum>
 							<enum name="sound_unmuted"><para>The sound played when the mute option it toggled off.</para></enum>
-							<enum name="sound_binaural_on"><para>The sound played when binaural auudio is turned on.</para></enum>
+							<enum name="sound_binaural_on"><para>The sound played when binaural audio is turned on.</para></enum>
 							<enum name="sound_binaural_off"><para>The sound played when the binaural audio is turned off.</para></enum>
 							<enum name="sound_only_person"><para>The sound played when the user is the only person in the conference.</para></enum>
 							<enum name="sound_only_one"><para>The sound played to a user when there is only one other
@@ -533,8 +542,22 @@
 								<para>The highest estimated maximum bitrate of all receivers in the bridge
 								is taken and sent to each sender.</para>
 							</enum>
+							<enum name="force">
+								<para>The bitrate configured in <literal>remb_estimated_bitrate</literal>
+								is sent to each sender.</para>
+							</enum>
 						</enumlist>
 					</description>
+					<see-also><ref type="configOption">remb_estimated_bitrate</ref></see-also>
+				</configOption>
+				<configOption name="remb_estimated_bitrate">
+					<synopsis>Sets the estimated bitrate sent to each participant in REMB reports</synopsis>
+					<description><para>
+					    When <literal>remb_behavior</literal> is set to <literal>force</literal>,
+					    this options sets the estimated bitrate (in bits per second) sent to each participant
+					    in REMB reports.
+					</para></description>
+					<see-also><ref type="configOption">remb_behavior</ref></see-also>
 				</configOption>
 				<configOption name="enable_events" default="no">
 					<synopsis>Enables events for this bridge</synopsis>
@@ -1336,7 +1359,7 @@ static int add_menu_entry(struct conf_menu *menu, const char *dtmf, const char *
 		comma = strchr(tmp_action_names, ',');
 
 		/* If the next action has brackets with comma delimited arguments in it,
-		 * make the delimeter ')' instead of a comma to preserve the argments */
+		 * make the delimeter ')' instead of a comma to preserve the arguments */
 		if (startbrace && endbrace && comma && (comma > startbrace && comma < endbrace)) {
 			delimiter = ")";
 		} else {
@@ -1420,10 +1443,7 @@ static int add_menu_entry(struct conf_menu *menu, const char *dtmf, const char *
 
 	/* if adding any of the actions failed, bail */
 	if (res) {
-		struct conf_menu_action *menu_action;
-		while ((menu_action = AST_LIST_REMOVE_HEAD(&menu_entry->actions, action))) {
-			ast_free(menu_action);
-		}
+		conf_menu_entry_destroy(menu_entry);
 		ast_free(menu_entry);
 		return -1;
 	}
@@ -1432,6 +1452,7 @@ static int add_menu_entry(struct conf_menu *menu, const char *dtmf, const char *
 	AST_LIST_TRAVERSE_SAFE_BEGIN(&menu->entries, cur, entry) {
 		if (!strcasecmp(cur->dtmf, menu_entry->dtmf)) {
 			AST_LIST_REMOVE_CURRENT(entry);
+			conf_menu_entry_destroy(cur);
 			ast_free(cur);
 			break;
 		}
@@ -1557,12 +1578,18 @@ static char *handle_cli_confbridge_show_user_profile(struct ast_cli_entry *e, in
 	ast_cli(a->fd,"Quiet:                   %s\n",
 		u_profile.flags & USER_OPT_QUIET ?
 		"enabled" : "disabled");
+	ast_cli(a->fd,"Hear Join:               %s\n",
+		u_profile.flags & USER_OPT_HEAR_OWN_JOIN_SOUND ?
+		"enabled" : "disabled");
 	ast_cli(a->fd,"Wait Marked:             %s\n",
 		u_profile.flags & USER_OPT_WAITMARKED ?
 		"enabled" : "disabled");
-	ast_cli(a->fd,"END Marked:              %s\n",
+	ast_cli(a->fd,"END Marked (All):        %s\n",
 		u_profile.flags & USER_OPT_ENDMARKED ?
 		"enabled" : "disabled");
+	ast_cli(a->fd,"END Marked (Any):        %s\n",
+		u_profile.flags & USER_OPT_ENDMARKEDANY ?
+		"enabled" : "disabled");
 	ast_cli(a->fd,"Drop_silence:            %s\n",
 		u_profile.flags & USER_OPT_DROP_SILENCE ?
 		"enabled" : "disabled");
@@ -1595,9 +1622,12 @@ static char *handle_cli_confbridge_show_user_profile(struct ast_cli_entry *e, in
 	ast_cli(a->fd,"Announce User Count all: %s\n",
 		u_profile.flags & USER_OPT_ANNOUNCEUSERCOUNTALL ?
 		"enabled" : "disabled");
-        ast_cli(a->fd,"Text Messaging:          %s\n",
-                u_profile.flags & USER_OPT_TEXT_MESSAGING ?
-                "enabled" : "disabled");
+	ast_cli(a->fd,"Text Messaging:          %s\n",
+			u_profile.flags & USER_OPT_TEXT_MESSAGING ?
+			"enabled" : "disabled");
+	ast_cli(a->fd,"Answer Channel:          %s\n",
+			u_profile.flags & USER_OPT_ANSWER_CHANNEL ?
+			"true" : "false");
 	ast_cli(a->fd, "\n");
 
 	return CLI_SUCCESS;
@@ -2159,7 +2189,9 @@ static int remb_behavior_handler(const struct aco_option *opt, struct ast_variab
 		BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST |
 		BRIDGE_OPT_REMB_BEHAVIOR_AVERAGE_ALL |
 		BRIDGE_OPT_REMB_BEHAVIOR_LOWEST_ALL |
-		BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST_ALL);
+		BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST_ALL |
+		BRIDGE_OPT_REMB_BEHAVIOR_FORCE
+		);
 
 	if (!strcasecmp(var->value, "average")) {
 		ast_set_flag(b_profile, BRIDGE_OPT_REMB_BEHAVIOR_AVERAGE);
@@ -2173,6 +2205,8 @@ static int remb_behavior_handler(const struct aco_option *opt, struct ast_variab
 		ast_set_flag(b_profile, BRIDGE_OPT_REMB_BEHAVIOR_LOWEST_ALL);
 	} else if (!strcasecmp(var->value, "highest_all")) {
 		ast_set_flag(b_profile, BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST_ALL);
+	} else if (!strcasecmp(var->value, "force")) {
+		ast_set_flag(b_profile, BRIDGE_OPT_REMB_BEHAVIOR_FORCE);
 	} else {
 		return -1;
 	}
@@ -2372,12 +2406,14 @@ int conf_load_config(void)
 	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, "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(&cfg_info, "hear_own_join_sound", ACO_EXACT, user_types, "yes", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_HEAR_OWN_JOIN_SOUND);
 	aco_option_register_custom(&cfg_info, "announce_user_count_all", ACO_EXACT, user_types, "no", announce_user_count_all_handler, 0);
 	aco_option_register(&cfg_info, "announce_user_count", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_ANNOUNCEUSERCOUNT);
 	/* Negative logic. Defaults to "yes" and evaluates with ast_false(). If !ast_false(), USER_OPT_NOONLYPERSON is cleared */
 	aco_option_register(&cfg_info, "announce_only_user", ACO_EXACT, user_types, "yes", OPT_BOOLFLAG_T, 0, FLDSET(struct user_profile, flags), USER_OPT_NOONLYPERSON);
 	aco_option_register(&cfg_info, "wait_marked", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_WAITMARKED);
 	aco_option_register(&cfg_info, "end_marked", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_ENDMARKED);
+	aco_option_register(&cfg_info, "end_marked_any", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_ENDMARKEDANY);
 	aco_option_register(&cfg_info, "talk_detection_events", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_TALKER_DETECT);
 	aco_option_register(&cfg_info, "dtmf_passthrough", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_DTMF_PASS);
 	aco_option_register(&cfg_info, "announce_join_leave", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_ANNOUNCE_JOIN_LEAVE);
@@ -2392,6 +2428,7 @@ int conf_load_config(void)
 	aco_option_register(&cfg_info, "jitterbuffer", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_JITTERBUFFER);
 	aco_option_register(&cfg_info, "timeout", ACO_EXACT, user_types, "0", OPT_UINT_T, 0, FLDSET(struct user_profile, timeout));
 	aco_option_register(&cfg_info, "text_messaging", ACO_EXACT, user_types, "yes", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_TEXT_MESSAGING);
+	aco_option_register(&cfg_info, "answer_channel", ACO_EXACT, user_types, "yes", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_ANSWER_CHANNEL);
 
 	/* This option should only be used with the CONFBRIDGE dialplan function */
 	aco_option_register_custom(&cfg_info, "template", ACO_EXACT, user_types, NULL, user_template_handler, 0);
@@ -2419,6 +2456,7 @@ int conf_load_config(void)
 	aco_option_register(&cfg_info, "video_update_discard", ACO_EXACT, bridge_types, "2000", OPT_UINT_T, 0, FLDSET(struct bridge_profile, video_update_discard));
 	aco_option_register(&cfg_info, "remb_send_interval", ACO_EXACT, bridge_types, "0", OPT_UINT_T, 0, FLDSET(struct bridge_profile, remb_send_interval));
 	aco_option_register_custom(&cfg_info, "remb_behavior", ACO_EXACT, bridge_types, "average", remb_behavior_handler, 0);
+	aco_option_register(&cfg_info, "remb_estimated_bitrate", ACO_EXACT, bridge_types, "0", OPT_UINT_T, 0, FLDSET(struct bridge_profile, remb_estimated_bitrate));
 	aco_option_register(&cfg_info, "enable_events", ACO_EXACT, bridge_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct bridge_profile, flags), BRIDGE_OPT_ENABLE_EVENTS);
 	/* This option should only be used with the CONFBRIDGE dialplan function */
 	aco_option_register_custom(&cfg_info, "template", ACO_EXACT, bridge_types, NULL, bridge_template_handler, 0);