.gitreview

 [gerrit]
-defaultbranch=13
+defaultbranch=16
 #
 # Intentional padding to ensure it is possible to point a commit
 # to an alternative gerrit server/repository without breaking

.version

-13.23.1
\ No newline at end of file
+16.1.1
\ No newline at end of file

CREDITS

@@ -154,6 +154,7 @@
 		(dialog-info), QUEUE_EXISTS function, device state provider
 		architecture, multiparking (together with mvanbaak), meetme and
 		parking device states, MiniVM - the small voicemail system,
+		RTP improvements, RTCP enhancements, DTMF timing fixes,
 		many documentation updates/corrections, and many bug fixes.
 		oej(AT)edvina.net, http://edvina.net
 

Makefile

@@ -76,9 +76,6 @@ export AST_DEVMODE_STRICT # Enables shadow warnings (-Wshadow)
 export _SOLINK            # linker flags for all shared objects
 export SOLINK             # linker flags for loadable modules
 export DYLINK             # linker flags for shared libraries
-export STATIC_BUILD       # Additional cflags, set to -static
-                          # for static builds. Probably
-                          # should go directly to ASTLDFLAGS
 
 #--- paths to various commands
 # The makeopts include below tries to set these if they're found during
@@ -167,20 +164,19 @@ LINKER_SYMBOL_PREFIX=
 # Default install directory for DAHDI hooks.
 DAHDI_UDEV_HOOK_DIR = /usr/share/dahdi/span_config.d
 
-# This Makefile previously contained a note about the ability to use .asterisk.makeopts
-# from your home directory or /etc/asterisk.makeopts to set defaults for menuselect.
-# These files have never worked in this branch of Asterisk.  The work around is to
-# manually copy the file containing defaults before running 'make menuselect':
-#
-# cp ${HOME}/.asterisk.makeopts menuselect.makeopts
-#   or
-# cp /etc/asterisk.makeopts menuselect.makeopts
-#
-# As an alternative, menuselect/menuselect can be used by a script to enable or disable
-# individual options or entire categories.  To use this feature you must first
-# compile menuselect using 'make menuselect.makeopts'.  For information about parameters
-# supported run:
-# menuselect/menuselect --help
+# If the file .asterisk.makeopts is present in your home directory, you can
+# include all of your favorite menuselect options so that every time you download
+# a new version of Asterisk, you don't have to run menuselect to set them.
+# The file /etc/asterisk.makeopts will also be included but can be overridden
+# by the file in your home directory.
+
+ifeq ($(wildcard menuselect.makeopts),)
+	USER_MAKEOPTS=$(wildcard ~/.asterisk.makeopts)
+	GLOBAL_MAKEOPTS=$(wildcard /etc/asterisk.makeopts)
+else
+	USER_MAKEOPTS=
+	GLOBAL_MAKEOPTS=
+endif
 
 
 MOD_SUBDIR_CFLAGS="-I$(ASTTOPDIR)/include"
@@ -358,10 +354,10 @@ makeopts: configure
 	@echo "****"
 	@exit 1
 
-menuselect.makeopts: menuselect/menuselect menuselect-tree makeopts build_tools/menuselect-deps
-ifeq ($(filter %menuselect,$(MAKECMDGOALS)),)
-	menuselect/menuselect --check-deps $@
+menuselect.makeopts: menuselect/menuselect menuselect-tree makeopts build_tools/menuselect-deps $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS)
+ifeq ($(filter %.menuselect,$(MAKECMDGOALS)),)
 	menuselect/menuselect --check-deps $@
+	menuselect/menuselect --check-deps $@ $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS)
 endif
 
 $(MOD_SUBDIRS_MENUSELECT_TREE):
@@ -950,7 +946,7 @@ config:
 		if [ -z "$(DESTDIR)" ] ; then \
 			/sbin/chkconfig --add asterisk ; \
 		fi ; \
-	elif [ -f /etc/os-release ] && [ "opensuse" = "$(shell . /etc/os-release && echo $$ID)" ] ; then \
+	elif [ -f /etc/os-release ] && [ "opensuse" = "$(shell . /etc/os-release 2>/dev/null && echo $$ID)" ] ; then \
 		./build_tools/install_subst contrib/init.d/rc.suse.asterisk  "$(DESTDIR)/etc/init.d/asterisk"; \
 		if [ ! -f /etc/sysconfig/asterisk ] ; then \
 			$(INSTALL) -m 644 contrib/init.d/etc_default_asterisk "$(DESTDIR)/etc/sysconfig/asterisk" ; \
@@ -962,7 +958,7 @@ config:
 		./build_tools/install_subst contrib/init.d/rc.archlinux.asterisk  "$(DESTDIR)/etc/init.d/asterisk"; \
 	elif [ -f /etc/slackware-version ]; then \
 		./build_tools/install_subst contrib/init.d/rc.slackware.asterisk  "$(DESTDIR)/etc/rc.d/rc.asterisk"; \
-	elif [ -f /etc/os-release ] && [ "slackware" = "$(shell . /etc/os-release && echo $$ID)" ] ; then \
+	elif [ -f /etc/os-release ] && [ "slackware" = "$(shell . /etc/os-release 2>/dev/null && echo $$ID)" ] ; then \
 		./build_tools/install_subst contrib/init.d/rc.slackware.asterisk  "$(DESTDIR)/etc/rc.d/rc.asterisk"; \
 	elif [ -d "$(DESTDIR)/Library/LaunchDaemons" ]; then \
 		if [ ! -f "$(DESTDIR)/Library/LaunchDaemons/org.asterisk.asterisk.plist" ]; then \

Makefile.moddir_rules

@@ -18,10 +18,6 @@
 # Also note that we can only set one variable per rule, so we have to
 # repeat the left hand side to set multiple variables.
 
-ifneq ($(findstring STATIC_BUILD,$(MENUSELECT_CFLAGS)),)
-  STATIC_BUILD=-static
-endif
-
 include $(ASTTOPDIR)/Makefile.rules
 
 # If MODULE_PREFIX is defined, use it to run the standard functions to set
@@ -52,7 +48,29 @@ LOADABLE_MODS:=$(C_MODS) $(CC_MODS)
 # is used to collect the required flags for a module... which can
 # then be used any place they are required.
 
-MOD_ASTCFLAGS=-DAST_MODULE=\"$(1)\" $(MENUSELECT_OPTS_$(1):%=-D%) $(foreach dep,$(MENUSELECT_DEPENDS_$(1)),$(value $(dep)_INCLUDE))
+MOD_ASTCFLAGS=\
+	-DAST_MODULE=\"$(1)\" \
+	-DAST_MODULE_SELF_SYM=__internal_$(1)_self \
+	$(MENUSELECT_OPTS_$(1):%=-D%) \
+	$(foreach dep,$(MENUSELECT_DEPENDS_$(1)),$(value $(dep)_INCLUDE))
+
+define MOD_ADD_SOURCE
+$(1).so: $$(subst $(3),$(5),$(2))
+$$(subst $(3),$(5),$(2)): _ASTCFLAGS+=$$(call MOD_ASTCFLAGS,$(1))
+.$(1).moduleinfo: MODULEINFO_EXTRA_OUTPUT=" $$(addprefix $$(SUBDIR)/,$$(subst $(3),$(5),$(2)) $$(subst $(3),$(4),$(2)))"
+# The use of wildcard ensures that 'make menuselect' will not fail for modules that
+# require additional source downloads.
+.$(1).moduleinfo: $(wildcard $(2))
+
+clean:: clean-$(1)$(3)
+
+clean-$(1)$(3):
+	rm -f $$(subst $(3),$(4),$(2)) $$(subst $(3),$(5),$(2)) $$(subst $(3),$(6),$(2)) $$(subst $(3),$(7),$(2))
+
+endef
+
+MOD_ADD_C=$(eval $(call MOD_ADD_SOURCE,$(1),$(2),.c,.i,.o,.gcda,.gcno))
+MOD_ADD_CC=$(eval $(call MOD_ADD_SOURCE,$(1),$(2),.cc,.ii,.oo,.gcda,.gcno))
 
 $(addsuffix .oo,$(CC_MODS)) $(addsuffix .o,$(C_MODS)):	\
 	_ASTCFLAGS+=$(call MOD_ASTCFLAGS,$*)
@@ -110,12 +128,12 @@ dist-clean::
 	rm -f *.exports
 
 .%.moduleinfo: %.c
-	@echo "<member name=\"$*\" displayname=\"$(shell $(GREP) -e AST_MODULE_INFO $< | head -n 1 | cut -d '"' -f 2)\" remove_on_change=\"$(SUBDIR)/$*.o $(SUBDIR)/$*.so\">" > $@
+	@echo "<member name=\"$*\" displayname=\"$(shell $(GREP) -e AST_MODULE_INFO $< | head -n 1 | cut -d '"' -f 2)\" remove_on_change=\"$(SUBDIR)/$*.o $(SUBDIR)/$*.i $(SUBDIR)/$*.so$(MODULEINFO_EXTRA_OUTPUT)\">" > $@
 	$(AWK) -f $(ASTTOPDIR)/build_tools/get_moduleinfo $^ >> $@
 	echo "</member>" >> $@
 
 .%.moduleinfo: %.cc
-	@echo "<member name=\"$*\" displayname=\"$(shell $(GREP) -e AST_MODULE_INFO $< | head -n 1 | cut -d '"' -f 2)\" remove_on_change=\"$(SUBDIR)/$*.oo $(SUBDIR)/$*.so\">" > $@
+	@echo "<member name=\"$*\" displayname=\"$(shell $(GREP) -e AST_MODULE_INFO $< | head -n 1 | cut -d '"' -f 2)\" remove_on_change=\"$(SUBDIR)/$*.oo $(SUBDIR)/$*.ii $(SUBDIR)/$*.so$(MODULEINFO_EXTRA_OUTPUT)\">" > $@
 	$(AWK) -f $(ASTTOPDIR)/build_tools/get_moduleinfo $^ >> $@
 	echo "</member>" >> $@
 

Makefile.rules

@@ -20,6 +20,9 @@
 # Helpful functions
 # call with $(call function,...)
 tolower = $(shell echo $(1) | tr '[:upper:]' '[:lower:]')
+# Takes a list of MENUSELECT_CFLAG Id and returns CFLAGS to declare
+# the ones which are enabled.
+get_menuselect_cflags=$(patsubst %,-D%,$(filter $1,$(MENUSELECT_CFLAGS)))
 
 .PHONY: dist-clean
 
@@ -176,17 +179,17 @@ ifeq ($(GNU_LD),1)
 	$(CMD_PREFIX) $(ASTTOPDIR)/build_tools/make_linker_version_script $* "$(LINKER_SYMBOL_PREFIX)" "$(ASTTOPDIR)"
 endif
 	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
-	$(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@ $(CC_LDFLAGS_SO) $^ $(CC_LIBS)
+	$(CMD_PREFIX) $(CC) -o $@ $(CC_LDFLAGS_SO) $^ $(CC_LIBS)
 
 %.so: %.oo
 ifeq ($(GNU_LD),1)
 	$(CMD_PREFIX) $(ASTTOPDIR)/build_tools/make_linker_version_script $* "$(LINKER_SYMBOL_PREFIX)" "$(ASTTOPDIR)"
 endif
 	$(ECHO_PREFIX) echo "   [LDXX] $^ -> $@"
-	$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(CXX_LDFLAGS_SO) $^ $(CXX_LIBS)
+	$(CMD_PREFIX) $(CXX) -o $@ $(CXX_LDFLAGS_SO) $^ $(CXX_LIBS)
 
 %: %.o
 	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
-	$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $^ $(CXX_LIBS) $(ASTLDFLAGS)
+	$(CMD_PREFIX) $(CXX) -o $@ $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $^ $(CXX_LIBS) $(ASTLDFLAGS)
 
 dist-clean:: clean

README.md

 # The Asterisk(R) Open Source PBX
 ```text
         By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
-        Copyright (C) 2001-2009 Digium, Inc. and other copyright holders.
+        Copyright (C) 2001-2016 Digium, Inc. and other copyright holders.
 ```
 ## SECURITY
 
@@ -69,7 +69,7 @@ list of new features in this version of Asterisk, see the [CHANGES] file.
 
   Ensure that your system contains a compatible compiler and development
 libraries.  Asterisk requires either the GNU Compiler Collection (GCC) version
-3.0 or higher, or a compiler that supports the C99 specification and some of
+4.1 or higher, or a compiler that supports the C99 specification and some of
 the gcc language extensions.  In addition, your system needs to have the C
 library headers available, and the headers and libraries for ncurses.
 

UPGRADE-13.txt

+===========================================================
+===
+=== Information for upgrading between Asterisk versions
+===
+=== These files document all the changes that MUST be taken
+=== into account when upgrading between the Asterisk
+=== versions listed below. These changes may require that
+=== you modify your configuration files, dialplan or (in
+=== some cases) source code if you have your own Asterisk
+=== modules or patches. These files also include advance
+=== notice of any functionality that has been marked as
+=== 'deprecated' and may be removed in a future release,
+=== along with the suggested replacement functionality.
+===
+=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
+=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
+=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
+=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
+=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
+=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
+=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
+===========================================================
+
+General Asterisk Changes:
+ - The asterisk command line -I option and the asterisk.conf internal_timing
+   option are removed and always enabled if any timing module is loaded.
+
+ - The per console verbose level feature as previously implemented caused a
+   large performance penalty.  The fix required some minor incompatibilities
+   if the new rasterisk is used to connect to an earlier version.  If the new
+   rasterisk connects to an older Asterisk version then the root console verbose
+   level is always affected by the "core set verbose" command of the remote
+   console even though it may appear to only affect the current console.  If
+   an older version of rasterisk connects to the new version then the
+   "core set verbose" command will have no effect.
+
+ - The asterisk compatibility options in asterisk.conf have been removed.
+   These options enabled certain backwards compatibility features for
+   pbx_realtime, res_agi, and app_set that made their behaviour similar to
+   Asterisk 1.4. Users who used these backwards compatibility settings should
+   update their dialplans to use ',' instead of '|' as a delimiter, and should
+   use the Set dialplan application instead of the MSet dialplan application.
+
+Build System:
+ - Sample config files have been moved from configs/ to a subfolder of that
+   directory, 'samples'.
+
+ - The menuselect utility has been pulled into the Asterisk repository. As a
+   result, the libxml2 development library is now a required dependency for
+   Asterisk.
+
+ - Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
+   objects will emit additional debug information to the refs log file located
+   in the standard Asterisk log file directory. This log file is useful in
+   tracking down object leaks and other reference counting issues. Prior to
+   this version, this option was only available by modifying the source code
+   directly. This change also includes a new script, refcounter.py, in the
+   contrib folder that will process the refs log file.
+
+Applications:
+
+ConfBridge:
+ - The sound_place_into_conference sound used in Confbridge is now deprecated
+   and is no longer functional since it has been broken since its inception
+   and the fix involved using a different method to achieve the same goal. The
+   new method to achieve this functionality is by using sound_begin to play
+   a sound to the conference when waitmarked users are moved into the conference.
+
+ - Added 'Admin' header to ConfbridgeJoin, ConfbridgeLeave, ConfbridgeMute,
+   ConfbridgeUnmute, and ConfbridgeTalking AMI events.
+
+ControlPlayback:
+ - The ControlPlayback and 'control stream file' AGI command will no longer
+   implicitly answer the channel. If you do not answer the channel prior to
+   using either this application or AGI command, you must send Progress
+   first.
+
+Queue:
+ - Queue rules provided in queuerules.conf can no longer be named "general".
+
+SetMusicOnHold:
+ - The SetMusicOnHold dialplan application was deprecated and has been removed.
+   Users of the application should use the CHANNEL function's musicclass
+   setting instead.
+
+WaitMusicOnHold:
+ - The WaitMusicOnHold dialplan application was deprecated and has been
+   removed. Users of the application should use MusicOnHold with a duration
+   parameter instead.
+
+CDR Backends:
+ - The cdr_sqlite module was deprecated and has been removed. Users of this
+   module should use the cdr_sqlite3_custom module instead.
+
+Channel Drivers:
+
+chan_dahdi:
+ - SS7 support now requires libss7 v2.0 or later.
+
+ - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
+   deal with switches that don't send an inband progress indication in the
+   SETUP ACKNOWLEDGE message.
+   Default is now no.
+
+chan_gtalk
+ - This module was deprecated and has been removed. Users of chan_gtalk
+   should use chan_motif.
+
+chan_h323
+ - This module was deprecated and has been removed. Users of chan_h323
+   should use chan_ooh323.
+
+chan_jingle
+ - This module was deprecated and has been removed. Users of chan_jingle
+   should use chan_motif.
+
+chan_pjsip:
+ - Added a 'force_avp' option to chan_pjsip which will force the usage of
+   'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
+   in SDP offers depending on settings, even when DTLS is used for media
+   encryption.
+
+ - Added a 'media_use_received_transport' option to chan_pjsip which will
+   cause the SDP answer to use the media transport as received in the SDP
+   offer.
+
+chan_sip:
+ - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
+   interoperability.
+
+ - The SIPPEER dialplan function no longer supports using a colon as a
+   delimiter for parameters. The parameters for the function should be
+   delimited using a comma.
+
+ - The SIPCHANINFO dialplan function was deprecated and has been removed. Users
+   of the function should use the CHANNEL function instead.
+
+ - Added a 'force_avp' option for chan_sip. When enabled this option will
+   cause the media transport in the offer or answer SDP to be 'RTP/AVP',
+   'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
+   configured. This option can be set to improve interoperability with WebRTC
+   clients that don't use the RFC defined transport for DTLS.
+
+ - The 'dtlsverify' option in chan_sip now has additional values besides
+   'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
+   will be verified. If 'no' is specified then neither the certificate or
+   fingerprint is verified. If 'certificate' is specified then only the
+   certificate is verified. If 'fingerprint' is specified then only the
+   fingerprint is verified.
+
+ - A 'dtlsfingerprint' option has been added to chan_sip which allows the
+   hash to be specified for the DTLS fingerprint placed in SDP. Supported
+   values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
+
+ - The 'progressinband=never' option is now more zealous in the persecution of
+   progress messages coming from Asterisk. Channels bridged with a SIP channel
+   that has 'progressinband=never' set will not be able to forward their
+   progress indications through to the SIP device. chan_sip will now turn such
+   progress indications into a 180 Ringing (if a 180 has not yet been
+   transmitted) if 'progressinband=never'.
+
+  - The codec preference order in an SDP during an offer is slightly different
+    than previous releases. Prior to Asterisk 13, the preference order of
+    codecs used to be:
+    (1) Our preferred codec
+    (2) Our configured codecs
+    (3) Any non-audio joint codecs
+
+    One of the ways the new media format architecture in Asterisk 13 improves
+    performance is by reference counting formats, such that they can be reused
+    in many places without additional allocation. To not require a large
+    amount of locking, an instance of a format is immutable by convention.
+    This works well except for formats with attributes. Since a media format
+    with an attribute is a different object than the same format without an
+    attribute, we have to carry over the formats with attributes from an
+    inbound offer so that the correct attributes are offered in an outgoing
+    INVITE request. This requires some subtle tweaks to the preference order
+    to ensure that the media format with attributes is offered to a remote
+    peer, as opposed to the same media format (but without attributes) that
+    may be stored in the peer object.
+
+    All of this means that our offer offer list will now be:
+    (1) Our preferred codec
+    (2) Any joint codecs offered by the inbound offer
+    (3) All other codecs that are not the preferred codec and not a joint
+        codec offered by the inbound offer
+
+chan_unistim:
+ - The unistim.conf 'dateformat' has changed meaning of options values to conform
+   values used inside Unistim protocol
+
+ - Added 'dtmf_duration' option with changing default operation to disable
+   receivied dtmf playback on unistim phone
+
+Core:
+
+Account Codes:
+ - accountcode behavior changed somewhat to add functional peeraccount
+   support.  The main change is that local channels now cross accountcode
+   and peeraccount across the special bridge between the ;1 and ;2 channels
+   just like channels between normal bridges.  See the CHANGES file for
+   more information.
+
+ARI:
+ - The ARI version has been changed to 1.5.0. This is to reflect backwards
+   compatible changes made since 12.0.0 was released.
+
+ - Added a new ARI resource 'mailboxes' which allows the creation and
+   modification of mailboxes managed by external MWI. Modules res_mwi_external
+   and res_stasis_mailbox must be enabled to use this resource.
+
+ - Added new events for externally initiated transfers. The event
+   BridgeBlindTransfer is now raised when a channel initiates a blind transfer
+   of a bridge in the ARI controlled application to the dialplan; the
+   BridgeAttendedTransfer event is raised when a channel initiates an
+   attended transfer of a bridge in the ARI controlled application to the
+   dialplan.
+
+ - Channel variables may now be specified as a body parameter to the
+   POST /channels operation. The 'variables' key in the JSON is interpreted
+   as a sequence of key/value pairs that will be added to the created channel
+   as channel variables. Other parameters in the JSON body are treated as
+   query parameters of the same name.
+
+ - A bug fix in bridge creation has caused a behavioural change in how
+   subscriptions are created for bridges. A bridge created through ARI, does
+   not, by itself, have a subscription created for any particular Stasis
+   application. When a channel in a Stasis application joins a bridge, an
+   implicit event subscription is created for that bridge as well. Previously,
+   when a channel left such a bridge, the subscription was leaked; this allowed
+   for later bridge events to continue to be pushed to the subscribed
+   applications. That leak has been fixed; as a result, bridge events that were
+   delivered after a channel left the bridge are no longer delivered. An
+   application must subscribe to a bridge through the applications resource if
+   it wishes to receive all events related to a bridge.
+
+AMI:
+ - The AMI version has been changed to 2.5.0. This is to reflect backwards
+   compatible changes made since 12.0.0 was released.
+
+ - The DialStatus field in the DialEnd event can now have additional values.
+   This includes ABORT, CONTINUE, and GOTO.
+
+ - The res_mwi_external_ami module can, if loaded, provide additional AMI
+   actions and events that convey MWI state within Asterisk. This includes
+   the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
+   MWIGetComplete events that occur in response to an MWIGet action.
+
+ - AMI now contains a new class authorization, 'security'. This is used with
+   the following new events: FailedACL, InvalidAccountID, SessionLimit,
+   MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
+   RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
+   InvalidPassword, ChallengeSent, and InvalidTransport.
+
+ - Bridge related events now have two additional fields: BridgeName and
+   BridgeCreator. BridgeName is a descriptive name for the bridge;
+   BridgeCreator is the name of the entity that created the bridge. This
+   affects the following events: ConfbridgeStart, ConfbridgeEnd,
+   ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
+   ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
+   AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
+
+ - MixMonitor AMI actions now require users to have authorization classes.
+   * MixMonitor - system
+   * MixMonitorMute - call or system
+   * StopMixMonitor - call or system
+
+ - Removed the undocumented manager.conf block-sockets option.  It interferes with
+   TCP/TLS inactivity timeouts.
+
+ - The response to the PresenceState AMI action has historically contained two
+   Message keys. The first of these is used as an informative message regarding
+   the success/failure of the action; the second contains a Presence state
+   specific message. Having two keys with the same unique name in an AMI
+   message is cumbersome for some client; hence, the Presence specific Message
+   has been deprecated. The message will now contain a PresenceMessage key
+   for the presence specific information; the Message key containing presence
+   information will be removed in the next major version of AMI.
+
+ - The manager.conf 'eventfilter' now takes an "extended" regular expression
+   instead of a "basic" one.
+
+CDRs:
+ - The "endbeforehexten" setting now defaults to "yes", instead of "no".
+   When set to "no", yhis setting will cause a new CDR to be generated when a
+   channel enters into hangup logic (either the 'h' extension or a hangup
+   handler subroutine). In general, this is not the preferred default: this
+   causes extra CDRs to be generated for a channel in many common dialplans.
+
+CLI commands:
+ - "core show settings" now lists the current console verbosity in addition
+   to the root console verbosity.
+
+ - "core set verbose" has not been able to support the by module verbose
+   logging levels since verbose logging levels were made per console.  That
+   syntax is now removed and a silence option added in its place.
+
+Logging:
+ - The 'verbose' setting in logger.conf still takes an optional argument,
+   specifying the verbosity level for each logging destination.  However,
+   the default is now to once again follow the current root console level.
+   As a result, using the AMI Command action with "core set verbose" could
+   again set the root console verbose level and affect the verbose level
+   logged.
+
+HTTP:
+ - Added http.conf session_inactivity timer option to close HTTP connections
+   that aren't doing anything.
+
+ - Added support for persistent HTTP connections.  To enable persistent
+   HTTP connections configure the keep alive time between HTTP requests.  The
+   keep alive time between HTTP requests is configured in http.conf with the
+   session_keep_alive parameter.
+
+Realtime Configuration:
+ - WARNING: The database migration script that adds the 'extensions' table for
+   realtime had to be modified due to an error when installing for MySQL.  The
+   'extensions' table's 'id' column was changed to be a primary key.  This could
+   potentially cause a migration problem.  If so, it may be necessary to
+   manually alter the affected table/column to bring it back in line with the
+   migration scripts.
+
+ - New columns have been added to realtime tables for 'support_path' on
+   ps_registrations and ps_aors and for 'path' on ps_contacts for the new
+   SIP Path support in chan_pjsip.
+
+ - The following new tables have been added for pjsip realtime: 'ps_systems',
+   'ps_globals', 'ps_tranports', 'ps_registrations'.
+
+ - The following columns were added to the 'ps_aors' realtime table:
+   'maximum_expiration', 'outbound_proxy', and 'support_path'.
+
+ - The following columns were added to the 'ps_contacts' realtime table:
+   'outbound_proxy', 'user_agent', and 'path'.
+
+ - New columns have been added to the ps_endpoints realtime table for the
+   'media_address', 'redirect_method' and 'set_var' options.  Also the
+   'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
+   'message_context' was added to let users configure how MESSAGE requests are
+   routed to the dialplan.
+
+ - A new column was added to the 'ps_globals' realtime table for the 'debug'
+   option.
+
+ - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
+   yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
+   changed from yes/no enumerators to integer values. PJSIP transport column
+   'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
+   been changed from a yes/no enumerator to an integer value.
+
+ - The 'queues' and 'queue_members' realtime tables have been added to the
+   config Alembic scripts.
+
+ - A new set of Alembic scripts has been added for CDR tables. This will create
+   a 'cdr' table with the default schema that Asterisk expects.
+
+ - A new upgrade script has been added that adds a 'queue_rules' table for
+   app_queue. Users of app_queue can store queue rules in a database. It is
+   important to note that app_queue only looks for this table on module load or
+   module reload; for more information, see the CHANGES file.
+
+Resources:
+
+res_odbc:
+- The compatibility setting, allow_empty_string_in_nontext, has been removed.
+  Empty column values will be stored as empty strings during realtime updates.
+
+res_jabber:
+ - This module was deprecated and has been removed. Users of this module should
+   use res_xmpp instead.
+
+res_http_websocket:
+ - Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
+   'websocket_write_timeout'. When a websocket connection exists where Asterisk
+   writes a substantial amount of data to the connected client, and the connected
+   client is slow to process the received data, the socket may be disconnected.
+   In such cases, it may be necessary to adjust this value.
+   Default is 100 ms.
+Scripts:
+
+safe_asterisk:
+ - The safe_asterisk script was previously not installed on top of an existing
+   version. This caused bug-fixes in that script not to be deployed. If your
+   safe_asterisk script is customized, be sure to keep your changes. Custom
+   values for variables should be created in *.sh file(s) inside
+   ASTETCDIR/startup.d/. See ASTERISK-21965.
+
+ - Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
+   you use tools to parse either of them, update your parse functions
+   accordingly. The changed strings are:
+   - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
+   - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
+
+Utilities:
+ - The refcounter program has been removed in favor of the refcounter.py script
+   in contrib/scripts.
+
+===========================================================
+===========================================================

UPGRADE-14.txt

+===========================================================
+===
+=== Information for upgrading between Asterisk versions
+===
+=== These files document all the changes that MUST be taken
+=== into account when upgrading between the Asterisk
+=== versions listed below. These changes may require that
+=== you modify your configuration files, dialplan or (in
+=== some cases) source code if you have your own Asterisk
+=== modules or patches. These files also include advance
+=== notice of any functionality that has been marked as
+=== 'deprecated' and may be removed in a future release,
+=== along with the suggested replacement functionality.
+===
+=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
+=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
+=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
+=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
+=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
+=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
+=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
+=== UPGRADE-13.txt  -- Upgrade info for 12 to 13
+===========================================================
+
+From 14.6.0 to 14.7.0:
+
+Core:
+ - ast_app_parse_timelen now returns an error if it encounters extra characters
+   at the end of the string to be parsed.
+
+From 14.4.0 to 14.5.0:
+
+Core:
+ - Support for embedded modules has been removed.  This has not worked in
+   many years.  LOADABLE_MODULES menuselect option is also removed as
+   loadable module support is now always enabled.
+
+From 14.3.0 to 14.4.0:
+
+res_rtp_asterisk:
+ - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
+   Data and Control Packets on a Single Port." For the PJSIP channel driver,
+   chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
+   to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
+   globally or on a per-peer basis in sip.conf.
+
+New in 14.0.0
+
+ARI:
+ - The policy for when to send "Dial" events has changed. Previously, "Dial"
+   events were sent on the calling channel's topic. However, starting in Asterisk
+   14, if there is no calling channel on which to send the event, the event is
+   instead sent on the called channel's topic. Note that for the ARI channels
+   resource's dial operation, this means that the "Dial" events will always be
+   sent on the called channel's topic.
+
+Channel Drivers:
+
+chan_dahdi:
+ - For users using the FXO port (FXS signaling) distinctive ring detection
+   feature, you will need to adjust the dringX count values.  The count
+   values now only record ring end events instead of any DAHDI event.  A
+   ring-ring-ring pattern would exceed the pattern limits and stop
+   Caller-ID detection.
+
+chan_sip:
+ - The SIP dial string has been extended past the [!dnid] option by another
+   exclamation mark: [!dnid[!fromuri].  An exclamation mark in the To-URI
+   will now mean changes to the From-URI.
+
+Core:
+ - The REF_DEBUG compiler flag is now used to enable refdebug by default.
+   The setting can be overridden in asterisk.conf by setting refdebug in
+   the options category.  No recompile is required to enable/disable it.
+
+ - Modified processing of command-line options to first parse only what
+   is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
+   the remaining options are processed.  The -X option now applies to
+   asterisk.conf only.  To enable #exec for other config files you must
+   set execincludes=yes in asterisk.conf.  Any other option set on the
+   command-line will now override the equivalent setting from asterisk.conf.
+
+AMI:
+ - The 'ModuleCheck' Action's Version key will no longer show the module
+   version. The value will always be blank.
+
+CLI:
+ - The 'core show file version' command has been removed. When Asterisk
+   moved to Git, the source control version support was removed. As a
+   result, the CLi command was no longer useful and was removed as well.
+
+Logging:
+ - The first callid created is now 1 instead of 0.  The value 0
+   is now reserved to represent a lack of callid.
+
+AMI:
+ - The Command action now sends the output from the CLI command as a series
+   of Output headers for each line instead of as a block of text with the
+   --END COMMAND-- delimiter to match the output from other actions.
+
+   Commands that fail to execute (no such command, invalid syntax etc.) now
+   return an Error response instead of Success.
+
+app_amd:
+ - The 'maximum_number_of_words' configuration option and parameter to the AMD
+   application previously did not match the documented functionality + variable
+   name.  In Asterisk 13, a value of '3' would mean that if '3' words were detected,
+   the result would be detection as a 'MACHINE'.  As of this version, the value
+   reflects the maximum words that if EXCEEDED (rather than reached), would
+   result in detection as a machine.  This means that you should update this
+   value to be one higher than your previos value, if your previous value
+   was working well for you.
+
+===========================================================
+===========================================================

UPGRADE-15.txt

+===========================================================
+===
+=== Information for upgrading between Asterisk versions
+===
+=== These files document all the changes that MUST be taken
+=== into account when upgrading between the Asterisk
+=== versions listed below. These changes may require that
+=== you modify your configuration files, dialplan or (in
+=== some cases) source code if you have your own Asterisk
+=== modules or patches. These files also include advance
+=== notice of any functionality that has been marked as
+=== 'deprecated' and may be removed in a future release,
+=== along with the suggested replacement functionality.
+===
+=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
+=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
+=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
+=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
+=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
+=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
+=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
+=== UPGRADE-13.txt  -- Upgrade info for 12 to 13
+=== UPGRADE-14.txt  -- Upgrade info for 13 to 14
+===========================================================
+
+From 15.2.0 to 15.3.0:
+
+res_pjsip
+------------------
+ * Users who are matching endpoints by SIP header need to reevaluate their
+   global "endpoint_identifier_order" option in light of the "ip" endpoint
+   identifier method split into the "ip" and "header" endpoint identifier
+   methods.
+
+res_pjsip_endpoint_identifier_ip
+------------------
+ * The endpoint identifier "ip" method previously recognized endpoints either
+   by IP address or a matching SIP header.  The "ip" endpoint identifier method
+   is now split into the "ip" and "header" endpoint identifier methods.  The
+   "ip" endpoint identifier method only matches by IP address and the "header"
+   endpoint identifier method only matches by SIP header.  The split allows the
+   user to control the relative priority of the IP address and the SIP header
+   identification methods in the global "endpoint_identifier_order" option.
+   e.g., If you have two type=identify sections where one matches by IP address
+   for endpoint alice and the other matches by SIP header for endpoint bob then
+   you can now predict which endpoint is matched when a request comes in that
+   matches both.
+
+New in 15.0.0:
+
+Build System:
+ - '--with-pjproject-bundled' is now the default when running ./configure
+   It can be disabled with '--without-pjproject-bundled'.
+
+Core:
+ - Multi-stream support has been added so a channel can have multiple
+   streams of the same type such as audio and video.
+
+ - The 'Data Retrieval API' has been removed. This API was not actively
+   maintained, was not added to new modules (such as res_pjsip), and there
+   exist better alternatives to acquire the same information, such as the
+   ARI. As a result, the 'DataGet' AMI action as well as the 'data get'
+   CLI command have been removed.

addons/Makefile

@@ -14,15 +14,17 @@
 MENUSELECT_CATEGORY=ADDONS
 MENUSELECT_DESCRIPTION=Add-ons (See README-addons.txt)
 
-H323OBJS:=ooCmdChannel.o ooLogChan.o ooUtils.o ooGkClient.o context.o \
-	ooDateTime.o decode.o dlist.o encode.o errmgmt.o \
-	memheap.o ootrace.o oochannels.o ooh245.o ooports.o \
-	ooq931.o ooCapability.o ooSocket.o perutil.o eventHandler.o \
-	ooCalls.o ooStackCmds.o ooh323.o ooh323ep.o printHandler.o \
-	rtctype.o ooTimer.o h323/H235-SECURITY-MESSAGESDec.o \
-	h323/H235-SECURITY-MESSAGESEnc.o h323/H323-MESSAGES.o h323/H323-MESSAGESDec.o \
-	h323/H323-MESSAGESEnc.o h323/MULTIMEDIA-SYSTEM-CONTROL.o \
-	h323/MULTIMEDIA-SYSTEM-CONTROLDec.o h323/MULTIMEDIA-SYSTEM-CONTROLEnc.o
+OOH323C:=ooCmdChannel.c ooLogChan.c ooUtils.c ooGkClient.c context.c \
+	ooDateTime.c decode.c dlist.c encode.c errmgmt.c \
+	memheap.c ootrace.c oochannels.c ooh245.c ooports.c \
+	ooq931.c ooCapability.c ooSocket.c perutil.c eventHandler.c \
+	ooCalls.c ooStackCmds.c ooh323.c ooh323ep.c printHandler.c \
+	rtctype.c ooTimer.c h323/H235-SECURITY-MESSAGESDec.c \
+	h323/H235-SECURITY-MESSAGESEnc.c h323/H323-MESSAGES.c h323/H323-MESSAGESDec.c \
+	h323/H323-MESSAGESEnc.c h323/MULTIMEDIA-SYSTEM-CONTROL.c \
+	h323/MULTIMEDIA-SYSTEM-CONTROLDec.c h323/MULTIMEDIA-SYSTEM-CONTROLEnc.c
+
+H323SOURCE:=$(addprefix ooh323c/src/,$(OOH323C)) ooh323cDriver.c
 
 H323CFLAGS:=-Iooh323c/src -Iooh323c/src/h323
 
@@ -57,22 +59,14 @@ endif
 
 include $(ASTTOPDIR)/Makefile.moddir_rules
 
-clean::
-	if [ -f mp3/Makefile ] ; then $(MAKE) -C mp3 clean ; fi
-	rm -f $(addprefix ooh323c/src/,$(H323OBJS))
-
-ifneq ($(wildcard mp3/Makefile),)
-format_mp3.so: mp3/common.o mp3/dct64_i386.o mp3/decode_ntom.o mp3/layer3.o mp3/tabinit.o mp3/interface.o
-endif
-
 chan_mobile.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
-chan_ooh323.o: _ASTCFLAGS+=$(H323CFLAGS)
 chan_ooh323.so: _ASTCFLAGS+=$(H323CFLAGS)
-chan_ooh323.so: $(addprefix ooh323c/src/,$(H323OBJS)) chan_ooh323.o ooh323cDriver.o
-$(addprefix ooh323c/src/,$(H323OBJS)) chan_ooh323.o ooh323cDriver.o: _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_ooh323)
+$(call MOD_ADD_C,chan_ooh323,$(H323SOURCE))
 
-.PHONY: check_mp3
+ifneq ($(wildcard mp3/Makefile),)
+$(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)
 
-ifeq ($(wildcard mp3/Makefile),)
-.PHONY: format_mp3.o format_mp3.so
+.PHONY: check_mp3
+else
+.PHONY: check_mp3 format_mp3.o format_mp3.so
 endif

addons/app_mysql.c

@@ -218,7 +218,7 @@ static int add_identifier(struct ast_channel *chan, int identifier_type, void *d
 		ast_log(LOG_WARNING, "Unable to lock identifiers list\n");
 		return -1;
 	} else {
-		i = malloc(sizeof(*i));
+		i = ast_malloc(sizeof(*i));
 		AST_LIST_TRAVERSE(headp, j, entries) {
 			if (j->identifier > maxidentifier) {
 				maxidentifier = j->identifier;
@@ -247,7 +247,7 @@ static int del_identifier(int identifier, int identifier_type)
 			if ((i->identifier == identifier) &&
 			    (i->identifier_type == identifier_type)) {
 				AST_LIST_REMOVE(headp, i, entries);
-				free(i);
+				ast_free(i);
 				found = 1;
 				break;
 			}

addons/cdr_mysql.c

@@ -42,8 +42,6 @@
 
 #include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
 #include <mysql/mysql.h>
 #include <mysql/errmsg.h>
 
@@ -127,7 +125,9 @@ static char *handle_cli_cdr_mysql_status(struct ast_cli_entry *e, int cmd, struc
 		return CLI_SHOWUSAGE;
 
 	if (connected) {
-		char status[256], status2[100] = "";
+		char status[256];
+		char status2[100] = "";
+		char buf[362]; /* 256+100+" for "+NULL */
 		int ctime = time(NULL) - connect_time;
 		if (dbport)
 			snprintf(status, 255, "Connected to %s@%s, port %d", ast_str_buffer(dbname), ast_str_buffer(hostname), dbport);
@@ -140,17 +140,10 @@ static char *handle_cli_cdr_mysql_status(struct ast_cli_entry *e, int cmd, struc
 			snprintf(status2, 99, " with username %s", ast_str_buffer(dbuser));
 		if (ast_str_strlen(dbtable))
 			snprintf(status2, 99, " using table %s", ast_str_buffer(dbtable));
-		if (ctime > 31536000) {
-			ast_cli(a->fd, "%s%s for %d years, %d days, %d hours, %d minutes, %d seconds.\n", status, status2, ctime / 31536000, (ctime % 31536000) / 86400, (ctime % 86400) / 3600, (ctime % 3600) / 60, ctime % 60);
-		} else if (ctime > 86400) {
-			ast_cli(a->fd, "%s%s for %d days, %d hours, %d minutes, %d seconds.\n", status, status2, ctime / 86400, (ctime % 86400) / 3600, (ctime % 3600) / 60, ctime % 60);
-		} else if (ctime > 3600) {
-			ast_cli(a->fd, "%s%s for %d hours, %d minutes, %d seconds.\n", status, status2, ctime / 3600, (ctime % 3600) / 60, ctime % 60);
-		} else if (ctime > 60) {
-			ast_cli(a->fd, "%s%s for %d minutes, %d seconds.\n", status, status2, ctime / 60, ctime % 60);
-		} else {
-			ast_cli(a->fd, "%s%s for %d seconds.\n", status, status2, ctime);
-		}
+
+		snprintf(buf, sizeof(buf), "%s%s for ", status, status2);
+		ast_cli_print_timestr_fromseconds(a->fd, ctime, buf);
+
 		if (records == totalrecords)
 			ast_cli(a->fd, "  Wrote %d records since last restart.\n", totalrecords);
 		else
@@ -455,6 +448,140 @@ static int my_load_config_number(struct ast_config *cfg, const char *category, c
 	return 0;
 }
 
+/** Connect to MySQL. Initializes the connection.
+ *
+ * * Assumes the read-write lock for columns is held.
+ * * Caller should allocate and free cfg
+ * */
+static int my_connect_db(struct ast_config *cfg)
+{
+	struct ast_variable *var;
+	char *temp;
+	MYSQL_ROW row;
+	MYSQL_RES *result;
+	char sqldesc[128];
+#if MYSQL_VERSION_ID >= 50013
+	my_bool my_bool_true = 1;
+#endif
+
+	mysql_init(&mysql);
+
+	if (timeout && mysql_options(&mysql, MYSQL_OPT_CONNECT_TIMEOUT, (char *)&timeout) != 0) {
+		ast_log(LOG_ERROR, "cdr_mysql: mysql_options returned (%d) %s\n", mysql_errno(&mysql), mysql_error(&mysql));
+	}
+
+#if MYSQL_VERSION_ID >= 50013
+	/* Add option for automatic reconnection */
+	if (mysql_options(&mysql, MYSQL_OPT_RECONNECT, &my_bool_true) != 0) {
+		ast_log(LOG_ERROR, "cdr_mysql: mysql_options returned (%d) %s\n", mysql_errno(&mysql), mysql_error(&mysql));
+	}
+#endif
+
+	if ((ssl_ca && ast_str_strlen(ssl_ca)) || (ssl_cert && ast_str_strlen(ssl_cert)) || (ssl_key && ast_str_strlen(ssl_key))) {
+		mysql_ssl_set(&mysql,
+			ssl_key ? ast_str_buffer(ssl_key) : NULL,
+			ssl_cert ? ast_str_buffer(ssl_cert) : NULL,
+			ssl_ca ? ast_str_buffer(ssl_ca) : NULL,
+			NULL, NULL);
+	}
+	temp = dbsock && ast_str_strlen(dbsock) ? ast_str_buffer(dbsock) : NULL;
+
+	configure_connection_charset();
+
+	if (!mysql_real_connect(&mysql, ast_str_buffer(hostname), ast_str_buffer(dbuser), ast_str_buffer(password), ast_str_buffer(dbname), dbport, temp, ssl_ca && ast_str_strlen(ssl_ca) ? CLIENT_SSL : 0)) {
+		ast_log(LOG_ERROR, "Failed to connect to mysql database %s on %s.\n", ast_str_buffer(dbname), ast_str_buffer(hostname));
+		connected = 0;
+		records = 0;
+
+		return AST_MODULE_LOAD_SUCCESS;	/* May be reconnected later */
+	}
+
+	ast_debug(1, "Successfully connected to MySQL database.\n");
+	connected = 1;
+	records = 0;
+	connect_time = time(NULL);
+
+	/* Get table description */
+	snprintf(sqldesc, sizeof(sqldesc), "DESC %s", dbtable ? ast_str_buffer(dbtable) : "cdr");
+	if (mysql_query(&mysql, sqldesc)) {
+		ast_log(LOG_ERROR, "Unable to query table description!!  Logging disabled.\n");
+		mysql_close(&mysql);
+		connected = 0;
+
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	if (!(result = mysql_store_result(&mysql))) {
+		ast_log(LOG_ERROR, "Unable to query table description!!  Logging disabled.\n");
+		mysql_close(&mysql);
+		connected = 0;
+
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	while ((row = mysql_fetch_row(result))) {
+		struct column *entry;
+		char *cdrvar = "", *staticvalue = "";
+
+		ast_debug(1, "Got a field '%s' of type '%s'\n", row[0], row[1]);
+		/* Check for an alias or a static value */
+		for (var = ast_variable_browse(cfg, "columns"); var; var = var->next) {
+			if (strncmp(var->name, "alias", 5) == 0 && strcasecmp(var->value, row[0]) == 0 ) {
+				char *alias = ast_strdupa(var->name + 5);
+				cdrvar = ast_strip(alias);
+				ast_verb(3, "Found alias %s for column %s\n", cdrvar, row[0]);
+				break;
+			} else if (strncmp(var->name, "static", 6) == 0 && strcasecmp(var->value, row[0]) == 0) {
+				char *item = ast_strdupa(var->name + 6);
+				item = ast_strip(item);
+				if (item[0] == '"' && item[strlen(item) - 1] == '"') {
+					/* Remove surrounding quotes */
+					item[strlen(item) - 1] = '\0';
+					item++;
+				}
+				staticvalue = item;
+			}
+		}
+
+		entry = ast_calloc(sizeof(char), sizeof(*entry) + strlen(row[0]) + 1 + strlen(cdrvar) + 1 + strlen(staticvalue) + 1 + strlen(row[1]) + 1);
+		if (!entry) {
+			ast_log(LOG_ERROR, "Out of memory creating entry for column '%s'\n", row[0]);
+			mysql_free_result(result);
+			return AST_MODULE_LOAD_DECLINE;
+		}
+
+		entry->name = (char *)entry + sizeof(*entry);
+		strcpy(entry->name, row[0]);
+
+		if (!ast_strlen_zero(cdrvar)) {
+			entry->cdrname = entry->name + strlen(row[0]) + 1;
+			strcpy(entry->cdrname, cdrvar);
+		} else { /* Point to same place as the column name */
+			entry->cdrname = (char *)entry + sizeof(*entry);
+		}
+
+		if (!ast_strlen_zero(staticvalue)) {
+			entry->staticvalue = entry->cdrname + strlen(entry->cdrname) + 1;
+			strcpy(entry->staticvalue, staticvalue);
+			ast_debug(1, "staticvalue length: %d\n", (int) strlen(staticvalue) );
+			entry->type = entry->staticvalue + strlen(entry->staticvalue) + 1;
+		} else {
+			entry->type = entry->cdrname + strlen(entry->cdrname) + 1;
+		}
+		strcpy(entry->type, row[1]);
+
+		ast_debug(1, "Entry name '%s'\n", entry->name);
+		ast_debug(1, "   cdrname '%s'\n", entry->cdrname);
+		ast_debug(1, "    static '%s'\n", entry->staticvalue);
+		ast_debug(1, "      type '%s'\n", entry->type);
+
+		AST_LIST_INSERT_TAIL(&columns, entry, list);
+	}
+	mysql_free_result(result);
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
 static int my_load_module(int reload)
 {
 	int res;
@@ -465,14 +592,7 @@ static int my_load_module(int reload)
 	 * rescan the table layout. */
 	struct ast_flags config_flags = { 0 };
 	struct column *entry;
-	char *temp;
 	struct ast_str *compat;
-	MYSQL_ROW row;
-	MYSQL_RES *result;
-	char sqldesc[128];
-#if MYSQL_VERSION_ID >= 50013
-	my_bool my_bool_true = 1;
-#endif
 
 	/* Cannot use a conditionally different flag, because the table layout may
 	 * have changed, which is not detectable by config file change detection,
@@ -569,129 +689,12 @@ static int my_load_module(int reload)
 		ast_debug(1, "Got DB charset of %s\n", ast_str_buffer(dbcharset));
 	}
 
-	mysql_init(&mysql);
-
-	if (timeout && mysql_options(&mysql, MYSQL_OPT_CONNECT_TIMEOUT, (char *)&timeout) != 0) {
-		ast_log(LOG_ERROR, "cdr_mysql: mysql_options returned (%d) %s\n", mysql_errno(&mysql), mysql_error(&mysql));
-	}
-
-#if MYSQL_VERSION_ID >= 50013
-	/* Add option for automatic reconnection */
-	if (mysql_options(&mysql, MYSQL_OPT_RECONNECT, &my_bool_true) != 0) {
-		ast_log(LOG_ERROR, "cdr_mysql: mysql_options returned (%d) %s\n", mysql_errno(&mysql), mysql_error(&mysql));
-	}
-#endif
-
-	if ((ssl_ca && ast_str_strlen(ssl_ca)) || (ssl_cert && ast_str_strlen(ssl_cert)) || (ssl_key && ast_str_strlen(ssl_key))) {
-		mysql_ssl_set(&mysql,
-			ssl_key ? ast_str_buffer(ssl_key) : NULL,
-			ssl_cert ? ast_str_buffer(ssl_cert) : NULL,
-			ssl_ca ? ast_str_buffer(ssl_ca) : NULL,
-			NULL, NULL);
-	}
-	temp = dbsock && ast_str_strlen(dbsock) ? ast_str_buffer(dbsock) : NULL;
-
-	configure_connection_charset();
-
-	if (!mysql_real_connect(&mysql, ast_str_buffer(hostname), ast_str_buffer(dbuser), ast_str_buffer(password), ast_str_buffer(dbname), dbport, temp, ssl_ca && ast_str_strlen(ssl_ca) ? CLIENT_SSL : 0)) {
-		ast_log(LOG_ERROR, "Failed to connect to mysql database %s on %s.\n", ast_str_buffer(dbname), ast_str_buffer(hostname));
-		connected = 0;
-		records = 0;
-	} else {
-		ast_debug(1, "Successfully connected to MySQL database.\n");
-		connected = 1;
-		records = 0;
-		connect_time = time(NULL);
-
-		/* Get table description */
-		snprintf(sqldesc, sizeof(sqldesc), "DESC %s", dbtable ? ast_str_buffer(dbtable) : "cdr");
-		if (mysql_query(&mysql, sqldesc)) {
-			ast_log(LOG_ERROR, "Unable to query table description!!  Logging disabled.\n");
-			mysql_close(&mysql);
-			connected = 0;
-			AST_RWLIST_UNLOCK(&columns);
-			ast_config_destroy(cfg);
-			free_strings();
-
-			return AST_MODULE_LOAD_DECLINE;
-		}
-
-		if (!(result = mysql_store_result(&mysql))) {
-			ast_log(LOG_ERROR, "Unable to query table description!!  Logging disabled.\n");
-			mysql_close(&mysql);
-			connected = 0;
-			AST_RWLIST_UNLOCK(&columns);
-			ast_config_destroy(cfg);
-			free_strings();
-
-			return AST_MODULE_LOAD_DECLINE;
-		}
-
-		while ((row = mysql_fetch_row(result))) {
-			struct column *entry;
-			char *cdrvar = "", *staticvalue = "";
-
-			ast_debug(1, "Got a field '%s' of type '%s'\n", row[0], row[1]);
-			/* Check for an alias or a static value */
-			for (var = ast_variable_browse(cfg, "columns"); var; var = var->next) {
-				if (strncmp(var->name, "alias", 5) == 0 && strcasecmp(var->value, row[0]) == 0 ) {
-					char *alias = ast_strdupa(var->name + 5);
-					cdrvar = ast_strip(alias);
-					ast_verb(3, "Found alias %s for column %s\n", cdrvar, row[0]);
-					break;
-				} else if (strncmp(var->name, "static", 6) == 0 && strcasecmp(var->value, row[0]) == 0) {
-					char *item = ast_strdupa(var->name + 6);
-					item = ast_strip(item);
-					if (item[0] == '"' && item[strlen(item) - 1] == '"') {
-						/* Remove surrounding quotes */
-						item[strlen(item) - 1] = '\0';
-						item++;
-					}
-					staticvalue = item;
-				}
-			}
-
-			entry = ast_calloc(sizeof(char), sizeof(*entry) + strlen(row[0]) + 1 + strlen(cdrvar) + 1 + strlen(staticvalue) + 1 + strlen(row[1]) + 1);
-			if (!entry) {
-				ast_log(LOG_ERROR, "Out of memory creating entry for column '%s'\n", row[0]);
-				res = -1;
-				break;
-			}
-
-			entry->name = (char *)entry + sizeof(*entry);
-			strcpy(entry->name, row[0]);
-
-			if (!ast_strlen_zero(cdrvar)) {
-				entry->cdrname = entry->name + strlen(row[0]) + 1;
-				strcpy(entry->cdrname, cdrvar);
-			} else { /* Point to same place as the column name */
-				entry->cdrname = (char *)entry + sizeof(*entry);
-			}
-
-			if (!ast_strlen_zero(staticvalue)) {
-				entry->staticvalue = entry->cdrname + strlen(entry->cdrname) + 1;
-				strcpy(entry->staticvalue, staticvalue);
-				ast_debug(1, "staticvalue length: %d\n", (int) strlen(staticvalue) );
-				entry->type = entry->staticvalue + strlen(entry->staticvalue) + 1;
-			} else {
-				entry->type = entry->cdrname + strlen(entry->cdrname) + 1;
-			}
-			strcpy(entry->type, row[1]);
-
-			ast_debug(1, "Entry name '%s'\n", entry->name);
-			ast_debug(1, "   cdrname '%s'\n", entry->cdrname);
-			ast_debug(1, "    static '%s'\n", entry->staticvalue);
-			ast_debug(1, "      type '%s'\n", entry->type);
-
-			AST_LIST_INSERT_TAIL(&columns, entry, list);
-		}
-		mysql_free_result(result);
-	}
+	res = my_connect_db(cfg);
 	AST_RWLIST_UNLOCK(&columns);
 	ast_config_destroy(cfg);
-	if (res < 0) {
+	if (res != AST_MODULE_LOAD_SUCCESS) {
 		my_unload_module(0);
-		return AST_MODULE_LOAD_DECLINE;
+		return res;
 	}
 
 	if (!reload) {
@@ -739,4 +742,5 @@ AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "MySQL CDR Backend",
 	.load = load_module,
 	.unload = unload_module,
 	.reload = reload,
+	.requires = "cdr",
 );

addons/chan_mobile.c

@@ -42,8 +42,6 @@
 
 #include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
 #include <pthread.h>
 #include <signal.h>
 
@@ -4818,8 +4816,8 @@ e_cleanup:
 }
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Bluetooth Mobile Device Channel Driver",
-		.support_level = AST_MODULE_SUPPORT_EXTENDED,
-		.load = load_module,
-		.unload = unload_module,
-		.load_pri = AST_MODPRI_CHANNEL_DRIVER,
+	.support_level = AST_MODULE_SUPPORT_EXTENDED,
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_CHANNEL_DRIVER,
 );

addons/chan_ooh323.c

@@ -359,6 +359,7 @@ static int  gDTMFCodec = 101;
 static int  gFAXdetect = FAXDETECT_CNG;
 static int  gT38Support = T38_FAXGW;
 static char gGatekeeper[100];
+static char gRASIP[2+8*4+7];	/* Max for IPv6 addr */
 static enum RasGatekeeperMode gRasGkMode = RasNoGatekeeper;
 
 static int  gIsGateway = 0;
@@ -590,7 +591,7 @@ static struct ooh323_pvt *ooh323_alloc(int callref, char *callToken)
 
 	pvt->call_reference = callref;
 	if (callToken)
-		pvt->callToken = strdup(callToken);
+		pvt->callToken = ast_strdup(callToken);
 
 	/* whether to use gk for this call */
 	if (gRasGkMode == RasNoGatekeeper)
@@ -697,8 +698,8 @@ static struct ast_channel *ooh323_request(const char *type, struct ast_format_ca
 	}
 
 	if (peer) {
-		p->username = strdup(peer->name);
-		p->host = strdup(peer->ip);
+		p->username = ast_strdup(peer->name);
+		p->host = ast_strdup(peer->ip);
 		p->port = peer->port;
 		/* Disable gk as we are going to call a known peer*/
 		/* OO_SETFLAG(p->flags, H323_DISABLEGK); */
@@ -759,9 +760,9 @@ static struct ast_channel *ooh323_request(const char *type, struct ast_format_ca
 		p->directrtp = gDirectRTP;
 		p->earlydirect = gEarlyDirect;
 
-		p->username = strdup(dest);
+		p->username = ast_strdup(dest);
 
-		p->host = strdup(dest);
+		p->host = ast_strdup(dest);
 		if (port > 0) {
 			p->port = port;
 		}
@@ -1016,22 +1017,22 @@ static int ooh323_call(struct ast_channel *ast, const char *dest, int timeout)
 	ast_mutex_lock(&p->lock);
 	ast_set_flag(p, H323_OUTGOING);
 	if (ast_channel_connected(ast)->id.number.valid && ast_channel_connected(ast)->id.number.str) {
-		free(p->callerid_num);
-		p->callerid_num = strdup(ast_channel_connected(ast)->id.number.str);
+		ast_free(p->callerid_num);
+		p->callerid_num = ast_strdup(ast_channel_connected(ast)->id.number.str);
 	}
 
 	if (ast_channel_connected(ast)->id.name.valid && ast_channel_connected(ast)->id.name.str) {
-		free(p->callerid_name);
-		p->callerid_name = strdup(ast_channel_connected(ast)->id.name.str);
+		ast_free(p->callerid_name);
+		p->callerid_name = ast_strdup(ast_channel_connected(ast)->id.name.str);
 	} else if (ast_channel_connected(ast)->id.number.valid && ast_channel_connected(ast)->id.number.str) {
-		free(p->callerid_name);
-		p->callerid_name = strdup(ast_channel_connected(ast)->id.number.str);
+		ast_free(p->callerid_name);
+		p->callerid_name = ast_strdup(ast_channel_connected(ast)->id.number.str);
 	} else {
 		ast_channel_connected(ast)->id.name.valid = 1;
-		free(ast_channel_connected(ast)->id.name.str);
-		ast_channel_connected(ast)->id.name.str = strdup(gCallerID);
-		free(p->callerid_name);
-		p->callerid_name = strdup(ast_channel_connected(ast)->id.name.str);
+		ast_free(ast_channel_connected(ast)->id.name.str);
+		ast_channel_connected(ast)->id.name.str = ast_strdup(gCallerID);
+		ast_free(p->callerid_name);
+		p->callerid_name = ast_strdup(ast_channel_connected(ast)->id.name.str);
 	}
 
 	/* Retrieve vars */
@@ -1044,7 +1045,7 @@ static int ooh323_call(struct ast_channel *ast, const char *dest, int timeout)
 	if ((val = pbx_builtin_getvar_helper(ast, "CALLER_H323DIALEDDIGITS"))) {
 		ast_copy_string(p->caller_dialedDigits, val, sizeof(p->caller_dialedDigits));
       		if(!p->callerid_num)
-			p->callerid_num = strdup(val);
+			p->callerid_num = ast_strdup(val);
 	}
 
 	if ((val = pbx_builtin_getvar_helper(ast, "CALLER_H323EMAIL"))) {
@@ -1165,7 +1166,7 @@ static int ooh323_answer(struct ast_channel *ast)
 	if (p) {
 
 		ast_mutex_lock(&p->lock);
-		callToken = (p->callToken ? strdup(p->callToken) : NULL);
+		callToken = (p->callToken ? ast_strdup(p->callToken) : NULL);
 		if (ast_channel_state(ast) != AST_STATE_UP) {
 			ast_channel_lock(ast);
 			if (!p->alertsent) {
@@ -1183,7 +1184,7 @@ static int ooh323_answer(struct ast_channel *ast)
 			ooAnswerCall(p->callToken);
 		}
 		if (callToken) {
-			free(callToken);
+			ast_free(callToken);
 		}
 		ast_mutex_unlock(&p->lock);
 	}
@@ -1290,7 +1291,7 @@ static int ooh323_indicate(struct ast_channel *ast, int condition, const void *d
 	if (!p) return -1;
 
 	ast_mutex_lock(&p->lock);
-	callToken = (p->callToken ? strdup(p->callToken) : NULL);
+	callToken = (p->callToken ? ast_strdup(p->callToken) : NULL);
 	ast_mutex_unlock(&p->lock);
 
 	if (!callToken) {
@@ -1456,7 +1457,7 @@ static int ooh323_indicate(struct ast_channel *ast, int condition, const void *d
 		ast_verb(0, "++++  ooh323_indicate %d on %s is %d\n", condition, callToken, res);
 	}
 
-   	free(callToken);
+   	ast_free(callToken);
 	return res;
 }
 
@@ -1862,25 +1863,25 @@ int ooh323_onReceivedSetup(ooCallData *call, Q931Message *pmsg)
 
 
 	if (call->remoteDisplayName) {
-		p->callerid_name = strdup(call->remoteDisplayName);
+		p->callerid_name = ast_strdup(call->remoteDisplayName);
 	}
 
 	if (ooCallGetCallingPartyNumber(call, number, OO_MAX_NUMBER_LENGTH) == OO_OK) {
-		p->callerid_num = strdup(number);
+		p->callerid_num = ast_strdup(number);
 	}
 
 	if (call->remoteAliases) {
 		for (alias = call->remoteAliases; alias; alias = alias->next) {
 			if (alias->type == T_H225AliasAddress_h323_ID) {
 				if (!p->callerid_name) {
-					p->callerid_name = strdup(alias->value);
+					p->callerid_name = ast_strdup(alias->value);
 				}
 				ast_copy_string(p->caller_h323id, alias->value, sizeof(p->caller_h323id));
 				}
          else if(alias->type == T_H225AliasAddress_dialedDigits)
          {
             if(!p->callerid_num)
-               p->callerid_num = strdup(alias->value);
+               p->callerid_num = ast_strdup(alias->value);
 				ast_copy_string(p->caller_dialedDigits, alias->value,
 															sizeof(p->caller_dialedDigits));
          }
@@ -1921,7 +1922,7 @@ int ooh323_onReceivedSetup(ooCallData *call, Q931Message *pmsg)
       	user = find_user(p->callerid_name, call->remoteIP);
       	if(user && (user->incominglimit == 0 || user->inUse < user->incominglimit)) {
 		ast_mutex_lock(&user->lock);
-		p->username = strdup(user->name);
+		p->username = ast_strdup(user->name);
  		p->neighbor.user = user->mUseIP ? ast_strdup(user->mIP) :
 						  ast_strdup(user->name);
 		ast_copy_string(p->context, user->context, sizeof(p->context));
@@ -1967,7 +1968,7 @@ int ooh323_onReceivedSetup(ooCallData *call, Q931Message *pmsg)
 		ast_mutex_unlock(&user->lock);
 	} else {
 	 if (!OO_TESTFLAG(p->flags,H323_DISABLEGK)) {
-		p->username = strdup(call->remoteIP);
+		p->username = ast_strdup(call->remoteIP);
 		p->directrtp = gDirectRTP;
 		p->earlydirect = gEarlyDirect;
 	} else {
@@ -2320,7 +2321,7 @@ int onCallCleared(ooCallData *call)
 		}
 		ast_mutex_unlock(&userl.lock);
 
-		free(user);
+		ast_free(user);
 	}
 
 	if (gH323Debug)
@@ -2352,13 +2353,13 @@ void ooh323_delete_peer(struct ooh323_peer *peer)
 			}
 		ast_mutex_unlock(&peerl.lock);
 
-      if(peer->h323id)   free(peer->h323id);
-      if(peer->email)    free(peer->email);
-      if(peer->url)      free(peer->url);
-      if(peer->e164)     free(peer->e164);
+		ast_free(peer->h323id);
+		ast_free(peer->email);
+		ast_free(peer->url);
+		ast_free(peer->e164);
 
-      ao2_cleanup(peer->cap);
-		free(peer);
+		ao2_cleanup(peer->cap);
+		ast_free(peer);
 	}
 
 	if (gH323Debug)
@@ -2725,7 +2726,7 @@ static int ooh323_do_reload(void)
 	if (gRasGkMode == RasUseSpecificGatekeeper ||
 		gRasGkMode == RasDiscoverGatekeeper) {
 		ooGkClientInit(gRasGkMode, (gRasGkMode == RasUseSpecificGatekeeper) ?
-								gGatekeeper : 0, 0);
+								gGatekeeper : 0, gRASIP, 0);
 		ooGkClientStart(gH323ep.gkClient);
 	}
 
@@ -2847,8 +2848,8 @@ int reload_config(int reload)
 		while (cur) {
 			prev = cur;
 	  		cur = cur->next;
-	  		free(prev->value);
-	  		free(prev);
+	  		ast_free(prev->value);
+	  		ast_free(prev);
 		}
 		gAliasList = NULL;
 		ooH323EpClearAllAliases();
@@ -2868,6 +2869,7 @@ int reload_config(int reload)
 	gTRCLVL = OOTRCLVLERR;
 	gRasGkMode = RasNoGatekeeper;
 	gGatekeeper[0] = '\0';
+	gRASIP[0] = '\0';
 	gRTPTimeout = 60;
 	gNat = FALSE;
 	gRTDRInterval = 0;
@@ -2963,7 +2965,7 @@ int reload_config(int reload)
 	  			ast_copy_string(gCallerID, v->value, sizeof(gCallerID));
 	 		}
 			pNewAlias->type =  T_H225AliasAddress_h323_ID;
-			pNewAlias->value = strdup(v->value);
+			pNewAlias->value = ast_strdup(v->value);
 			pNewAlias->next = gAliasList;
 			gAliasList = pNewAlias;
 			pNewAlias = NULL;
@@ -2984,7 +2986,7 @@ int reload_config(int reload)
 					return 1;
 				}
 				pNewAlias->type =  T_H225AliasAddress_dialedDigits;
-				pNewAlias->value = strdup(v->value);
+				pNewAlias->value = ast_strdup(v->value);
 				pNewAlias->next = gAliasList;
 				gAliasList = pNewAlias;
 				pNewAlias = NULL;
@@ -2999,7 +3001,7 @@ int reload_config(int reload)
 				return 1;
 			}
 			pNewAlias->type =  T_H225AliasAddress_email_ID;
-			pNewAlias->value = strdup(v->value);
+			pNewAlias->value = ast_strdup(v->value);
 			pNewAlias->next = gAliasList;
 			gAliasList = pNewAlias;
 			pNewAlias = NULL;
@@ -3028,6 +3030,9 @@ int reload_config(int reload)
 				gRasGkMode = RasUseSpecificGatekeeper;
             			ast_copy_string(gGatekeeper, v->value, sizeof(gGatekeeper));
 			}
+		} else if (!strcasecmp(v->name, "localras")) {
+		        ast_copy_string(gRASIP, v->value, sizeof(gRASIP));
+			ast_verb(3, "  == Setting RAS IP to %s\n", gRASIP);
 		} else if (!strcasecmp(v->name, "logfile")) {
          		ast_copy_string(gLogFile, v->value, sizeof(gLogFile));
 		} else if (!strcasecmp(v->name, "context")) {
@@ -3923,9 +3928,9 @@ static int load_module(void)
 
 		/* Gatekeeper */
 		if (gRasGkMode == RasUseSpecificGatekeeper)
-			ooGkClientInit(gRasGkMode, gGatekeeper, 0);
+			ooGkClientInit(gRasGkMode, gGatekeeper, gRASIP, 0);
 		else if (gRasGkMode == RasDiscoverGatekeeper)
-			ooGkClientInit(gRasGkMode, 0, 0);
+			ooGkClientInit(gRasGkMode, 0, gRASIP, 0);
 
 		/* Register callbacks */
 		ooH323EpSetH323Callbacks(h323Callbacks);
@@ -4012,7 +4017,7 @@ static void *do_monitor(void *data)
 			ooGkClientDestroy();
 			ast_verb(0, "Restart stopped gatekeeper client\n");
 			ooGkClientInit(gRasGkMode, (gRasGkMode == RasUseSpecificGatekeeper) ?
-									gGatekeeper : 0, 0);
+									gGatekeeper : 0, gRASIP, 0);
 			ooGkClientStart(gH323ep.gkClient);
 		}
 
@@ -4144,22 +4149,22 @@ int ooh323_destroy(struct ooh323_pvt *p)
 		}
 
 		if (cur->username) {
-			free(cur->username);
+			ast_free(cur->username);
 			cur->username = 0;
 		}
 
 		if (cur->host) {
-			free(cur->host);
+			ast_free(cur->host);
 			cur->host = 0;
 		}
 
 		if (cur->callerid_name) {
-			free(cur->callerid_name);
+			ast_free(cur->callerid_name);
 			cur->callerid_name = 0;
 		}
 
 		if (cur->callerid_num) {
-			free(cur->callerid_num);
+			ast_free(cur->callerid_num);
 			cur->callerid_num = 0;
 		}
 
@@ -4202,7 +4207,7 @@ int ooh323_destroy(struct ooh323_pvt *p)
 	  	user->inUse--;
 	  	ast_mutex_unlock(&user->lock);
 	  }
-	  free(cur->neighbor.user);
+	  ast_free(cur->neighbor.user);
 	 }
       } else {
 /* outgoing limit decrement here !!! */
@@ -4232,10 +4237,10 @@ int delete_peers()
 		cur = cur->next;
 
 		ast_mutex_destroy(&prev->lock);
-      if(prev->h323id)   free(prev->h323id);
-      if(prev->email)    free(prev->email);
-      if(prev->url)      free(prev->url);
-      if(prev->e164)     free(prev->e164);
+		ast_free(prev->h323id);
+		ast_free(prev->email);
+		ast_free(prev->url);
+		ast_free(prev->e164);
       if(prev->rtpmask) {
 		ast_mutex_lock(&prev->rtpmask->lock);
 		prev->rtpmask->inuse--;
@@ -4243,10 +4248,10 @@ int delete_peers()
 	 	if (prev->rtpmask->inuse == 0) {
 	  		regfree(&prev->rtpmask->regex);
 			ast_mutex_destroy(&prev->rtpmask->lock);
-	  		free(prev->rtpmask);
+	  		ast_free(prev->rtpmask);
       		}
       }
-		free(prev);
+		ast_free(prev);
 
 		if (cur == peerl.peers) {
 			break;
@@ -4274,11 +4279,11 @@ int delete_users()
 	 		if (prev->rtpmask->inuse == 0) {
 	  			regfree(&prev->rtpmask->regex);
 				ast_mutex_destroy(&prev->rtpmask->lock);
-	  			free(prev->rtpmask);
+	  			ast_free(prev->rtpmask);
       			}
       		}
       	ao2_cleanup(prev->cap);
-		free(prev);
+		ast_free(prev);
 		if (cur == userl.users) {
 			break;
 		}
@@ -4390,8 +4395,8 @@ static int unload_module(void)
 	while (cur) {
 	  prev = cur;
 	  cur = cur->next;
-	  free(prev->value);
-	  free(prev);
+	  ast_free(prev->value);
+	  ast_free(prev);
 	}
 	gAliasList = NULL;
 
@@ -4555,7 +4560,7 @@ static int ooh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance
 		memset(&p->redirip, 0, sizeof(p->redirip));
 	}
 
-	callToken = (p->callToken ? strdup(p->callToken) : NULL);
+	callToken = (p->callToken ? ast_strdup(p->callToken) : NULL);
 
 	if (!callToken) {
 		if (gH323Debug) {
@@ -4583,7 +4588,7 @@ static int ooh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance
 	}
 
 	ast_mutex_unlock(&p->lock);
-	free(callToken);
+	ast_free(callToken);
 	return 0;
 
 }
@@ -5236,9 +5241,10 @@ void ast_ooh323c_exit()
 #endif
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Objective Systems H323 Channel",
-		.support_level = AST_MODULE_SUPPORT_EXTENDED,
-			.load = load_module,
-			.unload = unload_module,
-			.reload = reload_module,
-			.load_pri = AST_MODPRI_CHANNEL_DRIVER
-			);
+	.support_level = AST_MODULE_SUPPORT_EXTENDED,
+	.load = load_module,
+	.unload = unload_module,
+	.reload = reload_module,
+	.load_pri = AST_MODPRI_CHANNEL_DRIVER,
+	.requires = "udptl",
+);

addons/chan_ooh323.h

@@ -38,7 +38,6 @@
 #include <errno.h>
 #include <fcntl.h>
 #include <signal.h>
-#include <sys/signal.h>
 
 #include "asterisk/lock.h"
 #include "asterisk/channel.h"

addons/format_mp3.c

@@ -34,8 +34,6 @@
 
 #include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
 #include "mp3/mpg123.h"
 #include "mp3/mpglib.h"