.gitreview

@@ -2,3 +2,4 @@
 host=gerrit.asterisk.org
 port=29418
 project=asterisk.git
+defaultbranch=13

.version

-13.18.5
\ No newline at end of file
+13.20.0
\ No newline at end of file

CHANGES

@@ -8,6 +8,105 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.19.0 to Asterisk 13.20.0 ----------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * During dialplan reload log messages are produced for each context,
+   extension and include.  These messages are no longer printed by the
+   verbose loggers, they are now only logged as debug messages.
+
+app_confbridge
+------------------
+ * Added the Muted header to the ConfbridgeJoin AMI event to indicate the
+   participant's starting mute status.
+
+ * Made the AMI ConfbridgeList action's ConfbridgeList events output all
+   the standard channel snapshot headers instead of a few hand-coded channel
+   snapshot headers.  The benefit is that the CallerIDName gets disruptive
+   characters like CR, LF, Tab, and a few others escaped.  However, an empty
+   CallerIDName is now output as "<unknown>" instead of "<no name>".
+
+app_followme
+------------------
+ * Added a new prompt, connecting-prompt, which will be played
+   (if configured) to the "winner" callee before connecting the call.
+
+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.
+
+ * The pjsip_transport_event feature introduced in 13.18.0 has been refactored.
+   Any external modules that may have used that feature (highly unlikey) will
+   need to be changed as the API has been altered slightly.
+
+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.
+
+res_pjsip_registrar_expire
+------------------
+ * The res_pjsip_registrar_expire module is now deprecated and will be removed
+   from Asterisk 16.  The existing functionality was moved into
+   res_pjsip_registrar.
+
+res_pjsip_pubsub
+------------------
+ * In an earlier release, inbound registrations on a reliable transport
+   were pruned on Asterisk restart since the TCP connection would have
+   been torn down and become unusable when Asterisk stopped.  This same
+   process is now also applied to inbound subscriptions.  Since this
+   required the addition of a new column to the ps_subscription_persistence
+   realtime table, users who store their subscriptions in a database will
+   need to run the "alembic upgrade head" process to add the column to
+   the schema.
+
+res_pjsip_transport_management
+------------------
+ * Since res_pjsip_transport_management provides several attack
+   mitigation features, its functionality moved to res_pjsip and
+   this module has been removed.  This way the features will always
+   be available if res_pjsip is loaded.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.18.0 to Asterisk 13.19.0 ----------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * Added the "cache_media_frames" option to asterisk.conf.  Disabling the option
+   helps track down media frame mismanagement when using valgrind or
+   MALLOC_DEBUG.  The cache gets in the way of determining if the frame is
+   used after free and who freed it.  NOTE: This option has no effect when
+   Asterisk is compiled with the LOW_MEMORY compile time option enabled because
+   the cache code does not exist.
+
+chan_sip
+------------------
+ * Calls to invalid extensions are now reported as an ACL failure security event
+   "no_extension_match".
+
+res_pjsip
+------------------
+ * The "identify_by" on endpoints can now be set to "ip" to restrict an endpoint
+   being matched based only on IP address. To ensure no behavior change the
+   default has been changed to "username,ip".
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.17.0 to Asterisk 13.18.0 ----------
 ------------------------------------------------------------------------------
@@ -308,7 +407,7 @@ ARI
    hasn't registered yet.
 
  * 'ari set debug' now displays requests and responses as well as events.
- 
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.12.0 to Asterisk 13.13.0 ----------
 ------------------------------------------------------------------------------
@@ -1190,7 +1289,7 @@ res_pjsip_endpoint_identifer_ip
 ------------------
  * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
    all configured PJSIP identify objects
-   
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
 ------------------------------------------------------------------------------
@@ -1458,7 +1557,7 @@ AMI
  * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
    endpoint as long as a default outbound endpoint is set. This also applies
    to the equivalent CLI command (pjsip send notify)
- 
+
  * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
    that give information on Asterisk's attempts to qualify the endpoint.
 
@@ -2801,7 +2900,7 @@ chan_agent
    has also been removed, for the same reason.
 
  * The endcall and enddtmf configuration options are removed.  Use the
-   dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
+   dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent
    channel before calling AgentLogin.
 
 chan_bridge

CREDITS

@@ -22,7 +22,7 @@
 	* John Todd, TalkPlus, Inc.  and JR Richardson, Ntegrated Solutions. 
 		for funding the development of SIP Session Timers support.
 
-	* Omnitor AB, Gunnar Hellstr�m, for funding work with videocaps, 
+	* Omnitor AB, Gunnar Hellström, for funding work with videocaps, 
 		T.140 RED, originate with video/text and many more 
 		contributions.
 
@@ -252,7 +252,7 @@
 		cdr_tds rewrite, countless other improvements, fixes, and good
 		ideas. sean(AT)malleable.com
 
-	* Jan Kal�b - Calendaring support for Exchange Server 2007+ via 
+	* Jan Kaláb - Calendaring support for Exchange Server 2007+ via 
 		Exchange Web Services.
 
 	* University of Oslo (uio.no), Norway - SIP Max-Forwards setting 

Makefile

@@ -104,8 +104,11 @@ export PATCH
 export SED
 export NM
 
-# makeopts is required unless the goal is clean or distclean
-ifeq ($(findstring clean,$(MAKECMDGOALS)),)
+# makeopts is required unless the goal is just {dist{-}}clean
+ifeq ($(MAKECMDGOALS),clean)
+else ifeq ($(MAKECMDGOALS),distclean)
+else ifeq ($(MAKECMDGOALS),dist-clean)
+else
 include makeopts
 endif
 
@@ -210,7 +213,9 @@ ifeq ($(AST_DEVMODE),yes)
   ifeq ($(AST_DEVMODE_STRICT),yes)
     _ASTCFLAGS+=-Wshadow
   endif
-  ADDL_TARGETS+=validate-docs
+  ifneq ($(DISABLE_XMLDOC),yes)
+    ADDL_TARGETS+=validate-docs
+  endif
 endif
 
 ifneq ($(findstring BSD,$(OSARCH)),)
@@ -248,6 +253,14 @@ ifneq ($(wildcard .svn),)
   ASTERISKVERSIONNUM:=999999
 endif
 
+ifneq ($(DISABLE_XMLDOC),yes)
+  CORE_XMLDOC=doc/core-en_US.xml
+  FULL_XMLDOC=doc/full-en_US.xml
+else
+  CORE_XMLDOC=
+  FULL_XMLDOC=
+endif
+
 _ASTCFLAGS+=$(OPTIONS)
 
 MOD_SUBDIRS:=third-party channels pbx apps codecs formats cdr cel bridges funcs tests main res addons $(LOCAL_MOD_SUBDIRS)
@@ -328,9 +341,9 @@ full: _full
 	@echo " +-------------------------------------------+"
 
 
-_all: makeopts $(SUBDIRS) doc/core-en_US.xml $(ADDL_TARGETS)
+_all: makeopts $(SUBDIRS) $(CORE_XMLDOC) $(ADDL_TARGETS)
 
-_full: makeopts $(SUBDIRS) doc/full-en_US.xml $(ADDL_TARGETS)
+_full: makeopts $(SUBDIRS) $(FULL_XMLDOC) $(ADDL_TARGETS)
 
 makeopts: configure
 	@echo "****"
@@ -425,7 +438,7 @@ distclean: $(SUBDIRS_DIST_CLEAN) _clean
 	rm -f doc/asterisk-ng-doxygen
 	rm -f build_tools/menuselect-deps
 
-datafiles: _all doc/core-en_US.xml
+datafiles: _all $(CORE_XMLDOC)
 	CFLAGS="$(_ASTCFLAGS) $(ASTCFLAGS)" build_tools/mkpkgconfig "$(DESTDIR)$(libdir)/pkgconfig";
 
 #	# Recursively install contents of the static-http directory, in case
@@ -437,8 +450,10 @@ datafiles: _all doc/core-en_US.xml
 			$(INSTALL) -m 644 $$x "$(DESTDIR)$(ASTDATADIR)/$$x" ; \
 		fi \
 	done
+ifneq ($(DISABLE_XMLDOC),yes)
 	$(INSTALL) -m 644 doc/core-en_US.xml "$(DESTDIR)$(ASTDATADIR)/static-http";
 	$(INSTALL) -m 644 doc/appdocsxml.xslt "$(DESTDIR)$(ASTDATADIR)/static-http";
+endif
 	if [ -d doc/tex/asterisk ] ; then \
 		$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/static-http/docs" ; \
 		for n in doc/tex/asterisk/* ; do \
@@ -567,9 +582,11 @@ bininstall: _all installdirs $(SUBDIRS_INSTALL) main-bininstall
 		for h in $(OLDHEADERS); do rm -f "$(DESTDIR)$(ASTHEADERDIR)/$$h"; done \
 	fi
 
+ifneq ($(DISABLE_XMLDOC),yes)
 	$(INSTALL) -m 644 doc/core-*.xml "$(DESTDIR)$(ASTDATADIR)/documentation"
 	$(INSTALL) -m 644 doc/appdocsxml.xslt "$(DESTDIR)$(ASTDATADIR)/documentation"
 	$(INSTALL) -m 644 doc/appdocsxml.dtd "$(DESTDIR)$(ASTDATADIR)/documentation"
+endif
 	$(INSTALL) -m 644 doc/asterisk.8 "$(DESTDIR)$(ASTMANDIR)/man8"
 	$(INSTALL) -m 644 doc/astdb*.8 "$(DESTDIR)$(ASTMANDIR)/man8"
 	$(INSTALL) -m 644 contrib/scripts/astgenkey.8 "$(DESTDIR)$(ASTMANDIR)/man8"
@@ -588,7 +605,12 @@ $(SUBDIRS_INSTALL):
 
 NEWMODS:=$(foreach d,$(MOD_SUBDIRS),$(notdir $(wildcard $(d)/*.so)))
 OLDMODS=$(filter-out $(NEWMODS) $(notdir $(DESTDIR)$(ASTMODDIR)),$(notdir $(wildcard $(DESTDIR)$(ASTMODDIR)/*.so)))
-BADMODS=$(strip $(filter-out $(shell ./build_tools/list_valid_installed_externals),$(OLDMODS)))
+ifneq ($(BASH),:)
+	FILMODS=$(filter-out $(shell ./build_tools/list_valid_installed_externals),$(OLDMODS))
+else
+	FILMODS=$(OLDMODS)
+endif
+BADMODS=$(strip $(FILMODS))
 
 oldmodcheck:
 	@if [ -n "$(BADMODS)" ]; then \
@@ -611,7 +633,7 @@ oldmodcheck:
 ld-cache-update:
 ifneq ($(LDCONFIG),)
 ifeq ($(DESTDIR),)  # DESTDIR means binary archive creation; ldconfig should be run on postinst
-	@if [ $${EUID} -eq 0 ] ; then \
+	@if [ $$(id -u) -eq 0 ] ; then \
 		$(LDCONFIG) "$(ASTLIBDIR)/" ; \
 	else \
 		echo " WARNING WARNING WARNING" ;\
@@ -956,7 +978,7 @@ ifeq ($(HAVE_DAHDI),1)
 endif
 	$(MAKE) -C sounds uninstall
 ifneq ($(LDCONFIG),)
-	$(LDCONFIG) || :
+	$(LDCONFIG) "$(ASTLIBDIR)/" || :
 endif
 
 uninstall: _uninstall
@@ -1076,11 +1098,9 @@ check-alembic: makeopts
 .PHONY: _all
 .PHONY: full
 .PHONY: _full
-.PHONY: prereqs
 .PHONY: uninstall
 .PHONY: _uninstall
 .PHONY: uninstall-all
-.PHONY: dont-optimize
 .PHONY: badshell
 .PHONY: installdirs
 .PHONY: validate-docs
@@ -1098,3 +1118,5 @@ check-alembic: makeopts
 
 FORCE:
 
+# This only stops targets within the root Makefile from building in parallel.
+.NOTPARALLEL:

Makefile.moddir_rules

@@ -38,7 +38,7 @@ ifneq ($(MODULE_PREFIX),)
     ALL_C_MODS+=$(foreach p,$(MODULE_PREFIX),$(patsubst %.c,%,$(wildcard $(p)_*.c)))
     ALL_CC_MODS+=$(foreach p,$(MODULE_PREFIX),$(patsubst %.cc,%,$(wildcard $(p)_*.cc)))
 endif
- 
+
 C_MODS:=$(filter-out $(MENUSELECT_$(MENUSELECT_CATEGORY)),$(ALL_C_MODS))
 CC_MODS:=$(filter-out $(MENUSELECT_$(MENUSELECT_CATEGORY)),$(ALL_CC_MODS))
 
@@ -93,7 +93,7 @@ ifneq ($(findstring :,$(XMLSTARLET)$(BASH)),:)
 	@if [ -f .moduleinfo ] ; then \
 		declare -A DISABLED_MODS ;\
 		for x in $(MENUSELECT_$(MENUSELECT_CATEGORY)) ; do DISABLED_MODS[$${x}]=1 ; done ;\
-		EXTERNAL_MODS=$$(xmlstarlet sel -t -m "/category/member[support_level = 'external']" -v "@name" -n .moduleinfo) ;\
+		EXTERNAL_MODS=$$($(XMLSTARLET) sel -t -m "/category/member[support_level = 'external']" -v "@name" -n .moduleinfo) ;\
 		for x in $${EXTERNAL_MODS} ; do \
 			if [ -z "$${DISABLED_MODS[$${x}]}" ] ; then \
 				$(ASTTOPDIR)/build_tools/download_externals $${x} ;\

Makefile.rules

 #
 # Asterisk -- An open source telephony toolkit.
-# 
+#
 # Makefile rules
 #
 # Copyright (C) 2006-2010, Digium, Inc.
@@ -55,7 +55,7 @@ ifeq ($(NOISY_BUILD),)
     ECHO_PREFIX=@
     CMD_PREFIX=@
 else
-    ECHO_PREFIX=@\# 
+    ECHO_PREFIX=@\#
     CMD_PREFIX=
 endif
 

README-addons.txt

@@ -9,18 +9,18 @@ potential licensing and/or patent implications that has on your usage and
 distribution of Asterisk.
 
     Even though Asterisk is released as open source under the terms of the
-GPLv2 (see LICENSE for details), no core functionality in Asterisk has any 
+GPLv2 (see LICENSE for details), no core functionality in Asterisk has any
 dependencies on libraries that are licensed under the GPL.  One reason a module
 may be in the add-ons category is that it may have a GPL dependency.  Since
 these dependencies are not compatible with dual licensing of Asterisk, the
-dependant modules are set aside to make it clear that they may not be used 
-with commercial versions of Asterisk, unless other licensing arrangements are 
+dependant modules are set aside to make it clear that they may not be used
+with commercial versions of Asterisk, unless other licensing arrangements are
 made with the copyright holders of those dependencies.
 
     Another reason that modules may be set aside is that there may be
 additional restrictions on the usage of the code imposed by the license or
 related patents.  The MySQL and MP3 modules are examples of this.
-       
+
     If you have any questions, contact your lawyer.
 
 ===============================================================================

UPGRADE-1.2.txt

@@ -37,9 +37,9 @@ Dialplan Expressions:
 
 Agents:
 
-* The default for ackcall has been changed to "no" instead of "yes" 
+* The default for ackcall has been changed to "no" instead of "yes"
   because of a bug which caused the "yes" behavior to generally act like
-  "no".  You may need to adjust the value if your agents behave 
+  "no".  You may need to adjust the value if your agents behave
   differently than you expect with respect to acknowledgement.
 
 * The AgentCallBackLogin application now requires a second '|' before
@@ -56,16 +56,16 @@ Parking:
 
 Dialing:
 
-* The Caller*ID of the outbound leg is now the extension that was 
-  called, rather than the Caller*ID of the inbound leg of the call.  The 
-  "o" flag for Dial can be used to restore the original behavior if 
+* The Caller*ID of the outbound leg is now the extension that was
+  called, rather than the Caller*ID of the inbound leg of the call.  The
+  "o" flag for Dial can be used to restore the original behavior if
   desired.  Note that if you are looking for the originating callerid
-  from the manager event, there is a new manager event "Dial" which 
+  from the manager event, there is a new manager event "Dial" which
   provides the source and destination channels and callerid.
 
-IAX: 
+IAX:
 
-* The naming convention for IAX channels has changed in two ways: 
+* The naming convention for IAX channels has changed in two ways:
    1. The call number follows a "-" rather than a "/" character.
    2. The name of the channel has been simplified to IAX2/peer-callno,
    rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
@@ -77,7 +77,7 @@ SIP:
   the other channel drivers and to avoid confusion with the "port"
   option for users/peers.
 
-* The "Registry" event now uses "Username" rather than "User" for 
+* The "Registry" event now uses "Username" rather than "User" for
   consistency with IAX.
 
 Applications:
@@ -85,7 +85,7 @@ Applications:
 * With the addition of dialplan functions (which operate similarly
   to variables), the SetVar application has been renamed to Set.
 
-* The CallerPres application has been removed.  Use SetCallerPres 
+* The CallerPres application has been removed.  Use SetCallerPres
   instead.  It accepts both numeric and symbolic names.
 
 * The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
@@ -145,7 +145,7 @@ Applications:
   moved to option 5.
 
 * The application VoiceMailMain now only matches the 'default' context if
-  none is specified in the arguments.  (This was the previously 
+  none is specified in the arguments.  (This was the previously
   documented behavior, however, we didn't follow that behavior.)  The old
   behavior can be restored by setting searchcontexts=yes in voicemail.conf.
 
@@ -153,7 +153,7 @@ Queues:
 
 * A queue is now considered empty not only if there are no members but if
   none of the members are available (e.g. agents not logged on).  To
-  restore the original behavior, use "leavewhenempty=strict" or 
+  restore the original behavior, use "leavewhenempty=strict" or
   "joinwhenempty=strict" instead of "=yes" for those options.
 
 * It is now possible to use multi-digit extensions in the exit context
@@ -165,8 +165,8 @@ Queues:
 Extensions:
 
 * By default, there is a new option called "autofallthrough" in
-  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier) 
-  behavior was to wait for an extension to be dialed after there were no 
+  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier)
+  behavior was to wait for an extension to be dialed after there were no
   more extensions to execute.  "autofallthrough" changes this behavior
   so that the call will immediately be terminated with BUSY,
   CONGESTION, or HANGUP based on Asterisk's best guess.  If you are
@@ -175,9 +175,9 @@ Extensions:
 
 AGI:
 
-* AGI scripts did not always get SIGHUP at the end, previously.  That 
-  behavior has been fixed.  If you do not want your script to terminate 
-  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to 
+* AGI scripts did not always get SIGHUP at the end, previously.  That
+  behavior has been fixed.  If you do not want your script to terminate
+  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
   be ignored within your application.
 
 * CallerID is reported with agi_callerid and agi_calleridname instead
@@ -199,13 +199,13 @@ chan_modem:
 MeetMe:
 
 * The conference application now allows users to increase/decrease their
-  speaking volume and listening volume (independently of each other and 
-  other users); the 'admin' and 'user' menus have changed, and new sound 
-  files are included with this release. However, if a user calling in 
-  over a Zaptel channel that does NOT have hardware DTMF detection 
-  increases their speaking volume, it is likely they will no longer be 
-  able to enter/exit the menu or make any further adjustments, as the  
-  software DTMF detector will not be able to recognize the DTMF coming 
+  speaking volume and listening volume (independently of each other and
+  other users); the 'admin' and 'user' menus have changed, and new sound
+  files are included with this release. However, if a user calling in
+  over a Zaptel channel that does NOT have hardware DTMF detection
+  increases their speaking volume, it is likely they will no longer be
+  able to enter/exit the menu or make any further adjustments, as the
+  software DTMF detector will not be able to recognize the DTMF coming
   from their device.
 
 GetVar Manager Action:

UPGRADE-10.txt

@@ -79,10 +79,10 @@ Asterisk Database:
    convert an existing astdb to the SQLite3 version automatically at runtime.
 
 Module Support Level
- - All modules in the addons, apps, bridge, cdr, cel, channels, codecs, 
+ - All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
    formats, funcs, pbx, and res have been updated to include MODULEINFO data
    that includes <support_level> tags with a value of core, extended, or deprecated.
-   More information is available on the Asterisk wiki at 
+   More information is available on the Asterisk wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
 
    Deprecated modules are now marked to not build by default and must be explicitly

UPGRADE-11.txt

@@ -185,7 +185,7 @@ SIP
    configuration option. Symptoms of this include one way media or no media flow.
 
 chan_unistim
- - Due to massive update in chan_unistim phone keys functions and on-screen 
+ - Due to massive update in chan_unistim phone keys functions and on-screen
    information changed.
 
 users.conf:
@@ -257,10 +257,10 @@ Manager:
    unchanged.
 
 Module Support Level
- - All modules in the addons, apps, bridge, cdr, cel, channels, codecs, 
+ - All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
    formats, funcs, pbx, and res have been updated to include MODULEINFO data
    that includes <support_level> tags with a value of core, extended, or deprecated.
-   More information is available on the Asterisk wiki at 
+   More information is available on the Asterisk wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
 
    Deprecated modules are now marked to not build by default and must be explicitly