[svn-commits] oej: branch oej/pinefrog-rtcp-11 r415190 - in /team/oej/pinefrog-rtcp-11: ./ ...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jun 5 04:20:10 CDT 2014
Author: oej
Date: Thu Jun 5 04:19:59 2014
New Revision: 415190
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=415190
Log:
Resetting branch
Added:
team/oej/pinefrog-rtcp-11/doc/astdb2bdb.8
- copied unchanged from r415171, branches/11/doc/astdb2bdb.8
team/oej/pinefrog-rtcp-11/doc/astdb2sqlite3.8
- copied unchanged from r415171, branches/11/doc/astdb2sqlite3.8
Modified:
team/oej/pinefrog-rtcp-11/ (props changed)
team/oej/pinefrog-rtcp-11/CHANGES
team/oej/pinefrog-rtcp-11/Makefile
team/oej/pinefrog-rtcp-11/README-SERIOUSLY.bestpractices.txt
team/oej/pinefrog-rtcp-11/UPGRADE.txt
team/oej/pinefrog-rtcp-11/addons/chan_ooh323.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCalls.h
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCapability.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCmdChannel.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooGkClient.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooGkClient.h
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooTimer.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/oochannels.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooh245.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooh323.c
team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooq931.c
team/oej/pinefrog-rtcp-11/addons/res_config_mysql.c
team/oej/pinefrog-rtcp-11/apps/app_adsiprog.c
team/oej/pinefrog-rtcp-11/apps/app_chanspy.c
team/oej/pinefrog-rtcp-11/apps/app_confbridge.c
team/oej/pinefrog-rtcp-11/apps/app_dial.c
team/oej/pinefrog-rtcp-11/apps/app_dumpchan.c
team/oej/pinefrog-rtcp-11/apps/app_festival.c
team/oej/pinefrog-rtcp-11/apps/app_forkcdr.c
team/oej/pinefrog-rtcp-11/apps/app_getcpeid.c
team/oej/pinefrog-rtcp-11/apps/app_jack.c
team/oej/pinefrog-rtcp-11/apps/app_meetme.c
team/oej/pinefrog-rtcp-11/apps/app_minivm.c
team/oej/pinefrog-rtcp-11/apps/app_mixmonitor.c
team/oej/pinefrog-rtcp-11/apps/app_playback.c
team/oej/pinefrog-rtcp-11/apps/app_queue.c
team/oej/pinefrog-rtcp-11/apps/app_saycounted.c (props changed)
team/oej/pinefrog-rtcp-11/apps/app_sms.c
team/oej/pinefrog-rtcp-11/apps/app_speech_utils.c
team/oej/pinefrog-rtcp-11/apps/app_stack.c
team/oej/pinefrog-rtcp-11/apps/app_transfer.c
team/oej/pinefrog-rtcp-11/apps/app_verbose.c
team/oej/pinefrog-rtcp-11/apps/app_voicemail.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_config_parser.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_state.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_state_empty.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_state_inactive.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_state_multi.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_state_multi_marked.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_state_single.c
team/oej/pinefrog-rtcp-11/apps/confbridge/conf_state_single_marked.c
team/oej/pinefrog-rtcp-11/apps/confbridge/include/confbridge.h
team/oej/pinefrog-rtcp-11/autoconf/ast_ext_lib.m4
team/oej/pinefrog-rtcp-11/bridges/bridge_softmix.c
team/oej/pinefrog-rtcp-11/build_tools/cflags.xml
team/oej/pinefrog-rtcp-11/build_tools/menuselect-deps.in
team/oej/pinefrog-rtcp-11/build_tools/prep_tarball
team/oej/pinefrog-rtcp-11/build_tools/sha1sum-sh (props changed)
team/oej/pinefrog-rtcp-11/cel/cel_custom.c
team/oej/pinefrog-rtcp-11/cel/cel_manager.c
team/oej/pinefrog-rtcp-11/cel/cel_odbc.c
team/oej/pinefrog-rtcp-11/cel/cel_pgsql.c
team/oej/pinefrog-rtcp-11/cel/cel_radius.c
team/oej/pinefrog-rtcp-11/channels/chan_alsa.c
team/oej/pinefrog-rtcp-11/channels/chan_dahdi.c
team/oej/pinefrog-rtcp-11/channels/chan_gtalk.c
team/oej/pinefrog-rtcp-11/channels/chan_h323.c
team/oej/pinefrog-rtcp-11/channels/chan_iax2.c
team/oej/pinefrog-rtcp-11/channels/chan_jingle.c
team/oej/pinefrog-rtcp-11/channels/chan_local.c
team/oej/pinefrog-rtcp-11/channels/chan_mgcp.c
team/oej/pinefrog-rtcp-11/channels/chan_misdn.c
team/oej/pinefrog-rtcp-11/channels/chan_motif.c
team/oej/pinefrog-rtcp-11/channels/chan_multicast_rtp.c (props changed)
team/oej/pinefrog-rtcp-11/channels/chan_oss.c
team/oej/pinefrog-rtcp-11/channels/chan_phone.c
team/oej/pinefrog-rtcp-11/channels/chan_sip.c
team/oej/pinefrog-rtcp-11/channels/chan_skinny.c
team/oej/pinefrog-rtcp-11/channels/chan_unistim.c
team/oej/pinefrog-rtcp-11/channels/iax2-parser.c
team/oej/pinefrog-rtcp-11/channels/misdn/isdn_msg_parser.c
team/oej/pinefrog-rtcp-11/channels/sig_analog.c
team/oej/pinefrog-rtcp-11/channels/sig_pri.c
team/oej/pinefrog-rtcp-11/channels/sig_pri.h
team/oej/pinefrog-rtcp-11/channels/sig_ss7.c (contents, props changed)
team/oej/pinefrog-rtcp-11/channels/sig_ss7.h (props changed)
team/oej/pinefrog-rtcp-11/channels/sip/config_parser.c
team/oej/pinefrog-rtcp-11/channels/sip/dialplan_functions.c
team/oej/pinefrog-rtcp-11/channels/sip/include/security_events.h (props changed)
team/oej/pinefrog-rtcp-11/channels/sip/include/sip.h
team/oej/pinefrog-rtcp-11/channels/sip/reqresp_parser.c
team/oej/pinefrog-rtcp-11/channels/sip/security_events.c (contents, props changed)
team/oej/pinefrog-rtcp-11/configure
team/oej/pinefrog-rtcp-11/configure.ac
team/oej/pinefrog-rtcp-11/default.exports
team/oej/pinefrog-rtcp-11/doc/Makefile (props changed)
team/oej/pinefrog-rtcp-11/doc/asterisk.8
team/oej/pinefrog-rtcp-11/funcs/func_audiohookinherit.c
team/oej/pinefrog-rtcp-11/funcs/func_blacklist.c
team/oej/pinefrog-rtcp-11/funcs/func_callcompletion.c
team/oej/pinefrog-rtcp-11/funcs/func_callerid.c
team/oej/pinefrog-rtcp-11/funcs/func_channel.c
team/oej/pinefrog-rtcp-11/funcs/func_config.c
team/oej/pinefrog-rtcp-11/funcs/func_curl.c
team/oej/pinefrog-rtcp-11/funcs/func_db.c
team/oej/pinefrog-rtcp-11/funcs/func_dialgroup.c
team/oej/pinefrog-rtcp-11/funcs/func_dialplan.c
team/oej/pinefrog-rtcp-11/funcs/func_env.c
team/oej/pinefrog-rtcp-11/funcs/func_frame_trace.c
team/oej/pinefrog-rtcp-11/funcs/func_global.c
team/oej/pinefrog-rtcp-11/funcs/func_groupcount.c
team/oej/pinefrog-rtcp-11/funcs/func_hangupcause.c
team/oej/pinefrog-rtcp-11/funcs/func_iconv.c
team/oej/pinefrog-rtcp-11/funcs/func_jitterbuffer.c
team/oej/pinefrog-rtcp-11/funcs/func_lock.c
team/oej/pinefrog-rtcp-11/funcs/func_math.c
team/oej/pinefrog-rtcp-11/funcs/func_odbc.c
team/oej/pinefrog-rtcp-11/funcs/func_pitchshift.c
team/oej/pinefrog-rtcp-11/funcs/func_presencestate.c
team/oej/pinefrog-rtcp-11/funcs/func_realtime.c
team/oej/pinefrog-rtcp-11/funcs/func_shell.c
team/oej/pinefrog-rtcp-11/funcs/func_speex.c
team/oej/pinefrog-rtcp-11/funcs/func_srv.c
team/oej/pinefrog-rtcp-11/funcs/func_strings.c
team/oej/pinefrog-rtcp-11/funcs/func_sysinfo.c
team/oej/pinefrog-rtcp-11/funcs/func_timeout.c
team/oej/pinefrog-rtcp-11/funcs/func_volume.c
team/oej/pinefrog-rtcp-11/include/asterisk.h
team/oej/pinefrog-rtcp-11/include/asterisk/astmm.h
team/oej/pinefrog-rtcp-11/include/asterisk/astobj.h
team/oej/pinefrog-rtcp-11/include/asterisk/astobj2.h
team/oej/pinefrog-rtcp-11/include/asterisk/autoconfig.h.in
team/oej/pinefrog-rtcp-11/include/asterisk/bridging_features.h (props changed)
team/oej/pinefrog-rtcp-11/include/asterisk/bridging_technology.h (props changed)
team/oej/pinefrog-rtcp-11/include/asterisk/channel.h
team/oej/pinefrog-rtcp-11/include/asterisk/devicestate.h
team/oej/pinefrog-rtcp-11/include/asterisk/frame.h
team/oej/pinefrog-rtcp-11/include/asterisk/lock.h
team/oej/pinefrog-rtcp-11/include/asterisk/logger.h
team/oej/pinefrog-rtcp-11/include/asterisk/message.h
team/oej/pinefrog-rtcp-11/include/asterisk/options.h
team/oej/pinefrog-rtcp-11/include/asterisk/pbx.h
team/oej/pinefrog-rtcp-11/include/asterisk/res_odbc.h
team/oej/pinefrog-rtcp-11/include/asterisk/rtp_engine.h
team/oej/pinefrog-rtcp-11/include/asterisk/select.h (props changed)
team/oej/pinefrog-rtcp-11/include/asterisk/test.h
team/oej/pinefrog-rtcp-11/include/asterisk/utils.h
team/oej/pinefrog-rtcp-11/main/Makefile
team/oej/pinefrog-rtcp-11/main/abstract_jb.c
team/oej/pinefrog-rtcp-11/main/acl.c
team/oej/pinefrog-rtcp-11/main/app.c
team/oej/pinefrog-rtcp-11/main/asterisk.c
team/oej/pinefrog-rtcp-11/main/asterisk.exports.in
team/oej/pinefrog-rtcp-11/main/astfd.c
team/oej/pinefrog-rtcp-11/main/astmm.c
team/oej/pinefrog-rtcp-11/main/astobj2.c
team/oej/pinefrog-rtcp-11/main/autoservice.c
team/oej/pinefrog-rtcp-11/main/bridging.c
team/oej/pinefrog-rtcp-11/main/ccss.c
team/oej/pinefrog-rtcp-11/main/cdr.c
team/oej/pinefrog-rtcp-11/main/channel.c
team/oej/pinefrog-rtcp-11/main/cli.c
team/oej/pinefrog-rtcp-11/main/config.c
team/oej/pinefrog-rtcp-11/main/data.c
team/oej/pinefrog-rtcp-11/main/db.c
team/oej/pinefrog-rtcp-11/main/devicestate.c
team/oej/pinefrog-rtcp-11/main/dsp.c
team/oej/pinefrog-rtcp-11/main/editline/readline.c
team/oej/pinefrog-rtcp-11/main/editline/term.c
team/oej/pinefrog-rtcp-11/main/enum.c
team/oej/pinefrog-rtcp-11/main/event.c
team/oej/pinefrog-rtcp-11/main/features.c
team/oej/pinefrog-rtcp-11/main/format_pref.c
team/oej/pinefrog-rtcp-11/main/frame.c
team/oej/pinefrog-rtcp-11/main/heap.c
team/oej/pinefrog-rtcp-11/main/http.c
team/oej/pinefrog-rtcp-11/main/indications.c
team/oej/pinefrog-rtcp-11/main/io.c
team/oej/pinefrog-rtcp-11/main/jitterbuf.c
team/oej/pinefrog-rtcp-11/main/loader.c
team/oej/pinefrog-rtcp-11/main/lock.c
team/oej/pinefrog-rtcp-11/main/logger.c
team/oej/pinefrog-rtcp-11/main/manager.c
team/oej/pinefrog-rtcp-11/main/message.c
team/oej/pinefrog-rtcp-11/main/pbx.c
team/oej/pinefrog-rtcp-11/main/say.c
team/oej/pinefrog-rtcp-11/main/sched.c
team/oej/pinefrog-rtcp-11/main/security_events.c
team/oej/pinefrog-rtcp-11/main/slinfactory.c
team/oej/pinefrog-rtcp-11/main/stdtime/localtime.c
team/oej/pinefrog-rtcp-11/main/taskprocessor.c
team/oej/pinefrog-rtcp-11/main/test.c
team/oej/pinefrog-rtcp-11/main/threadstorage.c
team/oej/pinefrog-rtcp-11/main/translate.c
team/oej/pinefrog-rtcp-11/main/udptl.c
team/oej/pinefrog-rtcp-11/main/utils.c
team/oej/pinefrog-rtcp-11/pbx/dundi-parser.c
team/oej/pinefrog-rtcp-11/pbx/pbx_config.c
team/oej/pinefrog-rtcp-11/pbx/pbx_dundi.c
team/oej/pinefrog-rtcp-11/pbx/pbx_loopback.c
team/oej/pinefrog-rtcp-11/pbx/pbx_lua.c
team/oej/pinefrog-rtcp-11/pbx/pbx_spool.c
team/oej/pinefrog-rtcp-11/sounds/Makefile
team/oej/pinefrog-rtcp-11/sounds/sounds.xml
team/oej/pinefrog-rtcp-11/tests/test_dlinklists.c
team/oej/pinefrog-rtcp-11/tests/test_expr.c (props changed)
team/oej/pinefrog-rtcp-11/tests/test_func_file.c (props changed)
team/oej/pinefrog-rtcp-11/tests/test_hashtab_thrash.c
team/oej/pinefrog-rtcp-11/tests/test_linkedlists.c
team/oej/pinefrog-rtcp-11/tests/test_locale.c (props changed)
team/oej/pinefrog-rtcp-11/tests/test_poll.c (props changed)
team/oej/pinefrog-rtcp-11/tests/test_substitution.c
team/oej/pinefrog-rtcp-11/tests/test_voicemail_api.c
Propchange: team/oej/pinefrog-rtcp-11/
('svnmerge-integrated' removed)
Modified: team/oej/pinefrog-rtcp-11/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-11/CHANGES?view=diff&rev=415190&r1=415189&r2=415190
==============================================================================
--- team/oej/pinefrog-rtcp-11/CHANGES (original)
+++ team/oej/pinefrog-rtcp-11/CHANGES Thu Jun 5 04:19:59 2014
@@ -7,6 +7,30 @@
=== and the other UPGRADE files for older releases.
===
==============================================================================
+
+------------------------------------------------------------------------------
+--- Functionality changes since Asterisk 11.8 --------------------------------
+------------------------------------------------------------------------------
+
+chan_sip
+-----------
+ * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
+ fields for prohibited callingpres information. Values are legacy, no, and
+ yes. By default, legacy is used.
+ trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
+ dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
+ headers are appended to outbound SIP messages just as they are with
+ allowed callingpres values, but data about the remote party's identity is
+ anonymized.
+ When sendrpid=rpid, only the remote party's domain is anonymized.
+ trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
+ headers are not sent.
+ trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
+ party information in tact even for prohibited callingpres information.
+ In the case of PAI, a Privacy: id header will be appended for prohibited
+ calling information to communicate that the private information should
+ not be relayed to untrusted parties.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------
Modified: team/oej/pinefrog-rtcp-11/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-11/Makefile?view=diff&rev=415190&r1=415189&r2=415190
==============================================================================
--- team/oej/pinefrog-rtcp-11/Makefile (original)
+++ team/oej/pinefrog-rtcp-11/Makefile Thu Jun 5 04:19:59 2014
@@ -360,15 +360,19 @@
+@$(SUBMAKE) $(MOD_SUBDIRS_EMBED_LDFLAGS)
+@$(SUBMAKE) $(MOD_SUBDIRS_EMBED_LIBS)
-$(SUBDIRS): makeopts cleantest main/version.c include/asterisk/build.h include/asterisk/buildopts.h defaults.h makeopts.embed_rules
+$(SUBDIRS): makeopts .lastclean main/version.c include/asterisk/build.h include/asterisk/buildopts.h defaults.h makeopts.embed_rules
ifeq ($(findstring $(OSARCH), mingw32 cygwin ),)
+ ifeq ($(shell grep ^MENUSELECT_EMBED=$$ menuselect.makeopts 2>/dev/null),)
# Non-windows:
# ensure that all module subdirectories are processed before 'main' during
# a parallel build, since if there are modules selected to be embedded the
# directories containing them must be completed before the main Asterisk
- # binary can be built
+ # binary can be built.
+ # If MENUSELECT_EMBED is empty, we don't need this and allow 'main' to be
+ # be built without building all dependencies first.
main: $(filter-out main,$(MOD_SUBDIRS))
+ endif
else
# Windows: we need to build main (i.e. the asterisk dll) first,
# followed by res, followed by the other directories, because
@@ -386,25 +390,25 @@
$(OTHER_SUBDIRS): makeopts
+ at _ASTCFLAGS="$(OTHER_SUBDIR_CFLAGS) $(_ASTCFLAGS)" ASTCFLAGS="$(ASTCFLAGS)" _ASTLDFLAGS="$(_ASTLDFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" $(SUBMAKE) --no-builtin-rules -C $@ SUBDIR=$@ all
-defaults.h: makeopts build_tools/make_defaults_h
+defaults.h: makeopts .lastclean build_tools/make_defaults_h
@build_tools/make_defaults_h > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
-main/version.c: FORCE
+main/version.c: FORCE .lastclean
@build_tools/make_version_c > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
-include/asterisk/buildopts.h: menuselect.makeopts
+include/asterisk/buildopts.h: menuselect.makeopts .lastclean
@build_tools/make_buildopts_h > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
-include/asterisk/build.h:
- @build_tools/make_build_h > $@.tmp
- @cmp -s $@.tmp $@ || mv $@.tmp $@
- @rm -f $@.tmp
+# build.h must depend on .lastclean, or parallel make may wipe it out after it's
+# been created.
+include/asterisk/build.h: .lastclean
+ @build_tools/make_build_h > $@
$(SUBDIRS_CLEAN):
+@$(SUBMAKE) -C $(@:-clean=) clean
@@ -458,7 +462,7 @@
done
$(MAKE) -C sounds install
-doc/core-en_US.xml: makeopts $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
+doc/core-en_US.xml: makeopts .lastclean $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
@printf "Building Documentation For: "
@echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
@echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
@@ -472,7 +476,7 @@
@echo
@echo "</docs>" >> $@
-doc/full-en_US.xml: makeopts $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
+doc/full-en_US.xml: makeopts .lastclean $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
ifeq ($(PYTHON),:)
@echo "--------------------------------------------------------------------------"
@echo "--- Please install python to build full documentation ---"
@@ -564,6 +568,7 @@
$(INSTALL) -m 644 doc/core-*.xml "$(DESTDIR)$(ASTDATADIR)/documentation"
$(INSTALL) -m 644 doc/appdocsxml.dtd "$(DESTDIR)$(ASTDATADIR)/documentation"
$(INSTALL) -m 644 doc/asterisk.8 "$(DESTDIR)$(ASTMANDIR)/man8"
+ $(INSTALL) -m 644 doc/astdb*.8 "$(DESTDIR)$(ASTMANDIR)/man8"
$(INSTALL) -m 644 contrib/scripts/astgenkey.8 "$(DESTDIR)$(ASTMANDIR)/man8"
$(INSTALL) -m 644 contrib/scripts/autosupport.8 "$(DESTDIR)$(ASTMANDIR)/man8"
$(INSTALL) -m 644 contrib/scripts/safe_asterisk.8 "$(DESTDIR)$(ASTMANDIR)/man8"
@@ -822,8 +827,8 @@
# .cleancount is the global clean count, and .lastclean is the
# last clean count we had
-cleantest:
- @cmp -s .cleancount .lastclean || $(MAKE) clean
+.lastclean: .cleancount
+ @$(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")
$(SUBDIRS_UNINSTALL):
@@ -901,19 +906,19 @@
CFLAGS="$(BUILD_CFLAGS)" LDFLAGS="$(BUILD_LDFLAGS)" \
$(MAKE) -C menuselect CONFIGURE_SILENT="--silent"
-menuselect/menuselect: menuselect/makeopts cleantest
+menuselect/menuselect: menuselect/makeopts .lastclean
+$(MAKE_MENUSELECT) menuselect
-menuselect/cmenuselect: menuselect/makeopts cleantest
+menuselect/cmenuselect: menuselect/makeopts .lastclean
+$(MAKE_MENUSELECT) cmenuselect
-menuselect/gmenuselect: menuselect/makeopts cleantest
+menuselect/gmenuselect: menuselect/makeopts .lastclean
+$(MAKE_MENUSELECT) gmenuselect
-menuselect/nmenuselect: menuselect/makeopts cleantest
+menuselect/nmenuselect: menuselect/makeopts .lastclean
+$(MAKE_MENUSELECT) nmenuselect
-menuselect/makeopts: makeopts cleantest
+menuselect/makeopts: makeopts .lastclean
+$(MAKE_MENUSELECT) makeopts
menuselect-tree: $(foreach dir,$(filter-out main,$(MOD_SUBDIRS)),$(wildcard $(dir)/*.c) $(wildcard $(dir)/*.cc)) build_tools/cflags.xml build_tools/cflags-devmode.xml sounds/sounds.xml build_tools/embed_modules.xml utils/utils.xml agi/agi.xml configure makeopts
@@ -944,7 +949,6 @@
.PHONY: full
.PHONY: _full
.PHONY: prereqs
-.PHONY: cleantest
.PHONY: uninstall
.PHONY: _uninstall
.PHONY: uninstall-all
Modified: team/oej/pinefrog-rtcp-11/README-SERIOUSLY.bestpractices.txt
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-11/README-SERIOUSLY.bestpractices.txt?view=diff&rev=415190&r1=415189&r2=415190
==============================================================================
--- team/oej/pinefrog-rtcp-11/README-SERIOUSLY.bestpractices.txt (original)
+++ team/oej/pinefrog-rtcp-11/README-SERIOUSLY.bestpractices.txt Thu Jun 5 04:19:59 2014
@@ -25,6 +25,9 @@
* Manager Class Authorizations:
Recognizing potential issues with certain classes of authorization
+
+* Avoid Privilege Escalations:
+ Disable the ability to execute functions that may escalate privileges
----------------
Additional Links
@@ -344,3 +347,24 @@
not running Asterisk as root, can prevent serious problems from arising when
allowing external connections to originate calls into Asterisk.
+===========================
+Avoid Privilege Escalations
+===========================
+
+External control protocols, such as Manager, often have the ability to get and
+set channel variables; which allows the execution of dialplan functions.
+
+Dialplan functions within Asterisk are incredibly powerful, which is wonderful
+for building applications using Asterisk. But during the read or write
+execution, certain diaplan functions do much more. For example, reading the
+SHELL() function can execute arbitrary commands on the system Asterisk is
+running on. Writing to the FILE() function can change any file that Asterisk has
+write access to.
+
+When these functions are executed from an external protocol, that execution
+could result in a privilege escalation. Asterisk can inhibit the execution of
+these functions, if live_dangerously in the [options] section of asterisk.conf
+is set to no.
+
+For backwards compatibility, live_dangerously defaults to yes, and must be
+explicitly set to no to enable this privilege escalation protection.
Modified: team/oej/pinefrog-rtcp-11/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-11/UPGRADE.txt?view=diff&rev=415190&r1=415189&r2=415190
==============================================================================
--- team/oej/pinefrog-rtcp-11/UPGRADE.txt (original)
+++ team/oej/pinefrog-rtcp-11/UPGRADE.txt Thu Jun 5 04:19:59 2014
@@ -20,12 +20,131 @@
===
===========================================================
+from 11.9 to 11.10
+ - The asterisk command line -I option and the asterisk.conf internal_timing
+ option are removed and always enabled if any timing module is loaded.
+
+ - 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.
+
+from 11.8 to 11.9
+- res_fax now returns the correct rates for V.27ter (4800 or 9600 bit/s).
+ Because of this the default settings would not load, so the minrate (minimum
+ transmission rate) option was changed to default to 4800 since that is the
+ minimum rate for v.27 which is included in the default modem options.
+
+- 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.
+
+- When communicating with a peer on an Asterisk 1.4 or earlier system, the
+ chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
+ prevents an incompatible connected line frame from an Astersik 1.8 or later
+ system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
+ this particular incompatibility has always existed between 1.4 and 1.8 and
+ later versions; this upgrade note is simply informing users of its existance.
+
+- A compatibility setting, allow_empty_string_in_nontext, has been added to
+ res_odbc.conf. When enabled (default behavior), empty column values are
+ stored as empty strings during realtime updates. Disabling this option
+ causes empty column values to be stored as NULLs for non-text columns.
+
+ Disable it for PostgreSQL backends in order to avoid errors caused by
+ updating integer columns with an empty string instead of NULL
+ (sippeers, sipregs, ..).
+
+From 11.7 to 11.8:
+- 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.
+
+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.
+
+Configuration Files:
+ - 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.
+
+From 11.6 to 11.7:
+ConfBridge
+ - ConfBridge now has the ability to set the language of announcements to the
+ conference. The language can be set on a bridge profile in confbridge.conf
+ or by the dialplan function CONFBRIDGE(bridge,language)=en.
+chan_sip - Clarify The "sip show peers" Forcerport Column And Add Comedia
+ - Under the "Forcerport" column, the "N" used to mean NAT (i.e. Yes). With
+ the additon of auto_* NAT settings, the meaning changed and there was a
+ certain combination of letters added to indicate the current setting. The
+ combination of using "Y", "N", "A" or "a", can be confusing. Therefore, we
+ now display clearly what the current Forcerport setting is: "Yes", "No",
+ "Auto (Yes)", "Auto (No)".
+ - Since we are clarifying the Forcerport column, we have added a column to
+ display the Comedia setting since this is useful information as well. We
+ no longer have a simple "NAT" setting like other versions before 11.
+
+* Certain dialplan functions have been marked as 'dangerous', and may only be
+ executed from the dialplan. Execution from extenal sources (AMI's GetVar and
+ SetVar actions; etc.) may be inhibited by setting live_dangerously in the
+ [options] section of asterisk.conf to no. SHELL(), channel locking, and direct
+ file read/write functions are marked as dangerous. DB_DELETE() and
+ REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
+ accept writes (which ignore the provided value).
+
+From 11.5 to 11.6:
+* res_agi will now properly indicate if there was an error in streaming an
+ audio file. The result code will be -1 and the result returned from the
+ the function will be RESULT_FAILURE instead of the prior behavior of always
+ returning RESULT_SUCCESS even if there was an error.
+* The libuuid development library is now optional for res_rtp_asterisk. If the
+ library is not present when building ICE and TURN support will not be present.
+* The option "register_retry_403" has been added to chan_sip to work around
+ servers that are known to erroneously send 403 in response to valid
+ REGISTER requests and allows Asterisk to continue attepmting to connect.
+ Due to a failed merge, this option is present, but non-functional until 11.8.0.
+
+From 11.4 to 11.5:
+* The default settings for chan_sip are now overriden properly by the general
+ settings in sip.conf. Please look over your settings upon upgrading.
+
+* It is now possible to play the Queue prompts to the first user waiting in a call queue.
+ Note that this may impact the ability for agents to talk with users, as a prompt may
+ still be playing when an agent connects to the user. This ability is disabled by
+ default but can be enabled on an individual queue using the 'announce-to-first-user'
+ option.
+
+* The libuuid development library is now required for res_rtp_asterisk. Consult
+ your distribution for the appropriate development library name.
+
From 11.3 to 11.4:
* Added the 'n' option to MeetMe to prevent application of the DENOISE function
to a channel joining a conference. Some channel drivers that vary the number
of audio samples in a voice frame will experience significant quality problems
if a denoiser is attached to the channel; this option gives them the ability
to remove the denoiser without having to unload func_speex.
+
+* The Registry AMI event for SIP registrations will now always include the
+ Username field. A previous bug fix missed an instance where it was not
+ included; that has been corrected in this release.
From 11.2.0 to 11.2.1:
* Asterisk would previously not output certain error messages when a remote
@@ -171,7 +290,7 @@
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:
@@ -243,10 +362,10 @@
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
Modified: team/oej/pinefrog-rtcp-11/addons/chan_ooh323.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-11/addons/chan_ooh323.c?view=diff&rev=415190&r1=415189&r2=415190
==============================================================================
--- team/oej/pinefrog-rtcp-11/addons/chan_ooh323.c (original)
+++ team/oej/pinefrog-rtcp-11/addons/chan_ooh323.c Thu Jun 5 04:19:59 2014
@@ -23,6 +23,7 @@
***/
#include "chan_ooh323.h"
+#include "asterisk/paths.h"
#include <math.h>
#define FORMAT_STRING_SIZE 512
@@ -30,7 +31,7 @@
/* Defaults */
#define DEFAULT_CONTEXT "default"
#define DEFAULT_H323ID "Asterisk PBX"
-#define DEFAULT_LOGFILE "/var/log/asterisk/h323_log"
+#define DEFAULT_LOGFILE "h323_log"
#define DEFAULT_H323ACCNT "ast_h323"
/* Flags */
@@ -294,8 +295,6 @@
static int usecnt = 0;
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
-AST_MUTEX_DEFINE_STATIC(ooh323c_cmd_lock);
-
static long callnumber = 0;
AST_MUTEX_DEFINE_STATIC(ooh323c_cn_lock);
@@ -307,6 +306,8 @@
int onCallEstablished(ooCallData *call);
int onCallCleared(ooCallData *call);
void onModeChanged(ooCallData *call, int t38mode);
+
+extern OOH323EndPoint gH323ep;
static char gLogFile[256] = DEFAULT_LOGFILE;
static int gPort = 1720;
@@ -644,6 +645,7 @@
ooh323_destroy(p);
ast_mutex_unlock(&iflock);
ast_log(LOG_ERROR, "Destination format is not supported\n");
+ *cause = AST_CAUSE_INVALID_NUMBER_FORMAT;
return NULL;
}
@@ -691,6 +693,10 @@
ooh323_destroy(p);
ast_mutex_unlock(&iflock);
return NULL;
+ } else if (!gH323ep.gkClient || (gH323ep.gkClient && gH323ep.gkClient->state != GkClientRegistered)) {
+ ast_log(LOG_ERROR, "Gatekeeper client is configured but not registered\n");
+ *cause = AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
+ return NULL;
}
p->g729onlyA = g729onlyA;
p->dtmfmode = gDTMFMode;
@@ -742,7 +748,6 @@
}
ast_mutex_unlock(&p->lock);
- ast_mutex_lock(&ooh323c_cmd_lock);
ast_cond_init(&p->rtpcond, NULL);
ooMakeCall(data, p->callToken, AST_MAX_EXTENSION, NULL);
ast_mutex_lock(&p->lock);
@@ -751,7 +756,6 @@
}
ast_mutex_unlock(&p->lock);
ast_cond_destroy(&p->rtpcond);
- ast_mutex_unlock(&ooh323c_cmd_lock);
}
restart_monitor();
@@ -1399,7 +1403,8 @@
}
break;
- case AST_CONTROL_PROCEEDING:
+ case AST_CONTROL_PROCEEDING:
+ case AST_CONTROL_PVT_CAUSE_CODE:
case -1:
break;
default:
@@ -2199,6 +2204,10 @@
ast_channel_unlock(p->owner);
p->owner = NULL;
ast_module_unref(myself);
+ }
+
+ if (!p->rtp) {
+ ast_cond_signal(&p->rtpcond);
}
ast_set_flag(p, H323_NEEDDESTROY);
@@ -2483,11 +2492,23 @@
return NULL;
}
} else if (!strcasecmp(v->name, "e164")) {
- if (!(peer->e164 = ast_strdup(v->value))) {
- ast_log(LOG_ERROR, "Could not allocate memory for e164 of "
+ int valid = 1;
+ const char *tmp;
+ for(tmp = v->value; *tmp; tmp++) {
+ if (!isdigit(*tmp)) {
+ valid = 0;
+ break;
+ }
+ }
+ if (valid) {
+ if (!(peer->e164 = ast_strdup(v->value))) {
+ ast_log(LOG_ERROR, "Could not allocate memory for e164 of "
"peer %s\n", name);
- ooh323_delete_peer(peer);
- return NULL;
+ ooh323_delete_peer(peer);
+ return NULL;
+ }
+ } else {
+ ast_log(LOG_ERROR, "Invalid e164: %s for peer %s\n", v->value, name);
}
} else if (!strcasecmp(v->name, "email")) {
if (!(peer->email = ast_strdup(v->value))) {
@@ -2618,7 +2639,8 @@
static int ooh323_do_reload(void)
{
- extern OOH323EndPoint gH323ep;
+ struct ooAliases * pNewAlias = NULL;
+ struct ooh323_peer *peer = NULL;
if (gH323Debug) {
ast_verb(0, "--- ooh323_do_reload\n");
@@ -2638,6 +2660,46 @@
gGatekeeper : 0, 0);
ooGkClientStart(gH323ep.gkClient);
}
+
+ /* Set aliases if any */
+ if (gH323Debug) {
+ ast_verb(0, "updating local aliases\n");
+ }
+
+ for (pNewAlias = gAliasList; pNewAlias; pNewAlias = pNewAlias->next) {
+ switch (pNewAlias->type) {
+ case T_H225AliasAddress_h323_ID:
+ ooH323EpAddAliasH323ID(pNewAlias->value);
+ break;
+ case T_H225AliasAddress_dialedDigits:
+ ooH323EpAddAliasDialedDigits(pNewAlias->value);
+ break;
+ case T_H225AliasAddress_email_ID:
+ ooH323EpAddAliasEmailID(pNewAlias->value);
+ break;
+ default:
+ ;
+ }
+ }
+
+ ast_mutex_lock(&peerl.lock);
+ peer = peerl.peers;
+ while (peer) {
+ if(peer->h323id) {
+ ooH323EpAddAliasH323ID(peer->h323id);
+ }
+ if(peer->email) {
+ ooH323EpAddAliasEmailID(peer->email);
+ }
+ if(peer->e164) {
+ ooH323EpAddAliasDialedDigits(peer->e164);
+ }
+ if(peer->url) {
+ ooH323EpAddAliasURLID(peer->url);
+ }
+ peer = peer->next;
+ }
+ ast_mutex_unlock(&peerl.lock);
if (gH323Debug) {
ast_verb(0, "+++ ooh323_do_reload\n");
@@ -2722,10 +2784,11 @@
free(prev);
}
gAliasList = NULL;
+ ooH323EpClearAllAliases();
}
/* Inintialize everything to default */
- strcpy(gLogFile, DEFAULT_LOGFILE);
+ snprintf(gLogFile, sizeof(gLogFile), "%s/%s", ast_config_AST_LOG_DIR, DEFAULT_LOGFILE);
gPort = 1720;
gIP[0] = '\0';
strcpy(gCallerID, DEFAULT_H323ID);
@@ -2838,17 +2901,29 @@
gAliasList = pNewAlias;
pNewAlias = NULL;
} else if (!strcasecmp(v->name, "e164")) {
- pNewAlias = ast_calloc(1, sizeof(struct ooAliases));
- if (!pNewAlias) {
- ast_log(LOG_ERROR, "Failed to allocate memory for e164 alias\n");
- ast_config_destroy(cfg);
- return 1;
+ int valid = 1;
+ const char *tmp;
+ for(tmp = v->value; *tmp; tmp++) {
+ if (!isdigit(*tmp)) {
+ valid = 0;
+ break;
+ }
}
- pNewAlias->type = T_H225AliasAddress_dialedDigits;
- pNewAlias->value = strdup(v->value);
- pNewAlias->next = gAliasList;
- gAliasList = pNewAlias;
- pNewAlias = NULL;
+ if (valid) {
+ pNewAlias = ast_calloc(1, sizeof(struct ooAliases));
+ if (!pNewAlias) {
+ ast_log(LOG_ERROR, "Failed to allocate memory for e164 alias\n");
+ ast_config_destroy(cfg);
+ return 1;
+ }
+ pNewAlias->type = T_H225AliasAddress_dialedDigits;
+ pNewAlias->value = strdup(v->value);
+ pNewAlias->next = gAliasList;
+ gAliasList = pNewAlias;
+ pNewAlias = NULL;
+ } else {
+ ast_log(LOG_ERROR, "Invalid e164: %s\n", v->value);
+ }
} else if (!strcasecmp(v->name, "email")) {
pNewAlias = ast_calloc(1, sizeof(struct ooAliases));
if (!pNewAlias) {
@@ -3372,7 +3447,6 @@
static char *handle_cli_ooh323_show_gk(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
char value[FORMAT_STRING_SIZE];
- extern OOH323EndPoint gH323ep;
switch (cmd) {
case CLI_INIT:
@@ -3420,6 +3494,9 @@
break;
case GkClientFailed:
ast_cli(a->fd, "%-20s%s\n", "GK state:", "Failed");
+ break;
+ case GkClientStopped:
+ ast_cli(a->fd, "%-20s%s\n", "GK state:", "Shutdown");
break;
default:
break;
@@ -3845,6 +3922,13 @@
if (reloading) {
ast_verb(1, "Reloading H.323\n");
ooh323_do_reload();
+ }
+ if (gH323ep.gkClient && gH323ep.gkClient->state == GkClientStopped) {
+ ooGkClientDestroy();
+ ast_verb(0, "Restart stopped gatekeeper client\n");
+ ooGkClientInit(gRasGkMode, (gRasGkMode == RasUseSpecificGatekeeper) ?
+ gGatekeeper : 0, 0);
+ ooGkClientStart(gH323ep.gkClient);
}
/* Check for interfaces needing to be killed */
Modified: team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCalls.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCalls.h?view=diff&rev=415190&r1=415189&r2=415190
==============================================================================
--- team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCalls.h (original)
+++ team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCalls.h Thu Jun 5 04:19:59 2014
@@ -228,7 +228,7 @@
char lastDTMF;
ASN1UINT nextDTMFstamp;
int rtdrInterval, rtdrCount; /* roundTripDelay interval and unreplied count */
- ASN1UINT rtdrSend, rtdrRecv; /* last sended/replied RTD request */
+ ASN1UINT8 rtdrSend, rtdrRecv; /* last sended/replied RTD request */
void *usrData; /*!<User can set this to user specific data*/
struct OOH323CallData* next;
struct OOH323CallData* prev;
Modified: team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCapability.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCapability.c?view=diff&rev=415190&r1=415189&r2=415190
==============================================================================
--- team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCapability.c (original)
+++ team/oej/pinefrog-rtcp-11/addons/ooh323c/src/ooCapability.c Thu Jun 5 04:19:59 2014
@@ -2940,6 +2940,22 @@
break;
+ case T_H245Capability_receiveAndTransmitUserInputCapability:
+ if((cap->u.receiveAndTransmitUserInputCapability->t ==
+ T_H245UserInputCapability_basicString) &&
+ (call->dtmfmode & OO_CAP_DTMF_H245_alphanumeric))
+ {
+ call->jointDtmfMode |= OO_CAP_DTMF_H245_alphanumeric;
+ return OO_OK;
+ }
+ else if((cap->u.receiveAndTransmitUserInputCapability->t ==
+ T_H245UserInputCapability_dtmf) &&
+ (call->dtmfmode & OO_CAP_DTMF_H245_signal))
+ {
+ call->jointDtmfMode |= OO_CAP_DTMF_H245_signal;
+ return OO_OK;
+ }
+
case T_H245Capability_receiveUserInputCapability:
if((cap->u.receiveUserInputCapability->t ==
[... 27932 lines stripped ...]
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