[svn-commits] oej: branch oej/sipregister r44726 - in
/team/oej/sipregister: ./ agi/ apps/ ...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Oct 7 12:05:17 MST 2006
Author: oej
Date: Sat Oct 7 14:05:15 2006
New Revision: 44726
URL: http://svn.digium.com/view/asterisk?rev=44726&view=rev
Log:
Trying to restore this branch...
Added:
team/oej/sipregister/build_tools/strip_nonapi
- copied unchanged from r44545, trunk/build_tools/strip_nonapi
team/oej/sipregister/channels/chan_gtalk.c
- copied unchanged from r44545, trunk/channels/chan_gtalk.c
team/oej/sipregister/channels/h323/Makefile.in
- copied unchanged from r44545, trunk/channels/h323/Makefile.in
team/oej/sipregister/channels/h323/ast_h323.cxx
- copied unchanged from r44545, trunk/channels/h323/ast_h323.cxx
team/oej/sipregister/channels/h323/caps_h323.cxx
- copied unchanged from r44545, trunk/channels/h323/caps_h323.cxx
team/oej/sipregister/channels/h323/caps_h323.h
- copied unchanged from r44545, trunk/channels/h323/caps_h323.h
team/oej/sipregister/channels/h323/cisco-h225.asn
- copied unchanged from r44545, trunk/channels/h323/cisco-h225.asn
team/oej/sipregister/channels/h323/cisco-h225.cxx
- copied unchanged from r44545, trunk/channels/h323/cisco-h225.cxx
team/oej/sipregister/channels/h323/cisco-h225.h
- copied unchanged from r44545, trunk/channels/h323/cisco-h225.h
team/oej/sipregister/channels/h323/compat_h323.cxx
- copied unchanged from r44545, trunk/channels/h323/compat_h323.cxx
team/oej/sipregister/channels/h323/compat_h323.h
- copied unchanged from r44545, trunk/channels/h323/compat_h323.h
team/oej/sipregister/channels/h323/noexport.map
- copied unchanged from r44545, trunk/channels/h323/noexport.map
team/oej/sipregister/doc/rtp-packetization.txt
- copied unchanged from r44545, trunk/doc/rtp-packetization.txt
team/oej/sipregister/funcs/func_blacklist.c
- copied unchanged from r44545, trunk/funcs/func_blacklist.c
team/oej/sipregister/funcs/func_vmcount.c
- copied unchanged from r44545, trunk/funcs/func_vmcount.c
team/oej/sipregister/include/jitterbuf.h
- copied unchanged from r44545, trunk/include/jitterbuf.h
team/oej/sipregister/pbx/ael/ael-test/ael-test15/
- copied from r44545, trunk/pbx/ael/ael-test/ael-test15/
team/oej/sipregister/pbx/ael/ael-test/ael-test15/extensions.ael
- copied unchanged from r44545, trunk/pbx/ael/ael-test/ael-test15/extensions.ael
team/oej/sipregister/pbx/ael/ael-test/ael-test16/
- copied from r44545, trunk/pbx/ael/ael-test/ael-test16/
team/oej/sipregister/pbx/ael/ael-test/ael-test16/extensions.ael
- copied unchanged from r44545, trunk/pbx/ael/ael-test/ael-test16/extensions.ael
team/oej/sipregister/pbx/ael/ael-test/ael-test3/telemarket_torture.ael2
- copied unchanged from r44545, trunk/pbx/ael/ael-test/ael-test3/telemarket_torture.ael2
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test15
- copied unchanged from r44545, trunk/pbx/ael/ael-test/ref.ael-test15
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test16
- copied unchanged from r44545, trunk/pbx/ael/ael-test/ref.ael-test16
Removed:
team/oej/sipregister/apps/app_hasnewvoicemail.c
team/oej/sipregister/apps/app_lookupblacklist.c
team/oej/sipregister/apps/app_lookupcidname.c
team/oej/sipregister/apps/app_random.c
team/oej/sipregister/apps/app_realtime.c
team/oej/sipregister/apps/app_setcdruserfield.c
team/oej/sipregister/apps/app_settransfercapability.c
team/oej/sipregister/channels/h323/Makefile
team/oej/sipregister/channels/h323/ast_h323.cpp
team/oej/sipregister/channels/h323/h323.conf.sample
team/oej/sipregister/funcs/func_language.c
team/oej/sipregister/funcs/func_moh.c
Modified:
team/oej/sipregister/CHANGES
team/oej/sipregister/CREDITS
team/oej/sipregister/LICENSE
team/oej/sipregister/Makefile
team/oej/sipregister/UPGRADE.txt
team/oej/sipregister/agi/ (props changed)
team/oej/sipregister/agi/Makefile
team/oej/sipregister/apps/app_adsiprog.c
team/oej/sipregister/apps/app_alarmreceiver.c
team/oej/sipregister/apps/app_amd.c
team/oej/sipregister/apps/app_authenticate.c
team/oej/sipregister/apps/app_chanspy.c
team/oej/sipregister/apps/app_db.c
team/oej/sipregister/apps/app_dial.c
team/oej/sipregister/apps/app_directed_pickup.c
team/oej/sipregister/apps/app_directory.c
team/oej/sipregister/apps/app_disa.c
team/oej/sipregister/apps/app_dumpchan.c
team/oej/sipregister/apps/app_echo.c
team/oej/sipregister/apps/app_externalivr.c
team/oej/sipregister/apps/app_festival.c
team/oej/sipregister/apps/app_followme.c
team/oej/sipregister/apps/app_getcpeid.c
team/oej/sipregister/apps/app_ices.c
team/oej/sipregister/apps/app_macro.c
team/oej/sipregister/apps/app_meetme.c
team/oej/sipregister/apps/app_mixmonitor.c
team/oej/sipregister/apps/app_mp3.c
team/oej/sipregister/apps/app_nbscat.c
team/oej/sipregister/apps/app_osplookup.c
team/oej/sipregister/apps/app_page.c
team/oej/sipregister/apps/app_playback.c
team/oej/sipregister/apps/app_privacy.c
team/oej/sipregister/apps/app_queue.c
team/oej/sipregister/apps/app_read.c
team/oej/sipregister/apps/app_record.c
team/oej/sipregister/apps/app_rpt.c
team/oej/sipregister/apps/app_setcallerid.c
team/oej/sipregister/apps/app_speech_utils.c
team/oej/sipregister/apps/app_stack.c
team/oej/sipregister/apps/app_talkdetect.c
team/oej/sipregister/apps/app_test.c
team/oej/sipregister/apps/app_url.c
team/oej/sipregister/apps/app_voicemail.c
team/oej/sipregister/apps/app_waitforring.c
team/oej/sipregister/apps/app_waitforsilence.c
team/oej/sipregister/apps/app_zapateller.c
team/oej/sipregister/apps/app_zapbarge.c
team/oej/sipregister/apps/app_zapras.c
team/oej/sipregister/apps/app_zapscan.c
team/oej/sipregister/bootstrap.sh
team/oej/sipregister/build_tools/cflags.xml
team/oej/sipregister/build_tools/embed_modules.xml
team/oej/sipregister/build_tools/make_version
team/oej/sipregister/build_tools/menuselect-deps.in
team/oej/sipregister/build_tools/prep_moduledeps
team/oej/sipregister/channels/Makefile
team/oej/sipregister/channels/chan_agent.c
team/oej/sipregister/channels/chan_alsa.c
team/oej/sipregister/channels/chan_features.c
team/oej/sipregister/channels/chan_h323.c
team/oej/sipregister/channels/chan_iax2.c
team/oej/sipregister/channels/chan_jingle.c
team/oej/sipregister/channels/chan_local.c
team/oej/sipregister/channels/chan_mgcp.c
team/oej/sipregister/channels/chan_misdn.c
team/oej/sipregister/channels/chan_nbs.c
team/oej/sipregister/channels/chan_oss.c
team/oej/sipregister/channels/chan_phone.c
team/oej/sipregister/channels/chan_sip.c
team/oej/sipregister/channels/chan_skinny.c
team/oej/sipregister/channels/chan_vpb.cc
team/oej/sipregister/channels/chan_zap.c
team/oej/sipregister/channels/h323/ (props changed)
team/oej/sipregister/channels/h323/README
team/oej/sipregister/channels/h323/TODO
team/oej/sipregister/channels/h323/ast_h323.h
team/oej/sipregister/channels/h323/chan_h323.h
team/oej/sipregister/channels/iax2-parser.c
team/oej/sipregister/channels/iax2-provision.c
team/oej/sipregister/channels/misdn/isdn_lib.c
team/oej/sipregister/channels/misdn/isdn_lib.h
team/oej/sipregister/configure
team/oej/sipregister/configure.ac
team/oej/sipregister/doc/CODING-GUIDELINES
team/oej/sipregister/doc/ael.txt
team/oej/sipregister/doc/app-sms.txt
team/oej/sipregister/doc/asterisk-conf.txt
team/oej/sipregister/doc/backtrace.txt
team/oej/sipregister/doc/billing.txt
team/oej/sipregister/doc/callfiles.txt
team/oej/sipregister/doc/chaniax.txt
team/oej/sipregister/doc/channelvariables.txt
team/oej/sipregister/doc/configuration.txt
team/oej/sipregister/doc/cygwin.txt
team/oej/sipregister/doc/dundi.txt
team/oej/sipregister/doc/enum.txt
team/oej/sipregister/doc/extensions.txt
team/oej/sipregister/doc/imapstorage.txt
team/oej/sipregister/doc/ip-tos.txt
team/oej/sipregister/doc/jabber.txt
team/oej/sipregister/doc/jingle.txt
team/oej/sipregister/doc/manager.txt
team/oej/sipregister/doc/misdn.txt
team/oej/sipregister/doc/mp3.txt
team/oej/sipregister/doc/osp.txt
team/oej/sipregister/doc/queues-with-callback-members.txt
team/oej/sipregister/doc/realtime.txt
team/oej/sipregister/doc/security.txt
team/oej/sipregister/doc/sms.txt
team/oej/sipregister/doc/speechrec.txt
team/oej/sipregister/funcs/func_callerid.c
team/oej/sipregister/funcs/func_curl.c
team/oej/sipregister/funcs/func_db.c
team/oej/sipregister/funcs/func_md5.c
team/oej/sipregister/funcs/func_odbc.c
team/oej/sipregister/funcs/func_rand.c
team/oej/sipregister/funcs/func_strings.c
team/oej/sipregister/funcs/func_timeout.c
team/oej/sipregister/include/asterisk/acl.h
team/oej/sipregister/include/asterisk/channel.h
team/oej/sipregister/include/asterisk/chanspy.h
team/oej/sipregister/include/asterisk/compat.h
team/oej/sipregister/include/asterisk/config.h
team/oej/sipregister/include/asterisk/monitor.h
team/oej/sipregister/include/asterisk/res_odbc.h
team/oej/sipregister/include/asterisk/rtp.h
team/oej/sipregister/include/asterisk/utils.h
team/oej/sipregister/pbx/Makefile
team/oej/sipregister/pbx/ael/ael-test/ael-test11/extensions.ael
team/oej/sipregister/pbx/ael/ael-test/ael-test3/extensions.ael
team/oej/sipregister/pbx/ael/ael-test/ael-test5/extensions.ael
team/oej/sipregister/pbx/ael/ael-test/ref.ael-ntest10
team/oej/sipregister/pbx/ael/ael-test/ref.ael-ntest12
team/oej/sipregister/pbx/ael/ael-test/ref.ael-ntest9
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test1
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test11
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test14
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test2
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test3
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test4
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test5
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test6
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test7
team/oej/sipregister/pbx/ael/ael-test/ref.ael-test8
team/oej/sipregister/pbx/ael/ael-test/ref.ael-vtest13
team/oej/sipregister/pbx/ael/ael.flex
team/oej/sipregister/pbx/ael/ael.tab.c
team/oej/sipregister/pbx/ael/ael.y
team/oej/sipregister/pbx/ael/ael_lex.c
team/oej/sipregister/pbx/pbx_ael.c
team/oej/sipregister/pbx/pbx_config.c
team/oej/sipregister/pbx/pbx_dundi.c
team/oej/sipregister/pbx/pbx_spool.c
team/oej/sipregister/sounds/Makefile
team/oej/sipregister/utils/Makefile
Modified: team/oej/sipregister/CHANGES
URL: http://svn.digium.com/view/asterisk/team/oej/sipregister/CHANGES?rev=44726&r1=44725&r2=44726&view=diff
==============================================================================
--- team/oej/sipregister/CHANGES (original)
+++ team/oej/sipregister/CHANGES Sat Oct 7 14:05:15 2006
@@ -1,127 +1,39 @@
-Changes since Asterisk 1.2.0-beta2:
+Changes since Asterisk 1.4-beta was branched:
- * Cygwin build system portability
- * Optional generation of outbound silence during channel recording
-
-Changes since Asterisk 1.2.0-beta1:
-
- * Many, many bug fixes
- * Documentation and sample configuration updates
- * Vastly improved presence/subscription support in the SIP channel driver
- * A new (experimental) mISDN channel driver
- * A new monitoring application (MixMonitor)
- * More portability fixes for non-Linux platforms
- * New dialplan functions replacing old applications
- * Significant deadlock and performance upgrades for the Manager interface
- * An upgrade to the 'new' dialplan expression parser for all users
- * New Zaptel echo cancellers with improved performance
- * Support for the latest OSP toolkit from TransNexus
- * Support user-controlled volume adjustment in MeetMe application
- * More dialplan applications now return status variables instead of priority jumping
- * Much more powerful ENUM support in the dialplan
- * SIP domain support for authentication and virtual hosting
- * Many PRI protocol updates and fixes, including more complete Q.SIG support
- * New applications: Pickup() and Page()
-
-Changes since Asterisk 1.0:
-
-This list currently only containts changes made from the end of November until
-March 26, 2005.
-
- * Add new applications:
- -- AgentMonitorOutgoing
- -- Curl
- -- ExecIf
- -- ExecIfTime
- -- IAX2Provision
- -- MacroExit
- -- MacroIf
- -- PauseQueueMember
- -- ReadFile
- -- SetRDNIS
- -- SIPAddHeader
- -- SIPGetHeader
- -- StartMusicOnHold
- -- StopMusicOnHold
- -- UnpauseQueueMember
- -- WaitForSilence
- -- While / EndWhile
- * app Answer
- -- added delay option
- * app ChanIsAvail
- -- added 's' option
- * app Dial
- -- add option to specify the class for musiconhold with m option
- * app EnumLookup
- -- added "reload enum" for configuration
- * app Goto
- -- added relative priorities
- * app GotoIf
- -- added relative priorities
- * app MeetMe
- -- added 'i' option
- -- added 'r' option
- -- added 'T' option
- -- added 'P' option
- -- added 'c' option
- -- added adminpin to meetme.conf
- -- added reload command
- * app PrivacyManager
- -- add config file privacy.conf
- * app queue
- -- queues.conf
- -- added persistentmembers option to queues.conf
- -- changed music option to musiconhold
- -- added weight option
- -- added note about why agent groups probably shouldn't be used
- -- added timeoutrestart option
- * app Read
- -- added attempts parameter
- -- added timeout parameter
- * app Record
- -- added 'q' option
- * app SendDTMF
- -- add timeout option
- * app SMS
- -- document alternative syntax for queueing messages
- * app Voicemail
- -- add info about VM_CATEGORY
- -- voicemail.conf
- -- added usedirectory option
- -- added VM_CIDNUM and VM_CIDNAME in message config
- * chan IAX2
- -- new jitterbuffer
- -- added setvar option
- -- added regex to iax2 show peers/users
- -- allow multiple bindaddr lines in iax.conf
- -- added reload command
- -- added forcejitterbuffer option
- -- added note about specifying bindport before bindaddr
- -- added trunktimestamps option
- * chan Agent
- -- added agent logoff CLI command
- * chan OSS
- -- added Flash CLI command
- * chan SIP
- -- added setvar option
- -- added compactheaders option
- -- added usereqphone option
- -- added registertimeout option
- -- added externhost option
- -- added sip notify CLI command
- -- added sip_notify.conf
- -- added allowguest option
- * chan Zap
- -- added hanguponplarityswitch option
- -- added sendcalleridafter option
- -- added priresetinterval option
- -- added TON/NPI config options (the ones right above the resetinterval option)
- -- added answeronpolarityswitch option
- -- added "never" for resetinterval
- * extensions
- -- allow '*' when including files (#include "sip-*.conf")
- -- added eswitch
- * General
- -- added #exec syntax for including output from a command
- -- added show features CLI command
- -- added configuration templates for category inheritance
+ * Argument support for Gosub application
+ * MailboxExists converted to dialplan function
+ * Ability to set process limits without restarting Asterisk
+ * SS7 support in chan_zap (via libss7 library)
+ * Proper codec support in chan_skinny.
+ * AEL upgraded to use the Gosub with Arguments instead
+ of Macro application, to hopefully reduce the problems
+ seen with the artificially low stack ceiling that
+ Macro bumps into. Macros can only call other Macros
+ to a depth of 7. Tests run using gosub, show depths
+ limited only by virtual memory. A small test demonstrated
+ recursive call depths of 100,000 without problems.
+ * Ability to use libcap to set high ToS bits when non-root
+ on Linux. If configure is unable to find libcap then you
+ can use --with-cap to specify the path.
+ * H323 remote hold notification support added (by NOTIFY message
+ and/or H.450 supplementary service)
+ * Added keepstats option to queues.conf which will keep queue
+ statistics during a reload.
+ * Added rotatetimestamp option to logger.conf which will use
+ the time to name the logger files instead of sequence number.
+ * The output of CallerID in Manager events is now more consistent.
+ CallerIDNum is used for number and CallerIDName for name.
+ * setinterfacevar option in queues.conf also now sets a variable
+ called MEMBERNAME which contains the member's name.
+ * Added Masquerade manager event for when a masquerade happens between
+ two channels.
+ * Added 'Strategy' field to manager event QueueParams which represents
+ the queue strategy in use.
+ * From the to-do lists: straighten out the app timeout args:
+ Wait() app now really does 0.3 seconds- was truncating arg to an int.
+ WaitExten() same as Wait().
+ Congestion() - Now takes floating pt. argument.
+ Busy() - now takes floating pt. argument.
+ Read() - timeout now can be floating pt.
+ WaitForRing() now takes floating pt timeout arg.
+ SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
Modified: team/oej/sipregister/CREDITS
URL: http://svn.digium.com/view/asterisk/team/oej/sipregister/CREDITS?rev=44726&r1=44725&r2=44726&view=diff
==============================================================================
--- team/oej/sipregister/CREDITS (original)
+++ team/oej/sipregister/CREDITS Sat Oct 7 14:05:15 2006
@@ -150,6 +150,8 @@
James Rothenberger - Support for IMAP storage integration added by OneBizTone LLC Work funded by University of Pennsylvania jar at onebiztone.com
+Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
+
=== OTHER CONTRIBUTIONS ===
John Todd - Monkey sounds and associated teletorture prompt
Michael Jerris - bug marshaling
Modified: team/oej/sipregister/LICENSE
URL: http://svn.digium.com/view/asterisk/team/oej/sipregister/LICENSE?rev=44726&r1=44725&r2=44726&view=diff
==============================================================================
--- team/oej/sipregister/LICENSE (original)
+++ team/oej/sipregister/LICENSE Sat Oct 7 14:05:15 2006
@@ -38,7 +38,8 @@
wish to use these trademarks for purposes other than simple
redistribution of Asterisk source code obtained from Digium, you
should contact our licensing department to determine the necessary
-steps you must take.
+steps you must take. For more information on this policy, please read
+http://www.digium.com/en/company/profile/trademarkpolicy.php
If you have any questions regarding our licensing policy, please
contact us:
Modified: team/oej/sipregister/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/sipregister/Makefile?rev=44726&r1=44725&r2=44726&view=diff
==============================================================================
--- team/oej/sipregister/Makefile (original)
+++ team/oej/sipregister/Makefile Sat Oct 7 14:05:15 2006
@@ -13,16 +13,51 @@
# All Makefiles use the following variables:
#
-# LDFLAGS - linker flags (not libraries), used for all links
+# ASTCFLAGS - compiler options
+# ASTLDFLAGS - linker flags (not libraries)
+# AST_LIBS - libraries to build binaries XXX
# LIBS - additional libraries, at top-level for all links,
# on a single object just for that object
# SOLINK - linker flags used only for creating shared objects (.so files),
# used for all .so links
#
-
-.EXPORT_ALL_VARIABLES:
-
-include makeopts
+# Default values fo ASTCFLAGS and ASTLDFLAGS can be specified in the
+# environment when running make, as follows:
+#
+# $ ASTCFLAGS="-Werror" make
+
+export ASTTOPDIR
+export ASTERISKVERSION
+export ASTERISKVERSIONNUM
+export INSTALL_PATH
+export ASTETCDIR
+export ASTVARRUNDIR
+export MODULES_DIR
+export ASTSPOOLDIR
+export ASTVARLIBDIR
+export ASTDATADIR
+export ASTLOGDIR
+export AGI_DIR
+export ASTCONFPATH
+export NOISY_BUILD
+export MENUSELECT_CFLAGS
+export CC
+export CXX
+export AR
+export RANLIB
+export HOST_CC
+export STATIC_BUILD
+export INSTALL
+export DESTDIR
+export PROC
+export SOLINK
+
+# even though we could use '-include makeopts' here, use a wildcard
+# lookup anyway, so that make won't try to build makeopts if it doesn't
+# exist (other rules will force it to be built if needed)
+ifneq ($(wildcard makeopts),)
+ include makeopts
+endif
#Uncomment this to see all build commands instead of 'quiet' output
#NOISY_BUILD=yes
@@ -74,7 +109,9 @@
ASTVARLIBDIR=$(localstatedir)/lib/asterisk
endif
endif
-ASTDATADIR?=$(ASTVARLIBDIR)
+ifeq ($(ASTDATADIR),)
+ ASTDATADIR:=$(ASTVARLIBDIR)
+endif
# Asterisk.conf is located in ASTETCDIR or by using the -C flag
# when starting Asterisk
@@ -87,8 +124,6 @@
# Determine by a grep 'ScriptAlias' of your Apache httpd.conf file
HTTP_CGIDIR=/var/www/cgi-bin
-ASTCFLAGS=
-
# Uncomment this to use the older DSP routines
#ASTCFLAGS+=-DOLD_DSP_ROUTINES
@@ -101,8 +136,8 @@
GLOBAL_MAKEOPTS=$(wildcard /etc/asterisk.makeopts)
USER_MAKEOPTS=$(wildcard ~/.asterisk.makeopts)
-MOD_SUBDIR_CFLAGS=-I../include -I../main
-OTHER_SUBDIR_CFLAGS=-I../include
+MOD_SUBDIR_CFLAGS=-I$(ASTTOPDIR)/include
+OTHER_SUBDIR_CFLAGS=-I$(ASTTOPDIR)/include
ifeq ($(OSARCH),linux-gnu)
ifeq ($(PROC),x86_64)
@@ -137,13 +172,6 @@
endif
endif
-ID=id
-
-ifeq ($(OSARCH),SunOS)
- M4=/usr/local/bin/m4
- ID=/usr/xpg4/bin/id
-endif
-
ASTCFLAGS+=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG)
ifeq ($(AST_DEVMODE),yes)
@@ -151,7 +179,8 @@
endif
ifneq ($(findstring BSD,$(OSARCH)),)
- ASTCFLAGS+=-I/usr/local/include -L/usr/local/lib
+ ASTCFLAGS+=-I/usr/local/include
+ ASTLDFLAGS+=-L/usr/local/lib
endif
ifneq ($(PROC),ultrasparc)
@@ -233,14 +262,15 @@
HAVEDOT=no
endif
-all: cleantest $(SUBDIRS)
+all: _all
@echo " +--------- Asterisk Build Complete ---------+"
- @echo " + Asterisk has successfully been built, but +"
- @echo " + cannot be run before being installed by +"
- @echo " + running: +"
+ @echo " + Asterisk has successfully been built, and +"
+ @echo " + can be installed by running: +"
@echo " + +"
@echo " + make install +"
@echo " +-------------------------------------------+"
+
+_all: cleantest $(SUBDIRS)
makeopts: configure
@echo "****"
@@ -276,10 +306,10 @@
main: $(filter-out main,$(MOD_SUBDIRS))
$(MOD_SUBDIRS):
- @CFLAGS="$(MOD_SUBDIR_CFLAGS)$(ASTCFLAGS)" $(MAKE) --no-print-directory -C $@ SUBDIR=$@ all
+ @ASTCFLAGS="$(MOD_SUBDIR_CFLAGS) $(ASTCFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" AST_LIBS="$(AST_LIBS)" $(MAKE) --no-print-directory -C $@ SUBDIR=$@ all
$(OTHER_SUBDIRS):
- @CFLAGS="$(OTHER_SUBDIR_CFLAGS)$(ASTCFLAGS)" $(MAKE) --no-print-directory -C $@ SUBDIR=$@ all
+ @ASTCFLAGS="$(OTHER_SUBDIR_CFLAGS) $(ASTCFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" $(MAKE) --no-print-directory -C $@ SUBDIR=$@ all
defaults.h: makeopts
@build_tools/make_defaults_h > $@.tmp
@@ -316,6 +346,7 @@
rm -f include/asterisk/version.h
rm -f .depend
@$(MAKE) -C menuselect clean
+ cp -f .cleancount .lastclean
dist-clean: distclean
@@ -331,7 +362,7 @@
rm -rf doc/api
rm -f build_tools/menuselect-deps
-datafiles: all
+datafiles: _all
if [ x`$(ID) -un` = xroot ]; then CFLAGS="$(ASTCFLAGS)" sh build_tools/mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
# Should static HTTP be installed during make samples or even with its own target ala
# webvoicemail? There are portions here that *could* be customized but might also be
@@ -365,7 +396,7 @@
NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
OLDHEADERS=$(filter-out $(NEWHEADERS),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
-bininstall: all
+bininstall: _all
mkdir -p $(DESTDIR)$(MODULES_DIR)
mkdir -p $(DESTDIR)$(ASTSBINDIR)
mkdir -p $(DESTDIR)$(ASTETCDIR)
@@ -429,7 +460,7 @@
echo " WARNING WARNING WARNING" ;\
fi
-install: all datafiles bininstall $(SUBDIRS_INSTALL)
+install: datafiles bininstall $(SUBDIRS_INSTALL)
@if [ -x /usr/sbin/asterisk-post-install ]; then \
/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
fi
@@ -456,7 +487,7 @@
@echo " +-------------------------------------------+"
@$(MAKE) -s oldmodcheck
-upgrade: all bininstall
+upgrade: bininstall
adsi:
mkdir -p $(DESTDIR)$(ASTETCDIR)
@@ -588,10 +619,10 @@
fi
$(MOD_SUBDIRS_DEPEND):
- @CFLAGS="$(MOD_SUBDIR_CFLAGS)$(ASTCFLAGS)" $(MAKE) --no-print-directory -C $(@:-depend=) depend
+ @ASTCFLAGS="$(MOD_SUBDIR_CFLAGS) $(ASTCFLAGS)" $(MAKE) --no-print-directory -C $(@:-depend=) depend
$(OTHER_SUBDIRS_DEPEND):
- @CFLAGS="$(OTHER_SUBDIR_CFLAGS)$(ASTCFLAGS)" $(MAKE) --no-print-directory -C $(@:-depend=) depend
+ @ASTCFLAGS="$(OTHER_SUBDIR_CFLAGS) $(ASTCFLAGS)" $(MAKE) --no-print-directory -C $(@:-depend=) depend
depend: include/asterisk/version.h include/asterisk/buildopts.h defaults.h $(SUBDIRS_DEPEND)
@@ -604,8 +635,7 @@
cleantest:
@if ! cmp -s .cleancount .lastclean ; then \
- $(MAKE) clean; cp -f .cleancount .lastclean;\
- $(MAKE) defaults.h;\
+ $(MAKE) clean;\
fi
$(SUBDIRS_UNINSTALL):
@@ -644,7 +674,7 @@
rm -rf $(DESTDIR)$(ASTLOGDIR)
menuselect: menuselect/menuselect menuselect-tree
- - at menuselect/menuselect $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts && echo "menuselect changes saved!" || echo "menuselect changes NOT saved!"
+ - at menuselect/menuselect $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
menuselect/menuselect: makeopts menuselect/menuselect.c menuselect/menuselect_curses.c menuselect/menuselect_stub.c menuselect/menuselect.h menuselect/linkedlists.h makeopts
@unset CC LD AR RANLIB && $(MAKE) -C menuselect CONFIGURE_SILENT="--silent"
Modified: team/oej/sipregister/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/oej/sipregister/UPGRADE.txt?rev=44726&r1=44725&r2=44726&view=diff
==============================================================================
--- team/oej/sipregister/UPGRADE.txt (original)
+++ team/oej/sipregister/UPGRADE.txt Sat Oct 7 14:05:15 2006
@@ -1,404 +1,8 @@
Information for Upgrading From Previous Asterisk Releases
=========================================================
-Build Process (configure script):
-
-Asterisk now uses an autoconf-generated configuration script to learn how it
-should build itself for your system. As it is a standard script, running:
-
-$ ./configure --help
-
-will show you all the options available. This script can be used to tell the
-build process what libraries you have on your system (if it cannot find them
-automatically), which libraries you wish to have ignored even though they may
-be present, etc.
-
-You must run the configure script before Asterisk will build, although it will
-attempt to automatically run it for you with no options specified; for most
-users, that will result in a similar build to what they would have had before
-the configure script was added to the build process (except for having to run
-'make' again after the configure script is run). Note that the configure script
-does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
-when your system configuration changes or you wish to build Asterisk with
-different options.
-
-Build Process (module selection):
-
-The Asterisk source tree now includes a basic module selection and build option
-selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
-In this tool, you can disable building of modules that you don't care about,
-turn on/off global options for the build and see which modules will not
-(and cannot) be built because your system does not have the required external
-dependencies installed.
-
-The resulting file from menuselect is called 'menuselect.makeopts'. Note that
-the resulting menuselect.makeopts file generally contains which modules *not*
-to build. The modules listed in this file indicate which modules have unmet
-dependencies, a present conflict, or have been disabled by the user in the
-menuselect interface. Compiler Flags can also be set in the menuselect
-interface. In this case, the resulting file contains which CFLAGS are in use,
-not which ones are not in use.
-
-If you would like to save your choices and have them applied against all
-builds, the file can be copied to '~/.asterisk.makeopts' or
-'/etc/asterisk.makeopts'.
-
-Build Process (Makefile targets):
-
-The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
-is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
-in the menuselect tool.
-
-It is now possible to run most make targets against a single subdirectory; from
-the top level directory, for example, 'make channels' will run 'make all' in the
-'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
-
-Sound (prompt) and Music On Hold files:
-
-Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
-use with Asterisk have been replaced with new versions produced from high quality
-master recordings, and are available in three languages (English, French and
-Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
-In addition, the music on hold files provided by FreePlay Music are now available
-in the same five formats, but no longer available in MP3 format.
-
-The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
-(as were supplied with previous releases) and the FreePlay MOH files in WAV format.
-All of the other variations can be installed by running 'make menuselect' and
-selecting the packages you wish to install; when you run 'make install', those
-packages will be downloaded and installed along with the standard files included
-in the tarball.
-
-If for some reason you expect to not have Internet access at the time you will be
-running 'make install', you can make your package selections using menuselect and
-then run 'make sounds' to download (only) the sound packages; this will leave the
-sound packages in the 'sounds' subdirectory to be used later during installation.
-
-WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
-instead of the alternate-language files being stored in subdirectories underneath
-the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
-etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
-language itself, then places all the sound files for that language under that
-directory and its subdirectories. This is the layout that will be created if you
-select non-English languages to be installed via menuselect, HOWEVER Asterisk does
-not default to this layout and will not find the files in the places it expects them
-to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
-/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
-installed.
-
-PBX Core:
-
-* The (very old and undocumented) ability to use BYEXTENSION for dialing
- instead of ${EXTEN} has been removed.
-
-* Builtin (res_features) transfer functionality attempts to use the context
- defined in TRANSFER_CONTEXT variable of the transferer channel first. If
- not set, it uses the transferee variable. If not set in any channel, it will
- attempt to use the last non macro context. If not possible, it will default
- to the current context.
-
-* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
- if your dialplan relies on the ability to 'run off the end' of an extension
- and wait for a new extension without using WaitExten() to accomplish that,
- you will need set autofallthrough to 'no' in your extensions.conf file.
-
-Command Line Interface:
-
-* 'show channels concise', designed to be used by applications that will parse
- its output, previously used ':' characters to separate fields. However, some
- of those fields can easily contain that character, making the output not
- parseable. The delimiter has been changed to '!'.
-
-Applications:
-
-* In previous Asterisk releases, many applications would jump to priority n+101
- to indicate some kind of status or error condition. This functionality was
- marked deprecated in Asterisk 1.2. An option to disable it was provided with
- the default value set to 'on'. The default value for the global priority
- jumping option is now 'off'.
-
-* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
- AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
- and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
- been removed in this version. You should use the equivalent dialplan
- function in places where you have previously used one of these applications.
-
-* The application SetGlobalVar has been deprecated. You should replace uses
- of this application with the following combination of Set and GLOBAL():
- Set(GLOBAL(name)=value). You may also access global variables exclusively by
- using the GLOBAL() dialplan function, instead of relying on variable
- interpolation falling back to globals when no channel variable is set.
-
-* The application SetVar has been renamed to Set. The syntax SetVar was marked
- deprecated in version 1.2 and is no longer recognized in this version.
-
-* app_read has been updated to use the newer options codes, using "skip" or
- "noanswer" will not work. Use s or n. Also there is a new feature i, for
- using indication tones, so typing in skip would give you unexpected results.
-
-* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
-
-* The CONNECT event in the queue_log from app_queue now has a second field
- in addition to the holdtime field. It contains the unique ID of the
- queue member channel that is taking the call. This is useful when trying
- to link recording filenames back to a particular call from the queue.
-
-* The old/current behavior of app_queue has a serial type behavior
- in that the queue will make all waiting callers wait in the queue
- even if there is more than one available member ready to take
- calls until the head caller is connected with the member they
- were trying to get to. The next waiting caller in line then
- becomes the head caller, and they are then connected with the
- next available member and all available members and waiting callers
- waits while this happens. This cycle continues until there are
- no more available members or waiting callers, whichever comes first.
- The new behavior, enabled by setting autofill=yes in queues.conf
- either at the [general] level to default for all queues or
- to set on a per-queue level, makes sure that when the waiting
- callers are connecting with available members in a parallel fashion
- until there are no more available members or no more waiting callers,
- whichever comes first. This is probably more along the lines of how
- one would expect a queue should work and in most cases, you will want
- to enable this new behavior. If you do not specify or comment out this
- option, it will default to "no" to keep backward compatability with the old
- behavior.
-
-* The app_queue application now has the ability to use MixMonitor to
- record conversations queue members are having with queue callers. Please
- see configs/queues.conf.sample for more information on this option.
-
-* The app_queue application strategy called 'roundrobin' has been deprecated
- for this release. Users are encouraged to use 'rrmemory' instead, since it
- provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
- 'rrmemory' will be renamed 'roundrobin'.
-
-* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
- the 'm' option now provides the functionality of "initially muted".
- In practice, most existing dialplans using the 'm' flag should not notice
- any difference, unless the keypad menu is enabled, allowing the user
- to unmute themsleves.
-
-* ast_play_and_record would attempt to cancel the recording if a DTMF
- '0' was received. This behavior was not documented in most of the
- applications that used ast_play_and_record and the return codes from
- ast_play_and_record weren't checked for properly.
- ast_play_and_record has been changed so that '0' no longer cancels a
- recording. If you want to allow DTMF digits to cancel an
- in-progress recording use ast_play_and_record_full which allows you
- to specify which DTMF digits can be used to accept a recording and
- which digits can be used to cancel a recording.
-
-* ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
- new ast_app_messagecount function which takes a single context/mailbox/folder
- mailbox specification and returns the message count for that folder only.
- This addresses the deficiency of not being able to count the number of
- messages in folders other than INBOX and Old.
-
-* The exit behavior of the AGI applications has changed. Previously, when
- a connection to an AGI server failed, the application would cause the channel
- to immediately stop dialplan execution and hangup. Now, the only time that
- the AGI applications will cause the channel to stop dialplan execution is
- when the channel itself requests hangup. The AGI applications now set an
- AGISTATUS variable which will allow you to find out whether running the AGI
- was successful or not.
-
- Previously, there was no way to handle the case where Asterisk was unable to
- locally execute an AGI script for some reason. In this case, dialplan
- execution will continue as it did before, but the AGISTATUS variable will be
- set to "FAILURE".
-
- A locally executed AGI script can now exit with a non-zero exit code and this
- failure will be detected by Asterisk. If an AGI script exits with a non-zero
- exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
- "SUCCESS".
-
-* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
- previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
- your table format using the schema provided in doc/odbcstorage.txt
-
Manager:
-* After executing the 'status' manager action, the "Status" manager events
- included the header "CallerID:" which was actually only the CallerID number,
- and not the full CallerID string. This header has been renamed to
- "CallerIDNum". For compatibility purposes, the CallerID parameter will remain
- until after the release of 1.4, when it will be removed. Please use the time
- during the 1.4 release to make this transition.
-
-* The AgentConnect event now has an additional field called "BridgedChannel"
- which contains the unique ID of the queue member channel that is taking the
- call. This is useful when trying to link recording filenames back to
- a particular call from the queue.
-
-* app_userevent has been modified to always send Event: UserEvent with the
- additional header UserEvent: <userspec>. Also, the Channel and UniqueID
[... 49046 lines stripped ...]
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