[asterisk-commits] oej: branch oej/astum r43935 - in /team/oej/astum: ./ apps/ build_tools/ cdr/...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Thu Sep 28 11:20:38 MST 2006 

Author: oej
Date: Thu Sep 28 13:20:37 2006
New Revision: 43935

URL: http://svn.digium.com/view/asterisk?rev=43935&view=rev
Log:
Reset automerge on this branch.

Don't really know what to do with this work, now that Mark made
it obsolete...

Added:
    team/oej/astum/build_tools/strip_nonapi
      - copied unchanged from r43921, trunk/build_tools/strip_nonapi
    team/oej/astum/channels/chan_gtalk.c
      - copied unchanged from r43921, trunk/channels/chan_gtalk.c
    team/oej/astum/channels/h323/Makefile.in
      - copied unchanged from r43921, trunk/channels/h323/Makefile.in
    team/oej/astum/channels/h323/ast_h323.cxx
      - copied unchanged from r43921, trunk/channels/h323/ast_h323.cxx
    team/oej/astum/channels/h323/caps_h323.cxx
      - copied unchanged from r43921, trunk/channels/h323/caps_h323.cxx
    team/oej/astum/channels/h323/caps_h323.h
      - copied unchanged from r43921, trunk/channels/h323/caps_h323.h
    team/oej/astum/channels/h323/cisco-h225.asn
      - copied unchanged from r43921, trunk/channels/h323/cisco-h225.asn
    team/oej/astum/channels/h323/cisco-h225.cxx
      - copied unchanged from r43921, trunk/channels/h323/cisco-h225.cxx
    team/oej/astum/channels/h323/cisco-h225.h
      - copied unchanged from r43921, trunk/channels/h323/cisco-h225.h
    team/oej/astum/channels/h323/compat_h323.cxx
      - copied unchanged from r43921, trunk/channels/h323/compat_h323.cxx
    team/oej/astum/channels/h323/compat_h323.h
      - copied unchanged from r43921, trunk/channels/h323/compat_h323.h
    team/oej/astum/channels/h323/noexport.map
      - copied unchanged from r43921, trunk/channels/h323/noexport.map
    team/oej/astum/configs/gtalk.conf.sample
      - copied unchanged from r43921, trunk/configs/gtalk.conf.sample
    team/oej/astum/configs/h323.conf.sample
      - copied unchanged from r43921, trunk/configs/h323.conf.sample
    team/oej/astum/doc/rtp-packetization.txt
      - copied unchanged from r43921, trunk/doc/rtp-packetization.txt
    team/oej/astum/funcs/func_blacklist.c
      - copied unchanged from r43921, trunk/funcs/func_blacklist.c
    team/oej/astum/funcs/func_vmcount.c
      - copied unchanged from r43921, trunk/funcs/func_vmcount.c
    team/oej/astum/include/jitterbuf.h
      - copied unchanged from r43921, trunk/include/jitterbuf.h
    team/oej/astum/res/res_limit.c
      - copied unchanged from r43921, trunk/res/res_limit.c
    team/oej/astum/res/res_realtime.c
      - copied unchanged from r43921, trunk/res/res_realtime.c
Removed:
    team/oej/astum/apps/app_hasnewvoicemail.c
    team/oej/astum/apps/app_lookupblacklist.c
    team/oej/astum/apps/app_lookupcidname.c
    team/oej/astum/apps/app_random.c
    team/oej/astum/apps/app_realtime.c
    team/oej/astum/apps/app_setcdruserfield.c
    team/oej/astum/apps/app_settransfercapability.c
    team/oej/astum/channels/h323/Makefile
    team/oej/astum/channels/h323/ast_h323.cpp
    team/oej/astum/channels/h323/h323.conf.sample
    team/oej/astum/funcs/func_language.c
    team/oej/astum/funcs/func_moh.c
    team/oej/astum/main/jitterbuf.h
Modified:
    team/oej/astum/   (props changed)
    team/oej/astum/.cleancount
    team/oej/astum/CHANGES
    team/oej/astum/CREDITS
    team/oej/astum/Makefile
    team/oej/astum/Makefile.moddir_rules
    team/oej/astum/UPGRADE.txt
    team/oej/astum/acinclude.m4
    team/oej/astum/apps/app_adsiprog.c
    team/oej/astum/apps/app_alarmreceiver.c
    team/oej/astum/apps/app_chanspy.c
    team/oej/astum/apps/app_dial.c
    team/oej/astum/apps/app_directory.c
    team/oej/astum/apps/app_festival.c
    team/oej/astum/apps/app_followme.c
    team/oej/astum/apps/app_getcpeid.c
    team/oej/astum/apps/app_meetme.c
    team/oej/astum/apps/app_mixmonitor.c
    team/oej/astum/apps/app_osplookup.c
    team/oej/astum/apps/app_playback.c
    team/oej/astum/apps/app_privacy.c
    team/oej/astum/apps/app_queue.c
    team/oej/astum/apps/app_rpt.c
    team/oej/astum/apps/app_setcallerid.c
    team/oej/astum/apps/app_stack.c
    team/oej/astum/apps/app_url.c
    team/oej/astum/apps/app_voicemail.c
    team/oej/astum/build_tools/cflags.xml
    team/oej/astum/build_tools/make_version
    team/oej/astum/build_tools/menuselect-deps.in
    team/oej/astum/build_tools/prep_moduledeps
    team/oej/astum/cdr/cdr_csv.c
    team/oej/astum/cdr/cdr_odbc.c
    team/oej/astum/cdr/cdr_pgsql.c
    team/oej/astum/cdr/cdr_radius.c
    team/oej/astum/cdr/cdr_tds.c
    team/oej/astum/channels/Makefile
    team/oej/astum/channels/chan_agent.c
    team/oej/astum/channels/chan_alsa.c
    team/oej/astum/channels/chan_features.c
    team/oej/astum/channels/chan_h323.c
    team/oej/astum/channels/chan_iax2.c
    team/oej/astum/channels/chan_jingle.c
    team/oej/astum/channels/chan_local.c
    team/oej/astum/channels/chan_mgcp.c
    team/oej/astum/channels/chan_misdn.c
    team/oej/astum/channels/chan_oss.c
    team/oej/astum/channels/chan_sip.c
    team/oej/astum/channels/chan_skinny.c
    team/oej/astum/channels/chan_zap.c
    team/oej/astum/channels/h323/   (props changed)
    team/oej/astum/channels/h323/README
    team/oej/astum/channels/h323/TODO
    team/oej/astum/channels/h323/ast_h323.h
    team/oej/astum/channels/h323/chan_h323.h
    team/oej/astum/channels/iax2-provision.c
    team/oej/astum/channels/misdn/isdn_lib.c
    team/oej/astum/channels/misdn/isdn_lib.h
    team/oej/astum/codecs/gsm/Makefile
    team/oej/astum/configs/extensions.ael.sample
    team/oej/astum/configs/logger.conf.sample
    team/oej/astum/configs/queues.conf.sample
    team/oej/astum/configs/res_odbc.conf.sample
    team/oej/astum/configs/sip.conf.sample
    team/oej/astum/configs/skinny.conf.sample
    team/oej/astum/configs/zapata.conf.sample
    team/oej/astum/configure
    team/oej/astum/configure.ac
    team/oej/astum/doc/ael.txt
    team/oej/astum/doc/channelvariables.txt
    team/oej/astum/doc/ip-tos.txt
    team/oej/astum/doc/jingle.txt
    team/oej/astum/doc/mp3.txt
    team/oej/astum/doc/realtime.txt
    team/oej/astum/doc/security.txt
    team/oej/astum/formats/format_g723.c
    team/oej/astum/formats/format_ogg_vorbis.c
    team/oej/astum/formats/format_pcm.c
    team/oej/astum/funcs/func_callerid.c
    team/oej/astum/funcs/func_md5.c
    team/oej/astum/funcs/func_odbc.c
    team/oej/astum/include/asterisk/acl.h
    team/oej/astum/include/asterisk/adsi.h
    team/oej/astum/include/asterisk/agi.h
    team/oej/astum/include/asterisk/autoconfig.h.in
    team/oej/astum/include/asterisk/channel.h
    team/oej/astum/include/asterisk/cli.h
    team/oej/astum/include/asterisk/compat.h
    team/oej/astum/include/asterisk/compiler.h
    team/oej/astum/include/asterisk/config.h
    team/oej/astum/include/asterisk/frame.h
    team/oej/astum/include/asterisk/jabber.h
    team/oej/astum/include/asterisk/lock.h
    team/oej/astum/include/asterisk/monitor.h
    team/oej/astum/include/asterisk/res_odbc.h
    team/oej/astum/include/asterisk/rtp.h
    team/oej/astum/include/asterisk/threadstorage.h
    team/oej/astum/main/Makefile
    team/oej/astum/main/acl.c
    team/oej/astum/main/asterisk.c
    team/oej/astum/main/astmm.c
    team/oej/astum/main/cdr.c
    team/oej/astum/main/channel.c
    team/oej/astum/main/cli.c
    team/oej/astum/main/config.c
    team/oej/astum/main/db.c
    team/oej/astum/main/dnsmgr.c
    team/oej/astum/main/file.c
    team/oej/astum/main/frame.c
    team/oej/astum/main/http.c
    team/oej/astum/main/image.c
    team/oej/astum/main/logger.c
    team/oej/astum/main/manager.c
    team/oej/astum/main/pbx.c
    team/oej/astum/main/rtp.c
    team/oej/astum/main/slinfactory.c
    team/oej/astum/main/strcompat.c
    team/oej/astum/main/translate.c
    team/oej/astum/main/udptl.c
    team/oej/astum/makeopts.in
    team/oej/astum/pbx/ael/ael-test/ael-test11/extensions.ael
    team/oej/astum/pbx/ael/ael-test/ael-test3/extensions.ael
    team/oej/astum/pbx/ael/ael-test/ael-test5/extensions.ael
    team/oej/astum/pbx/ael/ael-test/ref.ael-ntest10
    team/oej/astum/pbx/ael/ael-test/ref.ael-test1
    team/oej/astum/pbx/ael/ael-test/ref.ael-test2
    team/oej/astum/pbx/ael/ael-test/ref.ael-test3
    team/oej/astum/pbx/ael/ael-test/ref.ael-test4
    team/oej/astum/pbx/ael/ael-test/ref.ael-test5
    team/oej/astum/pbx/ael/ael-test/ref.ael-test6
    team/oej/astum/pbx/ael/ael-test/ref.ael-test7
    team/oej/astum/pbx/ael/ael-test/ref.ael-vtest13
    team/oej/astum/pbx/pbx_ael.c
    team/oej/astum/pbx/pbx_config.c
    team/oej/astum/pbx/pbx_dundi.c
    team/oej/astum/res/res_adsi.c
    team/oej/astum/res/res_agi.c
    team/oej/astum/res/res_clioriginate.c
    team/oej/astum/res/res_config_odbc.c
    team/oej/astum/res/res_config_pgsql.c
    team/oej/astum/res/res_convert.c
    team/oej/astum/res/res_crypto.c
    team/oej/astum/res/res_features.c
    team/oej/astum/res/res_indications.c
    team/oej/astum/res/res_jabber.c
    team/oej/astum/res/res_monitor.c
    team/oej/astum/res/res_musiconhold.c
    team/oej/astum/res/res_odbc.c
    team/oej/astum/sounds/Makefile
    team/oej/astum/utils/Makefile
    team/oej/astum/utils/muted.c

Propchange: team/oej/astum/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

Propchange: team/oej/astum/
            ('branch-1.2-blocked' removed)

Propchange: team/oej/astum/
            ('branch-1.2-merged' removed)

Propchange: team/oej/astum/
------------------------------------------------------------------------------
    branch-1.4-blocked = /branches/1.4:43484,43510,43582,43626,43703,43756

Propchange: team/oej/astum/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Propchange: team/oej/astum/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu Sep 28 13:20:37 2006
@@ -1,1 +1,1 @@
-/trunk:1-43176
+/trunk:1-43921

Modified: team/oej/astum/.cleancount
URL: http://svn.digium.com/view/asterisk/team/oej/astum/.cleancount?rev=43935&r1=43934&r2=43935&view=diff
==============================================================================
--- team/oej/astum/.cleancount (original)
+++ team/oej/astum/.cleancount Thu Sep 28 13:20:37 2006
@@ -1,1 +1,1 @@
-23
+25

Modified: team/oej/astum/CHANGES
URL: http://svn.digium.com/view/asterisk/team/oej/astum/CHANGES?rev=43935&r1=43934&r2=43935&view=diff
==============================================================================
--- team/oej/astum/CHANGES (original)
+++ team/oej/astum/CHANGES Thu Sep 28 13:20:37 2006
@@ -1,127 +1,19 @@
-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)

Modified: team/oej/astum/CREDITS
URL: http://svn.digium.com/view/asterisk/team/oej/astum/CREDITS?rev=43935&r1=43934&r2=43935&view=diff
==============================================================================
--- team/oej/astum/CREDITS (original)
+++ team/oej/astum/CREDITS Thu Sep 28 13:20:37 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/astum/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/astum/Makefile?rev=43935&r1=43934&r2=43935&view=diff
==============================================================================
--- team/oej/astum/Makefile (original)
+++ team/oej/astum/Makefile Thu Sep 28 13:20:37 2006
@@ -106,7 +106,7 @@
 GLOBAL_MAKEOPTS=$(wildcard /etc/asterisk.makeopts)
 USER_MAKEOPTS=$(wildcard ~/.asterisk.makeopts)
 
-MOD_SUBDIR_CFLAGS=-I../include -I../main
+MOD_SUBDIR_CFLAGS=-I../include
 OTHER_SUBDIR_CFLAGS=-I../include
 
 ifeq ($(OSARCH),linux-gnu)
@@ -140,13 +140,6 @@
       endif
     endif
   endif
-endif
-
-ID=id
-
-ifeq ($(OSARCH),SunOS)
-  M4=/usr/local/bin/m4
-  ID=/usr/xpg4/bin/id
 endif
 
 ASTCFLAGS+=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG)
@@ -238,13 +231,15 @@
   HAVEDOT=no
 endif
 
-all: cleantest $(SUBDIRS)
+all: _all
 	@echo " +--------- Asterisk Build Complete ---------+"  
 	@echo " + Asterisk has successfully been built, and +"  
 	@echo " + can be installed by running:              +"
 	@echo " +                                           +"
 	@echo " +               make install                +"  
 	@echo " +-------------------------------------------+"  
+
+_all: cleantest $(SUBDIRS)
 
 makeopts: configure
 	@echo "****"
@@ -336,7 +331,7 @@
 	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 $(DESTDIR)/usr/lib/pkgconfig; fi
 # Should static HTTP be installed during make samples or even with its own target ala
 # webvoicemail?  There are portions here that *could* be customized but might also be
@@ -370,7 +365,7 @@
 NEWHEADERS=$(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)
@@ -434,7 +429,7 @@
 		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 $(DESTDIR) . ; \
 	fi
@@ -461,7 +456,7 @@
 	@echo " +-------------------------------------------+"
 	@$(MAKE) -s oldmodcheck
 
-upgrade: all bininstall
+upgrade: bininstall
 
 adsi:
 	mkdir -p $(DESTDIR)$(ASTETCDIR)
@@ -648,7 +643,7 @@
 	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/astum/Makefile.moddir_rules
URL: http://svn.digium.com/view/asterisk/team/oej/astum/Makefile.moddir_rules?rev=43935&r1=43934&r2=43935&view=diff
==============================================================================
--- team/oej/astum/Makefile.moddir_rules (original)
+++ team/oej/astum/Makefile.moddir_rules Thu Sep 28 13:20:37 2006
@@ -34,7 +34,7 @@
 
 define module_so_template
 $(1)=$(1).so
-$(1).so: CFLAGS+=-fpic
+$(1).so: CFLAGS+=-fPIC
 $(1).so: LIBS+=$(foreach dep,$(MENUSELECT_DEPENDS_$(1)),$$(value $(dep)_LIB))
 $(1).so: LDFLAGS+=$(foreach dep,$(MENUSELECT_DEPENDS_$(1)),$$(value $(dep)_LDFLAGS))
 $(1).so: $(2)

Modified: team/oej/astum/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/oej/astum/UPGRADE.txt?rev=43935&r1=43934&r2=43935&view=diff
==============================================================================
--- team/oej/astum/UPGRADE.txt (original)
+++ team/oej/astum/UPGRADE.txt Thu Sep 28 13:20:37 2006
@@ -1,416 +1,3 @@
 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
-
-* app_waitforsilence: Fixes have been made to this application which changes the 
-  default behavior with how quickly it returns. You can maintain "old-style" behavior
-  with the addition/use of a third "timeout" parameter.
-  Please consult the application documentation and make changes to your dialplan 
-  if appropriate.
-
-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
-  headers are not automatically sent, unless you specify them as separate
-  arguments.  Please see the application help for the new syntax.
-
-* app_meetme: Mute and Unmute events are now reported via the Manager API.
-  Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
-  are easier to use than "Action Command:". The MeetMeStopTalking event has
-  also been deprecated in favor of the already existing MeetmeTalking event
-  with a "Status" of "on" or "off" added.
-
-Variables:
-
-* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
-  ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
-  and ${LANGUAGE} have all been deprecated in favor of their related dialplan
-  functions.  You are encouraged to move towards the associated dialplan
-  function, as these variables will be removed in a future release.
-
-* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now 
-  adjustable from cdr.conf, instead of recompiling.
-
-* OSP applications exports several new variables, ${OSPINHANDLE},
-  ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
-  ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
-  
-* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
-  created channel. This variables holds the channel name of the transferer.
-
-* The dial plan variable PRI_CAUSE will be removed from future versions 
-  of Asterisk.
-  It is replaced by adding a cause value to the hangup() application.
-
-Functions:
-
-* The function ${CHECK_MD5()} has been deprecated in favor of using an
-  expression: $[${MD5(<string>)} = ${saved_md5}].
-
-* The 'builtin' functions that used to be combined in pbx_functions.so are
-  now built as separate modules. If you are not using 'autoload=yes' in your
-  modules.conf file then you will need to explicitly load the modules that
-  contain the functions you want to use.
-
-* The ENUMLOOKUP() function with the 'c' option (for counting the number of 
-  records), but the lookup fails to match any records, the returned value will 
-  now be "0" instead of blank.
-
-* The REALTIME() function is now available in version 1.4 and app_realtime has
-  been deprecated in favor of the new function. app_realtime will be removed
-  completely with the version 1.6 release so please take the time between
-  releases to make any necessary changes
-
-* The QUEUEAGENTCOUNT() function has been deprecated in favor of
-  QUEUE_MEMBER_COUNT().
-
-The IAX2 channel:
-
-* The "mailboxdetail" option has been deprecated.  Previously, if this option
-  was not enabled, the 2 byte MSGCOUNT information element would be set to all
-  1's to indicate there there is some number of messages waiting.  With this
-  option enabled, the number of new messages were placed in one byte and the
-  number of old messages are placed in the other.  This is now the default
-  (and the only) behavior.
-
-The SIP channel:
-
-* The "incominglimit" setting is replaced by the "call-limit" setting in 
-  sip.conf.
-
-* OSP support code is removed from SIP channel to OSP applications. ospauth 
-  option in sip.conf is removed to osp.conf as authpolicy. allowguest option
-  in sip.conf cannot be set as osp anymore. 
-
-* The Asterisk RTP stack has been changed in regards to RFC2833 reception
-  and transmission. Packets will now be sent with proper duration instead of all
-  at once. If you are receiving calls from a pre-1.4 Asterisk installation you
-  will want to turn on the rfc2833compensate option. Without this option your
-  DTMF reception may act poorly.
-
-* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
-  in coming versions of Asterisk. Please use the dialplan function
-  SIPCHANINFO(useragent) instead.
-
-* The ALERT_INFO dialplan variable is deprecated and will be removed
-  in coming versions of Asterisk. Please use the dialplan application
-  sipaddheader() to add the "Alert-Info" header to the outbound invite.
-
-The Zap channel:
-
-* Support for MFC/R2 has been removed, as it has not been functional for some
-  time and it has no maintainer.
-
-The Agent channel:
-
-* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
-  it provided can be done using dialplan logic, without requiring additional
-  channel and module locks (which frequently caused deadlocks). An example of
-  how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
-
-The G726-32 codec:
-
-* It has been determined that previous versions of Asterisk used the wrong codeword
-  packing order for G726-32 data. This version supports both available packing orders,
-  and can transcode between them. It also now selects the proper order when
-  negotiating with a SIP peer based on the codec name supplied in the SDP. However,
-  there are existing devices that improperly request one order and then use another;
-  Sipura and Grandstream ATAs are known to do this, and there may be others. To
-  be able to continue to use these devices with this version of Asterisk and the
-  G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
-  to sip.conf, so that Asterisk can use the packing order expected by the device (even
-  though it requested a different order). In addition, the internal format number for
-  G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
-  result of this is that this version of Asterisk will be able to interoperate over
-  IAX2 with older versions of Asterisk, as long as this version is told to allow
-  'g726aal2' instead of 'g726' as the codec for the call.
-

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