[asterisk-commits] oej: branch oej/astum r99593 - in /team/oej/astum: ./ apps/ build_tools/ cdr/...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 22 11:55:31 CST 2008
Author: oej
Date: Tue Jan 22 11:35:07 2008
New Revision: 99593
URL: http://svn.digium.com/view/asterisk?view=rev&rev=99593
Log:
Reset, resolve
Added:
team/oej/astum/apps/app_jack.c
- copied unchanged from r99542, trunk/apps/app_jack.c
team/oej/astum/apps/app_pickupchan.c
- copied unchanged from r99542, trunk/apps/app_pickupchan.c
team/oej/astum/build_tools/make_version_c
- copied unchanged from r99542, trunk/build_tools/make_version_c
team/oej/astum/channels/chan_console.c
- copied unchanged from r99542, trunk/channels/chan_console.c
team/oej/astum/channels/console_board.c
- copied unchanged from r99542, trunk/channels/console_board.c
team/oej/astum/channels/console_gui.c
- copied unchanged from r99542, trunk/channels/console_gui.c
team/oej/astum/channels/vcodecs.c
- copied unchanged from r99542, trunk/channels/vcodecs.c
team/oej/astum/channels/vgrabbers.c
- copied unchanged from r99542, trunk/channels/vgrabbers.c
team/oej/astum/codecs/codec_resample.c
- copied unchanged from r99542, trunk/codecs/codec_resample.c
team/oej/astum/codecs/slin_resample_ex.h
- copied unchanged from r99542, trunk/codecs/slin_resample_ex.h
team/oej/astum/configs/console.conf.sample
- copied unchanged from r99542, trunk/configs/console.conf.sample
team/oej/astum/configs/phoneprov.conf.sample
- copied unchanged from r99542, trunk/configs/phoneprov.conf.sample
team/oej/astum/doc/siptls.txt
- copied unchanged from r99542, trunk/doc/siptls.txt
team/oej/astum/doc/tex/phoneprov.tex
- copied unchanged from r99542, trunk/doc/tex/phoneprov.tex
team/oej/astum/formats/format_sln16.c
- copied unchanged from r99542, trunk/formats/format_sln16.c
team/oej/astum/funcs/func_dialplan.c
- copied unchanged from r99542, trunk/funcs/func_dialplan.c
team/oej/astum/images/font.png
- copied unchanged from r99542, trunk/images/font.png
team/oej/astum/include/asterisk/libresample.h
- copied unchanged from r99542, trunk/include/asterisk/libresample.h
team/oej/astum/include/asterisk/tcptls.h
- copied unchanged from r99542, trunk/include/asterisk/tcptls.h
team/oej/astum/main/libresample/ (props changed)
- copied from r99542, trunk/main/libresample/
team/oej/astum/main/libresample/LICENSE.txt
- copied unchanged from r99542, trunk/main/libresample/LICENSE.txt
team/oej/astum/main/libresample/Makefile.asterisk
- copied unchanged from r99542, trunk/main/libresample/Makefile.asterisk
team/oej/astum/main/libresample/Makefile.in
- copied unchanged from r99542, trunk/main/libresample/Makefile.in
team/oej/astum/main/libresample/README.txt
- copied unchanged from r99542, trunk/main/libresample/README.txt
team/oej/astum/main/libresample/config.guess
- copied unchanged from r99542, trunk/main/libresample/config.guess
team/oej/astum/main/libresample/config.sub
- copied unchanged from r99542, trunk/main/libresample/config.sub
team/oej/astum/main/libresample/configure
- copied unchanged from r99542, trunk/main/libresample/configure
team/oej/astum/main/libresample/configure.in
- copied unchanged from r99542, trunk/main/libresample/configure.in
team/oej/astum/main/libresample/include/
- copied from r99542, trunk/main/libresample/include/
team/oej/astum/main/libresample/include/libresample.h
- copied unchanged from r99542, trunk/main/libresample/include/libresample.h
team/oej/astum/main/libresample/install-sh
- copied unchanged from r99542, trunk/main/libresample/install-sh
team/oej/astum/main/libresample/src/
- copied from r99542, trunk/main/libresample/src/
team/oej/astum/main/libresample/src/configtemplate.h
- copied unchanged from r99542, trunk/main/libresample/src/configtemplate.h
team/oej/astum/main/libresample/src/filterkit.c
- copied unchanged from r99542, trunk/main/libresample/src/filterkit.c
team/oej/astum/main/libresample/src/filterkit.h
- copied unchanged from r99542, trunk/main/libresample/src/filterkit.h
team/oej/astum/main/libresample/src/resample.c
- copied unchanged from r99542, trunk/main/libresample/src/resample.c
team/oej/astum/main/libresample/src/resample_defs.h
- copied unchanged from r99542, trunk/main/libresample/src/resample_defs.h
team/oej/astum/main/libresample/src/resamplesubs.c
- copied unchanged from r99542, trunk/main/libresample/src/resamplesubs.c
team/oej/astum/main/libresample/tests/
- copied from r99542, trunk/main/libresample/tests/
team/oej/astum/main/libresample/tests/compareresample.c
- copied unchanged from r99542, trunk/main/libresample/tests/compareresample.c
team/oej/astum/main/libresample/tests/resample-sndfile.c
- copied unchanged from r99542, trunk/main/libresample/tests/resample-sndfile.c
team/oej/astum/main/libresample/tests/testresample.c
- copied unchanged from r99542, trunk/main/libresample/tests/testresample.c
team/oej/astum/main/libresample/win/
- copied from r99542, trunk/main/libresample/win/
team/oej/astum/main/libresample/win/libresample.dsp
- copied unchanged from r99542, trunk/main/libresample/win/libresample.dsp
team/oej/astum/main/libresample/win/libresample.vcproj
- copied unchanged from r99542, trunk/main/libresample/win/libresample.vcproj
team/oej/astum/main/tcptls.c
- copied unchanged from r99542, trunk/main/tcptls.c
team/oej/astum/phoneprov/
- copied from r99542, trunk/phoneprov/
team/oej/astum/phoneprov/000000000000-directory.xml
- copied unchanged from r99542, trunk/phoneprov/000000000000-directory.xml
team/oej/astum/phoneprov/000000000000-phone.cfg
- copied unchanged from r99542, trunk/phoneprov/000000000000-phone.cfg
team/oej/astum/phoneprov/000000000000.cfg
- copied unchanged from r99542, trunk/phoneprov/000000000000.cfg
team/oej/astum/phoneprov/polycom.xml
- copied unchanged from r99542, trunk/phoneprov/polycom.xml
team/oej/astum/res/res_config_curl.c
- copied unchanged from r99542, trunk/res/res_config_curl.c
team/oej/astum/res/res_phoneprov.c
- copied unchanged from r99542, trunk/res/res_phoneprov.c
team/oej/astum/tests/ (props changed)
- copied from r99542, trunk/tests/
team/oej/astum/tests/Makefile
- copied unchanged from r99542, trunk/tests/Makefile
team/oej/astum/tests/test_skel.c
- copied unchanged from r99542, trunk/tests/test_skel.c
Removed:
team/oej/astum/build_tools/make_version_h
team/oej/astum/channels/answer.h
team/oej/astum/channels/busy.h
team/oej/astum/channels/gentone.c
team/oej/astum/channels/ring10.h
team/oej/astum/channels/ringtone.h
team/oej/astum/pbx/kdeconsole_main.cc
team/oej/astum/pbx/pbx_kdeconsole.cc
team/oej/astum/pbx/pbx_kdeconsole.h
Modified:
team/oej/astum/ (props changed)
team/oej/astum/CHANGES
team/oej/astum/CREDITS
team/oej/astum/Makefile
team/oej/astum/README
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_amd.c
team/oej/astum/apps/app_authenticate.c
team/oej/astum/apps/app_cdr.c
team/oej/astum/apps/app_dictate.c
team/oej/astum/apps/app_directory.c
team/oej/astum/apps/app_exec.c
team/oej/astum/apps/app_followme.c
team/oej/astum/apps/app_meetme.c
team/oej/astum/apps/app_osplookup.c
team/oej/astum/apps/app_queue.c
team/oej/astum/apps/app_rpt.c
team/oej/astum/apps/app_stack.c
team/oej/astum/apps/app_voicemail.c
team/oej/astum/apps/app_zapateller.c
team/oej/astum/apps/app_zapbarge.c
team/oej/astum/apps/app_zapras.c
team/oej/astum/apps/app_zapscan.c
team/oej/astum/build_tools/cflags.xml
team/oej/astum/build_tools/menuselect-deps.in
team/oej/astum/build_tools/prep_tarball
team/oej/astum/build_tools/strip_nonapi
team/oej/astum/cdr/cdr_adaptive_odbc.c
team/oej/astum/cdr/cdr_odbc.c
team/oej/astum/channels/ (props changed)
team/oej/astum/channels/Makefile
team/oej/astum/channels/chan_agent.c
team/oej/astum/channels/chan_alsa.c
team/oej/astum/channels/chan_gtalk.c
team/oej/astum/channels/chan_h323.c
team/oej/astum/channels/chan_iax2.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_usbradio.c
team/oej/astum/channels/chan_zap.c
team/oej/astum/channels/console_video.c
team/oej/astum/channels/console_video.h
team/oej/astum/codecs/Makefile
team/oej/astum/codecs/codec_g722.c
team/oej/astum/codecs/codec_speex.c
team/oej/astum/codecs/codec_zap.c
team/oej/astum/configs/cdr_adaptive_odbc.conf.sample
team/oej/astum/configs/extconfig.conf.sample
team/oej/astum/configs/extensions.conf.sample
team/oej/astum/configs/http.conf.sample
team/oej/astum/configs/manager.conf.sample
team/oej/astum/configs/modules.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/voicemail.conf.sample
team/oej/astum/configs/zapata.conf.sample
team/oej/astum/configure
team/oej/astum/configure.ac
team/oej/astum/doc/asterisk.8
team/oej/astum/doc/manager_1_1.txt
team/oej/astum/doc/tex/Makefile
team/oej/astum/doc/tex/asterisk.tex
team/oej/astum/doc/tex/imapstorage.tex
team/oej/astum/doc/tex/qos.tex
team/oej/astum/doc/tex/realtime.tex
team/oej/astum/funcs/func_cut.c
team/oej/astum/funcs/func_env.c
team/oej/astum/funcs/func_groupcount.c
team/oej/astum/funcs/func_iconv.c
team/oej/astum/funcs/func_odbc.c
team/oej/astum/funcs/func_strings.c
team/oej/astum/funcs/func_version.c
team/oej/astum/images/kpad2.jpg
team/oej/astum/include/asterisk/ (props changed)
team/oej/astum/include/asterisk/app.h
team/oej/astum/include/asterisk/autoconfig.h.in
team/oej/astum/include/asterisk/channel.h
team/oej/astum/include/asterisk/config.h
team/oej/astum/include/asterisk/frame.h
team/oej/astum/include/asterisk/http.h
team/oej/astum/include/asterisk/localtime.h
team/oej/astum/include/asterisk/manager.h
team/oej/astum/include/asterisk/module.h
team/oej/astum/include/asterisk/res_odbc.h
team/oej/astum/include/asterisk/translate.h
team/oej/astum/main/ (props changed)
team/oej/astum/main/Makefile
team/oej/astum/main/abstract_jb.c
team/oej/astum/main/acl.c
team/oej/astum/main/ast_expr2.c
team/oej/astum/main/ast_expr2.fl
team/oej/astum/main/ast_expr2.h
team/oej/astum/main/ast_expr2.y
team/oej/astum/main/ast_expr2f.c
team/oej/astum/main/asterisk.c
team/oej/astum/main/audiohook.c
team/oej/astum/main/autoservice.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/devicestate.c
team/oej/astum/main/dial.c
team/oej/astum/main/dsp.c
team/oej/astum/main/editline/readline.c
team/oej/astum/main/frame.c
team/oej/astum/main/http.c
team/oej/astum/main/io.c
team/oej/astum/main/loader.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/stdtime/localtime.c
team/oej/astum/main/translate.c
team/oej/astum/main/utils.c
team/oej/astum/makeopts.in
team/oej/astum/pbx/pbx_config.c
team/oej/astum/pbx/pbx_dundi.c
team/oej/astum/pbx/pbx_gtkconsole.c
team/oej/astum/pbx/pbx_loopback.c
team/oej/astum/pbx/pbx_realtime.c
team/oej/astum/pbx/pbx_spool.c
team/oej/astum/res/Makefile
team/oej/astum/res/ael/ael.flex
team/oej/astum/res/ael/ael_lex.c
team/oej/astum/res/res_agi.c
team/oej/astum/res/res_config_pgsql.c
team/oej/astum/res/res_features.c
team/oej/astum/res/res_jabber.c
team/oej/astum/res/res_odbc.c
team/oej/astum/res/res_smdi.c
team/oej/astum/res/snmp/agent.c
team/oej/astum/utils/ (props changed)
team/oej/astum/utils/astman.c
team/oej/astum/utils/extconf.c
Change Statistics:
0 files changed
Propchange: team/oej/astum/
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automerge = http://www.codename-pineapple.org/
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Binary property 'branch-1.4-blocked' - no diff available.
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--- svn:ignore (original)
+++ svn:ignore Tue Jan 22 11:35:07 2008
@@ -23,3 +23,4 @@
autom4te.cache
makeopts.embed_rules
aclocal.m4
+update.log
Propchange: team/oej/astum/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jan 22 11:35:07 2008
@@ -1,1 +1,1 @@
-/trunk:1-94749
+/trunk:1-99542
Modified: team/oej/astum/CHANGES
URL: http://svn.digium.com/view/asterisk/team/oej/astum/CHANGES?view=diff&rev=99593&r1=99592&r2=99593
==============================================================================
--- team/oej/astum/CHANGES (original)
+++ team/oej/astum/CHANGES Tue Jan 22 11:35:07 2008
@@ -35,29 +35,37 @@
* Added Masquerade manager event for when a masquerade happens between
two channels.
* Added "manager reload" command for the CLI
+ * Lots of commands that only provided information are now allowed under the
+ Reporting privilege, instead of only under Call or System.
+ * The IAX* commands now require either System or Reporting privilege, to
+ mirror the privileges of the SIP* commands.
Dialplan functions
------------------
* Added the DEVICE_STATE() dialplan function which allows retrieving any device
- state in the dialplan, as well as creating custom device states that are
- controllable from the dialplan.
+ state in the dialplan, as well as creating custom device states that are
+ controllable from the dialplan.
* Extend CALLERID() function with "pres" and "ton" parameters to
fetch string representation of calling number presentation indicator
and numeric representation of type of calling number value.
* MailboxExists converted to dialplan function
* A new option to Dial() for telling IP phones not to count the call
- as "missed" when dial times out and cancels.
+ as "missed" when dial times out and cancels.
* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
- mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
- held for any given channel. Also, locks are automatically freed when a
- channel is hung up.
+ mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
+ held for any given channel. Also, locks are automatically freed when a
+ channel is hung up.
* Added HINT() dialplan function that allows retrieving hint information.
- Hints are mappings between extensions and devices for the sake of
- determining the state of an extension. This function can retrieve the list
- of devices or the name associated with a hint.
+ Hints are mappings between extensions and devices for the sake of
+ determining the state of an extension. This function can retrieve the list
+ of devices or the name associated with a hint.
* Added EXTENSION_STATE() dialplan function which allows retrieving the state
of any extension.
* Added SYSINFO() dialplan function which allows retrieval of system information
+ * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
+ the existence of a dialplan target.
+ * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
+ upper and lower case, respectively.
CLI Changes
-----------
@@ -70,6 +78,10 @@
* Enhanced "agi debug" to print the channel name as a prefix to the debug
output to make debugging on busy systems much easier.
* New CLI commands "dialplan set extenpatternmatching true/false"
+ * New CLI command: "core set chanvar" to set a channel variable from the CLI.
+ * Added an easy way to execute Asterisk CLI commands at startup. Any commands
+ listed in the startup_commands file in the Asterisk configuration directory
+ will get executed.
SIP changes
-----------
@@ -107,13 +119,20 @@
states it is not needed. For phones, however, that do require it the "registertrying" option
has been added so it can be enabled.
* A new option called "callcounter" (global/peer/user level) enables call counters needed
- for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
- used to enable this functionality).
+ for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
+ used to enable this functionality).
* New settings for timer T1 and timer B on a global level or per device. This makes it
- possible to force timeout faster on non-responsive SIP servers. These settings are
- considered advanced, so don't use them unless you have a problem.
+ possible to force timeout faster on non-responsive SIP servers. These settings are
+ considered advanced, so don't use them unless you have a problem.
* Added a dial string option to be able to set the To: header in an INVITE to any
- SIP uri.
+ SIP uri.
+ * Added a new global and per-peer option, qualifyfreq, which allows you to configure
+ the qualify frequency.
+ * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
+ were not properly torn down due to network or endpoint failures during an established
+ SIP session.
+ * Added TCP and TLS support for SIP. See doc/siptls.txt and configs/sip.conf.sample for
+ more information on how it is used.
IAX2 changes
------------
@@ -136,9 +155,11 @@
------------
* Added separate settings for media QoS in mgcp.conf
-OSS Channel changes
+Console Channel Driver changes
-------------------
- * Added experimental support for video under X windows
+ * Added experimental support for video send & receive to chan_oss.
+ This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
+ a video source.
Phone channel changes (chan_phone)
----------------------------------
@@ -165,22 +186,37 @@
----------------------------------------
* SS7 support in chan_zap (via libss7 library)
* In India, some carriers transmit CID via dtmf. Some code has been added
- that will handle some situations. The cidstart=polarity_IN choice has been added for
- those carriers that transmit CID via dtmf after a polarity change.
+ that will handle some situations. The cidstart=polarity_IN choice has been added for
+ those carriers that transmit CID via dtmf after a polarity change.
* CID matching information is now shown when doing 'dialplan show'.
* Added zap show version CLI command to chan_zap.
* Added setvar support to zapata.conf channel entries.
* Added two new options: mwimonitor and mwimonitornotify. These options allow
you to enable MWI monitoring on FXO lines. When the MWI state changes,
the script specified in the mwimonitornotify option is executed. An internal
- event indicating the new state of the mailbox is also generated, so that
- the normal MWI facilities in Asterisk work as usual.
-
-A new channel driver: Unistim
------------------------------
+ event indicating the new state of the mailbox is also generated, so that
+ the normal MWI facilities in Asterisk work as usual.
+ * Added signalling type 'auto', which attempts to use the same signalling type
+ for a channel as configured in Zaptel. This is primarily designed for analog
+ ports, but will also work for digital ports that are configured for FXS or FXO
+ signalling types. This mode is also the default now, so if your zapata.conf
+ does not specify signalling for a channel (which is unlikely as the sample
+ configuration file has always recommended specifying it for every channel) then
+ the 'auto' mode will be used for that channel if possible.
+ * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
+ state for a channel; also ensured that the DNDState Manager event is
+ emitted no matter how the DND state is set or cleared.
+
+New Channel Drivers
+-------------------
* Added a new channel driver, chan_unistim. See doc/unistim.txt and
configs/unistim.conf.sample for details. This new channel driver allows
you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
+ * Added a new channel driver, chan_console, which uses portaudio as a cross
+ platform audio interface. It was written as a channel driver that would
+ work with Mac CoreAudio, but portaudio supports a number of other audio
+ interfaces, as well. Note that this channel driver requires v19 or higher
+ of portaudio; older versions have a different API.
DUNDi changes
-------------
@@ -228,6 +264,8 @@
future. The default is the old behavior, lockfile. However, there is a
new method, "flock", that uses a different method for situations where the
lockfile will not work, such as on SMB/CIFS mounts.
+ * Added the ability to backup deleted messages, to ease recovery in the case
+ that a user accidentally deletes a message, and discovers that they need it.
Queue changes
-------------
@@ -256,6 +294,9 @@
* Added new channel variable QUEUE_MIN_PENALTY
* QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
rules in queuerules.conf. See configs/queuerules.conf.sample for details
+ * Added a new parameter for member definition, called state_interface. This may be
+ used so that a member may be called via one interface but have a different interface's
+ device state reported.
MeetMe Changes
--------------
@@ -310,6 +351,8 @@
of asking for verification of each name, one at a time.
* Privacy() no longer uses privacy.conf, as all options are specifyable as
direct options to the app.
+ * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
+ for more details
Music On Hold Changes
---------------------
@@ -319,8 +362,8 @@
to this music on hold class.
* Support for realtime music on hold has been added.
* In conjunction with the realtime music on hold, a general section has
- been added to musiconhold.conf, its sole variable is cachertclasses. If this
- is set, then music on hold classes found in realtime will be cached in memory.
+ been added to musiconhold.conf, its sole variable is cachertclasses. If this
+ is set, then music on hold classes found in realtime will be cached in memory.
AEL Changes
-----------
@@ -342,11 +385,11 @@
fashion: Set(LOCAL(myvar)=someval); ("local" is now
an AEL keyword).
* utils/conf2ael introduced. Will convert an extensions.conf
- file into extensions.ael. Very crude and unfinished, but
- will be improved as time goes by. Should be useful for a
- first pass at conversion.
+ file into extensions.ael. Very crude and unfinished, but
+ will be improved as time goes by. Should be useful for a
+ first pass at conversion.
* aelparse will now read extensions.conf to see if a referenced
- macro or context is there before issueing a warning.
+ macro or context is there before issueing a warning.
Call Features (res_features) Changes
------------------------------------
@@ -384,6 +427,39 @@
command to be run after rotation. This is primarily useful with
rotatestrategry=rotate, to allow a limit on the number of logfiles kept
and to ensure that the oldest log file gets deleted.
+ * Added realtime support for the queue log
+
+Miscellaneous New Modules
+-------------------------
+ * Added a new CDR module, cdr_sqlite3_custom.
+ * Added a new realtime configuration module, res_config_sqlite
+ * Added a new codec translation module, codec_resample, which re-samples
+ signed linear audio between 8 kHz and 16 kHz to help support wideband
+ codecs.
+ * Added a new module, res_phoneprov, which allows auto-provisioning of phones
+ based on configuration templates that use Asterisk dialplan function and
+ variable substitution. It should be possible to create phone profiles and
+ templates that work for the majority of phones provisioned over http. It
+ is currently only intended to provision a single user account per phone.
+ An example profile and set of templates for Polycom phones is provided.
+ NOTE: Polycom firmware is not included, but should be placed in
+ AST_DATA_DIR/phoneprov/configs to match up with the included templates.
+ * Added a new module, app_jack, which provides interfaces to JACK, the Jack
+ Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
+ provided; there is a JACK() application, and a JACK_HOOK() function. Both
+ interfaces create an input and output JACK port. The application makes
+ these ports the endpoint of the call. The audio coming from the channel
+ goes out the output port and whatever comes back in on the input port is
+ what gets sent to the channel. The JACK_HOOK() function turns on a JACK
+ audiohook on the channel. This lets you run the audio coming from a
+ channel through JACK, and whatever comes back in is what gets forwarded
+ on as the channel's audio. This is very useful for building custom
+ vocoders or doing recording or analysis of the channel's audio in another
+ application.
+ * Added a new module, res_config_curl, which permits using a HTTP POST url
+ to retrieve, create, update, and delete realtime information from a remote
+ web server. Note that this module requires func_curl.so to be loaded for
+ backend functionality.
Miscellaneous
-------------
@@ -393,10 +469,8 @@
* Added maxfiles option to options section of asterisk.conf which allows you to specify
what Asterisk should set as the maximum number of open files when it loads.
* Added the jittertargetextra configuration option.
- * Added a new CDR module, cdr_sqlite3_custom.
* The cdr_manager module has a [mappings] feature, like cdr_custom,
to add fields to the manager event from the CDR variables.
- * Added a new realtime configuration module, res_config_sqlite
* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
configuration files for the IP channel drivers. The new option is "cos".
This information is also documented in doc/qos.tex, or the IP Quality of Service
@@ -408,14 +482,15 @@
* Added support for writing and running your dialplan in lua. See
configs/extensions.lua.sample for examples of how to do this.
* A new extension pattern matching algorithm, based on a trie, is introduced
- here, that could noticeably speed up mid-sized to large dialplans.
- It is NOT used by default, as duplicating the behaviour of the old pattern
- matcher is still under development. A config file option, in extensions.conf,
- in the [general] section, called "extenpatternmatchingnew", is by default
- set to false; setting that to true will force the use of the new algorithm.
- Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
- be used to switch the algorithms at run time.
+ here, that could noticeably speed up mid-sized to large dialplans.
+ It is NOT used by default, as duplicating the behaviour of the old pattern
+ matcher is still under development. A config file option, in extensions.conf,
+ in the [general] section, called "extenpatternmatchingnew", is by default
+ set to false; setting that to true will force the use of the new algorithm.
+ Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
+ be used to switch the algorithms at run time.
* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
- specifying which socket to use to connect to the running Asterisk daemon
- (-s)
-
+ specifying which socket to use to connect to the running Asterisk daemon
+ (-s)
+ * Added logging to 'make update' command. See update.log
+
Modified: team/oej/astum/CREDITS
URL: http://svn.digium.com/view/asterisk/team/oej/astum/CREDITS?view=diff&rev=99593&r1=99592&r2=99593
==============================================================================
--- team/oej/astum/CREDITS (original)
+++ team/oej/astum/CREDITS Tue Jan 22 11:35:07 2008
@@ -16,6 +16,9 @@
nic.at - ENUM support in Asterisk
Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
+
+John Todd, TalkPlus, Inc. and JR Richardson, Ntegrated Solutions. - for funding
+ the development of SIP Session Timers support.
=== WISHLIST CONTRIBUTERS ===
Jeremy McNamara - SpeeX support
@@ -53,7 +56,7 @@
and sip configs.
anthmct(AT)yahoo.com http://www.asterlink.com
-James Golovich - Innumerable contributions
+James Golovich - Innumerable contributions, including SIP TCP and TLS support.
You can find him and asterisk-perl at http://asterisk.gnuinter.net
Andre Bierwirth - Extension hints and status
@@ -106,7 +109,9 @@
simon(AT)slimey.org
Olle E. Johansson - SIP RFC compliance, documentation and testing, testing,
- testing; MiniVM - the small voicemail system, many documentation
+ SIP outbound proxy support, Manager 1.1 update, SIP transfer support,
+ SIP presence support, SIP call state updates (dialog-info),
+ MiniVM - the small voicemail system, many documentation
updates/corrections, and many bug fixes.
oej(AT)edvina.net, http://edvina.net
@@ -172,6 +177,11 @@
Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup,
and a bunch of infrastructure work (loader, new_cli, ...)
+
+Brett Bryant - digit option for musiconhold selection, ENUMQUERY and ENUMRESULT functions,
+ feature group configuration for features.conf, per-file CLI debug and verbose settings,
+ TCP and TLS support for SIP, and various bug fixes.
+ brettbryant(AT)gmail.com
=== OTHER CONTRIBUTIONS ===
John Todd - Monkey sounds and associated teletorture prompt
Modified: team/oej/astum/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/astum/Makefile?view=diff&rev=99593&r1=99592&r2=99593
==============================================================================
--- team/oej/astum/Makefile (original)
+++ team/oej/astum/Makefile Tue Jan 22 11:35:07 2008
@@ -101,7 +101,7 @@
# Some build systems, such as the one in openwrt, like to pass custom target
# CFLAGS and LDFLAGS in the COPTS and LDOPTS variables.
-ASTCFLAGS+=$(COPTS) -D_XPG4_2
+ASTCFLAGS+=$(COPTS)
ASTLDFLAGS+=$(LDOPTS)
#Uncomment this to see all build commands instead of 'quiet' output
@@ -257,7 +257,7 @@
endif
ifeq ($(OSARCH),SunOS)
- ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include
+ ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include -D_XPG4_2
endif
ASTERISKVERSION:=$(shell GREP=$(GREP) AWK=$(AWK) build_tools/make_version .)
@@ -275,11 +275,9 @@
# XXX MALLOC_DEBUG is probably unused, Makefile.moddir_rules adds the
# value directly to ASTCFLAGS
-# XXX BUSYDETECT is probably useless, the only similar reference is to
-# #ifdef BUSYDETECT in main/dsp.c
-ASTCFLAGS+=$(MALLOC_DEBUG)$(BUSYDETECT)$(OPTIONS)
-
-MOD_SUBDIRS:=channels pbx apps codecs formats cdr funcs main res $(LOCAL_MOD_SUBDIRS)
+ASTCFLAGS+=$(MALLOC_DEBUG)$(OPTIONS)
+
+MOD_SUBDIRS:=channels pbx apps codecs formats cdr funcs tests main res $(LOCAL_MOD_SUBDIRS)
OTHER_SUBDIRS:=utils agi
SUBDIRS:=$(OTHER_SUBDIRS) $(MOD_SUBDIRS)
SUBDIRS_INSTALL:=$(SUBDIRS:%=%-install)
@@ -309,10 +307,11 @@
# comment to print directories during submakes
#PRINT_DIR=yes
+SILENTMAKE:=$(MAKE) --quiet --no-print-directory
ifneq ($(PRINT_DIR)$(NOISY_BUILD),)
-SUBMAKE=$(MAKE) --quiet
+SUBMAKE:=$(MAKE) --quiet
else
-SUBMAKE=$(MAKE) --quiet --no-print-directory
+SUBMAKE:=$(MAKE) --quiet --no-print-directory
endif
# This is used when generating the doxygen documentation
@@ -352,13 +351,13 @@
menuselect/menuselect --check-deps $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts
$(MOD_SUBDIRS_EMBED_LDSCRIPT):
- @echo "EMBED_LDSCRIPTS+="`$(SUBMAKE) -C $(@:-embed-ldscript=) SUBDIR=$(@:-embed-ldscript=) __embed_ldscript` >> makeopts.embed_rules
+ @echo "EMBED_LDSCRIPTS+="`$(SILENTMAKE) -C $(@:-embed-ldscript=) SUBDIR=$(@:-embed-ldscript=) __embed_ldscript` >> makeopts.embed_rules
$(MOD_SUBDIRS_EMBED_LDFLAGS):
- @echo "EMBED_LDFLAGS+="`$(SUBMAKE) -C $(@:-embed-ldflags=) SUBDIR=$(@:-embed-ldflags=) __embed_ldflags` >> makeopts.embed_rules
+ @echo "EMBED_LDFLAGS+="`$(SILENTMAKE) -C $(@:-embed-ldflags=) SUBDIR=$(@:-embed-ldflags=) __embed_ldflags` >> makeopts.embed_rules
$(MOD_SUBDIRS_EMBED_LIBS):
- @echo "EMBED_LIBS+="`$(SUBMAKE) -C $(@:-embed-libs=) SUBDIR=$(@:-embed-libs=) __embed_libs` >> makeopts.embed_rules
+ @echo "EMBED_LIBS+="`$(SILENTMAKE) -C $(@:-embed-libs=) SUBDIR=$(@:-embed-libs=) __embed_libs` >> makeopts.embed_rules
$(MOD_SUBDIRS_MENUSELECT_TREE):
@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) moduleinfo
@@ -371,7 +370,7 @@
@$(MAKE) $(PRINT_DIR) $(MOD_SUBDIRS_EMBED_LDFLAGS)
@$(MAKE) $(PRINT_DIR) $(MOD_SUBDIRS_EMBED_LIBS)
-$(SUBDIRS): include/asterisk/version.h include/asterisk/build.h include/asterisk/buildopts.h defaults.h makeopts.embed_rules
+$(SUBDIRS): main/version.c include/asterisk/build.h include/asterisk/buildopts.h defaults.h makeopts.embed_rules
ifeq ($(findstring $(OSARCH), mingw32 cygwin ),)
# Non-windows:
@@ -402,8 +401,8 @@
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
-include/asterisk/version.h:
- @build_tools/make_version_h > $@.tmp
+main/version.c:
+ @build_tools/make_version_c > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
@@ -426,7 +425,7 @@
clean: $(SUBDIRS_CLEAN)
rm -f defaults.h
rm -f include/asterisk/build.h
- rm -f include/asterisk/version.h
+ rm -f main/version.c
@$(MAKE) -C menuselect clean
cp -f .cleancount .lastclean
@@ -449,10 +448,20 @@
# 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
# improved a lot. I'll put it here for now.
+ mkdir -p $(DESTDIR)$(ASTDATADIR)/phoneprov
+ for x in phoneprov/*; do \
+ $(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/phoneprov ; \
+ done
mkdir -p $(DESTDIR)$(ASTDATADIR)/static-http
for x in static-http/*; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/static-http ; \
done
+ if [ -d doc/tex/asterisk ] ; then \
+ mkdir -p $(DESTDIR)$(ASTDATADIR)/static-http/docs ; \
+ for n in doc/tex/asterisk/* ; do \
+ $(INSTALL) -m 644 $$n $(DESTDIR)$(ASTDATADIR)/static-http/docs ; \
+ done \
+ fi
mkdir -p $(DESTDIR)$(ASTDATADIR)/images
for x in images/*.jpg; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/images ; \
@@ -463,7 +472,10 @@
update:
@if [ -d .svn ]; then \
echo "Updating from Subversion..." ; \
+ fromrev="`svn info | $(AWK) '/Revision: / {print $$2}'`"; \
svn update | tee update.out; \
+ torev="`svn info | $(AWK) '/Revision: / {print $$2}'`"; \
+ echo "`date` Updated from revision $${fromrev} to $${torev}." >> update.log; \
rm -f .version; \
if [ `grep -c ^C update.out` -gt 0 ]; then \
echo ; echo "The following files have conflicts:" ; \
@@ -694,7 +706,7 @@
rpm: __rpm
-__rpm: include/asterisk/version.h include/asterisk/buildopts.h spec
+__rpm: main/version.c include/asterisk/buildopts.h spec
rm -rf /tmp/asterisk ; \
mkdir -p /tmp/asterisk/redhat/RPMS/i386 ; \
$(MAKE) DESTDIR=/tmp/asterisk install ; \
@@ -819,4 +831,4 @@
asterisk.pdf:
$(MAKE) -C doc/tex asterisk.pdf
-.PHONY: menuselect main sounds clean dist-clean distclean all prereqs cleantest uninstall _uninstall uninstall-all pdf dont-optimize $(SUBDIRS_INSTALL) $(SUBDIRS_DIST_CLEAN) $(SUBDIRS_CLEAN) $(SUBDIRS_UNINSTALL) $(SUBDIRS) $(MOD_SUBDIRS_EMBED_LDSCRIPT) $(MOD_SUBDIRS_EMBED_LDFLAGS) $(MOD_SUBDIRS_EMBED_LIBS)
+.PHONY: menuselect main sounds clean dist-clean distclean all prereqs cleantest uninstall _uninstall uninstall-all pdf dont-optimize $(SUBDIRS_INSTALL) $(SUBDIRS_DIST_CLEAN) $(SUBDIRS_CLEAN) $(SUBDIRS_UNINSTALL) $(SUBDIRS) $(MOD_SUBDIRS_EMBED_LDSCRIPT) $(MOD_SUBDIRS_EMBED_LDFLAGS) $(MOD_SUBDIRS_EMBED_LIBS) main/version.c
Modified: team/oej/astum/README
URL: http://svn.digium.com/view/asterisk/team/oej/astum/README?view=diff&rev=99593&r1=99592&r2=99593
==============================================================================
--- team/oej/astum/README (original)
+++ team/oej/astum/README Tue Jan 22 11:35:07 2008
@@ -1,65 +1,81 @@
-The Asterisk(R) Open Source PBX
-by Mark Spencer <markster at digium.com>
-and the Asterisk.org developer community
-
-Copyright (C) 2001-2006 Digium, Inc.
-and other copyright holders.
-================================================================
-
-* SECURITY
+===============================================================================
+=== The Asterisk(R) Open Source PBX
+===
+=== by Mark Spencer <markster at digium.com>
+=== and the Asterisk.org developer community
+===
+=== Copyright (C) 2001-2008 Digium, Inc.
+=== and other copyright holders.
+===============================================================================
+
+-------------------------------------------------------------------------------
+--- SECURITY ------------------------------------------------------------------
+
It is imperative that you read and fully understand the contents of
-the security information file (doc/security.txt) before you attempt
-to configure and run an Asterisk server.
-
-* WHAT IS ASTERISK ?
+the security information document before you attempt to configure and run
+an Asterisk server.
+
+ If you downloaded Asterisk as a tarball, see the security section in the PDF
+version of the documentation in doc/tex/asterisk.pdf. Alternatively, pull up
+the HTML version of the documentation in doc/tex/asterisk/index.html. The
+source for the security document is available in doc/tex/security.tex.
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- WHAT IS ASTERISK ? --------------------------------------------------------
+
Asterisk is an Open Source PBX and telephony toolkit. It is, in a
sense, middleware between Internet and telephony channels on the bottom,
-and Internet and telephony applications at the top. For more information
-on the project itself, please visit the Asterisk home page at:
+and Internet and telephony applications at the top. However, Asterisk supports
+more telephony interfaces than just Internet telephony. Asterisk also has a
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