[asterisk-commits] oej: branch oej/kill-the-user r99296 - in /team/oej/kill-the-user: ./ apps/ b...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 22 10:39:14 CST 2008
Author: oej
Date: Mon Jan 21 04:13:37 2008
New Revision: 99296
URL: http://svn.digium.com/view/asterisk?view=rev&rev=99296
Log:
Reset, resolve
Added:
team/oej/kill-the-user/doc/siptls.txt
- copied unchanged from r99280, trunk/doc/siptls.txt
team/oej/kill-the-user/include/asterisk/tcptls.h
- copied unchanged from r99280, trunk/include/asterisk/tcptls.h
team/oej/kill-the-user/main/tcptls.c
- copied unchanged from r99280, trunk/main/tcptls.c
team/oej/kill-the-user/res/res_config_curl.c
- copied unchanged from r99280, trunk/res/res_config_curl.c
Modified:
team/oej/kill-the-user/ (props changed)
team/oej/kill-the-user/CHANGES
team/oej/kill-the-user/CREDITS
team/oej/kill-the-user/Makefile
team/oej/kill-the-user/README
team/oej/kill-the-user/acinclude.m4
team/oej/kill-the-user/apps/app_queue.c
team/oej/kill-the-user/apps/app_voicemail.c
team/oej/kill-the-user/build_tools/cflags.xml
team/oej/kill-the-user/build_tools/menuselect-deps.in
team/oej/kill-the-user/cdr/cdr_adaptive_odbc.c
team/oej/kill-the-user/channels/chan_console.c
team/oej/kill-the-user/channels/chan_iax2.c
team/oej/kill-the-user/channels/chan_local.c
team/oej/kill-the-user/channels/chan_sip.c
team/oej/kill-the-user/channels/chan_zap.c
team/oej/kill-the-user/codecs/codec_speex.c
team/oej/kill-the-user/codecs/codec_zap.c
team/oej/kill-the-user/configs/cdr_adaptive_odbc.conf.sample
team/oej/kill-the-user/configs/console.conf.sample
team/oej/kill-the-user/configs/phoneprov.conf.sample
team/oej/kill-the-user/configs/sip.conf.sample
team/oej/kill-the-user/configs/zapata.conf.sample
team/oej/kill-the-user/configure
team/oej/kill-the-user/configure.ac
team/oej/kill-the-user/doc/tex/phoneprov.tex
team/oej/kill-the-user/funcs/func_cut.c
team/oej/kill-the-user/funcs/func_odbc.c
team/oej/kill-the-user/include/asterisk/autoconfig.h.in
team/oej/kill-the-user/include/asterisk/frame.h
team/oej/kill-the-user/include/asterisk/http.h
team/oej/kill-the-user/include/asterisk/translate.h
team/oej/kill-the-user/main/Makefile
team/oej/kill-the-user/main/abstract_jb.c
team/oej/kill-the-user/main/ast_expr2.c
team/oej/kill-the-user/main/ast_expr2.h
team/oej/kill-the-user/main/ast_expr2.y
team/oej/kill-the-user/main/asterisk.c
team/oej/kill-the-user/main/channel.c
team/oej/kill-the-user/main/dial.c
team/oej/kill-the-user/main/dsp.c
team/oej/kill-the-user/main/frame.c
team/oej/kill-the-user/main/http.c
team/oej/kill-the-user/main/manager.c
team/oej/kill-the-user/main/rtp.c
team/oej/kill-the-user/main/slinfactory.c
team/oej/kill-the-user/main/translate.c
team/oej/kill-the-user/main/utils.c
team/oej/kill-the-user/makeopts.in
team/oej/kill-the-user/res/res_features.c
team/oej/kill-the-user/res/res_odbc.c
team/oej/kill-the-user/res/res_phoneprov.c
Change Statistics:
team/oej/kill-the-user/CHANGES | 15
team/oej/kill-the-user/CREDITS | 14
team/oej/kill-the-user/Makefile | 7
team/oej/kill-the-user/README | 212 -
team/oej/kill-the-user/acinclude.m4 | 29
team/oej/kill-the-user/apps/app_queue.c | 2
team/oej/kill-the-user/apps/app_voicemail.c | 7
team/oej/kill-the-user/build_tools/cflags.xml | 11
team/oej/kill-the-user/build_tools/menuselect-deps.in | 1
team/oej/kill-the-user/cdr/cdr_adaptive_odbc.c | 45
team/oej/kill-the-user/channels/chan_console.c | 666 ++-
team/oej/kill-the-user/channels/chan_iax2.c | 31
team/oej/kill-the-user/channels/chan_local.c | 10
team/oej/kill-the-user/channels/chan_sip.c | 1827 +++++++++-
team/oej/kill-the-user/channels/chan_zap.c | 264 +
team/oej/kill-the-user/codecs/codec_speex.c | 1
team/oej/kill-the-user/codecs/codec_zap.c | 35
team/oej/kill-the-user/configs/cdr_adaptive_odbc.conf.sample | 4
team/oej/kill-the-user/configs/console.conf.sample | 24
team/oej/kill-the-user/configs/phoneprov.conf.sample | 15
team/oej/kill-the-user/configs/sip.conf.sample | 58
team/oej/kill-the-user/configs/zapata.conf.sample | 16
team/oej/kill-the-user/configure.ac | 10
team/oej/kill-the-user/doc/tex/phoneprov.tex | 26
team/oej/kill-the-user/funcs/func_cut.c | 6
team/oej/kill-the-user/funcs/func_odbc.c | 10
team/oej/kill-the-user/include/asterisk/autoconfig.h.in | 6
team/oej/kill-the-user/include/asterisk/frame.h | 17
team/oej/kill-the-user/include/asterisk/http.h | 86
team/oej/kill-the-user/include/asterisk/translate.h | 18
team/oej/kill-the-user/main/Makefile | 2
team/oej/kill-the-user/main/abstract_jb.c | 4
team/oej/kill-the-user/main/ast_expr2.c | 179
team/oej/kill-the-user/main/ast_expr2.h | 32
team/oej/kill-the-user/main/ast_expr2.y | 12
team/oej/kill-the-user/main/asterisk.c | 38
team/oej/kill-the-user/main/channel.c | 1
team/oej/kill-the-user/main/dial.c | 15
team/oej/kill-the-user/main/dsp.c | 73
team/oej/kill-the-user/main/frame.c | 52
team/oej/kill-the-user/main/http.c | 200 -
team/oej/kill-the-user/main/manager.c | 3
team/oej/kill-the-user/main/rtp.c | 6
team/oej/kill-the-user/main/slinfactory.c | 11
team/oej/kill-the-user/main/translate.c | 33
team/oej/kill-the-user/main/utils.c | 2
team/oej/kill-the-user/makeopts.in | 3
team/oej/kill-the-user/res/res_features.c | 8
team/oej/kill-the-user/res/res_odbc.c | 36
team/oej/kill-the-user/res/res_phoneprov.c | 73
50 files changed, 3221 insertions(+), 1035 deletions(-)
Propchange: team/oej/kill-the-user/
------------------------------------------------------------------------------
automerge = http://www.codename-pineapple.org/
Propchange: team/oej/kill-the-user/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/oej/kill-the-user/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/oej/kill-the-user/
------------------------------------------------------------------------------
--- svn:ignore (original)
+++ svn:ignore Mon Jan 21 04:13:37 2008
@@ -23,3 +23,4 @@
autom4te.cache
makeopts.embed_rules
aclocal.m4
+update.log
Propchange: team/oej/kill-the-user/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Jan 21 04:13:37 2008
@@ -1,1 +1,1 @@
-/trunk:1-98916
+/trunk:1-99295
Modified: team/oej/kill-the-user/CHANGES
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/CHANGES?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/CHANGES (original)
+++ team/oej/kill-the-user/CHANGES Mon Jan 21 04:13:37 2008
@@ -79,6 +79,9 @@
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
-----------
@@ -125,6 +128,11 @@
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
------------
@@ -448,6 +456,10 @@
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
-------------
@@ -480,4 +492,5 @@
* 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)
-
+ * Added logging to 'make update' command. See update.log
+
Modified: team/oej/kill-the-user/CREDITS
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/CREDITS?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/CREDITS (original)
+++ team/oej/kill-the-user/CREDITS Mon Jan 21 04:13:37 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/kill-the-user/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/Makefile?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/Makefile (original)
+++ team/oej/kill-the-user/Makefile Mon Jan 21 04:13:37 2008
@@ -275,9 +275,7 @@
# 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)$(OPTIONS)
+ASTCFLAGS+=$(MALLOC_DEBUG)$(OPTIONS)
MOD_SUBDIRS:=channels pbx apps codecs formats cdr funcs tests main res $(LOCAL_MOD_SUBDIRS)
OTHER_SUBDIRS:=utils agi
@@ -474,7 +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:" ; \
Modified: team/oej/kill-the-user/README
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/README?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/README (original)
+++ team/oej/kill-the-user/README Mon Jan 21 04:13:37 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
+vast amount of support for traditional PSTN telephony, as well. For more
+information on the project itself, please visit the Asterisk home page at:
http://www.asterisk.org
-In addition you'll find lots of information compiled by the Asterisk
+ In addition you'll find lots of information compiled by the Asterisk
community on this Wiki:
http://www.voip-info.org/wiki-Asterisk
-There is a book on Asterisk published by O'Reilly under the
-Creative Commons License. It is available in book stores as well
-as in a downloadable version on the http://www.asteriskdocs.org
-web site.
-
-* SUPPORTED OPERATING SYSTEMS
-
-== Linux ==
+ There is a book on Asterisk published by O'Reilly under the Creative Commons
+License. It is available in book stores as well as in a downloadable version on
+the http://www.asteriskdocs.org web site.
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- SUPPORTED OPERATING SYSTEMS -----------------------------------------------
+
+--- Linux
The Asterisk Open Source PBX is developed and tested primarily on the
GNU/Linux operating system, and is supported on every major GNU/Linux
distribution.
-== Others ==
+--- Others
Asterisk has also been 'ported' and reportedly runs properly on other
-operating systems as well, including Sun Solaris, Apple's Mac OS X, and
-the BSD variants.
-
-* GETTING STARTED
+operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin,
+and the BSD variants.
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- GETTING STARTED -----------------------------------------------------------
First, be sure you've got supported hardware (but note that you don't need
-ANY special hardware, not even a soundcard) to install and run Asterisk.
+ANY special hardware, not even a sound card) to install and run Asterisk.
Supported telephony hardware includes:
- * All Wildcard (tm) products from Digium (www.digium.com)
+ * All Analog and Digital Interface cards from Digium (www.digium.com)
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
- * any full duplex sound card supported by ALSA or OSS
+ * any full duplex sound card supported by ALSA, OSS, or PortAudio
* any ISDN card supported by mISDN on Linux (BRI)
* The Xorcom AstriBank channel bank
- * VoiceTronix OpenLine products
-
-The are several drivers for ISDN BRI cards available from third party sources.
-Check the voip-info.org wiki for more information on chan_capi and
-zaphfc.
-
-* UPGRADING FROM AN EARLIER VERSION
+ * VoiceTronix OpenLine products
+
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- UPGRADING FROM AN EARLIER VERSION -----------------------------------------
If you are updating from a previous version of Asterisk, make sure you
read the UPGRADE.txt file in the source directory. There are some files
@@ -67,29 +83,34 @@
made every effort possible to maintain backwards compatibility.
In order to discover new features to use, please check the configuration
-examples in the /configs directory of the source code distribution.
-To discover the major new features of Asterisk 1.2, please visit
-http://edvina.net/asterisk1-2/
-
-* NEW INSTALLATIONS
+examples in the /configs directory of the source code distribution. For a
+list of new features in this version of Asterisk, see the CHANGES file.
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- NEW INSTALLATIONS ---------------------------------------------------------
Ensure that your system contains a compatible compiler and development
libraries. Asterisk requires either the GNU Compiler Collection (GCC) version
3.0 or higher, or a compiler that supports the C99 specification and some of
the gcc language extensions. In addition, your system needs to have the C
-library headers available, and the headers and libraries for OpenSSL,
-ncurses and zlib.
-On many distributions, these files are installed by packages with names like
-'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
-
- So let's proceed:
+library headers available, and the headers and libraries for ncurses.
+
+ There are many modules that have additional dependencies. To see what
+libraries are being looked for, see ./configure --help, or run
+"make menuselect" to view the dependencies for specific modules.
+
+ On many distributions, these dependencies are installed by packages with names
+like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel'
+or similar.
+
+ So, let's proceed:
1) Read this README file.
- There are more documents than this one in the doc/ directory.
-You may also want to check the configuration files that contain
-examples and reference guides. They are all in the configs/
-directory.
+ There are more documents than this one in the doc/ directory. You may also
+want to check the configuration files that contain examples and reference
+guides. They are all in the configs/ directory.
2) Run "./configure"
@@ -98,30 +119,23 @@
3) Run "make menuselect" [optional]
- This is needed if you want to select the modules that will be
-compiled and to check modules dependencies.
+ This is needed if you want to select the modules that will be compiled and to
+check dependencies for various optional modules.
4) Run "make"
Assuming the build completes successfully:
5) Run "make install"
-
- Each time you update or checkout from the repository, you are strongly
-encouraged to ensure all previous object files are removed to avoid internal
-inconsistency in Asterisk. Normally, this is automatically done with
-the presence of the file .cleancount, which increments each time a 'make clean'
-is required, and the file .lastclean, which contains the last .cleancount used.
If this is your first time working with Asterisk, you may wish to install
the sample PBX, with demonstration extensions, etc. If so, run:
6) "make samples"
- Doing so will overwrite any existing config files you have.
-
- Finally, you can launch Asterisk in the foreground mode (not a daemon)
-with:
+ Doing so will overwrite any existing configuration files you have installed.
+
+ Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
# asterisk -vvvc
@@ -134,20 +148,22 @@
You can type "help" at any time to get help with the system. For help
with a specific command, type "help <command>". To start the PBX using
-your sound card, you can type "dial" to dial the PBX. Then you can use
-"answer", "hangup", and "dial" to simulate the actions of a telephone.
-Remember that if you don't have a full duplex sound card (and Asterisk
-will tell you somewhere in its verbose messages if you do/don't) then it
-won't work right (not yet).
+your sound card, you can type "console dial" to dial the PBX. Then you can use
+"console answer", "console hangup", and "console dial" to simulate the actions
+of a telephone. Remember that if you don't have a full duplex sound card
+(and Asterisk will tell you somewhere in its verbose messages if you do/don't)
+then it won't work right (not yet).
"man asterisk" at the Unix/Linux command prompt will give you detailed
information on how to start and stop Asterisk, as well as all the command
line options for starting Asterisk.
- Feel free to look over the configuration files in /etc/asterisk, where
-you'll find a lot of information about what you can do with Asterisk.
-
-* ABOUT CONFIGURATION FILES
+ Feel free to look over the configuration files in /etc/asterisk, where you
+will find a lot of information about what you can do with Asterisk.
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- ABOUT CONFIGURATION FILES -------------------------------------------------
All Asterisk configuration files share a common format. Comments are
delimited by ';' (since '#' of course, being a DTMF digit, may occur in
@@ -163,7 +179,7 @@
switchtype=national
-in order to indicate to Asterisk that the switch they are connecting to is
+ In order to indicate to Asterisk that the switch they are connecting to is
of the type "national". In general, the parameter will apply to
instantiations which occur below its specification. For example, if the
configuration file read:
@@ -174,7 +190,7 @@
switchtype = dms100
channel => 25-47
-the "national" switchtype would be applied to channels one through
+ The "national" switchtype would be applied to channels one through
four and channels 10 through 12, whereas the "dms100" switchtype would
apply to channels 25 through 47.
@@ -182,8 +198,10 @@
parameters. For example, the line "channel => 25-47" creates objects for
the channels 25 through 47 of the card, obtaining the settings
from the variables specified above.
-
-* SPECIAL NOTE ON TIME
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- SPECIAL NOTE ON TIME ------------------------------------------------------
Those using SIP phones should be aware that Asterisk is sensitive to
large jumps in time. Manually changing the system time using date(1)
@@ -206,8 +224,10 @@
Also note that this issue is separate from the clocking of TDM
channels, and is known to at least affect SIP registrations.
-
-* FILE DESCRIPTORS
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- FILE DESCRIPTORS ----------------------------------------------------------
Depending on the size of your system and your configuration,
Asterisk can consume a large number of file descriptors. In UNIX,
@@ -220,11 +240,13 @@
Most systems limit the number of file descriptors that Asterisk can
have open at one time. This can limit the number of simultaneous
calls that your system can handle. For example, if the limit is set
-at 1024 (a common default value) Asterisk can handle approxiately 150
+at 1024 (a common default value) Asterisk can handle approximately 150
SIP calls simultaneously. To change the number of file descriptors
follow the instructions for your system below:
-
-== PAM-based Linux System ==
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- PAM-based Linux System ----------------------------------------------------
If your system uses PAM (Pluggable Authentication Modules) edit
/etc/security/limits.conf. Add these lines to the bottom of the file:
@@ -242,21 +264,29 @@
If there are no instructions specifically adapted to your system
above you can try adding the command "ulimit -n 8192" to the script
that starts Asterisk.
-
-* MORE INFORMATION
+-------------------------------------------------------------------------------
+
+-------------------------------------------------------------------------------
+--- MORE INFORMATION ----------------------------------------------------------
See the doc directory for more documentation on various features. Again,
please read all the configuration samples that include documentation on
the configuration options.
+ If this release of Asterisk was downloaded from a tarball, then some
+additional documentation should have been included.
+ * doc/tex/asterisk.pdf --- PDF version of the documentation
+ * doc/tex/asterisk/index.html --- HTML version of the documentation
+
Finally, you may wish to visit the web site and join the mailing list if
you're interested in getting more information.
http://www.asterisk.org/support
Welcome to the growing worldwide community of Asterisk users!
-
-Mark Spencer
-
-----
-Asterisk is a trademark belonging to Digium, inc
+-------------------------------------------------------------------------------
+
+--- Mark Spencer, and the Asterisk.org development community
+
+-------------------------------------------------------------------------------
+Asterisk is a trademark of Digium, Inc.
Modified: team/oej/kill-the-user/acinclude.m4
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/acinclude.m4?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/acinclude.m4 (original)
+++ team/oej/kill-the-user/acinclude.m4 Mon Jan 21 04:13:37 2008
@@ -210,9 +210,9 @@
# Check for a package using $2-config. Similar to AST_EXT_LIB_CHECK,
# but use $2-config to determine cflags and libraries to use.
-# $3 and $4 can be used to replace --cflags and --libs in the request
-
-# AST_EXT_TOOL_CHECK([package], [tool name], [--cflags], [--libs])
+# $3 and $4 can be used to replace --cflags and --libs in the request
+
+# AST_EXT_TOOL_CHECK([package], [tool name], [--cflags], [--libs], [includes], [expression])
AC_DEFUN([AST_EXT_TOOL_CHECK],
[
if test "x${PBX_$1}" != "x1" -a "${USE_$1}" != "no"; then
@@ -223,8 +223,27 @@
$1_INCLUDE=$(${CONFIG_$1} $A)
if test x"$4" = x ; then A=--libs ; else A="$4" ; fi
$1_LIB=$(${CONFIG_$1} $A)
- PBX_$1=1
- AC_DEFINE([HAVE_$1], 1, [Define if your system has the $1 libraries.])
+ if test x"$5" != x ; then
+ saved_cppflags="${CPPFLAGS}"
+ if test "x${$1_DIR}" != "x"; then
+ $1_INCLUDE="-I${$1_DIR}/include"
+ fi
+ CPPFLAGS="${CPPFLAGS} ${$1_INCLUDE}"
+
+ AC_COMPILE_IFELSE(
+ [ AC_LANG_PROGRAM( [ $5 ],
+ [ $6; ]
+ )],
+ [ PBX_$1=1
+ AC_DEFINE([HAVE_$1], 1, [Define if your system has the $1 headers.])
+ ],
+ []
+ )
+ CPPFLAGS="${saved_cppflags}"
+ else
+ PBX_$1=1
+ AC_DEFINE([HAVE_$1], 1, [Define if your system has the $1 libraries.])
+ fi
fi
fi
])
Modified: team/oej/kill-the-user/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/apps/app_queue.c?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/apps/app_queue.c (original)
+++ team/oej/kill-the-user/apps/app_queue.c Mon Jan 21 04:13:37 2008
@@ -842,7 +842,7 @@
ast_copy_string(cur->membername, interface, sizeof(cur->membername));
if (!strchr(cur->interface, '/'))
ast_log(LOG_WARNING, "No location at interface '%s'\n", interface);
- cur->status = ast_device_state(interface);
+ cur->status = ast_device_state(cur->state_interface);
}
return cur;
Modified: team/oej/kill-the-user/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/apps/app_voicemail.c?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/apps/app_voicemail.c (original)
+++ team/oej/kill-the-user/apps/app_voicemail.c Mon Jan 21 04:13:37 2008
@@ -3466,7 +3466,7 @@
if (box == 1) return 10;
/* get the real IMAP message number for this message */
snprintf(sequence, sizeof(sequence), "%ld", vms->msgArray[msg]);
- ast_debug(3, "Copying sequence %s to mailbox %s\n",sequence,dbox);
+ ast_debug(3, "Copying sequence %s to mailbox %s\n",sequence,mbox(box));
res = mail_copy(vms->mailstream,sequence,(char *) mbox(box));
if (res == 1) return 0;
return 1;
@@ -5030,11 +5030,6 @@
return -1;
}
- /* Check Quota (here for now to test) */
- mail_parameters(NULL, SET_QUOTA, (void *) mm_parsequota);
- imap_mailbox_name(dbox, sizeof(dbox), vms, box, 1);
- imap_getquotaroot(vms->mailstream, dbox);
-
/* Check Quota */
if (box == 0) {
ast_debug(3, "Mailbox name set to: %s, about to check quotas\n", mbox(box));
Modified: team/oej/kill-the-user/build_tools/cflags.xml
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/build_tools/cflags.xml?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/build_tools/cflags.xml (original)
+++ team/oej/kill-the-user/build_tools/cflags.xml Mon Jan 21 04:13:37 2008
@@ -48,4 +48,15 @@
</member>
<member name="THREAD_CRASH" displayname="Crash on mutex errors">
</member>
+ <member name="BUSYDETECT_TONEONLY" displayname="Enable additional comparision of only the tone duration not the silence part">
+ <conflict>BUSYDETECT_COMPARE_TONE_AND_SILENCE</conflict>
+ <defaultenabled>no</defaultenabled>
+ </member>
+ <member name="BUSYDETECT_COMPARE_TONE_AND_SILENCE" displayname="Assume that tone and silence have the same duration">
+ <conflict>BUSYDETECT_TONEONLY</conflict>
+ <defaultenabled>no</defaultenabled>
+ </member>
+ <member name="BUSYDETECT_DEBUG" displayname="Enable additional busy detection debugging">
+ <defaultenabled>no</defaultenabled>
+ </member>
</category>
Modified: team/oej/kill-the-user/build_tools/menuselect-deps.in
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/build_tools/menuselect-deps.in?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/build_tools/menuselect-deps.in (original)
+++ team/oej/kill-the-user/build_tools/menuselect-deps.in Mon Jan 21 04:13:37 2008
@@ -29,6 +29,7 @@
PRI=@PBX_PRI@
RADIUS=@PBX_RADIUS@
SPEEX=@PBX_SPEEX@
+SPEEXDSP=@PBX_SPEEXDSP@
SQLITE3=@PBX_SQLITE3@
SQLITE=@PBX_SQLITE@
SS7=@PBX_SS7@
Modified: team/oej/kill-the-user/cdr/cdr_adaptive_odbc.c
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/cdr/cdr_adaptive_odbc.c?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/cdr/cdr_adaptive_odbc.c (original)
+++ team/oej/kill-the-user/cdr/cdr_adaptive_odbc.c Mon Jan 21 04:13:37 2008
@@ -56,6 +56,7 @@
struct columns {
char *name;
char *cdrname;
+ char *filtervalue;
SQLSMALLINT type;
SQLINTEGER size;
SQLSMALLINT decimals;
@@ -152,6 +153,31 @@
ast_verb(3, "Found adaptive CDR table %s@%s.\n", tableptr->table, tableptr->connection);
+ /* Check for filters first */
+ for (var = ast_variable_browse(cfg, catg); var; var = var->next) {
+ if (strncmp(var->name, "filter", 6) == 0) {
+ char *cdrvar = ast_strdupa(var->name + 6);
+ cdrvar = ast_strip(cdrvar);
+ ast_verb(3, "Found filter %s for cdr variable %s in %s@%s\n", var->value, cdrvar, tableptr->table, tableptr->connection);
+
+ entry = ast_calloc(sizeof(char), sizeof(*entry) + strlen(cdrvar) + 1 + strlen(var->value) + 1);
+ if (!entry) {
+ ast_log(LOG_ERROR, "Out of memory creating filter entry for CDR variable '%s' in table '%s' on connection '%s'\n", cdrvar, table, connection);
+ res = -1;
+ break;
+ }
+
+ /* NULL column entry means this isn't a column in the database */
+ entry->name = NULL;
+ entry->cdrname = (char *)entry + sizeof(*entry);
+ entry->filtervalue = (char *)entry + sizeof(*entry) + strlen(cdrvar) + 1;
+ strcpy(entry->cdrname, cdrvar);
+ strcpy(entry->filtervalue, var->value);
+
+ AST_LIST_INSERT_TAIL(&(tableptr->columns), entry, list);
+ }
+ }
+
while ((res = SQLFetch(stmt)) != SQL_NO_DATA && res != SQL_ERROR) {
char *cdrvar = "";
@@ -164,13 +190,14 @@
* really don't parse this file all that often, anyway.
*/
for (var = ast_variable_browse(cfg, catg); var; var = var->next) {
- if (strcasecmp(var->value, columnname) == 0) {
+ if (strncmp(var->name, "alias", 5) == 0 && strcasecmp(var->value, columnname) == 0) {
char *tmp = ast_strdupa(var->name + 5);
cdrvar = ast_strip(tmp);
ast_verb(3, "Found alias %s for column %s in %s@%s\n", cdrvar, columnname, tableptr->table, tableptr->connection);
break;
}
}
+
entry = ast_calloc(sizeof(char), sizeof(*entry) + strlen(columnname) + 1 + strlen(cdrvar) + 1);
if (!entry) {
ast_log(LOG_ERROR, "Out of memory creating entry for column '%s' in table '%s' on connection '%s'\n", columnname, table, connection);
@@ -341,6 +368,21 @@
strcasecmp(entry->cdrname, "end") == 0) ? 0 : 1);
if (colptr) {
+ /* Check first if the column filters this entry. Note that this
+ * is very specifically NOT ast_strlen_zero(), because the filter
+ * could legitimately specify that the field is blank, which is
+ * different from the field being unspecified (NULL). */
+ if (entry->filtervalue && strcasecmp(colptr, entry->filtervalue) != 0) {
+ ast_verb(4, "CDR column '%s' with value '%s' does not match filter of"
+ " '%s'. Cancelling this CDR.\n",
+ entry->cdrname, colptr, entry->filtervalue);
+ goto early_release;
+ }
+
+ /* Only a filter? */
+ if (ast_strlen_zero(entry->name))
+ continue;
+
LENGTHEN_BUF1(strlen(entry->name));
switch (entry->type) {
@@ -567,6 +609,7 @@
if (rows == 0) {
ast_log(LOG_WARNING, "cdr_adaptive_odbc: Insert failed on '%s:%s'. CDR failed: %s\n", tableptr->connection, tableptr->table, sql);
}
+early_release:
ast_odbc_release_obj(obj);
}
AST_RWLIST_UNLOCK(&odbc_tables);
Modified: team/oej/kill-the-user/channels/chan_console.c
URL: http://svn.digium.com/view/asterisk/team/oej/kill-the-user/channels/chan_console.c?view=diff&rev=99296&r1=99295&r2=99296
==============================================================================
--- team/oej/kill-the-user/channels/chan_console.c (original)
+++ team/oej/kill-the-user/channels/chan_console.c Mon Jan 21 04:13:37 2008
@@ -1,7 +1,7 @@
/*
* Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 2006 - 2007, Digium, Inc.
+ * Copyright (C) 2006 - 2008, Digium, Inc.
*
* Russell Bryant <russell at digium.com>
*
@@ -41,8 +41,6 @@
* in at least one of the other console channel drivers that are not yet
* implemented here are:
*
- * - Multiple device support
- * - with "active" CLI command
* - Set Auto-answer from the dialplan
* - transfer CLI command
* - boost CLI command and .conf option
@@ -68,6 +66,7 @@
#include "asterisk/cli.h"
#include "asterisk/musiconhold.h"
#include "asterisk/callerid.h"
+#include "asterisk/astobj2.h"
/*!
* \brief The sample rate to request from PortAudio
@@ -126,6 +125,8 @@
AST_DECLARE_STRING_FIELDS(
/*! Name of the device */
AST_STRING_FIELD(name);
+ AST_STRING_FIELD(input_device);
+ AST_STRING_FIELD(output_device);
/*! Default context for outgoing calls */
AST_STRING_FIELD(context);
/*! Default extension for outgoing calls */
@@ -157,14 +158,20 @@
unsigned int autoanswer:1;
/*! Ignore context in the console dial CLI command */
unsigned int overridecontext:1;
- /*! Lock to protect data in this struct */
- ast_mutex_t __lock;
+ /*! Set during a reload so that we know to destroy this if it is no longer
+ * in the configuration file. */
+ unsigned int destroy:1;
/*! ID for the stream monitor thread */
pthread_t thread;
-} console_pvt = {
- .__lock = AST_MUTEX_INIT_VALUE,
- .thread = AST_PTHREADT_NULL,
-};
+} globals;
+
+AST_MUTEX_DEFINE_STATIC(globals_lock);
+
+static struct ao2_container *pvts;
+#define NUM_PVT_BUCKETS 7
+
+static struct console_pvt *active_pvt;
+AST_RWLOCK_DEFINE_STATIC(active_lock);
/*!
* \brief Global jitterbuffer configuration
@@ -218,10 +225,32 @@
};
/*! \brief lock a console_pvt struct */
-#define console_pvt_lock(pvt) ast_mutex_lock(&(pvt)->__lock)
+#define console_pvt_lock(pvt) ao2_lock(pvt)
/*! \brief unlock a console_pvt struct */
-#define console_pvt_unlock(pvt) ast_mutex_unlock(&(pvt)->__lock)
+#define console_pvt_unlock(pvt) ao2_unlock(pvt)
+
+static inline struct console_pvt *ref_pvt(struct console_pvt *pvt)
+{
+ if (pvt)
+ ao2_ref(pvt, +1);
+ return pvt;
+}
+
+static inline struct console_pvt *unref_pvt(struct console_pvt *pvt)
+{
+ ao2_ref(pvt, -1);
+ return NULL;
+}
+
+static struct console_pvt *find_pvt(const char *name)
+{
+ struct console_pvt tmp_pvt = {
+ .name = name,
+ };
+
[... 6954 lines stripped ...]
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