[asterisk-commits] oej: branch oej/tdd-sip r101215 - in /team/oej/tdd-sip: ./ apps/ build_tools/...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 30 09:20:59 CST 2008
Author: oej
Date: Wed Jan 30 09:20:58 2008
New Revision: 101215
URL: http://svn.digium.com/view/asterisk?view=rev&rev=101215
Log:
Update branch
Added:
team/oej/tdd-sip/apps/app_jack.c
- copied unchanged from r99542, trunk/apps/app_jack.c
team/oej/tdd-sip/doc/siptls.txt
- copied unchanged from r99542, trunk/doc/siptls.txt
team/oej/tdd-sip/include/asterisk/tcptls.h
- copied unchanged from r99542, trunk/include/asterisk/tcptls.h
team/oej/tdd-sip/main/tcptls.c
- copied unchanged from r99542, trunk/main/tcptls.c
team/oej/tdd-sip/res/res_config_curl.c
- copied unchanged from r99542, trunk/res/res_config_curl.c
Modified:
team/oej/tdd-sip/ (props changed)
team/oej/tdd-sip/CHANGES
team/oej/tdd-sip/CREDITS
team/oej/tdd-sip/Makefile
team/oej/tdd-sip/README
team/oej/tdd-sip/acinclude.m4
team/oej/tdd-sip/apps/app_directory.c
team/oej/tdd-sip/apps/app_followme.c
team/oej/tdd-sip/apps/app_meetme.c
team/oej/tdd-sip/apps/app_queue.c
team/oej/tdd-sip/apps/app_voicemail.c
team/oej/tdd-sip/build_tools/cflags.xml
team/oej/tdd-sip/build_tools/menuselect-deps.in
team/oej/tdd-sip/cdr/cdr_adaptive_odbc.c
team/oej/tdd-sip/cdr/cdr_odbc.c
team/oej/tdd-sip/channels/chan_console.c
team/oej/tdd-sip/channels/chan_iax2.c
team/oej/tdd-sip/channels/chan_local.c
team/oej/tdd-sip/channels/chan_sip.c
team/oej/tdd-sip/channels/chan_skinny.c
team/oej/tdd-sip/channels/chan_zap.c
team/oej/tdd-sip/codecs/codec_g722.c
team/oej/tdd-sip/codecs/codec_resample.c
team/oej/tdd-sip/codecs/codec_speex.c
team/oej/tdd-sip/codecs/codec_zap.c
team/oej/tdd-sip/configs/cdr_adaptive_odbc.conf.sample
team/oej/tdd-sip/configs/console.conf.sample
team/oej/tdd-sip/configs/phoneprov.conf.sample
team/oej/tdd-sip/configs/queues.conf.sample
team/oej/tdd-sip/configs/res_odbc.conf.sample
team/oej/tdd-sip/configs/sip.conf.sample
team/oej/tdd-sip/configs/voicemail.conf.sample
team/oej/tdd-sip/configs/zapata.conf.sample
team/oej/tdd-sip/configure
team/oej/tdd-sip/configure.ac
team/oej/tdd-sip/doc/tex/Makefile
team/oej/tdd-sip/doc/tex/phoneprov.tex
team/oej/tdd-sip/doc/tex/qos.tex
team/oej/tdd-sip/doc/tex/realtime.tex
team/oej/tdd-sip/funcs/func_cut.c
team/oej/tdd-sip/funcs/func_odbc.c
team/oej/tdd-sip/include/asterisk/autoconfig.h.in
team/oej/tdd-sip/include/asterisk/frame.h
team/oej/tdd-sip/include/asterisk/http.h
team/oej/tdd-sip/include/asterisk/res_odbc.h
team/oej/tdd-sip/include/asterisk/translate.h
team/oej/tdd-sip/main/Makefile
team/oej/tdd-sip/main/abstract_jb.c
team/oej/tdd-sip/main/ast_expr2.c
team/oej/tdd-sip/main/ast_expr2.h
team/oej/tdd-sip/main/ast_expr2.y
team/oej/tdd-sip/main/asterisk.c
team/oej/tdd-sip/main/audiohook.c
team/oej/tdd-sip/main/channel.c
team/oej/tdd-sip/main/dial.c
team/oej/tdd-sip/main/dsp.c
team/oej/tdd-sip/main/frame.c
team/oej/tdd-sip/main/http.c
team/oej/tdd-sip/main/manager.c
team/oej/tdd-sip/main/pbx.c
team/oej/tdd-sip/main/rtp.c
team/oej/tdd-sip/main/slinfactory.c
team/oej/tdd-sip/main/translate.c
team/oej/tdd-sip/main/utils.c
team/oej/tdd-sip/makeopts.in
team/oej/tdd-sip/pbx/pbx_dundi.c
team/oej/tdd-sip/res/res_agi.c
team/oej/tdd-sip/res/res_features.c
team/oej/tdd-sip/res/res_odbc.c
team/oej/tdd-sip/res/res_phoneprov.c
Propchange: team/oej/tdd-sip/
------------------------------------------------------------------------------
automerge = http://www.codename-pineapple.org/
Propchange: team/oej/tdd-sip/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/oej/tdd-sip/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/oej/tdd-sip/
------------------------------------------------------------------------------
--- svn:ignore (original)
+++ svn:ignore Wed Jan 30 09:20:58 2008
@@ -23,3 +23,4 @@
autom4te.cache
makeopts.embed_rules
aclocal.m4
+update.log
Propchange: team/oej/tdd-sip/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Jan 30 09:20:58 2008
@@ -1,1 +1,1 @@
-/trunk:1-98175
+/trunk:1-99542
Modified: team/oej/tdd-sip/CHANGES
URL: http://svn.digium.com/view/asterisk/team/oej/tdd-sip/CHANGES?view=diff&rev=101215&r1=101214&r2=101215
==============================================================================
--- team/oej/tdd-sip/CHANGES (original)
+++ team/oej/tdd-sip/CHANGES Wed Jan 30 09:20:58 2008
@@ -43,27 +43,29 @@
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
-----------
@@ -76,6 +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
-----------
@@ -113,15 +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
------------
@@ -149,11 +160,6 @@
* 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.
- * 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.
Phone channel changes (chan_phone)
----------------------------------
@@ -180,8 +186,8 @@
----------------------------------------
* 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.
@@ -190,12 +196,27 @@
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
------------------------------
+ * 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
-------------
@@ -243,6 +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
-------------
@@ -273,7 +296,7 @@
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.
+ device state reported.
MeetMe Changes
--------------
@@ -339,8 +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
-----------
@@ -362,11 +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
------------------------------------
@@ -406,6 +429,38 @@
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
-------------
* Ability to use libcap to set high ToS bits when non-root
@@ -414,10 +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
@@ -429,24 +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)
- * 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.
+ 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/tdd-sip/CREDITS
URL: http://svn.digium.com/view/asterisk/team/oej/tdd-sip/CREDITS?view=diff&rev=101215&r1=101214&r2=101215
==============================================================================
--- team/oej/tdd-sip/CREDITS (original)
+++ team/oej/tdd-sip/CREDITS Wed Jan 30 09:20:58 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/tdd-sip/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/tdd-sip/Makefile?view=diff&rev=101215&r1=101214&r2=101215
==============================================================================
--- team/oej/tdd-sip/Makefile (original)
+++ team/oej/tdd-sip/Makefile Wed Jan 30 09:20:58 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/tdd-sip/README
URL: http://svn.digium.com/view/asterisk/team/oej/tdd-sip/README?view=diff&rev=101215&r1=101214&r2=101215
==============================================================================
--- team/oej/tdd-sip/README (original)
+++ team/oej/tdd-sip/README Wed Jan 30 09:20:58 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/tdd-sip/acinclude.m4
URL: http://svn.digium.com/view/asterisk/team/oej/tdd-sip/acinclude.m4?view=diff&rev=101215&r1=101214&r2=101215
==============================================================================
--- team/oej/tdd-sip/acinclude.m4 (original)
+++ team/oej/tdd-sip/acinclude.m4 Wed Jan 30 09:20:58 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/tdd-sip/apps/app_directory.c
[... 11089 lines stripped ...]
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