| Age | Commit message (Collapse) | Author |
|---|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines
Minor changes inspired by testing with latest GCC.
The latest GCC (what will become 4.5.x) has a few new warnings, that in these
cases found some either downright buggy code, or at least seriously poorly
designed code that could be improved.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Instead of specifying every object file, use make's magic to generate
it.
This will generate less conflicts in team branches when a new file is
added in trunk.
(closes issue #15226)
Reported by: eliel
Patches:
makefile uploaded by eliel (license 64)
Review: http://reviewboard.asterisk.org/r/269/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines
Add debugging mode for diagnosing file descriptor leaks.
(Related to issue #14625)
........
r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines
Oops, missed this file in the last commit.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar 2009) | 5 lines
Improve the build system to *properly* remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix.
With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This commit brings in the bridging core, bridging technologies,
and the ConfBridge application.
For usage information on the ConfBridge application please see
the output of "core show application ConfBridge" from the CLI.
For API documentation please see the doxygen page describing the
architecture and the documentation for each API call.
Review: http://reviewboard.digium.com/r/93/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
Trunk was already pretty 8-bit clean; but I'm still
removing the --full from the flex command so everything
is uniform.
........
r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
The real culprit was the --full argument to flex
in the Makefile! This causes a 7-bit scanner to be
generated.
I reviewed the rules and found one rule where I needed
to specifically include 8-bit chars for a token.
I tested against the text supplied by ibercom, and
all looks very well.
This has been there a surprisingly long time!
(closes issue #14498)
Reported by: ibercom
Patches:
14498.patch uploaded by murf (license 17)
Tested by: murf
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
A heap is a convenient data structure for implementing a priority queue.
Code from svn/asterisk/team/russell/heap/.
Review: http://reviewboard.digium.com/r/160/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Export the XML documentation API:
ast_xmldoc_build_synopsis()
ast_xmldoc_build_syntax()
ast_xmldoc_build_description()
ast_xmldoc_build_seealso()
ast_xmldoc_build_arguments()
ast_xmldoc_printable()
ast_xmldoc_load_documentation()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines
improve header inclusion process in a few small ways:
- it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose
- astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled
- simplify the usage of some of these headers in the AEL-related stuff in the utils directory
........
r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines
fix some minor issues with rev 144924
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
memory allocator. See doc/hoard.txt for more details.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
implementations. Asterisk has, for a long time,
had its own implementation of poll(2) which
just used the input arguments to call select(2).
In 1.4, this internal implementation was used
for Darwin systems. This was removed in Asterisk
trunk at some point, but it seems as though this
was not the right move to make.
On Mac OS X, it appears as though the poll used
to gather CLI input does not respond properly
when connecting via a remote Asterisk console.
Reverting to the use of Asterisk's poll fixed
the issue.
Also, there is now an option for the configure
script, --enable-internal-poll, which will allow
for anyone to use Asterisk's internal poll
implementation in case they suspect that their
system's poll implementation is buggy.
closes issue #11928)
Reported by: adriavidal
Patches:
1.6.0-configurev2.patch uploaded by putnopvut (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul 2008) | 2 lines
remove the dlfcn compatibility stuff, because no platforms that Asterisk currently runs on it use it, and it doesn't build anyway
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
steps are to convert channel.c and chan_iax2.c to use this new API, and to move all the DAHDI-specific timing source code into a new res_timing_dahdi module
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 lines
Add linux-gnueabi in.
(closes issue #12529)
Reported by: tzafrir
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
processing thread. Modules reference a taskprocessor, push tasks into the taskprocessor as needed, and unreference the taskprocessor when the taskprocessor is no longer needed.
A task wraps a callback function pointer and a data pointer and is managed internal to the taskprocessor subsystem. The callback function is responsible for releasing task data.
Taskprocessor API
* ast_taskprocessor_get(..) - returns a reference to a taskprocessor
* ast_taskprocessor_unreference(..) - releases reference to a taskprocessor
* ast_taskprocessor_push(..) - push a task into a taskprocessor queue
Check doxygen for more details
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines
Pass in the path to Zaptel for systems that install Zaptel headers in a separate location.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
an http post are not read into memory.
This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros.
If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar 2008) | 5 lines
check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4)
(closes issue #12179)
Reported by: Netview
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines
fix up various compiler warnings found with gcc-4.3:
- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)
- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement
- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur
- main/editline/readline.c had an unused variable
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008) | 7 lines
Compatibility fix for PPC64
(closes issue #12081)
Reported by: jcollie
Patches:
asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
Tested by: jcollie, Corydon76
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008) | 4 lines
Fix the last couple of issues related to building from a path that contains spaces.
(closes issue #11834)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line
This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
not work for everyone, but it did for some. This set of changes makes trunk
start again for those having problems. Instead of building libresample as a
static library, it just links the object files in directly with the asterisk
binary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
res_resample, and mark codec_resample as dependent upon res_resample. This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places. (I have another module
in a branch that needs it, too.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This commit imports libresample for use in Asterisk. It also adds a new codec
module, codec_resample. This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.
It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz
signed linear. But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(not complete yet - still need to sort out dependecies on res_*)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
committed to trunk. Forgive me in advance any disruption this may cause, and please, report any problems via the bugtracker. The upside is that this can speed up large dialplans by 20 times (or more). Context, extension, and priority matching are all fairly constant-time searches. I introduce here my hashtables (hashtabs), and a regression for them. I would have used the ast_obj2 tables, but mine are resizeable, and don't need the object destruction capability. The hashtab stuff is well tested and stable. I introduce a data structure, a trie, for extension pattern matching, in which knowledge of all patterns is accumulated, and all matches can be found via a single traversal of the tree. This is per-context. The trie is formed on the first lookup attempt, and stored in the context for future lookups. Destruction routines are in place for hashtabs and the pattern match trie. You can see the contents of the pattern match trie by using the 'dialplan show' cli command when 'core set debug' has been done to put it in debug mode. The pattern tree traversal only traverses those parts of the tree that are interesting. It uses a scoreboard sort of approach to find the best match. The speed of the traversal is more a function of the length of the pattern than the number of patterns in the tree. The tree also contains the CID matching patterns. See the source code comments for details on how everything works. I believe the approach general enough that any issues that might come up involving fine points in the pattern matching algorithm, can be solved by just tweaking things. We shall see. The current pattern matcher is fairly involved, and replicating every nuance of it is difficult. If you find and report problems, I will try to resolve than as quickly as I can. The trie and hashtabs are added to the existing context and exten structs, and none of the old machinery has been removed for the sake of the multitude of functions that use them. In the future, we can (maybe) weed out the linked lists and save some space.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
