Commit Graph

82 Commits

Author SHA1 Message Date
Russell Bryant cba19c8a67 Convert the ast_channel data structure over to the astobj2 framework.
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
2009-04-24 14:04:26 +00:00
Tilghman Lesher 8f28bfc63e Merged revisions 187300-187301 via svnmerge from
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 04:59:05 +00:00
Joshua Colp 63de834395 Merge in the RTP engine API.
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
2009-04-02 17:20:52 +00:00
Russell Bryant 0bdd99ad64 Merged revisions 182810 via svnmerge from
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/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
Kevin P. Fleming ab3e9ddad1 Merged revisions 182808 via svnmerge from
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
2009-03-18 02:21:23 +00:00
Joshua Colp 4c9ab0df8c Merge phase 1 support for the new bridging architecture.
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
2009-03-05 18:18:27 +00:00
Steve Murphy b5a8a85d35 Merged revisions 177540 via svnmerge from
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.

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  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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19 23:56:50 +00:00
Russell Bryant bb03ef8d47 Add an implementation of the heap data structure.
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
2009-02-17 20:51:10 +00:00
Tilghman Lesher c8223fc957 Merge ast_str_opaque branch (discontinue usage of ast_str internals)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-13 08:36:35 +00:00
Kevin P. Fleming 9a7c28cd5a we can now build with -Wformat=2, which found a couple of real bugs
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
2008-11-29 15:29:33 +00:00
Eliel C. Sardanons 23adb8e509 Move all the XML documentation API from pbx.c to xmldoc.c.
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
2008-11-10 13:53:23 +00:00
Sean Bright bc1629a9e8 Fix build errors.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04 16:50:34 +00:00
Russell Bryant 5b168ee34b Merge changes from team/group/appdocsxml
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
2008-11-01 21:10:07 +00:00
Kevin P. Fleming 629861a705 Merged revisions 144924-144925 via svnmerge from
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
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  r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines
  
  fix some minor issues with rev 144924
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-27 15:52:56 +00:00
Kevin P. Fleming 7df8b8b848 make datastore creation and destruction a generic API since it is not really 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
2008-08-05 16:56:11 +00:00
Sean Bright 6cf6d9eca5 Merge in changes that allow Asterisk to be built against the Hoard
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
2008-08-03 16:14:14 +00:00
Mark Michelson 99db9f65b5 This commit compensates for buggy poll(2)
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
2008-07-28 19:53:56 +00:00
Russell Bryant c87f901cfd Remove libresample from the Asterisk source tree. It is now available in its
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
2008-07-21 14:47:41 +00:00
Kevin P. Fleming 9a08061ea3 Merged revisions 131921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-18 16:16:12 +00:00
Kevin P. Fleming fd4a60c459 Merged revisions 125132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-25 23:05:28 +00:00
Kevin P. Fleming 191081e45f add infrastructure so that timing source can be a loadable module... next 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
2008-06-12 14:21:32 +00:00
Joshua Colp 5ec16b6d6b Merged revisions 116352 via svnmerge from
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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 18:54:16 +00:00
Dwayne M. Hubbard b0b72e89a8 A taskprocessor is an object that has a name, a task queue, and an event 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
2008-05-03 03:40:32 +00:00
Joshua Colp b7b2e732f0 Merged revisions 112711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-04 00:53:19 +00:00
Terry Wilson e727d15d34 Replace minimime with superior GMime library so that the entire contents of 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
2008-03-17 22:10:06 +00:00
Kevin P. Fleming c7eebb3db8 Merged revisions 107408 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 14:09:49 +00:00
Kevin P. Fleming 79c3038ee5 Merged revisions 107352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 11:36:51 +00:00
Tilghman Lesher 7d564048ed Merged revisions 104868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-28 00:11:31 +00:00
Russell Bryant 793edbecd0 Merged revisions 100932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-29 17:44:05 +00:00
Jason Parker 3bd33214b9 Move code from res_features into (new file) main/features.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-23 23:09:11 +00:00
Russell Bryant b995c78c31 Merge changes from team/group/sip-tcptls
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
2008-01-18 22:04:33 +00:00
Steve Murphy 33fadcc67c Merged revisions 97849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-10 20:45:05 +00:00
Russell Bryant 54bc2c20b6 Now that the version.h file was getting properly regenerated every time the svn
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
2008-01-05 22:09:06 +00:00
Russell Bryant 6cfd6009b1 For some odd reason, the last set of libresample build changes from Kevin did
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
2008-01-02 16:20:26 +00:00
Kevin P. Fleming 04a10c145b go back to including libresample in the main Asterisk binary, but this time 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
2008-01-02 14:05:30 +00:00
Russell Bryant 78f4b28552 Instead of linking libresample into the main Asterisk binary, build it as
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
2008-01-02 01:00:44 +00:00
Russell Bryant 21cb767db7 Merge changes from team/russell/codec_resample
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
2007-12-31 21:22:31 +00:00
Russell Bryant adf3b12e55 Fix a silly little typo :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:36:59 +00:00
Mark Michelson c52d8a1cd5 Merged revisions 90735 via svnmerge from
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
2007-12-04 17:08:36 +00:00
Luigi Rizzo a7a0ca9f93 initial makefile changes to build loadable modules under cygwin
(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
2007-11-20 07:42:38 +00:00
Luigi Rizzo a9395206f9 conditional targets for building the windows version
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17 12:33:15 +00:00
Luigi Rizzo 7cd78079ae more cygwin/mingw32 compatibility fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17 10:54:52 +00:00
Luigi Rizzo 5862c55451 use poll as detected by configure
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17 03:07:06 +00:00
Steve Murphy a897556f7f This is the perhaps the biggest, boldest, most daring change I've ever 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
2007-11-09 16:00:22 +00:00
Luigi Rizzo b80dc41707 Move the last instance of AST_LIBS to the only place it is used,
namely main/Makefile .

I am unclear where decisions on the build environment (CFLAGS,
LDFLAGS, LIBS and so on) should be made - right now they are
split here and there.

As a first step in cleaning up this situation, i am trying to at
least collect all instances of each variable in one place.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05 21:27:04 +00:00
Steve Murphy 50d5fd877e Merged revisions 86881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1 line

this update to Makefile corrects how ast_expr2f.c should be generated
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-23 20:44:58 +00:00
Russell Bryant d99440e2ed Merged revisions 81342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) | 3 lines

If chan_h323 is not being built, don't use g++ to do the final link of Asterisk.
(in response to a question on the asterisk-dev list)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-29 15:59:10 +00:00
Russell Bryant 50d7fc81aa Merged revisions 80362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | 34 lines

Merge changes from team/russell/iax_refcount.

This set of changes fixes problems with the handling of iax2_user and iax2_peer
objects.  It was very possible for a thread to still hold a reference to one of
these objects while a reload operation tries to delete them.  The fix here is to
ensure that all references to these objects are tracked so that they can't go away
while still in use.

To accomplish this, I used the astobj2 reference counted object model.  This
code has been in one of Luigi Rizzo's branches for a long time and was primarily
developed by one of his students, Marta Carbone.  I wanted to go ahead and bring
this in to 1.4 because there are other problems similar to the ones fixed by these
changes, so we might as well go ahead and use the new astobj if we're going to go
through all of the work necessary to fix the problems.

As a nice side benefit of these changes, peer and user handling got more efficient.
Using astobj2 lets us not hold the container lock for peers or users nearly as long
while iterating.  Also, by changing a define at the top of chan_iax2.c, the objects
will be distributed in a hash table, drastically increasing lookup speed in these
containers, which will have a very big impact on systems that have a large number of
users or peers.

The use of the hash table will be made the default in trunk.  It is not the default
in 1.4 because it changes the behavior slightly.  Previously, since peers and users
were stored in memory in the same order they were specified in the configuration file,
you could influence peer and user matching order based on the order they are specified
in the configuration.  The hash table does not guarantee any order in the container,
so this behavior will be going away.  It just means that you have to be a little
more careful ensuring that peers and users are matched explicitly and not forcing
chan_iax2 to have to guess which user is the right one based on secret, host, and
access list settings, instead of simply using the username.

If you have any questions, feel free to ask on the asterisk-dev list.

........


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2007-08-22 20:44:23 +00:00
Joshua Colp 602198c402 Merge audiohooks branch into trunk. This is a new API for developers to listen and manipulate the audio going through a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 19:30:52 +00:00
Steve Murphy 94b934c8f6 Merged revisions 72933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1 line

support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary.
........


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2007-07-02 21:50:15 +00:00