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${ noResults }
70 Commits (827f655dac2bc438e3fae3b1573324b8a503e024)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Kevin P. Fleming
|
c3347e79a5 |
Merged revisions 159818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@159851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Kevin P. Fleming
|
bc5b69d250 |
Merged revisions 144949-144951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r144949 | kpfleming | 2008-09-27 10:52:56 -0500 (Sat, 27 Sep 2008) | 17 lines 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 ........ r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines fix some minor issues with rev 144924 ........ ................ r144950 | kpfleming | 2008-09-27 11:10:33 -0500 (Sat, 27 Sep 2008) | 2 lines fix bugs caused by r144949 when MALLOC_DEBUG is defined ................ r144951 | kpfleming | 2008-09-27 11:17:43 -0500 (Sat, 27 Sep 2008) | 1 line remove incorrect comment ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@144993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
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 |
17 years ago |
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 |
17 years ago |
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 |
17 years ago |
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 |
17 years ago |
|
Kevin P. Fleming
|
9a08061ea3 |
Merged revisions 131921 via svnmerge from
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 |
17 years ago |
|
Kevin P. Fleming
|
fd4a60c459 |
Merged revisions 125132 via svnmerge from
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 |
17 years ago |
|
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 |
17 years ago |
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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116353 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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 |
17 years ago |
|
Joshua Colp
|
b7b2e732f0 |
Merged revisions 112711 via svnmerge from
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 |
17 years ago |
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 |
17 years ago |
|
Kevin P. Fleming
|
c7eebb3db8 |
Merged revisions 107408 via svnmerge from
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 |
18 years ago |
|
Kevin P. Fleming
|
79c3038ee5 |
Merged revisions 107352 via svnmerge from
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 |
18 years ago |
Tilghman Lesher
|
7d564048ed |
Merged revisions 104868 via svnmerge from
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 |
18 years ago |
|
Russell Bryant
|
793edbecd0 |
Merged revisions 100932 via svnmerge from
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 |
18 years ago |
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 |
18 years ago |
|
Russell Bryant
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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 |
18 years ago |
Steve Murphy
|
33fadcc67c |
Merged revisions 97849 via svnmerge from
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 |
18 years ago |
|
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 |
18 years ago |
|
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 |
18 years ago |
|
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 |
18 years ago |
|
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 |
18 years ago |
|
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 |
18 years ago |
|
Russell Bryant
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adf3b12e55 |
Fix a silly little typo :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
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 |
18 years ago |
Luigi Rizzo
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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 |
18 years ago |
|
Luigi Rizzo
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a9395206f9 |
conditional targets for building the windows version
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89377 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Luigi Rizzo
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7cd78079ae |
more cygwin/mingw32 compatibility fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89373 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Luigi Rizzo
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5862c55451 |
use poll as detected by configure
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89355 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Steve Murphy
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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 |
18 years ago |
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Luigi Rizzo
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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 |
18 years ago |
|
Steve Murphy
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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 |
18 years ago |
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Russell Bryant
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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 |
18 years ago |
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Russell Bryant
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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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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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 |
18 years ago |
|
Steve Murphy
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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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Luigi Rizzo
|
3caa225108 |
Make sure that we properly recurse in subdirectories to
check dependencies for libraries. Because these targets (e.g. minimime/libmmime.a) are real ones, declaring them .PHONY would cause them to be rebuilt every time (see e.g. SVN 64355). As a workaround I am using the following CHECK_SUBDIR target: CHECK_SUBDIR: # do nothing, just make sure that we recurse in the subdir/ minimime/libmmime.a: CHECK_SUBDIR @cd minimime && $(MAKE) libmmime.a which seems to do a better job than .PHONY (probably because .PHONY forces the rebuild even if the recursive make does not think it is necessary). If this turns out to be the correct approach, we can then merge it back into 1.4 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
42c63dd2ed |
Add the adsistub file to the Asterisk makefile, fix a stub definition, and no longer make the symbols from res_adsi global since they don't need to be.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65233 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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ee7301ac68 |
With libmmime.a as a .PHONY target, asterisk gets rebuilt every time, but without proper ASTCFLAGS.
This caused a problem with the buildinfo.o file not being able to find asterisk/build.h This was affecting DESTDIR, but I *think* that if asterisk had never been installed before, it would've failed also. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64355 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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b6b1bf3213 |
Merge changes from team/russell/events
This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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0a9750ef9f |
Merged revisions 60603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines To be able to achieve the things that we would like to achieve with the Asterisk GUI project, we need a fully functional HTTP interface with access to the Asterisk manager interface. One of the things that was intended to be a part of this system, but was never actually implemented, was the ability for the GUI to be able to upload files to Asterisk. So, this commit adds this in the most minimally invasive way that we could come up with. A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in the parser, and updated it to be thread-safe. The ability to check permissions of active manager sessions was added by Dwayne Hubbard. Then, hacking this all together and do doing the modifications necessary to the HTTP interface was done by me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60604 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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90b75256d9 |
Merged revisions 59573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59573 | qwell | 2007-03-30 17:50:31 -0500 (Fri, 30 Mar 2007) | 2 lines Add linux-uclibc host arch..."thingy". Sorry, I don't know what it's called... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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ef5aba253e |
Merged revisions 53464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53464 | russell | 2007-02-07 14:07:39 -0600 (Wed, 07 Feb 2007) | 4 lines The clean target actually needs to run "distclean" on editline. This is because we need to make sure that its configure script gets executed again, because the CFLAGS we want to pass to editline may have changed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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f351f4278a |
Merged revisions 52695 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52695 | russell | 2007-01-29 16:58:09 -0600 (Mon, 29 Jan 2007) | 2 lines Add a missing quotation mark. This was pointed out by jcmoore on #asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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5dfed82cb9 |
Merged revisions 52647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52647 | russell | 2007-01-29 15:36:56 -0600 (Mon, 29 Jan 2007) | 3 lines ASTLDFLAGS needs to be passed to the editline configure script as LDFLAGS. (issue #8928, zandbelt) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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9826fc599b |
Merged revisions 52049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 lines Merge in dialing API and the app_page that uses it. (issue #BE-118) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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80243d0f0c |
forgot to add AST_LIBS += $(BKTR_LIB)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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fa16f36aae |
Merged revisions 51262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) | 5 lines Ensure that the locations given to the Asterisk configure script for ncurses, curses, termcap, or tinfo are further passed along to the editline configure script. This fixes some cross-compilation environments. (issue #8637, reported by ovi, patch by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |