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${ noResults }
78 Commits (ebb266391fc54b47d561cc1a0046ecbf14d66e17)
Author | SHA1 | Message | Date |
---|---|---|---|
|
de10acaa07 |
Merged revisions 108084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait). (closes issue #11945) Reported by: xheliox ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108085 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
d5b454bf8d |
Convert ast_verbose to ast_verb.
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
99308dfb4e |
Conversions of free to ast_free, where applicable, and several other formatting fixes.
Reported by: eliel Patch by: eliel,tilghman (Closes issue #11209) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
9905034266 |
Merged revisions 89587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines Close the audio file before sending it to the post processing application. (closes issue #11357) Reported by: reformed Patches: mixmonitor.patch uploaded by reformed (license 330) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
51391e6b09 |
shuffle a little bit the content of header files to reduce dependencies.
In this commit: - move the ast_register/unregister_app functions to module.h to avoid the need to include pbx.h for the simpler apps; - move the ast_group structure to channel.h to remove the dependency of app.h on linkedlists.h Note, this is a long process that I am doing in small steps. The main difficulty is that now for each subsystem we have a single header (e.g. channel.h) included by the subsystem provider (usually one file, e.g. channel.c) and by its clients (dozens of them, e.g. we have some 70+ apps and 30+ functions). This requires the clients to include all the extra headers required by the provider (eg. lock.h, linkedlists.h, definitions of substructures...) even though many of the clients would be just happy with opaque struct declarations and function prototypes. The long term plan is to eventually rectify this structure so that the compilation can become faster, and also APIs are more stable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
ea2c54859d |
more removal of redundant headers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
7e8835e0d7 |
remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
a23c055c3d |
move asterisk/paths.h outside asterisk.h and into those files
who really need it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
0595b5e2aa |
include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
fdb7f7ba3d |
Start untangling header inclusion in a way that does not affect
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
e4bb565530 |
Merged revisions 89241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 Nov 2007) | 5 lines Reverting commit made in revision 89205 since it is unnecessary. Thanks to Kevin for pointing this out ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89242 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
13c49f6cce |
There is the potential to copy uninitialized memory into the mixmonitor->post_process
string. This fix prevents that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89207 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
1d4c579422 |
Merged revisions 89205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options and args.post_process strings are uninitialized and could contain garbage. This change handles this situation properly by only using arguments that we have parsed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89206 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
98429d37b2 |
Based on a note in asterisk-dev by Brian Capouch, I determined I too agressive in not initializing arrays passed to pbx_substitute_variables_xxxx; I reviewed the code (again) and hopefully found every possible spot where substitute_variables is called conditionally, and made sure the char array involved was set to a null string.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89186 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5a4867543d |
"show application <foo>" changes for clarity.
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
63f2f04cf4 |
This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
df6237eac3 |
Add volume adjustment in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
f066bfc38a |
Restore operation of the option that only writes when the channel is bridged.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87850 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
ebe4050128 |
Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b0f3e6097e |
Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e97a723cf1 |
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)
(closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
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 |
|
20bbd09de3 |
Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0e969271ae |
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
8a7732f067 |
via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
b8cd949cce |
Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
96a646734f |
It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
150c77c901 |
Merged revisions 72381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72381 | file | 2007-06-27 19:25:12 -0400 (Wed, 27 Jun 2007) | 10 lines Merged revisions 72378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2 lines Update documentation to clarify variable usage with MixMonitor. (issue #9494 reported by netoguy) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
0296bd9957 |
Issue 9970 - Ensure directory exists before trying to write an output file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
9d05ff8ed5 |
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
13a3661906 |
Add support for autocompleting start/stop options of the mixmonitor CLI command. (issue #9862 reported by eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
4b3a3fb14c |
Add a new API call for creating detached threads. Then, go replace all of the
places in the code where the same block of code for creating detached threads was replicated. (patch from bbryant) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
095b911df1 |
Merged revisions 52717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52717 | file | 2007-01-29 18:43:40 -0500 (Mon, 29 Jan 2007) | 10 lines Merged revisions 52716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 lines Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
48e707b6aa |
Merged revisions 52163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52163 | file | 2007-01-24 20:51:35 -0500 (Wed, 24 Jan 2007) | 10 lines Merged revisions 52162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 lines Add another note about audio files being played back to each bridged party. (issue #8718 reported by ppyy) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
ea17720d1e |
Merged revisions 51407 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51407 | file | 2007-01-22 14:13:44 -0500 (Mon, 22 Jan 2007) | 10 lines Merged revisions 51406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 lines Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51408 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
7b338004bf |
a quick fix to app_sms.c to get rid of cursed compiler warnings so I can compile under --enable-dev-mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48767 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
3c876af5cf |
Merged revisions 44378 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines update thread creation code a bit reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44379 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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fcb999c01c |
merge qwell's CLI verbification work
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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55d594e9da |
Make the difference clear about what the responsibilities of the core and a spy are when it comes to spying on a channel. The core is responsible for adding a spy to a channel, feeding frames into the spy, removing the spy from the channel, and notifying the spy that is has been detached. The spy is responsible for reading frames in, and cleaning itself up. Each side will not try to do the other's job.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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f7e7161607 |
Merge team/russell/frame_caching
There are some situations in Asterisk where ast_frame and/or iax_frame structures are rapidly allocatted and freed (at least 50 times per second for one call). This code significantly improves the performance of ast_frame_header_new(), ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping a thread-local cache of these structures and using frames from the cache whenever possible instead of calling malloc/free every time. This commit also converts the ast_frame and iax_frame structures to use the linked list macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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0a27d8bfe5 |
merge new_loader_completion branch, including (at least):
- restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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0af1edefb6 |
Merged revisions 33841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r33841 | kpfleming | 2006-06-13 08:30:06 -0500 (Tue, 13 Jun 2006) | 2 lines memory allocation optimizations ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
c311827dc9 |
Merged revisions 33724 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r33724 | file | 2006-06-12 18:34:38 -0300 (Mon, 12 Jun 2006) | 2 lines Greatly simply the mixmonitor thread, and move channel reference directly to spy structure so that the core can modify it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33725 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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472c1ca282 |
simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
04ecb29d03 |
remove almost all of the checks of the result from ast_strdupa() or alloca().
As it turns out, all of these checks were useless, because alloca will never return NULL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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03120e00ad |
use ast_channel_(un)lock
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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fb7757440f |
remove a check of the result from ast_mutex_lock
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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e43bc6634d |
This rather large commit changes the way modules are loaded.
As partly documented in loader.c and include/asterisk/module.h, modules are now expected to return all of their methods and flags into a structure 'mod_data', and are normally loaded with RTLD_NOW | RTLD_LOCAL, so symbols are resolved immediately and conflicts should be less likely. Only in a small number of cases (res_*, typically) modules are loaded RTLD_GLOBAL, so they can export symbols. The core of the change is only the two files loader.c and include/asterisk/module.h, all the rest is simply adaptation of the existing modules to the new API, a rather mechanical (but believe me, time and finger-consuming!) process whose detail you can figure out by svn diff'ing any single module. Expect some minor compilation issue after this change, please report it on mantis http://bugs.digium.com/view.php?id=6968 so we collect all the feedback in one place. I am just sorry that this change missed SVN version number 20000! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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f10f427d49 |
since the module API is changing, it's a good time to const-ify the description() and key() return values
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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c88e03113f |
Merged revisions 11778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r11778 | bweschke | 2006-03-04 06:45:37 -0500 (Sat, 04 Mar 2006) | 3 lines Substitute variables in the post_process string (if it exists) before those variables could possibly disappear (channel hangup) #6462 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11779 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |