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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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mixmonitor->post_process
string. This fix prevents that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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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
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(closes issue #11171, reported and patched by blitzrage)
Many thanks!
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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.
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(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
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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
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few other formatting cleanups, too.
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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
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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
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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
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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)
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guidelines changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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command. (issue #9862 reported by eliel)
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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)
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compile under --enable-dev-mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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.
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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
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- 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
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r33841 | kpfleming | 2006-06-13 08:30:06 -0500 (Tue, 13 Jun 2006) | 2 lines
memory allocation optimizations
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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again :-)
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
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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!
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description() and key() return values
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
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Rename and export ast_complete_channels for use by cli completion functions
that want to complete from the list of active channels
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(revisions 8378 through 8381)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7740 | kpfleming | 2006-01-03 11:24:56 -0600 (Tue, 03 Jan 2006) | 4 lines
revert incorrect fix for bug #6048 from revision 7709
put in correct (simpler) fix
add doxygen docs for channel spy 'state' values
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/branches/1.2
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