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#defines come out the same between the parser & lexer.
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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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(closes issue #11749)
Reported by: srt
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r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | 3 lines
Revert a change that introduces an unacceptable performance hit and is causing
memory leaks ... (from rev 97973)
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MSet. It is extremely hard to debug this issue so this should make it easier.
(closes issue #11759)
Reported by: caio1982
Patches:
setvar_space_warning1.diff uploaded by caio1982 (license 22)
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audiohooks. This causes an error when we attempt to destroy the lock later
when freeing the audiohook.
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variable (or function) on an active channel from the CLI.
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functions correctly. (Closes issue #11749)
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r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines
If the incoming RTP stream changes codec force the bridge to break if the other side does not support it.
(closes issue #11729)
Reported by: tsearle
Patches:
new_codec_patch_udiff.patch uploaded by tsearle (license 373)
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r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines
Properly report the hangup cause as no answer when someone does not answer
(closes issue #10574, reported by boch, patched by moy)
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sample length with g722. It is _2_ samples per byte, not 1. This was all
over the place, and I believed it, and it is what caused me to take so long
to figure out what was broken.
- Update copyright information on codec_g722.
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This fix was made in favor of the proposed patch since it doesn't involve changing
a core codec define.
(closes issue #11722, reported and initially patched by caio1982, final patch by me)
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Instead of using is16kHz(), implement a format_rate() function.
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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines
1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running. Don't just ignore it.
(Closes issue #11698)
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r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines
Fix various timing calculations that made assumptions that the audio being
processed was at a sample rate of 8 kHz.
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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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allowed, not nobody)
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1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
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based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
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(closes issue #11718)
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r97622 | qwell | 2008-01-09 14:28:43 -0600 (Wed, 09 Jan 2008) | 5 lines
Correctly display a message if a command could not be found.
Also fix a comment which may have led to this happening.
Issue 11718, reported by kshumard.
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r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 line
Fix some locking and return value funkiness. We really shouldn't be unlocking this lock inside of a function, unless we locked it there too.
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r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008) | 5 lines
Allow filename completion on zero-length modules, remove a memory leak, remove
a file descriptor leak, and make filename completion thread-safe.
Patched and tested by tilghman.
(Closes issue #11681)
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r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) | 3 lines
Increase constants to where we're less likely to hit them while debugging.
(Closes issue #11694)
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r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) | 3 lines
Apply multiple crash fixes, found in issue #11386, but not completely
closing that issue.
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mappings".
Closes issue #11704, patch by kshumard.
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Previously they would get registered twice because of the way manager.c operates.
(closes issue #11699)
Reported by: caio1982
Patches:
manager_module_commands1.diff uploaded by caio1982 (license 22)
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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.
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of applications when they get registered.
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r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) | 2 lines
Don't pass an empty string as the device name.
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r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) | 7 lines
Fix the problem of notification of a device state change to a device with a '-'
in the name. Could probably do with a better fix in trunk, but this bug has
been open way too long without a better solution.
Reported by: stevedavies
Patch by: tilghman
(Closes issue #9668)
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Reported and initially patched by: michael-fig
(Closes issue #11340)
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Report and fix by: mvanbaak
(Closes issue #11669)
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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.
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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)
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gdb dying while debugging asterisk. The problem seems to be related
with a race in the handling of module_list, which in turn is triggeded
by calling dlopen() on a system which uses initializers to create
locks.
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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.)
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r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines
Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then
calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is
dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension
into ast_merge_contexts_and_delete (sans the extra lock).
(this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the
problematic area experienced by the reporters of that issue)
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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.
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