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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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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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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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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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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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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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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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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r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | 9 lines
Don't report a syntax error when an empty string is passed to ast_get_group.
Just return 0.
(closes issue #11540)
Reported by: tzafrir
Patches:
group_empty.diff uploaded by tzafrir (license 46)
-- slightly changed by me
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r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines
Change ast_translator_best_choice() to only pay attention to audio formats.
This fixes a problem where Asterisk claims that a translation path can not be
found for channels involving video.
(closes issue #11638)
Reported by: cwhuang
Tested by: cwhuang
Patch suggested by cwhuang, with some additional changes by me.
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r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines
Use the constant that I really meant to use here ...
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r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | 4 lines
Just in case the AST_FLAG_END_DTMF_ONLY flag was already set before starting
autoservice, remember it and ensure that the channel has the same setting when
autoservice gets stopped. (pointed out by d1mas, patched up by me)
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r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | 4 lines
When a channel is in autoservice, mark a flag on the channel that says that
we only care about the END of a digit. That way, no magic digit emulation stuff
will happen when all we're doing is queueing up END frames.
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r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | 5 lines
Don't store DTMF BEGIN frames while a channel is in autoservice. It's just
going to make ast_read() do a lot of extra work when the channel comes back
out of autoservice.
(closes issue #11628, patched by me)
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r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) | 5 lines
Race: we need to wait to queue a NewChannel event until after the channel is
inserted into the channel list. The reason is because some manager users
immediately queue requests from the channel when they see that event and are
confused when Asterisk reports no such channel. (Closes issue #11632)
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r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec 2007) | 6 lines
Since we are freeing list elements within a list traversal, we need to use the safe
traversal and remove the item from the list before freeing it.
(closes issue 11612, reported by dtyoo)
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so that paths and filename are writable by asterisk.c without
causing segfaults.
This involves defining the variables as const char *,
and having them point to as static, writable buffer
defined in asterisk.c
On passing, fix some errors in using these variables
in some files in utils/ , and in res/snmp/agent.c
which was redefining a variable without using paths.h
(not applicable to 1.4)
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Mark Michelson
Russell Bryant
Tilghman Lesher
Steve Murphy
Terry Wilson
Jason Parker
Joshua Colp
Kevin P. Fleming
Luigi Rizzo