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
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and likely in other places too.
This is quite useful when placing mail/html stuff in config files.
/*!
\brief Convert some C escape sequences (\b\f\n\r\t) into the
equivalent characters.
\brief s The string to be converted (will be modified).
\return The converted string.
*/
char *ast_unescape_c(char *s);
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk goes into a busy loop, the machine will be recoverable. We'd still
need to do a restart to put Asterisk back into high priority mode, but at
least a reboot won't be required. (Closes issue #11559)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec 2007) | 6 lines
Rework deadlock avoidance used in ast_write, since it meant that agent channels which were being monitored
had one audio file recorded and one empty audio file saved.
(closes issue #11529, reported by atis patched by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
globally visible.
Document the fact that DEFAULT_TMP_DIR cannot be overridden
from the default configuration (this needs to be fixed, as you
could have a totally different spooldir configured at runtime,
and yet DEFAULT_TMP_DIR keeps the compile-time default).
Remove two unused entries for sounds and images.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7 lines
Do not try to access information about a lock when printing out a trylock attempt. It is possible for the lock that it references to no longer be valid. This would have caused segfaults or deadlocks.
(issue #BE-263)
(closes issue #11080)
Reported by: callguy
(closes issue #11100)
Reported by: callguy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
you need to reload after changes. Thanks YS.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(related to issue #11414)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
manager realtime implementation.
If you change accounts in manager.conf, you now need to reload to activate the
changes (deletions, additions). This was not the case with 1.4.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(closes issue #11414)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
equivalent to the check done by ast_verb, I wrote a macro, VERBOSITY_LEVEL, which does this
check. I did a quick look in the source and used this macro in some places where option_verbose
was used.
I also converted some verbose messages in logger.c to use ast_verb instead of ast_verbose.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10690)
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r92696 | qwell | 2007-12-12 18:11:09 -0600 (Wed, 12 Dec 2007) | 7 lines
If a typo is found in a config file, we previous continued on with what was already loaded.
We do not want to do this (see bug below for details).
This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded.
Issue 10690.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
