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scalability and is done in such a way that we should be able to add support for other poll() replacements.
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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microsecond, instead of only to the second
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r74572 | file | 2007-07-11 14:03:08 -0300 (Wed, 11 Jul 2007) | 2 lines
Instead of figuring out kernel versions that have compiler.h and not... let's just use autoconf to check for it's presence. (issue #10174 reported by francesco_r)
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r74515 | file | 2007-07-11 11:09:13 -0300 (Wed, 11 Jul 2007) | 2 lines
Only check if we need to do a SIGMA based tone generation if we have a card. (issue #10179 reported by mikowhy)
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r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007) | 2 lines
use ast_localtime() in every place localtime_r() was being used
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines
Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky)
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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reported by ywalther)
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r48088 | file | 2006-11-28 11:57:16 -0500 (Tue, 28 Nov 2006) | 10 lines
Merged revisions 48087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines
According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3)
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r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, 09 Nov 2006) | 10 lines
Merged revisions 47379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines
Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods.
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r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006) | 2 lines
apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process
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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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patch provided in bugnote, with minor changes.
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now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it
had a non static function when it should.
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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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)
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(issue #7710 reported by Mithraen)
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dialtone generation to the board. If you're using a quicknet board where this might work, give it a try as well. (issue #6092 reported by ywalther - minor mods by moi)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines
suppress a compiler warning about the usage of a potentially uninitialized variable
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r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines
Fix an issue that would cause a NewCallerID manager event to be generated
before the channel's NewChannel event. This was due to a somewhat recent
change that included using ast_set_callerid() where it wasn't before. This
function should not be used in the channel driver "new" functions.
(issue #7654, fixed by me)
Also, fix a couple minor bugs in usecount handling. chan_iax2 could have
increased the usecount but then returned an error. The place where chan_sip
increased the usecount did not call ast_update_usecount()
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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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r36725 | russell | 2006-07-03 00:19:09 -0400 (Mon, 03 Jul 2006) | 4 lines
use ast_set_callerid to be more consistent and to make sure that the
"callerid" option in the conf files is always handled the same way and sets ANI
(issue #7285, gkloepfer)
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again :-)
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->indicate()
update iax2_indicate to pass control frame payload to the connected channel
add an API call for sending an indication with payload, and use it for control frames with payload
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autoconf and menuselect tools for Asterisk!
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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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const-ify some more APIs
remove 'type' field from ast_channel, in favor of the one in the channel's tech structure
allow string field module users to specify the 'chunk size' for pool allocations
update chan_alsa to be compatible with recent const-ification patches
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some places where strlen is used instead of ast_strlen_zero
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of usses to use the new functions (bug #4504)
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pattern filtering
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base-10 parsing (bug #4110)
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(bug #4058)
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