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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41633 f38db490-d61c-443f-a65b-d21fe96a405b
Also fix some locking issues I found at the same time.
Issue 7770, original patch by alamantia
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40757 f38db490-d61c-443f-a65b-d21fe96a405b
- 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: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
channel to remove the theoretical race condition that the channel could get
bridged before the channel's jitterbuffer gets configured. This was pointed
out by PCadach on IRC. Thanks!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39964 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
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
........
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()
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38905 f38db490-d61c-443f-a65b-d21fe96a405b
inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38042 f38db490-d61c-443f-a65b-d21fe96a405b
Abstract out some common code into a single function
With the recent scheduler "issues", it pointed out a few things I might have been missing,
so I added some rudimentary vrtp and rtcp stuff
General cleanup...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37629 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36726 f38db490-d61c-443f-a65b-d21fe96a405b
- don't call the byte swapping macros on single byte numbers
- don't do a ++ increment in the argument in the argument to the byte swapping
macros. This gets expanded to incrementing the variable 4 times in a single
operation, which results in undefined (and obviously undesired) behavior. :)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35766 f38db490-d61c-443f-a65b-d21fe96a405b
allocated. These changes caused crashes when using a channel type that did
not support the jitterbuffer. Instead of fixing why it's crashing, I'm going
to implement this in a better way next week. The way I did it caused a
jitterbuffer to be allocated on every channel where the channel type supported
jitterbuffers, even if they were disabled.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35746 f38db490-d61c-443f-a65b-d21fe96a405b
so that channels not using a jitterbuffer don't waste as much memory
- ensure that the channel drivers that use jitterbuffers can handle a failure
from configuring a jitterbuffer on a new channel because of a memory
allocation error
- On passing through these channel drivers, configure the jitterbuffer before
starting the PBX thread instead of afterwards. If the pbx fails to start for
whatever reason, this would have caused a crash.
- Also on passing, move the increase of the usecount to after all of the
possible failure conditions in the function
- fix a place where ast_update_use_count() was not called
- ensure that the owner channel pointer of the channel pvt strcutures is set to
NULL in failure conditions
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35553 f38db490-d61c-443f-a65b-d21fe96a405b
and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257, north, with additional updates and modifications)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31413 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26417 f38db490-d61c-443f-a65b-d21fe96a405b
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!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20003 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9060 f38db490-d61c-443f-a65b-d21fe96a405b
mogorman