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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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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r19891 | tilghman | 2006-04-13 15:37:50 -0500 (Thu, 13 Apr 2006) | 2 lines
Bug 6947 - Allow vm broadcasts to more than 256 characters worth of mailboxes
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r19812 | kpfleming | 2006-04-13 12:40:21 -0500 (Thu, 13 Apr 2006) | 2 lines
oops... let's not set a variable and then immediately overwrite it while assuming its old value will magically return
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dialplan functions, and some other misc things.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r19397 | tilghman | 2006-04-11 17:39:59 -0500 (Tue, 11 Apr 2006) | 2 lines
Bug 6490 - telco intercept should report NOANSWER instead of CHANUNAVAIL
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r19394 | tilghman | 2006-04-11 17:24:46 -0500 (Tue, 11 Apr 2006) | 2 lines
Bug 6061 - Fix ODBC storage of VM on PGSQL and MSSQL
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r19348 | kpfleming | 2006-04-11 16:50:18 -0500 (Tue, 11 Apr 2006) | 2 lines
don't call the originating device as part of the Page() operation (issue #6932)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r19301 | kpfleming | 2006-04-11 15:11:01 -0500 (Tue, 11 Apr 2006) | 2 lines
handle call time limit properly when warning is requested _after_ call would hae already ended (issue #6356)
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r19008 | bweschke | 2006-04-10 20:05:01 -0500 (Mon, 10 Apr 2006) | 3 lines
When using the silence detector in ast_play_and_record() and ast_play_and_prepend(), the truncation code never gets called to remove the detected silence, because the value of res is zero when control gets to that point. #6903 w/some mods (softins)
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API! I wonder what we can do with this now...
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description() and key() return values
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r18494 | tilghman | 2006-04-08 14:20:45 -0500 (Sat, 08 Apr 2006) | 2 lines
Bug 6914 - .txt file fails to rename on operator out
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(issue #6904)
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- implement all functions through internal APIs in res_osp.c and app_osplookup.c
(homesick)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r18087 | kpfleming | 2006-04-06 17:04:58 -0500 (Thu, 06 Apr 2006) | 3 lines
don't try to support 'i' or 'r' options if chan_zap is not loaded, and warn the user when they attempt to use them (issue #6675)
update application help text to more clearly define when Zaptel and chan_zap are required
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application wouldn't block indefinitely looking for another frame from that channel. Don't try to do frame size analysis on a frame that isn't voice, only report DEBUG and VERBOSE msgs to the logger channels when the DEBUG and VERBOSE settings are high enough to require it, and some other minor cleanups.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r17905 | file | 2006-04-06 14:00:10 -0300 (Thu, 06 Apr 2006) | 2 lines
Update email address to my Digium one
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(STATIC_MODULE) to show what needs to be changed.
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rand() to threadsafe ast_random()
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returns differently
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- move "res=-1" out of verbose block.
- minor code cleanup
Imported from 1.2 branch
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https://svn.digium.com/svn/asterisk/branches/1.2
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r17283 | mogorman | 2006-04-04 10:24:50 -0500 (Tue, 04 Apr 2006) | 3 lines
Adds documentation to show what the w flag.
Patch from Ian Kinner at Digium.
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in pbx_exec is always 1 so it can be removed.
This change also takes away ast_exec_extension(), and lets all
switch functions (exists, canmatch, exec, matchmore) all use the same
prototype, which makes the code a bit cleaner.
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Thanks everyone involved for hard work, testing and testing!
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r16192 | tilghman | 2006-03-29 13:11:18 -0600 (Wed, 29 Mar 2006) | 2 lines
Bug 6830 - Let GosubIf work with the same conditions as a GotoIf (change in API approved by Russell)
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On passing, replace strdup with ast_strdup() and remove
the now useless checks for NULL since ast_strdup() can handle
it correctly.
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so that new apps will not try to use it by mistake.
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to S_OR functions. from bug note 6805 with minor
modifications.
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various code cleanups
don't steal copyright from author
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on reload.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14830 | tilghman | 2006-03-24 23:01:16 -0600 (Fri, 24 Mar 2006) | 2 lines
Fix incorrect size of zeroing (left over from when maxmsg was hardcoded at 100)
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r14821 | tilghman | 2006-03-24 22:09:04 -0600 (Fri, 24 Mar 2006) | 2 lines
Bug 6783 - When context is specified, voicemail should look for mailboxes in that context
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#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
to these files.
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r14659 | bweschke | 2006-03-23 22:53:42 -0600 (Thu, 23 Mar 2006) | 3 lines
Fix a typo in the app description
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