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were included almost everywhere.
Remove some of the instances.
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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few other formatting cleanups, too.
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applications
using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
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ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
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ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r66879 | russell | 2007-06-01 14:35:13 -0500 (Fri, 01 Jun 2007) | 2 lines
List app_meetme as a module that app_page depends on.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58992 | file | 2007-03-16 12:12:28 -0400 (Fri, 16 Mar 2007) | 2 lines
Wait for the async thread to exit when hanging up all of the paged phones under all circumstances. (issue #9181 reported by PhilSmith)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines
- Add the ability to register a callback to monitor state changes in an
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 lines
Merge in dialing API and the app_page that uses it. (issue #BE-118)
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(issue #8673 reported by sunder)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46347 | qwell | 2006-10-26 15:25:44 -0500 (Thu, 26 Oct 2006) | 2 lines
Fix small formatting issue, that causes misaligned line
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42783 | tilghman | 2006-09-11 16:47:23 -0500 (Mon, 11 Sep 2006) | 4 lines
When paging, only wait 5 seconds for the marked user to enter the conference.
After that, assume the paging already completed by the time the channel entered
the conference and drop back out. (Issue 7275)
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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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again :-)
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24639 f38db490-d61c-443f-a65b-d21fe96a405b
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(JeffSaxe)
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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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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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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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description() and key() return values
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rand() to threadsafe ast_random()
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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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(revisions 8378 through 8381)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7265 | oej | 2005-12-01 17:18:14 -0600 (Thu, 01 Dec 2005) | 2 lines
Changing bug report address to the Asterisk issue tracker
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r7266 | kpfleming | 2005-12-01 17:18:29 -0600 (Thu, 01 Dec 2005) | 3 lines
Makefile 'update' target now supports updating from Subversion repositories (issue #5875)
remove support for 'patches' subdirectory, it's no longer useful
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r7268 | kpfleming | 2005-12-01 17:34:58 -0600 (Thu, 01 Dec 2005) | 2 lines
ensure channel's scheduling context is freed (issue #5788)
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r7269 | kpfleming | 2005-12-01 17:49:44 -0600 (Thu, 01 Dec 2005) | 2 lines
don't block waiting for the Festival server forever when it goes away (issue #5882)
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r7270 | kpfleming | 2005-12-01 18:26:12 -0600 (Thu, 01 Dec 2005) | 2 lines
allow variables to exist on both 'halves' of the Local channel (issue #5810)
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r7271 | kpfleming | 2005-12-01 18:28:48 -0600 (Thu, 01 Dec 2005) | 2 lines
protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879)
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r7272 | kpfleming | 2005-12-01 18:39:00 -0600 (Thu, 01 Dec 2005) | 3 lines
properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870)
reformat password changing code to conform to coding guidelines (issue #5870)
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r7273 | kpfleming | 2005-12-01 18:42:40 -0600 (Thu, 01 Dec 2005) | 2 lines
allow previous context-searching behavior to be used if desired (issue #5899)
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r7274 | kpfleming | 2005-12-01 18:51:15 -0600 (Thu, 01 Dec 2005) | 2 lines
inherit channel variables into channels created by Page() application (issue #5888)
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r7275 | oej | 2005-12-01 18:52:13 -0600 (Thu, 01 Dec 2005) | 2 lines
Bug #5907. Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
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In general, LOCAL_USER_ADD/REMOVE should be the first/last thing called in an
application. An exception is if there is some *fast* setup code that might
halt the execution of the application, such as checking to see if an argument
exists.
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STANDARD_HANGUP_LOCALUSERS
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