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https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232012 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14859)
Reported by: atis
Patches:
20090821__issue14859.diff.txt uploaded by tilghman (license 14)
20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: atis, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221044 f38db490-d61c-443f-a65b-d21fe96a405b
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The 'pglobal' tool is quite handy indeed :-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200620 f38db490-d61c-443f-a65b-d21fe96a405b
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This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191140 f38db490-d61c-443f-a65b-d21fe96a405b
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This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
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channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89425 f38db490-d61c-443f-a65b-d21fe96a405b
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Reproduce with
grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89348 f38db490-d61c-443f-a65b-d21fe96a405b
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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.).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
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autoservice in func_lock and func_shell.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87233 f38db490-d61c-443f-a65b-d21fe96a405b
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3 seconds while waiting on a lock when other locks are currently held to
avoid deadlocks. Change the code to reflect this.
* Since trying to grab a lock may block for some time, put the channel in
autoservice so that audio is still read from the channel and that any
active generators on the channel don't pause.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84143 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79859 f38db490-d61c-443f-a65b-d21fe96a405b
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deadlock avoidance
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79813 f38db490-d61c-443f-a65b-d21fe96a405b
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requested by Russell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77838 f38db490-d61c-443f-a65b-d21fe96a405b
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are now deprecated.
(Closes issue #10264)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77834 f38db490-d61c-443f-a65b-d21fe96a405b
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