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2011-07-14Merged revisions 328247 via svnmerge from lmadsen1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ 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 ........ 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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-29Merge str_substitution branch.tilghman1-0/+1
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
2008-11-01Merge changes from team/group/appdocsxmlrussell1-5/+18
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
2007-11-22remove some useless includesrizzo1-5/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89521 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19another bunch of include removals (errno.h and asterisk/logger.h)rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89425 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16remove a bunch of duplicate includesrizzo1-1/+0
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
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+0
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
2007-01-24Doxygen updateoej1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51920 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-06finish const-ifying ast_func_read()kpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@49741 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-30show, list, view, display... whatever.oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46514 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-30Adding dialplan function IFMODULE, so you can create dialplans that handleoej1-0/+77
various PBX installations and checks if a module is loaded before using it. example IFMODULE(chan_sip3.so) issue #6671 in the bug tracker, finally gone. Thanks to mithraen for keeping it updated. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46513 f38db490-d61c-443f-a65b-d21fe96a405b