Commit Graph

14 Commits

Author SHA1 Message Date
Corey Farrell
824c8d4b6b Fix dialplan function NULL channel safety issues
(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-27 19:06:13 +00:00
Sean Bright
61f2bc9adc Minor spelling fix to the VOLUME documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-30 17:04:34 +00:00
Mark Michelson
a64586e5c2 Fix a deadlock that occurs when func_volume is used on a local channel.
This was discovered by trying to perform a call forward to an extension
that makes use of func_volume. When the local channel is optimized away,
the datastore on the local;2 channel would have its audiohook destroyed
rather than detaching the audiohook from the channel and then destroying
it.

With this patch, func_volume's datastore destructor takes the proper
route of detaching the audiohook and then destroying it.

(closes issue ASTERISK-19611)
reported by Volker Sauer
Patches:
	ASTERISK-19611.patch uploaded by Mark Michelson (license #5049)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-13 20:59:01 +00:00
Mark Michelson
095054e4a1 Fix Coverity-reported ARRAY_VS_SINGLETON error.
As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none
took arguments. The proper thing to do for this case is to
pass NULL for the "args" parameter here. We were instead passing
a seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume.

(closes issue ASTERISK-19656)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01 23:08:20 +00:00
Leif Madsen
d4938a111e 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@328209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14 20:13:06 +00:00
Jonathan Rose
7ea558865a Merged revisions 310585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

........
  r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines
  
  Adds 'p' as an option to func_volume.  When it is on, the old behavior with DTMF controlling volume adjustment will be enforced.
  When it is off, DTMF will not be processed by the function.
  
  Programmed by Jonathan Rose
  Reviewed by David Vossel, Leif Madsen, and Russell Bryant
  
  http://reviewboard.digium.internal/r/93/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-14 15:27:57 +00:00
Tilghman Lesher
d8e0c58437 Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Olle Johansson
9b12df5731 By copying this code I got bad comments in reviewboard... Better fix the original.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-31 18:17:38 +00:00
Russell Bryant
5b168ee34b Merge changes from team/group/appdocsxml
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: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01 21:10:07 +00:00
Kevin P. Fleming
7df8b8b848 make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-05 16:56:11 +00:00
Luigi Rizzo
9c2aaeb701 remove some unnecessary includes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 20:42:06 +00:00
Luigi Rizzo
fdb7f7ba3d Start untangling header inclusion in a way that does not affect
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: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 20:04:58 +00:00
Russell Bryant
040a5f20f9 * Constify the uid field of channel datastores
* Convert some spaces to tabs in func_volume
* Add a note in channel.h making it clear that none of the datastore API calls
  lock the channel they are given, so the channel should be locked before
  calling the functions that take a channel argument.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-28 18:32:56 +00:00
Joshua Colp
602198c402 Merge audiohooks branch into trunk. This is a new API for developers to listen and manipulate the audio going through a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 19:30:52 +00:00