Age | Commit message (Collapse) | Author |
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output to remove consoles. The prototypes added to logger.h still need
doxygen documentation, as well.
- Add the new command line option to the man page
- make the mute option a flag instead of an int since it is only a binary
option
- remove useless extern keywords for prototypes added to logger.h
- rename ast_console_mute() to ast_console_toggle_mute() since that is what
it actually does
- actually apply the mute option to newly created remote consoles instead of
only working when the CLI command is used
- don't imply the NO_FORK option if the mute command line option is provided
- place the new CLI command in the correct place in the list which has to be
in alphabetical order
- Finally, clean up a few spacing issues to conform to the coding guidelines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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reported by mavetju)
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using the Hangup application. (issue #7160 reported by kmilitzer branch by jcollie)
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is an error executing the AGI script, or the AGI script itself returns a
non-zero value, the AGISTATUS variable will now be set to FAILURE instead of
SUCCESS.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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update ast_mutex_init to allow mutexes that are all zero bytes to be initialized (in the case of a dynamically-allocated structure containing a mutex)
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/trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- with contributions from miconda, jcollie, and sb
- branch maintained by oej
Thanks everyone!
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messagecount2 to messagecount.
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than for 1.2)
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allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example)
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So, I have removed all of the uses of AST_LIST_HEAD_INIT and replaced them
with the equivalent static initializations.
- On passing, fix a memory leak in the unload_module() function of chan_agent.
The agents list mutex was never destroyed, and the elements in the agents
list were not freed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Thanks Dinesh for tracking the bug and fixing it)
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support in SIP.
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instead of being added to the compiler commands. This header file will be
installed and modules built outside of the main tree will be able to use the
same build options used to build the rest of Asterisk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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->indicate()
update iax2_indicate to pass control frame payload to the connected channel
add an API call for sending an indication with payload, and use it for control frames with payload
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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goes through the existing path.
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asterisk to answer an outbound call
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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support sending control frames with payload
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of warnings
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- remove some checks of the result of ast_mutex_lock, since it is not necessary
(this would be a good project to add to the janitor projects list).
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- we can't use ast_true here because non-empty strings would no longer be
evaluated as true
document the return values of pbx_checkcondition() in doxygen format
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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res_agi.c no longer depend on manager.h (issue #6397, casper)
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- add information about subscriptions and realtime dial plans in sip.conf.sample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- add missing locking of the functions list in the "show functions" CLI command
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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rename a flag to have the proper name
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- protect against multiple includes of ael_structs.h
- remove prototypes for unused or undeclared functions
pbx/ael/ael.y
- staticize functions as appropriate
- constify arguments
- remove useless extern
pbx/ael/ael.flex
- ifdef out unused functions
pbx/pbx_ael.c
- constify some variables and arguments
- ifdef out unused functions
- staticize functions as appropriate
update generated files accordingly
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no longer considered experimental :-)
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autoconf and menuselect tools for Asterisk!
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in app_dial to support the use of this mode.
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code is replicated in way too many places not to have a global
function for that.
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