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my concerns are listed in issue #6305
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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change their resolved IP (aka dynamic dns setups) (issue #6305 reported and fixed by ivanfm)
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Thanks to everyone involved in working with this!
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31738 | bweschke | 2006-06-03 10:48:13 -0400 (Sat, 03 Jun 2006) | 3 lines
Fix doxygen comment about AST_LIST_HEAD_INIT_NOLOCK
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31520 | kpfleming | 2006-06-01 15:27:50 -0500 (Thu, 01 Jun 2006) | 2 lines
handle Zap transfers behind chan_agent properly so the agent doesn't get stuck waiting for the call to hang up
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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as well as being more efficient, patch from jcollie's
base64 branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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and fixes bug with pruneregister
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
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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
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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.
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- with contributions from miconda, jcollie, and sb
- branch maintained by oej
Thanks everyone!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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res_agi.c no longer depend on manager.h (issue #6397, casper)
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