and some patches (all disclaimed).
- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments
Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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
- 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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
on the frame counters. Document it in the header file.
- provide a single exit point for a function;
- mark XXX some unclear parts of the code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
equivalent (the reason is, when passing these strings through a
statically allocated buffer, we have no way to tell between NULL and ""
so we would be unable to preserve the difference, if any).
No code changes yet.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
wrappers around the basic 'say' functions, and redeclare these
wrappers as ordinary functions rather than function pointers.
This way, alternative implementations of the 'say' functions
will only have to implement the basic functions and not the
wrappers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
don't transcode via SLINEAR when the option is enabled but there is a direct path from the source to the destination
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Update lock.h with definitions of ast_channel_lock, ast_channel_unlock and ast_channel_trylock
- Convert some functions (but not all) in channel.c
- Fix some bugs in chan_sip.c
- Convert rest of chan_sip.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when you have channel locking issues.
(Part of the SIP transfer patch, where I had a *lot* of
channel locking problems)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As partly documented in loader.c and include/asterisk/module.h,
modules are now expected to return all of their methods and flags
into a structure 'mod_data', and are normally loaded with RTLD_NOW
| RTLD_LOCAL, so symbols are resolved immediately and conflicts
should be less likely. Only in a small number of cases (res_*,
typically) modules are loaded RTLD_GLOBAL, so they can export
symbols.
The core of the change is only the two files loader.c and
include/asterisk/module.h, all the rest is simply adaptation of the
existing modules to the new API, a rather mechanical (but believe
me, time and finger-consuming!) process whose detail you can figure
out by svn diff'ing any single module.
Expect some minor compilation issue after this change, please
report it on mantis http://bugs.digium.com/view.php?id=6968
so we collect all the feedback in one place.
I am just sorry that this change missed SVN version number 20000!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes the compilation on OS/X (the change exposed a wrong
assumption on mutex types on OS/X), but still leaves open the
bugs in initializing mutex on bsd systems, which you will see
reported as 'locking failures' on certain operations.
I need to investigate the issue further, but the best thing
i can do now is leave things as they have been for months.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These are momstly debugging tools for developers,
a bit documented in the header files (utils.h),
although more documentation is definitely necessary.
The performance impact is close to zero(*) so there is no
need to compile it conditionally.
(*) not completely true - thread destruction still needs
to search a list _but_ this can be easily optimized if we
end up with hundreds of active threads (in which case, though,
the problem is clearly elsewhere).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- misspelled ast_mutex_logger() instead of __ast_mutex_logger()
- misplaced #define ast_mutex_init(pmutex)
- wrong arguments to __ast_mutex_logger() in one instance.
Clearly this code is too spaghetti!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_register_atexit()/ ast_unregister_atexit() into asterisk.h
These are general functions, not restricted to modules, so move
them in a more proper place.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
new-style modules using static symbols.
Everything will still work as before, but new-style modules
can now be defined by putting a '#define STATIC_MODULE' somewhere
before including module.h, then declaring STATIC_MODULE the
various methods (load, unload, key...) that the module is
supposed to supply, and adding a 'STD_MOD(MOD_1, reload_fn, NULL, NULL)'
macro call at the end.
A module compiled in this way will be loaded RTLD_NOW|RTLD_LOCAL
so symbol pollution is reduced, and symbols are resolved immediately.
Removing just the '#define STATIC_MODULE' will restore the old
behaviour.
In order for a module to be loaded RTLD_NOW|RTLD_LOCAL, it must not
export any symbol[1], and all the modules it depends on (e.g. res_*)
must be loaded already.
[1] Mechanisms are in place, and will be enabled later, to still
allow such modules to 'export' symbols and resolving the dependencies
irrespective of the load order.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
collecting common functions in a single place and removing
them from the individual handlers.
The full description is on mantis,
http://bugs.digium.com/view.php?id=6375
and only the ogg_vorbis handler needs to be converted to
the new structure.
As a result of this change, format_au.c and format_pcm_alaw.c
should go away (in a separate commit) as their functionality
(trivial) has been merged in another file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
in pbx_exec is always 1 so it can be removed.
This change also takes away ast_exec_extension(), and lets all
switch functions (exists, canmatch, exec, matchmore) all use the same
prototype, which makes the code a bit cleaner.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_walk_indications(), to walk through the list of indications.
The new method returns an unlocked record, which is no different from the
behaviour of other existing methods in indications.c
(i.e. they all need to be fixed, with refcounts or some similar
method).
Note that ast_walk_indications() uses the pointer argument only as a
search key, so its implementation is completely safe.
In turn, this change allows the removal of AST_MUTEX_DEFINE_EXPORTED.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
#define DEFAULT_SAMPLE_RATE 8000
#define DEFAULT_SAMPLES_PER_MS ((DEFAULT_SAMPLE_RATE)/1000)
to the main header, and remove equivalent ones from plc.[ch]
This will simplify the cleanup of the codec/ and formats/ files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
which is the basis for several simplifications and fixes
to the CLI interfaces.
The core is in cli.c, some documentation on a new function
to help command completion is in cli.h, and one line of
glue code in the other two files.
Next step is to bring in the patches described in #6066 and
other simplifications.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and updated to today's version.
The core of the patch is only two files, loader.c
and include/asterisk/module.h, with the other files
touched only to adapt non-standard usages of the
reference counts and localuser lists.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
writing of common constructs like
chan = !ast_strlen_zero(cdr->channel) ? cdr->channel : "<unknown>";
(or the if/then/else form) into
chan = S_OR(cdr->channel, "<unknown>");
The name can be changed if we find a better (and not too long) one;
currently, it is S as String, OR as it mimics the behaviour of
the || operator, but applied to strings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
can override them.
On passing, fix a potential problem in the top level Makefile:
if a static library is not referenced by any of the core objects,
it is not linked in the main program, and will not be available
to modules, which leads to failure at runtime when the modules
are loaded.
This is the case of stdtime/localtime.o, which supplies some core
symbolx, but is only linked in as a library. Fix the problem by
linking in the object.
NOTE: this is intended as a temporary aid to replace the
existing say.c with a newer implementation. Once the
task is completed, we may decide whether or not the ast_say*()
functions should be pluggable or not and possibly revert
part of this change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In this specific case the problem triggered on app_amd.c,
but it keeps coming out from time to time so it is better
to fix it in a more central place.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
it doesn't report that all allocations are coming from utils.h. Also, add some
more information to the error message astmm reports when a memory allocation
failure occurs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
representation into pbx.c so that every file that includes pbx.h does not
unnecessarily get a copy of it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
was improperly managed when doing removals or insertions.
also solved issues with app_voicemail init. and reload
solves bug #6557
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Rename and export ast_complete_channels for use by cli completion functions
that want to complete from the list of active channels
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10287 65c4cc65-6c06-0410-ace0-fbb531ad65f3