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
Kevin P. Fleming
Matthew Fredrickson
Russell Bryant
Luigi Rizzo
Mark Spencer
Olle Johansson
BJ Weschke
Jim Dixon
Joshua Colp