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/trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29467 f38db490-d61c-443f-a65b-d21fe96a405b
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- with contributions from miconda, jcollie, and sb
- branch maintained by oej
Thanks everyone!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29094 f38db490-d61c-443f-a65b-d21fe96a405b
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messagecount2 to messagecount.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@28745 f38db490-d61c-443f-a65b-d21fe96a405b
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than for 1.2)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@28300 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@27559 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@26990 f38db490-d61c-443f-a65b-d21fe96a405b
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(Thanks Dinesh for tracking the bug and fixing it)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26954 f38db490-d61c-443f-a65b-d21fe96a405b
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support in SIP.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26920 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@26808 f38db490-d61c-443f-a65b-d21fe96a405b
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26451 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@26417 f38db490-d61c-443f-a65b-d21fe96a405b
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goes through the existing path.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26314 f38db490-d61c-443f-a65b-d21fe96a405b
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asterisk to answer an outbound call
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26103 f38db490-d61c-443f-a65b-d21fe96a405b
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support sending control frames with payload
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26093 f38db490-d61c-443f-a65b-d21fe96a405b
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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).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@25443 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@25411 f38db490-d61c-443f-a65b-d21fe96a405b
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res_agi.c no longer depend on manager.h (issue #6397, casper)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@25026 f38db490-d61c-443f-a65b-d21fe96a405b
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- add information about subscriptions and realtime dial plans in sip.conf.sample
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24342 f38db490-d61c-443f-a65b-d21fe96a405b
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- add missing locking of the functions list in the "show functions" CLI command
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23950 f38db490-d61c-443f-a65b-d21fe96a405b
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rename a flag to have the proper name
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23675 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22654 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22526 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22504 f38db490-d61c-443f-a65b-d21fe96a405b
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no longer considered experimental :-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22273 f38db490-d61c-443f-a65b-d21fe96a405b
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autoconf and menuselect tools for Asterisk!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22267 f38db490-d61c-443f-a65b-d21fe96a405b
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in app_dial to support the use of this mode.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22128 f38db490-d61c-443f-a65b-d21fe96a405b
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code is replicated in way too many places not to have a global
function for that.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22075 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@21933 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@21743 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@21338 f38db490-d61c-443f-a65b-d21fe96a405b
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update header file comments to reflect new usage of structure field
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@21207 f38db490-d61c-443f-a65b-d21fe96a405b
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(from tilghman's developer branch, res_odbc_rewrite)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@21181 f38db490-d61c-443f-a65b-d21fe96a405b
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be acquired
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: http://svn.digium.com/svn/asterisk/trunk@20962 f38db490-d61c-443f-a65b-d21fe96a405b
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- New get_sip_pvt_byid function (not really used correctly yet...)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20424 f38db490-d61c-443f-a65b-d21fe96a405b
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