https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines
Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, 16 Feb 2007) | 10 lines
Merged revisions 54999 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42133 | bweschke | 2006-09-06 14:16:41 -0400 (Wed, 06 Sep 2006) | 6 lines
Look ma! No more deadlocks! <sic>
As posted from #7458 and others similar to it in Mantis:
p->app_lock was a mutex really designed for use with agents not in callback mode. That being the case, I've tried to code it so that when callback mode is used, the app_lock mutex will not be locked/unlocked at all. Please let me know how you make out - and if you continue to deadlock now, please reproduce the deadlock logging information and post to Mantis.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it
had a non static function when it should.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r36725 | russell | 2006-07-03 00:19:09 -0400 (Mon, 03 Jul 2006) | 4 lines
use ast_set_callerid to be more consistent and to make sure that the
"callerid" option in the conf files is always handled the same way and sets ANI
(issue #7285, gkloepfer)
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More ideas for developing better video support in Asterisk?
Join the asterisk-video mailing list to help out in the
Asterisk Video Task Force!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35365 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
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
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
const-ify some more APIs
remove 'type' field from ast_channel, in favor of the one in the channel's tech structure
allow string field module users to specify the 'chunk size' for pool allocations
update chan_alsa to be compatible with recent const-ification patches
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7448 | kpfleming | 2005-12-12 22:25:14 -0600 (Mon, 12 Dec 2005) | 2 lines
use the stream's current point when pausing/unpausing, instead of elapsed time (which doesn't work when the stream has been skipped forward or backward) (issue #5897)
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r7449 | kpfleming | 2005-12-12 22:43:38 -0600 (Mon, 12 Dec 2005) | 2 lines
only report AGENT_IDLE for callback mode agents when they are actually idle (issue #5902)
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r7451 | kpfleming | 2005-12-12 23:14:27 -0600 (Mon, 12 Dec 2005) | 2 lines
ensure that hangups while incoming calls are in early state are handled properly (issue #5919)
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r7453 | kpfleming | 2005-12-12 23:53:00 -0600 (Mon, 12 Dec 2005) | 2 lines
restore ability of caller to hangup calls that are still ringing (issue #5839)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This should prevent us from unintentionally changing variable
values when they're returned from pbx_builtin_getvar_helper.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
specified extension do not conflict. This also makes the application not
assume that the extension is numeric. While we're at it, include some code
cleanup - strcpy to ast_copy_string, and cleanups to LOCAL_USER handling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Steve Murphy
Joshua Colp
Kevin P. Fleming
Luigi Rizzo
Tilghman Lesher
Matt O'Gorman
Mark Spencer
BJ Weschke
Olle Johansson