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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 lines
(closes issue #5866)
Reported by: tyler
Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues.
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@76654 f38db490-d61c-443f-a65b-d21fe96a405b
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using GCC 4.2.
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instead of loginstart
since loginstart is used after logoff. Thanks to makoto for pointing this out and suggesting the fix.
(closes issue #10178, reported and patched by makoto, with modification by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 Jul 2007) | 5 lines
The cli command "agent logoff Agent/x soft" did not work...at all. Now it does.
(closes issue #10178, reported and patched by makoto, with slight modification for 1.4 and trunk by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4 lines
Fix an issue with wrapuptime not working when using AgentLogin.
Issue 10169, patch by makoto, with a minor mod by me to not re-break issue 9618
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 Jul 2007) | 5 lines
Fixed a bug wherein agents get stuck busy. (issue 9618, reported by jiddings, patched by moi)
closes issue #9618
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bugs while half asleep. (issue #10046 reported by dimas)
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unhold frame on it. (issue #10046 reported by dimas)
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queues.
Also, remove an unneeded leading slash for the astdb family.
(issue #9911, patch by atis)
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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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refinement, but this won't have as many folks bothered.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60989 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines
Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines
Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155)
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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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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...)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51311 f38db490-d61c-443f-a65b-d21fe96a405b
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cases
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solve a fairly small problem... such is life.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47303 f38db490-d61c-443f-a65b-d21fe96a405b
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consistent with "category verb arguments"
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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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future this should be expanded to the channel driver (just like all the other channel drivers) (issue #7975 reported by andrewt)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43199 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@41633 f38db490-d61c-443f-a65b-d21fe96a405b
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41507 f38db490-d61c-443f-a65b-d21fe96a405b
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- 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: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37988 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37212 | bweschke | 2006-07-06 15:38:45 -0500 (Thu, 06 Jul 2006) | 3 lines
Don't do wierd things on a callback agent that has attempted logoff while still on the phone.
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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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- please check
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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: http://svn.digium.com/svn/asterisk/trunk@35365 f38db490-d61c-443f-a65b-d21fe96a405b
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but was lost at some point in /trunk. #7374 (jstorm)
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- small reformatting changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34093 f38db490-d61c-443f-a65b-d21fe96a405b
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again :-)
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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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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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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r25563 | bweschke | 2006-05-08 10:12:20 -0400 (Mon, 08 May 2006) | 3 lines
Don't show agents as available when they are in wrap-up time. #6726 (ZX81)
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cli for agents, show agents online , patch from bug
7096
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@25441 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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by pressing '*'. This is still on by default. (issue #6897)
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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!
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description() and key() return values
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
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rand() to threadsafe ast_random()
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strlen(word), localize variables and normalize the test
for finding the candidate string.
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to S_OR functions. from bug note 6805 with minor
modifications.
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