being #included twice. This was due to the fact that #exec created a temporary file which
was then #included. The name of the temporary file was the name of the #exec'd file, with
the Unix timestamp and thread ID concatenated. The issue was that if multiple #exec statements
of the same file were reached in the same second, then the result was that the temporary files
would have duplicate names. To resolve this, the temporary file now has microsecond resolution
for the timestamp portion.
(closes issue #12574)
Reported by: jmls
Patches:
12574.patch uploaded by putnopvut (license 60)
Tested by: jmls, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A task wraps a callback function pointer and a data pointer and is managed internal to the taskprocessor subsystem. The callback function is responsible for releasing task data.
Taskprocessor API
* ast_taskprocessor_get(..) - returns a reference to a taskprocessor
* ast_taskprocessor_unreference(..) - releases reference to a taskprocessor
* ast_taskprocessor_push(..) - push a task into a taskprocessor queue
Check doxygen for more details
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
party to be spoken instead of the channel name or number.
This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.
This change comes as a suggestion from Switchvox, which already has this feature. AST-23
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add some additional features to the core park_call_full() function, and expose
them as options to the Park() application. The functionality being added is the
ability to specify a custom return extension/context/priority, a custom timeout,
and a couple of options. The options are to play ringing instead of MOH to the
parked caller, and to randomize parking spot selection.
(code inspired by the patch in AST-17, code from switchvox)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) | 6 lines
Improve some broken cookie parsing code. Previously, manager login over HTTP
would only work if the mansession_id cookie was first. Now, the code builds
a list of all of the cookies in the Cookie header. This fixes a problem
observed by users of the Asterisk GUI.
(closes AST-20)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) | 5 lines
Store the manager session ID explicitly as 4 byte ID instead of a ulong. The
mansession_id cookie is coded to be limited to 8 characters of hex, and this
could break logins from 64-bit machines in some cases.
(inspired by AST-20)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 lines
Instead of stopping dialplan execution when SayNumber attempts to say a large number that it can not print out a message informing the user and continue on.
(closes issue #12502)
Reported by: bcnit
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: triccyx
I had a bit a problem reproducing this in my setup (trying not to disturb my other stuff)
but finally, I got it. The problem appears to be that the extension is being added in
replace mode, which kinda assumes that the pattern trie has been formed, when in fact,
in this case, it was not. The checks being done are not nec. when the tree is not yet
formed, as changes like this will be summarized when the trie is formed in the future.
I tested the fix, and the crash no longer happens. Feel free to open the bug again if
this fix doesn't cure the problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I gathered the code that filled the string, and put it in a different func which
I cryptically call "append_lock_information()".
Now, both log_show_lock(), and handle_show_locks() both call this code to do
the work. Tested, seems to work fine.
Also, log_show_lock was modified to use the ast_str stuff, along with checking
for successful ast_str creation, and freeing the ast_str obj when finished.
A break was inserted to terminate the search for the lock; we should never
see it twice.
An example usage in chan_sip.c was created as a comment, for instructional
purposes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: atis
Tested by: murf
This upgrade adds the ~~ (concatenation) string operator to expr2.
While not needed in normal runtime pbx operation, it is needed when
raw exprs are being syntax checked. This plays into future syntax-
unification plans. By permission of atis, this addition in trunk
and the reason of why things are as they are will suffice to close
this bug.
I also added a short note about the previous addition of "sip show sched"
to the CLI in CHANGES, which I discovered I forgot in a previous commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines
It was possible for a reference to a frame which was part of a freed DSP to still be
referenced, leading to memory corruption and eventual crashes. This code change ensures
that the dsp is freed when we are finished with the frame. This change is very similar
to a change Russell made with translators back a month or so ago.
(closes issue #11999)
Reported by: destiny6628
Patches:
11999.patch uploaded by putnopvut (license 60)
Tested by: destiny6628, victoryure
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a. fix a self-found problem with SPAWN-ing an extension,
where matches were not being found
b. correct some wording in a comment
c. Add some debug for future debugging.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr 2008) | 11 lines
Increase the retry count when attempting to show channels. This apparently
cleared an issue someone was seeing when attempting to show channels when
the load was high.
(closes issue #11667)
Reported by: falves11
Patches:
11677.txt uploaded by russell (license 2)
Tested by: falves11
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr 2008) | 5 lines
Save a local copy of the generate callback prior to unlocking the channel in
case the generate callback goes NULL on us after the channel is unlocked. Thanks
to Russell for pointing this need out to me.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines
Only try to prefix language if we are not using an absolute path (suffix it otherwise).
en/var/lib/asterisk/sounds/blah.gsm is a very silly path.
(closes issue #12379)
Reported by: kuj
Patches:
12379-absolutepath.diff uploaded by qwell (license 4)
Tested by: kuj, qwell
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
were handled. In 1.4, if the absolute timeout were reached on a call, no matter what
the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T'
extension or hangup otherwise. The rearrangement of this function in trunk made this check
only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned
1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite
loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will
behave as they did in 1.4
(closes issue #11550)
Reported by: pj
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines
If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute.
(closes issue #12296)
Reported by: jvandal
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines
This fix prevents a deadlock that was experienced in chan_local. There was
deadlock prevention in place in chan_local, but it would not work in a specific
case because the channel was recursively locked. By unlocking the channel prior
to calling the generator's generate callback in ast_read_generator_actions(), we
prevent the recursive locking, and therefore the deadlock.
(closes issue #12307)
Reported by: callguy
Patches:
12307.patch uploaded by putnopvut (license 60)
Tested by: callguy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113066 65c4cc65-6c06-0410-ace0-fbb531ad65f3