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r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines
Merged revisions 304729 via svnmerge from
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r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines
Make sure that we unref the correct object when ejecting the most recent caller.
Currently, when we kick the last user to enter, we decrement our own reference
count which results in a crash when we kick another user or when we exit the
conference ourselves.
This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in
1.6.2.
(closes issue #18225)
Reported by: kenji
Patches:
issue18225.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines
Merged revisions 304726 via svnmerge from
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r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines
Fix user reference leak in MeetMe.
We were unlinking the user from the conferences user container, but not
decrementing the reference count of the user as well, resulting in a leak.
(closes issue #18444)
Reported by: junky
Tested by: seanbright
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r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines
Merged revisions 304659,304682 via svnmerge from
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r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) | 5 lines
Don't leak references if we can't create a pseudo channel for mixing in MeetMe.
If there was a problem allocating a pseudo channel when building our meetme, we
weren't destroying our user container or destroying the mutexes that we created.
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r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) | 2 lines
Revert part of the previous commit that snuck in.
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Followme answers an incoming call if it hasn't already been answered and starts
MOH. Some poorly designed autodialers see the answer and start playing their
message to the hold music. The 'N' option has been added to indicate ringing and
not answer until the call is accepted.
(closes issue #18479)
Reported by: ianc
Patches:
trunk_followme.diff uploaded by ianc (license 998)
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r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines
Merged revisions 303677 via svnmerge from
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r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines
Merged revisions 303676 via svnmerge from
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r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines
Fix voicemail sequencing for file based storage.
A previous change was made to account for when the number of voicemail messages
exceeds the max limit to be handled properly, but it caused gaps in the messages
to not be properly handled. This has now been resolved.
In later non 1.4 branches, it appears that resequencing wasn't even occurring
due from what appears and accidental code removal.
(closes issue #18498)
Reported by: JJCinAZ
Patches:
bug18498v2.patch uploaded by jpeeler (license 325)
(closes issue #18486)
Reported by: bluefox
Patches:
bug18486.patch uploaded by jpeeler (license 325)
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r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
Merged revisions 303548 via svnmerge from
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r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
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r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
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r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines
Merged revisions 303008 via svnmerge from
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r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines
Merged revisions 303007 via svnmerge from
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r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines
Add new queue strategy to preserve behavior for when queue members moved to ao2.
Add queue strategy called "rrordered" to mimic old behavior from when queue
members were stored in a linked list.
ABE-2707
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r302921 | russell | 2011-01-20 10:12:15 -0600 (Thu, 20 Jan 2011) | 9 lines
Merged revisions 302920 via svnmerge from
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r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) | 2 lines
Resolve a compiler warning.
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r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines
Merged revisions 302917 via svnmerge from
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r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines
Option L() is milliseconds, not seconds.
> Change the verbose output of option L() to say milliseconds and not seconds
> as the value is in milliseconds.
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> (closes issue #18264)
> Reported by: jacco
> Patches:
> app_dial_patch.txt uploaded by lmadsen (license 10)
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r302834 | seanbright | 2011-01-19 18:49:00 -0500 (Wed, 19 Jan 2011) | 14 lines
Merged revisions 302833 via svnmerge from
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r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) | 7 lines
Support greetingsfolder as documented in voicemail.conf.sample.
(closes issue #17870)
Reported by: edhorton
Patches:
__20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10)
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r301177 | pabelanger | 2011-01-08 17:00:12 -0500 (Sat, 08 Jan 2011) | 14 lines
Merged revisions 301176 via svnmerge from
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r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) | 7 lines
Indicate log level argument for Log() is not optional
(closes issue #18586)
Reported by: kshumard
Patches:
app_verbose.c.patch uploaded by kshumard (license 92)
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r301090 | qwell | 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) | 15 lines
Merged revisions 301089 via svnmerge from
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r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines
Initialize useropts/adminopts in case there is no column in the realtime DB.
(closes issue #18182)
Reported by: dimas
Patches:
v1-18182.patch uploaded by dimas (license 88)
Tested by: dimas
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r301047 | jpeeler | 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) | 15 lines
Merged revisions 301046 via svnmerge from
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r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) | 8 lines
Fix regression causing forwarding voicemails to not work with file storage.
I had actually already fixed this in 295200 in 1.4 and thought it wasn't
missing in the other branches for some reason.
(closes issue #18358)
Reported by: cabal95
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r300955 | jpeeler | 2011-01-07 11:24:14 -0600 (Fri, 07 Jan 2011) | 21 lines
Merged revisions 300951 via svnmerge from
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r300951 | jpeeler | 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) | 14 lines
Merged revisions 300918 via svnmerge from
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r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines
Ensure good bye prompt in voicemail is played at the correct time.
Specifically in the case of timing out but not leaving voicemail nothing
should be heard. And when leaving voicemail it should be heard.
ABE-2647
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playing_silence was not initialized with the struct
was initialized, it was being set after the fact
which caused problems if something that relied on
playing_silence being set was called too quickly
(closes issue #18430)
Reported by: stevebrandli
Patches:
externalivr.patch uploaded by thedavidfactor (license 903)
Tested by: thedavidfactor, stevebrandli
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r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) | 4 lines
Quote arguments, just in case there's a space in a pathname.
(Diagnosed by pabelanger on #asterisk-dev, fixed by me.)
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r299865 | pabelanger | 2010-12-28 13:53:37 -0500 (Tue, 28 Dec 2010) | 9 lines
Merged revisions 299864 via svnmerge from
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r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec 2010) | 2 lines
Documentation typo
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r298685 | jpeeler | 2010-12-16 17:31:50 -0600 (Thu, 16 Dec 2010) | 16 lines
Merged revisions 298684 via svnmerge from
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r298684 | jpeeler | 2010-12-16 17:30:59 -0600 (Thu, 16 Dec 2010) | 9 lines
Merged revisions 298683 via svnmerge from
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r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines
After recording only silence for a voicemail prepending, restore backup files.
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r298598 | jpeeler | 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) | 21 lines
Merged revisions 298597 via svnmerge from
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r298597 | jpeeler | 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) | 14 lines
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r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines
Fix improper hangup when doing an attended transfer to queue.
Had to indicate ringing in wait_for_answer so the attended transfer code would
not try and hang up the local channel it created, which would kill the call.
ABE-2624
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r297733 | tilghman | 2010-12-06 18:29:26 -0600 (Mon, 06 Dec 2010) | 22 lines
Merged revisions 297713 via svnmerge from
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r297713 | tilghman | 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) | 15 lines
Merged revisions 297689 via svnmerge from
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r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines
Don't create a Local channel if the target extension does not exist.
(closes issue #18126)
Reported by: junky
Patches:
followme.diff uploaded by junky (license 177)
(partially restructured by me to avoid a possible memory leak)
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r297245 | russell | 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) | 20 lines
Merged revisions 297229 via svnmerge from
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r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines
Merged revisions 297228 via svnmerge from
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r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines
Add "DAHDI" to a couple of app_meetme error messages.
This is in response to some questions on IRC. To the user, there was nothing
that made it obvious that this error had anything to do with DAHDI not being
loaded.
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r296870 | jpeeler | 2010-11-30 18:28:16 -0600 (Tue, 30 Nov 2010) | 18 lines
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r296869 | jpeeler | 2010-11-30 18:24:58 -0600 (Tue, 30 Nov 2010) | 11 lines
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r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines
Properly restore backup information file when hanging up during message prepending.
ABE-2654
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r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) | 2 lines
DOC: Conference number can be omitted; if omitted, all users in a meetme are listed.
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r296467 | tilghman | 2010-11-27 04:40:22 -0600 (Sat, 27 Nov 2010) | 12 lines
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r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) | 5 lines
18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision).
(closes issue #18369)
Reported by: tnakonz
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Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.
Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
meetme_final_patch_v.diff uploaded by parisioa (license 1153)
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r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines
Merged revisions 296001 via svnmerge from
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r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
Merged revisions 296000 via svnmerge from
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r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
Handle failures building translation paths more effectively.
The problem scenario occurred on a heavily loaded system that was using the
codec_dahdi module and exceeded the hardware transcoding capacity. The failure
mode at that point was not good. The report came in to us as an Asterisk
lock-up. The "core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is in this
state, the CPU was maxed out. The CPU was being consumed by the Asterisk
logger spewing messages on every audio frame for calls set up after transcoder
capacity was reached.
The purpose of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't translate, then the
call just needs to be dropped, as it's not going to work. These are the
changes:
1) In set_format() of channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if ast_translator_build_path() failed and
returned NULL. It now pays attention to that case and returns a result
reflecting failure. With this change in place, the bridging code will
immediately detect a failure and end the bridge instead of proceeding to try to
bridge frames that can't be translated and making channel drivers freak out by
sending them frames in a format they weren't expecting.
2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
ignored. It is now reflected in the return value of the function. This didn't
turn out to have any affect on the bug, but seemed like a good change to leave
in.
3) In app_dial(), when only sending a call to a single endpoint, it will
attempt to do some bridging of its own of early audio. It uses
make_compatible() when it's going to do this. However, it ignored failure from
make compatible. So, even with the fix from #1, if there was early audio going
through app_dial, there would still be a period of invalid frames passing
through. After detecting failure here, Dial() exits.
ABE-2658
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r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines
Merged revisions 295843 via svnmerge from
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r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
Merged revisions 295790 via svnmerge from
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r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
To recreate the problem:
1) Party A calls Party B
2) Invoke CLI "channel redirect" command to redirect channel call leg
associated with A.
3) All associated channels are hung up.
Note that if the CLI command were done on the channel call leg associated
with B it works.
This regression was a result of the fix for issue #16946
(https://reviewboard.asterisk.org/r/740/).
The regression affects all features that use an async goto to execute the
dialplan because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection.
The struct ast_channel._softhangup code is a mess. The variable is used
for several purposes that do not necessarily result in the call being hung
up. I have added doxygen comments to describe how the various _softhangup
bits are used. I have corrected all the places where the variable was
tested in a non-bit oriented manner.
The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak
hangup request so the soft hangup requests that do not normally result in
a hangup do not hangup.
JIRA SWP-2470
JIRA SWP-2489
(closes issue #18171)
Reported by: SantaFox
(closes issue #18185)
Reported by: kwemheuer
(closes issue #18211)
Reported by: zahir_koradia
(closes issue #18230)
Reported by: vmarrone
(closes issue #18299)
Reported by: mbrevda
(closes issue #18322)
Reported by: nerbos
Review: https://reviewboard.asterisk.org/r/1013/
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r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) | 8 lines
Patch for deadlock from ordering issue between channel/queue locks in app_queue
(set_queue_variables).
(closes issue #18031)
Reported by: rain
Review: https://reviewboard.asterisk.org/r/1018/
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r294911 | jpeeler | 2010-11-12 15:14:43 -0600 (Fri, 12 Nov 2010) | 11 lines
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r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) | 4 lines
Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent.
Reported by alecdavis in asterisk-dev.
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r294905 | jpeeler | 2010-11-12 14:52:06 -0600 (Fri, 12 Nov 2010) | 30 lines
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r294904 | jpeeler | 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) | 23 lines
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r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines
Fix regression causing abort in voicemail after opening a mailbox with no mesgs.
In order to be more safe, some error handling code was changed to respect more
error conditions including the potential memory allocation failure for deleted
and heard message tracking introduced in 293004. However, last_message_index
returns -1 for zero messages (perhaps as expected) and was triggering the
stricter error checking. Because last_message_index is only called directly
in one place, just return 0 from open_mailbox (for file based storage) when no
messages are detected unless a real error has occurred.
(closes issue #18240)
Reported by: leobrown
Patches:
bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
Tested by: pabelanger
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r293119 | jpeeler | 2010-10-26 13:49:08 -0500 (Tue, 26 Oct 2010) | 43 lines
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r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines
Merged revisions 293004 via svnmerge from
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r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines
Fix inprocess_container in voicemail to correctly restrict max messages.
The comparison function logic was off, so the number of sessions for a given
mailbox were not being incremented properly. This problem caused the maximum
number of messages per folder to not be respected when simultaneously leaving
multiple voicemails just below the threshold.
These problems should be fixed by the above, but just in case:
Fixed resequence_mailbox to rely on the actual number of detected number of
files in a directory rather than just assuming only 10 messages more than the
maximum had been left. Also if more messages than the maximum are deleted they
are actually removed now.
The second purpose of this commit should have been separated out probably, but
is related to the above. Again, if the number of messages in a given voicemail
folder exceeds the maximum set limit make sure to allocate enough space for the
deleted and heard index tracking array.
A few random fixes:
There was a forgotten decrement of the inprocess count in imap_store_file.
When using IMAP storage, do not look in the directory where file based storage
messages may still reside and influence the message count.
Ensure to use only the first format in sendmail.
ABE-2516
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r292436 | pabelanger | 2010-10-20 20:21:59 -0400 (Wed, 20 Oct 2010) | 8 lines
Application not properly unregister in voicemail
(closes issue #18128)
Reported by: junky
Patches:
vm_unregister.diff uploaded by junky (license 177)
Tested by: pabelanger, lmadsen
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r292413 | pabelanger | 2010-10-20 20:07:17 -0400 (Wed, 20 Oct 2010) | 24 lines
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r292412 | pabelanger | 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) | 17 lines
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r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines
Record priv-recordintro as sln, not gsm
This removes the gsm->sln step when transcoding
priv-recordintro.
(closes issue #18176)
Reported by: pabelanger
Patches:
chan_sip.diff uploaded by pabelanger (license 224)
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r292227 | jpeeler | 2010-10-18 16:55:46 -0500 (Mon, 18 Oct 2010) | 25 lines
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r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines
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r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines
Fix improper operator key acceptance and clean up temp recording files.
This is a fix for when pressing the operator key after recording an unavailable,
busy, name, or temporary message in mailbox options. The operator key should not
be accepted here, but should be allowed during the message recording. If the
operator key is pressed during ensure the file is saved or deleted as
apporopriate. Also, ensure removal of temporary recorded files after an early
hang up or when message acceptance confirmation times out.
ABE-2518
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Add the extension and context of the calling channel to the log output if a macro could not be found.
(closes issue #18112)
Reported by: prado
Patches:
app_macro-info.diff uploaded by prado (license 510)
Tested by: schmidts
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r290614 | rmudgett | 2010-10-06 13:50:37 -0500 (Wed, 06 Oct 2010) | 12 lines
Merged revision 290613 from
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r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, 06 Oct 2010) | 5 lines
Eliminate a redundant test for AST_CONTROL_REDIRECTING.
Eliminate redundant test for AST_CONTROL_REDIRECTING that prevents running
the redirecting interception macro if it is defined.
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r290376 | dvossel | 2010-10-05 14:56:29 -0500 (Tue, 05 Oct 2010) | 16 lines
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r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) | 10 lines
Fixes PickupChan() not working with full channel name.
(closes issue #18011)
Reported by: schern
Patches:
app_directed_pickup.c.2.patch uploaded by schern (license 995)
app_directed_pickup.c.trunk.patch uploaded by schern (license 995)
Tested by: schern, dvossel
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r289875 | tilghman | 2010-10-01 23:46:43 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289874 via svnmerge from
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r289874 | tilghman | 2010-10-01 23:45:49 -0500 (Fri, 01 Oct 2010) | 15 lines
Merged revisions 289873 via svnmerge from
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r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines
When forwarding a message, a prepend means that the filesystem will always have a better copy.
(closes issue #17803)
Reported by: dpetersen
Patches:
20100923__issue17803.diff.txt uploaded by tilghman (license 14)
Tested by: dpetersen
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r289426 | russell | 2010-09-30 10:39:45 -0500 (Thu, 30 Sep 2010) | 22 lines
Merged revisions 289425 via svnmerge from
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r289425 | russell | 2010-09-30 10:37:29 -0500 (Thu, 30 Sep 2010) | 15 lines
Merged revisions 289424 via svnmerge from
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r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines
Fix a crash in app_sms.
Since the data being passed to the generator callback is on the stack of the
SMS() application, we must ensure that the generator is stopped before the
application exits.
ABE-2587
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r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) | 4 lines
Solaris compatibility fixes
Review: https://reviewboard.asterisk.org/r/942/
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r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines
Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code.
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r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines
Simplify locking code for REDIRECTING interception macro when forwarding a call.
Simplified the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and app_queue.c:wait_for_answer()
for launching the REDIRECTING interception macro when a call is forwarded.
Reduced the lock time of the 'o->chan' and 'in' channels.
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r287760 | bbryant | 2010-09-20 20:00:23 -0400 (Mon, 20 Sep 2010) | 30 lines
Merged revisions 287759 via svnmerge from
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r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines
Merged revisions 287758 via svnmerge from
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r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines
Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag.
When using the 'a' MeetMe flag and having a user and admin pin setup for your
conference, using the user pin would gain you admin priviledges. Also, when no
user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the
user tried to enter a conference then they were still prompted for a pin and
forced to hit #.
(closes issue #17908)
Reported by: kuj
Patches:
pins_2.patch uploaded by kuj (license 1111)
Tested by: kuj
Review: [full review board URL with trailing slash]
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r287388 | tilghman | 2010-09-17 16:08:54 -0500 (Fri, 17 Sep 2010) | 21 lines
Merged revisions 287387 via svnmerge from
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r287387 | tilghman | 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) | 14 lines
Merged revisions 287386 via svnmerge from
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r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines
Blank columns should get set on reload, not ignored.
(closes issue #16893)
Reported by: haakon
Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
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r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) | 4 lines
Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf.
Review: https://reviewboard.asterisk.org/r/922/
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r287015 | jpeeler | 2010-09-15 15:32:52 -0500 (Wed, 15 Sep 2010) | 21 lines
Merged revisions 286998 via svnmerge from
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r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
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r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
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r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines
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r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines
Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart.
(closes issue #17408)
Reported by: sysreq
Patches:
asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009)
Tested by: sysreq
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r285197 | bbryant | 2010-09-07 13:54:21 -0400 (Tue, 07 Sep 2010) | 24 lines
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r285196 | bbryant | 2010-09-07 13:49:07 -0400 (Tue, 07 Sep 2010) | 17 lines
Merged revisions 285194 via svnmerge from
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r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines
Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages.
(closes issue #15726)
Reported by: 298
Patches:
M15726.diff uploaded by junky (license 177)
Tested by: junky
Review: [full review board URL with trailing slash]
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r284921 | twilson | 2010-09-03 11:28:18 -0500 (Fri, 03 Sep 2010) | 19 lines
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r284897 | twilson | 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) | 12 lines
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r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines
Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant file. The existing
code treated missing files as though they existed.
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r284632 | tilghman | 2010-09-02 00:31:02 -0500 (Thu, 02 Sep 2010) | 14 lines
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r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines
Don't reset queue stats on a module reload.
(closes issue #17535)
Reported by: raarts
Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
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r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
When optional_api is non-optional, force dependent modules to be loaded.
(closes issue #17707)
Reported by: ira
Patches:
20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/876/
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