Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear
their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber.
This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape.
If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour.
Reported by: alecdavis
Tested by: alecdavis
Patch
vm_a_extension.diff2.txt uploaded by alecdavis (license 585)
Review: https://reviewboard.asterisk.org/r/489/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We nicely detect the right flags on each system for building Asterisk with
pthreads, then ignore it for every other build option that requires us to
build with pthreads. This caused some items to return a false negative.
Also cleanup some minor naming issues that caused "library library" redundancy
in the output.
(closes issue #17303)
Reported by: stuarth
Patches:
20100507__issue17303.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The pri_dchannel thread currently violates locking order by locking the private
and then attempting to queue a frame, which needs to lock the channel. Queueing
a frame is unneccesary though and is actually a regression since sig_pri.
All the places that currently use ast_softhangup_nolock now will just set the
softhangup value directly as before.
(closes issue #17216)
Reported by: lmsteffan
Patches:
bug17216.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made more places use pri_queue_control() instead of pri_queue_frame()
and a local frame variable.
* Made pri_queue_frame() use sig_pri_lock_owner(). pri_queue_frame() no
longer releases the libpri access lock unless it is required.
* Made the pri_queue_frame() and pri_queue_control() parameter list
similar to sig_pri_lock_owner().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines
Only allow the operator key to be accepted after leaving a voicemail.
Or rather disallow the operator key from being accepted when not offered,
such as after finishing a recording from within the mailbox options menu.
ABE-2121
SWP-1267
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The example given within the related issue showed 120 lines, which was mostly
a result of the body being XML.
(closes issue #17179)
Reported by: khw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I modified the original patch for trunk to use the unit test API.
(issue #17277)
Reported by: cappucinoking
Patches:
test_heap.diff uploaded by cappucinoking (license 1036)
Tested by: cappucinoking, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This bug surfaced in 1.6.2 and does not affect code in any other released
version of Asterisk. It manifested itself as SIP qualify not happening when
it should, causing peers to go unreachable. This was debugged down to scheduler
entries sometimes not getting executed when they were supposed to, which was in
turn caused by an error in the heap code.
The problem only sometimes occurs, and it is due to the logic for removing an entry
in the heap from an arbitrary location (not just popping off the top). The scheduler
performs this operation frequently when entries are removed before they run (when
ast_sched_del() is used).
In a normal pop off of the top of the heap, a node is taken off the bottom,
placed at the top, and then bubbled down until the max heap property is restored
(see max_heapify()). This same logic was used for removing an arbitrary node
from the middle of the heap. Unfortunately, that logic is full of fail. This
patch fixes that by fully restoring the max heap property when a node is thrown
into the middle of the heap. Instead of just pushing it down as appropriate, it
first pushes it up as high as it will go, and _then_ pushes it down.
Lastly, fix a minor problem in ast_heap_verify(), which is only used for
debugging. If a parent and child node have the same value, that is not an
error. The only error is if a parent's value is less than its children.
A huge thanks goes out to cappucinoking for debugging this down to the scheduler,
and then producing an ast_heap test case that demonstrated the breakage. That
made it very easy for me to focus on the heap logic and produce a fix. Open source
projects are awesome.
(closes issue #16936)
Reported by: ib2
Tested by: cappucinoking, crjw
(closes issue #17277)
Reported by: cappucinoking
Patches:
heap-fix.rev2.diff uploaded by russell (license 2)
Tested by: cappucinoking, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes some breakage in the test suite, that uses the contents of asterisk.conf
to discover the install layout on the system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Refer-To header field containing the Replaces header in the URI
was not being decoded properly. This caused invalid parsing between
the caller id field and the domain resulting in a failed transfer.
(closes issue #17284)
Reported by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a recognized media type was present, but the media type was not
enabled for the channel, then a warning would be emitted. For instance,
attempting to get CHANNEL(rtpsource,video) on a call with no video would
cause a warning message to appear.
With this change, the warning will only appear if the stream argument
is not recognized as being a media type that can be specified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This simply moves the functionality from the Makefile (cleaning it up) into an external
asterisk.conf.samples file. Also updates formatting (easier to read) and grammar
changes to asterisk.conf.samples.
(closes issue #17027)
Reported by: pabelanger
Patches:
0017027.asterisk.conf.v6.patch uploaded by pabelanger (license 224)
Tested by: qwell, lmadsen, pabelanger, chappell
Review: https://reviewboard.asterisk.org/r/616/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) | 7 lines
Protect against overflow, when calculating how long to wait for a frame.
(closes issue #17128)
Reported by: under
Patches:
d.diff uploaded by under (license 914)
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r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 lines
Add a tiny corner case to the previous commit
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CLI "dahdi show channel" command was not correctly reporting the
InAlarm status.
The inalarm flag is now consistently passed between chan_dahdi and
submodules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines
Voicemail transfer to operator should occur immediately, not after main menu.
There were two scenarios in the advanced options that while using the
operator=yes and review=yes options, the transfer occurred only after exiting
the main menu (after sending a reply or leaving a message for an extension).
Now after the audio is processed for the reply or message the transfer occurs
immediately as expected.
ABE-2107
ABE-2108
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the following in-conference admin DTMF features:
*81 - Roll call (or simply user count if INTROUSER isn't enabled)
*82 - Eject all non-admins
*83 - Mute/unmute all non-admins
*84 - Start recording the conference on the fly
FWIW, this code uses newly recorded prompts.
(closes issue #16379)
Reported by: rfinnie
Patches:
meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940)
modified slightly by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We have some functions inside the AstData API to get the tree
in XML form, but it is not required at the moment to compile
asterisk and we can disable that part of the API if we don't have
libxml2 support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines
Ensure channel state is not incorrectly set in the case of a very early answer.
The needringing bit was being read in dahdi_read after answering thereby
setting the state to ringing from up. This clears needringing upon answering
so that is no longer possible.
(closes issue #17067)
Reported by: tzafrir
Patches:
needringing.diff uploaded by tzafrir (license 46)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created
SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS
SIG_MFCR2_MAX_CHANNELS
Also fixed the declaration of pollers[] in mfcr2_monitor(). It was
dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to
the same dimension of the struct dahdi_mfcr2.pvts[].
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr 2010) | 18 lines
Fix potential crash from race condition due to accessing channel data without the channel locked.
In res_musiconhold.c, there are several places where a channel's
stream's existence is checked prior to calling ast_closestream on it. The issue
here is that in several cases, the channel was not locked while checking the
stream. The result was that if two threads checked the state of the channel's
stream at approximately the same time, then there could be a situation where
both threads attempt to call ast_closestream on the channel's stream. The result
here is that the refcount for the stream would go below 0, resulting in a crash.
I have added proper channel locking to res_musiconhold.c to ensure that
we do not try to check chan->stream without the channel locked. A Digium customer
has been using this patch for several weeks and has not had any crashes since
applying the patch.
ABE-2147
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a specific position is specified for the queue, the idea
was that the caller cannot be placed ahead of higher-priority
callers. Unfortunately, the logic was reversed so that the caller
could ONLY be placed ahead of higher priority callers.
Discovered while writing a unit test.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This saves time, when, even though the system allows the process limit to be
that high, the practical limit is much lower. Also introduce an additional
optimization, in the form of using the CLOEXEC flag to close descriptors at
the right time.
(closes issue #17223)
Reported by: dbackeberg
Patches:
20100423__issue17223.diff.txt uploaded by tilghman (license 14)
Tested by: dbackeberg
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines
DTMF CallerID detection problems.
The code handling DTMF CallerID drops digits on long CallerID numbers and
may timeout waiting for the first ring with shorter numbers.
The DTMF emulation mode was not turned off when processing DTMF CallerID.
When the emulation code gets behind in processing the DTMF digits it can
skip a digit.
For shorter numbers, the timeout may have been too short. I increased it
from 2 seconds to 4 seconds. Four seconds is a typical time between rings
for many countries.
(closes issue #16460)
Reported by: sum
Patches:
issue16460.patch uploaded by rmudgett (license 664)
issue16460_v1.6.2.patch uploaded by rmudgett (license 664)
Tested by: sum, rmudgett
Review: https://reviewboard.asterisk.org/r/634/
JIRA SWP-562
JIRA AST-334
JIRA SWP-901
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines
Fixes crash in audiohook_write_list
The middle_frame in the audiohook_write_list function was
being freed if a audiohook manipulator returned a failure.
This is incorrect logic. This patch resolves this and
adds detailed descriptions of how this function should work
and why manipulator failures must be ignored.
(closes issue #17052)
Reported by: dvossel
Tested by: dvossel
(closes issue #16196)
Reported by: atis
Review: https://reviewboard.asterisk.org/r/623/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines
resolves deadlocks in chan_local
Issue_1.
In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan,
and pvt->owner. Proper deadlock avoidance is done when the channel to hangup
is the outbound chan_local channel, but when it is not the outbound channel we
have an issue... We attempt to do deadlock avoidance only on the tech pvt, when
both the tech pvt and the pvt->owner are locked coming into that loop. By
never giving up the pvt->owner channel deadlock avoidance is not entirely possible.
This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt
when trying to get the pvt->chan lock.
Issue_2.
ast_prod() is used in ast_activate_generator() to queue a frame on the channel
and make the channel's read function get called. This function is used in
ast_activate_generator() while the channel is locked, which mean's the channel
will have a lock both from the generator code and the frame_queue code by the
time it gets to chan_local.c's local_queue_frame code... local_queue_frame
contains some of the same crazy deadlock avoidance that local_hangup requires,
and this recursive lock prevents that deadlock avoidance from happening correctly.
This patch removes ast_prod() from the channel lock so only one lock is held during
the local_queue_frame function.
(closes issue #17185)
Reported by: schmoozecom
Patches:
issue_17185_v1.diff uploaded by dvossel (license 671)
issue_17185_v2.diff uploaded by dvossel (license 671)
Tested by: schmoozecom, GameGamer43
Review: https://reviewboard.asterisk.org/r/631/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) | 6 lines
Update config.guess.
Updating config.guess because after installing Ubuntu Server 9.10 and
running all the update scripts, running ./configure would not continue
because it was unable to determine what kind of system I had. After
updating config.guess things started working again.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | 7 lines
Remove usage of `id` since it isn't useful and was causing breakge.
Solaris `id` doesn't support the -u argument. Instead of figuring out how to
fix this to work on Solaris, I decided to check why it was necessary and where
else it was used. It was only used in one place, and it hasn't been needed
for a very long time (I question whether it was ever needed).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Alec L Davis
Leif Madsen
Jeff Peeler
Richard Mudgett
Jason Parker
Russell Bryant
Tzafrir Cohen
David Vossel
Paul Belanger
Mark Michelson
Eliel C. Sardanons