This will save a considerable amount of CPU on the BSDs, including Mac OS X,
as it eliminates several places in the code that we previously used a busy
loop. Additionally, this adds a res_timing interface, using kqueue timers.
Review: https://reviewboard.asterisk.org/r/543/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Marquis42 suggested a better method of doing what I wanted because I ended up
removing the WARNING message for all instances when really I just wanted to
remove it for the 'return' keyword, not everything.
(issue #17145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines
fixes app_meetme dsp error
We attempted to detect silence after translating a frame
from signed linear. This caused a flooding of errors. To
resolve this the code to detect silence was moved before the
translation.
(closes issue #17133)
Reported by: jsdyer
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise,
when the duplicate pseudo channel is destroyed, it frees the original
pseudo channel cc_params. The original pseudo channel is then left with a
dangling pointer for when the next duplicated pseudo channel is created.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, 12 May 2010) | 4 lines
Forgot some conditionals around the callrerouting facility help text.
JIRA ABE-2223
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r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) | 22 lines
Add mISDN Call rerouting facility for point-to-point ISDN lines (exchange line)
In the case of ISDN point-to-multipoint (multidevice) you can use the
mISDN "facility calldeflect" application for call diversions from external
(PSTN) to external (PSTN). In that case this is the only way to get rid
of the two call legs to the PBX and let the calling number at the C party
become the number of the A party. In the case of ISDN point-to-point
(exchange line) the call deflection facility may not be used. Instead a
call rerouting facility has to be used.
This patch for chan_misdn.c is an extension to realize this service
(facility rerouting application). It can accept either spelling:
"callrerouting" or "callrerouteing".
The patch is tested towards Deutsche Telekom and requires a modified
version of mISDN from Digium, Inc.
Patches:
misdn_rerouteing_corrected.patch (Slightly modified.)
JIRA ABE-2223
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Clean up chan_sip.c to use new AST_CLI functions
(closes issue #17287)
Reported by: pabelanger
Patches:
issue17287.patch uploaded by pabelanger (license 224)
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931
Section 5.3.2. However, this resulted in an unexpected behaviour change
to the upper layer (Asterisk).
This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2
call hangup better if the version of libpri supports it.
(issue #17104)
Reported by: shawkris
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines
Use a less silly method for modifying a flex-generated file.
The sed syntax that was used wasn't actually valid, causing some versions to
choke. This is the method that is used in 1.6.x+ for similar changes.
(closes issue #16696)
Reported by: bklang
Patches:
16696-sedfix.diff uploaded by qwell (license 4)
Tested by: qwell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As usual, the way they do things is different, so we need to account for that.
automake is versioned ala BSD/Linux, but autoconf is not. We don't actually
need to specify a version there, since AC_PREREQ will cover it for us. Things
will fail pretty loudly if AC_PREREQ isn't met.
(closes issue #16341)
Reported by: bklang
Patches:
opensolaris_bootstrap.sh uploaded by bklang (license 919)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a race condition between console_hangup()
and start_stream(). It is possible for console_hangup()
to be called and then the stream thread to begin after the hangup.
To avoid this a check in start_stream() to make sure the pvt-owner
still exists while the pvt lock is held is made. If the owner
is gone that means the channel hung up and start_stream should
be aborted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
