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Merged revisions 265611 via svnmerge from
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r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May 2010) | 15 lines
Merged revisions 265610 via svnmerge from
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r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines
Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670.
(closes issue #17334)
Reported by: jvandal
Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
Tested by: aragon, jvandal
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Tick tock on the clock.
Shoutouts to kpfleming and DJ Funky Fresh.
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(issue #17277)
Reported by: cappucinoking
Patches:
test_heap.diff uploaded by cappucinoking (license 1036)
Tested by: cappucinoking, russell
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r261496 | russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
Fix handling of removing nodes from the middle of a heap.
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
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r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) | 4 lines
When failing to configure, don't destroy 'cfg' twice
Fixes a crash when some config section had an incorrect channel config.
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r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May 2010) | 19 lines
Merged revisions 261274 via svnmerge from
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r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines
Registration fix for SIP realtime.
Make sure realtime fields are not empty.
(closes issue #17266)
Reported by: Nick_Lewis
Patches:
chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis, sberney
Review: https://reviewboard.asterisk.org/r/643/
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r261232 | pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10 lines
'queue reset stats' erroneously clears wrapuptime configuration.
Resets each member's lastcall to 0 now.
(closes issue #17262, #16519)
Reported by: rain
Patches:
wrapuptime_reset_fix.diff uploaded by rain (license 327)
Tested by: rain
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r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010) | 18 lines
Merged revisions 261093-261094 via svnmerge from
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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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r260924 | jpeeler | 2010-05-04 13:51:28 -0500 (Tue, 04 May 2010) | 18 lines
Merged revisions 260923 via svnmerge from
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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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r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 Mar 2010) | 49 lines
Improve handling of T.38 re-INVITEs that arrive before a T.38-capable
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
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r260802 | qwell | 2010-05-04 10:49:57 -0500 (Tue, 04 May 2010) | 9 lines
Merged revisions 260801 via svnmerge from
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r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May 2010) | 1 line
Fix fallout from removing from configure script. Pointed out by philipp64 on #asterisk-dev
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r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May 2010) | 10 lines
non-root make install PREFIX=/tmp fails.
Prepend libdir when executing mkpkgconfig allowing non-root installs to work.
(closes issue #17268)
Reported by: pabelanger
Patches:
issue17268.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
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r260662 | pabelanger | 2010-05-03 12:54:41 -0400 (Mon, 03 May 2010) | 3 lines
Should have removed /usr/lib/ part. Thanks Qwell.
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r260570 | lmadsen | 2010-05-03 09:58:23 -0500 (Mon, 03 May 2010) | 9 lines
Merged revisions 260569 via svnmerge from
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r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line
Minor typo pointed out by pabelanger on IRC.
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r260437 | jpeeler | 2010-04-30 17:36:49 -0500 (Fri, 30 Apr 2010) | 18 lines
Merged revisions 260434 via svnmerge from
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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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r260346 | mmichelson | 2010-04-30 15:11:02 -0500 (Fri, 30 Apr 2010) | 24 lines
Merged revisions 260345 via svnmerge from
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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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r260292 | tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13 lines
Don't allow file descriptors to go above 64k, when we're closing them in a fork(2).
This saves time, when, even though the system allows the process limit to be
that high, the practical limit is much lower.
(closes issue #17223)
Reported by: dbackeberg
Patches:
20100423__issue17223.diff.txt uploaded by tilghman (license 14)
Tested by: dbackeberg
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r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30 Apr 2010) | 7 lines
Logic fixups for a sample FREENUM dialplan context.
(closes issue #17263)
Reported by: pprindeville
Patches:
freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
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r260231 | rmudgett | 2010-04-29 17:44:14 -0500 (Thu, 29 Apr 2010) | 33 lines
Merged revisions 260195 via svnmerge from
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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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r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29 Apr 2010) | 2 lines
Pattern match fail.
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r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) | 21 lines
Merged revisions 260049 via svnmerge from
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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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r259957 | mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 lines
Don't override peer context with domain context.
(closes issue #17040)
Reported by: pprindeville
Patches:
asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347)
Tested by: pprindeville
Review: https://reviewboard.asterisk.org/r/565/
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r259870 | dvossel | 2010-04-28 16:20:03 -0500 (Wed, 28 Apr 2010) | 39 lines
Merged revisions 259858 via svnmerge from
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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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r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010) | 14 lines
Merged revisions 259852 via svnmerge from
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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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r259848 | qwell | 2010-04-28 15:32:14 -0500 (Wed, 28 Apr 2010) | 9 lines
Merged revisions 259847 via svnmerge from
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r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr 2010) | 1 line
Add AC_CONFIG_AUX_DIR to configure script, so systems without install can use install-sh from our source dir.
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r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) | 9 lines
Merged revisions 259833 via svnmerge from
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r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | 1 line
Missed this when removing $ID
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r259760 | qwell | 2010-04-28 14:19:54 -0500 (Wed, 28 Apr 2010) | 14 lines
Merged revisions 259748 via svnmerge from
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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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r259672 | jpeeler | 2010-04-28 12:18:43 -0500 (Wed, 28 Apr 2010) | 11 lines
Merged revisions 259664 via svnmerge from
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r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) | 4 lines
Do not play goodbye prompt after timeout of message review.
ABE-2124
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r259538 | rmudgett | 2010-04-27 17:18:09 -0500 (Tue, 27 Apr 2010) | 18 lines
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r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines
DAHDI "WARNING" message is confusing and vague
"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success"
Changed the warning to "Failed to decode CallerID on channel 'name'". The
message before it is likely more specific about why the CallerID decode
failed.
SWP-501
AST-283
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r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010) | 23 lines
Merged revisions 259526 via svnmerge from
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r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) | 15 lines
Update sounds files.
* Add additional sounds prompts for say_enumeration
* Update the English conference sounds prompts so they are better
quality and all sound more consistent
* Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files to
include all present sound files
Both core (en, fr, es) and extra (en, fr) sounds files have been updated.
(closes issue #16200)
Reported by: murf
(closes issue #17137)
Reported by: lmadsen
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r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) | 5 lines
Add gar to the check for AR for those silly OSes (Solaris) that don't have ar.
autoconf2.13 couldn't handle AC_PROG_GREP, so I removed it. This is fine,
since we don't need to use anything that the configure script doesn't.
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r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) | 5 lines
Add gar to the check for AR for those silly OSes (Solaris) that don't have ar.
autoconf2.13 couldn't handle AC_PROG_GREP, so I removed it. This is fine,
since we don't need to use anything that the configure script doesn't.
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r259353 | qwell | 2010-04-27 14:31:55 -0500 (Tue, 27 Apr 2010) | 12 lines
Merged revisions 259352 via svnmerge from
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r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | 5 lines
Support the silly OSes that don't have ar and strip.
Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't specified, and
AC_PATH_TOOLS doesn't exist, we'll just switch to AC_CHECK_TOOLS.
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r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) | 21 lines
Merged revisions 259270 via svnmerge from
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r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines
hidecalleridname parameter in chan_dahdi.conf
Issue #7321 implements a new chan_dahdi configuration option. However, a
change mentioned in the issue was never implemented. This is the change
that will allow the feature to work.
I added a note to chan_dahdi.conf.sample about the feature.
(closes issue #17143)
Reported by: djensen99
Patches:
diff.txt uploaded by djensen99 (license NA) (One line change)
Tested by: djensen99
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r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr 2010) | 9 lines
Merged revisions 259104 via svnmerge from
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r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr 2010) | 3 lines
Let compilation succeed warning-free when DONT_OPTIMIZE is turned off.
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r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr 2010) | 19 lines
Merged revisions 259018 via svnmerge from
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r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr 2010) | 13 lines
Prevent Newchannel manager events for dummy channels.
No Newchannel manager event will be fired for channels that are
allocated to not match a registered technology type. Thus bogus
channels allocated solely for variable substitution or CDR
operations do not result in a Newchannel event.
(closes issue #16957)
Reported by: atis
Review: https://reviewboard.asterisk.org/r/601
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r258934 | lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
Small error in the T.140 RTP port verbose log.
(closes issue #16988)
Reported by: frawd
Patches:
chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610)
Tested by: russell
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r258595 | eliel | 2010-04-22 16:04:23 -0400 (Thu, 22 Apr 2010) | 3 lines
Pass interactive = 0 and fix a compile error.
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r258517 | eliel | 2010-04-22 14:07:02 -0400 (Thu, 22 Apr 2010) | 14 lines
Asterisk data retrieval API.
This module implements an abstraction for retrieving and exporting
asterisk data.
Developed by:
Brett Bryant <brettbryant@gmail.com>
Eliel C. Sardanons (LU1ALY) <eliels@gmail.com>
For the Google Summer of code 2009 Project.
Documentation can be found in doxygen format and inside the
header include/asterisk/data.h
Review: https://reviewboard.asterisk.org/r/275/
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r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010) | 13 lines
Merged revisions 258775 via svnmerge from
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r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines
When StopMonitor is called, ensure that it will not be restarted by a channel event.
(closes issue #16590)
Reported by: kkm
Patches:
resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888)
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r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr 2010) | 32 lines
Merged revisions 193391,258670 via svnmerge from
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r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May 2009) | 8 lines
Set the proper disposition on originated calls.
(closes issue #14167)
Reported by: jpt
Patches:
call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
Tested by: dlotina, rmartinez, mnicholson
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r258670 | mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 lines
Fix broken CDR behavior.
This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER.
Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call(). To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call().
(closes issue #16797)
Reported by: VarnishedOtter
Tested by: mnicholson
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(closes issue #16222)
Reported by: telles
Tested by: mnicholson
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r258675 | mnicholson | 2010-04-22 17:11:23 -0500 (Thu, 22 Apr 2010) | 2 lines
Fix previous commit.
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For 1.6.2, only merge the bug fixes, not the unit test.
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r258632 | russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
Add ast_event subscription unit test and fix some ast_event API bugs.
This patch introduces another test in test_event.c that exercises most of the
subscription related ast_event API calls. I made some minor additions to the
existing event allocation test to increase API coverage by the test code.
Finally, I made a list in a comment of API calls not yet touched by the test
module as a to-do list for future test development.
During the development of this test code, I discovered a number of bugs in
the event API.
1) subscriptions to AST_EVENT_ALL were not handled appropriately in a couple
of different places. The API allows a subscription to all event types,
but with IE parameters, just as if it was a subscription to a specific
event type. However, the parameters were being ignored. This affected
ast_event_check_subscriber() and event distribution to subscribers.
2) Some of the logic in ast_event_check_subscriber() for checking subscriptions
against query parameters was wrong.
Review: https://reviewboard.asterisk.org/r/617/
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