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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@261735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After finishing a recording from within the mailbox options menu, pressing 0
exhibited strange behavior with operator=yes turned on. Pressing 0 was not
even advertised as an option and the options from the vm-saveoper prompt:
"Press 1 to accept this recording. Otherwise, please continue to hold" did
not function correctly. While this of course could be fixed, it didn't really
seem to make sense even if it was working properly.
ABE-2121
SWP-1267
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@261698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@258670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Specifically, prompting for an extension (when leaving or forwarding a message)
or when prompting for a digit (when saving a message or changing folders).
ABE-2122
SWP-1268
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@258432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a user's mailbox was full and a message was attempted to be forwarded to
said box, warnings on the console would indicate failure. However, the played
prompt was that of success (vm-msgsaved). Now storage failure is taken into
account and the correct prompt (vm-mailboxfull) is played when appropriate.
ABE-2123
SWP-1262
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@258029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mantis issue 17078 reports MixMonitor recordings have shorter durations than
the call duration. This was because the mixmonitor thread was not processing
frames from the audiohook fast enough. The mixmonitor thread would slowly fall
behind the most recent audio frame and when the channel hangs up, the mixmonitor
thread would exit without processing the same number of frames as the channel;
leaving the mixmonitor recording shorter than actual call duration.
This revision fixes this issue by moving the ast_audiohook_trigger_wait() and
the subsequent audiohook.status check into the block where the
ast_audiohook_read_frame() function returns NULL.
(closes issue #17078)
Reported by: geoff2010
Patches:
dw-M17078.patch uploaded by dhubbard (license 733)
Tested by: dhubbard, geoff2010
Review: https://reviewboard.asterisk.org/r/611/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@257686 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a reply to a deregistration is lost in transmit, the client retries the
deregistration. Previously, this would cause a realtime/autocreate peer to be
loaded back into memory, after it had already been correctly purged. Instead,
we just want to resend the reply without loading the peer.
(closes issue #16908)
Reported by: kkm
Patches:
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
Tested by: kkm
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@257467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update the backtrace.txt documentation so it conforms to the same layout as
other documents we've been working on recently. Additionally, add a bunch of
new information about gathering backtraces for crashes and deadlocks, along
with ways of verifying your file before uploading it. Create a couple of one
line commands for people to generate the files we need.
(closes issue #17190)
Reported by: lmadsen
Patches:
backtrace.txt.patch-2 uploaded by lmadsen (license 10)
Tested by: lmadsen, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@257426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested.
(closes issue #17023)
Reported by: nblasgen
Review: https://reviewboard.asterisk.org/r/602/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@257070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Paul Belanger has been helping a lot with bug tracking recently and created
this document that we can now point to when additional debugging information
is required. This document will help those filing issues to know how to get
the information required when filing their issues. This will make things
easier on the developers.
Initial text and changes by pabelanger. Tweaks and editing by myself.
(closes issue #17159)
Reported by: pabelanger
Patches:
HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10)
Tested by: tzafrir, pabelanger, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@256900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes an issue with certain Mail Transport Agents, where attachments are not
interpreted correctly.
(closes issue #16557)
Reported by: jcovert
Patches:
20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: ebroad, zktech
Reviewboard: https://reviewboard.asterisk.org/r/544/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@255591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
