https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"Bad Things" would happen if Asterisk was compiled with DEBUG_THREADS, but a
loaded module was not (or vice versa). This also immensely simplifies the
lock code, since there are no longer 2 separate versions of them.
Review: https://reviewboard.asterisk.org/r/508/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines
Fix looping forever when no input received in certain voicemail menu scenarios.
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
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These changes add the ability to run 'make asterisk.txt' just like the existing
'make asterisk.pdf' commands to generate a text document from the TeX files we
have in the doc/tex/ directory. I've also updated a few of the .tex files because
they weren't properly escaping certain characters so they would show up as Unicode
characters (like [U+021C]). Made changes to the configure scripts so it would
detect the catdvi program which is required to convert the .dvi file generated
by latex.
I've also added a few lines to the build_tools/prep_tarball script so that the
text documentation gets generated and added to future tarballs of Asterisk
releases.
(closes issue #17220)
Reported by: lmadsen
Patches:
asterisk.txt.patch uploaded by lmadsen (license 10)
asterisk.txt.patch-v4 uploaded by pabelanger (license 224)
Tested by: lmadsen, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This directs users to documents which can help explain the
concepts and configuration options settable with the function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a clear mistake in logic. Future discussions
about how to avoid having to handle uri's like this
should take place in the future, but this fix needs
to go in for now.
(closes issue #15847)
Reported by: ebroad
Patches:
doublesip.patch uploaded by ebroad (license 878)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added a new manager command to mute/unmute MixMonitor audio on a channel.
Added a new feature to audiohooks so that you can mute either read / write
(or both) types of frames - this allows for MixMonitor to mute either side
of the conversation without affecting the conversation itself.
(closes issue #16740)
Reported by: jmls
Review: https://reviewboard.asterisk.org/r/487/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines
Play correct prompt when voicemail store failure occurs after attempted forward.
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
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Updated the doxygen \arg line after looking at the file for some other Asterisk documentation
and noticing they weren't up to date. Thanks to seanbright for looking at the code for me :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were many changes in revision 176627 which would avoid the error that a
missing config would have caused. Other than this, there are no other config
files (including asterisk.conf, surprisingly) that are required.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for multiple SRV queries to be done
from the dialplan for the same service on a single call while
still allowing one to bypass the call to SRVQUERY if they so
please.
Taking action since no comments had been left for a while.
This can easily be reverted if needed. External tests
still pass.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Because res/res_features.c was removed and main/cdr.c added, these changes
didn't make it to trunk and the 1.6.x branches
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines
Make the mixmonitor thread process audio frames faster
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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision 1072 of libpri added SERVICE message support for the 'national'
switchtype. The attached patch enables the use of 'pri service' CLI commands
on dahdi channels that are configured for the 'national' switchtype.
(closes issue #17142)
Reported by: dhubbard
Patches:
dw-ni2.patch uploaded by dhubbard (license 733)
Tested by: elguero, dhubbard
Review: https://reviewboard.asterisk.org/r/612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines
Don't recreate peer, when responding to a repeated deregistration attempt.
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
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r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines
Update backtrace.txt documentation.
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Application arguments within the feature map could possibly contain a comma,
which conflicts with the syntax of the features.conf configuration file. This
patch allows the argument to be wrapped in parentheses or quoted, to allow the
application arguments to be interpreted as a single configuration parameter.
(closes issue #16646)
Reported by: pinga-fogo
Patches:
20100414__issue16646.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/547/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines
Add an option to restore past broken behavor of the Events manager action
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/
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