Uses the VT100 method of clearing the line from the cursor position to the
end of the line: Esc-0K
(closes issue #17160)
Reported by: coolmig
Patches:
20100531__issue17160.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a T.38 re-INVITE failed with an 488 or 606 answer, we should
fallback to audio fax by send a re-re-INVITE without T.38. The
function is backported from 1.6 asterisk.
(closes issue #16795)
Reported by: vrban
(closes issue #16692)
Reported by: vrban
Patches:
t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
Tested by: lmadsen, vrban, haggard
https://reviewboard.asterisk.org/r/514/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you call signal() in a Solaris signal handler, instead of just resetting
the signal handler, it causes the signal to refire, because the signal is not
marked as handled prior to the signal handler being called. This effectively
causes Solaris to immediately exceed the threadstack in recursive signal
handlers and crash.
(closes issue #17000)
Reported by: rmcgilvr
Patches:
20100526__issue17000.diff.txt uploaded by tilghman (license 14)
Tested by: rmcgilvr
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
At times, the "Member" field was not specified during the event.
It's there now.
(closes issue #15638)
Reported by: elbriga
Patches:
patchAppQueueAgentComplete.diff uploaded by elbriga (license 482)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
Add the FullyBooted AMI event
It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can listen for the
FullyBooted manager event. It will be sent upon connection if all modules have
been loaded, or as soon as loading is complete. The event:
Event: FullyBooted
Privilege: system,all
Status: Fully Booted
Review: https://reviewboard.asterisk.org/r/639/
........
r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
Merge the rest of the FullyBooted patch
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@265570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This also fixes a documentation mistake in file.h that made my original attempt
to correct this problem not work correctly.
(closes issue #17061)
Reported by: RoadKill
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@265089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Several callers of ast_callerid_parse() do not initialize the name
parameter before calling thus there is the potential to use an
uninitialized pointer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The reason for ensuring DAHDI 2.3 or above is that this version ensures that
a timer is always available, whereas in previous versions, it was possible
for DAHDI to be loaded, but have no drivers to actually generate timing. If
internal_timing was turned on in this circumstance, a complete lack of audio
would result. This is the reason why internal_timing was not on by default.
However, now that DAHDI ensures the availability of a timer, there is no
reason for this setting to be off (and in fact, it solves a great many initial
user problems).
(closes issue #15932)
Reported by: dimas
Patches:
20100519__issue15932.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the case of accidentally entering the wrong first three letters for the
reading, users could be very frustrated if the name listing is very long. This
allows interrupting the reading by pressing 0 or #. 0 will attempt to execute
a configured operator (o) extension and # will exit and proceed in the
dialplan.
ABE-2200
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@263769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using strsep, if one of the list of specified separators is not found,
it is the first parameter to strsep which is now NULL, not the pointer returned
by strsep.
This issue isn't especially severe in that the worst it is likely to do is waste
some cycles when a device with no '/' and no ':' is passed to ast_device_state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@263639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Version field in the cookies we're setting contain quotes around the version
number which is not compatible with RFC2109 and breaks some implementations.
(closes issue #17231)
Reported by: ecarruda
Patches:
manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559)
manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559)
Tested by: ecarruda, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@263456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The latest version of the core sounds files 1.4.19 now includes the missing
queue-minute sound file which is called by app_queue but which has been
missing.
(closes issue #17123)
Reported by: n8ideas
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@263374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
dahdi_compat.h was not being included in channel.c when used with
Zaptel and wasn't in file.c at all.
(closes issue #15250)
Reported by: mneuhauser
Patches:
dahdi_compat.patch uploaded by mneuhauser (license 425)
Tested by: IgorG
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@263112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@262662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@262421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
