Actions (or failures to act) by external clients should not cause memory leaks
in Asterisk, especially when those continued leaks could cause Asterisk to
misbehave later.
(closes issue #17234)
Reported by: mav3rick
Patches:
20100510__issue17234.diff.txt uploaded by tilghman (license 14)
20100517__issue17234__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: mav3rick, davidw
(closes issue #17365)
Reported by: davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) | 11 lines
Prevent CLI prompt from distorting output of lines shorter than the prompt.
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Newer versions of libical (which we require) store the header file in a
libical/ subfolder and include an ical.h file that does a #warning for
deprecation and then #includes <libical/ical.h>. Since we now test for
libical/ical.h, we can change the #includes back to <libical/ical.h> and
remove the test which specifically adds /usr/include/libical as an include
directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Queries from the PBX core come in 3's. Caching avoids the additional
performance penalty from those two additional queries hitting the database.
(closes issue #16521)
Reported by: tilghman
Patches:
20091229__issue16521.diff.txt uploaded by tilghman (license 14)
Tested by: Hubguru, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines
Use sigaction for signals which should persist past the initial trigger, not signal.
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch breaks up every part of the sip registry string during
config parsing and removes all parsing from transmit_register().
Thanks to Nick_Lewis for contributing this patch!
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657)
chan_sip.c.patch uploaded by Nick Lewis (license 657)
chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657)
nicklewispatch.diff uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
Review: https://reviewboard.asterisk.org/r/628/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r266004 | mmichelson | 2010-05-26 13:21:10 -0500 (Wed, 26 May 2010) | 12 lines
Make AgentComplete message more consistent.
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This uses a modified version of pabelanger's patch that checks for NTLM support
instead, which was added in 0.29.0 which is what is required for
res_calendar_ews.
(closes issue #17391)
Reported by: loloski
Patches:
issue17391.patch.v2 uploaded by pabelanger (license 224)
Tested by: twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This ensures cross-platform compatibility, even among Linux distributions,
which don't always put headers in the same place.
(closes issue #17391)
Reported by: loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The logic used in transmit_register to get the outboundproxy for a peer
was flawed since this value would be overridden shortly afterwards when
create_addr was called.
In addition, this also fixes some logic used when parsing users.conf so
that the peer name is placed in the internally-generated register string
so that an outboundproxy set in the Asterisk GUI will be used for outbound
REGISTERs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The handling of the control subclass AST_CONTROL_READ_ACTION frame leaked
connected line string memory in __ast_read().
Also in __ast_read() the frame type switch should not have had a case for
AST_CONTROL_READ_ACTION. AST_CONTROL_READ_ACTION is not a frame type.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Patch supplied by reporter was modified to use autoservice and
prevent a potential channel ref leak but is otherwise as the
reporter uploaded it.
(closes issue #17182)
Reported by: rcasas
Patches:
app_senddtmf.c.patch_trunk uploaded by rcasas (license 641)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The decoding of the replace_id did not need to be broken
up in this instance. This was brought to my attention
again because it caused a segfault when the from or to
tags were not present in the "Replaces" header.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
__ast_channel_alloc_ap() has several points in the initialization of a new
channel structure where it could fail. Since the channel structure is now
an ao2 object, the destructor callback needs to be able to handle clean up
when the structure setup is incomplete.
Problems corrected:
1) Failing to setup the alertpipe would not unreference the structure but
free it directly. Doing this to an ao2_object is very bad.
2) File descriptors need to be initialized to -1 before a construction
failure could occur so the destructor will not close unopened descriptors.
3) The destructor needs to check that the string field has been
initialized before using any string field values. Crashes expected.
4) The destructor should not notify devstate if the device name is empty.
It is a waste of cycles and a couple ERROR log messages are generated.
Review: https://reviewboard.asterisk.org/r/675/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines
Don't hang up on a queue caller if the file we attempt to play does not exist.
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The same code was repeated in lots of different places, so I made a utility
fuction for it. This should make the merge in the v6-new branch easier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264905 65c4cc65-6c06-0410-ace0-fbb531ad65f3