This patch adds the ability to create a serializer from a thread pool. A
serializer is a ast_taskprocessor with the same contract as a default
taskprocessor (tasks execute serially) except instead of executing out
of a dedicated thread, execution occurs in a thread from a
ast_threadpool. Think of it as a lightweight thread.
While it guarantees that each task will complete before executing the
next, there is no guarantee as to which thread from the pool individual
tasks will execute. This normally only matters if your code relys on
thread specific information, such as thread locals.
This patch also fixes a bug in how the 'was_empty' parameter is computed
for the push callback, and gets rid of the unused 'shutting_down' field.
Review: https://reviewboard.asterisk.org/r/2323/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Caching a struct ast_app pointer is not a good idea because someone could
unload the application. After the applicaiton unload the cached ast_app
pointer is no longer valid. Only pbx.c can cache the pointer because it
knows when the application is unloaded and removes the pointer.
* Fixed one-touch Monitor and MixMonitor to not cache the ast_app pointer
and not use the silly monitor_ok/mixmonitor_ok/stopmixmonitor_ok flags.
* Extracted bridge_check_monitor() from ast_bridge_call() and use propper
locking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Taking advantage of the sorted order of the registered functions container
requires that they are actually inserted in the expected sort order.
* Insert the registered functions into the container in case sensitive
position. As a result, only the complete_functions() routine needs to
search the entire container because it does a case insensitive search for
convenience.
Caught by the unit tests.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed "core show function" tab completion and token count checking.
* Refactored function and application container handling code to reduce
redundancy.
* Made __ast_pbx_run() return using the defines the caller should expect.
Doesn't change the returned values. Just made use the defines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These two variables were previously not being set when comebacktoorigin=yes
and the example configs seemed to imply that they should be. Since there
is no harm in this and since calls that are sent back to origin are capable
of continuing in the dialplan, this seemed like a no-brainer. Also it
supports some bridging tests I've been working on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A regression was introduced which removed automatic fallback behavior from
the PBX. This behavior was used by call parking (or at least documented as
how the feature works) in order to select an extension when the flat channel
extension wasn't available from the comebackcontext. Parking now handles
the fallbacks internally in order to keep behavior matching with how it is
documented.
(closes issue ASTERISK-20716)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2296/
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This patch came about due to a problem observed where wav files had an
empty header. The header is supposed to be updated in wav_close(). It
turns out that this was broken when the cache_record_files option from
asterisk.conf was enabled. The cleanup code was moving the file to its
final destination *before* running the close() method of the file
destructor, so the header didn't get updated.
Another problem here is that the move was being done before actually
closing the FILE *.
Finally, the last bug fixed here is that I noticed that wav_close()
checks for stream->filename to be non-NULL. In the previous cleanup
order, it's checking a pointer to freed memory. This doesn't actually
cause anything to break, but it's treading on dangerous waters. Now the
free() of stream->filename is happening after the format module's
close() method gets called, so it's safer.
Review: https://reviewboard.asterisk.org/r/2286/
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* Made multiplexed_bridge_destroy() check if anything to destroy and
cleared bridge_pvt pointer after destruction.
* Made multiplexed_add_or_remove() handling of the chans array simpler.
* Extracted bridge_channel_poke().
* Simplified bridge_array_remove() handling of the bridge->array[]. The
array does not have a NULL sentinel pointer.
* Made ast_bridge_new() not create a temporary bridge just to see if it
can be done. Only need to check if there is an appropriate bridge tech
available.
* Made ast_bridge_new() clean up on allocation failures.
* Made destroy_bridge() free resources in the opposite order of creation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sorcery is a unifying data access layer which provides a pluggable mechanism to allow
object creation, retrieval, updating, and deletion using different backends (or wizards).
This is a fancy way of saying "one interface to rule them all" where them is configuration,
realtime, and anything else that comes along.
Review: https://reviewboard.asterisk.org/r/2259/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.
Review: https://reviewboard.asterisk.org/r/2265/
(closes issue ASTERISK-20882)
Reported by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When r376428 was commited to re-order start up sequences to be more tolerant of
forking with thread primitives, a few items were changed that caused changes
in behavior on some distros. This includes:
* Not displaying the splash screen on a remote console.
* Displaying an error message on stderr when a remote console cannot connect
to a running instance of Asterisk.
In the first case, the splash screen was re-added (thanks to Michael L. Young).
In the second case, the various init.d scripts were modified to pipe stderr
to /dev/null, as the error message is useful - if you execute a remote
console or a remote console command execution and it fail, it should tell
you. Note that the error message was always present, it just failed to be
printed prior to r376428.
Much thanks to the folks who quickly reported this problem, provided solutions,
and promptly tested the various init.d scripts on a variety of distros.
(closes issue ASTERISK-20945)
Reported by: Warren Selby
Tested by: Michael L. Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan
patches:
asterisk-20945-remote-intro-msg.diff uploaded by elguero (license 5026)
ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan (license 6283)
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When Asterisk forks itself into the background via a call to daemon, it must
re-set the pid value of the new process. Otherwise, astcanary gets the pid
value of the process before the fork, which prevents it from running. Asterisk
eventually starts lowering its priority, as it can no longer communicate
with the proverbial canary in the coal mine.
This patch ensures that the correct process identifier is used by astcanary.
Note that this is getting committed to 10 as a regression fix.
(closes issue ASTERISK-20947)
Reported by: Jakob Hirsch
Tested by: mjordan
patches:
asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch (license 6113)
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* This allows us to remove some special-case build logic.
* 10.5 is down to less that 8% of the OS X market share. 10.4 is down to
under 2%.
* Apple is no longer releasing security updates for 10.5 and earlier.
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* Add a max_size option for threadpools. Also added a test for this option.
* Fixed comments to be more accurate and have fewer typos.
* Updated copyright dates on new files.
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r379229 | mjordan | 2013-01-16 11:46:15 -0600 (Wed, 16 Jan 2013) | 10 lines
Let documentation reference links specify which module they're linking to
Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref
links have to specify which reference they're referring to. The various
documentation parsers can interpret the module attribute however they want
in order to construct the appropriate links.
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r379231 | rmudgett | 2013-01-16 11:49:52 -0600 (Wed, 16 Jan 2013) | 10 lines
chan_misdn: Fix compile error.
(issue ASTERISK-15456)
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r379233 | rmudgett | 2013-01-16 12:09:28 -0600 (Wed, 16 Jan 2013) | 8 lines
Reduce call-id logging resource usage.
Since there is no need for the call-id logging ao2 object to have a lock,
don't create it with one.
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* Clarify some documentation
* Change copyright date of taskprocessor files
* Address potential issue of creating taskprocessor with listener if
taskprocessor with that name exists already
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Now user data is allocated by the creator of the taskprocessor
listener and that user data is passed into ast_taskprocessor_listener_alloc().
Similarly, freeing of the user data is left up to the user himself. He can
free the data when the taskprocessor shuts down, or he can choose to hold
onto it if it makes sense to do so.
This, unsurprisingly, makes threadpool allocation a LOT cleaner now.
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r379021 | dlee | 2013-01-14 09:29:22 -0600 (Mon, 14 Jan 2013) | 15 lines
Fix XML encoding of 'identity display' in NOTIFY messages, continued.
When r378933 was merged into 1.8, it should have also escaped
remote_display, since it will have the same XML encoding problem when
the caller/callee roles are reversed.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
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r379023 | dlee | 2013-01-14 09:58:01 -0600 (Mon, 14 Jan 2013) | 20 lines
Masquerades are an insane implementation detail within Asterisk. It generates
a number of useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly thorough review of
masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ.
While ultimately it makes the most sense to abandon masquerades altogether,
it will take some time to completely irradicate. Even then, there may always
be code that's not worth rewriting to get rid of the masquerade.
This patch does two things to make masquerades slightly less insane:
* When swapping the names of the original and clone channel, only emit a
single rename event of original -> original<ZOMBIE>. The original code
issued three rename events to accomplish the same end.
* In addition to swapping the names of the channels, also swap their
uniqueid's. This allows the 'Uniqueid' field to be used as a stable
identifier for a channel from and external interface, such as AMI.
Review: https://reviewboard.asterisk.org/r/2266/
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r378935 | dlee | 2013-01-12 00:43:37 -0600 (Sat, 12 Jan 2013) | 41 lines
Fix XML encoding of 'identity display' in NOTIFY messages.
XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.
This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.
Several things to note:
* The Right Thing(TM) to do would probably be to replace the
ast_build_string stuff with building an ast_xml_doc. That's a much
bigger change, and out of scope for the original ticket, so I
refrained myself.
* It is with great sadness that I wrote my own ast_xml_escape
function. There's one in libxml2, but it's knee-deep in
libxml2-ness, and not easily used to one-off escape a
string.
* I only escaped the string we know is causing problems
(local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
Review: http://reviewboard.digium.internal/r/365/
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r378915 | dlee | 2013-01-11 16:31:42 -0600 (Fri, 11 Jan 2013) | 21 lines
Add JSON API for Asterisk.
This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.
* Some names have been asteriskified to protect the innocent.
* Jansson provides both reference-stealing and reference-borrowing
versions of several API's. The Asterisk API is exclusively
reference-stealing for operations that put elements into arrays and
objects.
* No support for doubles, since we usually don't need that.
* Coming along for the ride is the ast_test_validate macro, which made
the unit tests much easier to write.
[1]: http://www.digip.org/jansson/
(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/
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r378918 | file | 2013-01-11 17:05:38 -0600 (Fri, 11 Jan 2013) | 11 lines
Retain XMPP filters across reconnections so external modules continue to function as expected.
Previously if an XMPP client reconnected any filters added by an external module were lost.
This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling.
(closes issue ASTERISK-20916)
Reported by: kuj
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r378889 | rmudgett | 2013-01-09 20:40:50 -0600 (Wed, 09 Jan 2013) | 8 lines
* Simplify native bridge code in ast_channel_bridge().
* Fix an unbalanced manager_bridge_event(unlink) call if
AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge().
* Make ast_channel_bridge() use common cleanup code when leaving the
bridge.
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r378783 | dlee | 2013-01-09 14:30:33 -0600 (Wed, 09 Jan 2013) | 14 lines
Fix end condition in ast_rtp_lookup_mime_multiple2.
The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag
in the debug output.
(closes issue ASTERISK-20772)
Reported by: Xavier Hienne
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r378789 | rmudgett | 2013-01-09 14:56:23 -0600 (Wed, 09 Jan 2013) | 4 lines
* Found some more places to use ast_channel_lock_both().
* Minor optimization in ast_rtp_instance_early_bridge().
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r378790 | rmudgett | 2013-01-09 15:14:39 -0600 (Wed, 09 Jan 2013) | 4 lines
* Whitespace changes.
* Made ast_test_init() match its prototype.
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* Remove extraneous whitespace
* Bump up debug levels of messages and add identifying info to messages.
* Account for potential failures of ao2_link()
* Add additional test and some more test data
* Add some comments in places where they could be useful
* Make threadpool listeners and their callbacks optional
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r378458 | rmudgett | 2013-01-03 12:47:29 -0600 (Thu, 03 Jan 2013) | 18 lines
chan_agent: Misc code cleanup.
* Fix off-nominal path resource cleanup in agent_request().
* Create agent_pvt_destroy() to eliminate inlined versions in many places.
* Pull invariant code out of loop in add_agent().
* Remove redundant module user references in login_exec().
* Remove unused struct agent_pvt logincallerid[] member.
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r378460 | kmoore | 2013-01-03 12:51:43 -0600 (Thu, 03 Jan 2013) | 13 lines
Add missing test event
This test event was missing from channel.c causing the dial_LS_options
test to fail intermittently because of a race condition where most code
paths emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
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r378374 | rmudgett | 2013-01-02 15:23:16 -0600 (Wed, 02 Jan 2013) | 33 lines
Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged
together. There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.
* Made the bridge wait for both channels to be redirected before exiting.
* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.
* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding. Previously the code fell back to a single channel
redirect operation.
(closes issue ASTERISK-18975)
Reported by: Ben Klang
(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
Review: https://reviewboard.asterisk.org/r/2243/
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r378377 | mjordan | 2013-01-02 16:10:32 -0600 (Wed, 02 Jan 2013) | 24 lines
Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.
This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.
(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
* issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
* issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
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r378384 | mjordan | 2013-01-02 16:19:32 -0600 (Wed, 02 Jan 2013) | 11 lines
Clean up app_mysql's application entry points to properly parse arguments
When parsing arguments, application entry points should not attempt to
directly modify the parameters to the function. This patch properly duplicates
the passed in parameters before attempting to parse them.
(issue ASTERISK-20658)
Reported by: wdoekes
patches:
issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)
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r378322 | mjordan | 2013-01-02 12:11:59 -0600 (Wed, 02 Jan 2013) | 33 lines
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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r378288 | mjordan | 2013-01-02 09:39:42 -0600 (Wed, 02 Jan 2013) | 36 lines
Resolve crashes due to large stack allocations when using TCP
Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a stack
allocation
* For XMPP (in res_jabber), the allocation has been eliminated since it was
unnecesary.
Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.
(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
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r378072 | rmudgett | 2012-12-17 14:34:25 -0600 (Mon, 17 Dec 2012) | 9 lines
chan_local: Misc lock and ref tweaks.
* awesome_locking() does not need to thrash the pvt lock as much.
* local_setoption() does not need to check for NULL pvt on cleanup since
it will never be NULL.
* Made ref the pvt before locking for consistency.
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r378074 | qwell | 2012-12-17 14:59:51 -0600 (Mon, 17 Dec 2012) | 10 lines
Make libasteriskssl.so symlink use a relative path.
This was causing issues when using DESTDIR, since the path to which the link
pointed is not likely to exist (and not useful to exist) on the target system.
(issue ASTNOW-284)
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r378063 | jrose | 2012-12-14 16:34:18 -0600 (Fri, 14 Dec 2012) | 8 lines
Features: BRIDGE_FEATURES variable automixmonitor support and use proper party
BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
does. In addition, the BRIDGE_FEATURES variable would not apply features to
the proper party based on whether the feature option letter was in caps or
in lowercase (both ways would apply it to the caller). Now uppercase applies
to the caller while lowercase applies to the callee (like with the dial option)
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r378064 | rmudgett | 2012-12-14 16:45:03 -0600 (Fri, 14 Dec 2012) | 4 lines
chan_agent: Remove some duplicated code.
No need to check for an agent twice. Santa does that.
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r377915 | elguero | 2012-12-11 22:43:18 -0600 (Tue, 11 Dec 2012) | 17 lines
Convert Dynamic Features Buffer To Use ast_str
Currently, the buffer for the dynamic features list is set to a fixed size of
128. If the list is bigger than that, it results in the dynamic feature(s) not
being recognized.
This patch changes the buffer from a fixed size to a dynamic one.
(closes issue ASTERISK-20680)
Reported by: Clod Patry
Tested by: Michael L. Young
Patches:
asterisk-20680-dynamic-features-v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2221/
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r377906 | mmichelson | 2012-12-11 16:42:11 -0600 (Tue, 11 Dec 2012) | 3 lines
Add test events necessary for bridging tests to be able to properly run.
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The documentation for taskprocessors was incorrect with
regards to when a listener's alloc callback was called.
I also made the names of queued function calls in the
threadpool more uniform.
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r377743 | rmudgett | 2012-12-10 20:13:37 -0600 (Mon, 10 Dec 2012) | 25 lines
Cleanup indications on exit.
* Made ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to select
the tone zone being unregistered again.
* Ringcadence is no longer parsed twice in store_config_tone_zone().
* Cleanup CLI commands and destroy default_tone_zone on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Unfortunately, this required a taskprocessor listener change that makes listener allocation
utterly silly. I'm going to change the scheme so that allocation of taskprocessor listeners
is done internally within taskprocessor code. This will make it parallel with threadpool
code, which is a good thing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The only test added so far is an idle thread timeout
option. This will greatly aid threadpool users who wish
to maintain a threadpool by allowing for idle threads to
die out as necessary.
Test passes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This helps tests to pass more often than before.
They are far less likely to queue extra processes
into the control taskprocessor since they are prevented
once the threadpool begins to shut down.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This one involves shrinking the threadpool in such
a way that both idle and active threads are affected.
This test made me re-realize why the zombie state exists,
so I re-added it. We don't want to clog up the control
taskprocessor by waiting on active threads to complete
what they are doing. Instead, we mark them as zombies so
that when they are done, they can clean themselves up
properly.
Without the zombie state available, the new test actually
will deadlock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new thread creation test fails because Asterisk locks up
while trying to lock a taskprocessor.
While trying to debug that, I found a race condition during taskprocessor
creation where a default taskprocessor listener could try to operate on
a partially started taskprocessor. This was fixed by adding a new callback
to taskprocessor listeners.
Then while testing that change, I found some bugs in the taskprocessor
tests where I was not properly unlocking when done with a lock. Scoped
locks have spoiled me a bit.
I still have not figured out why the threadpool thread creation test
is locking up.
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r377324 | mjordan | 2012-12-06 08:26:13 -0600 (Thu, 06 Dec 2012) | 13 lines
Fix memory leak in 'manager show event' when command entered incorrectly
When the CLI command 'manager show event' was run incorrectly and its usage
instructions returned, a reference to the event container was leaked. This
would prevent the container from being reclaimed when Asterisk exits. We now
properly decrement the count on the ao2 object using the nifty RAII_VAR macro.
Thanks to Russell for helping me stumble on this, and Terry for writing that
ridiculously helpful macro.
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r377329 | russell | 2012-12-06 09:06:47 -0600 (Thu, 06 Dec 2012) | 7 lines
Add CLI tab completion to 'acl show'.
The 'acl show' CLI command allows you to show the details about a specific
named ACL in acl.conf. This patch adds tab completion to the command.
Review: https://reviewboard.asterisk.org/r/2230/
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r377330 | russell | 2012-12-06 09:13:37 -0600 (Thu, 06 Dec 2012) | 6 lines
Minor code cleanup in named_acl.c.
This patch makes a few little cleanups to named_acl.c. A couple non-public
functions were made static and an opening brace for a function was moved to
its own line, per the coding guidelines.
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After giving it some consideration, there's no real
use for zombie threads. Listeners can't really use the
current number of zombie threads as a way of gauging activity,
zombifying threads is just an extra step before they die that
really serves no purpose, and since there's no way to re-animate
zombies, the operation does not need to be around.
I also fixed up some miscellaneous compilation errors that
were lingering from some past revisions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since threadpool shutdown is very strictly controlled,
there is no need to be so precise with reference counts
in queued operations. Since the threadpool shuts down its
own control taskprocessor before doing anything else destructive,
it can be guaranteed that all queued tasks will have a valid
pointer to the pool. This meant that some destructor functions
for helper structs could be removed entirely.
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r377168 | rmudgett | 2012-12-03 17:00:08 -0600 (Mon, 03 Dec 2012) | 21 lines
Cleanup ast_run_atexits() atexits list.
* Convert atexits list to a mutex instead of a rd/wr lock. The lock is
only write locked.
* Move CLI verbose Asterisk ending message to where AMI message is output
in really_quit() to avoid further surprises about using stuff already
shutdown.
(issue ASTERISK-20649)
Reported by: Corey Farrell
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David M. Lee
Mark Michelson
Richard Mudgett
Joshua Colp
Jonathan Rose
Russell Bryant
Matthew Jordan
Walter Doekes
Automerge script