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/
........
Merged revisions 380348 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
........
Merged revisions 380210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 380211 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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)
........
Merged revisions 379760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379777 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 379790 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 379509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379510 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 379513 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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.
........
Merged revisions 379228 from http://svn.asterisk.org/svn/asterisk/branches/11
................
r379231 | rmudgett | 2013-01-16 11:49:52 -0600 (Wed, 16 Jan 2013) | 10 lines
chan_misdn: Fix compile error.
(issue ASTERISK-15456)
........
Merged revisions 379226 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379230 from http://svn.asterisk.org/svn/asterisk/branches/11
................
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.
........
Merged revisions 379232 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Clarify some documentation
* Change copyright date of taskprocessor files
* Address potential issue of creating taskprocessor with listener if
taskprocessor with that name exists already
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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
........
Merged revisions 379001 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379020 from http://svn.asterisk.org/svn/asterisk/branches/11
................
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/
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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/
........
Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........
Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378934 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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/
................
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
........
Merged revisions 378917 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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
........
Merged revisions 378776 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378780 from http://svn.asterisk.org/svn/asterisk/branches/11
................
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().
................
r378790 | rmudgett | 2013-01-09 15:14:39 -0600 (Wed, 09 Jan 2013) | 4 lines
* Whitespace changes.
* Made ast_test_init() match its prototype.
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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.
........
Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378457 from http://svn.asterisk.org/svn/asterisk/branches/11
................
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.
........
Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378459 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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/
........
Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378358 from http://svn.asterisk.org/svn/asterisk/branches/11
................
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)
........
Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378376 from http://svn.asterisk.org/svn/asterisk/branches/11
................
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)
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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)
........
Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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)
........
Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378287 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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.
................
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)
........
Merged revisions 378073 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
........
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)
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
........
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/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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
........
Merged revisions 377740 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377741 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377742 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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.
........
Merged revisions 377319 from http://svn.asterisk.org/svn/asterisk/branches/11
................
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/
................
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.
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
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
........
Merged revisions 377165 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377166 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377167 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
........
r376820 | pkiefer | 2012-11-29 10:44:42 -0600 (Thu, 29 Nov 2012) | 14 lines
Fix chan_sip websocket payload handling
Websocket by default doesn't return an ast_str for the payload received. When
converting it to an ast_str on chan_sip the last character was being omitted,
because ast_str functions expects that the given length includes the trailing
0x00. payload_len only has the actual string length without counting the
trailing zero.
For most cases this passed unnoticed as most of SIP messages ends with \r\n.
(closes issue ASTERISK-20745)
Reported by: I?\195?\177aki Baz Castillo
Review: https://reviewboard.asterisk.org/r/2219/
........
r376821 | dlee | 2012-11-29 11:16:50 -0600 (Thu, 29 Nov 2012) | 5 lines
Fixed ast_random's comment about locking.
The original comment was separated from the code at some point, and didn't
reflect the use of libc's other than glibc for Linux.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376791 | rmudgett | 2012-11-28 18:48:12 -0600 (Wed, 28 Nov 2012) | 32 lines
Add MALLOC_DEBUG atexit unreleased malloc memory summary.
* Adds the following CLI commands to control MALLOC_DEBUG reporting of
unreleased malloc memory when Asterisk is shut down.
memory atexit list on
memory atexit list off
memory atexit summary byline
memory atexit summary byfunc
memory atexit summary byfile
memory atexit summary off
* Made check all remaining allocated region blocks atexit for fence
violations.
* Increased the allocated region hash table size by about three times. It
still isn't large enough considering the number of malloced blocks
Asterisk uses.
* Made CLI "memory show allocations anomalies" use
regions_check_all_fences().
Review: https://reviewboard.asterisk.org/r/2196/
........
Merged revisions 376788 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376789 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376790 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376761 | rmudgett | 2012-11-28 18:07:55 -0600 (Wed, 28 Nov 2012) | 25 lines
Enhance MALLOC_DEBUG CLI commands.
* Fixed CLI "memory show allocations" misspelling of anomalies option.
The command will still accept the original misspelling.
* Miscellaneous tweaks to CLI "memory show allocations" command output
format.
* Made CLI "memory show summary" summarize by line number instead of by
function if a filename is given.
* Made CLI "memory show summary" sort its output by filename or
function-name/line-number depending upon request.
* Miscellaneous tweaks to CLI "memory show summary" command output format.
........
Merged revisions 376758 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376759 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376760 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376728 | jrose | 2012-11-28 10:47:44 -0600 (Wed, 28 Nov 2012) | 22 lines
manager: Make challenge work with allowmultiplelogin=no
Prior to this patch, challenge would yield a multiple logins error if used
without providing the username (which isn't really supposed to be an argument
to challenge) if allowmultiplelogin was set to no because allowmultiplelogin
finds a user with a zero length login name. This check is simply disabled for
the challenge action when the username is empty by this patch.
(closes issue ASTERISK-20677)
Reported by: Vladimir
Patches:
challenge_action_nomultiplelogin.diff uploaded by Jonathan Rose (license 6182)
........
Merged revisions 376725 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376726 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376727 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376691 | rmudgett | 2012-11-27 18:13:10 -0600 (Tue, 27 Nov 2012) | 39 lines
Fix extension matching with the '-' char.
The '-' char is supposed to be ignored by the dialplan extension matching.
Unfortunately, it's treatment is not handled consistently throughout the
extension matching code.
* Made the old exten matching code consistently ignore '-' chars.
* Made the old exten matching code consistently handle case in the
matching.
* Made ignore empty character sets.
* Fixed ast_extension_cmp() to return -1, 0, or 1 as documented. The only
user of it in pbx_lua.c was testing for -1. It was originally returning
the strcmp() value for less than which is not usually going to be -1.
* Fix character set sorting if the sets have the same number of characters
and start with the same character. Character set [0-9] now sorts before
[02-9a] as originally intended.
* Updated some extension label and priority already in use warnings to
also indicate if the extension is aliased.
(closes issue ASTERISK-19205)
Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2201/
........
Merged revisions 376688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376689 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376690 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376589 | mjordan | 2012-11-22 18:02:23 -0600 (Thu, 22 Nov 2012) | 29 lines
Re-initialize logmsgs mutex upon logger initialization to prevent lock errors
Similar to the patch that moved the fork earlier in the startup sequence to
prevent mutex errors in the recursive mutex surrounding the read/write thread
registration lock, this patch re-initializes the logmsgs mutex. Part of the
start up sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to daemon in order
to read startup parameters. When reading in a conf file, log statements can
be generated. Since this can't be avoided, the mutex instead is
re-initialized to ensure a reset of any thread tracking information.
This patch also includes some additional debugging to catch errors when
locking or unlocking the recursive mutex that surrounds locks when the
DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will
cause an abort() if a mutex error is detected.
(issue ASTERISK-19463)
Reported by: mjordan
Tesetd by: mjordan
........
Merged revisions 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376587 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376588 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
........
r376575 | rmudgett | 2012-11-21 12:33:16 -0600 (Wed, 21 Nov 2012) | 20 lines
Add red-black tree container type to astobj2.
* Add red-black tree container type.
* Add CLI command "astobj2 container dump <name>"
* Added ao2_container_dump() so the container could be dumped by other
modules for debugging purposes.
* Changed ao2_container_stats() so it can be used by other modules like
ao2_container_check() for debugging purposes.
* Updated the unit tests to check red-black tree containers.
(closes issue ASTERISK-19970)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2110/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376447 | mjordan | 2012-11-18 14:27:45 -0600 (Sun, 18 Nov 2012) | 55 lines
Reorder startup sequence to prevent lockups when process is sent to background
Although it is very rare and timing dependent, the potential exists for the
call to 'daemon' to cause what appears to be a deadlock in Asterisk during
startup. This can occur when a recursive mutex is obtained prior to the
daemon call executing. Since daemon uses fork to send the process into the
background, any threading primitives are unsafe to re-use after the call.
Implementations of pthread recursive mutexes are highly likely to store the
thread identifier of the thread that previously obtained the mutex. If
the mutex was locked prior to the fork, a subsequent unlock operation will
potentially fail as the thread identifier is no longer valid. Since the
mutex is still locked, all subsequent attempts to grab the mutex by other
threads will block.
This behavior exhibited itself most often when DEBUG_THREADS was enabled, as
this compile time option surrounds the mutexes in Asterisk with another
recursive mutex that protects the storage of thread related information. This
made it much more likely that a recursive mutex would be obtained prior to
daemon and unlocked after the call.
This patch does the following:
a) It backports a patch from Asterisk 11 that prevents the spawning of the
localtime monitoring thread. This thread is now spawned after Asterisk has
fully booted.
b) It re-orders the startup sequence to call daemon earlier during Asterisk
startup. This limits the potential of threading primitives being accessed
by initialization calls before daemon is called.
c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called.
Developers should send error messages directly to stderr prior to daemon,
as calls to ast_log may access recursive mutexes that store thread related
information.
d) It reorganizes when thread local storage is created for storing lock
information during the creation of threads. Prior to this patch, the
read/write lock protecting the list of threads in ast_register_thread would
utilize the lock in the thread local storage prior to it being initialized;
this patch prevents that.
On a very related note, this patch will *greatly* improve the stability of the
Asterisk Test Suite.
Review: https://reviewboard.asterisk.org/r/2197
(closes issue ASTERISK-19463)
Reported by: mjordan
Tested by: mjordan
........
Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This helps account for the fact that it is unknown just
how many references may exist for a given taskprocessor
listener, so simply unreffing it from the taskprocessor
shutdown function is not enough to convey the gravity
of the situation.
By putting in a shutdown callback, it now becomes clear
to the listener not to try to do any further operations
on the taskprocessor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376291 | beagles | 2012-11-15 08:35:01 -0600 (Thu, 15 Nov 2012) | 14 lines
Patch to prevent stopping the active generator when it is not the silence
generator.
This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.
(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
........
Merged revisions 376217 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376148 | elguero | 2012-11-12 14:18:47 -0600 (Mon, 12 Nov 2012) | 26 lines
Fix Dynamic Hints Variable Substition - Underscore Problem
When adding a dynamic hint, if an extension contains an underscore no variable
subsitution is being performed.
This patch changes from checking if the extension contains an underscore to
checking if the extension begins with an underscore.
(closes issue ASTERISK-20639)
Reported by: Steven T. Wheeler
Tested by: Steven T. Wheeler, Michael L. Young
Patches:
asterisk-20639-dynamic-hint-underscore.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2188/
........
Merged revisions 376142 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376143 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376144 from http://svn.asterisk.org/svn/asterisk/branches/11
................
r376169 | file | 2012-11-12 14:46:51 -0600 (Mon, 12 Nov 2012) | 20 lines
Properly check if the "Context" and "Extension" headers are empty in a ShowDialPlan action.
The code which handles the ShowDialPlan action wrongly assumed that a non-NULL return value
from the function which retrieves headers from an action indicates that the header has a
value. This is incorrect and the contents must be checked to see if they are blank.
(closes issue ASTERISK-20628)
Reported by: jkroon
Patches:
asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
........
Merged revisions 376166 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376167 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376168 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This includes changing the taskprocessor to use its builtin
ao2_lock instead of having a separate mutex. It can do this
now since there is no longer an ast_cond_t associated with
the taskprocessor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Taskprocessors are now divided into two units: the task queue
and their listeners.
When a task is added to the queue, the listener is notified and
can take whatever action is desired. This means that taskprocessors
are no longer confined to having their tasks executed within a
single thread.
A default taskprocessor listener has been added that mirrors the
old taskprocessor behavior.
I've tested it by running Asterisk and placing calls. It appears
to work as expected. I'm going to do some cleaning up first and
then write some unit tests to be sure everything works as expected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
................
r376049 | rmudgett | 2012-11-08 11:38:31 -0600 (Thu, 08 Nov 2012) | 41 lines
Add MALLOC_DEBUG enhancements.
* Makes malloc() behave like calloc(). It will return a memory block
filled with 0x55. A nonzero value.
* Makes free() fill the released memory block and boundary fence's with
0xdeaddead. Any pointer use after free is going to have a pointer
pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid
memory address so a crash is expected.
* Puts the freed memory block into a circular array so it is not reused
immediately.
* When the circular array rotates out a memory block to the heap it checks
that the memory has not been altered from 0xdeaddead.
* Made the astmm_log message wording better.
* Made crash if the DO_CRASH menuselect option is enabled and something is
found.
* Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms.
* Extracted region_check_fences() from __ast_free_region() and
handle_memory_show().
* Updated handle_memory_show() CLI usage help.
Review: https://reviewboard.asterisk.org/r/2182/
........
Merged revisions 376029 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376030 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376048 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
........
r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
........
Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
........
Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375895 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Manager's tcp/tls objects have a periodic function that purge old manager
sessions periodically. During shutdown, the underlying container holding
those sessions can be disposed of and set to NULL before the tcp/tls periodic
function is stopped. If the periodic function fires, it will attempt to
iterate over a NULL container.
This patch checks for whether or not the sessions container exists before
attempting to purge sessions out of it. If the sessions container is NULL,
we simply return.
Note that this error was also caught by the Asterisk Test Suite.
........
Merged revisions 375800 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375801 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375802 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AstDB uses prepared SQLite3 statements to retrieve data from the SQLite3
database. These statements should be finalized during Asterisk shutdown so
that the SQLite3 database can be properly closed. Failure to finalize the
statements results in a memory leak and a failure when closing the database.
This patch fixes those issues by ensuring that all prepared statements are
properly finalized at shutdown.
(closes issue ASTERISK-20647)
Reported by: Corey Farrell
patches:
astdb-sqlite3_close.patch uploaded by Corey Farrell (license 5909)
........
Merged revisions 375761 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375763 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two memory leaks:
1) When building XML documentation items, the 'name' attribute was extracted
from XML elements but not properly freed after being copied into the item
being built.
2) When unloading XML documentation, the doctree container objects were not
properly freed.
This patch corrects these memory leaks. Note that this patch was modified
slightly for this commmit, as the case where the 'name' attribute doesn't
exist also wasn't handled in the item construction. This patch also checks
for that attribute not existing.
(closes issue ASTERISK-20648)
Reported by: Corey Farrell
Tested by: mjordan
patches:
xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)
........
Merged revisions 375756 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Asterisk Test Suite caught an error condition where a scheduled CDR batch
write can be deleted twice if two channels attempt to post their CDRs at the
same time. The batch CDR mutex is locked while the CDRs are appended to the
current batch list; however, it is unlocked prior to actually scheduling the
CDR write. As such, two threads can attempt to remove the currently scheduled
batch write at the same time, resulting in an assertion error.
This patch extends the time that the mutex is locked to encompass actually
scheduling the write. This prevents two threads from unscheduling the
currently scheduled write at the same time.
........
Merged revisions 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375728 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375729 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing. Update title that was left behind many years ago.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Upon further examination, this code was causing compliation problems on
CentOS at the least (possibly on any machine without curses) and also
the local value of COLS is used even with a remote console, so it is
less than ideal.
(issue ASTERISK-20396)
Reported by: Johan Wilfer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Remove the "RTP Read too short" warning for RTP keepalives. Remove the
the warning about the application delimiter switch from pipe to comma.
(You should've done this by now.) Make cdr_odbc report more when an
insert fails. Make chan_sip warn less when the peer wants SRTP (and we
don't) or sends a zero port to disable a media type.
Review: https://reviewboard.asterisk.org/r/2167
(closes issue ASTERISK-20538)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
manager show commands now shows the full name of the command being displayed
regardless of size. The privilege column has also been removed from this
display. It will also now use the full length of the terminal if curses is
available. Manager show command will now always display the privilege of
the manager command within the CLI.
(closes ASTERISK-20396)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/2143/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
........
Merged revisions 375025 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 375026 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 375027 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update title that was left behind many years ago. Used revision 6596 as my guide for what it should be.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add app_skel.c as an example in app.c and fix some formating for the "Dial Privacy scripts" so it actually shows up in the Doxygen output.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is used to solve an issue where a poll on a file
descriptor does not necessarily correspond to the readiness
of a FILE handle to be read.
This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead.
Because TCP does not guarantee that an entire message or even
just one single message will arrive during a read, a loop has
been introduced to ensure that we only attempt to handle a
single message at a time. The tcptls_session_instance structure
has also had an overflow buffer added to it so that if more
than one TCP message arrives in one go, there is a place to
throw the excess.
Huge thanks goes out to Walter Doekes for doing extensive review
on this change and finding edge cases where code could fail.
(closes issue ASTERISK-20212)
reported by Phil Ciccone
Review: https://reviewboard.asterisk.org/r/2123
........
Merged revisions 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 374906 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374914 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During testing, it was discovered that having chan_sip
export global symbols was problematic.
The biggest problem was that load order was affected.
Trying to use realtime could be problematic since in
all likelihood the necessary realtime driver(s) would
not be loaded before chan_sip.
In addition, it was found that it was impossible to
use the Digium Phone Module for Asterisk since it
must be loaded before chan_sip since it must hook
into chan_sip's configuration parsing.
The solution is to use a virtual table in the same
manner that other modules in Asterisk do, like
app_voicemail.
(closes issue ASTERISK-20545)
Reported by: kmoore
........
Merged revisions 374842 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Similar to r369351, the billing duration can be skewed when batch mode is
enabled. This happened much more rarely than the duration, as it only
occured when the call was answered (thereby indicating an actual answer
time) and immediately hung up on (indicating a billsec of 0). Since
a billing time of '0' can either mean that the call immediately ended
or that the CDR was improperly answered, we have to use additional information
to know whether or not we can trust the CDR billsec value. Prior to this
patch, we looked to see if we had a valid answer time. If we did, and
billsec was zero, we used the current time to calculate what billsec value
we could from the CDR being written. If batch mode is enabled, this will
incorrectly report a billsec value being much greater than the actual
duration of the call.
Instead of relying on the presence of an answer time to know whether or not
we can re-calculate the billsec for the CDR, we now also use the presence
of the CDR's end time to know if we need to re-calculate or whether we can
trust the billsec value that we have. This prevents erroneous jumps in the
billsec value, while still making sure that in the worst case, some billing
time will be calculated.
(closes issue AST-1016)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
........
Merged revisions 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 374844 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374845 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to usage of ast_hook_send_action, AMI action handling code should
be able to handle a NULL mansession->session. This would cause a crash
on NULL dereference if action_originate was called from
ast_hook_send_action.
(closes issue ASTERISK-20544)
........
Merged revisions 374792 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes trivial build error on Solaris:
acl.c: In function `get_local_address':
acl.c:196: error: `best_score' undeclared (first use in this function)
acl.c:196: error: (Each undeclared identifier is reported only once
acl.c:196: error: for each function it appears in.)
make[2]: *** [acl.o] Error 1
(issue ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang
patches:
0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch by Shaun Ruffell (license 5417)
........
Merged revisions 374632 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using the channel technology agnostic application/AMI command MessageSend,
the "From" field is technically optional for the SIP channel driver. However,
if being sent by the XMPP resource module (either res_xmpp or res_jabber), the
"From" field is necessary, and must correspond to a defined account. This
patch updates the documentation for this application/AMI command to reflect
this.
(closes issue ASTERISK-20405)
Reported by: Leif Madsen
........
Merged revisions 374611 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines
Improve AMI long line error handling
In AMI's parser, when it receives a long line (> 1024 characters), it discards
that line, but continues to process the message normally.
Typically, this is not a problem because a) who has lines that long and b)
usually a discarded line results in an invalid message. But if that line is
specifying an optional field, then the message will be processed, you get a
'Response: Success', but things don't work the way you expected them to.
This patch changes the behavior when a line-too-long parse error occurs.
* Changes the log message to avoid way-too-long (and truncated anyways) log
messages
* Adds a 'parsing' status flag to Response: Success
* Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long
* Responds with an appropriate error if parsing != MESSAGE_OKAY
(closes issue AST-961)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2142/
........
r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
I've committed too much. Reverting part of r374570.
........
Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI DBDelTree command will return Success/Key tree deleted successfully even
if the given key does not exist. The CLI command 'database deltree' had a
similar problem, but was saved because it actually responded with '0 database
entries removed'. AGI had a slightly different error, where it would return
success if the database was unavailable.
This came from confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted (including 0 for
deleting nothing).
* Changed some poorly named res variables to num_deleted
* Specified specific errors when calling ast_db_deltree (database unavailable
vs. entry not found vs. success)
* Fixed similar bug in AGI database deltree, where 'Database unavailable'
results in successful result
(closes issue AST-967)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2138/
........
Merged revisions 374426 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 374427 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374428 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries.
Various countries have different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies.
Power level difference between frequencies for different Administrations/RPOAs
NTT = Max. 5 dB
AT&T = 4dB(reverse) to 8dB(normal)
Danish = Max. 6 dB
Australian = Max. 10 dB
Brazilian = Max. 9 dB
ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03)
Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB.
Default is AT&T specifications
Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31
;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31
;relax_dtmf_reverse_twist=3.98
(closes issue ASTERISK-20442)
Reported by: tbsky
Tested by: tbsky,alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2141/
........
Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 374385 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374386 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For each item in core_instances disposed of in the shutdown of ccss, any
generic monitor instances referenced by the objects will be removed from
generic_monitors during their destruction. Hilarity ensues if
generic_monitors no longer exists.
Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly
when it ran into this.
........
Merged revisions 374300 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374301 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard pointed out two problems with the check-in from r374177:
* The ast_msg_shutdown function declaration doesn't match the prototype
in main/message.c.
* The ref/alloc function usage in astobj2 (in trunk) can use the ao2_t_*
variants of the functions to allow the REF_DEBUG flag to enable/disable
their debug counterparts.
........
Merged revisions 374210 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374211 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Greenlight in #asterisk brought up that he was receiving an error message "Could
not create persistent member string, out of space" when running app_queue in
Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to
store the generated string, but with queues that have large member lists this is
not always the case. This patch removes the limitation and uses ast_str instead
of a fixed sized buffer.
The complicating factor comes from the fact that ast_db_get requires a buffer
and buffer size argument, which doesn't let us pull back more than what we pass
in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d
copy of the value from astdb.
As an aside, I did some testing on the maximum size of data that we can store in
the BDB library we distribute and was able to store a 10MB string and retrieve
it with no problems, so I feel this is a safe patch.
Review: https://reviewboard.asterisk.org/r/2136/
........
Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 374135 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374150 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CLI "core show help" output leaves something to be desired.
1) The command is truncated to a maximum of 30 characters.
2) The output columns are mirrored from the 31st column.
Current output format:
logger mute Toggle logging output to a console
logger reload Reopens the log files
logger rotate Rotates and reopens the log files
logger set level {DEBUG|NOTICE Enables/Disables a specific logging level for this console
logger show channels List configured log channels
New format:
logger mute -- Toggle logging output to a console
logger reload -- Reopens the log files
logger rotate -- Rotates and reopens the log files
logger set level {DEBUG|NOTICE|WARNING|ERROR|VERBOSE|DTMF} {on|off} -- Enables/Disables a specific logging level for this console
logger show channels -- List configured log channels
Review: https://reviewboard.asterisk.org/r/2133/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The following dialplan applications now recognize 'W' to pause sending
DTMF for one second in addition to the previously existing 'w' that paused
sending DTMF for half a second. Dial, ExternalIVR, and SendDTMF.
* The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now
distinguishes between 'w' and 'W'. The 'w' pauses dialing for half a
second. The 'W' pauses dialing for one second.
* Created dahdi_dial_str() in chan_dahdi that eliminated a lot of
duplicated dialing code and diagnostic messages for the channel driver.
(closes issue ASTERISK-20039)
Reported by: Jeremiah Gowdy
Patches:
jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy
Expanded patch to add support in chan_dahdi.
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made ast_dtmf_stream() wait after starting the silence generator rather
than before.
* Made ast_dtmf_stream() put the peer in autoservice for the whole time
things are being done to the chan.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an Asterisk module specifies a dependency in ast_module_info.nonoptreq, it
is possible for Asterisk to skip calling the modules's .load function.
Asterisk was loading and linking the module via load_dynamic_module() but was
not adding the module to the resource_heap. Therefore the module was not
initialized based on it's priority along with the other modules in the heap.
Now use load_resource() instead of load_dynamic_module() for non-optional
requirement. This will add the module to the resource_heap so the module can
be properly initialized in the correct order.
This is required if there are any module global data structures initialized in
the .load() callback for the module on platforms which do not support weak
references.
(issue ASTERISK-20439)
Reported by: sruffell
Patches:
0001-loader-Ensure-dependent-modules-are-properly-initial.patch uploaded by sruffell (license 5417)
........
Merged revisions 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 373910 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 373911 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.
The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.
(closes issue AST-942)
reported by Malcolm Davenport
(closes issue AST-943)
reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/2101
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen.
Further updates coming.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
janitor cleanup. No functional change.
1). ast_dsp_call_progress: use 'short samp' instead of s[x] inside loop.
apply same casting as other _init, dsp->energy = (int32_t) samp * (int32_t) samp
2). ast_dtmf_detect_init: move repeated setting of s->energy to outside of loop.
do goertzel_init loop first before setting s->lasthit and s->current_hit, consistant with ast_dsp_digitreset()
3). ast_mf_detect_init:
do goertzel_init loop first before setting s->hits[] and s->current_hit, consistant with ast_dsp_digitreset()
4). Don't chain init different variables, as the type may change
Review https://reviewboard.asterisk.org/r/2107/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.
Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.
Review: https://reviewboard.asterisk.org/r/2113/
........
Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* ASTERISK-20383
Missing named call pickup group features:
CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.
* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up. In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.
Regression because of the named call pickup group feature.
* See ASTERISK-20386 for the implementation improvements. These are the
changes in channel.c and channel.h.
* Fixed some locking issues in CHANNEL().
(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2112/
........
Merged revisions 373220 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple re-initializations) of
portions of the SSL library. Asterisk calls SSL_library_init and
SSL_load_error_strings during SSL initialization; collectively this
obviates the need for calling any of the following during initialization
or client connection handling:
* ERR_load_crypto_strings (handled by SSL_load_error_strings)
* OpenSSL_add_all_algorithms (synonym for SSL_library_init)
* SSLeay_add_ssl_algorithms (synonym for SSL_library_init)
2) Failure to completely clean up all memory allocated by Asterisk and by
the SSL library for TLS clients. This included not freeing the SSL_CTX
object in the SIP channel driver, as well as not clearing the error
stack when the TLS client exited.
Note that these memory leaks were found by Thomas Arimont, and this patch
was essentially written by him with some minor tweaks.
(closes issue AST-889)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
patches:
(bugAST-889.patch) by Thomas Arimont (license 5525)
Review: https://reviewboard.asterisk.org/r/2105
........
Merged revisions 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 373062 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 373079 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new API allows for sorted containers, insertion options, duplicate
handling options, and traversal order options.
* Adds the ability for containers to be sorted when they are created.
* Adds container creation options to handle duplicates when they are
inserted.
* Adds container creation option to insert objects at the beginning or end
of the container traversal order.
* Adds OBJ_PARTIAL_KEY to allow searching with a partial key. The partial
key works similarly to the OBJ_KEY flag. (The real search speed
improvement with this flag will come when red-black trees are added.)
* Adds container traversal and iteration order options: Ascending and
Descending.
* Adds an AST_DEVMODE compile feature to check the stats and integrity of
registered containers using the CLI "astobj2 container stats <name>" and
"astobj2 container check <name>". The channels container is normally
registered since it is one of the most important containers in the system.
* Adds ao2_iterator_restart() to allow iteration to be restarted from the
beginning.
* Changes the generic container object to have a v_method table pointer to
support other types of containers.
* Changes the container nodes holding objects to be ref counted.
The ref counted nodes and v_method table pointer changes pave the way to
allow other types of containers.
* Includes a large astobj2 unit test enhancement that tests the new
features.
(closes issue ASTERISK-19969)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/2078/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this option in use, it may be necessary to regulate your log files
externally.
(closes issue ASTERISK-20189)
Reported by: Jaco Kroon
Patches:
asterisk-logger-norotate-trunk.patch uploaded by Jaco Kroon (license 5671)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
message.c makes use of a special message queue channel that exists
in thread storage. This channel never goes away due to the fact that
the taskprocessor used by message.c does not get shut down, meaning
that it never ends the thread that stores the channel.
This patch fixes the problem by shutting down the taskprocessor when
Asterisk is shut down. In addition, the thread storage has a destructor
that will release the channel reference when the taskprocessor is destroyed.
(closes issue AST-937)
Reported by Jason Parker
Patches:
AST-937.patch uploaded by Mark Michelson (License #5049)
Tested by Jason Parker
........
Merged revisions 372885 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 372888 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous fix still would look in the static_RTP_PT table, which
is inappropriate since we specifically want to find a codec that has
been negotiated.
(closes issue ASTERISK-20296)
reported by NITESH BANSAL
Patches:
codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
........
Merged revisions 372311 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI Originate action can allow a remote user to specify information that can
be used to execute shell commands on the system hosting Asterisk. This can
result in an unwanted escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to perform actions
that would typically require the "system" class authorization. Previous attempts
to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought
to do so by inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched a predefined
set of values, rejecting the command if the user lacked the "system" class
authorization. As noted by IBM X-Force Research, the "ExternalIVR"
application is not listed in the predefined set of values. The solution for
this particular vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class authorization.
Unfortunately, the approach of inspecting fields in the Originate action against
known applications/functions has a significant flaw. The predefined set of
values can be bypassed by creative use of the Originate action or by certain
dialplan configurations, which is beyond the ability of Asterisk to analyze at
run-time. Attempting to work around these scenarios would result in severely
restricting the applications or functions and prevent their usage for legitimate
means. As such, any additional security vulnerabilities, where an
application/function that would normally require the "system" class
authorization can be executed by users with the "originate" class authorization,
will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has
been updated to reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper system
configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132)
Reported by: Zubair Ashraf of IBM X-Force Research
........
Merged revisions 371998 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 371999 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 372000 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes numerous doxygen warnings across Asterisk. It also updates
the makefile to regenerate the doxygen configuration on the local system
before running doxygen to help prevent warnings/errors on the local system.
Much thanks to Andrew for tackling one of the Asterisk janitor projects!
(issue ASTERISK-20259)
Reported by: Andrew Latham
Patches:
doxygen_partial.diff uploaded by Andrew Latham (license 5985)
make_progdocs.diff uploaded by Andrew Latham (license 5985)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original implementations simply wrap pthread functions, which take
absolute time as an argument. The spinlock version for systems without
those functions treated the argument as a delta. This patch fixes the
spinlock version to be consistent with the pthread version.
(closes issue ASTERISK-20240)
Reported by: Egor Gorlin
Patches:
lock.c.patch uploaded by Egor Gorlin (license 6416)
........
Merged revisions 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 371720 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With the SCOPED_LOCK macro, you can create a variable
that locks a specific lock and unlocks the lock when the
variable goes out of scope. This is useful for situations
where many breaks, continues, returns, or other interruptions
would require separate unlock statements. With a scoped lock,
these aren't necessary.
There are specializations for mutexes, read locks, write locks,
ao2 locks, ao2 read locks, ao2 write locks, and channel locks.
Each of these is a SCOPED_LOCK at heart though.
Review: https://reviewboard.asterisk.org/r/2060
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.
(issue ASTERISK-20090)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When formatting documentation fields, the XML documentation parser calls
xmldoc_get_formatted. This function allocates a string buffer at the
beginning of its routine. Unfortunately, on certain code paths, it also
calls xmldoc_string_cleanup, which assumes that it will create the string
buffer. The previously allocated string buffer is then leaked by the
xmldoc_string_cleanup routine.
Now: we don't do that.
(closes issue AST-932)
Reported by: Alexander Homig
........
Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 371491 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 371492 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the feature "Private representation of caller, connected
and redirecting party ids", as previously discussed with us (DATUS) and
Digium.
1. Feature motivation
Until now it is quite difficult to modify a party number or name which can
only be seen by exactly one particular instantiated technology channel
subscriber. One example where a modified party number or name on one
channel is spread over several channels are supplementary services like
call transfer or pickup. To implement these features Asterisk internally
copies caller and connected ids from one channel to another. Another
example are extension subscriptions. The monitoring entities (watchers)
are notified of state changes and - if desired - of party numbers or names
which represent the involving call parties. One major feature where a
private representation of party names is essentially needed, i.e. where a
party name shall be exclusively signaled to only one particular user, is a
private user-specific name resolution for party numbers. A lookup in a
private destination-dependent telephone book shall provide party names
which cannot be seen by any other user at any time.
2. Feature Description
This feature comes along with the implementation of additional private
party id elements for caller id, connected id and redirecting ids inside
Asterisk channels.
The private party id elements can be read or set by the user using
Asterisk dialplan functions.
When a technology channel is initiating a call, receives an internal
connected-line update event, or receives an internal redirecting update
event, it merges the corresponding public id with the private id to create
an effective party id. The effective party id is then used for protocol
signaling.
The channel technologies which initially support the private id
representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and
PRI (chan_dahdi).
Once a private name or number on a channel is set and (implicitly) made
valid, it is generally used for any further protocol signaling until it is
rewritten or invalidated.
To simplify the invalidation of private ids all internally generated
connected/redirecting update events and also all connected/redirecting
update events which are generated by technology channels -- receiving
regarding protocol information - automatically trigger the invalidation of
private ids.
If not using the private party id representation feature at all, i.e. if
using only the 'regular' caller-id, connected and redirecting related
functions, the current characteristic of Asterisk is not affected by the
new extended functionality.
3. User interface Description
To grant access to the private name and number representation from the
Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan
functions are extended by the following data types. The formats of these
data types are equal to the corresponding regular 'non-private' already
existing data types:
CALLERID:
priv-all
priv-name priv-name-valid priv-name-charset priv-name-pres
priv-num priv-num-valid priv-num-plan priv-num-pres
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
CONNECTEDLINE:
priv-name priv-name-valid priv-name-pres priv-name-charset
priv-num priv-num-valid priv-num-pres priv-num-plan
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
REDIRECTING:
priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset
priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd
priv-orig-tag
priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset
priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan
priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd
priv-from-tag
priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset
priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan
priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
priv-to-tag
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2030/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
You cannot unref a pointer and then expect to ref it again later.
* Fix potential NULL pointer deref if the call pickup search fails.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Quote from review board:
This patch extends the extension state callbacks so that monitoring channels
(as chan_sip) get more information of the devices which are responsible for
an extension state change. The additional information is needed by chan_sip
to present names/numbers of the caller and callee in an early-state SIP
notification. Users of extenstion state callback not interested in the
additional information are not affected by the changes.
Motivation: to present the involved party's name/number in an early-state
nofification (used by the notified device as a pickup offer) one after another
so that a user can see which call he will pick up in an undirected pickup.
Such a pickup offer to a user shall indicate the same call (number/name-A calls
number/name-B) as the call which would be picked up when an undirected pickup
is executed.
Users interested in additional state info must use the new functions
ast_extension_state_add_extended() resp.
ast_extension_state_add_destroy_extended() to register an extended state
callback. When the callback is registered this way, an extra member
device_state_info of struct ast_state_cb_info is passed to the callback in
addition to the aggregated extension state. This container holds an object for
every device of the monitored extension hint consisting of the device name, the
device state and a channel reference to the channel which (presumably) caused
the device state.
The information is used by chan_sip for early-state notifications. When the
state of a device changes and the new state contains AST_EVENT_RINGING, an
early-state notification is sent to the subscribed devices with the
caller/callee names/numbers of the oldest ringing channel of the monitored
extension. The notified user may then invoke a direct pickup, which will pickup
exactly this channel.
Users of the old non-extended callbacks will only be called when the aggregated
state did change (same behavior as before). Users of the extended callback will
also be called when the state is unchanged but does contain AST_EVENT_RINGING.
That could be the case if two channels are ringing at one device and one of
them hangs up, so the aggregated state does not change. This way the monitoring
channel can create a new early-state notification with the now ringing
party-ids.
Review: https://reviewboard.asterisk.org/r/2048
This contribution comes from Guenther Kelleter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is based on the work done by Olle Johansson on review board.
The idea is that the channel specified in an AMI originate or call
file is typically not connected to the outgoing extension until the
channel has been answered. With this change, an EarlyMedia header can
be specified for AMI originates and an early_media option can
be specified in call files. With this option set, once early media is
received on a channel, it will be connected with the outgoing extension.
(closes issue ASTERISK-18644)
Reported by Olle Johansson
Review: https://reviewboard.asterisk.org/r/1472
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Implementation of a dialplan function for checking manager accounts. Right now
it only returns the number of logged in sessions for a manager account, but
other attributes can be added later.
Patch by: Olle Johansson
Review: https://reviewboard.asterisk.org/r/421/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation. However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.
Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup". Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.
Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.
Review: https://reviewboard.asterisk.org/r/2043
Uploaded by:
Guenther Kelleter(license #6372)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this option set, channel variables can be set on
every manager originate. The Variable header can still
be used to set additional channel variables for individual
calls if desired.
This work was completed by Olle Johansson on review board.
I have applied the review feedback and am committing it in
order to get this into trunk before Asterisk 11 is branched.
Review: https://reviewboard.asterisk.org/r/1412
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, updating the device state cache was done by the thread
that originated the event. It would update the cache and then queue the event
up for another thread to dispatch. This thread moves the cache updating part
to be in the same thread as event dispatching.
I was working with someone on a heavily loaded Asterisk system and while
reviewing backtraces of the system while it was having problems, I noticed that
there were a lot of threads contending for the lock on the event cache. By
simply moving this into a single thread, this helped performance *a lot* and
alleviated some deadlock-like symptoms.
Review: https://reviewboard.asterisk.org/r/2066/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new CLI command to the res_corosync module. It is primarily
used as a debugging tool. It lets you fire off an event which will cause
res_corosync on other nodes in the cluster to place messages into the logger if
everything is working ok. It verifies that the corosync communication is
working as expected.
I didn't put anything in the CHANGES file for this, because this module is new
in Asterisk 11. There is already a generic "res_corosync new module" entry in
there so I figure that covers it just fine.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous change to the build system for using a system-provided editline
library was missing a crucial include directory for building against the
copy of the library in the Asterisk source tree.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes the build system to refer to the embedded editline directory
using an absolute path, which will resolve a problem seen on the CentOS
automated build agents.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes the Asterisk configure script and build system to detect
the presence of the NetBSD editline library (libedit) on the system. If it is
found, it will be used in preference to the version included in the Asterisk
source tree.
(closes issue ASTERISK-18725)
Reported by: Jeffrey C. Ollie
Review: https://reviewboard.asterisk.org/r/1528/
Patches:
0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) There is no such function as ast_ref()
2) The patch was originally credited as the one uploaded by Guenther
Kelleter (license 6372) via issue AST-921, but the patch committed
was not the patch referenced on the issue.
3) Guenther Kelleter's patch was actually correct. It moved the
ast_free above the presencechange_cleanup label. I am not
committing his change as it is not technically necesary--calling
ast_free(NULL) is perfectly safe and I worry that moving the
ast_free outside of the label could lead to future bugs if
someone ever adds another failure conditional and expects
'goto presencechange_cleanup;' to clean up after everything.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
items (separated by commas), and items in the rule can be negated by prefixing
them with '!'. This simplifies Asterisk Realtime configurations, since it is no
longer necessray to control the order that the 'permit' and 'deny' columns are
returned from queries.
Review: https://reviewboard.asterisk.org/r/1592/
Initial patch contributed by Tilghman Lesher
Unit tests written by Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes the following:
* Unit tests for the abstract Jitter Buffer API. This includes both fixed
and adaptive flavors, testing nominal creation, frame input, frame retrieval,
resyncing; off nominal frame input overflow, out of order, and others.
* Tweaks to the abstract_jb API to remove the unnecessary resync_threshold
parameter from the create function (resync_threshold is already in the
struct passed into the create function)
* Ensure the fixed jitter buffer is empty before destroying it, to avoid an
ASSERT
* Don't "resync" the adaptive jitter buffer. The mechanism that was being
used actually causes the jitter buffer to think its being overflowed by going
around the jitterbuf API and attempting to 'resynch' it improperly. If a
resync is needed, the jitter buffer will do it properly by itself. Note that
this is only an optimization needed for trunk, as the worst that happens is
the loss of three voice packets before the adaptive jitter buffer will resync
anyway.
Review: https://reviewboard.asterisk.org/r/2035
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The value "none" is specified in the config file as a valid value for
the "video_mode" option. The code prior to the ACO conversion did not
check for "none", but just ignored it and relied on the default zero
value. The parsing with ACO is more strict, so without handling
"none" specifically, parsing would fail.
When parsing failed, but the module loaded anyway, the config info
would never be stored, and one place in the code did not check for
this case and would segfault. It was also possible that the
aco_info struct's internals would be destroyed and used as well.
This patch keeps the module from loading after parse failures, adds
the "none" option to "video_mode", registers CLI functions only
after parsing has completed, checks the config data for NULL before
accessing it, and returns -1 on some allocation failures when
initializing.
(closes issue ASTERISK-20159)
Reported by: Birger "WIMPy" Harzenetter
Tested by: Birger "WIMPy" Harzenetter
Patches:
confbridge_fix3.txt uploaded by Terry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.
(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It's harder to correlate the Newchannel and Hangup AMI events without
specifying "AccountCode" in both.
(closes issue ASTERISK-19963)
Reported by: Oleg A. Arkhangelsky
Patches:
hangup_acctcode.diff (license #6397) patch uploaded by Oleg A. Arkhangelsky
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A number of applications/AMI commands in Asterisk have specific behavioral
differences depending on the resource or channel technology those
applications are executed on. For example, the MessageSend application/
command is technology agnostic, but how the channel drivers that support
that functionality behave is dependant on the protocols and channel
driver implementation. Prior to this patch, those details were either
documented in the application/command documentation itself, or were left
undocumented.
This patch adds a new element to the documentation schema, <info/>. An info
node is essentially a piece of technology specific reference information that
can be included by any top level XML documentation node. For example, the
MessageSend application can now include XMPP/SIP specific information, where
that technology specific information can be defined in chan_motif/res_xmpp/
chan_sip. Likewise, that information can also be included in the MessageSend
AMI command.
Review: https://reviewboard.asterisk.org/r/2049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Jonathan Rose
Joshua Colp
Russell Bryant
Matthew Jordan
Walter Doekes
David M. Lee
Mark Michelson
Automerge script
Andrew Latham
Kinsey Moore
Alec L Davis
Sean Bright
Brent Eagles
Terry Wilson
Kevin P. Fleming