When a module's configuration is not loadable, we still load the module but it
is not in a running state. When trying to troubleshoot, let's say, why
chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a
loaded module is not currently running.
(closes issue ASTERISK-21108)
Reported by: Rusty Newton
Tested by: Michael L. Young
Patches:
asterisk-21108_add_status-v2.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2331/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does two things:
1. It disables (temporarily) strict XML documentation checking for module
configurations. We should re-enable it before making any release from
trunk.
2. Pass the module flag AST_MODULE through sorcery. This means several of the
API calls are now macros and will do this automatically for you. The config
framework needs the module that objects are registering to so it can
properly construct the documentation. (This was already a required field,
but sorcery was getting by without it)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While autoservice is running and servicing a channel the callid is being stored
and removed in the thread's local storage for each iteration of the thread loop.
If debug was set to a sufficient level the log file would be spammed with callid
thread local storage debug messages.
Added a new function that checks to see if the callid to be stored is different
than what is already contained (if anything). If it is different then
store/replace and log, otherwise just leave as is. Also made it so all logging
of debug messages pertaining to the callid thread storage outputs only when
TEST_FRAMEWORK is defined.
(issue ASTERISK-21014)
(closes issue ASTERISK-21014)
Report by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2324/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:
1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
options that use the configuration framework
This patch include configuration documentation for the following modules:
* chan_motif
* res_xmpp
* app_confbridge
* app_skel
* udptl
Two new CLI commands have been added:
* config show help - show configuration help by module, category, and item
* xmldoc dump - dump the in-memory representation of the XML documentation to
a new XML file.
Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058
patches:
on review 2058 uploaded by twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The core module related to coloring terminal output was old and needed
some love. The main thing here was an attempt to get rid of the
obscene number of stack-local buffers that were allocated for no other
reason than to colorize some output. Instead, this uses a simple trick
to allocate several buffers within threadlocal storage, then
automatically rotates between them, so that you can make multiple calls
to the colorization routine within one function and not need to
allocate multiple buffers.
Review: https://reviewboard.asterisk.org/r/2241/
Patches:
bug.patch uploaded by Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability to create a serializer from a thread pool. A
serializer is a ast_taskprocessor with the same contract as a default
taskprocessor (tasks execute serially) except instead of executing out
of a dedicated thread, execution occurs in a thread from a
ast_threadpool. Think of it as a lightweight thread.
While it guarantees that each task will complete before executing the
next, there is no guarantee as to which thread from the pool individual
tasks will execute. This normally only matters if your code relys on
thread specific information, such as thread locals.
This patch also fixes a bug in how the 'was_empty' parameter is computed
for the push callback, and gets rid of the unused 'shutting_down' field.
Review: https://reviewboard.asterisk.org/r/2323/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the red/black tree work was committed, there was an extra ", " in
the REF_DEBUG definition of ao2_container_alloc_rbtree.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With ptlib 2.10.9, the configure script fails due to grep returning multiple
matches for the pattern it searches for. This patch updates the pattern
matching to return only the actual version for the symbol searched for,
PTLIB_VERSION.
(closes issue ASTERISK-20980)
Reported by: Stefan Reuter
patches:
ASTERISK-20980-1.patch uploaded by Stefan Reuter (license 5339)
........
Merged revisions 380297 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 380298 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380299 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
* 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
* 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
................
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
................
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
................
r378000 | seanbright | 2012-12-13 15:20:32 -0600 (Thu, 13 Dec 2012) | 8 lines
Make generate_exchange_uuid() always return the passed ast_str pointer.
I changed this code earlier to return NULL if it wasn't able to generate a UUID,
whereas the earlier code would always return the ast_str that was passed in.
Switch back to returning the ast_str, only set it to the empty string instead if
UUID generation fails. We still do a validity check later which will catch this
and blow up if necessary.
................
r378001 | wedhorn | 2012-12-13 15:25:31 -0600 (Thu, 13 Dec 2012) | 9 lines
Minor fixes for chan_skinny
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and
correct len of 2 strcmp in skinny_setdebug(). (see opticron's review
on https://reviewboard.asterisk.org/r/2240/)
........
Merged revisions 377991 from http://svn.asterisk.org/svn/asterisk/branches/11
................
r378002 | rmudgett | 2012-12-13 15:28:15 -0600 (Thu, 13 Dec 2012) | 35 lines
confbridge: Fix MOH on simultaneous user entry to a new conference.
When two users entered a new conference simultaneously, one of the callers
hears MOH. This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.
* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.
* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.
* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference. This way any pre-join file playback does not
need to worry about MOH.
* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.
(closes issue ASTERISK-20606)
Reported by: Eugenia Belova
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2232/
........
Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377993 from http://svn.asterisk.org/svn/asterisk/branches/11
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378003 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
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
Richard Mudgett
Jason Parker
Michael L. Young
Joshua Colp
Matthew Jordan
Kevin Harwell
Kinsey Moore
Sean Bright
David M. Lee
Walter Doekes
Mark Michelson
Automerge script