Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.
Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Often times, when subscribing to a topic, one wants to handle
different message types differently. While one could cascade if/else
statements through the subscription handler, it is much cleaner to
specify a different callback for each message type. The
stasis_message_router is here to help!
A stasis_message_router is constructed for a particular stasis_topic,
which is subscribes to. Call stasis_message_router_unsubscribe() to
cancel that subscription.
Once constructed, routes can be added using
stasis_message_router_add() (or stasis_message_router_set_default()
for any messages not handled by other routes). There may be only one
route per stasis_message_type. The route's callback is invoked just as
if it were a callback for a subscription; but it only gets called for
messages of the specified type.
(issue ASTERISK-20887)
Review: https://reviewboard.asterisk.org/r/2390/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Unsubscribing things in Asterisk seems to very commonly follow with
NULLing out the variable that was unsubscribed. This change makes that
a bit simpler.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AMI, HTTP, and chan_sip all support TLS in some way, but none of them
support all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from setting
TLS/SSL options that they do not support.
This also gets tlsverifyclient closer to a working state by requesting
the client certificate when tlsverifyclient is set. Currently, there is
no consumer of main/tcptls.c in Asterisk that supports this feature and
so it can not be properly tested.
Review: https://reviewboard.asterisk.org/r/2370/
Reported-by: John Bigelow
Patch-by: Kinsey Moore
(closes issue AST-1093)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The cache dump mechanism allows the developer to retreive multiple
items of a given type (or of all types) from the cache residing in a
stasis caching topic in addition to the existing single-item cache
retreival mechanism. This also adds to the caching unit tests to
ensure that the new cache dump mechanism is functioning properly.
Review: https://reviewboard.asterisk.org/r/2367/
(issue ASTERISK-21097)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the initial use of this bus, I took some work kmoore did creating
channel snapshots. So rather than create AMI events directly in the
channel code, this patch generates Stasis events, which manager.c uses
to then publish the AMI event.
This message bus provides a generic publish/subscribe mechanism within
Asterisk. This message bus is:
- Loosely coupled; new message types can be added in seperate modules.
- Easy to use; publishing and subscribing are straightforward
operations.
In addition to basic publish/subscribe, the patch also provides
mechanisms for message forwarding, and for message caching.
(issue ASTERISK-20887)
(closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Often, Asterisk may realize that a change in the source of an RTP stream is
about to occur and ask that the RTP engine reset it's lock on the current RTP
source. In certain scenarios, it may take awhile for the new remote system to
send RTP packets, while the old remote system may continue providing RTP during
that time period. This causes Asterisk to re-lock onto the old source, thereby
rejecting the new source when the old source stops sending RTP and the new
source begins.
This patch prevents that by having a constant secondary, 'secret' probation
mode enabled when an RTP source has been chosen. RTP packets from other sources
are always considered, but never chosen unless the current RTP source stops
sending RTP.
Review: https://reviewboard.asterisk.org/r/2364
(closes issue AST-1124)
Reported by: John Bigelow
Tested by: John Bigelow
(closes issue AST-1125)
Reported by: John Bigelow
Tested by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If ast_threadpool_set_size with a size equal to the current size, a
reference to a set_size_data structure would be leaked.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ownership of the listener reference is not transferred because the
listener is reffed when placed into the taskprocessor. Ensure that the
listener is dereffed properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When converting AMI class authorizations to a string representation, the
method always appends the ALL class authorization. This is especially
important for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as a class
authorization for itself. (Events have always assumed that the ALL class
authorization is implied when they are raised)
Unfortunately, this did mean that specifying a user with restricted class
authorizations would show up in the 'manager show user' CLI command as
having the ALL class authorization.
Rather then modifying the existing string manipulation function, this patch
adds a function that will only return a string if the field being compared
explicitly matches class authorization field it is being compared against.
This prevents ALL from being returned unless it is actually specified for
the user.
(closes issue ASTERISK-20397)
Reported by: Johan Wilfer
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The incorrect callid was being written to the "data1" field in queue_log table
for transfer events. The callid of the queue was being written instead of the
transfer target's callid. This now gets the correct "transfer to" number and
places that in the "data1" field of the queue_log table when a transfer event
is triggered.
(closes issue ASTERISK-19960)
Reported by: vladimir shmagin
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 fixes a crash in Asterisk that could be caused by using the
PresenceState AMI action while providing an invalid provider. This patch
also adds some additional warnings when a user attempts to provide the
PresenceState action with invalid data, and removes some NOTICE statements
that were still lurking in the code from testing.
(closes issue AST-1084)
Reported by: John Bigelow
Tested by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381595 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
Caching a struct ast_app pointer is not a good idea because someone could
unload the application. After the applicaiton unload the cached ast_app
pointer is no longer valid. Only pbx.c can cache the pointer because it
knows when the application is unloaded and removes the pointer.
* Fixed one-touch Monitor and MixMonitor to not cache the ast_app pointer
and not use the silly monitor_ok/mixmonitor_ok/stopmixmonitor_ok flags.
* Extracted bridge_check_monitor() from ast_bridge_call() and use propper
locking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Taking advantage of the sorted order of the registered functions container
requires that they are actually inserted in the expected sort order.
* Insert the registered functions into the container in case sensitive
position. As a result, only the complete_functions() routine needs to
search the entire container because it does a case insensitive search for
convenience.
Caught by the unit tests.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed "core show function" tab completion and token count checking.
* Refactored function and application container handling code to reduce
redundancy.
* Made __ast_pbx_run() return using the defines the caller should expect.
Doesn't change the returned values. Just made use the defines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These two variables were previously not being set when comebacktoorigin=yes
and the example configs seemed to imply that they should be. Since there
is no harm in this and since calls that are sent back to origin are capable
of continuing in the dialplan, this seemed like a no-brainer. Also it
supports some bridging tests I've been working on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A regression was introduced which removed automatic fallback behavior from
the PBX. This behavior was used by call parking (or at least documented as
how the feature works) in order to select an extension when the flat channel
extension wasn't available from the comebackcontext. Parking now handles
the fallbacks internally in order to keep behavior matching with how it is
documented.
(closes issue ASTERISK-20716)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2296/
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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/
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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
Kinsey Moore
David M. Lee
Matthew Jordan
Richard Mudgett
Jason Parker
Joshua Colp
Tzafrir Cohen
Kevin Harwell
Michael L. Young
Mark Michelson
Jonathan Rose
Russell Bryant