The option had not been converted to use the replacement for
ast_bridged_channel(). One touch mixmonitor now records files again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In addition to porting those features, they now enjoy greater feature parity
with one another. Specifically, AutoMixMon now has a start and stop
message that can be specified with TOUCH_MIXMONITOR_MESSAGE_START and
TOUCH_MIXMONITOR_MESSAGE_STOP.
(closes issue ASTERISK-21553)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2620/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was a misunderstanding about ast_bridge_impart()'s handling of the
imparted channel's reference. The channel reference is passed by the
caller unless ast_bridge_impart() returns an error.
* Fixed a memory leak in conf_announce_channel_push() if the impart
failed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change removes AST_CEL_BRIDGE_UPDATE since it should no longer be
used because masquerade situations are now accounted for in other ways.
This also refactors usage of AST_CEL_FORWARD to be produced by a Dial
message which has been extended with a "forward" field.
(closes issue ASTERISK-21566)
Review: https://reviewboard.asterisk.org/r/2635/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The menuselect parser is very simple. It looks for AST_MODULE_INFO and
uses any quoted string on that line as the module summary display.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
This means CDRs track well with what an actual channel is doing - which
is useful in transfer scenarios (which were previously difficult to pin
down). It does, however, mean that CDRs cannot be 'fooled'. Previous
behavior in Asterisk allowed for CDR applications, channels, and other
properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
be what everyone wants, but it is a defined behavior and as such, it is
predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
changes have been made to ResetCDR and ForkCDR in particular. Many of the
options for these two applications no longer made any sense with the new
framework and the (slightly) more immutable nature of CDRs.
There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.
(closes issue ASTERISK-21196)
Review: https://reviewboard.asterisk.org/r/2486/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r386792, the ability to play prompts to the first caller in a call queue was
added. While this is arguably a bug fix for those who expect the first caller
to continue receiving prompts while the agent is dialed, it has the side effect
of preventing the first caller from hearing the agent immediately upon
bridging. This may not be a problem for those who really want this option, but
for those who didn't care whether or not the first caller in queue heard their
position, it was an issue.
This patch disables the ability for the first caller in the queue to hear
prompts and adds a new option, announce-to-first-user, to queues.conf. Those
who the behavior can enable it by setting this value to True.
Note that if we ever implement the ability to have the prompts be stopped
upon bridging, this option can be removed.
(closes issue ASTERISK-21782)
Reported by: Remi Quezada
........
Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 391241 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Features configuration is handled in its own API in
features_config.h and features_config.c. This way, features
configuration is accessible to anything that needs it.
In addition, features configuration has been altered to
be more channel-oriented. Most callers of features API
code will be supplying a channel so that the individual
channel's settings will be acquired rather than the global
setting.
Missing from this commit is XML documentation for the
features configuration. That will be handled in a separate
commit.
Review: https://reviewboard.asterisk.org/r/2578/
(issue ASTERISK-21542)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Refactor some channel blob publishing code to use
ast_channel_publish_blob now that it is available and fix a JSON
reference leak that was occurring during varset publishing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ast_channel_cached_blob_create was merged,
ast_channel_blob_create_from_cache was partially removed in an
unresolved merge conflict. This restores ast_channel_blob_create_from_cache
and refactors usage of ast_channel_cached_blob_create (requires an
ast_channel) to use ast_channel_blob_create_from_cache (requires a
channel uniqueid) instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The attended transfer API call can complete the attended transfer in a number of ways
depending on the current bridged states of the channels involved.
The hiding of masquerades is done in some bridging-related functions, such as the manager
Bridge action and the Bridge dialplan application. In addition, call pickup was edited
to "move" a channel rather than masquerade it.
Review: https://reviewboard.asterisk.org/r/2511
(closes issue ASTERISK-21334)
Reported by Matt Jordan
(closes issue Asterisk-21336)
Reported by Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
* ChanSpyStart/Stop
* MonitorStart/Stop
* MusicOnHoldStart/Stop
* FullyBooted/Reload
* All Voicemail/MWI related events
In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.
Review: https://reviewboard.asterisk.org/r/2532
(closes issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
New in 12 are the ConfBridgeMute/Unmute events, which are triggered when a user
changes their mute/unmute state. This was typically triggered when a user hit a
DTMF key that triggered the mute/unmute menu handler. Forgotten in this is when an
AMI action or CLI command triggers the mute/unmute. This patch now raises the
events in those situations as well.
(closes issue ASTERISK-21802)
Reported by: Birger "WIMPy" Harzenetter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Breaks many things until they can be reworked. A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When this option was added, it was noted in CHANGES, but was missing
the XML documentation that this patch adds.
(closes issue ASTERISK-21780)
Patch-by: Brad Latus (snuffy)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r388005, macros were introduced to consistently define message
types. This added an assert if a message type was used either before
it was initialized or after it had been cleaned up. It turns out that
this assertion fires during shutdown.
This actually exposed a hidden shutdown ordering problem. Since
unsubscribing is asynchronous, it's possible that the message types
used by the subscription could be freed before the final message of
the subscription was processed.
This patch adds stasis_subscription_join(), which blocks until the
last message has been processed by the subscription. Since joining was
most commonly done right after an unsubscribe, a
stasis_unsubscribe_and_join() convenience function was also added.
Similar functions were also added to the stasis_caching_topic and
stasis_message_router, since they wrap subscriptions and have similar
problems.
Other code in trunk was refactored to join() where appropriate, or at
least verify that the subscription was complete before being
destroyed.
Review: https://reviewboard.asterisk.org/r/2540
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does two things:
* It fixes a bug where the outbound channel's application/data set by the
dialing API/app_dial is not communicated until the channel is hung up.
If that happens, AMI would incorrectly send a NewExten event immediately
after a Hangup. This isn't really AMI's fault, as the dialing APIs never
communicated the 'helpful' app/data on the outbound channel until it was
hungup.
* It makes public sending a stasis message about a change in channel state.
This is useful enough that - for now at least - it should be public. If
operations on a channel go to being more coarse-grained, this function
could be made private again.
Review: https://reviewboard.asterisk.org/r/2548
Note that this problem was found and reported by Matt DiMeo.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The snapshot API contains an option that allow for combining of new
and old messages within a single snapshot. New messages, however,
include options beyond just 'INBOX' - it also includes the Urgent
folder. A previous patch that combined INBOX and Urgent accidentally
impacted snapshots that attempted to gain messages from just the Old
folder. This patch fixes the snapshot gathering such that the API
returns the appropriate messages for the folder selected, with and
without the combine option.
This should make it more clear about what's happening.
Review: https://reviewboard.asterisk.org/r/2539/
........
Merged revisions 388816 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way.
This patch breaks the major components of res_stasis.c into individual
files.
* res/stasis/app.c - Stasis application tracking
* res/stasis/control.c - Channel control objects
* res/stasis/command.c - Channel command object
This refactoring also allows res_stasis applications to be loaded as
independent modules, such as the new res_stasis_answer module.
The bulk of this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary.
Review: https://reviewboard.asterisk.org/r/2530/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I've noticed when doing a graceful shutdown that the res_stasis_http.so
module gets unloaded before the modules that use it, which causes some
asserts during their unload.
While r386928 was a quick hack to get it to not assert and die, this
patch increases the use counts on res_stasis.so and res_stasis_http.so
properly. It's a bigger change than I expected, hence the review instead
of just committing it.
Review: https://reviewboard.asterisk.org/r/2489/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a framework in res_stasis for handling events from
channel topics. JSON event generation and validation code is created
from event documentation in rest-api/api-docs/events.json to assist in
JSON event generation, ensure consistency, and ensure that accurate
documentation is available for ALL events that are received by
res_stasis applications.
The userevent application has been refactored along with the code that
handles userevent channel blob events to pass the headers as key/value
pairs in the JSON blob. As a side-effect, app_userevent now handles
duplicate keys by overwriting the previous value.
Review: https://reviewboard.asterisk.org/r/2428/
(closes issue ASTERISK-21180)
Patch-By: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the "ignorebusy" setting was deprecated, we added some code to allow us to
be compatible with older setups that are still using the "ignorebusy" setting
instead of "ringinuse". We set a char *variable with the column name to use,
which helps the realtime functions to use the correct column in their SQL
queries. When "persistentmembers" is enabled, we are not setting this variable
before the realtime functions were called to load members. This results in the
variable being NULL and therefore causing a segfault when loading members during
the module's process of loading.
The solution was to move the code that sets that variable to be before these
realtime functions are called during the loading of the module.
(closes issue ASTERISK-21738)
Reported by: JoshE
Tested by: JoshE
Patches:
asterisk-21738-rt-ringinuse-field-not-set.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2499/
........
Merged revisions 388108 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, a call to ast_load_realtime_multientry could get away with
passing a NULL parameter to the function, even though it really isn't
supposed to do that. After the change over to using ast_variable instead
of variadic arguments, the realtime engine gets unhappy if you do this.
This was always an unintended function call in app_directory anyway - now,
we just don't call into the realtime function calls if we don't have anything
to query on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we first introduced the channel blob types, the JSON blobs were
self identifying by a required "type" field in the JSON object
itself. This, as it turns out, was a bad idea.
When we introduced the message router, it was useless for routing based
on the JSON type. And messages had two type fields to check: the
stasis_message_type() of the message itself, plus the type field in the
JSON blob (but only if it was a blob message).
This patch corrects that mistake by removing the required type field
from JSON blobs, and introducing first class stasis_message_type objects
for the actual message type.
Since we now will have a proliferation of message types, I introduced a
few macros to help reduce the amount of boilerplate necessary to set
them up.
Review: https://reviewboard.asterisk.org/r/2509
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reload support was originally not included for SLA. It was added later,
but in a fairly non-traditional way. It basically sets a flag
indicating that a reload is pending, and then waits for a time where it
thinks everything SLA related is idle and unused, and *then* executes
the reload. It does this because the reload process is destructive. It
starts by throwing everything away and starting over.
There are a number of problems with this approach. One of them is that
the check to see if anything in use was incomplete. This patch makes it
more complete and thus less likely for a crash to occur during reload
processing. However, this approach still has problems so some much more
significant reworking of this code will need to come in as a next step.
Patch credit and testing by CoreDial, LLC.
........
Merged revisions 387688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 387689 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves VarSet events for local variables raised by GoSub
over to Stasis-Core. It also tweaks up the post-processing documentation
scripts to not combine parameters if both parameters are already documented.
(issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Make confbridge config parsing user profile, bridge profile, and menu
container hash/cmp functions correctly check the OBJ_POINTER, OBJ_KEY, and
OBJ_PARTIAL_KEY flags.
* Made confbridge load_module()/unload_module() free all resources on
failure conditions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
sla.lock was already locked in the only place that sla_check_reload() was called.
Remove the redundant locking of sla.lock done in this function. Less recursive
locking is A Good Thing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the move from Asterisk's event system to Stasis, this makes
distributed device state aggregation always-on, removes unnecessary
task processors where possible, and collapses aggregate and
non-aggregate states into a single cache for ease of retrieval. This
also removes an intermediary step in device state aggregation.
Review: https://reviewboard.asterisk.org/r/2389/
(closes issue ASTERISK-21101)
Patch-by: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After some discussion on asterisk-dev, it was decided that the bulk of
the logic in app_stasis actually belongs in a resource module instead
of the application module.
This patch does that, leaves the app specific stuff in app_stasis, and
fixes up everything else to be consistent with that change.
* Renamed test_app_stasis to test_res_stasis
* Renamed app_stasis.h to stasis_app.h
* This is still stasis application support, even though it's no
longer in an app_ module. The name should never have been tied to
the type of module, anyways.
* Now that json isn't a resource module anymore, moved the
ast_channel_snapshot_to_json function to main/stasis_channels.c,
where it makes more sense.
Review: https://reviewboard.asterisk.org/r/2430/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
refactored to use these events rather than producing the events directly
in channel.c. Finally, the code was added to app_stasis to produce
DTMF events on the WebSocket.
The AMI events are completely backward compatible, including sending
events on transmitted DTMF, and sending DTMF start events.
The Stasis-HTTP events are somewhat simplified. Since DTMF start and
DTMF send events are generally less useful, Stasis-HTTP will only send
events on received DTMF end.
(closes issue ASTERISK-21282)
(closes issue ASTERISK-21359)
Review: https://reviewboard.asterisk.org/r/2439
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original report was that app_voicemail would crash. This was caused by
ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being
performed for that return status. After adding the initial patch to fix this
issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered.
During review, Walter Doekes (wdoekes) suggested adding a helper function in
order to determine if we had a valid configuration or not.
This patch does the following:
* Creates a helper function to check if the configuration is valid
* Adds calls to the new helper function where appropiate
* Fixes memory leaks where the code returned without running
ast_config_destroy() on the configuration that was loaded
(closes issue ASTERISK-21302)
Reported by: Jaco Kroon
Tested by: Jaco Kroon, Michael L. Young
Patches:
asterisk-11.3.0-app_voicemail-ast_config-fixes.patch
Jaco Kroon (license 5671)
asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2443/
........
Merged revisions 385551 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 385557 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The hash and compare functions for the control container was reusing
the wrong ones, causing some problems. I fixed it, but in the wrong
branch. Oh well, it happens.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* A new Stasis payload has been defined for multi-channel messages. This
payload can store multiple ast_channel_snapshot objects along with a single
JSON blob. The payload object itself is opaque; the snapshots are stored
in a container keyed by roles. APIs have been provided to query for and
retrieve the snapshots from the payload object.
* The Dial AMI events have been refactored onto Stasis. This includes dial
messages in app_dial, as well as the core dialing framework. The AMI events
have been modified to send out a DialBegin/DialEnd events, as opposed to
the subevent type that was previously used.
* Stasis messages, types, and other objects related to channels have been
placed in their own file, stasis_channels. Unit tests for some of these
objects/messages have also been written.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support.
This module registers a dialplan function named Stasis, which is used
to put a channel into the named Stasis app. As a channel enters and
leaves the Stasis diaplan application, the Stasis app receives a
'stasis-start' and 'stasis-end' events.
Stasis apps register themselves using the stasis_app_register and
stasis_app_unregister functions. Messages are sent to an application
using stasis_app_send.
Finally, Stasis apps control channels through the use of the
stasis_app_control object, and the family of stasis_app_control_*
functions.
Other changes along for the ride are:
* An ast_frame_dtor function that's RAII_VAR safe
* Some common JSON encoders for name/number, timeval, and
context/extension/priority
Review: https://reviewboard.asterisk.org/r/2361/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch migrates the TestEvent AMI events to first be dispatched over the
Stasis-Core message bus. This helps to preserve the ordering of the events
with other events in the AMI system, such as the various channel related
events.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
At least one call to run_externnotify provides a NULL context parameter and
because the snprintf statement doesn't account for a NULL context parameter,
it simply writes '(null)' to the arguments string instead. This patch makes
it write two quotes back to back for that argument instead in the event of
a NULL context.
(closes issue ASTERISK-18207)
Reported by: Barry L. Kline
Patches:
modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930)
........
Merged revisions 384325 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 384326 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an SLA trunk is ringing (inbound call on the trunk) Asterisk will
make outbound calls to the stations that have that trunk. If more than
one station answers the call at the same time, all channels other than
the first one to answer are left in a bad state. The channel gets
leaked, is not connected to anything, and there's no way to get rid of
it.
We now properly clean up these losing channels by hanging up on them.
Since they lost the race, as we process their answer, there is no
ringing trunk for them to answer.
........
Merged revisions 383835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 383836 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A regression was accidentally introduced when allowing an optional ID to be used
when calling StopMixMonitor. When we are unable to stop MixMonitor on a
channel, -1 is being returned which triggers the hangup of the channel.
This patch restores the prior behavior by returning 0 whether we were successful
or not. It also allows the call from the manager to use the return code when
the action fails.
(closes issue ASTERISK-21294)
Reported by: daroz
Tested by: daroz
Patches:
asterisk-21294-stop_mixmonitor_hangingup.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2404/
........
Merged revisions 383631 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.
To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.
I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.
* Move JSON support from res_json.c to main/json.c
* Made libjansson-dev a required dependency
* Added an ast_channel_blob message type, which has a channel
snapshot and JSON blob of data.
* Changed UserEvent and Newexten events so that they are dispatched
via ast_channel_blob messages on the channel's topic.
* Got rid of the ast_channel_varset message; used ast_channel_blob
instead.
* Extracted the manager functions converting Stasis channel events to
AMI events into manager_channel.c.
(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot
that Urgent also "counts" as new messages. This fixed the problem when any of
the three folders was specified and the combine option was used.
It missed the case where the folder isn't specified and we build a snapshot of
all folders. This patch corrects that.
........
Merged revisions 382617 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fix checks to make sure that if a confbridge record start command is issued
from the CLI it will always use the file name given on the CLI even if it
changes between start/stop records for a conference. Previously it had been
reusing the same file between start/stops even if a new filename was given.
(issue AST-1088)
Reported by: John Bigelow
........
Merged revisions 382385 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some channel drivers, specifically those that have a varying rate in the
number of audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the DENOISE
function in func_speex to channels joining the conference.
The denoiser function in the speex library is initialized with the number of
audio samples in each sample that will be provided to it. If the number of
audio samples changes, the denoiser has to be thrown away and re-initialized.
While this could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the system.
This patches does the following:
* Checks for the presence of func_speex as opposed to codec_speex when
determining if the DENOISE function is present (which is where the function
is actually implemented)
* Adds an option to MeetMe 'n' that causes the denoiser to not be applied
to a channel when it joins. This keeps the current behavior the default, but
let's users disable the denoiser if it causes problems on their system.
Review: https://reviewboard.asterisk.org/r/2358
(closes issue AST-1062)
Reported by: Thomas Arimont
........
Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 382230 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ConfBridge was refactored to better handle the concept of marked,
wait_marked, and normal users co-existing in a conference (thereby implementing
a state machine for the conference), the wait_marked users were put into their
own list of conference participants, separate from the active users. This list
is used for wait_marked users when they are waiting in a conference but no
marked user has joined; normal users may have joined at this point however.
There are several AMI/CLI commands that affect conference users that were not
checking the wait_marked users list:
* CLI/AMI commands that mute/unmute a participant. In this case, wait_marked
users have to remain in their particular state and should not be affected -
however, the commands would return "Channel not found" as opposed to the
appropriate error condition.
* CLI/AMI commands that kick a participant. An admin should always be able to
kick a participant out of the conference.
This patch fixes both sets of commands, and cleans up the CLI commands slightly
by allowing them to complete a participant name (this was supposed to have been
added, but the function call was commented out and wasn't implemented).
Review: https://reviewboard.asterisk.org/r/2346/
(closes issue AST-1114)
Reported by: John Bigelow
Tested by: John Bigelow
........
Merged revisions 382068 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge and Page require that there always be a default bridge and user
profile available. While properties of the default profiles can be overriden
in the configuration file, removing them can create situations where neither
application can function properly.
This patch ensures that if an administrator removes the profiles from the
confbridge.conf configuration file, the profiles are added upon load.
Documentation clarifying this has been added to the confbridge.conf.sample file.
Review: https://reviewboard.asterisk.org/r/2356/
(closes issue AST-1115)
Reported by: John Bigelow
Tested by: John Bigelow
........
Merged revisions 382066 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ParkAndAnnounce application documentation for the optional return_context
parameter states the following:
return_context
The goto-style label to jump the call back into after timeout. Default
'priority+1'.
Unfortunately, the application was sending the channel back into the dialplan
at 'priority', which is the ParkAndAnnounce application call. This causes an
infinite loop of the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those callers you
wish to drive to the end of madness, this was not the intent of the
application.
(closes issue ASTERISK-20113)
Reported by: serginuez
patches:
app_parkandannounce.diff uploaded by serginuez (License 6405)
........
Merged revisions 381916 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 381917 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Apparently this feature became broken in 11, probably as a result
of the Hangup Cause project.
(closes issue ASTERISK-21113)
Reprted by: Heiko Wundram
Patches:
app_dial.patch uploaded by Heiko Wundram (license 5822)
........
Merged revisions 381880 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you started/stopped recording of a conference multiple times channels
would remain active even when all participants left the conference. This
was due to the fact that a reference to the confbridge was being added
every time a start record command was issued, but when the recording was
stopped there was no matching de-reference thus keeping the conference alive.
Made sure only a single reference is added for the record thread no matter how
many times recording is started/stopped. A de-reference is issued upon thread
ending.
Note, this issue is being fixed under AST-1088 since it relates to it and
should have been corrected along with those modifications.
(issue AST-1088)
Reported by: John Bigelow
........
Merged revisions 381737 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if one starts, stops, and then starts a recording again for a
conference the recorded data is appended to the file originally created
on the first record start. An option record_file_append has been added
that defaults to "yes", but when set to "no" will force creation of a new
file between every record start/stop.
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A deadlock occurred after starting/stopping and then restarting a confbridge
recording. Upon starting a recording a record thread is created that holds a
lock until just before exiting. Stopping the recording does not stop/exit the
thread or release the lock. The thread waits until recording begins again.
Starting a stopped recording signals the thread to continue and start recording
again. However restarting the recording also created another record thread
resulting in a deadlock. The fix was to make sure the record thread was only
created once.
Also it was noted that filenames for the recordings were being concatenated for
each start/stop. This was fixed by creating a new file for each conference
session and appending the actual recorded data within the file (e.g. passing
the 'a' option to MixMonitor).
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
........
Merged revisions 381702 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added the following flags to the CLI "confbridge list <conference>" output:
A - The user is an admin
M - The user is a marked user
W - The user must wait for a marked user to join
E - The user will be kicked after the last marked user leaves the conference
w - The user is waiting for a marked user to join
* Added the following header to the AMI ConfbridgeList events:
WaitMarked, EndMarked, and Waiting.
(closes issue AST-1101)
Reported by: John Bigelow
Patches:
confbridge-show-admin3.txt (license #5091) patch uploaded by John Bigelow
Modified
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381644 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
The function call ast_db_deltree returns the number of row deleted, or a
negative number if it failed. DBdeltree was treating any non-zero return
as an error, causing a spurious verbose error message to be displayed.
This patch handles the return code of ast_db_deltree correctly.
(closes issue ASTERISK-21070)
Reported by: ianc
patches:
dbdeltree.diff uploaded by ianc (License #5955)
........
Merged revisions 381364 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 381365 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, ConfBridge does not send manager events for ConfbridgeMute,
ConfbridgeUnmute, ConfbridgeStartRecord and ConfbridgeStopRecord. This patch
adds these events to the manager.
The reporter's patch moves some other events up to the beginning of the file.
The patch being committed is based on the patch contributed from the reporter of
this issue. I have made a lot of modifications to the patch in order for it to
fit in better with what we currently are doing in the code when it comes to
manager events. I also made a few changes to the <see-also> elements on some of
the events.
(closes issue ASTERISK-20827)
Reported by: Clint Davis
Tested by: Clint Davis, Michael L. Young
Patches:
20827.diff uploaded by Clint Davis (license 6453)
asterisk-20827-confbridge-events.diff uploaded by
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2309/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If say.conf did not exists prior to originally loading module app_playback it
would not load on subsequent reloads of the module once it had been created.
This occurred because upon reload of the app_playback module it would only
load a new configuration if an old one had previously existed. This fix simply
removed the association between checking if an old configuration existed and
the loading of the new one.
(closes issue ASTERISK-20800)
Reported by: pgoergler
........
Merged revisions 381216 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 381217 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Unloading ConfBridge caused the next AMI action received to crash
Asterisk.
* Add the missing unregister of AMI action ConfbridgeSetSingleVideoSrc
when ConfBridge is unloaded.
(closes issue ASTERISK-20994)
Reported by: Jeremy Kister
Patches:
jira_asterisk_20994_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: Rusty Newton, Jeremy Kister
........
Merged revisions 381067 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Page and ConfBridge custom announcement did not play when users
entered the conference.
* Fix the CONFBRIDGE(user,announcement) file not getting played. The code
to do this got removed accidentally when the ConfBridge code was
restructured to be more state machine like.
* Fixed play_prompt_to_user() doxygen comments.
* Fixed the Page A(x) and n options for the caller. The caller never
played the announcement file and totally ignored the n option. The code
to do this was lost when the application was converted to use ConfBridge.
* Factored out setup_profile_bridge(), setup_profile_paged(), and
setup_profile_caller() routines to setup ConfBridge profiles. Made each
profile setup routine use the default template if one has not already been
setup by dialplan.
(closes issue ASTERISK-20990)
Reported by: Jeremy Kister
Tested by: rmudgett
........
Merged revisions 380894 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A marked user ending a conference with only end_marked users generates
error messages:
ERROR[0000][C-00000000]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user ''
* The MULTI_MARKED state was doing too much when it was kicking out the
end_marked users from the conference. The kicked out users will clean up
after themselves when they exit the conference.
(closes issue ASTERISK-20991)
Reported by: Jeremy Kister
Tested by: rmudgett
........
Merged revisions 380892 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "sound_only_one" sound was not being set even though it was configured. In
looking into this, I found that the "join" and "leave" prompts were not being
set either.
(closes issue ASTERISK-20898)
Reported by: Stephan
Tested by: Stephan
Patches:
asterisk-20898-custom-sounds-ignored.diff uploaded by
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2289/
........
Merged revisions 380193 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation for ConfbridgeList states that the Conference field is
optional. That's not really the case: if you fail to provide a Conference
number, the command will kick back an error.
(closes issue AST-1090)
Reported by: John Bigelow
........
Merged revisions 380028 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The old prompts for the administrator menu were inadequate. They didn't mention
that the menu had additional options through the 8 key and pressing the 8 key
wouldn't reveal what those options were. This patch fixes all of that while
also organizing code pertaining to each individual menu type which was
previously all stored in one gigantic function along with many of the basic
conference functions.
(closes issue AST-996)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/360/
........
Merged revisions 379885 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379892 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379912 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
This patch fixes two bugs:
* If an outbound call is made from a SLA phone using SLAStation, then there is
no ringtone audible to the phone that originates the call. The indication of
the ringing was not being passed to the SLA station; this patch fixes that
by passing through the progress indications.
* If an SLA station hangs up before the called party answers, then the channel
to the called party continues to ring until a timeout occurs. If the called
party manages to answer, Asterisk attempts to connect the called party to
a non-existant MeetMe room. This patch corrects the behavior by abandoning
the call attempt if it detects that the SLA station is no longer in use
while attempting to call the called party.
Review: https://reviewboard.asterisk.org/r/2275/
(closes issue ASTERISK-20462)
Reported by: dkerr
patches:
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20462.patch uploaded by dkerr (license 5558)
(closes issue ASTERISK-20440)
Reported by: dkerr
patches:
asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
........
Merged revisions 379825 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379826 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Generate a warning message if sound files do not exist when trying to
play the user count to the conference. Use the new helper routine
sound_file_exists() for consistency.
* Put the new user into autoservice when playing user counts to the
conference.
* Check the return value of ast_bridge_impart().
........
Merged revisions 379808 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An incorrect string initializations was left in ast_str_encode_mime from the
patch that converted string manipulations to use ast_str strings (r191140).
The string initialization causes a crash when ast_str_set is called on
the string later on in the function.
(closes issue ASTERISK-18697)
Reported by: Chris Boot
patches:
minivm-null-pointer-dereference-fix.patch uploaded by bootc (license 6309)
(issue ASTERISK-20854)
Reported by: Chris Warr
Tested by: Chris Warr
........
Merged revisions 379608 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379609 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the restructuring work got committed to Confbridge in r375470 to
fix many open issues, it caused a regression in the reported count of
users when conference information was requested via CLI or manager.
This corrects the user count and user information displayed when
listing conference information from the CLI and manager.
(closes issue ASTERISK-20938)
Reported By: Timo Teras
Patches:
confbridge-list.patch uploaded by Timo Teras (license 5409)
........
Merged revisions 379478 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
app_voicemail will no longer issue error messages when it retrieves an msg_id
with a NULL value from realtime and will instead simply populate the msg_id
field with a newly generated msg_id. In addition, this patch changes the way
msg_ids are generated to eliminate certain causes of duplicate IDs appearing
within a single system. In addition, when messages are copied, they will now
receive a new msg_id.
(closes issue ASTERISK-20717)
Reported by: Alec Davis
Review: https://reviewboard.asterisk.org/r/2220/
........
Merged revisions 379460 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.
* Fix so a queue member does not receive more than one call from a queue.
NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.
* Did some refactoring to eliminate some code redundancy.
(issue ASTERISK-16115)
Reported by: nik600
Patches:
jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
Modified
* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem. The fix did not need to be optional. The fix should not have
tried to explicitly set the device state. Setting the device state by
something other than the device introduces a race condition. I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
........
Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378687 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the "h" extension is present within the context of the queue, all calls
are being reported COMPLETECALLER even when the agent is hanging up the call.
This patch checks to see if the agent hung-up or not instead of only relying on
checking if the queue (caller) channel hung-up or not. It would appear that
having the h extension in the mix, the pbx goes to the h extension,
"hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.
(closes issue ASTERISK-20743)
Reported by: call
Tested by: call, Michael L. Young
Patches:
asterisk-20743-q-cmplt-caller.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2256/
........
Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378515 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is an interesting feature that allows additional strings to be used to
search the Directory, primarily intended to be used with nicknames, but could
be used with affiliations and the like. Because the name field is used in
more than one place (such as email notifications), it is important that these
additional strings not be placed in the name field, but be specified
separately.
Review: https://reviewboard.asterisk.org/r/2244/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With the option ringinuse=no set, the patch committed for ASTERISK-16115
causes non-SIP queue members to never be called because the device state
is checked after a channel is created to determine if the member is busy.
These queue members always get the "Member %s is busy, cannot dial"
message.
Most channel drivers other than chan_sip use the default device state
handling. The default device-state state is considered in use or unknown
if the channel exists or not respectively.
(closes issue ASTERISK-20801)
Reported by: rmudgett
Patches:
jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett
........
Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378037 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378038 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@378002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Jonathan Rose
Kinsey Moore
Matthew Jordan
Jason Parker
Mark Michelson
David M. Lee
Michael L. Young
Russell Bryant
Olle Johansson
Joshua Colp
Kevin Harwell
Tilghman Lesher