This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.
Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The cause code needs to be passed from the disconnecting channel to the
bridge peers if the disconnecting channel dissolves the bridge.
* Made the call to an app_agent_pool agent disconnect with the busy cause
code if the agent does not ack the call in time or hangs up before acking
the call.
(closes issue ASTERISK-22042)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2772/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This essentially makes app_queue usable again. From reviewboard:
* Reporting of transfers and call completion is done by creating stasis
subscriptions and listening for specific events in order to determine
when the call is finished (either via a transfer or hangup).
* Dial end messages have been added where they were previously missing.
* Queue stats are properly being updated again once calls have finished.
* AgentComplete stasis messages and AMI events are now occurring again.
* Mixmonitor starting has been factored into its own function and uses the
Mixmonitor API now instead of using ast_pbx_run()
In addition to the changes in app_queue, there are several supplementary changes as well:
* Queue logging now differentiates between attended and blind transfers. A
note about this is in the CHANGES file.
* Local channel optimization events now report more information. This
includes which of the two local channels involved is the destination of
the optimization, the channel that is replacing the destination local channel,
and an identifier so that begin and end events can be matched to each other.
The end events are now sent whether the optimization was successful or not and
includes an indicator of whether the optimization was successful.
* Changes were made to features and bridging_basic so that additional flags may
be set on a bridge. This is necessary because the queue requires that its
bridge only allows move-swap local channel optimizations into the bridge.
(closes issue ASTERISK-21517)
Reported by Matt Jordan
(closes issue ASTERISK-21943)
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2694
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this change, we would reject SUBSCRIBE requests that had no Accept
headers. Now event package handlers that handle the default type for the
event package indicate that they do so. Therefore, if we have a handler that
can handle the default type, we can allow SUBSCRIBEs for the handler's event
package that have no Accept headers.
(closes issue ASTERISK-22067)
reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/2774
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This modifies the behavior of the CEL engine to conform to documented
behavior for Asterisk 12 as defined on the wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification
The primary changes deal with removal of the peer field from function
calls since it is no longer directly relevant to the bridging system
and removal of the layer of CDR-like business logic that was providing
a partial emulation of Asterisk 11 CEL functionality. With this change,
there is no longer a distinction between "bridges" and "conferences"
and all participation changes are denoted with bridge enter and bridge
exit messages.
This updates the CEL unit tests to handle these changes and simplifies
some of the macros used in the process.
This also fixes a segfault when attempting to ref a configuration that
failed to load.
Review: https://reviewboard.asterisk.org/r/2788/
(issue ASTERISK-21567)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an endpoint fails to include the T38MaxBitRate attribute during negotiation,
Asterisk will negotiate a bit rate of 2400 instead of the ITU recommended
bit rate of 14400. This patch fixes this by making AST_T38_RATE_14400 the
'default' value of the enum by assigning it a value of 0, such that if an
endpoint fails to include the attribute, the default will be 14400.
Note that Walter Doekes included the nice comment in frame.h about why we are
purposefully assigning AST_T38_RATE_14400 a value of 0.
(closes issue ASTERISK-22275)
Reported by: Andreas Steinmetz
patches:
fax-fix.patch uploaded by anstein (License 6523)
........
Merged revisions 397256 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 397257 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added an option flags parameter to interval hooks. Interval hooks now
can specify if the callback will affect the media path or not.
* Added an option flags parameter to the bridge action custom callback.
The action callback now can specify if the callback will affect the media
path or not.
* Made the holding bridge technology reexamine the participant idle mode
option whenever the entertainment is restarted.
* Fixed app_agent_pool waiting agents needlessly starting and stopping MOH
every second by specifying the heartbeat interval hook as not affecting
the media path.
* Fixed app_agent_pool agent alert from restarting the MOH after the alert
beep. The agent entertainment is now changed from MOH to silence after
the alert beep.
* Fixed holding bridge technology to defer starting the entertainment. It
was previously a mixture of immediate and deferred.
* Fixed holding bridge technology to immediately stop the entertainment.
It was previously a mixture of immediate and deferred. If the channel
left the bridging system, any deferred stopping was discarded before
taking effect.
* Miscellaneous holding bridge technology rework coding improvements.
Review: https://reviewboard.asterisk.org/r/2761/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is more-or-less a reversion of previous ACL behavior so that
it is more self-contained. ACL sections are now only parsed if res_pjsip_acl.so
is loaded. Moreover, the configuration section is now "type=acl" instead of
"type=security".
The original reason for having ACLs configured in a "type=security" section
was to lump ACLs and other security-related items into the same section. The
problem is that ACLs really should be in their own sections and there are
no other security-related options implemented anyways.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.
This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.
Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP/foo -- Local;1==Local;2 -- .... -- Local;1==Local;2 -- SIP/bar
Kick a ;1 channel and the chain toward SIP/foo goes away.
Kick a ;2 channel and the chain toward SIP/bar goes away.
This can leave a local channel chain between the kicked ;1 and ;2 channels
that are orphaned until you manually request one of those channels to
hangup or request the bridge to dissolve.
* Added ast_bridge_kick() as a companion to ast_bridge_remove(). The
functional difference is that ast_bridge_kick() may dissolve the bridge as
a result of the channel leaving the bridge.
* Made CLI "bridge kick <bridge> <channel>" use ast_bridge_kick() instead
of ast_bridge_remove() so the bridge can dissolve if needed.
* Renamed bridge_channel_handle_hangup() to ast_bridge_channel_kick() and
made it accessible to other files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change r395954 reordered some stasis object destruction, which should
have been fine. Unfortunately, it caused some hard to reproduce issues
related to objects being accessed after they had been destroyed. The
patch in r396329 fixed the destruction order problem; this patch
addresses the underlying issue. A few other stasis-related fixes were
also added.
* Add ref-bumps around areas where objects may get transitively
destroyed. (For example, where we lock a topic, unref a subscription,
which unrefs the topic, which explodes the topic when we try to
unlock it.)
* Wrote an extensive doxygen page about Stasis implementation,
relationships between objects, lifecycles of objects, how the
refcounting works, etc. Many other comments were added, corrected, or
cleaned up.
* Added an assert to the topic dtor to catch extra ref decrements.
* Fixed type used after destruction errors for graceful shutdown in
stasis_channels.c.
* I added two unit tests in an attempt to catch destruction order
issues. Since the underlying cause is a race condition, though, the
tests rarely failed even when the code was wrong.
* Fixed a leak in stasis_cache_pattern.c.
(closes issue ASTERISK-22243)
Review: https://reviewboard.asterisk.org/r/2746/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When asterisk has run out of memory (for whatever reason), the alloc
function logs a message. Logging requires memory. A recipe for
infinite recursion.
Stop the recursion by comparing the function call depth for sane values
before attempting another OOM log message.
Review: https://reviewboard.asterisk.org/r/2743/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Bridge API DTMF hook matching would not deal with DTMF end events
only. It required a DTMF begin event to start matching the DTMF hooks.
There are many places in Asterisk where code only generates DTMF end
events without the corresponding begin event. One such place is the AMI
action Atxfer.
* Fixed DTMF hook matching if there is a string of DTMF frames in the read
queue. We could potentially miss some of them before.
* Fixed AMI Atxfer action documentation.
(closes issue ASTERISK-22037)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2752/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes ARI bridging to allow other channel operations to
happen while the channel is bridged.
ARI channel operations are designed to queue up and execute
sequentially. This meant, though, that while a channel was bridged,
any other channel operations would queue up and execute only after the
channel left the bridge.
This patch changes ARI bridging so that channel commands can execute
while the channel is bridged. For most operations, things simply work
as expected. The one thing that ended up being a bit odd is recording.
The current recording implementation will fail when one attempts to
record a channel that's in a bridge. Note that the bridge itself may
be recording; it's recording a specific channel in the bridge that
fails. While this is an annoying limitation, channel recording is
still very useful for use cases such as voice mail, and bridge
recording makes up much of the difference for other use cases.
(closes issue ASTERISK-22084)
Review: https://reviewboard.asterisk.org/r/2726/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The lonely flag is an optional flag for bridge channels that will
make them leave a bridge when a channel leaves if only lonely
channels are in the bridge at that point. This is useful for things
like ending recording and playback channels when they cease to be
interacting with other channels in the bridge.
(closes issue ASTERISK-22117)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2721/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Changed ast_manager_build_bridge_state_string() to assume an empty
prefix string just like ast_manager_build_channel_state_string().
* Created ast_manager_build_bridge_state_string_prefix() to work just like
ast_manager_build_channel_state_string_prefix().
* Made BridgeMerge AMI event use To/From prefixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does three things:
1. It provides a Surrogate channel technology with a consolidated
"implementation detail flag" on the channel technology. This tells
consumers of Stasis that the creation of this channel is an implementation
detail in Asterisk and can be ignored (if they so choose). This
consolidates the conference recorder/announcer flags as well - these flags
had no additional meaning beyond "ignore this channel please".
2. It modifies allocation of a channel in two ways:
(a) If a channel technology can be determined from the name, we set it
directly in the allocation routine. This prevents the initial
publication of the message from going out with a NULL channel technology
where possible. This lets Stasis consumers get the right channel
technology on the first publication.
(b) It reorganizes allocation to make use of the 'finalized' property on the
channel. This was already used to know that a channel had completely
finished its construction in the masquerade routine; now we also use it
to know whether or not the setting of certain channel properties is
occurring during or post construction. The various set routines were
modified accordingly as well.
3. The masquerade event is now dead, Jim. It no longer served any purpose
whatsoever - if you perform a call pickup you'll get a Pickup event;
if you perform an attended transfer you will still get those events; if you
steal a channel to put it elsewhere you'll get the corresponding NewExten or
BridgeEnter events.
Review: https://reviewboard.asterisk.org/r/2740
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch implements the controls from ARI recordings. The controls
are:
* DELETE /recordings/live/{recordingName} - stop recording and
discard it
* POST /recordings/live/{recordingName}/stop - stop recording
* POST /recordings/live/{recordingName}/pause - pause recording
* POST /recordings/live/{recordingName}/unpause - resume recording
* POST /recordings/live/{recordingName}/mute - mute recording (record
silence to the file)
* POST /recordings/live/{recordingName}/unmute - unmute recording.
Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.
(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Stasis changes in r395954 had an unanticipated side effect: messages
published directly to an _all topic does not get forwarded to the
corresponding caching topic.
This patch fixes that by changing how caching topics forward messages,
and how the caching pattern forwards are setup.
For the caching pattern, the all_topic is forwarded to the
all_topic_cached. This forwards messages published directly to the
all_topic to all_topic_cached.
In order to avoid duplicate messages on all_topic_cached, caching topics
were changed to no longer forward uncached messages. Subscribers to an
individual caching topic should only expect to receive cache updates,
and subscription change messages. Since individual caching topics are
new, this shouldn't be a problem.
There are a few minor changes to the pre-cache split behavior.
* For topics changed to use the caching pattern, the all_topic_cached
will forward snapshots in addition to cache updates. Since
subscribers by design ignore unexpected messages, this should be
fine.
* Caching topics that don't use the caching pattern no longer forward
non-cache updates. This makes no difference for the current caching
topics.
* mwi_topic_cached, channel_by_name_topic and
presence_state_topic_cached have no subscribers
* device_state_topic_cached's only subscriber only processes cache
udpates
(issue ASTERISK-22243)
Review: https://reviewboard.asterisk.org/r/2738
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dial and Queue would previously apply a new set of features whenever
bridging. These options would be based purely on the options supplied
to the dial/queue applications. This patch changes the function those
applications use to bridge calls so that the features will be added
to the set of existing features for each channel rather than having
them override the existing features.
(closes issue ASTERISK-22209)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2713/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Roles are now cleared with each entry into a bridge with addChannel.
If the roles parameter is present, the role specified will be applied
to all channels being added with the addChannel command.
(closes issue ASTERISK-21973)
Reported by: Matt Jordan
https://reviewboard.asterisk.org/r/2691/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.
This patch renames the variables, adding the ast_ prefix so they will
be exported.
Review: https://reviewboard.asterisk.org/r/2737
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* BridgeEnter now contains the unique ID of the channel that is to be swapped out, if applicable.
* There is a ParkedCallSwap event that is sent when a parked channel has a new channel take its place.
(closes issue ASTERISK-22193)
reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/2712
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This moves ast_str_container_alloc, ast_str_container_add,
ast_str_container_remove, and related private functions into
strings.c/h since they really don't belong in astobj2.c/h.
As a result of this move, utils also had to be updated.
Review: https://reviewboard.asterisk.org/r/2719/
(closes issue ASTERISK-22041)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.
(closes issue ASTERISK-22039)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2717
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It moves the pickup code out of features.c and into pickup.c
* It removes the vast majority of dead code out of features.c. In particular,
this includes the parking code.
(issue ASTERISK-22134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds support for externally initiated parking requests. In particular,
chan_skinny has a protocol level message that initiates a call park.
This patch now supports that option, as well as the protocol specific
mechanisms in chan_dahdi/sig_analog and chan_mgcp.
* A parking bridge features virtual table has been added that provides
access to the parking functionality that the Bridging API needs. This
includes requests to park an entire 'call' (with little or no additional
information, thank you chan_skinny), perform a blind transfer to a parking
extension, determine if an extension is a parking extension, as well as the
actual "do the parking" request from the Bridging API.
* Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
functions
* The removal of some - but not all - dead parking code from features.c
This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)
Review: https://reviewboard.asterisk.org/r/2710
(closes issue ASTERISK-22134)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This renames all files and API calls from several variants of
Stasis-HTTP to ARI including:
* Stasis-HTTP -> ARI
* STASIS_HTTP -> ARI
* stasis_http -> ari (ast_ari for global symbols, file names as well)
* stasis http -> ARI
Review: https://reviewboard.asterisk.org/r/2706/
(closes issue ASTERISK-22136)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most hook callbacks did not need the bridge parameter. The pointer value
could become invalid if the channel is moved to another bridge while it is
executing.
* Fixed some issues in feature_attended_transfer() as a result.
* Reduce the bridge inhibit count in
attended_transfer_properties_shutdown() after it has restored the bridge
channel hooks.
* Removed basic bridge requirement on feature_blind_transfer(). It does
not require the basic bridge like feature_attended_transfer().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed feature limits to not use special members of struct
ast_bridge_features.
* Fixed memory leak in off nominal paths of bridge_builtin_set_limits().
* Fixed off nominal path in ast_bridge_features_limits_construct() freeing
unallocated memory if it was not called by bridge_builtin_set_limits().
* Made bridge_builtin_interval_features.so unloadable.
* Simplified parking's use of its duration interval hook.
* Made BridgeWait S option not depend upon another module being loaded.
(closes issue ASTERISK-22107)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2701/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes arguments for BridgeWait from BridgeWait(role, options) to
BridgeWait(bridge_name, role, options). Now multiple holding bridges may
be created and referenced by this application.
(closes issue ASTERISK-21922)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2642/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Convert interval timers to use the ast_waitfor_nandfds() timeout.
* Remove bridge channel action for intervals. Now the main loop handles
running interval hooks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Reduced the number of hook containers to just dtmf_hooks,
interval_hooks, and other_hooks. As a result, several functions dealing
with the different hook containers could be combined.
* Extended the generic hook struct for DTMF and interval hooks instead of
using a variant record.
* Merged the special talk detector hook into the other_hooks container.
* Replaced ast_bridge_features_set_talk_detector() with
ast_bridge_talk_detector_hook().
(issue ASTERISK-22107)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It pulls out bridge_channel and puts it into its own translation unit
* It adds public and protected headers for bridging_channel. Protected
functions are appropriate only for the Bridging API and sub-classes of a
bridge.
(issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This greatly modifies the operation of DTMF attended transfers so that
the full range of options from features.conf applies.
In addition, a new option has been added that allows for a transferer
to switch between bridges during a transfer before completing the
transfer.
(closes issue ASTERISK-21543)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2654
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch introduces DTLS-SRTP support to chan_pjsip and the options
necessary to configure it including an option to allow choosing between
32 and 80 byte SRTP tag lengths.
During the implementation and testing of this patch, three other bugs
were found and their fixes are included with this patch. The two in
chan_sip were a segfault relating to DTLS setup and mistaken call
rejection. The third bug fix prevents chan_pjsip from attempting to
perform bridge optimization between two endpoints if either of them is
running any form of SRTP.
Review: https://reviewboard.asterisk.org/r/2683/
(closes issue ASTERISK-21419)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The stasis_cache_update messages are somewhat cumbersome to handle
with the stasis_message_router. Since all updates have the same
message type, they are normally handled with the same route.
Since caching itself is a first class component of stasis-core, it
makes sense for the router to handle the cache update messages itself.
This patch adds stasis_message_router_add_cache_update() and
stasis_message_router_remove_cache_update() to handle the routing of
stasis_cache_update messages.
This patch also corrects an issue with manager_{bridging,channels}.c,
where events might be reordered. The reordering occurs because the
components use different message routers, which they needed because
they both needed to route cache update messages. They now both use
manager's router, and add cache routes for just the cache updates they
are interested in.
(closes issue ASTERISK-22038)
Review: https://reviewboard.asterisk.org/r/2677/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent
local channel optimizations. Local channel optimizations were one of
several things conveyed by the now defunct BRIDGE_UPDATE event type.
This also adds a unit test to test generation of this new CEL event.
Review: https://reviewboard.asterisk.org/r/2676/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.
This adds tests for blind transfers, several types of attended
transfers, and call pickup.
The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.
Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags together into a
field and renamed them to OBJ_SEARCH_OBJECT, OBJ_SEARCH_KEY, and
OBJ_SEARCH_PARTIAL_KEY respectively. The values were selected to keep
existing code compiling and working until the codebase can be changed to
stop using these values as bit flags and use them as an enum field.
The old names are defined to the new names for backward compatibility.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made ast_audiohook_detach_list() and ast_audiohook_write_list_empty()
NULL tolerant.
* Made ast_audiohook_detach_list() return void since it is a destructor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.
(closes issue ASTERISK-21592)
Reported by: Matt Jordan
(closes issue ASTERISK-21593)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2670/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds new flags to the channel tech properties that flag it as
different types of implementation detail used exclusively to provide a
feature. Examples of channels that would have these flags include the
announcement and recording channels used by confbridge which are the
only two marked as such by this patch.
Review: https://reviewboard.asterisk.org/r/2633/
(closes issue ASTERISK-21873)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since ast_hangup() is effectively a channel destructor, it should be a
void function.
* Make the few silly callers checking the return value no longer do so.
Only the CDR and CEL unit tests checked the return value.
* Make all callers take advantage of the NULL safe change and remove the
NULL check before the call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2657/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds a virtual table of callbacks to core_unreal. These callbacks can be
supplied by concrete implementations of "unreal" channel drivers, which lets
the unreal channel driver call specific functionality when it performs some
action. Currently, this is done to notify implementations when an
optimization operation has begun, and when an optimization operation has
succeeded.
* It adds Stasis-Core messages for Local channel bridging and Local channel
optimization. Local channel optimization is now two events: a Begin and an
End. Some consumers of Stasis-Core may want to know when an operation is
beginning so that they can 'prepare' their information; others will be more
concerned about when the operation has completed, so that they can 'fix up'
information. Stasis-Core allows for both, as does AMI.
Review: https://reviewboard.asterisk.org/r/2552
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a
channel is executing dialplan hangup logic, i.e., the 'h' extension or a
hangup handler. Stasis messages now also convey the soft hangup flag so
consumers of the messages can know when a channel is executing said
hangup logic.
* It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is
well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs,
and other consumers of Stasis have been updated to look for this flag to
know when the channel should by lying six feet under.
* The CDR engine has been updated to better handle a channel entering and
leaving a bridge. Previously, a new CDR was automatically created when a
channel left a bridge and put into the 'Pending' state; however, this
way of handling CDRs made it difficult for the 'endbeforehexten' logic to
work correctly - there was always a new CDR waiting in the hangup logic
and, even if 'ended', wouldn't be the CDR people wanted to inspect in the
hangup routine. This patch completely removes the Pending state and instead
defers creation of the new CDR until it gets a new message that requires
a new CDR.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
information in the RTCP events. Because Stasis provides a cache, Jaco's
patch was modified to pass the channel uniqueid to the RTP layer as
opposed to a pointer to the channel. This has the following benefits:
(1) It keeps the RTP engine 'clean' of references back to channels
(2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
Potentially, other implementations (such as res_rtp_multicast) could also
raise RTCP information. The engine provides structs to represent RTCP headers
and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
raise an event when we sent a RR report.
Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.
Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.
Review: https://reviewboard.asterisk.org/r/2603/
(closes issue ASTERISK-20574)
Reported by: Jaco Kroon
patches:
asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)
(closes issue ASTERISK-21471)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Legacy channel drivers often include the ability to set a default parking lot
on an endpoint basis; when channels are created for that endpoint, they inherit
the parkinglot option. Parking used to use this option more frequently; while
it is still supported, other options (such as using channel variables or
creation of a custom parkinglot) are supported. More importantly, conveying the
parkinglot information through a channel snapshot isn't terribly useful - it
is rarely (if ever) changed on a channel and some consumers of channel
snapshots, such as ARI, will never use the information.
(closes issue ASTERISK-21968)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This process also involved a large amount of rework regarding how to redial
the Parker when a channel leaves a parking lot due to timeout. An attended
transfer channel variable has been added to attended transfers to extensions
that will eventually park (but haven't at the time of transfer) as well.
This resolves one of the two BUGBUG comments remaining in res_parking.
(issues ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2638/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The OneTouchRecord feature has historically been a toggle. This patch
adds the ability to make the OneTouchRecord hook optionally start/stop
recording only. If OneTouchRecord is already doing what is requested then
only the invoker hears the courtesy tone and/or start/stop recording
message.
The new feature is written so we could easily add explicit start/stop
recording DTMF hooks for Monitor and MixMonitor.
The majority of the changes in bridge_builtin_features.c is a refactoring
of the OneTouchRecord code (Monitor and MixMonitor versions) so it is easy
to direct the toggle/start/stop functionality.
Review: https://reviewboard.asterisk.org/r/2655/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the first step in adding recording support to the
Asterisk REST Interface.
Recordings are stored in /var/spool/recording. Since recordings may be
destructive (overwriting existing files), the API rejects attempts to
escape the recording directory (avoiding issues if someone attempts to
record to ../../lib/sounds/greeting, for example).
(closes issue ASTERISK-21594)
(closes issue ASTERISK-21581)
Review: https://reviewboard.asterisk.org/r/2612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The appropriate settings for the Stasis threadpool is very system
specific, depending upon both workload and system configuration.
This patch adds a stasis.conf file which can be used to configure the
key attributes of the threadpool for the Stasis message bus.
(closes issue ASTERISK-21280)
Review: https://reviewboard.asterisk.org/r/2651/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started with the simple idea of changing the /events data
model to be more sane. The original model would send out events like:
{ "stasis_start": { "args": [], "channel": { ... } } }
The event discriminator was the field name instead of being a value in
the object, due to limitations in how Swagger 1.1 could model objects.
While technically sufficient in communicating event information, it was
really difficult to deal with in terms of client side JSON handling.
This patch takes advantage of a proposed extension[1] to Swagger which
allows type variance through the use of a discriminator field. This had
a domino effect that made this a surprisingly large patch.
[1]: https://groups.google.com/d/msg/wordnik-api/EC3rGajE0os/ey_5dBI_jWcJ
In changing the models, I also had to change the swagger_model.py
processor so it can handle the type discriminator and subtyping. I took
that a big step forward, and using that information to generate an
ari_model module, which can validate a JSON object against the Swagger
model.
The REST and WebSocket generators were changed to take advantage of the
validators. If compiled with AST_DEVMODE enabled, JSON objects that
don't match their corresponding models will not be sent out. For REST
API calls, a 500 Internal Server response is sent. For WebSockets, the
invalid JSON message is replaced with an error message.
Since this took over about half of the job of the existing JSON
generators, and the .to_json virtual function on messages took over the
other half, I reluctantly removed the generators.
The validators turned up all sorts of errors and inconsistencies in our
data models, and the code. These were cleaned up, with checks in the
code generator avoid some of the consistency problems in the future.
* The model for a channel snapshot was trimmed down to match the
information sent via AMI. Many of the field being sent were not
useful in the general case.
* The model for a bridge snapshot was updated to be more consistent
with the other ARI models.
Another impact of introducing subtyping was that the swagger-codegen
documentation generator was insufficient (at least until it catches up
with Swagger 1.2). I wanted it to be easier to generate docs for the API
anyways, so I ported the wiki pages to use the Asterisk Swagger
generator. In the process, I was able to clean up many of the model
links, which would occasionally give inconsistent results on the wiki. I
also added error responses to the wiki docs, making the wiki
documentation more complete.
Finally, since Stasis-HTTP will now be named Asterisk REST Interface
(ARI), any new functions and files I created carry the ari_ prefix. I
changed a few stasis_http references to ari where it was non-intrusive
and made sense.
(closes issue ASTERISK-21885)
Review: https://reviewboard.asterisk.org/r/2639/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves the RESTful URL's around to more appropriate
locations for release.
The /stasis URL's are moved to /ari, since Asterisk REST Interface was
a more appropriate name than Stasis-HTTP. (Most of the code still has
stasis_http references, but they will be cleaned up after there are no
more outstanding branches that would have merge conflicts with such a
change).
A larger change was moving the ARI events WebSocket off of the shared
/ws URL to its permanent home on /ari/events. The Swagger code
generator was extended to handle "upgrade: websocket" and
"websocketProtocol:" attributes on an operation.
The WebSocket module was modified to better handle WebSocket servers
that have a single registered protocol handler. If a client
connections does not specify the Sec-WebSocket-Protocol header, and
the server has a single protocol handler registered, the WebSocket
server will go ahead and accept the client for that subprotocol.
(closes issue ASTERISK-21857)
Review: https://reviewboard.asterisk.org/r/2621/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I ended up using a bridge blob, so this structure was
unused. Keeping it in the header would just cause confusion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If no matching endpoint is found for the incoming request Asterisk will respond
with a 401 Unauthorized (rejecting the request), but will first challenge if
no authorization creditials are given.
Changes also included moving ACL options into a new global 'security'
configuration section in res_sip.conf.
(closes issue ASTERISK-21433)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2554/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds several unit tests for CEL functionality and provides the
requisite framework for creating additional unit tests.
This also cleans up some reference leaks that were occurring in
Stasis-Core message callback code.
Review: https://reviewboard.asterisk.org/r/2646/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the ability to send unsolicited NOTIFY requests to a particular endpoint
with a configured payload. Added both CLI and AMI support. For a given
endpoint, this module will iterate over all its contacts sending the appropriate
NOTIFY request to each.
(closes issue ASTERISK-21436)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2623/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The originate APIs allow callers to provide a pointer to a channel that will
point to the originated channel if the function call succeeds. This is used by AMI
to provide channel information when the originate is performed synchronously.
Unfortunately, if the originate fails in certain ways, the outbound channel is
already disposed of during the dialing itself. This results in the channel being
improperly dereferenced by the internal originate function in pbx.c.
This patch ref bumps the channel to prevent this from occurring. Callers must now
unlock and unref the channel (which is more in line with general channel management
guidelines anyway).
This only affects manager, as it is the only consumer of this API function that
actually passes in a channel pointer.
Review: https://reviewboard.asterisk.org/r/2617/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393361 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
Things may no longer behave in an unexpected fashion. Local channel
optimization to holding bridges will work again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Parking typically occurs when a channel is transferred to a parking extension.
When this occurs, the channel never actually hits the dialplan if the extension
it was transferred to was a "parking extension", that is, the extension in
the first priority calls the Park application. Instead, the channel is
immediately sent into the holding bridge acting as the parking bridge.
This is problematic.
Because we never go out to the dialplan, the CDRs won't transition properly
and the application field will not be set to "Park". CDRs typically swallow
holding bridges, so the CDR itself won't even be generated.
This patch handles this by pulling out the holding bridge handling into its
own CDR state. CDRs now have an explicit parking state that accounts for this
specific subclass of the holding bridge. In addition, we handle the parking
stasis message to set application specific data on the CDR such that the
last known application for the CDR properly reflects "Park".
This is a bit sad since we're working around the odd internal implementation
of parking that exists in Asterisk (and that we had to maintain in order to
continue to meet some odd use cases of parking), but at least the code to
handle that is where it belongs: in CDRs as opposed to sprinkled liberally
throughout the codebase.
This patch also properly clears the OUTBOUND channel flag from a channel when
it leaves a bridge, and tweaks up dialing handling to properly compare the
correct CDR with the channel calling/being dialed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add config framework OPT_CHAR_ARRAY_T and OPT_STRINGFIELD_T non-empty
requirement option. There are cases were you don't want a config option
string to be empty. To require the option string to be non-empty, just
set the aco_option_register() flags parameter to non-zero.
* Updated some config framework enum aco_option_type comments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393034 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
This adds support for stasis/sounds and stasis/sounds/{ID} queries via
the Asterisk RESTful Interface (ARI, formerly Stasis-HTTP).
The following changes have been made to accomplish this:
* A modular indexer was created for local media.
* A new function to get an ast_format associated with a file extension
was added.
* Modifications were made to the built-in HTTP server so that URI
decoding could be deferred to the URI handler when necessary.
* The Stasis-HTTP sounds JSON documentation was modified to handle
cases where multiple languages are installed in different formats.
* Register and Unregister events for formats were added to the system
topic.
(closes issue ASTERISK-21584)
(closes issue ASTERISK-21585)
Review: https://reviewboard.asterisk.org/r/2507/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sorcery specific object information is now opaque and allocated with the object.
This means that modules do not need to be recompiled if the sorcery specific part
is changed. It also means that sorcery can store additional information on objects
and ensure it is freed or the reference count decreased when the object goes away.
To facilitate the above a generic sorcery allocator function has been added which
also ensures that allocated objects do not have a lock.
Extended fields have been added thanks to all of the above which allows specific fields
to be marked as extended, and thus simply stored as-is within the object. Type safety
is *NOT* enforced on these fields. A consumer of them has to query and ultimately perform
their own safety check. What does this mean? Extra modules can extend already defined
structures without having to modify them.
Tests have also been included to verify extended field functionality.
Review: https://reviewboard.asterisk.org/r/2585/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. Security events
2. Websocket support
3. Diversion header + redirecting support
4. An anonymous endpoint identifier
5. Inbound extension state subscription support
6. PIDF notify generation
7. One touch recording support (special thanks Sean Bright!)
8. Blind and attended transfer support
9. Automatic inbound registration expiration
10. SRTP support
11. Media offer control dialplan function
12. Connected line support
13. SendText() support
14. Qualify support
15. Inband DTMF detection
16. Call and pickup groups
17. Messaging support
Thanks everyone!
Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Extract a useful routine from the softmix bridge technology for other
technologies. Make other technologies use it if they can.
* Made native and 1-1 bridges write to all parties if the bridge channel
writing the frame into the bridge is NULL. Softmix will also do the same
for frame types that make sense.
* Tweak the bridge write routine return value meaning and adjust the
bridge technologies to match.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The bridge frame queue functions need to return an error status if the
frame failed to be queued because of an error condition. The main calls
that needed to return the status are:
ast_bridge_channel_queue_action_data() and
ast_bridge_channel_write_action_data(). The other return changes are
ripple effects.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This finishes moving all CEL linkedid tracking entirely within cel.c
since that is now possible with channel snapshots.
This also removes another CEL linkedid manipulation function from cel.h
that has already been internalized and is neither called nor available
to link against.
Review: https://reviewboard.asterisk.org/r/2632/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392241 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
While very handy, this macro didn't occur until a later version of libjansson.
We'd prefer to be compatible with older versions still - as such, iteration
over key/value pairs in a JSON object have to be done with a little bit more
manual work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This pulls bridge-related CEL event triggers out of the code in which
they were residing and pulls them into cel.c where they are now
triggered by changes in bridge snapshots. To get access to the
Stasis-Core parking topic in cel.c, the Stasis-Core portions of parking
init have been pulled into core Asterisk init.
This also adds a new CEL event (AST_CEL_BRIDGE_TO_CONF) that indicates
a two-party bridge has transitioned to a multi-party conference. The
reverse cannot occur in CEL terms even though it may occur in actuality
and two party bridges which receive a AST_CEL_BRIDGE_TO_CONF will be
treated as multi-party conferences for the duration of the bridge.
Review: https://reviewboard.asterisk.org/r/2563/
(closes issue ASTERISK-21564)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This uses the channel state change events from Stasis-Core to determine
when channel-related CEL events should be raised. Those refactored in
this patch are:
* AST_CEL_CHANNEL_START
* AST_CEL_ANSWER
* AST_CEL_APP_START
* AST_CEL_APP_END
* AST_CEL_HANGUP
* AST_CEL_CHANNEL_END
Retirement of Linked IDs is also refactored.
CEL configuration has been refactored to use the config framework.
Note: Some HANGUP events are not generated correctly because the bridge
layer does not propagate hangupcause/hangupsource information yet.
Review: https://reviewboard.asterisk.org/r/2544/
(closes issue ASTERISK-21563)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When checking compatability for the native RTP bridge technology there is a
race condition between clearing framehooks that are destroyed when leaving
certain bridges with certain technologies (such as bridge_native_rtp) and
joining bridges with the bridge_native_rtp technology. Yes, that means a
channel in a native RTP bridge could move to another native RTP bridge and
be considered incompatible with the new native RTP bridge causing it to
revert to a simple bridge technology0. This fixes that bug by ignoring
framehooks that have been marked for destruction when checking for
compatibility with the bridge_native_rtp technology.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a Stasis message type is defined in a loadable module, handling
those messages for AMI and res_stasis events can be cumbersome.
This patch adds a vtable to stasis_message_type, with to_ami and
to_json virtual functions. These allow messages to be handled
abstractly without putting module-specific code in core.
As an example, the VarSet AMI event was refactored to use the to_ami
virtual function.
(closes issue ASTERISK-21817)
Review: https://reviewboard.asterisk.org/r/2579/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds support for Stasis applications to receive bridge-related
messages when the application shows interest in a given bridge.
To supplement this work and test it, this also adds support for the
following bridge-related Stasis-HTTP functionality:
* GET stasis/bridges
* GET stasis/bridges/{bridgeId}
* POST stasis/bridges
* DELETE stasis/bridges/{bridgeId}
* POST stasis/bridges/{bridgeId}/addChannel
* POST stasis/bridges/{bridgeId}/removeChannel
Review: https://reviewboard.asterisk.org/r/2572/
(closes issue ASTERISK-21711)
(closes issue ASTERISK-21621)
(closes issue ASTERISK-21622)
(closes issue ASTERISK-21623)
(closes issue ASTERISK-21624)
(closes issue ASTERISK-21625)
(closes issue ASTERISK-21626)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows astmm to access the backtrace generation code in Asterisk.
When memory is allocated, a backtrace is created and stored with the memory
region that tracks the allocation. If a memory corruption is detected, the
backtrace is printed to the astmm log. The backtrace will make use of the
BETTER_BACKTRACES build option if available.
As a result, this patch moves the backtrace generation code into its own file
and uses the non-wrapped versions of the C library memory allocation routines.
This allows the memory allocation code to safely use the backtrace generation
routines without infinitely recursing.
Review: https://reviewboard.asterisk.org/r/2567
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added a start technology callback that technologies can use to start
bridging operations. It is expected that native bridges will find this
useful.
* Factored out bridge_channel_complete_join().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A three party bridge uses the softmix bridging technology. This
technology has a dedicated thread used to perform the analog mixing. When
one of these parties leaves the bridge, the bridge technology is changed
from the softmix technology to a two-party mixing technology. Changing
technologies is done by removing channels from the old technology and
adding them to the new technology. Since the remaining channels do not
leave the bridge, the softmix mixing thread could continue to process all
channels in the bridge. If the bridge code is not able to start
destruction of the softmix technology before the softmix mixing thread
wakes up, a crash happens.
* Added a stop technology callback that technologies can use to request
any helper threads to stop in preparation for being destroyed.
(closes issue AST-1156)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Stasis cache clear message payloads now consist of a stasis_message
representative of the message to be cleared from the cache. This allows
multiple parallel caches to coexist and be cleared properly by the same
cache clear message even when keyed on different fields.
This change fixes a bug where multiple cache clears could be posted for
channels. The cache clear is now produced in the destructor instead of
ast_hangup.
Additionally, dummy channels are no longer capable of producing channel
snapshots.
Review: https://reviewboard.asterisk.org/r/2596
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer
channel driver specific. If the channel variable is set on the
transferrer channel, the sound will be played to the target of an attended
transfer.
* The channel variable BRIDGEPEER becomes a comma separated list of peers
in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10
peers listed. Any more peers in the bridge will not be included in the
list. BRIDGEPEER is not valid in holding bridges like parking since those
channels do not talk to each other even though they are in a bridge.
* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.
* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and
is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name
that activated the dynamic feature.
* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are
set only on the channel executing the dynamic feature. Executing a
dynamic feature on the bridge peer in a multi-party bridge will execute it
on all peers of the activating channel.
(closes issue ASTERISK-21555)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 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
This change is used to make bridge hook removal more generic. This way,
depending on the circumstance, the appropriate bridge hooks may be
removed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The library that provides UUID support varies greatly from system to
system. On most Linux distros, it's in libuuid. On OpenBSD, it's in
libe2fs-uuid. On OS X, it is in libsystem.
This patch plays hide-and-seek with UUID support, looking for it in the
three places we know about. It also corrects the Makefile so that it uses
the configured library name and include path.
(closes issue ASTERISK-21816)
Reported by: Brad Latus (snuffy)
Tested by: Brad Latus (snuffy)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* DTMF attended and blind transfers have hold/unhold behavior restored.
* External attended and blind transfers unhold the transfered party when
the transfer is initiated.
* Made prohibit blind transferring a bridge marked as masquerade only.
(ConfBridge bridges)
* Made running an application or playing a file inside a bridge post the
hold/unhold messages if MOH is requested.
Review: https://reviewboard.asterisk.org/r/2574/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses issues during immediate shutdowns, where modules
are not unloaded, but Asterisk atexit handlers are run.
In the typical case, this usually isn't a big deal. But the
introduction of the Stasis message bus makes it much more likely for
asynchronous activity to be happening off in some thread during
shutdown.
During an immediate shutdown, Asterisk skips unloading modules. But
while it is processing the atexit handlers, there is a window of time
where some of the core message types have been cleaned up, but the
message bus is still running. Specifically, it's still running
module subscriptions that might be using the core message types. If a
message is received by that subscription in that window, it will
attempt to use a message type that has been cleaned up.
To solve this problem, this patch introduces ast_register_cleanup().
This function operates identically to ast_register_atexit(), except
that cleanup calls are not invoked on an immediate shutdown. All of
the core message type and topic cleanup was moved from atexit handlers
to cleanup handlers.
This ensures that core type and topic cleanup only happens if the
modules that used them are first unloaded.
This patch also changes the ast_assert() when accessing a cleaned up
or uninitialized message type to an error log message. Message type
functions are actually NULL safe across the board, so the assert was a
bit heavy handed. Especially for anyone with DO_CRASH enabled.
Review: https://reviewboard.asterisk.org/r/2562/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 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
Asterisk REST interface.
This adds the /playback/{playbackId}/control resource, which may be
POSTed to to pause, unpause, reverse, forward or restart the media
playback.
Attempts to control a playback that is not currently playing will
either return a 404 Not Found (because the playback object no longer
exists) or a 409 Conflict (because the playback object is still in the
queue to be played).
This patch also adds skipms and offsetms parameters to the
/channels/{channelId}/play resource.
(closes issue ASTERISK-21587)
Review: https://reviewboard.asterisk.org/r/2559
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and GET /playback/{playbackId}.
This allows an external application to initiate playback of a sound on a
channel while the channel is in the Stasis application.
/play commands are issued asynchronously, and return immediately with
the URL of the associated /playback resource. Playback commands queue up,
playing in succession. The /playback resource shows the state of a
playback operation as enqueued, playing or complete. (Although the
operation will only be in the 'complete' state for a very short time,
since it is almost immediately freed up).
(closes issue ASTERISK-21283)
(closes issue ASTERISK-21586)
Review: https://reviewboard.asterisk.org/r/2531/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389587 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
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
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
The debug versions of ao2_ref() should only be used if REF_DEBUG is
enabled so nothing is written to /tmp/refs unexpectedly.
(closes issue ASTERISK-21785)
Reported by: abelbeck
Patches:
jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: abelbeck
........
Merged revisions 388700 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388701 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
A GCC bug[1] can, in some cases, pop up an unsuppressible pedwarn when
using a static inline standard library function from a non-static
inline function.
This normally doesn't show up, but can occur if you're running an
upgrade version of GCC (such as GCC 4.8 on OS X, which normally runs
GCC 4.2).
[1]: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388046 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
An endpoint is an external device/system that may offer/accept
channels to/from Asterisk. While this is a very useful concept for end
users, it is surprisingly not a core concept within Asterisk itself.
This patch defines ast_endpoint as a separate object, which channel
drivers may use to expose their concept of an endpoint. As the channel
driver creates channels, it can use ast_endpoint_add_channel() to
associate channels to the endpoint. This updated the endpoint
appropriately, and forwards all of the channel's events to the
endpoint's topic.
In order to avoid excessive locking on the endpoint object itself, the
mutable state is not accessible via getters. Instead, you can create a
snapshot using ast_endpoint_snapshot_create() to get a consistent
snapshot of the internal state.
This patch also includes a set of topics and messages associated with
endpoints, and implementations of the endpoint-related RESTful
API. chan_sip was updated to create endpoints with SIP peers, but the
state of the endpoints is not updated with the state of the peer.
Along for the ride in this patch is a Stasis test API. This is a
stasis_message_sink object, which can be subscribed to a Stasis
topic. It has functions for blocking while waiting for conditions in
the message sink to be fulfilled.
(closes issue ASTERISK-21421)
Review: https://reviewboard.asterisk.org/r/2492/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds the ability for modules to add themselves as observers
to sorcery object types. Observers can be notified when objects are
created, updated, or deleted as well as when the object type is loaded or
reloaded. Observer notifications are done using a thread pool in a serialized
fashion so the caller of the sorcery API calls is minimally impacted.
This also adds the ability to create JSON changesets of a sorcery object.
Tests are also present to confirm all of the above functionality.
Review: https://reviewboard.asterisk.org/r/2477/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch:
* Cleans up some doxygen
* Prevents leaking the system level Stasis topics and messages
on exit (users of valgrind will be happier)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change does the following:
1. Adds the sorcery realtime module
2. Adds unit tests for the sorcery realtime module
3. Changes the realtime core to use an ast_variable list instead of variadic arguments
4. Changes all realtime drivers to accept an ast_variable list
Review: https://reviewboard.asterisk.org/r/2424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to get people familiar with the Stasis message bus, it would
be useful to have something of a tutorial. Since I'm not clever enough
to think of some cool integration we could do with Twitter, I settled
for something that might actually be useful.
This patch adds a res_statsd.so module, which implements a basic
statsd[1] client. Statsd is a very simple statistics gathering server,
which can publish its results to a backend graphing engine, like
Graphite[2]. There are several different Statsd server
implementations[3], so you can pick what works best for your
environment.
The actual example of how to use the Stasis message bus is in
res_chan_stats.so. This module demonstrates how to use subscriptions
and the message router by monitoring messages and posting channels
stats to the statsd server.
A wiki page walking through res_chan_stats.so is forthcoming.
[1]: https://github.com/etsy/statsd/
[2]: http://graphite.readthedocs.org/en/latest/
[3]: http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/
Review: https://reviewboard.asterisk.org/r/2460/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.
SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.
API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The API itself is documented using Swagger, a lightweight mechanism for
documenting RESTful API's using JSON. This allows us to use swagger-ui
to provide executable documentation for the API, generate client
bindings in different languages, and generate a lot of the boilerplate
code for implementing the RESTful bindings. The API docs live in the
rest-api/ directory.
The RESTful bindings are generated from the Swagger API docs using a set
of Mustache templates. The code generator is written in Python, and
uses Pystache. Pystache has no dependencies, and be installed easily
using pip. Code generation code lives in rest-api-templates/.
The generated code reduces a lot of boilerplate when it comes to
handling HTTP requests. It also helps us have greater consistency in the
REST API.
(closes issue ASTERISK-20891)
Review: https://reviewboard.asterisk.org/r/2376/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the concept of ast_websocket_server to
res_http_websocket, allowing WebSocket connections on URL's more more
than /ws.
The existing funcitons for managing the WebSocket subprotocols on /ws
still work, so this patch should be completely backward compatible.
(closes issue ASTERISK-21279)
Review: https://reviewboard.asterisk.org/r/2453/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386020 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
These functions are already used in one branch (jrose's parking branch)
and will soon be used in other branches as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385522 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 fixes a bug introduced in r76703, wherein Asterisk could only parse
arguments in the so-called 'recommended' way, e.g., NoOp(foo,bar). The proper
syntax of NoOp,foo|bar is now parsed correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384452 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
* Updated test_uuid.c to test the new API call.
* Made system use the new API call to eliminate "10's of lines" where
used.
* Fixed untested ast_strdup() return in stasis_subscribe() by eliminating
the need for it. struct stasis_subscription now contains the uniqueid[]
string.
* Fixed some issues in exchangecal_write_event():
Create uid with enough space for a UUID string to avoid a realloc.
Fix off by one error if the calendar event provided a UUID string.
There is no need to check for NULL before calling ast_free().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
messages, with the cause code as an optional field in the blob.
NewCallerid now simply watches for changes in the callerid information
in channel snapshots, and creates the AMI event appropriately.
Since the original NewCallerid event honored the channelvars setting
in manager.conf, the channel variables configured there had to become
a part of the channel snapshot. These are now a part of every snapshot
based event, making the configuration description "every time a
channel-oriented event is emitted" less of a lie.
There a a few other changes wrapped up in here as well.
* When ast_channel_topic() is given NULL for a channel, it returns
the ast_channel_topic_all() topic instead of NULL. This can clean
up a lot of NULL checking we're doing currently.
* The fields Cause and Cause-txt were removed from the base channel
information and put only on the Hangup events, since those fields
are meaningless outside of a Hangup event.
* Removed the pipe-delimiter processing of the channelvars field,
since that's been deprecated forever.
(closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When I moved res_json.c to json.c, I left the MODULE_INFO stuff in there,
which was interesting if you ran module show. I also forgot to call what
was in module_load() from asterisk main().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383611 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
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
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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Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 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 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)
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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.
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* Clarify some documentation
* Change copyright date of taskprocessor files
* Address potential issue of creating taskprocessor with listener if
taskprocessor with that name exists already
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.
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r379021 | dlee | 2013-01-14 09:29:22 -0600 (Mon, 14 Jan 2013) | 15 lines
Fix XML encoding of 'identity display' in NOTIFY messages, continued.
When r378933 was merged into 1.8, it should have also escaped
remote_display, since it will have the same XML encoding problem when
the caller/callee roles are reversed.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
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r379023 | dlee | 2013-01-14 09:58:01 -0600 (Mon, 14 Jan 2013) | 20 lines
Masquerades are an insane implementation detail within Asterisk. It generates
a number of useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly thorough review of
masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ.
While ultimately it makes the most sense to abandon masquerades altogether,
it will take some time to completely irradicate. Even then, there may always
be code that's not worth rewriting to get rid of the masquerade.
This patch does two things to make masquerades slightly less insane:
* When swapping the names of the original and clone channel, only emit a
single rename event of original -> original<ZOMBIE>. The original code
issued three rename events to accomplish the same end.
* In addition to swapping the names of the channels, also swap their
uniqueid's. This allows the 'Uniqueid' field to be used as a stable
identifier for a channel from and external interface, such as AMI.
Review: https://reviewboard.asterisk.org/r/2266/
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r378935 | dlee | 2013-01-12 00:43:37 -0600 (Sat, 12 Jan 2013) | 41 lines
Fix XML encoding of 'identity display' in NOTIFY messages.
XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.
This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.
Several things to note:
* The Right Thing(TM) to do would probably be to replace the
ast_build_string stuff with building an ast_xml_doc. That's a much
bigger change, and out of scope for the original ticket, so I
refrained myself.
* It is with great sadness that I wrote my own ast_xml_escape
function. There's one in libxml2, but it's knee-deep in
libxml2-ness, and not easily used to one-off escape a
string.
* I only escaped the string we know is causing problems
(local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
Review: http://reviewboard.digium.internal/r/365/
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r378915 | dlee | 2013-01-11 16:31:42 -0600 (Fri, 11 Jan 2013) | 21 lines
Add JSON API for Asterisk.
This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.
* Some names have been asteriskified to protect the innocent.
* Jansson provides both reference-stealing and reference-borrowing
versions of several API's. The Asterisk API is exclusively
reference-stealing for operations that put elements into arrays and
objects.
* No support for doubles, since we usually don't need that.
* Coming along for the ride is the ast_test_validate macro, which made
the unit tests much easier to write.
[1]: http://www.digip.org/jansson/
(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/
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r378918 | file | 2013-01-11 17:05:38 -0600 (Fri, 11 Jan 2013) | 11 lines
Retain XMPP filters across reconnections so external modules continue to function as expected.
Previously if an XMPP client reconnected any filters added by an external module were lost.
This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling.
(closes issue ASTERISK-20916)
Reported by: kuj
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r378783 | dlee | 2013-01-09 14:30:33 -0600 (Wed, 09 Jan 2013) | 14 lines
Fix end condition in ast_rtp_lookup_mime_multiple2.
The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag
in the debug output.
(closes issue ASTERISK-20772)
Reported by: Xavier Hienne
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r378789 | rmudgett | 2013-01-09 14:56:23 -0600 (Wed, 09 Jan 2013) | 4 lines
* Found some more places to use ast_channel_lock_both().
* Minor optimization in ast_rtp_instance_early_bridge().
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r378790 | rmudgett | 2013-01-09 15:14:39 -0600 (Wed, 09 Jan 2013) | 4 lines
* Whitespace changes.
* Made ast_test_init() match its prototype.
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* Remove extraneous whitespace
* Bump up debug levels of messages and add identifying info to messages.
* Account for potential failures of ao2_link()
* Add additional test and some more test data
* Add some comments in places where they could be useful
* Make threadpool listeners and their callbacks optional
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r378374 | rmudgett | 2013-01-02 15:23:16 -0600 (Wed, 02 Jan 2013) | 33 lines
Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged
together. There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.
* Made the bridge wait for both channels to be redirected before exiting.
* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.
* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding. Previously the code fell back to a single channel
redirect operation.
(closes issue ASTERISK-18975)
Reported by: Ben Klang
(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
Review: https://reviewboard.asterisk.org/r/2243/
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r378377 | mjordan | 2013-01-02 16:10:32 -0600 (Wed, 02 Jan 2013) | 24 lines
Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.
This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.
(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
* issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
* issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
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r378384 | mjordan | 2013-01-02 16:19:32 -0600 (Wed, 02 Jan 2013) | 11 lines
Clean up app_mysql's application entry points to properly parse arguments
When parsing arguments, application entry points should not attempt to
directly modify the parameters to the function. This patch properly duplicates
the passed in parameters before attempting to parse them.
(issue ASTERISK-20658)
Reported by: wdoekes
patches:
issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)
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r378322 | mjordan | 2013-01-02 12:11:59 -0600 (Wed, 02 Jan 2013) | 33 lines
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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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.
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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/)
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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/
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The documentation for taskprocessors was incorrect with
regards to when a listener's alloc callback was called.
I also made the names of queued function calls in the
threadpool more uniform.
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
The new thread creation test fails because Asterisk locks up
while trying to lock a taskprocessor.
While trying to debug that, I found a race condition during taskprocessor
creation where a default taskprocessor listener could try to operate on
a partially started taskprocessor. This was fixed by adding a new callback
to taskprocessor listeners.
Then while testing that change, I found some bugs in the taskprocessor
tests where I was not properly unlocking when done with a lock. Scoped
locks have spoiled me a bit.
I still have not figured out why the threadpool thread creation test
is locking up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After giving it some consideration, there's no real
use for zombie threads. Listeners can't really use the
current number of zombie threads as a way of gauging activity,
zombifying threads is just an extra step before they die that
really serves no purpose, and since there's no way to re-animate
zombies, the operation does not need to be around.
I also fixed up some miscellaneous compilation errors that
were lingering from some past revisions.
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r376589 | mjordan | 2012-11-22 18:02:23 -0600 (Thu, 22 Nov 2012) | 29 lines
Re-initialize logmsgs mutex upon logger initialization to prevent lock errors
Similar to the patch that moved the fork earlier in the startup sequence to
prevent mutex errors in the recursive mutex surrounding the read/write thread
registration lock, this patch re-initializes the logmsgs mutex. Part of the
start up sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to daemon in order
to read startup parameters. When reading in a conf file, log statements can
be generated. Since this can't be avoided, the mutex instead is
re-initialized to ensure a reset of any thread tracking information.
This patch also includes some additional debugging to catch errors when
locking or unlocking the recursive mutex that surrounds locks when the
DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will
cause an abort() if a mutex error is detected.
(issue ASTERISK-19463)
Reported by: mjordan
Tesetd by: mjordan
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r376575 | rmudgett | 2012-11-21 12:33:16 -0600 (Wed, 21 Nov 2012) | 20 lines
Add red-black tree container type to astobj2.
* Add red-black tree container type.
* Add CLI command "astobj2 container dump <name>"
* Added ao2_container_dump() so the container could be dumped by other
modules for debugging purposes.
* Changed ao2_container_stats() so it can be used by other modules like
ao2_container_check() for debugging purposes.
* Updated the unit tests to check red-black tree containers.
(closes issue ASTERISK-19970)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2110/
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Kinsey Moore
Richard Mudgett
Mark Michelson
Matthew Jordan
David M. Lee
Walter Doekes
Jonathan Rose
Joshua Colp
Jason Parker
Kevin Harwell
Sean Bright
Olle Johansson
Andrew Latham
Michael L. Young
Automerge script