Whenever possible, audiohooks and framehooks will now be copied over
to the channel that the masquerading channel gets cloned into. This
should occur for all audiohooks and most framehooks. As a result,
in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
deprecated and its behavior is essentially the new default for all
audiohooks, plus some additional audiohooks/framehooks.
Review: https://reviewboard.asterisk.org/r/3721/
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This patch is a re-do of r414122.
When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft
hangup flags have a catastrophic effect on the pbx core if they leak out from
the bridge layer: the channel gets hung up. With the number of threads
involved in a blind transfer, and with the initial patch, it was likely that
this would occur. This caused a large number of test failures
This patch is nearly identical with the one proposed in r414122, save for the
following changes:
- We explicitly clear the UNBRIDGE flag when setting an after goto on a
channel in a bridge
- Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it
https://reviewboard.asterisk.org/r/3585/
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The bridge_native_rtp module currently uses the bridge result of the first
channel that joins a bridge as the ultimate result. This means that if the
first channel has direct media enabled but the second does not a direct
media bridge will still occur.
This change makes it so that both sides are taken into account. If either
side forbids the bridge or responds with a local bridge result then
either a generic or local bridge occurs.
ASTERISK-23541 #close
Reported by: Justin E
Review: https://reviewboard.asterisk.org/r/3577/
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This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
* A number of chatty verbose messages were removed or demoted to DEBUG
messages. Verbose messages with a verbosity level of 5 or higher were -
if kept as verbose messages - demoted to level 4. Several messages
that were emitted at verbose level 3 were demoted to 4, as announcement
of dialplan applications being executed occur at level 3 (and so the
effects of those applications should generally be less).
* Some verbose messages that only appear when their respective 'debug'
options are enabled were bumped up to always be displayed.
* Prefix/timestamping of verbose messages were moved to the verboser
handlers. This was done to prevent duplication of prefixes when the
timestamp option (-T) is used with the CLI.
* Verbose magic is removed from messages before being emitted to
non-verboser handlers. This prevents the magic in multi-line verbose
messages (such as SIP debug traces or the output of DumpChan) from
being written to files.
* _Slightly_ better support for the "light background" option (-W) was
added. This includes using ast_term_quit in the output of XML
documentation help, as well as changing the "Asterisk Ready" prompt to
bright green on the default background (which stands a better chance of
being displayed properly than bright white).
Review: https://reviewboard.asterisk.org/r/3547/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes issues with direct media bridges that occur after a blind
transfer. These issues were caught by the (currently failing)
pjsip/transfers/blind_transfer/caller_direct_media test.
The test currently fails primarily for two reasons:
(1) When Bob and Charlie (the transfer target and the transfer destination)
enter a bridge together, the framehook remains on the transfer target
channel until both channels are in the bridge. As it consumes voice frames,
the initial bridge type is a simple bridge. The framehook is removed when
both channels are in the bridge; however, this does not currently cause the
bridging framework to re-evaluate the bridge. This patch adds a
AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a
framehook is removed so the bridge can re-evaluate itself.
(2) When a channel leaves a native RTP bridge, it may be leaving due to being
hung up. Sending a re-INVITE to a channel that is about to be hung up is
not nice - in fact, there's a good chance we'll send the BYE request before
the channel has had a chance to send back a 200 OK. To be somewhat nicer,
this patch adds a function to channel.h that allows the bridging framework
to query for exactly why a channel is leaving a bridge via the channel's
soft hangup flags. This allows it to only send the re-INVITE if there's a
chance the channel will survive the native bridging experience.
Review: https://reviewboard.asterisk.org/r/3535/
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In the past framehooks have had no capability to determine what frame types a hook
is actually interested in consuming. This has meant that code has had to assume they
want all frames, thus preventing native bridging.
This change adds a callback which allows a framehook to be queried for whether it
is consuming a frame of a specific type. The native RTP bridging module has also
been updated to take advantange of this, allowing native bridging to occur when
previously it would not.
ASTERISK-23497 #comment Reported by: Etienne Lessard
ASTERISK-23497 #close
Review: https://reviewboard.asterisk.org/r/3522/
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In the past framehooks have had no capability to determine what frame types a hook
is actually interested in consuming. This has meant that code has had to assume they
want all frames, thus preventing native bridging.
This change adds a callback which allows a framehook to be queried for whether it
is consuming a frame of a specific type. The native RTP bridging module has also
been updated to take advantange of this, allowing native bridging to occur when
previously it would not.
ASTERISK-23497 #comment Reported by: Etienne Lessard
ASTERISK-23497 #close
Review: https://reviewboard.asterisk.org/r/3522/
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Add an option to enable a periodic beep to be played into a call if it
is being recorded. If enabled, it uses the PERIODIC_HOOK() function
internally to play the 'beep' prompt into the call at a specified
interval. This option is provided for both Monitor() and
MixMonitor().
Review: https://reviewboard.asterisk.org/r/3424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.
The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
not to occur (the bridge dies, the channel is removed from the bridge), then we would
never be notified.
The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.
Review: https://reviewboard.asterisk.org/r/3338
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for a channel to be masqueraded out of a bridge which
means it may no longer have RTP glue to check upon leaving said bridge.
If this situation occurred (it's possible at least during dial and call
pickup) then Asterisk would crash. This change makes sure the glue is
checked before use.
(closes issue AST-1290)
Reported by: John Bigelow
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The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
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When a bridge transitions away from one tech to another, the tech going
away is provided a dummy bridge with no channels in it to tear down.
Currently this means that the teardown code exits prematurely and does
not tear anything down. This change tears down RTP bridging for the
channel provided in the leave bridge tech callback.
This also reverts the majority of r400403 since it is now redundant.
(closes issue ASTERISK-22628)
(closes issue ASTERISK-22676)
Reported by: John Bigelow
Reported by: Kevin Harwell
Tested by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2905/
Patches:
native_rtp_fix.diff uploaded by Kinsey Moore (License 6273)
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The crash is caused by a race condition when switching between native RTP
and softmix bridging technologies. In this situation, the bridging
technology is switched from native RTP to softmix, and then back to native
RTP fast enough that the softmix private data gets destroyed before the
softmix mixing thread gets started.
Thanks to Kinsey Moore for the crash analysis.
* Fix race condition when starting the softmix mixing thread and switching
to another bridge technology.
(closes issue ASTERISK-22678)
Reported by: John Bigelow
Patches:
jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett
Tested by: John Bigelow
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When a party leaves a bridge, there may be more participants in the bridge than expected.
As such, it is important not to make assumptions regarding the list of channels in a
bridge.
This change makes it so that when a party leaves a native RTP bridge, we unbridge it and
the party it was bridged with. Previously, the first and last channels in the list were
unbridged since it was assumed that these were the two channels that had been bridged. As
previously stated, a new party had been inserted into the bridge, so this logic did not
work properly.
(closes issue ASTERISK-22615)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2899
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Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
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These refleaks were causing bridged calls not to close their RTP ports. Thus
a call would leave open 4 ports (RTP for party A, RTCP for party A, RTP for party
B, and RTCP for party B). This led to an eventual depletion of available RTP
ports.
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Issuing hold/unhold would lead to odd behavior. Between two chan_sip devices,
a hold could cause an endless chain of updates while with pjsip a similar chain
would begin but then end somewhat randomly. This patch fixes that by no longer
tweaking the RTP glue on both sides of the call for every
HOLD/UNHOLD/UPDATE_RTP_PEER frame.
(issue ASTERISK-22217)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2794/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DTMF start/end and hold/unhold events have state because a DTMF begin
event and hold event must be ended by something.
The following cases need to be handled when a channel is moved around in
the system.
* When a channel leaves a bridge it may owe a DTMF end event to the
bridge.
* When a channel leaves a bridge it may owe an UNHOLD event to the bridge.
(This case is explicitly ignored because things like transfers need
explicit control over this.)
* When a channel leaves the bridging system it may need to simulate a DTMF
end event to the channel.
* When a channel leaves the bridging system it may need to simulate an
UNHOLD event to the channel.
The patch also fixes the following:
* Fixes playing a file and restarting MOH using the latest MOH class used.
(closes issue ASTERISK-22043)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2791/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 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
* 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
Suspend and unsuspend callbacks are added to the holding bridge so
that entertainment can be disabled and re-enabled when operations
would suspend a channel on the bridge (such as playback operations).
This fixes entertainment so that when those operations end, the
entertainment can pick back up and it also serves as an optimization.
Also, this patch fixes a bug caused by triggering ringing frames
immediately instead of pushing them to the queue which created a race
condition where sometimes parking with ringing during attended
transfers would cause the ringing to be interrupted by an unhold
frame.
(closes issue ASTERISK-22006)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2711/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396189 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
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
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
Created a native_dahdi bridging technology for use with the new bridging
API.
The new bridging technology is part of the chan_dahdi channel driver
because it is very specific to that driver. Rather than include the new
code directly into chan_dahdi.c the new bridge technology is in its own
file and linked into chan_dahdi.so. A large part of this change is the
mechanical process of moving declarations around so chan_dahdi.c can be
split up into more files later.
* Changed the bridging core to pass NULL frames into the channel
technologies instead of discarding them. The channel technologies may
need the proding to determine if their configuration is still valid.
(closes issue ASTERISK-21886)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2681/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395154 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 adds more entertainment options to holding bridges and the
bridge_wait application. Also, holding bridges will now use music on
hold as the default entertainment option instead of none. The
parameters for app_bridgewait have changed to (role, options) from
the previous (options) and the options themselves have changed as
well (entertainment options are now contained in an enumerator, role
specification is handled by the role parameter, etc)
(closes issue ASTERISK-21923)
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/2679/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394731 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
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
Matthew Jordan
Jonathan Rose
Joshua Colp
Kinsey Moore
Russell Bryant
Mark Michelson
Kevin Harwell
Richard Mudgett