When a sip session is refreshed, the stream topology is looped
through, checking each stream for compatible formats. This would
cause a crash if the stream state was AST_STREAM_STATE_REMOVED,
since the formats would never be set for this stream, causing
a NULL value to be returned from ast_stream_get_formats. This
commit adds a check for streams with removed states.

Also removed a stray semicolon.

Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42

If an error occurs during a bridge impart it's possible that
the "bridge_after" callback might try to run before
control_swap_channel_in_bridge has been signalled to continue.
Since control_swap_channel_in_bridge is holding the control lock
and the callback needs it, a deadlock will occur.

* control_swap_channel_in_bridge now only holds the control
  lock while it's actually modifying the control structure and
  releases it while the bridge impart is running.
* bridge_after_cb is now tolerant of impart failures.

Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3

This change fixes a few locking issues and some video misrouting.

1. When accessing the stream topology of a channel the channel lock
must be held to guarantee the topology remains valid.

2. When a channel was joined to a bridge the bridge specific
implementation for stream mapping was not invoked, causing video
to be misrouted for a brief period of time.

ASTERISK-27182

Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03

This change does a few things to improve packet loss and renegotiation:

1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.

2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.

3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.

4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.

5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.

ASTERISK-27143

Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45

In an earlier version of Asterisk a local channel [un]lock all functions were
added in order to keep a crash from occurring when a channel hung up too early
during an attended transfer. Unfortunately, when a transfer failure occurs and
depending on the timing, the local channels sometime do not get properly
unlocked and deref'ed after being locked and ref'ed. This happens because the
underlying local channel structure gets NULLed out before unlocking.

This patch reworks those [un]lock functions and makes sure the values that get
locked and ref'ed later get unlocked and deref'ed.

ASTERISK-27074 #close

Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09

If an attended transfer failed it was possible for some of the channels
involved to get "stuck" because Asterisk was not hanging up the transfer target.

This patch ensures Asterisk hangs up the transfer target when an attended
transfer failure occurs.

ASTERISK-27075 #close

Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9

This change adds a deferred queue to bridging. If a bridge
technology determines that a frame can not be written and
should be deferred it can indicate back to bridging to do so.
Bridging will then requeue any deferred frames upon a new
channel joining the bridge.

This change has been leveraged for T.38 request negotiate
control frames. Without the deferred queue there is a race
condition between the bridge receiving the T.38 request
negotiate and the second channel joining and being in the
bridge. If the channel is not yet in the bridge then the T.38
negotiation fails.

A unit test has also been added that confirms that a T.38
request negotiate control frame is deferred when no other
channel is in the bridge and that it is requeued when a new
channel joins the bridge.

ASTERISK-26923

Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415

A previous commit added plumbing to bridge_softmix to allow for an SFU
experience with Asterisk. This commit adds an option to app_confbridge
that allows for a confbridge to actually make use of the SFU video mode.

SFU mode is implemented in a "set it and forget it" kind of way. That
is, when the bridge is created, if SFU mode is enabled, then the video
mode gets set to SFU and cannot be changed. Future improvements may
allow for a hybrid experience (e.g. forward multiple video streams,
specifically those of the most recent talkers), but for this addition,
no such capability is present.

Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020

This sets up the "plumbing" in bridge_softmix to
be able to accommodate Asterisk asking as an SFU
(selective forwarding unit) for conferences.

The way this works is that whenever a channel enters or leaves a
conference, all participants in the bridge get sent a stream topology
change request. The topologies consist of the channels' original
topology, along with video destination streams corresponding to each
participants' source video streams. So for instance, if Alice, Bob, and
Carol are in the conference, and each supplies one video stream, then
the topologies for each would look like so:

Alice:
Audio,
Source video(Alice),
Destination Video(Bob),
Destination video (Carol)

Bob:
Audio,
Source video(Bob)
Destination Video(Alice),
Destination video (Carol)

Carol:
Audio,
Source video(Carol)
Destination Video(Alice),
Destination video (Bob)

This way, video that arrives from a source video stream can then be
copied out to the destination video streams on the other participants'
channels.

Once the bridge gets told that a topology on a channel has changed, the
bridge constructs a map in order to get the video frames routed to the
proper destination streams. This is done using the bridge channel's
stream_map.

This change is bare-bones with regards to SFU support. Some key features
are missing at this point:

* Stream limits. This commit makes no effort to limit the number of
  streams on a specific channel. This means that if there were 50 video
  callers in a conference, bridge_softmix will happily send out topology
  change requests to every channel in the bridge, requesting 50+
  streams.

* Configuration. The plumbing has been added to bridge_softmix, but
  there has been nothing added as of yet to app_confbridge to enable SFU
  video mode.

* Testing. Some functions included here have unit tests.
  However, the functionality as a whole has only been verified by
  hand-tracing the code.

* Selectivenss. For a "selective" forwarding unit, this does not
  currently have any means of being selective.

* Features. Presumably, someone might wish to only receive video from
  specific sources. There are no external-facing functions at the moment
  that allow for users to select who they receive video from.

* Efficiency. The current scheme treats all video streams as being
  unidirectional. We could be re-using a source video stream as a
  desetnation, too. But to simplify things on this first round, I did it
  this way.

Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d

This patch is the first cut at adding stream support to the bridging framework.
Changes were made to the framework that allows mapping of stream topologies to
a bridge's supported media types.

The first channel to enter a bridge initially defines the media types for a
bridge (i.e. a one to one mapping is created between the bridge and the first
channel). Subsequently added channels merge their media types into the bridge's
adding to it when necessary. This allows channels with different sized
topologies to map correctly to each other according to media type. The bridge
drops any frame that does not have a matching index into a given write stream.

For now though, bridge_simple will align its two channels according to size or
first to join. Once both channels join the bridge the one with the most streams
will indicate to the other channel to update its streams to be the same as that
of the other. If both channels have the same number of streams then the first
channel to join is chosen as the stream base.

A topology change source was also added to a channel when a stream toplogy
change request is made. This allows subsystems to know whether or not they
initiated a change request. Thus avoiding potential recursive situations.

ASTERISK-26966 #close

Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163

Adds binaural synthesis to bridge_softmix (via convolution using libfftw3).
Binaural synthesis is conducted at 48kHz.
For a conference, only one spatial representation is rendered.
The default rendering is applied for mono-capable channels.

ASTERISK-26292

Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf

* app_minivm: Use built-in completion facilities to complete optional
arguments.

* app_voicemail: Use built-in completion facilities to complete
optional arguments.

* app_confbridge: Add missing colons after 'Usage' text.

* chan_alsa: Use built-in completion facilities to complete optional
arguments.

* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'

* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'

* func_odbc: Correct completions for 'odbc read' and 'odbc write'

* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.

* main/bridge: Correct completions for 'bridge kick.'

* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.

* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'

* main/pbx_app: Remove redundant completions for 'core show
applications.'

* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'

* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.

Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca

In multi-party bridges, Asterisk currently supports two video modes:
 * Follow the talker, in which the speaker with the most energy is shown
   to all participants but the speaker, and the speaker sees the
   previous video source
 * Explicitly set video sources, in which all participants see a locked
   video source

Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.

This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
    Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
    Removes any explicit video source, and sets the video mode to talk
    detection

ASTERISK-26595 #close

Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621

It's actually quite useful to see the source of a video stream change.
This doesn't happen terribly often, even with talk detection - but when
it does, it's nice to know which channel is now providing your video
stream.

As a verbose 5 level message, it shouldn't be terribly spammy or costly
to have, and is 'lower level' then most other verbose messages that the
bridge system emits.

ASTERISK-26555

Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c

ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966

Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85

Invisible bridges function the same as normal bridges, but they have the
following restrictions:

* They never show up in CLI, AMI, or ARI queries.
* They do not have Stasis messages published about them.

Invisible bridges' main use is for when use of the bridging system is
desired, but the bridge should not be known to users of the Asterisk
system.

ASTERISK-25925

Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670

You cannot reference the passed in features struct after calling
ast_bridge_impart().  Even if the call fails.

Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21

softmix_bridge_join() failed because of an allocation failure.  To address
this, the softmix bridge technology now checks if the channel failed to
join softmix successfully.  In addition, the bridge now begins the process
of kicking the channel out of the bridge so we don't have channels
partially in the bridge for very long.

* Fix the test_channel_feature_hooks.c unit tests.  The test channel must
have a valid codec to join the simple_bridge technology.  This patch makes
joining a bridge more strict by not allowing partially joined channels to
remain in the bridge.

Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b

An earlier patch blocked the ast_bridge_impart() call until the channel
either entered the target bridge or it failed.  Unfortuantely, if the
target bridge is stasis and the imprted channel is not a stasis channel,
stasis bounces the channel out of the bridge to come back into the bridge
as a proper stasis channel.  When the channel is bounced out, that
released the block on ast_bridge_impart() to continue.  If the impart was
a result of a transfer, then it became a race to see if the swap channel
would get hung up before the imparted channel could come back into the
stasis bridge.  If the imparted channel won then everything is fine.  If
the swap channel gets hung up first then the transfer will fail because
the swap channel is leaving the bridge.

* Allow a chain of ast_bridge_impart()'s to happen before any are
unblocked to prevent the race condition described above.  When the channel
finally joins the bridge or completely fails to join the bridge then the
ast_bridge_impart() instances are unblocked.

ASTERISK-25947
Reported by: Richard Mudgett

ASTERISK-24649
Reported by: John Bigelow

ASTERISK-24782
Reported by: John Bigelow

Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1

It's possible for the transferer channel to get hung up early during the
attended transfer process. For instance, a phone may send a "bye" immediately
upon receiving a sip notify that contains a sip frag 100 (I'm looking at you
Jitsi). When this occurs a race begins between the transferer being hung up
and completion of the transfer code.

If the channel hangs up too early during a transfer involving stasis bridging
for instance, then when the created local channel goes to look up its swap
channel (and associated datastore) it can't find it (since it is no longer in
the bridge) thus it fails to enter the stasis application. Consequently, the
created local channel(s) hang up as well. If the timing is just right then the
bridging code attempts to add the message link with missing local channel(s).
Hence the crash.

Unfortunately, there is no great way to solve the problem of the unexpected
"bye". While we can't guarantee we won't receive an early hangup, and in this
case still fail to enter the stasis application, we can make it so asterisk
does not crash.

This patch does just that by locking the local channel structure, checking
that the local channel's peer has not been lost, and then continuing. This
keeps the local channel's peer from being ripped out from underneath it by
the local/unreal hangup code while attempting to set the stasis message link.

ASTERISK-25771

Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880

Local channel optimization could cause DTMF digits to be duplicated.
Pending DTMF end events would be posted to a bridge when the local channel
optimizes out and is replaced by the channel further down the chain.  When
the real digit ends, the channel would get another DTMF end posted to the
bridge.

A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B

1) LocalA has the /n flag to prevent optimization.
2) B is sending DTMF to A through the local channel chain.
3) When LocalB optimizes out it can move B to the position of LocalB;1
4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would
settle an owed DTMF end to the bridge toward LocalA;2.
5) When B finally ends its DTMF it sends the DTMF end down the chain.
6) Without this patch, A would hear the DTMF digit end when LocalB
optimizes out and when B ends the original digit.

ASTERISK-25582

Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251

Currently if a channel is transferred out of a bridge, the BRIDGEPEER
variable (also BRIDGEPVTCALLID) remain set even once the channel is
out of the bridge. This patch removes these variables when leaving
the bridge.

ASTERISK-25600 #close
Reported by: Mark Michelson

Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da

During an attended transfer a thread is started that handles imparting the
bridge channel. From the start of the thread to when the bridge channel is
ready exists a gap that can potentially cause problems (for instance, the
channel being swapped is hung up before the replacement channel enters the
bridge thus stopping the transfer). This patch adds a condition that waits
for the impart thread to get to a point of acceptable readiness before
allowing the initiating thread to continue.

ASTERISK-24782
Reported by: John Bigelow

Change-Id: I08fe33a2560da924e676df55b181e46fca604577

After completing an attended transfer the transfer target channel was not being
hung up after leaving the bridge. Added an explicit softhangup to hangup said
channel, but only if it was previously bridged.

ASTERISK-24782 #close
Reported by: John Bigelow

Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada

Due to a race condition there was a chance that during an attended transfer the
channel's application would return NULL. This, of course, would cause a crash
when attempting to access the memory. This patch retrieves the channel's app
at an earlier time in processing in hopes that the app name is available.
However, if it is not then "unknown" is used instead. Since some string value
is now always present the crash can no longer occur.

ASTERISK-24869 #close
Reported by: viniciusfontes
Review: https://gerrit.asterisk.org/#/c/133/

Change-Id: I5134b84c4524906d8148817719d76ffb306488ac

Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e

When a channel enters the bridging system it is first made compatible with
the bridge and then the bridge technology makes the channel compatible
with the technology.  For all but the DAHDI native and softmix bridge
technologies the make channel compatible with the bridge step is an
effective noop because the other technologies allow all audio formats.
For the DAHDI native bridge technology it doesn't matter because it is not
an initial bridge technology and chan_dahdi allows only one native format
per channel.  For the softmix bridge technology, it is a noop at best and
harmful at worst because the wrong translation path could be setup if the
channel's native formats allow more than one audio format.

This is an intermediate patch for a series of patches aimed at improving
translation path choices.

* Removed code dealing with the unnecessary step of making the channel
compatible with the bridge.

ASTERISK-24841
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/4600/
........

Merged revisions 434424 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434430 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After completing an attended transfer the transfer target channel (the one that
gets swapped out) was not being hung up after leaving the bridge. This resulted
in a channel possibly being left around. Added an explicit softhangup for the
channel in question after the transfer is successfully completed in order to
make sure the channel is hung up.

ASTERISK-24782 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4575/
........

Merged revisions 434240 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434241 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups.  Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe.  ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.

Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.

ASTERISK-24142 #close
Reported by: David Brillert

ASTERISK-24683 #close
Reported by: Peter Katzmann

ASTERISK-24805 #close
Reported by: Badalian Vyacheslav

ASTERISK-24881 #close
Reported by: Corey Farrell

Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
........

Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Switch logger callid's from AO2 objects to simple integers.
This helps in two ways.  Copying integers is faster than
referencing AO2 objects, so this will result in a small
reduction in logger overhead.  This also erases the possibility
of an infinate loop caused by an invalid callid in
threadstorage.

ASTERISK-24833 #comment Committed callid conversion to trunk. 
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4466/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are three CLI commands to stop and restart Asterisk each.

1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
New channels are prevented while the shutdown request is pending.

2) core stop/restart gracefully - Stop or restart Asterisk when there are
no calls remaining in the system.  New channels are prevented while the
shutdown request is pending.

3) core stop/restart when convenient - Stop or restart Asterisk when there
are no calls in the system.  New calls are not prevented while the
shutdown request is pending.

ARI has made stopping/restarting Asterisk more problematic.  While a
shutdown request is pending it is desirable to continue to process ARI
HTTP requests for current calls.  To handle the current calls while a
shutdown request is pending, a new committed to shutdown phase is needed
so ARI applications can deal with the calls until the system is fully
committed to shutdown.

* Added a new shutdown committed phase so ARI applications can deal with
calls until the final committed to shutdown phase is reached.

* Made refuse new HTTP requests when the system has reached the final
system shutdown phase.  Starting anything while the system is actively
releasing resources and unloading modules is not a good thing.

* Split the bridging framework shutdown to not cleanup the global bridging
containers when shutting down in a hurry.  This is similar to how other
modules prevent crashes on rapid system shutdown.

* Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
ast_shutting_down().  You should not have to include channel.h just to
access these system functions.

ASTERISK-24752 #close
Reported by: Matthew Jordan

Review: https://reviewboard.asterisk.org/r/4399/
........

Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When swapping a Local channel in place of one already
in a bridge (to complete a bridge attended transfer),
the channel that was swapped out can actually be hung
up before the stasis bridge push callback executes on
the independant transfer thread.  This results in the
stasis app loop dropping out and removing the control
that has the the app name which the local replacement
channel needs so it can re-enter stasis.

To avoid this race condition a new push_peek callback
has been added, and called from the ast_bridge_impart
thread before it launches the independant thread that
will complete the transfer.  Now the stasis push_peek
callback can copy the stasis app name before the swap
channel can hang up.

ASTERISK-24649
Review: https://reviewboard.asterisk.org/r/4382/
........

Merged revisions 431450 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431451 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When code imparts a channel into a bridge to swap with another channel, a
ref needs to be held on the swap channel to ensure that it cannot
dissapear before finding it in the bridge.

* The ast_bridge_join() swap channel parameter now always steals a ref for
the swap channel.  This is the only change to the bridge framework's
public API semantics.

* bridge_channel_internal_join() now requires the bridge_channel->swap
channel to pass in a ref.

ASTERISK-24649
Reported by: John Bigelow

Review: https://reviewboard.asterisk.org/r/4354/
........

Merged revisions 430975 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-24049


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Follow-up issue to -r430435 from reviewboard review.

ASTERISK-24049
Review: https://reviewboard.asterisk.org/r/4315/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList

* Incremented the AMI version to 2.7.0.

* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start".  The corresponding complete event always used "Complete".

* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.

* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots().  Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.

* Fixed minor protocol error in action_getconfig() when no requested
categories are found.  Each line needs to be formatted as "Header: text".

* Fixed off-nominal memory leak in manager_build_parked_call_string().

* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().

ASTERISK-24049 #close
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/4315/
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Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Any partially collected DTMF digits for a DTMF hook need to be pushed into
the bridge when a channel leaves the bridging system as if there were a
timeout.

Review: https://reviewboard.asterisk.org/r/4199/
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Merged revisions 428601 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 428602 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

From reviewboard:

"During blind transfer testing, it was noticed that tests were failing
occasionally because the ARI blind transfer event was not being sent.
After investigating, I detected a race condition in the blind transfer
code. When blind transferring a single channel, the actual transfer
operation (i.e. removing the transferee from the bridge and directing
them to the proper dialplan location) is queued onto the transferee
bridge channel. After queuing the transfer operation, the blind transfer
Stasis message is published. At the time of publication, snapshots of
the channels and bridge involved are created. The ARI subscriber to the
blind transfer Stasis message then attempts to determine if the bridge
or any of the involved channels are subscribed to by ARI applications.
If so, then the blind transfer message is sent to the applications. The
way that the ARI blind transfer message handler works is to first see
if the transferer channel is subscribed to. If not, then iterate over
all the channel IDs in the bridge snapshot and determine if any of
those are subscribed to. In the test we were running, the lone
transferee channel was subscribed to, so an ARI event should have been
sent to our application. Occasionally, though, the bridge snapshot did
not have any channels IDs on it at all. Why?

The problem is that since the blind transfer operation is handled by a
separate thread, it is possible that the transfer will have completed and
the channels removed from the bridge before we publish the blind transfer
Stasis message. Since the blind transfer has completed, the bridge on
which the transfer occurred no longer has any channels on it, so the
resulting bridge snapshot has no channels on it. Through investigation of
the code, I found that attended transfers can have this issue too for the
case where a transferee is transferred to an application."

The fix employed here is to decouple the creation of snapshots for the transfer
messages from the publication of the transfer messages. This way, snapshots
can be created to reflect what they are at the time of the transfer operation.

Review: https://reviewboard.asterisk.org/r/4135
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Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 427870 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427873 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a channel is imparted to a bridge, the invocation of the function may
provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart,
the caller must assume that ownership has passed to the function, as in all
paths the function destroys the struct prior to returning (as its purpose is
to configure the behavior of the channel while in the bridge). On one off
nominal path - where the channel already has a PBX thread - the struct was not
being destroyed.

This patch fixes that glitch.

ASTERISK-24437 #close
Reported by: Scott Griepentrog
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Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 426432 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426433 65c4cc65-6c06-0410-ace0-fbb531ad65f3