This reverts commit 163e9e53.

Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b

Queue member will get stuck in pending_members if queue calls a device
that is different from the one observed for state changes.

This patch removes members from pending_members as a result of channel stasis
events such as blind or attended transfers and hangup.

ASTERISK-26862 #close

Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727

The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.

Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088

Thanks to Chris Howard for pointing this out on the wiki.

Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705

Dereferencing struct ast_autochan.chan without first calling
ast_autochan_channel_lock() is unsafe because the pointer could change at
any time due to a masquerade.  Unfortunately, ast_autochan_channel_lock()
itself uses struct ast_autochan.chan unsafely and can result in a deadlock
if the original channel happens to get destroyed after a masquerade in
addition to the pointer getting changed.

The problem is more likely to happen with v11 and earlier because
masquerades are used to optimize out local channels on those versions.
However, it could still happen on newer versions if the channel is
executing a dialplan application when the channel is transferred or
redirected.  In this situation a masquerade still must be used.

* Added a lock to struct ast_autochan to safely be able to use
ast_autochan.chan while trying to get the channel lock in
ast_autochan_channel_lock().  The locking order is the channel lock then
the autochan lock.  Locking in the other direction requires deadlock
avoidance.

* Fix unsafe ast_autochan.chan usages in app_mixmonitor.c.

* Fix unsafe ast_autochan.chan usages in app_chanspy.c.

* app_chanspy.c: Removed unused autochan parameter from next_channel().

ASTERISK-26867

Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592

A caller can leave the Queue() application after being bridged with a
member in a few ways:

  * Caller or member hangup
  * Caller is transferred somewhere else (blind or atx)
  * Caller is externally redirected elsewhere

The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.

This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.

ASTERISK-26400 #close
Reported by: Etienne Lessard

Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334

* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.

ASTERISK-24562 #close

Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe

DTMF configuration options for the binaural softmix bridge:
toggle binaural rendering (per channel).

ASTERISK-26292

Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8

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

ast_load_realtime_multientry() returns an ast_config structure whose
ast_categorys are keyed with the empty strings. Several modules were
giving semantic meaning to the category names causing problems at
runtime.

* app_directory: Treated the category name as the mailbox name, and
  would fail to direct calls to the appropriate extension after an
  entry was chosen.

* app_queue: Queues, queue members, and queue rules were all affected
  and needed to be updated.

* pbx_realtime: Pattern matching would never succeed because the
  extension entered by the user was always compared to the empty
  string.

Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7

vm_authenticate doesn't always set the passed ast_vm_user argument, so
we initialize to 0 before passing it in.

ASTERISK-25893 #close
Reported by: Filip Jenicek

Change-Id: Ia3cc0128f93d352ed9add8d5c2f0f7232c2cbe4a

Original patch by John Covert, slight modifications by me.

ASTERISK-17428 #close
Reported by: John Covert
Patches:
	app_voicemail.c.patch (license #5512) patch uploaded by
        John Covert

Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6

When attempting to use VoiceMailPlayMsg with a realtime data backend
the message is located, but never retrieved. This patch adds the
required RETRIEVE and DISPOSE calls that will fetch the message from
the database (and IMAP storage as well for that matter).

Also, removed extraneous make_file call.

ASTERISK-26723 #close

Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c

When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.

This patch adds the 'u' option to Record() to override that behavior.

ASTERISK-18286 #close
Reported by: var
Patches:
	app_record-1.8.7.1.diff (license #6184) patch uploaded by var

Change-Id: Ia1cd163483235efe2db05e52f39054288553b957

ASTERISK-26775 #close

Change-Id: I86de4b1a699d6edc77fea9b70d839440e4088284

* 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 Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c496. This patch
restores that behavior.

This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.

ASTERISK-26115 #close
Reported by: Nasir Iqbal

Change-Id: I8afbd06725e99610e02adb529137d4800c05345d

We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.

* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging.  Made hold the channel lock after the called
party answers while updating the caller channel staging.

* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.

* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.

* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.

Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a

With 500+ queues and a reload every minute, a random queue disappears
upon reload. The cause is mususe of the 'dead' flag. Namely, all queues
were marked dead up front, and then "resurrected" by dropping this flag
for those found in the configuration. But a queue marked dead can be
removed also when control leaves the app entry point on a PBX thread.

With this change, the queue is marked only not found, and at the end of
reload only the queues that are still not found are actually marked as
dead, so the dead flag is never reset, and set only on positively dead
queues.

ASTERISK-26755

Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf

Some (voicemail-related) tests API symlinks beep.gsm and other files
from ast_config_AST_VAR_DIR. It should use ast_config_AST_DATA_DIR.

ASTERISK-26740 #close

Change-Id: Id49c56fb9e16df64b1a2b829693ca7601252df89

QueueLog did not log ringnoanswer when the caller abandoned call
before first timeout. It was impossible to get agent membername
and ringing duration for this short calls. After some discusions
it seems that the best way is to add new event RINGCANCELED,
which is generated after caller hangup during ringing.

ASTERISK-26665

Change-Id: Ic70f7b0f32fc95c9378e5bcf63865519014805d3

Add an application that allows tracking outbound calls
using app_queue.

ASTERISK-19862

Change-Id: Ia0ab64aed934c25b2a25022adcc7c0624224346e

Adds a new formula for SL2 and documentation

ASTERISK-26559

Change-Id: I0970c620460507cd9d45b0d43600779c8915e770

In some cases member is added to pending_members, and the channel
is hung up before any extension state change. So the member would
stay in pending_members forever. So when we call do_hang, we
should also remove member from pending.

ASTERISK-26621 #close

Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54

Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992
that requested ability to add callerid into app_originate.
Comments in that issue suggested that it was better solved by
adding an option to gosub prior to originating the call.  The
attached patch implements this much like app_dial with two
options one to gosub on the originating channel and one to gosub
on the newly created channel and behaves just like app_dial.
I have tested this patch by adding callerid info to the new
channel and also SIPAddHeader (to e.g. add header to force auto
answer) and confirmed it works.  Have also tested both 'exten'
and 'app' versions of app_originate.

Opened by: dkerr
Patch by: dkerr

Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57

fopencookie/funclose is a non-standard API and should not be used
in portable software. Additionally, the way FILE's fd is used in
non-blocking mode is undefined behaviour and cannot be relied on.

This introduces internal abstraction for io streams, that allows
implementing the desired virtualization of read/write operations
with necessary timeout handling.

ASTERISK-24515 #close
ASTERISK-24517 #close

Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85

In 9785e8d0, app_echo was updated to relay video source updates to the
channel for the purposes of displaying video in WebRTC tests.
Unfortunately, this can cause a Kafkaesque nightmare if two or more
Local channels are in a bridge together where their ends are in
app_echo. When this situation occurs, a video update sent into app_echo
will cause the video update to be relayed to the other Local channels,
causing another round of video updates, etc. In not much time at all,
the channel length queues will be overwhelmed, channel alert pipes will
fail, and all hell will break loose as Asterisk merrily continues to
throw more video update requests onto the channels.

This patch updates app_echo to *only* relay a single video update. Once
a video update has been made, all further video updates are dropped.
This meets the intended purpose of the original patch: if we get a video
update and we're in app_echo, go ahead and ask the sender to update
themselves. However, once we've got that video stream sync'd up, don't
keep spamming the world.

Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74

sets the variable ABANDONED to TRUE if the call was not answered.

ASTERISK-26558

Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3

Given the scenario where multiple channels are dialed using Dial()
but the caller is picked up using PickupChan() all outgoing channels
except the channel specified to PickupChan() would be marked
as ringing until the call had been hung up.

When using the PickupChan application the channel executing the
application is swapped into place of another channel. As part
of this process the channel is answered. The Dial application
has explicit logic which checks if the channel is answered,
cancels all other outgoing channels, and bridges. This logic is
different than the normal logic that is executed when an outgoing
channel is answered. This different logic failed to publish dial
events stating that the other outgoing channels had been canceled.
As a result references to the outgoing channels were held onto by
the dial masquerade process until the call had been ended and
the channels had gone away. This would result in the channels
appearing in the "core show channels" list despite not being present
anymore and would also result in incorrect device state.

This change makes it so that this logic also publishes
dial events stating that the other outgoing channels have been
canceled.

ASTERISK-26549

Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f

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

When executing the MailboxExists dialplan application and
MAILBOX_EXISTS dialplan function the passed in temporary voice
mailbox was not cleared, causing it to try to free garbage.

ASTERISK-26503 #close

Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3

Adds setting to confbridge.conf (binaural_active) that determines if binaural
synthesis can be available in bridge_softmix.

ASTERISK-26292

Change-Id: I59dfcb8e55fe1df4ef32045882fea5bb58fc71db

When using Asterisk Realtime Architecture, empty fields are skipped and the
default values are used. If the "context" parameter in queue was set and then
cleared from the database, the old value remains in memory and it continues
to be used. This change initialize the "context" parameter with an empty value,
allowing clearing the parameter.

ASTERISK-26462 #close

Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905

Added needed UTF-8 checks before constructing json objects in various
files for strings obtained outside the system.  In this case string values
from a channel driver's peer and not from the user setting channel
variables.

* aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
object construction.

ASTERISK-26466
Reported by: Richard Mudgett

Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096

Change-Id: I082b239022fac462666e52a14a44304748908dc0

The pause reason is not always cleared when it should be cleared.

* Made set_queue_member_pause() always clear pause reason if not pausing
with a reason string.

Change-Id: I993dad19626ec017478a230e980989438b778c53

The "Q" option will set the cause on the unanswered channels when
another channel answers.  It overrides the default of
ANSWERED_ELSEWHERE.

NOTE:  chan_sip does not support setting the cause on a CANCEL to
anything other than ANSWERED_ELSEWHERE.

ASTERISK-26446 #close

Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47

Previously, when reloading the members of a queue, the members added statically
(i.e. defined in queues.conf) would see their "ringinuse" value updated but not
the members added dynamically.

This change makes dynamic members ringuse value to be updated on reload.

Note that it's impossible to add a dynamic member with a specific ringinuse
value. For both static and dynamic members, the ringinuse value can always be
changed later on with command like "queue set ringinuse" or with the AMI action
"QueueMemberRingInUse". So it's possible this commit could break a user workflow
if he was changing the ringinuse value of dynamic members via such commands and
was also relying on the fact that a queue reload would not update the dynamic
members ringinuse value.

ASTERISK-26330

Change-Id: I3745cc9a06ba7e02c399636f1ee9e58c04081f3f

The output of CLI "queue show" and AMI Queues action is truncated and
"failed to extend from 240 to 327" messages are generated if the queue
member and interface names are lengthy.

* Increase the string buffer size from 240 to 512 in order to accommodate
for more information fields added to the output since v1.8.

ASTERISK-26360 #close
Reported by: Richard Mudgett

Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d

Confbridge announcements tend to block a channel while they are being
played. In some circumstances, this is warranted since you want that
particular channel not to hear the announcement (Example: "John Doe has
entered the conference"). For others it makes less sense.

This change first introduces methods for playing sounds asynchronously
into the conference. This is very similar to how synchronous sounds are
played, except the channel initiating the playback does not wait for the
sound to complete before moving on.

Asynchronous announcements are used for two circumstances:
* Sounds played for a user after they have left the bridge
* Sounds that play first to a single user and then the rest of the
  conference (if the channel and conference use the same language)

ASTERISK-26289 #close
Reported by Mark Michelson

Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a