While the 'interface' column is a NOT NULL, the empty string is still
allowed. res_config_odbc treats the empty string as a NULL and we crash
when trying to dereference.

Also cleaned up an adjacent error message for consistency.

ASTERISK-28168 #close

Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202

This change add ability to set the wrapuptime per-member using the
AddQueueMember application.

The feature to set wrapuptime per member was include in the issue
ASTERISK-27483 for static member by configuration file and was not
added to set from AddQueueMember.

ASTERISK-28055 #close

Change-Id: I7c7ee4a6f804922cd7c42cb02eea26eb3806c6cf

When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.

This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.

This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.

ASTERISK-28277

Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6

You can now define an "aliases" context in voicemail.conf
whose entries point to actual mailboxes.  These can be used anywhere
the mailbox is specified.

Example:
[general]
aliasescontext = myaliases

[default]
1234 = yadayada

[myaliases]
4321@devices = 1234@default

Now you can use 4321@devices to refer to the 1234@default mailbox.

This can be useful to provide channel drivers with constant
mailbox specifications such as <extension>@devices leaving
app_voicemail to control exactly which mailbox the alias points to.
Now, only voicemail has to be reloaded to make changes instead of
individual channel drivers which are usually more expensive to
reload.

Change-Id: I395b9205c91523a334fe971be0d1de4522067b04

If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is
not updated correctly since urgent messages are in a different directory. The
fix is to update the channel variable when the path to the urgent message is
created.

ASTERISK-28225

Change-Id: I8efbace06e6122ea0793f7bdb073d4378e8274ca

When using the 'b' option to Queue with a queue that was not configured
for ring all a crash would occur as the wrong pointer would be used.

ASTERISK-28218

Change-Id: If1390f64e321047dff24fd2410c95dde74904980

The free_user function was automatically deleting the stasis mailbox
state but this only makes sense when the mailbox is actually
deleted, not just the structure freed.  This was causing issues
where leave_voicemail would publish the mwi message to stasis and
delete the state before the message could be processed by
res_pjsip_mwi.

* Removed the delete of state from free_user().

* Created a new free_user_final() function that both frees the data
  structure and deletes the state.  This function is only called
  during module load/unload where it's appropriate to delete the
  state.

ASTERISK-28215

Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd

Currently the file sound_only_person is not played when a marked
user (with announce_only_user=yes) joins an empty conference.

This patch fixes it.

ASTERISK-28201 #close

Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4

Revert commit 6409e7b1, and add
NULL checks for all app_queue event handling code.

Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844

ASTERISK-28125

Change-Id: I37334ea184ebb56e54471496b82937d4927815a0

This reverts commit 29115e23.

That commit closed a long standing hole which allowed subscriptions
to mailboxes that weren't configured in voicemail.conf.  This
caused an issue with FreePBX which depdended on that behavior.
The commit is being reverted until FreePBX can handle the new
behavior.

ASTERISK-28151
Reported by: Ronald Raikes

Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15

* The bridging core no longer uses the stasis cache for bridge
  snapshots.  The latest bridge snapshot is now stored on the
  ast_bridge structure itself.

* The following APIs are no longer available since the stasis cache
  is no longer used:
    ast_bridge_topic_cached()
    ast_bridge_topic_all_cached()

* A topic pool is now used for individual bridge topics.

* The ast_bridge_cache() function was removed since there's no
  longer a separate container of snapshots.

* A new function "ast_bridges()" was created to retrieve the
  container of all bridges.  Users formerly calling
  ast_bridge_cache() can use the new function to iterate over
  bridges and retrieve the latest snapshot directly from the
  bridge.

* The ast_bridge_snapshot_get_latest() function was renamed to
  ast_bridge_get_snapshot_by_uniqueid().

* A new function "ast_bridge_get_snapshot()" was created to retrieve
  the bridge snapshot directly from the bridge structure.

* The ast_bridge_topic_all() function now returns a normal topic
  not a cached one so you can't use stasis cache functions on it
  either.

* The ast_bridge_snapshot_type() stasis message now has the
  ast_bridge_snapshot_update structure as it's data.  It contains
  the last snapshot and the new one.

* cdr, cel, manager and ari have been updated to use the new
  arrangement.

Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369

When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.

This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.

The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.

Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.

ASTERISK-28119

Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423

Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.

As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()

The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.

The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.

The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.

ast_channel_snapshot_get_latest() still returns the latest snapshot.

The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.

ASTERISK-28102

Change-Id: I9334febff60a82d7c39703e49059fa3a68825786

Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.  Remove ao2_container_alloc macro.

Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088

This replaces the inline functions with macros.  This removes the need
to directly use __ao2_ref, opts instead for standard ao2_bump and
ao2_cleanup macros.

Change-Id: If4e04e9bab2e3c883188437cb9f487b3e498a21b

This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.

This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.

There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.

ASTERISK-28103

Change-Id: I99bee23895baa0a117985d51683f7963b77aa190

The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates
from the called parties to the caller.

This patch also blocks updates in the other direction before call is
answered.

ASTERISK-27980

Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01

Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel.  Only bridge_softmix has that
data so now it's set when the bridge topology is changed.

ASTERISK-28107

Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499

* Update the post-answer documentation and example.  The Dial example was
incorrect and misleading for the post-answer subroutine useage.

* Fix note and warning paragraphs in option descriptions.  They don't show
up in the wiki.

Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14

Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push.  This will help
ensure we perform the necessary cleanup upon failure.

Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d

Change-Id: I1946509f518961d23fb21229d91676ee3e441921

Declining the queue_member_status_type stasis message in stasis.conf
causes these messages to leak json objects.

* Add missing ast_json_unref() if the type is NULL in
queue_publish_member_blob().

ASTERISK-28084

Change-Id: I691ecf49bd1f7d9c29182e1eee8c4bb7103be9fc

The first attempt at publishing confbridge events to participants
involved publishing them at the same time stasis events were
created.  This caused issues with bridge and channel locks.  The
second attempt involved publishing them when the stasis events
were received by the code that published the confbridge AMI events.
This caused timing issues because, depending on resources available,
the event could be received before channels actually joined the
bridge and would therefore fail to send messages to the participant.

This attempt reverts to the original mechanism with one exception.
The join and leave events are published via bridge join and leave
hooks.  This guarantees the states of the channels and bridge and
provides deterministic timing for event publishing.

Change-Id: I2660074f8a30a5224cb953d5e047ee84484a9036

This issue related to setting of holdtime, announcements, member delays.
It works well if we set the member delays to "0" and no announcements
and no holdtime.This issue will happen if we set member delays to "1",
"2"... or announcements or holdtime and hangs up the call during
processing it.

And here is the reason:
(At the step of answering a phone.)
It takes care any holdtime, announcements, member delays,
or other options after a call has been answered if it exists.

Normally, After the call has been aswered,
and we wait for the processing one of the cases of the member delays
or hold time or announcements finished, "if (ast_check_hangup(peer))"
will be not executed, then queue will be updated at update_queue().
Here, pending member will be removed.

However, after the call has been aswered,
if we hangs up the call during one of the cases of the member delays
or hold time or announcements, "if (ast_check_hangup(peer))"
will be executed.
outgoing = NULL and at hangupcalls, pending members will not be removed.

* This fixed patch will remove the pending member from container
before hanging up the call with outgoing is NULL.

ASTERISK-27920

Reported by: Cao Minh Hiep
Tested by: Cao Minh Hiep

Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855

app_voicemail wasn't properly cleaning up the stasis cache or the
mwi topic pool when the module was unloaded or when a user was
deleted as a result of a reload.  This resulted in leaks in both
areas.

* app_voicemail now calls ast_delete_mwi_state_full when it frees
  a user structure and ast_delete_mwi_state_full in turn now calls
  the new stasis_topic_pool_delete_topic function to clear the topic
  from the pool.

Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8

The append_mailbox function wasn't calculating the correct length
to pass to ast_alloca and it wasn't handling the case where context
might be empty.

Found by the Address Sanitizer.

Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161

app_voicemail was using the stasis cache to build and maintain a
list of mailboxes that had subscribers.  It then used this list
to determine if a mailbox should be polled for new messages if
polling was enabled.  For this to work, stasis had to cache every
subscription and unsubscription to the mailbox which caused a lot of
overhead, both cpu and memory related.

Since polling is only required when changes are being made to
mailboxes outside of app_voicemail and since the number of mailboxes
that don't have any subscribers is likely to be very low, all
mailboxes are now polled instead of just the ones with subscribers.

This paves the way for disabling the caching of stasis subscription
change messages.

Also fixed cleanup in some of the unit tests that not only left
test users in the users list but also caused segfaults if the tests
were run more than once.

ASTERISK-27121

Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee

This ensures the most up-to-date information is used for the next
call attempt.

ASTERISK-28032

Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce

Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8

I'm only seeing an error in 14+, so I assume it is due to different
compiler options:

app_queue.c: In function ‘handle_queue_add_member’:
app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11
    bytes into a region of size 3 [-Werror=format-overflow=]
     sprintf(num, "%d", state);
                   ^~
app_queue.c:10234:18: note: directive argument in the range
    [-2147483648, 99]
     sprintf(num, "%d", state);
                  ^~~~

Compiler: gcc version 8.0.1 20180414 (experimental)
    [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2) 

Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10

When a call leaves a queue on leaveempty condition, QUEUESTATUS
must be set to LEAVEEMPTY, no matter whether Queue was executed with or
without the "c" (continue) option.

The regression was introduced in the fix for ASTERISK_25665.
The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was
overwritten in case when "c" is set, regardless of what was the cause
for leaving the queue.

ASTERISK-27973 #close
Reported-by: Valentin Safonov

Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c

I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.

ASTERISK-27965

Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7

Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.

Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for.  If it finds it, it adds a "label"
attribute to the current sdp.

Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5

If a conference is ended very quickly after it was created (i.e., the
first user immediately hangs up) then the conference bridge and announcer
channels are not removed.

When a conference is created, the push_announcer() function is added to
the playback queue task processor and the conference object reference is
bumped.  If a conference is ended while the push_announcer() function is
still going then the ao2_cleanup(conference) at the end of
push_announcer() will call the destructor function -
destroy_conference_bridge().

The destroy_conference_bridge() function will then add the
hangup_playback() task to the playback queue and will wait for it to end.
Since it is already a current task of the playback queue it will wait
forever.

This patch makes the conference thread call push_announcer() directly.
This way the conference object reference bump is not needed.  Since the
playback queue task processor is only used by the conference thread
itself, there is no danger of trying to play announcements before the
announcer is pushed to the bridge.

ASTERISK-27870 #close

Change-Id: I947a50fb121422d90fd1816d643a54d75185a477

With the participant info code in app_confbridge, we were still
in the process of adding the channel to the bridge when trying to send
an in-dialog MESSAGE.  This caused 2 threads to grab the channel
blocking flag at the same time.  To mitigate this, the participant
info code was moved to confbridge_manager so it runs after all
channel/bridge actions have finished.

Change-Id: I228806ac153074f45e0b35d5236166e92e132abd

Add predial handler support to app_queue.  app_dial (ASTERISK_19548) and
app_originate (ASTERISK_26587) have the ability to execute predial
handlers on caller and callee channels.  This patch adds predial handlers
to app_queue and uses the same options as Dial and Originate (b and B).
The caller routine gets executed when the caller first enters the queue.
The callee routine gets executed for each queue member when they are about
to be called.

ASTERISK-27912

Change-Id: I5acf5c32587ee008658d12e8a8049eb8fa4d0f24

There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.

ASTERISK-27625

Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905

This patch changes the way asterisk polls output from mpg123, instead
of waiting for 10 seconds(when playing an http url) it now uses a
timeout of one second and iterates 10 times using this same timeout.

The main difference is that for every timeout asterisk receives it now
checks if mpg123 is still running before poll again.

ASTERISK-27752

Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620

ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc.  In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.

For all but the ConfbridgeWelcome event, the JSON message contains
information about the bridge, such as its id and name, and information
about the channel that triggered the event such as channel name,
callerid info, mute status, and the MSID labels for their audio and
video tracks. You can use the labels to correlate callerid and mute
status to specific video elements in a webrtc client.

To control this behavior, the following options have been added to
confbridge.conf:

bridge_profile/enable_events:  This must be enabled on any bridge where
events are desired.

user_profile/send_events:  This must be set for a user profile to send
events.  Different user profiles connected to the same bridge can have
different settings.  This allows admins to get events but not normal
users for instance.

user_profile/echo_events:  In some cases, you might not want the user
triggering the event to get the event sent back to them.  To prevent it,
set this to false.

A change was also made to res_pjsip_sdp_rtp to save the generated msid
to the stream so it can be re-used.  This allows participant A's video
stream to appear as the same label to all other participants.

Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e

There was no real reason to limit the conteny type to text/plain other
than that's what it was limited to before.  Now any text/* content
type will be allowed for channel drivers that don't support enhanced
messaging and any type will be allowed for channel drivers that do
support enhanced messaging.

Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9