Fix some warnings found with clang.

Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd

Member lastcall time is updated later than member status. There was chance to
check wrapuptime for available member with wrong (old) lastcall time.
New boolean flag "in_call" is set to true right before connecting call, and
reset to false after update of lastcall time. Members with "in_call" set to true
are treat as unavailable.

ASTERISK-19820 #close

Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500

The menuselect conflict between app_voicemail and res_mwi_external
makes it hard to package 1 version of Asterisk.  There no actual
build dependencies between the 2 so moving this check to runtime
seems like a better solution.

The ast_vm_register and ast_vm_greeter_register functions in app.c
were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there
is already a voicemail module registered. The modules' load_module
functions were then modified to return DECLINE instead of -1 to the
loader.  Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE,
the modules were incorrectly causing Asterisk to stop so this needed
to be cleaned up anyway.

Now you can build both and use modules.conf to decide which voicemail
implementation to load.

The default menuselect options still build app_voicemail and not
res_mwi_external but if both ARE built, res_mwi_external will load
first and become the voicemail provider unless modules.conf rules
prevent it.  This is noted in CHANGES.

Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247

If a caller hangs up before dial is executed within an AGI then the AGI
has likely eaten all queued frames before executing the dial in DeadAGI
mode.  With the caller hung up and no pending frames from the caller's
read queue, dial would not know that the call has hung up until a called
channel answers.  It is rather annoying to whoever just answered the
non-existent call.

Dial should not continue execution in DeadAGI mode, hangup handlers, or
the h exten.

* Added a check early in dial to abort dialing if the caller has hungup.

ASTERISK-25307 #close
Reported by: David Cunningham

Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418

- The maximum_number_of_words was previously documented as being
the number of words that when exceeded, would result in the AMD
application returning that the audio represents a machine.

This was inconsistent with its actual functionality - it was
a number of words that when REACHED, would result in determination
as a machine.

This update corrects the functionality to match the previously
documented functionality.  This is a backwards incompatible change
in configuration file, and has been added to UPGRADE.txt as a result.

The sample configuration file and application defaults have been updated
so that the default value is now 2, which reflects the same default
functionality as previous versions.

- Update documentation for silence_threshold, which previously implied
that it was measuring time, rather than noise averages in the sample.

- Update the comments in amd.conf.sample.

ASTERISK-25639 #close
Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093

If a call enters on a queue and the members on that queue are updated in
realtime (ex: using mysql inserting a new agent) the queue members are
never refreshed and the call will stay in the queue until other event occurs.
This happens only if this is the first call of the queue and there is no
agents servicing.
This patch prevent this issue, ensuring realtime members are updated if
there is one call in the queue and no available agents

ASTERISK-25442 #close

Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682

The default value was never set for audio_buffers, causing bad
audio quality.  This ensures the default is always set.

ASTERISK-25569 #close

Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44

Add value of pause reason when is paused on CLI command "queue show"

ASTERISK-25581 #close

Report by: Rodrigo Ramírez Norambuena

Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864

To be able to barge into a call by dialling a prefix+extension that maps
to the extensions device.

Senario is that DECT headset users may be away from their desks and need
to transfer the call, the goal is that from any phone they dial a prefix
then their extension and are added to the bridge that they are in, from
there they can drop the headset call, as it's also on the handset,
and transfer the caller.

The dialplan would look like, where prefix=73, extension = 8512;
exten => _738512,1,BridgeAdd(SIP/cisco0001)

ASTERISK-25551 #close
Reported By: Alec Davis

Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540

Implemented support for the StatsD sample rate parameter,
which is a parameter for determining when to send computed
statistics to a client.

Valid sample rate values are:
Less than or equal to 0.0 will never be sent.
Between 0.0 and 1.0 will randomly be sent.
Greater than or equal to 1.0 will always be sent.

ASTERISK-25419
Reported By: Ashley Sanders

Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52

commit aae45acb (Mark Michelson 2015-04-15 10:38:02 -0500 6525)
refer ASTERISK-24958

above commit removed ast_channel_lock(qe->chan);
but failed to remove corresponding ast_channel_unlock(qe->chan);

ASTERISK-25561 #close
Reported Alec Davis

Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a

This option adds the ability to specify a timeout, in seconds, for a
participant in a ConfBridge. When the user's timeout has been reached,
the user is ejected from the conference with the CONFBRIDGE_RESULT
channel variable set to "TIMEOUT".

The rationale for this change is that there have been times where we
have seen channels get "stuck" in ConfBridge because a network issue
results in a SIP BYE not being received by Asterisk. While these
channels can be hung up manually via CLI/AMI/ARI, adding some sort of
automatic cleanup of the channels is a nice feature to have.

ASTERISK-25549 #close
Reported by Mark Michelson

Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98

ASTERISK-25533 #close

Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a

Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7

Added code to allow the StatsD dialplan application to
send data to the server specified in statsd.conf.

ASTERISK-25419

Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922

Added code to accept user input and validate it before
allowing it to be sent to the StatsD server.

ASTERISK-25419
Reported By: Ashley Sanders

Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3

In app_queue added value Paused Reason on QueueMemberStatus when a member
on queue is paused and the reason was set.

ASTERISK-25480 #close
Reporte by: Rodrigo Ramírez Norambuena

Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e

Wrote the skeleton framework for the Asterisk StatsD dialplan
application. This includes a load function, unload function, a
callback for execution, and XML documentation.

ASTERISK-25419
Reported By: Ashley Sanders

Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363

* When a call is answered and the outgoing channel name has changed then
force a connected line update because the channel is no longer the same.
The channel was masqueraded into by another channel.  This is usually
because of a call pickup.

Note: Forwarded calls are handled in a controlled manner so the original
channel name is replaced with the forwarded channel.

ASTERISK-25423 #close
Reported by: John Hardin

Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172

Replace inlined code with update_connected_line_from_peer().

ASTERISK-25423
Reported by: John Hardin

Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3

While the 'A' option is playing the announcement file allow the caller and
peer to exchange COLP update frames.

ASTERISK-25423
Reported by: John Hardin

Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9

* When a call is answered and the outgoing channel name has changed then
force a connected line update because the channel is no longer the same.
The channel was masqueraded into by another channel.  This is usually
because of a call pickup.

Note: Forwarded calls are handled in a controlled manner so the original
channel name is replaced with the forwarded channel.

ASTERISK-25423
Reported by: John Hardin

Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c

Replace inlined code with update_connected_line_from_peer().

ASTERISK-25423
Reported by: John Hardin

Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091

Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54

Page uses the async method of dialing with the dial API.  When a call gets
forwarded there is no calling channel available.  If the predial handler
was set then the calling channel could not be put into auto-service
for the forwarded call because it doesn't exist.  A crash is the result.

* Moved the callee predial parameter string processing to before the
string is passed to the dial API rather than having the dial API do it.
There are a few benefits do doing this.  The first is the predial
parameter string processing doesn't need to be done for each channel
called by the dial API.  The second is in async mode and the forwarded
channel is to have the predial handler executed on it then the
non-existent calling channel does not need to be present to process the
predial parameter string.

* Don't start auto-service on a non-existent calling channel to execute
the predial handler when the dial API is in async mode and forwarding a
call.

ASTERISK-25384 #close
Reported by: Chet Stevens

Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981

The RECORDED_FILE variable is empty unless a '%d' is specified in the filename.
This patch makes it so the variable is always set to the filename.

ASTERISK-25410 #close

Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653

When a queued caller transfers an agent to another extension sometimes the
raised AgentComplete event has a reason of "caller" and sometimes "transfer".
Since a transfer has taken place this should always be transfer. This occurs
because sometimes the stasis hangup event arrives before the transfer event
thus writing a different reason out.

With this patch, when a hangup event is received during a transfer it will
check to see if the channel that is hanging up is part of a transfer. If so
it will return and let the subsequently received transfer event handler take
care of the cleanup.

ASTERISK-25399 #close

Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d

During some transfer scenarios involving queues Asterisk would sometimes
crash when trying to obtain a channel snapshot (could happen on caller or
member channels). This occurred because the underlying channel had already
disappeared when trying to obtain the latest snapshot.

This patch adds a reference to both the member and caller channels that
extends to the lifetime of the queue'd call, thus making sure the channels
will always exist when retrieving the latest snapshots.

ASTERISK-25185 #close
Reported by: Etienne Lessard

Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128

Setting the 'paused' and 'ringinuse' options on a queue member using the
dialplan function QUEUE_MEMBER did not behave the same way as the
equivalent dialplan applications or AMI actions.

* Made queue_function_mem_write() call the set_member_paused() and
set_member_value() for the 'paused' and 'ringinuse' options respectively.
A beneficial side effect is that the queue name is now optional and sets
the value in all queues the interface is a member.

* Update QUEUE_MEMBER XML documentation.

* Fix error checking in QUEUE_MEMBER() write.

ASTERISK-25215 #close
Reported by: Lorne Gaetz

Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb

* Extract set_queue_member_pause() from set_member_paused() for simpler
and more consistent code.

* Extract set_queue_member_ringinuse() from
set_member_ringinuse_help_members() for simpler code.

Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306

Change-Id: I7294e13d27875851c2f4ef6818adba507509d224

Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian
(GameGamer43) for pointing that out.

Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106

Currently when requesting a channel the native formats of the
calling channel are provided to the core for usage when dialing
the outbound channel. This occurs without holding the channel lock
or keeping a reference to the formats. This is problematic as
the channel driver may end up changing the formats during this time.
In the case of chan_sip this happens when an SDP negotiation
completes.

This change makes it so app_dial keeps a reference to the native
formats of the calling channel which guarantees that they will
remain valid for the period of time needed.

ASTERISK-25172 #close

Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db

The voicemail.conf mailbox key/value pair is defined as:
<mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]]
Where all fields in the value including the field values are optional.

Since the parsing code for the mailbox key/value pair is sloppy, this
patch tightens the parsing for the directory information.

* Renamed the 'pos' and 'bufptr' variables to 'name' and 'options'
respectively in search_directory_sub().  Those names make more sense.

* Made sure that search_directory_sub() is dealing with the voicemail.conf
mailbox options field if it even exists when looking for the 'hidefromdir'
and 'alias' options.

* Fix crash if a voicemail.conf mailbox is just
<mailbox>=<password>,<name> when the 'a' option is used.  If there were no
fields after the name then the 'options' pointer was not checked for NULL.

* Fix users.conf alias processing if the 'a' option is used.  The wrong
variable was used.

ASTERISK-25087 #close
Reported by: Chet Stevens

Change-Id: I86052ea77307beddddba5279824d39dc0d593374

Although ast_context_find, ast_context_find_or_create and
ast_context_destroy perform locking of the contexts table,
any context pointer can become invalid at any time that the
contexts table is unlocked. This change adds locking around
all complete operations involving these functions.

Places where ast_context_find was followed by ast_context_destroy
have been replaced with calls ast_context_destroy_by_name.

ASTERISK-25094 #close
Reported by: Corey Farrell

Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa

If a channel hangs up while an audio file is playing, there's
no need to clutter up the logs with a warning so suppress it
if ast_check_hangup returns true.

Also, change warning to debug/2 in file.c if writing a frame
fails.  Same reasoning.

Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>

Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723

When completing voicemail playback of a message in the 'INBOX', the
message gets moved to the 'Old' messages folder. Without this patch, if
the 'Old' folder is already at its set limit, then the 'INBOX' message will
simply be deleted. With this patch, the flag to delete the message will be
removed if the save_to_folder function indicates that the message could
not be moved due to a full folder.

ASTERISK-25082 #close
Reported by: Jonathan Rose
Review: https://gerrit.asterisk.org/#/c/448/

Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f

This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3
parameters: position, original position and waiting time.

ASTERISK-25038 #close
Reported by: Etienne Lessard

Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618

* The REF_DEBUG compiler flag no longer has any effect on code that uses
  Astobj2.  It is used to determine if reference debugging is enabled by
  default.  Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
  This was possible now that we no longer require a dual ABI.

ASTERISK-24974 #close
Reported by: Corey Farrell

Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1