The configuration unsigned integer option handler sets flags for the
parser as if the option should be a signed integer (PARSE_INT32),
leading to errors on "out of range" values. Fix flags (PARSE_UINT32).

A fix to res_pjsip is also present which stops invalid flags from
being passed when registering sorcery object fields for qualify
status.

ASTERISK-25612 #close

Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e

This prevents pbx_core from hanging up the channel if the app isn't
registered.

ASTERISK-25846 #close

Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce

Channel masquerading had a conflict with autochannel locking.

When locking autochannel->channel, the channel is fetched from the
autochannel and then locked. During the fetch, the autochannel -- which
has no locks itself -- can be modified by someone who owns the channel
lock. That means that the value of autochan->channel cannot be trusted
until you hold the lock.

In practice, this caused problems with Local channels getting
masqueraded away while the ChanSpy attempted to get info from that
channel. The old channel which was about to get removed got locked, but
the new (replaced) channel got unlocked (no-op). Because the replaced
channel was now locked (and would never get unlocked), it couldn't get
removed from the channel list in a timely manner, and would now cause
deadlocks when iterating over the channel list.

This change checks the autochannel after locking the channel for changes
to the autochannel. If the channel had been changed, the lock is
reobtained on the new channel.

In theory it seems possible that after this fix, the lock attempt on the
old (wrong) channel can be on an already destroyed lock, maybe causing
a crash. But that hasn't been observed in the wild and is harder induce
than the current deadlock.

Thanks go to Filip Frank for suggesting a fix similar to this and
especially to IRC user hexanol for pointing out why this deadlock was
possible and testing this fix. And to Richard for catching my rookie
while loop mistake ;)

ASTERISK-25321 #close

Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def

Fix calculate of average time for talktime is wrong when is completed the
first call beacuse the time for talked would be that call.

ASTERISK-25800 #close

Change-Id: I94f79028935913cd9174b090b52bb300b91b9492

A user cannot set new bridge options after the conference is created by
the first user.  Attempting to do so is documented as undefined behavior.

This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.

Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266

Fix some warnings found with clang.

Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd

Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51

* Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.

* Added Muted header to AMI ConfbridgeListRooms action response list
events to indicate the muted conference state.

* Added Muted column to CLI "confbridge list" output to indicate the muted
conference state and made the locked column a yes/no value instead of a
locked/unlocked value.

ASTERISK-20987
Reported by: hristo

Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1

Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7

ASTERISK-20987 #close
Reported by: hristo

Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38

Add time when started a the last pause for a queue member for
QueueMemberStatus ami event.

Also show accumulate time in seconds when started a pause for a queue
member to CLI command 'queue show'.

ASTERISK-16394 #close

Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c

Change-Id: I2734392b131f1fb0949515d538f83f30fbc15d8c

Change-Id: Idc2ae8a6bd869a66544916906744a5678622262d

When the Asterisk is restared is not preseved reason paused of members.
This patch fixed this cases, retain data on astdb and set when Asterisk
is started.

ASTERISK-25732 #close

Report by: Rodrigo Ramírez Norambuena

Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5

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