Most issues were in stringfields and had to do with comparing
a pointer to an constant/interned string with NULL.  Since the
string was a constant, a pointer to it could never be NULL so
the comparison was always "true".  gcc now complains about that.

There were also a few issues where determining if there was
enough space for a memcpy or s(n)printf which were fixed
by defining some of the involved variables as "volatile".

There were also a few other miscellaneous fixes.

ASTERISK-30044

Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570

Documents the Dial syntax for DAHDI, namely the channel group,
distinctive ring, answer confirmation, and digital call options
that are specified in the resource itself.

ASTERISK-24827 #close

Change-Id: Ib95e78497fb00dc5cbfde1c93a69f034bfd08c30

Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio
file.

ASTERISK-29931
Added by Michael Cargile

Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2

The admin_exec function in app_meetme is used by the SLA
applications for internal bridging. However, in these cases,
chan is NULL. Currently, this function will set some status
variables that are intended for a channel, but since channel
is NULL, this is erroneously creating meaningless global
variables, which shouldn't be happening. This sets these
variables only if chan is not NULL.

ASTERISK-30002 #close

Change-Id: I817df6c26f5bda131678e56791b0b61ba64fc6f7

Adds version information for applications, functions,
and manager events/actions.

This is not completely exhaustive by any means but
covers most new things added that have release
versioning information in the issue tracker.

ASTERISK-29940 #close

Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131

When executing dial, the topology of the incoming channel is cloned and
used for the outgoing channel. This creates issues when an incoming
stream is sendonly or recvonly as the stream state of the outgoing
channel will be the same as the stream state of the incoming channel.

Now the stream state is flipped for the outgoing stream in
dial_exec_full if the incoming stream topology is recvonly or sendonly.

ASTERISK-29655
Reported by: Michael Auracher

ASTERISK-29638
Reported by: Michael Auracher

Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1

The ReceiveMF and ReceiveSF applications currently always
return 0, even if a channel has hung up. The call will still
end but generally applications are expected to return -1 if
the channel has hung up.

We now return -1 if a hangup occured to bring this behavior
in line with this norm. This has no functional impact, but
merely increases conformity with how these modules interact
with the PBX core.

ASTERISK-29951 #close

Change-Id: I234d755050ab8ed58f197c6925b968ba26b14033

Adds the m option to the Queue application, which allows a
music on hold class to be specified at runtime which will
override the class configured in queues.conf.

This option functions like the m option to Dial.

ASTERISK-29876 #close

Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7

Currently, if a user tries to access a non-dynamic
MeetMe conference and the conference is not found,
the call simply silent hangs up. There is no indication
to the user that anything went wrong at all.

This changes the relevant debug message to a warning
so that the user is notified of this invalidity.

ASTERISK-29954 #close

Change-Id: Iebcfae3755d00f2150d676ee211c57bc59530048

Adds documentation for all of the possible return values
for the DIALSTATUS variable in the Dial application.

ASTERISK-25716

Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0

This adds a new AMI action called QueueWithdrawCaller.
This AMI action makes it possible to withdraw a caller from a queue,
in a safe and a generic manner.
This can be useful for retrieving a specific call and
dispatching it to a specific extension.
It works by signaling the caller to exit the queue application
whenever it can. Therefore, it is not guaranteed
that the call will leave the queue.

ASTERISK-29909 #close

Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec

Adds the since tag to the documentation DTD so
that individual applications, functions, etc.
can now specify when they were added to Asterisk.

This tag is added at the individual application,
function, etc. level as opposed to at the module
level because modules can expand over time as new
functionality is added, and granularity only
to the module level would generally not be useful.

This enables the ability to more easily determine
when new functionality was added to Asterisk, down
to minor version as opposed to just by major version.
This makes it easier for users to write more portable
dialplan if desired to not use functionality that may
not be widely available yet.

ASTERISK-29896 #close

Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5

Currently, if VoiceMailMain is called with a mailbox, if that
mailbox doesn't exist, then the application silently falls back
to prompting the user for the mailbox, as if no arguments were
provided.

However, if a specific mailbox is requested and it doesn't exist,
then no warning at all is emitted.

This fixes this behavior to now warn if a specifically
requested mailbox could not be accessed, before falling back to
prompting the user for the correct mailbox.

ASTERISK-29920 #close

Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4

mpg123 doesn't support HTTPS, but the MP3Player application
doesn't document this or warn the user about this. HTTPS
streams have become more common nowadays and users could
reasonably try to play them without being aware they should
use the HTTP stream instead.

This adds documentation to note this limitation. It also
throws a warning if users try to use the HTTPS stream to
tell them to use the HTTP stream instead.

ASTERISK-29900 #close

Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827

Adds an option to the ReceiveMF application to allow specifying a
maximum number of digits.

Originally, this capability was not added to ReceiveMF as it was
with ReceiveSF because typically a ST digit is used to denote that
sending of digits is complete. However, there are certain signaling
protocols which simply transmit a digit (such as Expanded In-Band
Signaling) and for these, it's necessary to be able to read a
certain number of digits, as opposed to until receiving a ST digit.

This capability is added as an option, as opposed to as a parameter,
to remain compatible with existing usage (and not shift the
parameters).

ASTERISK-29877 #close

Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b

There are a lot of Queue AMI actions and Queue applications
which do not load queue and queue members from Realtime.

AMI actions
QueuePause - if queue not in memory - response "Interface not found".
QueueStatus/QueueSummary - if queue not in memory - empty response.

Applications:
PauseQueueMember - if queue not in memory
	Attempt to pause interface %s, not found
UnpauseQueueMember - if queue not in memory
	Attempt to unpause interface xxxxx, not found

This patch adds a new function load_realtime_queues
which loads queue and queue members for desired queue
or all queues and all members if param 'queuename' is NULL or empty.
Calls the function load_realtime_queues when needed.

Also this patch fixes leak of ast_config in function set_member_value.

Also this patch fixes incorrect LOG_WARNING when pausing/unpausing
already paused/unpaused member.
The function ast_update_realtime returns 0 when no record modified.
So 0 is not an error to warn about.

ASTERISK-29873 #close
ASTERISK-18416 #close
ASTERISK-27597 #close

Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c

if holdtime is (0 min, 0 sec) there is no hold time announcements
we should then also not playing queue-thankyou

ASTERISK-29831

Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c

gethostbyname() and gethostbyname_r() are deprecated in favor of
getaddrinfo() which we use in the ast_sockaddr family of functions.

ASTERISK-29819 #close

Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7

adding support for playing the correct en/et for nordic languages
by adding 'n' for neuter gender in the relevant ast_say_number

ASTERISK-29827

Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1

Currently, the MP3Player application doesn't
emit a warning if attempting to play a stream
which no longer exists. This can be a common
scenario as many mp3 streams are valid at some
point but can disappear at any time.

Now a warning is thrown if attempting to play
a nonexistent MP3 stream, instead of silently
exiting.

ASTERISK-29829 #close

Change-Id: I53a0bf1ed1740166655eb66fe7675f6f808bf535

Adds missing documentation for some channel,
bridge, and queue events.

ASTERISK-24427
ASTERISK-29515

Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64

Adds tech-agnostic support for SF signaling
by adding SF sender and receiver applications
as well as Dial integration.

ASTERISK-29802 #close

Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad

A previous patch for ASTERISK_29578 caused a 'leak' of
extension state information across queues, causing the
state of the first member of unrelated queues to be
updated in addition to the correct member. Which queues
and members depended on the order of queues in the
iterator.

Additionally, it is possible to use the same 'hint:' on
multiple queue members, so the update cannot break out
of the update loop early when a match is found.

ASTERISK-29806 #close

Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad

added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial

ASTERISK-29795

Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6

added support for playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code

ASTERISK-29797

Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2

Adds a ReceiveText application that can be used in
conjunction with SendText. Currently, there is no
way in Asterisk to receive text in the dialplan
(or anywhere else, really). This allows for Asterisk
to be the recipient of text instead of just the sender.

ASTERISK-29759 #close

Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d

Adds tech-agnostic support for MF signaling by adding
MF sender and receiver applications as well as Dial
integration.

ASTERISK-29496-mf #do-not-close

Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4

Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016

changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING

ASTERISK-28053
Reported by: roadkill

Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89

ASTERISK-29779

Change-Id: If338163488498f65fa7248b60e80299c0a928e4b

Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.

This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.

ASTERISK-29777 #close

Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b

Refactors generic functions used for email generation
into utils.c so that they can be used by multiple
modules, including app_voicemail and app_minivm,
to avoid code duplication.

ASTERISK-29715 #close

Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee

ASTERISK-29740

Change-Id: Icb6fbcfea0a5f1c82caa5001902b6a786adbf307

Fixes a deadlock in app_morsecode caused by locking
the channel twice when reading variables from the
channel. The duplicate lock is simply removed.

ASTERISK-29744 #close

Change-Id: I204000701f123361d7f85e0498fedc90243c75e4

Currently, when the t option is specified with no arguments,
the # character is still treated as a terminator, even though
no character should be treated as a terminator.

This is because a previous regression fix was modified to
remove the use of NULL as a default altogether. However,
NULL and an empty string actually refer to different
arrangements and should be treated differently. NULL is the
default terminator (#), while an empty string removes the
terminator altogether. This is the behavior being used by
the rest of the core.

Additionally, since S_OR catches empty strings as well as
NULL (not intended), this is changed to a ternary operator
instead, which fixes the behavior.

ASTERISK-29705 #close

Change-Id: I9b6b72196dd04f5b1e0ab5aa1b0adf627725e086

Correct typos of the following word families:

simultaneously
administrator
directforward
attachfmt
dailplan
automatically
applicable
nouns
explicit
outside
sponsored
attachment
audio
spied
doesn't
counting
encoded
implements
recursively
emailaddress
arguments
queuerules
members
priority
output
advanced
silencethreshold
brazilian
debugging
argument
meadmin
formatting
integrated
sneakiness

ASTERISK-29714

Change-Id: Ie5ecaec91c00b26309da4e51cfc0991a5bb7d092

If users are able to press # for options while leaving
a message and then press 3 to rerecord the message, if
the caller hangs up during the rerecord prompt but before
Asterisk starts recording a message, then an "empty"
voicemail gets processed whereby an email gets sent out
notifying the user of a 0:00 duration message. The file
doesn't actually exist, so playback will fail since there
was no message to begin with.

This adds a check after the streaming of the rerecord
announcement to see if the caller has hung up. If so,
we bail out early so that we can clean up properly.

ASTERISK-29391 #close

Change-Id: Id965d72759a2fd3b39afb76fec08aaebebe75c31

Add a time_t logintime to storage a time when a member is added into a
queue.

Also, includes show this time (in seconds) using a 'queue show' command
and the field LoginTime for response for AMI events.

ASTERISK-18069 #close

Change-Id: Ied6c3a300f78d78eebedeb3e16a1520fc3fff190

I am adding a mix option that will play by filename and say.conf unlike
say option that will only play with say.conf. It
will look on the format of the name, if it is like say it play with
say.conf if not it will play the file name.

ASTERISK-29662

Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a

If the terminator character is not explicitly specified
and an indications tone is used for reading a digit,
there is no null pointer check so Asterisk crashes.
This prevents null usage from occuring.

ASTERISK-29673 #close

Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac