Follow-up issue to -r430435 from reviewboard review.

ASTERISK-24049
Review: https://reviewboard.asterisk.org/r/4315/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList

* Incremented the AMI version to 2.7.0.

* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start".  The corresponding complete event always used "Complete".

* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.

* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots().  Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.

* Fixed minor protocol error in action_getconfig() when no requested
categories are found.  Each line needs to be formatted as "Header: text".

* Fixed off-nominal memory leak in manager_build_parked_call_string().

* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().

ASTERISK-24049 #close
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/4315/
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Summaries can't contain XML nodes, as they are defined to contain only text
data.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the ability to pass options and a command to MixMontor when
recording a conference using ConfBridge.

New options are -

* record_options: Options to MixMontor, eg: m(), W() etc.
* record_command: The command to execute when recording is over.
* record_file_timestamp: Append the start time to the file name.

These options can also be used with the CONFBRIDGE function, e.g.,
Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))

Review: https://reviewboard.asterisk.org/r/4023

ASTERISK-24351 #close
Reported by: Gareth Palmer
patches:
  record_command-428838.patch uploaded by Gareth Palmer (License 5169)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:

    -out += sprintf(out, "%%%02X", (unsigned char) *ptr);
    +out += sprintf(out, "%%%02X", (unsigned) *ptr);

That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.

This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)

Review: https://reviewboard.asterisk.org/r/4263/

ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
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When the app_meetme module is loaded without its configuration file, the
module settings aren't initialized. In particular, this impacts the use
of logging realtime members. This patch guarantees that we always set the
default module settings on initial load.

Review: https://reviewboard.asterisk.org/r/4242/

ASTERISK-24572 #close
Reported by: Nuno Borges
patches:
  24572.patch uploaded by Nuno Borges (License 6116)
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The UW IMAP library is instrinsically not thread-safe, and relies upon higher
level applications to guarantee thread safety. For the most part, this is
provided by the vms object, which provides locking for individual streams.
Unfortunately, this is not sufficient for calls to mail_open which create the
IMAP stream. mail_open can, on some systems, call into a UW IMAP specific
function for determining the address of a system based on a hostname,
ip_nametoaddr.

In the ip6_unix implementation of this function, static variables are used
to hold parsing buffers. This can cause a crash if multiple threads attempt
to convert a hostname to an address at the same time. Locking on a single
mail stream is not sufficient to prevent simultaneous access to these static
variables.

In the IMAP library, this function can be called from the mail_open and
imap_status functions. As the imap_status function is not used by
app_voicemail, locking on access to mail_open is sufficient to prevent
any mangling of the buffers.

Review: https://reviewboard.asterisk.org/r/4188/

ASTERISK-24516 #close
Reported by: David Duncan Ross Palmer
Tested by: David Duncan Ross Palmer
patches:
  ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660)
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Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.

For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
  a single message - the subscription is created, a message is published, the
  delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.

This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.

Review: https://reviewboard.asterisk.org/r/4193

ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording.

The Record dialplan function trims 1/4 of a second from the end of recordings in case
they are terminated because of DTMF. When hanging up, however, you don't want this to happen.
This change makes it so on hangup this does not occur.

ASTERISK-24530 #close
Reported by: Ben Smithurst
patches:
 app_record_v2.diff submitted by Ben Smithurst (license 6529)

Review: https://reviewboard.asterisk.org/r/4201/
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Confbridge dialplan function permission escalation via AMI and inappropriate
class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan
function when executed from an external protocol (for instance AMI), could
result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord”
could also be used to execute arbitrary system commands without first checking
for system access. The AMI “ConfbridgeStopRecord” has also been updated to
only run under a system authorization.

Asterisk now inhibits the CONFBRIDGE function from being executed from an
external interface if the live_dangerously option is set to no.  Also, the
“ConfbridgeStartRecord” AMI action is now only allowed to execute under a
user with system level access.

ASTERISK-24490
Reported by: Gareth Palmer
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When r428077 was made for ASTERISK-24522, it failed to take into account users
who are neither wait_marked nor end_marked. These users are *also* supposed to
hear the 'leader has left the conference' message. Granted, this behaviour is
a bit odd; however, that is how it used to work... and behaviour changes are
not good.

This patch ensures that if there are any 'normal' users present when the last
marked user leaves the conference, the message will still be played to them.

Note that this regression was caught by the Asterisk Test Suite's
confbridge_nominal test, which has a quirky combination of users.
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Consider the following:
- A marked user in a conference
- One or more end_marked only users in the conference

When the marked users leaves, we will be in the conf_state_multi_marked state.
This currently will traverse the users, kicking out any who have the end_marked
flags. When they are kicked, a full ast_bridge_remove is immediately called on
the channels. At this time, we also unilaterally set the need_prompt flag.

When the need_prompt flag is set, we then playback a sound to the bridge
informing everyone that the leader has left; however, no one is left in the
bridge. This causes some odd behaviour for the end_marked users - they are
stuck waiting for the bridge to be unlocked. This results in them waiting for
5 or 6 seconds of dead air before hearing that they've been kicked.

Unfortunately, we do have to keep the bridge locked while we're playing back
the 'leader-has-left' prompt. If there are any wait_marked users in the
conference, this behaviour can't be easily changed - but we do make the case
of the end_marked users better with this patch.

Review: https://reviewboard.asterisk.org/r/4184/

ASTERISK-24522 #close
Reported by: Matt Jordan
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From reviewboard:

"During blind transfer testing, it was noticed that tests were failing
occasionally because the ARI blind transfer event was not being sent.
After investigating, I detected a race condition in the blind transfer
code. When blind transferring a single channel, the actual transfer
operation (i.e. removing the transferee from the bridge and directing
them to the proper dialplan location) is queued onto the transferee
bridge channel. After queuing the transfer operation, the blind transfer
Stasis message is published. At the time of publication, snapshots of
the channels and bridge involved are created. The ARI subscriber to the
blind transfer Stasis message then attempts to determine if the bridge
or any of the involved channels are subscribed to by ARI applications.
If so, then the blind transfer message is sent to the applications. The
way that the ARI blind transfer message handler works is to first see
if the transferer channel is subscribed to. If not, then iterate over
all the channel IDs in the bridge snapshot and determine if any of
those are subscribed to. In the test we were running, the lone
transferee channel was subscribed to, so an ARI event should have been
sent to our application. Occasionally, though, the bridge snapshot did
not have any channels IDs on it at all. Why?

The problem is that since the blind transfer operation is handled by a
separate thread, it is possible that the transfer will have completed and
the channels removed from the bridge before we publish the blind transfer
Stasis message. Since the blind transfer has completed, the bridge on
which the transfer occurred no longer has any channels on it, so the
resulting bridge snapshot has no channels on it. Through investigation of
the code, I found that attended transfers can have this issue too for the
case where a transferee is transferred to an application."

The fix employed here is to decouple the creation of snapshots for the transfer
messages from the publication of the transfer messages. This way, snapshots
can be created to reflect what they are at the time of the transfer operation.

Review: https://reviewboard.asterisk.org/r/4135
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Currently if the leader of a conference bridge leaves any participant
that has musiconhold enabled will not hear the "leader has left" sound.
This is because musiconhold is started and THEN the sound is played.

This change makes it so that the sound is played and THEN musiconhold
is started. This provides a better experience for users as they may not
have known previously why they went back to musiconhold.

Review: https://reviewboard.asterisk.org/r/4177/
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An issue existed in r420577, which added multiple recipients to voicemail
emails. The patch, when looking at the intended recipients, looked ahead for
the '|' character inside a while loop which already had pulled out the
appropriate field parsing on the '|' character. This would cause it to skip
the recipients.

This patch fixes it such that it relies completely on the while loop to parse
through the e-mail fields.

Note that the original author of the patch looked at this fix and approved it.

ASTERISK-24250 #close
Reported by: abelbeck
patches:
  voicemail-420577-to-comma-fix.diff uploaded by abelbeck (License 5903)
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Made agent able to interrupt the alerting beep playback with DTMF.  Any
digit can interrupt if the call does not need to be acknowledged.  Only
the first digit of the acknowledgement can interrupt if the call needs to
be acknowledged.  The agent interrupting the alerting playback builds on
the ASTERISK-24447 patch because it knows what digit interrupted the
playback and needs to be able to pass that digit to the DTMF hook digit
collection code.

ASTERISK-24257 #close
Reported by: Steve Pitts

Review: https://reviewboard.asterisk.org/r/4123/
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There is no procedure called ast_closeframe, fix code to use
ast_closestream.

Reported By: Matt Jordan
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Fix cleanup in __ast_play_and_record where others[x] may be leaked.
This was caught where prepend != NULL && outmsg != NULL, once
realfile[x] == NULL any further others[x] would be leaked. A cleanup
block was also added for prepend != NULL && outmsg == NULL.

11+: Fix leak of ast_writestream recording_fs in
app_voicemail:leave_voicemail.

ASTERISK-24476 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4138/
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set_member_value has a couple leaks to references in the variable q
found through testsuite tests/queues/set_penalty.  Also remove the
REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible
with the updated REF_DEBUG code.

ASTERISK-24466 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4125/
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In update_messages_by_imapuser(), messages were appended to a finite
array which resulted in a crash when an IMAP mailbox contained more
than 256 entries. This memory is now dynamically increased as needed.

Observe that this patch adds a bunch of XXX's to questionable code. See
the review (url below) for more information.

ASTERISK-24190 #close
Reported by: Nick Adams
Tested by: Nick Adams

Review: https://reviewboard.asterisk.org/r/4126/
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Clean ao2_iterator, resolving reference leak to queue members.

ASTERISK-24454 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4111/
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This patch provides the capability to manipulate templates and categories
with non-unique names via AMI.

Summary of changes:

GetConfig and GetConfigJSON: Added "Filter" parameter:  A comma separated list
of name_regex=value_regex expressions which will cause only categories whose
variables match all expressions to be considered.  The special variable name
TEMPLATES can be used to control whether templates are included.  Passing
'include' as the value will include templates along with normal categories.
Passing 'restrict' as the value will restrict the operation to ONLY templates.
Not specifying a TEMPLATES expression results in the current default behavior
which is to not include templates.

UpdateConfig: NewCat now includes options for allowing duplicate category
names, indicating if the category should be created as a template, and
specifying templates the category should inherit from.  The rest of the
actions now accept a filter string as defined above.  If there are non-unique
category names, you can now update specific ones based on variable values.

To facilitate the new capabilities in manager, corresponding changes had to be
made to config, most notably the addition of filter criteria to many of the
APIs.  In some cases it was easy to change the references to use the new
prototype but others would have required touching too many files for this
patch so a wrapper with the original prototype was created.  Macros couldn't
be used in this case because it would break binary compatibility with modules
such as res_digium_phone that are linked to real symbols.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4033/
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Adding a mixmonitor to a channel causes the bridge to change technologies
from native to simple_bridge so the call can be recorded.  However, when
the mixmonitor is stopped the bridge does not switch back to the native
technology.

* Added unbridge requests to reevaluate the bridge when a channel
audiohook is removed.

* Moved the unbridge request into ast_audiohook_attach() ensure that the
bridge reevaluates whenever an audiohook is attached.  This simplified the
mixmonitor and chan_spy start code as well.

* Added defensive code to stop_mixmonitor_full() in case additional
arguments are ever added to the StopMixMonitor application.

* Made ast_framehook_detach() not do an unbridge request if the framehook
does not exist.

* Made ast_framehook_list_fixup() do an unbridge request if there are any
framehooks.  Also simplified the loop.

ASTERISK-24195 #close
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/4046/
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The QUEUE_GET_CHANNEL function returns the caller's channel name at the
specified position in a queue.

QUEUE_GET_CHANNEL(<queuename>[,<position>])

The queue position parameter defaults to 1 if not specified.

Noop(${QUEUE_GET_CHANNEL(queuename, 2)})
"SIP/peer-00000002", if queue exist and have at least 2 callers

Noop(${QUEUE_GET_CHANNEL(queuename, 1)})
Noop(${QUEUE_GET_CHANNEL(queuename)})
"SIP/peer-00000000", if queue exist and have at least 1 caller

ASTERISK-24365 #close
Reported by: Kristian Hogh
Patches:
      queue_get_firstchannel.patch (license #6639) patch uploaded by Kristian Hogh
      rb4035.patch (license #6639) patch uploaded by Kristian Hogh
      Patch morphed from QUEUE_GET_FIRSTCHANEL to the more general QUEUE_GET_CHANNEL
      on reviewbord.

Review: https://reviewboard.asterisk.org/r/4035/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424493 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This gets rid of most old libc free/malloc/realloc and replaces them
with ast_free and friends. When compiling with MALLOC_DEBUG you'll
notice it when you're mistakenly using one of the libc variants. For
the legacy cases you can define WRAP_LIBC_MALLOC before including
asterisk.h.

Even better would be if the errors were also enabled when compiling
without MALLOC_DEBUG, but that's a slightly more invasive header
file change.

Those compiling addons/format_mp3 will need to rerun
./contrib/scripts/get_mp3_source.sh.

ASTERISK-24348 #related
Review: https://reviewboard.asterisk.org/r/4015/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Changes made during format improvements resulted in the
recording to voicemail option 'm' of the MixMonitor app
writing a zero length duration in the msgXXXX.txt file.

This change introduces a new function ast_ratestream(),
which provides the sample rate of the format associated
with the stream, and updates the app_voicemail function
for ast_app_copy_recording_to_vm to calculate the right
duration.

Review: https://reviewboard.asterisk.org/r/3996/
ASTERISK-24328 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The context/extension in a CDR is generally considered the destination of a
call. When looking at a 2-party call CDR, users will typically be presented
with the following:

context    exten      channel     dest_channel app  data
default    1000       SIP/8675309 SIP/1000     Dial SIP/1000,,20

However, if the Dial actually takes place in a Macro, the current behaviour
in 12 will result in the following CDR:

context    exten      channel     dest_channel app  data
macro-dial s          SIP/8675309 SIP/1000     Dial SIP/1000,,20

The same is true of a GoSub:

context    exten      channel     dest_channel app  data
subs       dial_stuff SIP/8675309 SIP/1000     Dial SIP/1000,,20

This generally makes the context/exten fields less than useful.

It isn't hard to preserve these values in the CDR state machine; however, we
need to have something that informs us when a channel is executing a
subroutine. Prior to this patch, there isn't anything that does this.

This patch solves this problem by adding a new channel flag,
AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a
Macro or a GoSub. The CDR engine looks for this value when updating a Party A
snapshot; if the flag is present, we don't override the context/exten on the
main CDR object. In a funny quirk, executing a hangup handler must *not* abide
by this logic, as the endbeforehexten logic assumes that the user wants to see
data that occurs in hangup logic, which includes those subroutines. Since
those execute outside of a typical Dial operation (and will typically have
their own dedicated CDR anyway), this is unlikely to cause any heartburn.

Review: https://reviewboard.asterisk.org/r/3962/

ASTERISK-24254 #close
Reported by: tm1000, Tony Lewis
Tested by: Tony Lewis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422720 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ConfbridgeList event doesn't include how long the user has been a
member of the conference.  This patch adds Duration (seconds) which
is based on user->chan->answertime.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3955/
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Currently there's no way to tell if a user is an admin or not when receiving
the join, leave, mute, unmute and talking events.  This patch adds that
capability.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3950/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422178 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Kick, mute and unmute were a little inconsistent in their handling of channel
targets.  This patch cleans that up by insuring they all handle the 'all'
target consistently and adds the 'participants' target which acts on
non-admins.  Documentation for kick was also cleaned up as it never
supported partial channel names.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3944/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We need to unlock the audiohook before trying to lock
the channel, since the correct locking order is channel
then audiohook.
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This patch addresses a few issues:

1) The order of Dial events have been changed when performing a call forward.
   The order has now been altered to
    1) Dial begins dialing channel A.
    2) When A forwards the call to B, we issue the dial end event to channel
       A, indicating the dial is being canceled due to a forward to B.
    3) When the call to channel B occurs, we then issue a new dial begin to
       channel B.

2) Call forwards are now reported on the calling channel, not the peer channel.

3) AMI DialEnd events have been altered to display the extension the call is
   being forwarded to when relevant.

4) You can now get the values of channel variables for channels that are not
   currently in the Stasis application. This brings the retrieval of channel
   variables more in line with the rest of channel read operations since they
   may be performed on channels not in Stasis.

ASTERISK-24134 #close
Reported by Matt Jordan

ASTERISK-24138 #close
Reported by Matt Jordan

Patches:
	forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

Review: https://reviewboard.asterisk.org/r/3899
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The same function, meetme_stasis_generate_msg, handles creating and publishing
Stasis message both when there are channels in the MeetMe conference and when
there are no channels in the conference. When the performance improvement was
made to use cached snapshots, this created a situation where Asterisk would
crash: obtaining a cached snapshot is not NULL tolerant.

This patch restores the previous implementation, which used a NULL safe set
of routines to produce a blob containing the channel snapshot (if available)
and information about the MeetMe conference.

ASTERISK-24234 #close
Reported by: Shaun Ruffell
Tested by: Shaun Ruffell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The 'z' option is supposed to disable the dial timeout in the case of a call
forward. Unfortunately, the wrong timeout timer was passed to the do_forward
function, resulting in the option not working.

ASTERISK-24225 #close
Reported by: dimitripietro
Tested by: dimitripietro
patches:
  jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621)
  jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421236 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.

Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.

ASTERISK-24027 #close
Reported by: mjordan
Review: https://reviewboard.asterisk.org/r/3900/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch gives the optional ability to keep queue rules in RealTime. It is
important to note that with this patch:
 (a) Queue rules in RealTime are only examined on module load/reload
 (b) Queue rules are loaded both from the queuerules.conf file as well as the
     RealTime backend
To inform app_queue to examine RealTime for queue rules, a new setting has been
added to queuerules.conf's general section "realtime_rules". RealTime queue
rules will only be used when this setting is set to "yes".

The schema for the database table supports a rule_name, time, min_penalty, and
max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or
'+' literal is provided. Otherwise, the penalties are treated as constants.

For example:
rule_name, time, min_penalty, max_penalty
'default', '10', '20', '30'
'test2', '20', '30', '55'
'test2', '25', '-11', '+1111'
'test2', '400', '112', '333'
'test3', '0', '4564', '46546'
'test_rule', '40', '15', '50'

which would result in :

Rule: default
 - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust
   QUEUE_MIN_PENALTY to 20
Rule: test2
 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
   QUEUE_MIN_PENALTY to 30
 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust
   QUEUE_MIN_PENALTY by -11
 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
   QUEUE_MIN_PENALTY to 112
Rule: test3
 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust
   QUEUE_MIN_PENALTY to 4564
Rule: test_rule
 - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust
   QUEUE_MIN_PENALTY to 15

If you use RealTime, the queue rules will be always reloaded on a module
reload, even if the underlying file did not change. With the option disabled,
the rules will only be reloaded if the file was modified.

Review: https://reviewboard.asterisk.org/r/3607/

ASTERISK-23823 #close
Reported by: Michael K
patches:
  app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-24045
Reported by: Jacob Barber
Review: https://reviewboard.asterisk.org/r/3833/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3