This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html



The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint.  Only dynamic contact add/delete actions
update the endpoint.  Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.

This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...

1.  A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message.  The default is 3000ms.  When the timer expires, the contact is
marked unavailable.

2.  Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'.  When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'.  The
existing endpoint events are generated appropriately.

ASTERISK-24863 #close

Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>

This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html



Since we currently have no control over pjproject transaction timeout, this
patch pulls the pjsip_endpt_send_request function out of pjproject and into
res_pjsip/endpt_send_transaction in order to implement that capability.

Now when the transaction is initiated, we also schedule our own pj_timer with
our own desired timeout.

If the transaction completes before either timeout, pjproject cancels its timer,
and calls our tsx callback where we cancel our timer and run the app callback.

If the pjproject timer times out first, pjproject calls our tsx callback where
we cancel our timer and run the app callback.

If our timer times out first, we terminate the transaction which causes
pjproject to cancel its timer and call our tsx callback where we run the app
callback.

Regardless of the scenario, pjproject is calling the tsx callback inside the
group_lock and there are checks in the callback to make sure it doesn't run
twice.

As part of this patch ast_sip_send_out_of_dialog_request was created to replace
its similarly named private function.  It takes a new timeout argument in
milliseconds (<= 0 to disable the timeout).

ASTERISK-24863 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>

Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747

This new macro allows a single line to add all additional
sources to a module.  This helps prevent modules from
missing steps, and makes future changes easier since
they can be made in a single place.

ASTERISK-24960 #close
Reported by: Corey Farrell

Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b

Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e

Add the .gitignore and .gitreview files to the asterisk repo.

NB:  You can add local ignores to the .git/info/exclude file
without having to do a commit.

Common ignore patterns are in the top-level .gitignore file.
Subdirectory-specific ignore patterns are in their own .gitignore
files.

Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69
Tested-by: George Joseph

Prior to this patch, the far_max_datagram value on the UDPTL structure would
remain -1 if the remote endpoint fails to provide the SDP media attribute
T38FaxMaxDatagram. This can result in the INVITE request being rejected. With
this patch, we will now properly initialize the value with either the default
value or with the value provided by pjsip.conf's t38_udptl_maxdatagram
parameter.

Review: https://reviewboard.asterisk.org/r/4589

ASTERISK-24928 #close
Reported by: Juergen Spies
Tested by: Juergen Spies
patches:
  pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698)
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With this patch, chan_pjsip/res_pjsip now sets the native formats to the
codecs negotiated by a call.

* The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native
formats to include all the negotiated audio codecs instead of only the
initial preferred audio codec and later the currently received audio
codec.

* The audio frame handling in channel.c:ast_read() is more streamlined and
will automatically adjust to changes in received frame formats.  The new
policy is to remove translation and pass the new frame format to the
receiver except if the translation was to a signed linear format.  A more
long winded version is commented in ast_read() along with some caveats.

* The audio frame handling in channel.c:ast_write() is more streamlined
and will automatically adjust any needed translation to changes in the
frame formats sent.  Frame formats sent can change for many reasons such
as a recording is being played back or the bridged peer changed the format
it sends.  Since it is a normal expectation that sent formats can change,
the codec mismatch warning message is demoted to a debug message.

* Removed the short circuit check in
channel.c:ast_channel_make_compatible_helper().  Two party bridges need to
make channels compatible with each other.  However, transfers and moving
channels among bridges can result in otherwise compatible channels having
sub-optimal translation paths if the make compatible check is short
circuited.  A result of forcing the reevaluation of channel compatibility
is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc
options take effect consistently now.  It is unfortunate that these two
options are enabled by default and negate some of the benefits to the
changes in channel.c:ast_read() by forcing translation through signed
linear on a two party bridge.

* Improved the softmix bridge technology to better control the translation
of frames to the bridge.  All of the incoming translation is now normally
handled by ast_read() instead of splitting any translation steps between
ast_read() and the slin factory.  If any frame comes in with an unexpected
format then the translation path in ast_read() is updated for the next
frame and the slin factory handles the current frame translation.

This is the final patch in a series of patches aimed at improving
translation path choices.  The other patches are on the following reviews:
https://reviewboard.asterisk.org/r/4600/
https://reviewboard.asterisk.org/r/4605/

ASTERISK-24841 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/4609/
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This patch adds support for automatically detecting the type of DTMF that a
PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
the channel created for an endpoint will attempt to determine if RFC 4733
DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
for the channel will be set to inband.

Review: https://reviewboard.asterisk.org/r/4438

ASTERISK-24706 #close
Reported by: yaron nahum
patches:
  yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
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While investigating other unload issues I realized that the load/unload process 
for the config wizard was pretty ugly so I've refactored it as follows...

When the res_pjsip sorcery instance is created the config_wizard bumps it's own 
module reference to prevent it from unloading while the sorcery instance is 
still active.  When res_pjsip unloads and it's sorcery instance is destroyed, 
the config wizard unrefs itself which then allows itself to unload cleanly.  
Since the config wizard now can't load after res_pjsip or unload before it 
(which should have been the correct behavior all along), I was able to remove 
the chunks of code in both load_module and unload_module that handled that case.

Ran the testsuite tests to insure there were no functional changes and REF_DEBUG 
to insure that Asterisk was shutting down cleanly with no FRACKs or leaks.

Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4610/
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When the ChannelHold event was added, the 'musicclass' parameter was
erroneously removed. This caused the ChannelHold events to be rejected as
they failed model validation. This patch updates the Swagger schema such that
it now properly reflects the event that is being created.

Hooray for tests that catch things like this.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434598 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to 
a missing OBJ_NODATA in an ao2_callback in load_users().  Rather than adding the 
OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator.  
This plugged the leak but exposed an unload order issue between 
res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip.

res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip.  
Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it 
unloads, it's objects are still in the sorcery instance.  When res_pjsip 
unloads, it destroys all its objects including res_pjsip_phoneprov_provider's.  
The phoneprov destructor then attempts to unregister the extension from 
res_phoneprov but because res_phoneprov is already cleaned up, its users 
container is gone and we get a FRACK.

Simple solution, check for the NULL users container before attempting to remove 
the entry. Duh.

Ran tests/res_phoneprov/res_phoneprov_provider.  No leaks in 
res_pjsip_phoneprov_provider and no FRACKs.

Reported-by: Corey Farrell
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4608/
ASTERISK-24935 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434547 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added a new CLI command for res_pjsip that shows both global and system
configuration settings: pjsip show settings

ASTERISK-24918 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/4597/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434528 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new session supplement that handles in-dialog OPTIONS
requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup
for the OPTIONS request would already have been done by the time the
session supplement receives the inbound request.

ASTERISK-24862 #close
Reported by: yaron nahum
patches:
  res_pjsip_dlg_options.c submitted by yaron nahum (License 6676)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes autological comparison warnings in the following:
 * chan_skinny: letohl may return a signed or unsigned value, depending on the
   macro chosen
 * func_curl: Provide a specific cast to CURLoption to prevent mismatch
 * cel: Fix enum comparisons where the enum can never be negative
 * enum: Fix comparison of return result of dn_expand, which returns a signed
   int value
 * event: Fix enum comparisons where the enum can never be negative
 * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be
   negative
 * presencestate: Use the actual enum value for INVALID state
 * security_events: Fix enum comparisons where the enum can never be negative
 * udptl: Don't bother to check if the return value from encode_length is less
   than 0, as it returns an unsigned int
 * translate: Since the parameters are unsigned int, don't bother checking
   to see if they are negative. The cast to unsigned int would already blow
   past the matrix bounds.
 * res_pjsip_exten_state: Use a temporary value to cache the return of
   ast_hint_presence_state
 * res_stasis_playback: Fix enum comparisons where the enum can never be
   negative
 * res_stasis_recording: Add an enum value for the case where the recording
   operation is in error; fix enum comparisons
 * resource_bridges: Use enum value as opposed to -1
 * resource_channels: Use enum value as opposed to -1

Review: https://reviewboard.asterisk.org/r/4533
ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4533.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Without this patch, if a PJSIP endpoint with udptl enabled and authentication
set attempted to use sendFax, the FAX session would fail during setup. This
was because the invite issued in response to being auth challenged would cause
the PJSIP channel performing the FAX to receive a second T38 framehook and
this would cause frames to be consumed in an inappropriate manner.

ASTERISK-24933 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4577/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434431 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes several warnings pointed out by the clang compiler.
* chan_pjsip: Removed check for data->text, as it will always be non-NULL.
* app_minivm: Fixed evaluation of etemplate->locale, which will always
  evaluate to 'true'. This patch changes the evaluation to use
  ast_strlen_zero.
* app_queue:
  - Fixed evaluation of qe->parent->monfmt, which always evaluates to
    true. Instead, we just check to see if the dereferenced pointer
    evaluates to true.
  - Fixed evaluation of mem->state_interface, wrapping it with a call to
    ast_strlen_zero.
* res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.

Review: https://reviewboard.asterisk.org/r/4541

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4541.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If we receive a MESSAGE request that we cannot send a response
to, we should not send the incoming MESSAGE to the dialplan.

This commit should help the bouncing message_retrans test to
pass consistently.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For some applications - such as SLA - a phone pressing hold should not behave
in the fashion that the Asterisk core would like it to. Instead, the hold
action has some application specific behaviour associated with it - such as
disconnecting the channel that initiated the hold; only playing MoH to channels
in the bridge if the channels are of a particular type, etc.

One way of accomplishing this is to use a framehook to intercept the
hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
accomplishes that using a new dialplan function, HOLD_INTERCEPT.

In addition, some general cleanup of raising hold/unhold Stasis messages was
done, including removing some RAII_VAR usage.

Review: https://reviewboard.asterisk.org/r/4549/

ASTERISK-24922 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434217 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Clang will flag errors when a char pointer is set to '\0', as opposed to a
value that the char pointer points to. This patch fixes this warning
in a variety of locations.

Review: https://reviewboard.asterisk.org/r/4551

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4551.patch submitted by dkdegroot (License 6600)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434189 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When setting the configuration option 'timers' equal to 'no' the bit flag was
not properly negated. This patch clears all associated flags and only sets the
specified one. pjsip will handle any necessary flag combinations. Also went
ahead and did similar for the '100rel' option.

ASTERISK-24910 #close
Reported by: Ray Crumrine
Review: https://reviewboard.asterisk.org/r/4582/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes a warning caught by clang, in which it detected that large
chunks of extconf were unused. Frankly, I wish we could pretend that all of
extconf was unused, but alas, that is not yet the case.

A few extraneous functions in the parking tests were removed as well, for
the same reason.

Review: https://reviewboard.asterisk.org/r/4553

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4553.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434099 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds NAPTR record allocation and sorting, as well as
unit tests that verify that NAPTR records are parsed and
sorted correctly.

Review: https://reviewboard.asterisk.org/r/4542



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434068 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_pjsip_phoneprov_provider is using ao2_callback with OBJ_MULTIPLE, then
ignoring the return.  OBJ_NODATA flag was to prevent a reference leak, but
this caused the module to FRACK on unload.  Revert change until this can
be investigated further.

ASTERISK-24935
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4578/
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This is a change to align behavior with that of Asterisk 11 and previous versions.
In those versions, if a parked call were retrieved, and the call ended, the parked
call retriever would be hung up after the ParkedCall application ran. Prior to this
patch, in Asterisk 13, the same situation would result in the parked call retriever
falling through to additional priorities in the extension where the ParkedCall
application was called. With this patch, the behavior between Asterisk 11 and 13
aligns.

ASTERISK-24899 #close
Reported by Malcolm Davenport
Patches:
	ASTERISK-24899.patch uploaded by Mark Michelson(license #5049)
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res_pjsip_phoneprov_provider was using ao2_callback with OBJ_MULTIPLE, then
ignoring the return.  Added OBJ_NODATA flag to prevent a reference leak.

ASTERISK-24935 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4578/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Outbound SIP MESSAGEs had the potential to be sent out
of order from how they were specified in a set of
dialplan steps.

This change creates a serializer for sending outbound
MESSAGE requests on. This ensures that the MESSAGEs are
sent by Asterisk in the same order that they were sent
from the dialplan.

ASTERISK-24937 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/4579
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds support for parsing SRV records and consuming their values
in an easy fashion. It also adds automatic sorting of SRV records according
to RFC 2782.

Tests have also been included which cover parsing, sorting, and off-nominal
cases where the record is corrupted.

ASTERISK-24931 #close
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/4528/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes several warnings caught by clang - in this case, usage of the
abs function on non-integer values. This patch uses labs and fabs, as
appropriate, in the various affected files.

Review: https://reviewboard.asterisk.org/r/4525

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4525.patch submitted by dkdegroot (License 6600)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes some invalid enum conversion warnings caught by clang. In
particular:
* chan_sip: Several functions mixed usage of the st_refresher_param
  enum and st_refresher enum. This patch corrects the functions to use the
  right enum.
* chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state.
* strings: Fixed incorrect usage of AO2 flags with strings container.
* res_stasis: Change a return enumeration to stasis_app_user_event_res.

Review: https://reviewboard.asterisk.org/r/4535

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4535.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Clang will treat ((a == b)) as a warning, as it reasonably expects that the
developer may have intended to write (a == b) or ((a = b)). This patch cleans
up all instances where equality, not assignment, was intended between two
parantheses.

Review: https://reviewboard.asterisk.org/r/4531/

ASTERISK-24917
Repoted by: dkdegroot
patches:
  rb4531.patch submitted by dkdegroot (License 6600)
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Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes clange compiler warnings for initializer overrides.
Specifically:

res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration
value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing
those enum values, we therefore initialize the value twice to two different
values, "tlsv1" and "default". This patch changes it to just initialize
the index in the array to "tlsv1".

Review: https://reviewboard.asterisk.org/r/4539/

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4539.patch submitted by dkdegroot (License 6600)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433685 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage.

* Converted the contact_autoexpire container to use the ao2 template hash
and cmp functions.  Also made use the OBJ_SEARCH_xxx names instead of the
deprecated names.

* Eliminates several unnecessary uses of RAII_VAR().

Review: https://reviewboard.asterisk.org/r/4524/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk had an issue where retransmissions of MESSAGE requests resulted in
Asterisk processing the retransmission as if it were a new MESSAGE request.

This patch fixes the issue by creating a transaction in PJSIP on the incoming
request. This way, if a retransmission arrives, the PJSIP transaction layer
will resend the response and Asterisk will not ever see the retransmission.

ASTERISK-24920 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/4532/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433620 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Contact expiration object refs were leaked when the module was unloaded.

* Made empty the scheduler of entries before destroying it to release the
object ref held by the scheduler entry.

Review: https://reviewboard.asterisk.org/r/4523/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch updates the kqueue timing module to conform to current timer API.

This fixes issues with using the kqueue timing source on Asterisk 13 on
FreeBSD 10. These issues include:

- Remove support for kevent64().  The values used to support Asterisk timers
  fit within 32bits and so can be handled on all platforms via kevent().

- Provide debug logging for, but do not track, unacked events.  This matches
  the behavior of all other timer implementations.

- Implement continuous mode by triggering and leaving active, a user event.
  This ensures that the file descriptor for the timer returns immediately from
  poll(), without placing the load of a high speed timer on the kernel.

- In kqueue_timer_get_max_rate(), don't overstate the capability of the timer.
  On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer
  type kqueue supports for timers.

- In kqueue_timer_get_event(), assume the caller woke up from poll() and just
  return the mode the timer is currently in. This matches all other timer
  implementations.

- Adjust the test code now that unacked events are not tracked.

Review: https://reviewboard.asterisk.org/r/4465/

ASTERISK-24857 #close
Reported by: scsiguy
Tested by: Ed Hynan
patches:
  rb4465.patch submitted by scsiguy (License 6692)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433575 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
  caused by running PJSIP functions from non-PJSIP threads.
* Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
  crashes in some cases.  In theory pj_shutdown() should take care of this.
* Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
  shutdown.
* Resolve leaked config global in res_pjsip_notify.
* Unregister pubsub pjsip service module.
* Implement cleanup for res_pjsip_session.

ASTERISK-24731 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4498/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* In res/res_sorcery_realtime.c: Broke long line.

* In main/bucket.c: Eliminated unnecessary NULL check as
ast_sorcery_unref() is NULL tolerant and set the global object to NULL
after unref in the system shutdown bucket_cleanup().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433421 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Due to a quirk in the configuration handling of res_xmpp, the 'autoregister'
setting was never actually processed. This was due to not properly copying
over the global settings to the client settings when applying the
configuration to the run-time object.

Review: https://reviewboard.asterisk.org/r/4496/

ASTERISK-14233
ASTERISK-24780 #close
Reported by: Simon Arlott
patches:
  asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds an abstracted core DNS API which resembles the API described
here[1]. The API provides a pluggable mechanism for resolvers and also a
consistent view for records. Both synchronous and asynchronous queries are
supported.

This change also adds a res_resolver_unbound module which uses the libunbound
library to provide resolution.

Unit tests have also been written for all of the above to confirm the API and
functionality.

ASTERISK-24834 #close
Reported by: Matt Jordan

ASTERISK-24836 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/4474/
Review: https://reviewboard.asterisk.org/r/4512/

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens.  If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.

Consequences of these unnecessary messages:

* The caller can start hearing ringback before the far end even gets the
call.

* Many phones tend to grab the first connected line information and refuse
to update the display if it changes.  The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.

When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled.  When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.

* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages.  The default is "no" to disable sending the
unnecessary messages.

ASTERISK-24781 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4473/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3