This makes the 'bt' parameter unconditional for ast_store_lock_info and
ast_remove_lock_info.  The 'bt' parameter is unused when HAVE_BKTR is
undefined.

Change-Id: Ieced0e920928b735a39c3b5952b806c473d67453

ASTERISK-23556
Reported by: Marcello Ceschia

Change-Id: Ic27e88e0336a0d83877dc857938659dc5560b93c

On second run the config_hook test was unexpectedly failing to load
test_config.conf because it was still unmodified since the last load.
This is fixed by not passing CONFIG_FLAG_FILEUNCHANGED for the initial
loads, only using it when we are tested that a reload of unmodified
files do not initiate the hook.

ASTERISK-25960

Change-Id: Ifd679509a23ed163e5cc647490bf7df4ae3cd856

Change-Id: Ib0bc95fd0ec288c78c313823254d7a84ebfc4429

This matches the behavior of the other SIP channel driver, chan_pjsip.

ASTERISK-27365

Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6

Update patches included in bundled PJPROJECT for the new version.

ASTERISK-27355

Change-Id: I9ac5dbbffaadca25ad24fac8b9ab615e5ace6083

Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.  This required a reorder of the module_load so that
the non-reversable pjsip functions are not called until all potential
errors have been ruled out.

ASTERISK-24483

Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7

Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.

ASTERISK-24483

Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d

When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'.  This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.

ASTERISK-23462 #close

Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318

* Mark the module deprecated.
* Disable the module by default.
* Produce a warning the first time a macro is used.
* Note deprecation related options in app_dial and app_queue.

ASTERISK-27350

Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc

Beside allowing AES-GCM again, this adds AES-192 again.

ASTERISK-27356

Change-Id: Ia97a435faf26300335d9552fa676b5d17e5f7233

As channels join and leave an SFU the bridge_softmix module
needs to renegotiate to add and remove their streams from
the other participants. Previously this was done by constructing
the ideal stream topology every time but in the case of leave
this was incomplete.

This change makes it so bridge_softmix keeps an ideal stream
topology for each channel and uses it when making changes. This
ensures that when we request a renegotiation we are always
certain that we are aiming for the best stream topology
possible. In the case of a channel leaving this ensures that
we try to have an existing participant fill their place if
a participant has a fixed limit on the maximum number of video
streams they allow.

ASTERISK-27354

Change-Id: I58070f421ddeadd2844a33b869b052630cf2e514

* Rename the Party A CDR container from active_cdrs_by_channel to
active_cdrs_master.

* Renamed the support functions associated with active_cdrs_master
appropriately.

ASTERISK-27335

Change-Id: I6104bb3edc3a0b7243ce502e45e8832b0cff14f7

The CDR performance gets worse the further it gets behind in processing
stasis messages.  One of the reasons is because of a n*m loop used when
processing Party B information.

* Added a new CDR container that is keyed to Party B so we don't need such
a large loop when processing Party B information.

NOTE: To reduce the size of the patch I deferred to another patch the
renaming of the Party A active_cdrs_by_channel container to
active_cdrs_master and renaming the container's hash and cmp functions
appropriately.

ASTERISK-27335

Change-Id: I0bf66e8868f8adaa4b5dcf9e682e34951c350249

when 'all' is specified in an allow or disallow section, it should erase
all values from the inverse section in the default config. E.G.
allow=all should erase any deny values from default config &
vice-versa

ASTERISK-27333 #close

Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6

The sys/sysmacros.h include file does not exist in BSD systems and
is not required to build this module there.
Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section
already exist I moved that include line inside it's #else branch.

ASTERISK-27343 #close

Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1

PJSIP allows a domain name as external_media_address. This allows chan_pjsip to
be used behind a NAT with changing IP addresses. The IP address of that domain
is resolved to the c= line already. This change sets also the o= line to that
domain.

ASTERISK-27341 #close

Change-Id: I690163b6e762042ec38b3995aa5c9bea909d8ec4

When making channels compatible the bridge_simple module
will renegotiate one to better match the other. Some
endpoints incorrectly terminate the call if this process
fails.

To better handle this scenario the audio streams present
on the new requested topology will include any existing
negotiated formats that happen to exist on the first
valid audio stream. This ensures formats are persent that
are known to be acceptable to the remote endpoint.

ASTERISK-27259

Change-Id: I8fc0cc03e8bcfd0be8302f13b9f32d8268977f43