* Updated unit test as ast_json_name_number() is now NULL tolerant.

ASTERISK-26466 #close
Reported by: Richard Mudgett

Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6

Since the json library does not make the check function public we
recreate/copy the function in our interface module.

ASTERISK-26466
Reported by: Richard Mudgett

Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99

Adds libfftw3 to the build chain that is is going to be used for binaural
synthesis by bridge_softmix.

ASTERISK-26292

Change-Id: Iedc2f174e4ccb39ae5d9e698e339c6a17155867b

When T.38 gets rejected and G711 failback occurs there is a period of
time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set,
leading to a crash.

Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982

The "Q" option will set the cause on the unanswered channels when
another channel answers.  It overrides the default of
ANSWERED_ELSEWHERE.

NOTE:  chan_sip does not support setting the cause on a CANCEL to
anything other than ANSWERED_ELSEWHERE.

ASTERISK-26446 #close

Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47

Small fix. It is necessary to double-check
the index that we just removed because there
is a new element.

ASTERISK-26453 #close

Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7

If a bridge switched to P2P when a DTMF was in progress it
was possible for the DTMF to continue being sent indefinitely.

Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29

Remote asterisk consoles should only display verbose log messages
created by the daemon.  The first patch for ASTERISK-26410 caused
a couple verbose messages to be printed when the rasterisk process
ended.

ASTERISK-26410

Change-Id: Ie2a1bb3753ad2724c0349ec1a336f52f7117b52a

The main frame read and write handlers in main/channel.c don't use the
optimum placement in the processing flow for calling audiohooks
callbacks, as far as codec translation is concerned. This change places
the audiohooks callback code:
 * After the channel read translation if the frame is not linear before
the translation, thereby increasing the chance that the frame is linear
as required by audiohooks
 * Before the channel write translation if the frame is linear at this
point
This prevents the audiohooks code from instantiating additional
translation paths to/from linear where a linear frame format is already
available, saving valuable CPU cycles

ASTERISK-26419

Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f

Fixed a memory leak. It removes only the first element.
Added a useful feature in vector.h to remove all items
under the CMP through a callback function / macro.

ASTERISK-26453 #close

Change-Id: I84508353463456d2495678f125738e20052da950

We use a lot res_calendar, we are very happy with that, especially
because you use libical, the almost alone opensource library that
supports really ical format with all types of recurrency.

Nevertheless, some features are missed for our business use cases.

This first patch adds a new option in calendar.conf:
fetch_again_at_reload. Be my guest for a better name.

If it's true, when you'll launch "module reload res_calendar.so",
Asterisk will download again the calendar.

The business use case is that we have a WebUI with a scheduler planner,
we know when the calendars are modified.

For now, we need to define 1 minute of timeout to have a chance that
our user doesn't wait too long between the modification and the real
test.  But it generates a lot of useless HTTP traffic.


ASTERISK-26422 #close

Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077

Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715

Added tests for bzip2, tar, patch, sed and nm to configure.ac.

Set DOWNLOAD_TO_STDOUT to a working command line regardless of
whether the download program is wget, curl or fetch.

Added a 'configure.m4' file to the third-party directory which takes
care of calling any third-party project setup.  Had to move some
pjproject_bundled stuff up in configure.ac so it was called before
the third-party configure macro.

The pjproject tarball is now downloaded to the externals_cache_dir if
it was specified on the ./configure command line

Removed regeneration of the pjproject aconfigure file.  It was only
needed for an old patch that no longer applies.

Converted the tests for symbols to explicit tests since we know that
they're now available in the bundled version.  Saves a little time
during configure.

ASTERISK-26416 #close
Reported-by: Corey Farrell

Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b
(cherry picked from commit e6b0053d)
(cherry picked from commit a0d02f38)

This change introduced some fax test failures
that have not yet been addressed. So this is
not forgotten I'm submitting a change which
reverts it.

This reverts:
d56fc3b3.

ASTERISK-25629

Change-Id: Ibc2f23c38643f5a2c89cf8915ae2d805b81bc3d5

pjproject_bundled will now use the asterisk memory debugging APIs
if MALLOC_DEBUG is turned on in menuselect.

Because this required stubs for the executable programs and the python
bindings, some Makefile reorganization was needed to properly handle
the dependencies.  As a result, the makefile now individually makes
each of the pjproject libraries separately instead of making them all
in 1 shot.  The only visible change is that there are separate status
lines printed for each library instead oif 1 for all libs.  Also, the
making of the pjproject dependency files was eliminated.  They're not
needed for building unless you're actively modifying pjproject source
files and it makes the build process faster.  Finally, any issues with
parallel builds should be resolved again making the build faster.

Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0

cdr, config and voicemail are all separate alembic trees.  Because
alembic's default is to use a table named 'alembic_version' to store
the current tree revision, the 3 trees can't exist in the same schema
without stepping on each other.

Now each tree uses 'alembic_version_<tree_name>' as the version table.
Each tree's env.py script now first checks for 'alembic_version'.  If
it finds it AND its revision is in the tree's history, the script
renames it to 'alembic_version_<tree_name>'.  Regardless, the script
then continues with the migration using 'alembic_version_<tree_name>'
and creates that table if it's not found.  The result is that if an
existing 'alembic_version' table was found but it didn't belong to this
tree, it's left alone and 'alembic_version_<tree_name>' is used or
created.

WARNING:  If multiple trees are using the same schema, they MUST NOT
CRU or D any objects with names that might exist in the other trees.
An example would be 'yesno_values' type.  If two trees perform
operations on it, one tree could pull it out from under the other.
Thankfully we currently don't share any names among cdr, config and
voicemail.

NOTE:  Since the env.py scripts in each tree were identical, a common
env.py has been placed in the ast-db-manage directory and a symlink
to it has been placed in each tree directory.

ASTERISK-24311 #close
Reported-by: Dafi Ni

Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898

In the SIP channel driver chan_sip, the default is "auto_force_rport". When no
NAT was detected, for example in case of IPv6, Asterisk uses the IP address
from the headers within the SIP-REGISTER for subsequent SIP signaling. When
the remote party specifies support for Symmetric Response (RFC 3581) via the
parameter "rport", Asterisk should not extract the port from the SIP headers
but reuse the port of the transport. This did not happen because of a typo.

ASTERISK-26438 #close

Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6

Asterisk only supports mono audio at the moment.
This patch adds interleaved two-channel audio to Asterisk's channels.

ASTERISK-26292

Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a

* Compile __ast_assert_failed unconditionally.
* Use __ast_assert_failed to log messages from log_bad_ao2
* Remove calls to ast_assert(0) that happen after log_bad_ao2 was run.

Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751

Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636

Previously, when reloading the members of a queue, the members added statically
(i.e. defined in queues.conf) would see their "ringinuse" value updated but not
the members added dynamically.

This change makes dynamic members ringuse value to be updated on reload.

Note that it's impossible to add a dynamic member with a specific ringinuse
value. For both static and dynamic members, the ringinuse value can always be
changed later on with command like "queue set ringinuse" or with the AMI action
"QueueMemberRingInUse". So it's possible this commit could break a user workflow
if he was changing the ringinuse value of dynamic members via such commands and
was also relying on the fact that a queue reload would not update the dynamic
members ringinuse value.

ASTERISK-26330

Change-Id: I3745cc9a06ba7e02c399636f1ee9e58c04081f3f

This reverts commit 40aa2813.

ASTERISK-26426 #close

Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5

This allows asterisk to compiled with LOW_MEMORY to load modules built
without LOW_MEMORY.

ASTERISK-26398 #close

Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d