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

Verbose messages should be printed to the console if the sublevel is
less than option_verbose.  This fix ensures the welcome message with
copyright and license are printed at daemon and interactive rasterisk
startup.

ASTERISK-26410 #close

Change-Id: Ia44235e30ec328aba92ea2c8a837b094e65c9a03

Updated codecs/codecs.xml to add codec_opus to the external
download list.

ASTERISK-26409

Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4
(cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c)
(cherry picked from commit e9684f3acd0e8def0df582c1505dd39dd3fd1610)

Preparation

ASTERISK-26409

Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3
(cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9)

Add Ogg/Opus playback support.

This uses libopusfile in order to be able to read .opus files and play
them back.

Writing/recording support is not present at this time.

ASTERISK-26409

Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955
(cherry picked from commit daee8bbd5209b4158bc1785eede845a26e6cbeaa)

Some external packages have multiple variants that apply to different
builds of asterisk.  The DPMA for instance has a "bundled" variant that
needs to be downloaded if asterisk was configured with
--with-pjproject-bundled.

There are 2 ways to specify variants:

If you need the user to make the decision about which variant to
download, simply create multiple menuselect "member" entries like so...

<member name="res_digium_phone" displayname="..snipped..">
  <support_level>external</support_level>
  <depend>xmlstarlet</depend>
  <depend>bash</depend>
  <defaultenabled>no</defaultenabled>
</member>

<member name="res_digium_phone-bundled" displayname="..snipped..">
  <support_level>external</support_level>
  <depend>xmlstarlet</depend>
  <depend>bash</depend>
  <defaultenabled>no</defaultenabled>
</member>

Note that the second entry has "-<variant>" appended to the name.
You can then use the existing menuselect facilities to restrict which
members to enable or disable.  Youy probably don't want the user to
enable multiple at the same time.

If you want to hide the details of the variants, the better way to
do it is to create 1 member with "variant" elements.

<member name="res_digium_phone" displayname="..snipped..">
  <support_level>external</support_level>
  <depend>xmlstarlet</depend>
  <depend>bash</depend>
  <defaultenabled>no</defaultenabled>
  <member_data>
    <downloader>
      <variants>
        <variant tag="bundled"
          condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/>
      </variants>
    </downloader>
  </member_data>
</member>

The condition must be a bash expression suitable for use with an "if"
statement.  Any environment variable can be used plus those available
in makeopts.

In this case, if asterisk was configured with --with-pjproject-bundled
the bundled variant will be automatically downloaded.  Otherwise the
normal version will be downloaded.

Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e

For the channel driver chan_sip, you specify externhost=example.com in sip.conf
when your Asterisk is behind a NAT and your IP address is assigned dynamically.
Or stated differently: You do not have a static IP address to use "externaddr"
directly. This NAT support is quite handy but just about IPv4. Previously,
Asterisk resolved "externhost" to any IP version. When the first DNS answer
resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and
connection (c=). This happened in outgoing SIP-REGISTER and while answering
SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an
IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost".

ASTERISK-18232 #close
Reported by: Jacek Kowalski
Tested by: Alexander Traud
patches:
 changes.patch submitted by Alessandro Crespi

Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac

Users upgrading from asterisk 13.5 to a later version and who use
realtime with peers that have mailboxes were experiencing runaway
situations that manifested as a continuous stream of taskprocessor
congestion errors, memory leaks and an unresponsive chan_sip.

A related issue was that setting rtcachefriends=no NEVER worked in
asterisk 13 (since the move to stasis).  In 13.5 and earlier, when a
peer tried to register, all of the stasis threads would block and
chan_sip would again become unresponsive.  After 13.5, the runaway
would happen.

There were a number of causes...
* mwi_event_cb was (indirectly) calling build_peer even though calls to
  mwi_event_cb are often caused by build_peer.
* In an effort to prevent chan_sip from being unloaded while messages
  were still in flight, destroy_mailboxes was calling
  stasis_unsubscribe_and_join but in some cases waited forever for the
  final message.
* add_peer_mailboxes wasn't properly marking the existing mailboxes
  on a peer as "keep" so build_peer would always delete them all.
* add_peer_mwi_subs was unsubscribing existing mailbox subscriptions
  then just creating them again.

All of this was causing a flood of subscribes and unsubscribes on
multiple threads all for the same peer and mailbox.

Fixes...
* add_peer_mailboxes now marks mailboxes correctly and build_peer only
  deletes the ones that really are no longer needed by the peer.
* add_peer_mwi_subs now only adds subscriptions marked as "new" instead
  of unsubscribing and resubscribing everything.  It also adds the peer
  object's address to the mailbox instead of its name to the subscription
  userdata so mwi_event_cb doesn't have to call build_peer.

With these changes, with rtcachefriends=yes (the most common setting),
there are no leaks, locks, loops or crashes at shutdown.

rtcachefriends=no still causes leaks but at least it doesn't lock, loop
or crash.  Since making rtcachefriends=no work wasnt in scope for this
issue, further work will have to be deferred to a separate patch.

Side fixes...
 * The ast_lock_track structure had a member named "thread" which gdb
   doesn't like since it conflicts with it's "thread" command.  That
   member was renamed to "thread_id".

ASTERISK-25468 #close

Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0

HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered.
This patch change the call order of ast_queue_control_data
and ast_queue_control in chan_pjsip_incoming_response.

ASTERISK-26396 #close
Reported by: AaronAn
Tested by: AaronAn

Change-Id: Ide2d31723d8d425961e985de7de625694580be61

When retrieving presence state information there is no
guarantee that the subtype and message passed in are
set to NULL. This change ensures they are.

ASTERISK-26397 #close

Change-Id: If38cd730e409e9a9b6eb9adef6591d15a9e61f86

This changes the notice for the deprecation of the old
pooling options to point to the new option for doing
pooling. This gives a clearer direction as to what to
look into.

ASTERISK-26389 #close

Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10

The pooling, shared_connection, limit, and idlecheck options
are no longer used in res_odbc.

ASTERISK-26389

Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6

Routines responsible for managing ast_callid's are overly complicated.
This is left-over code from when ast_callid was an AO2 object.  Now that
it is an integer the code can be reduced.

ast_callid handler code no longer prints it's own error message upon failure
to allocate threadstorage as ast_calloc would have already printed a
message.  Debug messages that were printed when TEST_FRAMEWORK was
enabled have been also been removed.

Change-Id: I65a768a78dc6cf3cfa071e97f33ce3dce280258e

Move the function outside the conditional block that excludes
LOW_MEMORY.

ASTERISK-26273 #close

Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4

Previous versions of Asterisk did not require verbose to be specified in
logger.conf for the console channel, if it was requested by command line
or asterisk.conf it just worked.  This change causes Asterisk to always
enable verbose in the console channel level mask.  Verbose is displayed
on consoles if requested by command line, option_verbose or 'core set
verbose'.

This also delays initialization of the logger until after threadstorage
is initialized.  Initializing too early can cause messages to be printed
multiple times to the console (stdout).

ASTERISK-26391 #close

Change-Id: I52187d67c2fcb3efd5561bf04b3e5e23e5ee8a04

When logger.conf is missing or invalid we should be printing notices,
warnings and errors to the console.  The logmask was incorrectly
calculated.

Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3

Without this change, a 'core restart' would kill the astcanary forever
if you're not running as root. Both with and without this patch, the
scheduling priority was still SCHED_RR after restart.

Additionally, the astcanary is now spawned if you start with high
priority and Asterisk doesn't get a chance to lower it. For example
through: `chrt -r 10 sudo -u asterisk asterisk -c`

Also reap killed astcanary processes on core restart.

ASTERISK-26352 #close

Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55

Previously only the canary checking thread itself had its priority set
to SCHED_OTHER. Now all threads are traversed and adjusted.

ASTERISK-19867 #close
Reported by: Xavier Hienne

Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39

Creating ODBC SQL queries resulted in queries too large to fit into the
supplied buffer.  The resulting truncated buffer contained an invalid SQL
query.

* Made SQL query generation code use a thread storage buffer that can
increase in size as needed.

* Fixed bad multi-line warning messages.

ASTERISK-26263 #close
Reported by: Jeppe Ryskov Larsen

Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae

When receiving an SDP offer with multiple payloads for
the same format we would generate an answer with the first
payload, but during the payload crossover operation
(to set the payloads for receiving) we would remove all
payloads but the last. This would result in incoming
traffic being matched against the wrong format and outgoing
traffic being sent using the wrong payload.

This change makes it so that once a format has a payload
number put into the mapping all subsequent ones are ignored.
This ensures there is only ever one payload in the mapping
and that it is the payload placed into the answer SDP.

ASTERISK-26365 #close

Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60