ChangeLog

	  arguments are all integers and the response is
	  a whole number, in which case the user is likely
	  expecting an integer. This casts the float to an integer
	  before printing into the response buffer if possible.

	  ASTERISK-29495

	  Change-Id: I902d29eacf3ecd0f8a6a5e433c97f0421d205488

2021-08-04 09:46 +0000 [9cac1c16da]  Naveen Albert <asterisk@phreaknet.org>

	* app_morsecode: Add American Morse code

	  Previously, the Morsecode application only supported international
	  Morse code. This adds support for American Morse code and adds an
	  option to configure the frequency used in off intervals.

	  Additionally, the application checks for hangup between tones
	  to prevent application execution from continuing after hangup.

	  ASTERISK-29541

	  Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4

2021-08-04 14:16 +0000 [3eec5b8c5c]  Naveen Albert <asterisk@phreaknet.org>

	* func_scramble: Audio scrambler function

	  Adds a function to scramble audio on a channel using
	  whole spectrum frequency inversion. This can be used
	  as a privacy enhancement with applications like
	  ChanSpy or other potentially sensitive audio.

	  ASTERISK-29542

	  Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e

2021-08-04 19:28 +0000 [cb1dfecc11]  Naveen Albert <asterisk@phreaknet.org>

	* app_originate: Add ability to set codecs

	  A list of codecs to use for dialplan-originated calls can
	  now be specified in Originate, similar to the ability
	  in call files and the manager action.

	  Additionally, we now default to just using the slin codec
	  for originated calls, rather than all the slin* codecs up
	  through slin192, which has been known to cause issues
	  and inconsistencies from AMI and call file behavior.

	  ASTERISK-29543

	  Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883

2021-08-16 11:11 +0000 [a8e8b3aaff]  Alexander Traud <pabstraud@compuserve.com>

	* BuildSystem: Remove two dead exceptions for compiler Clang.

	  Commit 305ce3d added -Wno-parentheses-equality to Makefile.rules,
	  turning the previous two warning suppressions from commit e9520db
	  redundant. Let us remove the latter.

	  Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac

2021-08-10 12:41 +0000 [121860e3f6]  Sean Bright <sean.bright@gmail.com>

	* mgcp: Remove dead debug code

	  ASTERISK-20339 #close

	  Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5

2021-08-11 06:15 +0000 [13fd0789a2]  Joshua C. Colp <jcolp@sangoma.com>

	* policy: Add deprecation and removal versions to modules.

	  app_meetme is deprecated in 19, to be removed in 21.
	  app_osplookup is deprecated in 19, to be removed in 21.
	  chan_alsa is deprecated in 19, to be removed in 21.
	  chan_mgcp is deprecated in 19, to be removed in 21.
	  chan_skinny is deprecated in 19, to be removed in 21.
	  res_pktccops is deprecated in 19, to be removed in 21.
	  cdr_mysql was deprecated in 1.8, to be removed in 19.
	  app_mysql was deprecated in 1.8, to be removed in 19.
	  app_ices was deprecated in 16, to be removed in 19.
	  app_macro was deprecated in 16, to be removed in 21.
	  app_fax was deprecated in 16, to be removed in 19.
	  app_url was deprecated in 16, to be removed in 19.
	  app_image was deprecated in 16, to be removed in 19.
	  app_nbscat was deprecated in 16, to be removed in 19.
	  app_dahdiras was deprecated in 16, to be removed in 19.
	  cdr_syslog was deprecated in 16, to be removed in 19.
	  chan_oss was deprecated in 16, to be removed in 19.
	  chan_phone was deprecated in 16, to be removed in 19.
	  chan_sip was deprecated in 17, to be removed in 21.
	  chan_nbs was deprecated in 16, to be removed in 19.
	  chan_misdn was deprecated in 16, to be removed in 19.
	  chan_vpb was deprecated in 16, to be removed in 19.
	  res_config_sqlite was deprecated in 16, to be removed in 19.
	  res_monitor was deprecated in 16, to be removed in 21.
	  conf2ael was deprecated in 16, to be removed in 19.
	  muted was deprecated in 16, to be removed in 19.

	  ASTERISK-29548
	  ASTERISK-29549
	  ASTERISK-29550
	  ASTERISK-29551
	  ASTERISK-29552
	  ASTERISK-29553
	  ASTERISK-29554
	  ASTERISK-29555
	  ASTERISK-29557
	  ASTERISK-29558
	  ASTERISK-29559
	  ASTERISK-29560
	  ASTERISK-29561
	  ASTERISK-29562
	  ASTERISK-29563
	  ASTERISK-29564
	  ASTERISK-29565
	  ASTERISK-29566
	  ASTERISK-29567
	  ASTERISK-29568
	  ASTERISK-29569
	  ASTERISK-29570
	  ASTERISK-29571
	  ASTERISK-29572
	  ASTERISK-29573
	  ASTERISK-29574

	  Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131

2021-08-12 11:00 +0000 [288d018fb7]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.6.0
2021-06-16 15:30 +0000 [118d848238]  Naveen Albert <asterisk@phreaknet.org>

	* func_frame_drop: New function

	  Adds function to selectively drop specified frames
	  in the TX or RX direction on a channel, including
	  control frames.

	  ASTERISK-29478

	  Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec

2021-08-02 12:33 +0000 [0b1a629ecd]  Alexander Traud <pabstraud@compuserve.com>

	* aelparse: Accept an included context with timings.

	  With Asterisk 1.6.0, in the main parser for the configuration file
	  extensions.conf, the separator was changed from vertical bar to comma.
	  However, the first separator was not changed in aelparse; it still had
	  to be a vertical bar, and no comma was allowed.

	  Additionally, this change allows the vertical bar for the first and
	  last parameter again, even in the main parser, because the vertical bar
	  was still accepted for the other parameters.

	  ASTERISK-29540

	  Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81

2021-08-03 11:30 +0000 [628830921e]  Kevin Harwell <kharwell@sangoma.com>

	* format_ogg_speex: Implement a "not supported" write handler

	  This format did not specify a "write" handler, so when attempting to write
	  to it (ast_writestream) a crash would occur.

	  This patch adds a default handler that simply issues a "not supported"
	  warning, thus no longer crashing.

	  ASTERISK-29539

	  Change-Id: I8f6ddc7cc3b15da30803be3b1cf68e2ba0fbce91

2021-08-05 14:28 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.6.0-rc1 Released.

2021-06-28 08:48 +0000 [adf707f2ae]  Naveen Albert <asterisk@phreaknet.org>

	* cdr_adaptive_odbc: Prevent filter warnings

	  Previously, if CDR filters were used so that
	  not all CDR records used all sections defined
	  in cdr_adaptive_odbc.conf, then warnings will
	  always be emitted (if each CDR record is unique
	  to a particular section, n-1 warnings to be
	  specific).

	  This turns the offending warning log into
	  a verbose message like the other one, since
	  this behavior is intentional and not
	  indicative of anything wrong.

	  ASTERISK-29494

	  Change-Id: Ifd314fa9298722bc99494d5ca2658a5caa94a5f8

2021-07-25 16:53 +0000 [940f6c4a03]  Naveen Albert <asterisk@phreaknet.org>

	* app_queue: Allow streaming multiple announcement files

	  Allows multiple files comprising an agent announcement
	  to be played by separating on the ampersand, similar
	  to the multi-file support in other Asterisk applications.

	  ASTERISK-29528

	  Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a

2021-04-13 02:36 +0000 [1e4ed61a2b]  Igor Goncharovsky <igorg@iqtek.ru>

	* res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern

	  PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request.
	  It may be used to get all X- headers in case the actual set and names of headers unknown.

	  ASTERISK-29389

	  Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b

2021-07-08 07:34 +0000 [71dd1d91ad]  Rijnhard Hessel <rijnhard@teleforge.co.za>

	* res_statsd: handle non-standard meter type safely

	  Meter types are not well supported,
	  lacking support in telegraf, datadog and the official statsd servers.
	  We deprecate meters and provide a compliant fallback for any existing usages.

	  A flag has been introduced to allow meters to fallback to counters.


	  ASTERISK-29513

	  Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7

2021-07-22 11:39 +0000 [feb1e06ac5]  under <pcapdump@gmail.com>

	* codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother

	  If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps.
	  This makes the audio stream not-playable at the receiver side.
	  Linphone isn't able to play such an audio - lots of disruptions are heard.
	  Also I had complains of bad audio from users which use other types of phones.

	  After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c).

	  Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c).

	  However, this flag is never set in Asterisk-12 and newer.
	  Previously it has been set (see Asterisk-11).

	  ASTERISK-29526 #close

	  Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d

2021-06-16 15:26 +0000 [016f6a0e14]  Naveen Albert <asterisk@phreaknet.org>

	* app_dtmfstore: New application to store digits

	  Adds application to asynchronously collect digits
	  dialed on a channel in the TX or RX direction
	  using a framehook and stores them in a specified
	  variable, up to a configurable number of digits.

	  ASTERISK-29477

	  Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f

2021-07-27 07:53 +0000 [9117f09d28]  Joshua C. Colp <jcolp@sangoma.com>

	* docs: Remove embedded macro in WaitForCond XML documentation.

	  Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c

2021-07-22 16:56 +0000 [993b3ba919]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.5.1
2021-06-14 13:28 +0000 [3025ef4f6e]  Kevin Harwell <kharwell@sangoma.com>

	* AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS

	  If an SSL socket parent/listener was destroyed during the handshake,
	  depending on timing, it was possible for the handling callback to
	  attempt access of it after the fact thus causing a crash.

	  ASTERISK-29415 #close

	  Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d

2021-05-10 17:59 +0000 [2a141a58b6]  Kevin Harwell <kharwell@sangoma.com>

	* AST-2021-008 - chan_iax2: remote crash on unsupported media format

	  If chan_iax2 received a packet with an unsupported media format, for
	  example vp9, then it would set the frame's format to NULL. This could
	  then result in a crash later when an attempt was made to access the
	  format.

	  This patch makes it so chan_iax2 now ignores/drops frames received
	  with unsupported media format types.

	  ASTERISK-29392 #close

	  Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1

2021-04-28 07:36 +0000 [523a795289]  Joshua C. Colp <jcolp@sangoma.com>

	* AST-2021-007 - res_pjsip_session: Don't offer if no channel exists.

	  If a re-INVITE is received after we have sent a BYE request then it
	  is possible for no channel to be present on the session. If this
	  occurs we allow PJSIP to produce the offer instead. Since the call
	  is being hung up if it produces an incorrect offer it doesn't
	  actually matter. This also ensures that code which produces SDP
	  does not need to handle if a channel is not present.

	  ASTERISK-29381

	  Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042

2021-06-29 11:07 +0000 [2c3defc6c6]  Andre Barbosa <andre.emanuel.barbosa@gmail.com>

	* res_stasis_playback: Check for chan hangup on play_on_channels

	  Verify `ast_check_hangup` before looping to the next sound file.
	  If the call is already hangup we just break the cycle.
	  It also ensures that the PlaybackFinished event is sent if the call was hangup.

	  This is also use-full when we are playing a big list of file for a channel that is hangup.
	  Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played.

	  With the patch we just break the playback cycle when the chan is hangup.

	  ASTERISK-29501 #close

	  Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8

2021-07-15 15:04 +0000 [30feaadabf]  Sean Bright <sean.bright@gmail.com>

	* res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup.

	  From RFC 8225 Section 5.2.1:

	      The "dest" claim is a JSON object with the claim name of "dest"
	      and MUST have at least one identity claim object.  The "dest"
	      claim value is an array containing one or more identity claim JSON
	      objects representing the destination identities of any type
	      (currently "tn" or "uri").  If the "dest" claim value array
	      contains both "tn" and "uri" claim names, the JSON object should
	      list the "tn" array first and the "uri" array second.  Within the
	      "tn" and "uri" arrays, the identity strings should be put in
	      lexicographical order, including the scheme-specific portion of
	      the URI characters.

	  Additionally, make it clear that there was a failure to sign the JWT
	  payload and not necessarily a memory allocation failure.

	  Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9

2021-06-30 17:15 +0000 [4bd975f415]  Sebastien Duthil <sduthil@wazo.community>

	* stun: Emit warning message when STUN request times out

	  Without this message, it is not obvious that the reason is STUN timeout.

	  ASTERISK-29507 #close

	  Change-Id: I26e4853c23a1aed324552e1b9683ea3c05cb1f74

2021-07-02 10:15 +0000 [76c09b1cfd]  Sean Bright <sean.bright@gmail.com>

	* res_http_media_cache.c: Parse media URLs to find extensions.

	  Use the URI parsing functions to parse playback URLs in order to find
	  their file extensions.

	  For backwards compatibility, we first look at the full URL, then at
	  any Content-Type header, and finally at just the path portion of the
	  URL.

	  ASTERISK-27871 #close

	  Change-Id: I16d0682f6d794be96539261b3e48f237909139cb

2021-07-13 10:31 +0000 [fcebc4d24a]  Sean Bright <sean.bright@gmail.com>

	* main/cdr.c: Correct Party A selection.

	  This appears to just have been a copy/paste error from 6258bbe7. Fix
	  suggested by Ross Beer in ASTERISK~29166.

	  Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37

2021-05-26 12:09 +0000 [a41d192e99]  Naveen Albert <asterisk@phreaknet.org>

	* app_reload: New Reload application

	  Adds an application to reload modules
	  from within the dialplan.

	  ASTERISK-29454

	  Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774

2021-07-08 09:32 +0000 [b9bb96ffed]  Igor Goncharovsky <igorg@iqtek.ru>

	* res_ari: Fix audiosocket segfault

	  Add check that data parameter specified when audiosocket used for externalMedia.

	  ASTERISK-29514 #close

	  Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617

2021-06-30 08:07 +0000 [146b59df3f]  Sean Bright <sean.bright@gmail.com>

	* res_pjsip_config_wizard.c: Add port matching support.

	  In f8b0c2c9 we added support for port numbers in 'match' statements
	  but neglected to include that support in the PJSIP config wizard.

	  The removed code would have also prevented IPv6 addresses from being
	  successfully used in the config wizard as well.

	  ASTERISK-29503 #close

	  Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d

2021-05-22 09:31 +0000 [1b21b1abf7]  Naveen Albert <mail@interlinked.x10host.com>

	* app_waitforcond: New application

	  While several applications exist to wait for
	  a certain event to occur, none allow waiting
	  for any generic expression to become true.
	  This application allows for waiting for a condition
	  to become true, with configurable timeout and
	  checking interval.

	  ASTERISK-29444

	  Change-Id: I08adf2824b8bc63405778cf355963b5005612f41

2021-06-04 06:11 +0000 [283812e492]  Andre Barbosa <andre.emanuel.barbosa@gmail.com>

	* res_stasis_playback: Send PlaybackFinish event only once for errors

	  When we try to play a list of sound files in the same Play command,
	  we get only one PlaybackFinish event, after all sounds are played.

	  But in the case where the Play fails (because channel is destroyed
	  for example), Asterisk will send one PlaybackFinish event for each
	  sound file still to be played. If the list is big, Asterisk is
	  sending many events.

	  This patch adds a failed state so we can understand that the play
	  failed. On that case we don't send the event, if we still have a
	  list of sounds to be played.

	  When we reach the last sound, we send the PlaybackFinish with
	  the failed state.

	  ASTERISK-29464 #close

	  Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322

2021-06-17 07:57 +0000 [88da59efe7]  George Joseph <gjoseph@digium.com>

	* jitterbuffer:  Correct signed/unsigned mismatch causing assert

	  If the system time has stepped backwards because of a time
	  adjustment between the time a frame is timestamped and the
	  time we check the timestamps in abstract_jb:hook_event_cb(),
	  we get a negative interval, but we don't check for that there.
	  abstract_jb:hook_event_cb() then calls
	  fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed)
	  and the first thing that does is assert(interval >= 0).

	  There are several issues with this...

	   * abstract_jb:hook_event_cb() saves the interval in a variable
	     named "now" which is confusing in itself.

	   * "now" is defined as an unsigned int which converts the negative
	     value returned from ast_tvdiff_ms() to a large positive value.

	   * fixed_jb_get()'s parameter is defined as a signed int so the
	     interval gets converted back to a negative value.

	   * fixed_jb_get()'s assert is NOT an ast_assert but a direct define
	     that points to the system assert() so it triggers even in
	     production mode.

	  So...

	   * hook_event_cb()'s "now" was renamed to "relative_frame_start" and
	     changed to an int64_t.
	   * hook_event_cb() now checks for a negative value right after
	     retrieving both the current and framedata timestamps and just
	     returns the frame if the difference is negative.
	   * fixed_jb_get()'s local define of ASSERT() was changed to call
	     ast_assert() instead of the system assert().

	  ASTERISK-29480
	  Reported by: Dan Cropp

	  Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9

2021-05-21 19:08 +0000 [c4236dcff2]  Naveen Albert <mail@interlinked.x10host.com>

	* app_dial: Expanded A option to add caller announcement

	  Hitherto, the A option has made it possible to play
	  audio upon answer to the called party only. This option
	  is expanded to allow for playback of an audio file to
	  the caller instead of or in addition to the audio
	  played to the answerer.

	  ASTERISK-29442

	  Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e

2021-06-21 06:31 +0000 [5e1cb3253c]  Joshua C. Colp <jcolp@sangoma.com>

	* core: Don't play silence for Busy() and Congestion() applications.

	  When using the Busy() and Congestion() applications the
	  function ast_safe_sleep is used by wait_for_hangup to safely
	  wait on the channel. This function may send silence if Asterisk
	  is configured to do so using the transmit_silence option.

	  In a scenario where an answered channel dials a Local channel
	  either directly or through call forwarding and the Busy()
	  or Congestion() dialplan applications were executed with the
	  transmit_silence option enabled the busy or congestion
	  tone would not be heard.

	  This is because inband generation of tones (such as busy
	  and congestion) is stopped when other audio is sent to
	  the channel they are being played to. In the given
	  scenario the transmit_silence option would result in
	  silence being sent to the channel, thus stopping the
	  inband generation.

	  This change adds a variant of ast_safe_sleep which can be
	  used when silence should not be played to the channel. The
	  wait_for_hangup function has been updated to use this
	  resulting in the tones being generated as expected.

	  ASTERISK-29485

	  Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133

2021-05-07 01:18 +0000 [6b041d1092]  Bernd Zobl <b.zobl@commend.com>

	* res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress

	  With the fix for ASTERISK_28754 channels are no longer put on hold if an
	  outbound INVITE is answered with a "Session Progress" containing
	  "inactive" audio.

	  The previous change moved the evaluation of the media attributes to
	  `negotiate_incoming_sdp_stream()` to have the `remotely_held` status
	  available when building the SDP in `create_outgoing_sdp_stream()`.
	  This however means that an answer to an outbound INVITE, which does not
	  traverse `negotiate_incoming_sdp_stream()`, cannot set the
	  `remotely_held` status anymore.

	  This change moves the check so that both, `negotiate_incoming_sdp_stream()` and
	  `apply_negotiated_sdp_stream()` can do the checks.

	  ASTERISK-29479

	  Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75

2021-06-17 14:44 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.5.0-rc1 Released.

2021-06-17 09:39 +0000 [0747162d4f]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.5.0
2021-06-16 08:50 +0000 [702e1d33b5]  George Joseph <gjoseph@digium.com>

	* res_pjsip_messaging: Overwrite user in existing contact URI

	  When the MessageSend destination is in the form
	  PJSIP/<number>@<endpoint> and the endpoint's contact
	  URI already has a user component, that user component
	  will now be replaced with <number> when creating the
	  request URI.

	  ASTERISK_29404

	  Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5

2021-03-16 11:45 +0000 [804788037e]  Bernd Zobl <b.zobl@commend.com>

	* res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter

	  Set preferred transport when querying the local address to use in
	  filter_on_tx_messages(). This prevents the module to erroneously select
	  the wrong transport if more than one transports of the same type (TCP or
	  TLS) are configured.

	  ASTERISK-29241

	  Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6

2021-06-10 09:34 +0000 [2b174a38fe]  Naveen Albert <asterisk@phreaknet.org>

	* pbx_builtins: Corrects SayNumber warning

	  Previously, SayNumber always emitted a warning if the caller hung up
	  during execution. Usually this isn't correct, so check if the channel
	  hung up and, if so, don't emit a warning.

	  ASTERISK-29475

	  Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594

2021-05-22 07:53 +0000 [6b67821098]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Prevent module unloading in-use module.

	  The scenario where a channel still has an associated datastore we
	  cannot unload since there is a function pointer to the destroy and fixup
	  functions in play.  Thus increase the module ref count whenever we
	  allocate a datastore, and decrease it during destroy.

	  In order to tighten the race that still exists in spite of this (below)
	  add some extra failure cases to prevent allocations in these cases.

	  Race:

	  If module ref is zero, an LOCK or TRYLOCK is invoked (near)
	  simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if
	  in such a case the datastore is created *prior* to unloading being set
	  to true (first step in module unload) then it's possible that the module
	  will unload with the destructor being called (and segfault) post the
	  module being unloaded.  The module will however wait for such locks to
	  release prior to unloading.

	  If post that we can recheck the module ref before returning the we can
	  (in theory, I think) eliminate the last of the race.  This race is
	  mostly theoretical in nature.

	  Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-22 07:29 +0000 [6f303335d3]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Add "dialplan locks show" cli command.

	  For example:

	  arthur*CLI> dialplan locks show
	  func_lock locks:
	  Name                                     Requesters Owner
	  uls-autoref                              0          (unlocked)
	  1 total locks listed.

	  Obviously other potentially useful stats could be added (eg, how many
	  times there was contention, how many times it failed etc ... but that
	  would require keeping the stats and I'm not convinced that's worth the
	  effort.  This was useful to troubleshoot some other issues so submitting
	  it.

	  Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-22 07:42 +0000 [a3df5d7de8]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Fix memory corruption during unload.

	  AST_TRAVERSE accessess current as current = current->(field).next ...
	  and since we free current (and ast_free poisons the memory) we either
	  end up on a ast_mutex_lock to a non-existing lock that can never be
	  obtained, or a segfault.

	  Incidentally add logging in the "we have to wait for a lock to release"
	  case, and remove an ineffective statement that sets memory that was just
	  cleared by ast_calloc to zero.

	  Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-22 07:48 +0000 [6bd741b77d]  Jaco Kroon <jaco@uls.co.za>

	* func_lock: Fix requesters counter in error paths.

	  In two places we bail out with failure after we've already incremented
	  the requesters counter, if this occured then it would effectively result
	  in unload to wait indefinitely, thus preventing clean shutdown.

	  Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-05-25 10:36 +0000 [a611a0cd42]  Naveen Albert <asterisk@phreaknet.org>

	* app_originate: Allow setting Caller ID and variables

	  Caller ID can now be set on the called channel and
	  Variables can now be set on the destination
	  using the Originate application, just as
	  they can be currently using call files
	  or the Manager Action.

	  ASTERISK-29450

	  Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66

2021-06-10 16:24 +0000 [26059f8616]  Sean Bright <sean.bright@gmail.com>

	* menuselect: Fix description of several modules.

	  The text description needs to be the last thing on the AST_MODULE_INFO
	  line to be pulled in properly by menuselect.

	  Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832

2021-05-23 19:20 +0000 [a40e58a4da]  Naveen Albert <asterisk@phreaknet.org>

	* app_confbridge: New ConfKick() application

	  Adds a new ConfKick() application, which may
	  be used to kick a specific channel, all channels,
	  or all non-admin channels from a specified
	  conference bridge, similar to existing CLI and
	  AMI commands.

	  ASTERISK-29446

	  Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b

2021-06-02 08:11 +0000 [6873c5f3e4]  Naveen Albert <asterisk@phreaknet.org>

	* sip_to_pjsip: Fix missing cases

	  Adds the "auto" case which is valid with
	  both chan_sip dtmfmode and chan_pjsip's
	  dtmf_mode, adds subscribecontext to
	  subscribe_context conversion, and accounts
	  for cipher = ALL being invalid.

	  ASTERISK-29459

	  Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2

2021-06-02 08:25 +0000 [99573f9540]  Naveen Albert <asterisk@phreaknet.org>

	* res_pjsip_dtmf_info: Hook flash

	  Adds hook flash recognition support
	  for application/hook-flash.

	  ASTERISK-29460

	  Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea

2021-05-20 09:51 +0000 [a861522467]  Naveen Albert <mail@interlinked.x10host.com>

	* app_confbridge: New option to prevent answer supervision

	  A new user option, answer_channel, adds the capability to
	  prevent answering the channel if it hasn't already been
	  answered yet.

	  ASTERISK-29440

	  Change-Id: I26642729d0345f178c7b8045506605c8402de54b

2021-04-22 13:07 +0000 [8e2672d2a4]  George Joseph <gjoseph@digium.com>

	* res_pjsip_messaging: Refactor outgoing URI processing

	   * Implemented the new "to" parameter of the MessageSend()
	     dialplan application.  This allows a user to specify
	     a complete SIP "To" header separate from the Request URI.

	   * Completely refactored the get_outbound_endpoint() function
	     to actually handle all the destination combinations that
	     we advertized as supporting.

	   * We now also accept a destination in the same format
	     as Dial()...  PJSIP/number@endpoint

	   * Added lots of debugging.

	  ASTERISK-29404
	  Reported by Brian J. Murrell

	  Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce

2021-05-16 10:21 +0000 [9106c9d1f1]  Naveen Albert <mail@interlinked.x10host.com>

	* func_math: Three new dialplan functions

	  Introduces three new dialplan functions, MIN and MAX,
	  which can be used to calculate the minimum or
	  maximum of up to two numbers, and ABS, an absolute
	  value function.

	  ASTERISK-29431

	  Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d

2021-05-19 13:45 +0000 [26a38c4084]  Ben Ford <bford@digium.com>

	* STIR/SHAKEN: Add Date header, dest->tn, and URL checking.

	  STIR/SHAKEN requires a Date header alongside the Identity header, so
	  that has been added. Still on the outgoing side, we were missing the
	  dest->tn section of the JSON payload, so that has been added as well.
	  Moving to the incoming side, URL checking has been added to the public
	  cert URL to ensure that it starts with http.

	  https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

	  Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c

2021-05-24 13:38 +0000 [16e4a9d8cf]  Joshua C. Colp <jcolp@sangoma.com>

	* res_pjsip: On partial transport reload also move factories.

	  For connection oriented transports PJSIP uses factories to
	  produce transports. When doing a partial transport reload
	  we need to also move the factory of the transport over so
	  that anything referencing the transport (such as an endpoint)
	  has the factory available.

	  ASTERISK-29441

	  Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161

2021-05-20 08:18 +0000 [033c2a2283]  Naveen Albert <mail@interlinked.x10host.com>

	* func_volume: Add read capability to function.

	  Up until now, the VOLUME function has been write
	  only, so that TX/RX values can be set but not
	  read afterwards. Now, previously set TX/RX values
	  can be read later.

	  ASTERISK-29439

	  Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f

2021-04-13 02:57 +0000 [59d15c4c2a]  Evgenios_Greek <jone1984@hotmail.com>

	* stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing

	  When unsubscribing from an endpoint technology a FRACK
	  would occur due to incorrect reference counting. This fixes
	  that issue, along with some other issues.

	  Fixed a typo in get_subscription when calling ao2_find as it
	  needed to pass the endpoint ID and not the entire object.

	  Fixed scenario where a subscription would get returned when
	  it shouldn't have been when searching based on endpoint
	  technology.

	  A doulbe unreference has also been resolved by only explicitly
	  releasing the reference held by tech_subscriptions.

	  ASTERISK-28237 #close
	  Reported by: Lucas Tardioli Silveira

	  Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729

2021-05-20 02:15 +0000 [b21d4d1b87]  Joseph Nadiv <ynadiv@corpit.xyz>

	* res_pjsip.c: Support endpoints with domain info in username

	  In multidomain environments, it is desirable to create
	  PJSIP endpoints with the domain info in the endpoint name
	  in pjsip_endpoint.conf.  This resulted in an error with
	  registrations, NOTIFY, and OPTIONS packet generation.

	  This commit will detect if there is an @ in the endpoint
	  identifier and generate the URI accordingly so NOTIFY and
	  OPTIONS From headers will generate correctly.

	  ASTERISK-28393

	  Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619

2021-05-20 07:51 +0000 [3aed363716]  Joshua C. Colp <jcolp@sangoma.com>

	* res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates.

	  RTCP ICE candidates use a base address derived from the RTP
	  candidate. The port on the base address was not being updated to
	  the RTCP port.

	  This change sets the base port to the RTCP port and all is well.

	  ASTERISK-29433

	  Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040

2021-05-25 05:38 +0000 [60ed1847b8]  Joshua C. Colp <jcolp@sangoma.com>

	* asterisk: We've moved to Libera Chat!

	  Change-Id: I48c1933dd79b50ddc0a6793acec4754b4e95c575

2021-05-19 13:13 +0000 [0f8e2174a7]  Jeremy Lainé <jeremy.laine@m4x.org>

	* res_rtp_asterisk: make it possible to remove SOFTWARE attribute

	  By default Asterisk reports the PJSIP version in a SOFTWARE attribute
	  of every STUN packet it sends. This may not be desired in a production
	  environment, and RFC5389 recommends making the use of the SOFTWARE
	  attribute a configurable option:

	  https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2

	  This patch adds a `stun_software_attribute` yes/no option to make it
	  possible to omit the SOFTWARE attribute from STUN packets.

	  ASTERISK-29434

	  Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b

2021-04-15 10:43 +0000 [655ee680cd]  George Joseph <gjoseph@digium.com>

	* res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs

	  RFC7616 and RFC8760 allow more than one WWW-Authenticate or
	  Proxy-Authenticate header per realm, each with different digest
	  algorithms (including new ones like SHA-256 and SHA-512-256).
	  Thankfully however a UAS can NOT send back multiple Authenticate
	  headers for the same realm with the same digest algorithm.  The
	  UAS is also supposed to send the headers in order of preference
	  with the first one being the most preferred.  We're supposed to
	  send an Authorization header for the first one we encounter for a
	  realm that we can support.

	  The UAS can also send multiple realms, especially when it's a
	  proxy that has forked the request in which case the proxy will
	  aggregate all of the Authenticate headers and then send them all
	  back to the UAC.

	  It doesn't stop there though... Each realm can require a
	  different username from the others.  There's also nothing
	  preventing each digest algorithm from having a unique password
	  although I'm not sure if that adds any benefit.

	  So now... For each Authenticate header we encounter, we have to
	  determine if we support the digest algorithm and, if not, just
	  skip the header.  We then have to find an auth object that
	  matches the realm AND the digest algorithm or find a wildcard
	  object that matches the digest algorithm. If we find one, we add
	  it to the results vector and read the next Authenticate header.
	  If the next header is for the same realm AND we already added an
	  auth object for that realm, we skip the header. Otherwise we
	  repeat the process for the next header.

	  In the end, we'll have accumulated a list of credentials we can
	  pass to pjproject that it can use to add Authentication headers
	  to a request.

	  NOTE: Neither we nor pjproject can currently handle digest
	  algorithms other than MD5.  We don't even have a place for it in
	  the ast_sip_auth object. For this reason, we just skip processing
	  any Authenticate header that's not MD5.  When we support the
	  others, we'll move the check into the loop that searches the
	  objects.

	  Changes:

	   * Added a new API ast_sip_retrieve_auths_vector() that takes in
	     a vector of auth ids (usually supplied on a call to
	     ast_sip_create_request_with_auth()) and populates another
	     vector with the actual objects.

	   * Refactored res_pjsip_outbound_authenticator_digest to handle
	     multiple Authenticate headers and set the stage for handling
	     additional digest algorithms.

	   * Added a pjproject patch that allows them to ignore digest
	     algorithms they don't support.  This patch has already been
	     merged upstream.

	   * Updated documentation for auth objects in the XML and
	     in pjsip.conf.sample.

	   * Although res_pjsip_authenticator_digest isn't affected
	     by this change, some debugging and a testsuite AMI event
	     was added to facilitate testing.

	  Discovered during OpenSIPit 2021.