ChangeLog

	* main/utils: Implement ast_get_tid() for NetBSD

	  Implement the ast_get_tid() function for NetBSD system.  NetBSD supports
	  getting the TID via _lwp_self().

	  ASTERISK-29850

	  Change-Id: If57fd3f9ea15ef5d010bfbdcbbbae9b379f72f8c

2021-11-10 20:05 +0000 [7b1e5fa34a]  Michał Górny <mgorny@NetBSD.org>

	* BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD

	  Fix the configure script not to detect the presence of gethostbyname_r()
	  on NetBSD incorrectly.  NetBSD includes it as an internal libc symbol
	  that is not exposed in system headers and that is incompatible with
	  other implementations.  In order to avoid misdetecting it, perform
	  the symbol check only if the declaration is found in the public header
	  first.

	  ASTERISK-29817

	  Change-Id: Iafa359b09908251bcd299ff54be003ea129b9eda

2021-11-10 22:06 +0000 [eef29d24e1]  Michał Górny <mgorny@NetBSD.org>

	* include: Remove unimplemented HMAC declarations

	  Remove the HMAC declarations from the includes.  They are
	  not implemented nor used anywhere, and their presence breaks the build
	  on NetBSD that delivers an incompatible hmac() function in <stdlib.h>.

	  ASTERISK-29818

	  Change-Id: I0c4b88645e30174b1b63846a6b328625b69c2ea7

2022-01-11 12:41 +0000 [18c257b44d]  Naveen Albert <asterisk@phreaknet.org>

	* frame.h: Fix spelling typo

	  Fixes CNG description from "noice" to "noise".

	  ASTERISK-29855 #close

	  Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d

2022-01-11 12:46 +0000 [a9e9e15c3a]  Naveen Albert <asterisk@phreaknet.org>

	* res_rtp_asterisk: Fix typo in flag test/set

	  The code currently checks to see if an RFC3389
	  warning flag is set, except if it is, it merely
	  sets the flag again, the logic of which doesn't
	  make any sense.

	  This adjusts the if comparison to check if the
	  flag has NOT been set, and if so, emit a notice
	  log event and set the flag so that future frames
	  do not cause an event to be logged.

	  ASTERISK-29856 #close

	  Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1

2022-01-18 08:04 +0000 [cc38ed9c21]  George Joseph <gjoseph@digium.com>

	* bundled_pjproject: Fix srtp detection

	  Reverted recent change that set '--with-external-srtp' instead
	  of '--without-external-srtp'.  Since Asterisk handles all SRTP,
	  we don't need it enabled in pjproject at all.

	  ASTERISK-29867

	  Change-Id: I2ce1bdd30abd21c062eac8f8fefe9b898787b801

2022-01-10 07:44 +0000 [f55886a72c]  George Joseph <gjoseph@digium.com>

	* res_pjsip: Make message_filter and session multipart aware

	  Neither pjsip_message_filter's filter_on_tx_message() nor
	  res_pjsip_session's session_outgoing_nat_hook() were multipart
	  aware and just assumed that an SDP would be the only thing in
	  a message body.  Both were changed to use the new
	  pjsip_get_sdp_info() function which searches for an sdp in
	  both single- and multi- part message bodies.

	  ASTERISK-29813

	  Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56

2022-01-06 13:05 +0000 [59cf9f0047]  George Joseph <gjoseph@digium.com>

	* res_pjsip: Add utils for checking media types

	  Added two new functions to assist checking media types...

	  * ast_sip_are_media_types_equal compares two pjsip_media_types.
	  * ast_sip_is_media_type_in tests if one media type is in a list
	    of others.

	  Added static definitions for commonly used media types to
	  res_pjsip.h.

	  Changed several modules to use the new functions and static
	  definitions.

	  ASTERISK_29813
	  (not ready to close)

	  Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9

2022-01-12 11:12 +0000 [b59bd3d3e4]  George Joseph <gjoseph@digium.com>

	* build: Fix issues building pjproject

	  The change to allow easier hacking on bundled pjproject created
	  a few issues:

	  * The new Makefile was trying to run the bundled make even if
	    PJPROJECT_BUNDLED=no.  third-party/Makefile now checks for
	    PJPROJECT_BUNDLED and JANSSON_BUNDLED and skips them if they
	    are "no".

	  * When building with bundled, config_site.h was being copied
	    only if a full make or a "make main" was done.  A "make res"
	    would fail all the pjsip modules because they couldn't find
	    config_site.h.  The Makefile now copies config_site.h and
	    asterisk_malloc_debug.h into the pjproject source tree
	    when it's "configure" is performed.  This is how it used
	    to be before the big change.

	  ASTERISK-29858

	  Change-Id: I9427264fa3cb8b3f59a95e5f9693eac236a6f76d

2022-01-12 07:16 +0000 [f10947ecc2]  George Joseph <gjoseph@digium.com>

	* bundled_pjproject: Create generic pjsip_hdr_find functions

	  pjsip_msg_find_hdr(), pjsip_msg_find_hdr_by_name(), and
	  pjsip_msg_find_hdr_by_names() require a pjsip_msg to be passed in
	  so if you need to search a header list that's not in a pjsip_msg,
	  you have to do it yourself.  This commit adds generic versions of
	  those 3 functions that take in the actual header list head instead
	  of a pjsip_msg so if you need to search a list of headers in
	  something like a pjsip_multipart_part, you can do so easily.

	  Change-Id: I6f2c127170eafda48e5e0d5d4d187bcd52b4df07

2022-01-12 13:20 +0000 [3fd47840c9]  Sean Bright <sean.bright@gmail.com>

	* say.c: Prevent erroneous failures with 'say' family of functions.

	  A regression was introduced in ASTERISK~29531 that caused 'say'
	  functions to fail with file lists that would previously have
	  succeeded. This caused affected channels to hang up where previously
	  they would have continued.

	  We now explicitly check for the empty string to restore the previous
	  behavior.

	  ASTERISK-29859 #close

	  Change-Id: Ia2e5769868e2792313c2d7c07996efe009c6f8d5

2022-01-08 09:09 +0000 [e006d2d2a6]  Naveen Albert <asterisk@phreaknet.org>

	* pbx_variables: add missing ASTSBINDIR variable

	  Every config variable in the directories
	  section of asterisk.conf currently has a
	  counterpart built-in variable containing
	  the value of the config option, except
	  for the last one, astsbindir, which should
	  have an ASTSBINDIR variable.

	  However, the actual corresponding ASTSBINDIR
	  variable is missing in pbx_variables.c.

	  This adds the missing variable so that all
	  the config options have their corresponding
	  variable.

	  ASTERISK-29847 #close

	  Change-Id: I36006faf471825b36ebc8aa5e87a3bcb38d446fc

2022-01-08 14:35 +0000 [707f32170c]  Naveen Albert <asterisk@phreaknet.org>

	* documentation: Document built-in system and channel vars

	  Documentation for built-in special system and channel
	  vars is currently outdated, and updating is a manual
	  process since there is no XML documentation for these
	  anywhere.

	  This adds documentation for system vars to func_env
	  and for channel vars to func_channel so that they
	  appear along with the corresponding fields that would
	  be accessed using a function.

	  ASTERISK-29848 #close

	  Change-Id: I6997f925c4a45fffe71321861f5898a8b7182fa9

2021-11-30 16:35 +0000 [3f093b8dda]  George Joseph <gjoseph@digium.com>

	* bundled_pjproject:  Make it easier to hack

	  There are times when you need to troubleshoot issues with bundled
	  pjproject or add new features that need to be pushed upstream
	  but...

	  * The source directory created by extracting the pjproject tarball
	    is not scanned for code changes so you have to keep forcing
	    rebuilds.
	  * The source directory isn't a git repo so you can't easily create
	    patches, do git bisects, etc.
	  * Accidentally doing a make distclean will ruin your day by wiping
	    out the source directory, and your changes.
	  * etc.

	  This commit makes that easier.
	  See third-party/pjproject/README-hacking.md for the details.

	  ASTERISK-29824

	  Change-Id: Idb1251040affdab31d27cd272dda68676da9b268

2021-12-24 10:26 +0000 [0bbef4d8c5]  Sean Bright <sean.bright@gmail.com>

	* utils.c: Remove all usages of ast_gethostbyname()

	  gethostbyname() and gethostbyname_r() are deprecated in favor of
	  getaddrinfo() which we use in the ast_sockaddr family of functions.

	  ASTERISK-29819 #close

	  Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7

2021-12-13 09:53 +0000 [54f2f1e027]  Naveen Albert <asterisk@phreaknet.org>

	* say.conf: fix 12pm noon logic

	  Fixes 12pm noon incorrectly returning 0/a.m.
	  Also fixes a misspelling typo in the config.

	  ASTERISK-29695 #close

	  Change-Id: Ie40f9618636eb4c483b449bd707a5dcffca5c406

2022-01-04 08:08 +0000 [ee69441fbd]  Sean Bright <sean.bright@gmail.com>

	* pjproject: Fix incorrect unescaping of tokens during parsing

	  ASTERISK-29664 #close

	  Change-Id: I29dcde52e9faeaf2609c604eada61c6a9e49d8f5

2021-12-30 07:02 +0000 [dea71ddbbf]  Mark Petersen <bugs.digium.com@zombie.dk>

	* app_queue.c: Support for Nordic syntax in announcements

	  adding support for playing the correct en/et for nordic languages
	  by adding 'n' for neuter gender in the relevant ast_say_number

	  ASTERISK-29827

	  Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1

2021-12-23 08:50 +0000 [6d7161820e]  Naveen Albert <asterisk@phreaknet.org>

	* dsp: Add define macro for DTMF_MATRIX_SIZE

	  Adds the macro DTMF_MATRIX_SIZE to replace
	  the magic number 4 sprinkled throughout
	  dsp.c.

	  ASTERISK-29815 #close

	  Change-Id: Ie3bddb92c6b16204ece0f758009e9490eb33b9ba

2022-01-03 11:10 +0000 [f133ae6ca2]  Naveen Albert <asterisk@phreaknet.org>

	* ami: Add AMI event for Wink

	  Adds an AMI event for a wink frame.

	  ASTERISK-29830 #close

	  Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56

2021-12-15 08:23 +0000 [1c2f311ba3]  Naveen Albert <asterisk@phreaknet.org>

	* cli: Add module refresh command

	  Adds a command to the CLI to unload and then
	  load a module. This makes it easier to perform
	  these operations which are often done
	  subsequently to load a new version of a module.

	  "module reload" already refers to reloading of
	  configuration, so the name "refresh" is chosen
	  instead.

	  ASTERISK-29807 #close

	  Change-Id: I595f6f11774a0de2565a1fba38da22309ce93a2c

2022-01-02 19:13 +0000 [775c371d09]  Naveen Albert <asterisk@phreaknet.org>

	* app_mp3: Throw warning on nonexistent stream

	  Currently, the MP3Player application doesn't
	  emit a warning if attempting to play a stream
	  which no longer exists. This can be a common
	  scenario as many mp3 streams are valid at some
	  point but can disappear at any time.

	  Now a warning is thrown if attempting to play
	  a nonexistent MP3 stream, instead of silently
	  exiting.

	  ASTERISK-29829 #close

	  Change-Id: I53a0bf1ed1740166655eb66fe7675f6f808bf535

2021-12-13 08:29 +0000 [b37feb42ae]  Naveen Albert <asterisk@phreaknet.org>

	* documentation: Add missing AMI documentation

	  Adds missing documentation for some channel,
	  bridge, and queue events.

	  ASTERISK-24427
	  ASTERISK-29515

	  Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64

2021-11-15 16:13 +0000 [06f9227ac5]  Kevin Harwell <kharwell@sangoma.com>

	* tcptls.c: refactor client connection to be more robust

	  The current TCP client connect code, blocks and does not handle EINTR
	  error case.

	  This patch makes the client socket non-blocking while connecting,
	  ensures a connect does not immediately fail due to EINTR "errors",
	  and adds a connect timeout option.

	  The original client start call sets the new timeout option to
	  "infinite", thus making sure old, orginal behavior is retained.

	  ASTERISK-29746 #close

	  Change-Id: I907571843a83e43c0742b95a64785f4411f02671

2021-12-13 10:59 +0000 [dd6df42534]  Naveen Albert <asterisk@phreaknet.org>

	* app_sf: Add full tech-agnostic SF support

	  Adds tech-agnostic support for SF signaling
	  by adding SF sender and receiver applications
	  as well as Dial integration.

	  ASTERISK-29802 #close

	  Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad

2021-12-15 06:23 +0000 [16a63027c0]  Steve Davies <steve@one47.co.uk>

	* app_queue: Fix hint updates, allow dup. hints

	  A previous patch for ASTERISK_29578 caused a 'leak' of
	  extension state information across queues, causing the
	  state of the first member of unrelated queues to be
	  updated in addition to the correct member. Which queues
	  and members depended on the order of queues in the
	  iterator.

	  Additionally, it is possible to use the same 'hint:' on
	  multiple queue members, so the update cannot break out
	  of the update loop early when a match is found.

	  ASTERISK-29806 #close

	  Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad

2021-12-23 15:57 +0000 [4fe94bab09]  Sean Bright <sean.bright@gmail.com>

	* say.c: Honor requests for DTMF interruption.

	  SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal,
	  and SayPhonetic all claim to allow DTMF interruption if the
	  SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we
	  are failing to break out of a given 'say' application if DTMF actually
	  occurs.

	  ASTERISK-29816 #close

	  Change-Id: I6a96e0130560831d2cb45164919862b9bcb6287e

2021-11-16 06:32 +0000 [4e204db2bf]  Florentin Mayer <f.mayer@commend.com>

	* res_pjsip_sdp_rtp: Preserve order of RTP codecs

	  The ast_rtp_codecs_payloads functions do not preserve the order in which
	  the payloads were specified on an incoming SDP media line. This leads to
	  a problem with the codec negotiation functionality, as the format
	  capabilities of the stream are extracted from the ast_rtp_codecs. This
	  commit moves the ast_rtp_codec to ast_format conversion to the place
	  where the order is still known.

	  ASTERISK-28863
	  ASTERISK-29320

	  Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25

2021-12-27 07:28 +0000 [d83a46869e]  Joshua C. Colp <jcolp@sangoma.com>

	* bridge: Unlock channel during Local peer check.

	  It's not safe to keep the channel locked while locking
	  the peer Local channel, as it can result in a deadlock.

	  This change unlocks it during this time but keeps the
	  bridge locked to ensure nothing changes about the bridge.

	  ASTERISK-29821

	  Change-Id: Ib68eb7037e5a479bcc2aceee77337cdde1fbdde6

2021-11-07 09:32 +0000 [a5cdee36a7]  Josh Soref <jsoref@gmail.com>

	* test_time.c: Tolerate DST transitions

	  When test_timezone_watch runs very near a DST transition,
	  two time zones that would otherwise be expected to report the same
	  time can differ because of the DST transition.

	  Instead of having the test fail when this happens, report the
	  times, time zones, and dst flags.

	  ASTERISK-29722

	  Change-Id: Id59bdac8b277e14343ccdf0c99b89e92f79f316a

2021-12-14 11:39 +0000 [0cf4e325aa]  George Joseph <gjoseph@digium.com>

	* bundled_pjproject:  Add more support for multipart bodies

	  Adding upstream patch for pull request...
	  https://github.com/pjsip/pjproject/pull/2920
	  ---------------------------------------------------------------

	  sip_inv:  Additional multipart support (#2919)

	  sip_inv.c:inv_check_sdp_in_incoming_msg() deals with multipart
	  message bodies in rdata correctly. In the case where early media is
	  involved though, the existing sdp has to be retrieved from the last
	  tdata sent in this transaction. This, however, always assumes that
	  the sdp sent is in a non-multipart body. While there's a function
	  to retrieve the sdp from multipart and non-multpart rdata bodies,
	  no similar function for tdata exists.  So...

	  * The existing pjsip_rdata_get_sdp_info2 was refactored to
	    find the sdp in any body, multipart or non-multipart, and
	    from either an rdata or tdata.  The new function is
	    pjsip_get_sdp_info.  This new function detects whether the
	    pjsip_msg->body->data is the text representation of the sdp
	    from an rdata or an existing pjmedia_sdp_session object
	    from a tdata, or whether pjsip_msg->body is a multipart
	    body containing either of the two sdp formats.

	  * The exsting pjsip_rdata_get_sdp_info and pjsip_rdata_get_sdp_info2
	    functions are now wrappers that get the body and Content-Type
	    header from the rdata and call pjsip_get_sdp_info.

	  * Two new wrappers named pjsip_tdata_get_sdp_info and
	    pjsip_tdata_get_sdp_info2 have been created that get the body
	    from the tdata and call pjsip_get_sdp_info.

	  * inv_offer_answer_test.c was updated to test multipart scenarios.

	  ASTERISK-29804

	  Change-Id: I483c7c3d413280c9e247a96ad581278347f9c71b

2021-12-09 02:55 +0000 [965f4abd9a]  Frederic Van Espen <frederic.ve@gmail.com>

	* ast_coredumper: Fix deleting results when output dir is set

	  When OUTPUTDIR is set to another directory and the
	  --delete-results-after is set, the resulting txt files are
	  not deleted.

	  ASTERISK-29794 #close

	  Change-Id: I1c0071f6809a1e3f5cfc455d6eb08378bc0d7286

2021-12-13 16:49 +0000 [bb27d5e1fe]  Naveen Albert <asterisk@phreaknet.org>

	* pbx_variables: initialize uninitialized variable

	  The variable cp4 in a variable substitution function
	  can potentially be used without being initialized
	  currently. This causes Asterisk to no longer compile.

	  This initializes cp4 to NULL to make the compiler
	  happy.

	  ASTERISK-29803 #close

	  Change-Id: I392579cbb76db2795d5820c9427cf55fbcee9e72

2021-12-08 05:24 +0000 [3d71bcd2f4]  Mark Petersen <bugs.digium.com@zombie.dk>

	* app_queue.c: added DIALEDPEERNUMBER on outgoing channel

	  added that we set DIALEDPEERNUMBER on the outgoing channels
	  so it is avalible in b(content^extension^line)
	  this add the same behaviour as Dial

	  ASTERISK-29795

	  Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6

2021-11-15 15:35 +0000 [05afa061f5]  Kevin Harwell <kharwell@sangoma.com>

	* http.c: Add ability to create multiple HTTP servers

	  Previously, it was only possible to have one HTTP server in Asterisk.
	  With this patch it is now possible to have multiple HTTP servers
	  listening on different addresses.

	  Note, this behavior has only been made available through an API call
	  from within the TEST_FRAMEWORK. Specifically, this feature has been
	  added in order to allow unit test to create/start and stop servers,
	  if one has not been enabled through configuration.

	  Change-Id: Ic5fb5f11e62c019a1c51310f4667b32a4dae52f5

2021-12-12 18:08 +0000 [030f7d4131]  Naveen Albert <asterisk@phreaknet.org>

	* app.c: Throw warnings for nonexistent options

	  Currently, Asterisk doesn't throw warnings if options
	  are passed into applications that don't accept them.
	  This can confuse users if they're unaware that they
	  are doing something wrong.

	  This adds an additional check to parse_options so that
	  a warning is thrown anytime an option is parsed that
	  doesn't exist in the parsing application, so that users
	  are notified of the invalid usage.

	  ASTERISK-29801 #close

	  Change-Id: Id029274a57135caca193c913307a63fd75e24679

2021-12-08 12:07 +0000 [a4c42e70c1]  Mark Petersen <bugs.digium.com@zombie.dk>

	* app_voicemail.c: Support for Danish syntax in VM

	  added support for playing the correct plural sound file
	  dependen on where you have 1 or multipe messages
	  based on the existing SE/NO code

	  ASTERISK-29797

	  Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2

2021-12-11 20:11 +0000 [86bc3eef99]  Naveen Albert <asterisk@phreaknet.org>

	* strings: Fix enum names in comment examples

	  The enum values for ast_strsep_flags includes
	  AST_STRSEP_STRIP. However, some comments reference
	  AST_SEP_STRIP, which doesn't exist. This fixes
	  these comments to use the correct value.

	  ASTERISK-29800 #close

	  Change-Id: If7bbd0c0e6226a211d25ddf9d1629347e2674943

2021-11-17 15:16 +0000 [c6410dc4ed]  Naveen Albert <asterisk@phreaknet.org>

	* configs: Updates to sample configs

	  Includes some minor updates to extensions.conf
	  and iax.conf. In particular, the demonstration
	  of macros in extensions.conf is removed, as
	  Macro is deprecated and will be removed soon.
	  These examples have been replaced with examples
	  demonstrating the usage of Gosub instead.

	  The older exten => ...,n syntax is also mostly
	  replaced with the same keyword to demonstrate the
	  newer, more concise way of defining extensions.

	  IAXTEL no longer exists, so this example is replaced
	  with something more generic.

	  Some documentation is also added to extensions.conf
	  and iax.conf to clarify some of the new expanded
	  encryption capabilities with IAX2.

	  ASTERISK-29758 #close

	  Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46

2021-11-17 15:39 +0000 [2f1eb56116]  Naveen Albert <asterisk@phreaknet.org>

	* app_sendtext: Add ReceiveText application

	  Adds a ReceiveText application that can be used in
	  conjunction with SendText. Currently, there is no
	  way in Asterisk to receive text in the dialplan
	  (or anywhere else, really). This allows for Asterisk
	  to be the recipient of text instead of just the sender.

	  ASTERISK-29759 #close

	  Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d

2021-11-20 14:37 +0000 [c6309af560]  Naveen Albert <asterisk@phreaknet.org>

	* pbx_variables: Increase parsing capabilities of MSet

	  Currently MSet can only parse a maximum of 24 variables.
	  If more variables are provided to MSet, the 24th variable
	  will simply contain the remainder of the string and the
	  remaining variables thereafter will never get set.

	  This increases the number of variables that can be parsed
	  in one go from 24 to 99. Additionally, documentation is added
	  since this limitation is currently undocumented and is
	  confusing to users who encounter this limitation.

	  ASTERISK-29766 #close

	  Change-Id: I3fe35b462dedec0a452fd9ea7f92c920a3939f16

2021-11-23 20:21 +0000 [3108457d8f]  Naveen Albert <asterisk@phreaknet.org>

	* chan_sip: Fix crash when accessing RURI before initiating outgoing call

	  Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before
	  initiating an outgoing call will cause Asterisk to crash. This is
	  because a null field is accessed, resulting in an offset from null and
	  subsequent memory access violation.

	  Since RURI is not guaranteed to exist, we now check if the base
	  pointer is non-null before calculating an offset.

	  ASTERISK-29772

	  Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83

2021-10-25 16:19 +0000 [c0cdaf0246]  Naveen Albert <asterisk@phreaknet.org>

	* func_json: Adds JSON_DECODE function

	  Adds the JSON_DECODE function for parsing JSON in the
	  dialplan. JSON parsing already exists in the Asterisk
	  core and is used for many different things. This
	  function exposes the basic parsing capability to
	  the user in the dialplan, for instance, in conjunction
	  with CURL for using API responses.

	  ASTERISK-29706 #close

	  Change-Id: Iea60c49a7358dfdc2db60803cdc9a742f808ba2c

2021-12-11 18:45 +0000 [087f25d3fd]  Sean Bright <sean.bright@gmail.com>

	* CHANGES: Correct reference to configuration file.

	  Change-Id: I22a788ebf11168fff7fbf9ea956ebcd705ab63dd

2021-11-15 15:08 +0000 [cc1418ef47]  Naveen Albert <asterisk@phreaknet.org>

	* pbx: Add variable substitution API for extensions

	  Currently, variable substitution involving dialplan
	  extensions is quite clunky since it entails obtaining
	  the current dialplan location, backing it up, storing
	  the desired variables for substitution on the channel,
	  performing substitution, then restoring the original
	  location.

	  In addition to being clunky, things could also go wrong
	  if an async goto were to occur and change the dialplan
	  location during a substitution.

	  Fundamentally, there's no reason it needs to be done this
	  way, so new API is added to allow for directly passing in
	  the dialplan location for the purposes of variable
	  substitution so we don't need to mess with the channel
	  information anymore. Existing API is not changed.

	  ASTERISK-29745 #close

	  Change-Id: I23273bf27fa0efb64a606eebf9aa8e2f41a065e4

2021-09-21 19:18 +0000 [2b2b708d43]  Naveen Albert <asterisk@phreaknet.org>

	* app_mf: Add full tech-agnostic MF support

	  Adds tech-agnostic support for MF signaling by adding
	  MF sender and receiver applications as well as Dial
	  integration.

	  ASTERISK-29496-mf #do-not-close

	  Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4

2021-12-06 04:25 +0000 [55dd77b921]  Alexander Traud <pabstraud@compuserve.com>

	* xmldoc: Avoid whitespace around value for parameter/required.

	  Otherwise, the value 'false' was not found in the enumerated set of
	  the XML DTD for the XML attribute 'required' in the XML element
	  'parameter'. Therefore, DTD validation of the runtime XML failed.

	  ASTERISK-29790

	  Change-Id: Id13f230ad65a70dd8c2e3ae9ac85d1e841aed03e

2021-12-04 02:36 +0000 [4b6c72572c]  Alexander Traud <pabstraud@compuserve.com>

	* progdocs: Fix Doxygen left-overs.

	  Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016

2021-12-06 05:17 +0000 [5b24edeb7c]  Alexander Traud <pabstraud@compuserve.com>

	* xmldoc: Correct definition for XML element 'matchInfo'.

	  ASTERISK-29791

	  Change-Id: I7c656498427fcadd0a5d61a54ff67e6036609725

2021-11-23 08:05 +0000 [d914e14420]  Alexander Traud <pabstraud@compuserve.com>

	* progdocs: Update Makefile.

	  In developer mode, use internal documentation as well.
	  This should produce no warnings. Fix yours!

	  In noisy mode, output all possible warnings of Doxygen.
	  This creates zillion of warnings. Double-check your current module!

	  Any warnings are in the file './doxygen.log'. Beside that, this change
	  avoids deprecated parameters because the configuration file for Doxygen
	  contains only those parameters which differ from the default. This
	  avoids the need to update the file on each run. Furthermore, it adds
	  AST_VECTOR to be expanded. Finally, the default name for that file is
	  Doxyfile. Therefore, let us use that!

	  ASTERISK-26991
	  ASTERISK-20259

	  Change-Id: I4129092a199d5e24c319a09cd088614b121015af

2021-12-03 07:38 +0000 [a103956fc9]  Alexander Traud <pabstraud@compuserve.com>

	* res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites.

	  res_sdp_crypto_parse_offer(.) emits many log messages already.

	  ASTERISK-29785

	  Change-Id: I1a191ebe4fec1102946d4e31887e5197ca02dfe8

2021-11-30 14:16 +0000 [c37cc5d3bc]  Sean Bright <sean.bright@gmail.com>

	* channel: Short-circuit ast_channel_get_by_name() on empty arg.

	  We know that passing a NULL or empty argument to
	  ast_channel_get_by_name() will never result in a matching channel and
	  will always result in an error being emitted, so just short-circuit
	  out in that case.

	  ASTERISK-28219 #close

	  Change-Id: I88eadc748e9c6996fc17467b0a05881bbfd00bce

2021-10-26 16:12 +0000 [04d00c203c]  Mike Bradeen <mbradeen@sangoma.com>

	* res_rtp_asterisk: Addressing possible rtp range issues

	  res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined
	  that rtpstart was configured to be an odd value. Also adding a loop
	  counter to prevent a possible infinite loop when looking for a free
	  port.

	  ASTERISK-27406

	  Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b

2021-08-24 09:56 +0000 [6c9e8afd4e]  Mark Petersen <bugs.digium.com@zombie.dk>

	* apps/app_dial.c: HANGUPCAUSE reason code for CANCEL is set to AST_CAUSE_NORMAL_CLEARING

	  changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING

	  ASTERISK-28053
	  Reported by: roadkill

	  Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89

2021-11-19 02:54 +0000 [178cb0ffe4]  Alexander Traud <pabstraud@compuserve.com>

	* res: Fix for Doxygen.

	  These are the remaining issues found in /res.

	  ASTERISK-29761

	  Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d

2021-11-08 18:30 +0000 [b2e71b82e7]  Dustin Marquess <jailbird@fdf.net>

	* res_fax_spandsp: Add spandsp 3.0.0+ compatibility

	  Newer versions of spandsp did refactoring of code to add new features
	  like color FAXing. This refactoring broke backwards compatibility.
	  Add support for the new version while retaining support for 0.0.6.

	  ASTERISK-29729 #close

	  Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1

2021-11-19 09:47 +0000 [20d9158c9c]  Alexander Traud <pabstraud@compuserve.com>

	* main: Fix for Doxygen.

	  ASTERISK-29763

	  Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e

2021-12-02 18:26 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.9.0-rc1 Released.

2021-12-02 11:59 +0000 [868d2d5e53]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.9.0
2021-11-28 04:29 +0000 [f946b92553]  Alexander Traud <pabstraud@compuserve.com>

	* progdocs: Fix for Doxygen, the hidden parts.

	  ASTERISK-29779

	  Change-Id: If338163488498f65fa7248b60e80299c0a928e4b

2021-11-12 10:05 +0000 [751bbf4b97]  Alexander Traud <pabstraud@compuserve.com>

	* progdocs: Fix grouping for latest Doxygen.

	  Since Doxygen 1.8.16, a special comment block is required. Otherwise
	  (pure C comment), the group command is ignored. Additionally, several
	  unbalanced group commands were fixed.

	  ASTERISK-29732

	  Change-Id: I4687857b9d56e6f44fd440b73af156691660202e

2021-11-25 12:41 +0000 [bcb7aee723]  Naveen Albert <asterisk@phreaknet.org>

	* documentation: Standardize examples

	  Most examples in the XML documentation use the
	  example tag to demonstrate examples, which gets
	  parsed specially in the Wiki to make it easier
	  to follow for users.

	  This fixes a few modules to use the example
	  tag instead of vanilla para tags to bring them
	  in line with the standard syntax.

	  ASTERISK-29777 #close

	  Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b

2021-11-28 14:52 +0000 [04ac4fe509]  Sean Bright <sean.bright@gmail.com>

	* config.c: Prevent UB in ast_realtime_require_field.

	  A backend's implementation of the realtime 'require' function may call
	  va_arg() and then fail, leaving the va_list in an undefined
	  state. Pass a copy of the va_list instead.

	  ASTERISK-29771 #close

	  Change-Id: I555565a72af84e96d49f62fe8cb66ba5a78461f4

2021-11-01 10:40 +0000 [70cdb0f9a8]  Naveen Albert <asterisk@phreaknet.org>

	* app_voicemail: Refactor email generation functions

	  Refactors generic functions used for email generation
	  into utils.c so that they can be used by multiple
	  modules, including app_voicemail and app_minivm,
	  to avoid code duplication.

	  ASTERISK-29715 #close

	  Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee

2021-11-25 10:34 +0000 [b290bb1251]  Alexander Traud <pabstraud@compuserve.com>

	* stir/shaken: Avoid a compiler extension of GCC.

	  ASTERISK-29776

	  Change-Id: I86e5eca66fb775a5744af0c929fb269e70575a73

2021-11-20 04:00 +0000 [858c9e1d80]  Alexander Traud <pabstraud@compuserve.com>

	* chan_misdn: Fix for Doxygen.

	  ASTERISK-29764

	  Change-Id: I6e5466cce03e25695c5c7d8b68c305184dcf5375

2021-11-23 07:45 +0000 [422f5389f6]  Alexander Traud <pabstraud@compuserve.com>

	* progdocs: Remove outdated references in doxyref.h.

	  ASTERISK-29773

	  Change-Id: Ica93160d9158cc0e80c5fda829b80d1b49a6b9b9

2021-11-20 06:05 +0000 [31e385bebb]  Alexander Traud <pabstraud@compuserve.com>

	* xmldoc: Fix for Doxygen.

	  ASTERISK-29765

	  Change-Id: I654ba0debe8351038d4433716434a09370f04c9d

2021-10-28 02:28 +0000 [89237be105]  Jaco Kroon <jaco@uls.co.za>

	* logger: use __FUNCTION__ instead of __PRETTY_FUNCTION__

	  This avoids a few long-name overflows, at the cost of less instructive
	  names in the case of C++ (specifically overloaded functions and class
	  methods).  This in turn is offset against the fact that we're logging
	  the filename and line numbers in any case.

	  Change-Id: I54101a0bb5f8cb9ef63ec12c5e0d4c8edafff9ed
	  Signed-off-by: Jaco Kroon <jaco@uls.co.za>

2021-11-16 16:34 +0000 [ea941032ff]  Mike Bradeen <mbradeen@sangoma.com>

	* astobj2.c: Fix core when ref_log enabled

	  In the AO2_ALLOC_OPT_LOCK_NOLOCK case the referenced obj
	  structure is freed, but is then referenced later if ref_log is
	  enabled. The change is to store the obj->priv_data.options value
	  locally and reference it instead of the value from the freed obj

	  ASTERISK-29730

	  Change-Id: I60cc5dc1f5a4330e7ad56976fc38a42de0ab6072

2021-11-19 03:46 +0000 [3f86c95cf5]  Alexander Traud <pabstraud@compuserve.com>

	* channels: Fix for Doxygen.

	  ASTERISK-29762

	  Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7

2021-11-16 04:06 +0000 [7d4e37a180]  Joshua C. Colp <jcolp@sangoma.com>

	* bridge: Deny full Local channel pair in bridge.

	  Local channels are made up of two pairs - the 1 and 2
	  sides. When a frame goes in one side, it comes out the
	  other. Back and forth. When both halves are in a
	  bridge this creates an infinite loop of frames.

	  This change makes it so that bridging no longer
	  allows both of these sides to exist in the same
	  bridge.

	  ASTERISK-29748

	  Change-Id: I29928b6de87cd9be996a77daccefd7c360fef651

2021-11-06 18:35 +0000 [ca2e13e18f]  Naveen Albert <asterisk@phreaknet.org>

	* res_tonedetect: Add call progress tone detection

	  Makes basic call progress tone detection available