ChangeLog

	  If users are able to press # for options while leaving
	  a message and then press 3 to rerecord the message, if
	  the caller hangs up during the rerecord prompt but before
	  Asterisk starts recording a message, then an "empty"
	  voicemail gets processed whereby an email gets sent out
	  notifying the user of a 0:00 duration message. The file
	  doesn't actually exist, so playback will fail since there
	  was no message to begin with.

	  This adds a check after the streaming of the rerecord
	  announcement to see if the caller has hung up. If so,
	  we bail out early so that we can clean up properly.

	  ASTERISK-29391 #close

	  Change-Id: Id965d72759a2fd3b39afb76fec08aaebebe75c31

2021-10-25 19:47 +0000 [bea08a563b]  Naveen Albert <asterisk@phreaknet.org>

	* chan_iax2: Allow both secret and outkey at dial time

	  Historically, the dial syntax for IAX2 has held that
	  an outkey (used only for RSA authenticated calls)
	  and a secret (used only for plain text and MD5 authenticated
	  calls, historically) were mutually exclusive, and thus
	  the same position in the dial string was used for both
	  values.

	  Now that encryption is possible with RSA authentication,
	  this poses a limitation, since encryption requires a
	  secret and RSA authentication requires an outkey. Thus,
	  the dial syntax is extended so that both a secret and
	  an outkey can be specified.

	  The new extended syntax is backwards compatible with the
	  old syntax. However, a secret can now be specified after
	  the outkey, or the outkey can be specified after the secret.
	  This makes it possible to spawn an encrypted RSA authenticated
	  call without a corresponding peer being predefined in iax.conf.

	  ASTERISK-29707 #close

	  Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e

2021-10-28 07:09 +0000 [95da40cd50]  Alexander Traud <pabstraud@compuserve.com>

	* res_snmp: As build tool, prefer pkg-config over net-snmp-config.

	  ASTERISK-29709

	  Change-Id: Ie169df878bdfc3a06b3097c5c38d185b480f54d4

2021-10-28 07:41 +0000 [8b76a3cd3b]  Alexander Traud <pabstraud@compuserve.com>

	* stasis: Avoid 'dispatched' as unused variable in normal mode.

	  ASTERISK-29710

	  Change-Id: Ia849f1172e4e694c5d5d7f0cad449f936ee12216

2021-10-29 10:05 +0000 [2c03f73016]  Sean Bright <sean@seanbright.com>

	* various: Fix GCC 11.2 compilation issues.

	  * Initialize some variables that are never used anyway.

	  * Use valid pointers instead of integers cast to void pointers when
	    calling pthread_setspecific().

	  ASTERISK-29711 #close
	  ASTERISK-29713 #close

	  Change-Id: I8728cd6f2f4b28e0e48113c5da450b768c2a6683

2021-09-09 09:39 +0000 [08cb67251f]  George Joseph <gjoseph@digium.com>

	* ast_coredumper:  Refactor to better find things

	  The search for a running asterisk when --running is used
	  has been greatly simplified and in the event it doesn't
	  work, you can now specify a pid to use on the command
	  line with --pid.

	  The search for asterisk modules when --tarball-coredumps
	  is used has been enhanced to have a better chance of finding
	  them and in the event it doesn't work, you can now specify
	  --libdir on the command line to indicate the library directory
	  where they were installed.

	  The DATEFORMAT variable was renamed to DATEOPTS and is now
	  passed to the 'date' utility rather than running DATEFORMAT
	  as a command.

	  The coredump and output files are now renamed with DATEOPTS.
	  This can be disabled by specifying --no-rename.

	  Several confusing and conflicting options were removed:
	  --append-coredumps
	  --conffile
	  --no-default-search
	  --tarball-uniqueid

	  The script was re-structured to make it easier for follow.

	  Change-Id: I674be64bdde3ef310b6a551d4911c3b600ffee59

2021-10-21 12:29 +0000 [ae97aaedb0]  Kevin Harwell <kharwell@sangoma.com>

	* strings/json: Add string delimter match, and object create with vars methods

	  Add a function to check if there is an exact match a one string between
	  delimiters in another string.

	  Add a function that will create an ast_json object out of a list of
	  Asterisk variables. An excludes string can also optionally be passed
	  in.

	  Also, add a macro to make it easier to get object integers.

	  Change-Id: I5f34f18e102126aef3997f19a553a266d70d6226

2021-09-21 12:09 +0000 [2e55c0fded]  Ben Ford <bford@digium.com>

	* STIR/SHAKEN: Option split and response codes.

	  The stir_shaken configuration option now has 4 different choices to pick
	  from: off, attest, verify, and on. Off and on behave the same way they
	  do now. Attest will only perform attestation on the endpoint, and verify
	  will only perform verification on the endpoint.

	  Certain responses are required to be sent based on certain conditions
	  for STIR/SHAKEN. For example, if we get a Date header that is outside of
	  the time range that is considered valid, a 403 Stale Date response
	  should be sent. This and several other responses have been added.

	  Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7

2021-08-25 08:15 +0000 [a203769c9d]  Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>

	* app_queue: Add LoginTime field for member in a queue.

	  Add a time_t logintime to storage a time when a member is added into a
	  queue.

	  Also, includes show this time (in seconds) using a 'queue show' command
	  and the field LoginTime for response for AMI events.

	  ASTERISK-18069 #close

	  Change-Id: Ied6c3a300f78d78eebedeb3e16a1520fc3fff190

2021-10-21 12:49 +0000 [859f579504]  Kevin Harwell <kharwell@sangoma.com>

	* res_speech: Add a type conversion, and new engine unregister methods

	  Add a new function that converts a speech results type to a string.
	  Also add another function to unregister an engine, but returns a
	  pointer to the unregistered engine object instead of a success/fail
	  integer.

	  Change-Id: I0f7de17cb411021c09fb03988bc2b904e1380192

2021-10-07 13:07 +0000 [0b2646aee6]  Mike Bradeen <mbradeen@sangoma.com>

	* various: Fix GCC 11 compilation issues.

	  test_voicemail_api: Use empty char* for empty_msg_ids.
	  chan_skinny: Fix size of calledParty to be maximum extension.
	  menuselect: Change Makefile to stop deprecated warnings. Added comments
	  test_linkedlist: 'bogus' variable was manually allocated from a macro
	  and the test fails if this happens but the compiler couldn't 'see' this
	  and returns a warning. memset to all 0's after allocation.
	  chan_ooh323: Fixed various indentation issues that triggered misleading
	   indentation warnings.

	  ASTERISK-29682
	  Reported by: George Joseph

	  Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe

2021-09-20 11:10 +0000 [63c8d12e95]  Shloime Rosenblum <shloimerosenblum@gmail.com>

	* apps/app_playback.c: Add 'mix' option to app_playback

	  I am adding a mix option that will play by filename and say.conf unlike
	  say option that will only play with say.conf. It
	  will look on the format of the name, if it is like say it play with
	  say.conf if not it will play the file name.

	  ASTERISK-29662

	  Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a

2021-10-19 11:35 +0000 [c07e3c2f4d]  George Joseph <gjoseph@digium.com>

	* BuildSystem: Check for alternate openssl packages

	  OpenSSL is one of those packages that often have alternatives
	  with later versions.  For instance, CentOS/EL 7 has an
	  openssl package at version 1.0.2 but there's an openssl11
	  package from the epel repository that has 1.1.1.  This gets
	  installed to /usr/include/openssl11 and /usr/lib64/openssl11.
	  Unfortunately, the existing --with-ssl and --with-crypto
	  ./configure options expect to point to a source tree and
	  don't work in this situation.  Also unfortunately, the
	  checks in ./configure don't use pkg-config.

	  In order to make this work with the existing situation, you'd
	  have to run...
	  ./configure --with-ssl=/usr/lib64/openssl11 \
	      --with-crypto=/usr/lib64/openssl11 \
	      CFLAGS=-I/usr/include/openssl11

	  BUT...  those options don't get passed down to bundled pjproject
	  so when you run make, you have to include the CFLAGS again
	  which is a big pain.

	  Oh...  To make matters worse, although you can specify
	  PJPROJECT_CONFIGURE_OPTS on the ./configure command line,
	  they don't get saved so if you do a make clean, which will
	  force a re-configure of bundled pjproject, those options
	  don't get used.

	  So...

	  * In configure.ac... Since pkg-config is installed by install_prereq
	    anyway, we now use it to check for the system openssl >= 1.1.0.
	    If that works, great.  If not, we check for the openssl11
	    package. If that works, great.  If not, we fall back to just
	    checking for any openssl.  If pkg-config isn't installed for some
	    reason, or --with-ssl=<dir> or --with-crypto=<dir> were specified
	    on the ./configure command line, we fall back to the existing
	    logic that uses AST_EXT_LIB_CHECK().

	  * The whole OpenSSL check process has been moved up before
	    THIRD_PARTY_CONFIGURE(), which does the initial pjproject
	    bundled configure, is run.  This way the results of the above
	    checks, which may result in new include or library directories,
	    is included.

	  * Although not strictly needed for openssl, We now save the value of
	    PJPROJECT_CONFIGURE_OPTS in the makeopts file so it can be used
	    again if a re-configure is triggered.

	  ASTERISK-29693

	  Change-Id: I341ab7603e6b156aa15a66f43675ac5029d5fbde

2021-10-14 14:38 +0000 [8c2720e540]  Sean Bright <sean.bright@gmail.com>

	* func_talkdetect.c: Fix logical errors in silence detection.

	  There are 3 separate changes here:

	  1. The documentation erroneously stated that the dsp_talking_threshold
	     argument was a number of milliseconds when it is actually an energy
	     level used by the DSP code to classify talking vs. silence.

	  2. Fixes a copy paste error in the argument handling code.

	  3. Don't erroneously switch to the talking state if we aren't actively
	     handling a frame we've classified as talking.

	  Patch inspired by one provided by Moritz Fain (License #6961).

	  ASTERISK-27816 #close

	  Change-Id: I5953fd570b98b49c41cee55bfe3b941753fb2511

2021-10-14 10:15 +0000 [e3466893e9]  Sebastien Duthil <sduthil@wazo.community>

	* main/stun.c: fix crash upon STUN request timeout

	  Some ast_stun_request users do not provide a destination address when
	  sending to a connection-mode socket.

	  ASTERISK-29691

	  Change-Id: Idd9114c3380216ba48abfc3c68619e79ad37defc

2021-10-12 13:17 +0000 [bac66e9743]  Mike Bradeen <mbradeen@sangoma.com>

	* build: prevent binary downloads for non x86 architectures

	  download_externals: Add check for i686 and i386 (in addition
	  to the current x86_64) and exit if not one of the three.

	  ASTERISK-26497

	  Change-Id: Ia4d429fcefa5b2f5b6e99159d4607de8e8325b2f

2021-10-11 14:04 +0000 [482281deff]  Sean Bright <sean.bright@gmail.com>

	* configure: Remove unused OpenSSL SRTP check.

	  Discovered while looking at ASTERISK~29684. Usage was removed in change
	  I3c77c7b00b2ffa2e935632097fa057b9fdf480c0.

	  Change-Id: Iaf2f7a16ea5a7eee6375319347e4b40b8e7b10e3

2021-10-13 10:26 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.8.0-rc1 Released.

2021-10-13 05:21 +0000 [9063680148]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.8.0
2021-10-07 12:50 +0000 [804b1987fb]  Sean Bright <sean.bright@gmail.com>

	* Makefile: Use basename in a POSIX-compliant way.

	  If you aren't using GNU coreutils, chances are that your basename
	  doesn't know about the -s argument. Luckily for us, basename does what
	  we need it do even without the -s argument.

	  Change-Id: I8b81a429bb037b997ee6640ff8a2b5e860962bb7

2021-10-05 19:59 +0000 [e091aa2763]  Mark Murawski <markm@intellasoft.net>

	* pbx_ael:  Fix crash and lockup issue regarding 'ael reload'

	  Avoid infinite recursion and crash

	  Change-Id: I8ed05ec3aa2806c50c77edc5dd0cd4e4fa08b3f4

2021-05-24 13:04 +0000 [437b2bfbd6]  Naveen Albert <asterisk@phreaknet.org>

	* chan_iax2: Add encryption for RSA authentication

	  Adds support for encryption to RSA-authenticated
	  calls. Also prevents crashes if an RSA IAX2 call
	  is initiated to a switch requiring encryption
	  but no secret is provided.

	  ASTERISK-20219

	  Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40

2021-07-19 11:34 +0000 [15e432220c]  Matthew Kern <mkern@alconconstruction.com>

	* res_pjsip_t38: bind UDPTL sessions like RTP

	  In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
	  fallback use of the transport's bind address solve problems sending
	  media on systems that cannot send ipv4 packets on ipv6 sockets, and
	  certain other situations. This change extends both of these behaviors
	  to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
	  problems on these systems, introducing a new option
	  endpoint/t38_bind_udptl_to_media_address.

	  ASTERISK-29402

	  Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557

2021-09-29 12:58 +0000 [5a6f140765]  Naveen Albert <asterisk@phreaknet.org>

	* app_read: Fix null pointer crash

	  If the terminator character is not explicitly specified
	  and an indications tone is used for reading a digit,
	  there is no null pointer check so Asterisk crashes.
	  This prevents null usage from occuring.

	  ASTERISK-29673 #close

	  Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac

2021-09-29 04:32 +0000 [0ab4e7491d]  Jean Aunis <jean.aunis@prescom.fr>

	* res_rtp_asterisk: fix memory leak

	  Add missing reference decrement in rtp_deallocate_transport()

	  ASTERISK-29671

	  Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9

2021-09-19 15:08 +0000 [29c44caecb]  Shloime Rosenblum <shloimerosenblum@gmail.com>

	* main/say.c: Support future dates with Q and q format params

	  The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today

	  ASTERISK-29637

	  Change-Id: I1fb1cef0ce3c18d87b1fc94ea309d13bc344af02

2021-07-21 16:36 +0000 [4368764032]  Joseph Nadiv <ynadiv@corpit.xyz>

	* res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts

	  The behavior of max_contacts and remove_existing are connected.  If
	  remove_existing is enabled, the soonest expiring contacts are removed.
	  This may occur when there is an unavailable contact.  Similarly,
	  when remove_existing is not enabled, registrations from good
	  endpoints are rejected in favor of retaining unavailable contacts.

	  This commit adds a new AOR option remove_unavailable, and the effect
	  of this setting will depend on remove_existing.  If remove_existing
	  is set to no, we will still remove unavailable contacts when they
	  exceed max_contacts, if there are any. If remove_existing is set to
	  yes, we will prioritize the removal of unavailable contacts before
	  those that are expiring soonest.

	  ASTERISK-29525

	  Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784

2021-09-23 09:13 +0000 [ea36473c45]  Joshua C. Colp <jcolp@sangoma.com>

	* ari: Ignore invisible bridges when listing bridges.

	  When listing bridges we go through the ones present in
	  ARI, get their snapshot, turn it into JSON, and add it
	  to the payload we ultimately return.

	  An invisible "dial bridge" exists within ARI that would
	  also try to be added to this payload if the channel
	  "create" and "dial" routes were used. This would ultimately
	  fail due to invisible bridges having no snapshot
	  resulting in the listing of bridges failing.

	  This change makes it so that the listing of bridges
	  ignores invisible ones.

	  ASTERISK-29668

	  Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a

2021-09-19 06:14 +0000 [484da42d6c]  Naveen Albert <asterisk@phreaknet.org>

	* func_vmcount: Add support for multiple mailboxes

	  Allows multiple mailboxes to be specified for VMCOUNT
	  instead of just one.

	  ASTERISK-29661 #close

	  Change-Id: I9108528300795fd5b607efa9d4dd7b74be031813

2021-09-21 09:58 +0000 [e98839b73c]  Sean Bright <sean.bright@gmail.com>

	* message.c: Support 'To' header override with AMI's MessageSend.

	  The MessageSend AMI action has been updated to allow the Destination
	  and the To addresses to be provided separately. This brings the
	  MessageSend manager command in line with the capabilities of the
	  MessageSend dialplan application.

	  ASTERISK-29663 #close

	  Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c

2021-09-15 13:21 +0000 [cf0d656ae6]  Naveen Albert <asterisk@phreaknet.org>

	* func_channel: Add CHANNEL_EXISTS function.

	  Adds a function to check for the existence of a channel by
	  name or by UNIQUEID.

	  ASTERISK-29656 #close

	  Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb

2021-09-05 13:11 +0000 [cfd0246d11]  Naveen Albert <asterisk@phreaknet.org>

	* app_queue: Fix hint updates for included contexts

	  Previously, if custom hints were used with the hint:
	  format in app_queue, when device state changes occured,
	  app_queue would only do a literal string comparison of
	  the context used for the hint in app_queue and the context
	  of the hint which just changed state. This caused hints
	  to not update and become stale if the context associated
	  with the agent included the context which actually changes
	  state, essentially completely breaking device state for
	  any such agents defined in this manner.

	  This fix adds an additional check to ensure that included
	  contexts are also compared against the context which changed
	  state, so that the behavior is correct no matter whether the
	  context is specified to app_queue directly or indirectly.

	  ASTERISK-29578 #close

	  Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78

2021-09-10 09:40 +0000 [b2c834e349]  Sean Bright <sean.bright@gmail.com>

	* res_http_media_cache.c: Compare unaltered MIME types.

	  Rather than stripping parameters from Content-Type headers before
	  comparison, first try to compare the whole string. If no match is
	  found, strip the parameters and try that way.

	  ASTERISK-29275 #close

	  Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f

2021-07-25 17:19 +0000 [a65bb134f5]  Naveen Albert <asterisk@phreaknet.org>

	* logger: Add custom logging capabilities

	  Adds the ability for users to log to custom log levels
	  by providing custom log level names in logger.conf. Also
	  adds a logger show levels CLI command.

	  ASTERISK-29529

	  Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702

2021-09-17 10:57 +0000 [dce142baa4]  Sean Bright <sean.bright@gmail.com>

	* app_externalivr.c: Fix mixed leading whitespace in source code.

	  No functional changes.

	  Change-Id: I46514152c0af67f395526374aaa847ccd6a85378

2021-09-17 14:58 +0000 [03377c35fc]  Guido Falsi <madpilot@freebsd.org>

	* res_rtp_asterisk.c: Fix build failure when not building with pjproject.

	  Some code has been added referencing symbols defined in a block
	  protected by #ifdef HAVE_PJPROJECT. Protect those code parts in
	  ifdef blocks too.

	  ASTERISK-29660

	  Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f

2021-09-16 13:43 +0000  Asterisk Development Team <asteriskteam@digium.com>

	* asterisk 18.7.0-rc1 Released.

2021-09-16 08:39 +0000 [00cf86dafe]  Asterisk Development Team <asteriskteam@digium.com>

	* Update CHANGES and UPGRADE.txt for 18.7.0
2021-09-13 10:18 +0000 [e8f7b53023]  Carlos Oliva <carlos.oliva@invoxcontact.com>

	* app_mp3: Force output to 16 bits in mpg123

	  In new mpg123 versions (since 1.26) the default output is 32 bits
	  Asterisk expects the output in 16 bits, so we force the output to be on 16 bits.
	  It will work wit new and old versions of mpg123.
	  Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key!

	  ASTERISK-29635 #close

	  Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816

2021-09-14 12:02 +0000 [0947c30224]  George Joseph <gjoseph@digium.com>

	* pjproject: Add patch to fix trailing whitespace issue in rtpmap

	  An issue was found where a particular manufacturer's phones add a
	  trailing space to the end of the rtpmap attribute when specifying
	  a payload type that has a "param" after the format name and clock
	  rate. For example:

	  a=rtpmap:120 opus/48000/2 \r\n

	  Because pjmedia_sdp_attr_get_rtpmap currently takes everything after
	  the second '/' up to the line end as the param, the space is
	  included in future comparisons, which then fail if the param being
	  compared to doesn't also have the space.

	  We now use pj_scan_get() to parse the param part of rtpmap so
	  trailing whitespace is automatically stripped.

	  ASTERISK-29654

	  Change-Id: Ibd0a4e243a69cde7ba9312275b13ab62ab86bc1b

2021-06-08 15:44 +0000 [1a23c9c047]  Naveen Albert <asterisk@phreaknet.org>

	* res_pjsip_caller_id: Add ANI2/OLI parsing

	  Adds parsing of ANI II digits (Originating
	  Line Information) to PJSIP, on par with
	  what currently exists in chan_sip.

	  ASTERISK-29472

	  Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847

2021-06-28 10:37 +0000 [60daa8f761]  Naveen Albert <asterisk@phreaknet.org>

	* app_mf: Add channel agnostic MF sender

	  Adds a SendMF application and PlayMF manager
	  event to send arbitrary R1 MF tones on the
	  current or specified channel.

	  ASTERISK-29496

	  Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4

2021-09-10 09:56 +0000 [847349853a]  Sean Bright <sean.bright@gmail.com>

	* test_http_media_cache.c: Fix copy/paste error during test deregistration.

	  Change-Id: I9a3a978b2f818be464e062d97b93831b127ef28c

2021-09-02 18:20 +0000 [c736cef310]  Naveen Albert <asterisk@phreaknet.org>

	* app_stack: Include current location if branch fails

	  Previously, the error emitted when app_stack tries
	  to branch to a dialplan location that doesn't exist
	  has included only the information about the attempted
	  branch in the error log. This adds the current location
	  as well so users can see where the branch failed in
	  the logs.

	  ASTERISK-29626

	  Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce

2021-09-03 13:27 +0000 [d9747104ff]  Sungtae Kim <pchero21@gmail.com>

	* resource_channels.c: Fix external media data option

	  Fixed the external media creation handle to handle the 'data' option correctly.

	  ASTERISK-29629

	  Change-Id: I22e57fe8ebf3d3e08fb2121aa4a8a52cc62e8129

2021-09-02 18:57 +0000 [6198c1d28c]  Naveen Albert <asterisk@phreaknet.org>

	* func_strings: Add STRBETWEEN function

	  Adds the STRBETWEEN function, which can be used to insert a
	  substring between each character in a string. For instance,
	  this can be used to insert pauses between DTMF tones in a
	  string of digits.

	  ASTERISK-29627

	  Change-Id: Ice23009d4a8e9bb9718d2b2301d405567087d258

2021-09-08 14:29 +0000 [ee62a07914]  Sean Bright <sean.bright@gmail.com>

	* test_abstract_jb.c: Fix put and put_out_of_order memory leaks.

	  We can't rely on RAII_VAR(...) to properly clean up data that is
	  allocated within a loop.

	  ASTERISK-27176 #close

	  Change-Id: Ib575616101230c4f603519114ec62ebf3936882c

2021-09-02 19:00 +0000 [19de228e8b]  Naveen Albert <asterisk@phreaknet.org>

	* func_env: Add DIRNAME and BASENAME functions

	  Adds the DIRNAME and BASENAME functions, which are
	  wrappers around the corresponding C library functions.
	  These can be used to safely and conveniently work with
	  file paths and names in the dialplan.

	  ASTERISK-29628 #close

	  Change-Id: Id3aeb907f65c0ff96b6e57751ff0cb49d61db7f3

2021-07-26 12:46 +0000 [b6b7b1490b]  Naveen Albert <asterisk@phreaknet.org>

	* func_sayfiles: Retrieve say file names

	  Up until now, all of the logic used to translate
	  arguments to the Say applications has been
	  directly coupled to playback, preventing other
	  modules from using this logic.

	  This refactors code in say.c and adds a SAYFILES
	  function that can be used to retrieve the file
	  names that would be played. These can then be
	  used in other applications or for other purposes.

	  Additionally, a SayMoney application and a SayOrdinal
	  application are added. Both SayOrdinal and SayNumber
	  are also expanded to support integers greater than
	  one billion.

	  ASTERISK-29531

	  Change-Id: If9718c89353b8e153d84add3cc4637b79585db19

2021-08-09 12:41 +0000 [a6eb1b6f95]  Naveen Albert <asterisk@phreaknet.org>

	* res_tonedetect: Tone detection module

	  dsp.c contains arbitrary tone detection functionality
	  which is currently only used for fax tone recognition.
	  This change makes this functionality publicly
	  accessible so that other modules can take advantage
	  of this.

	  Additionally, a WaitForTone and TONE_DETECT app and
	  function are included to allow users to do their
	  own tone detection operations in the dialplan.

	  ASTERISK-29546

	  Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26

2021-09-08 09:36 +0000 [2806a45034]  George Joseph <gjoseph@digium.com>

	* res_snmp: Add -fPIC to _ASTCFLAGS

	  With gcc 11, res/res_snmp.c and res/snmp/agent.c need the
	  -fPIC option added to its _ASTCFLAGS.

	  ASTERISK-29634

	  Change-Id: I34649c85e075fd954e578378fabf798c3f038f50

2021-09-04 12:07 +0000 [858cb386fd]  Sean Bright <sean.bright@gmail.com>

	* term.c: Add support for extended number format terminfo files.

	  ncurses 6.1 introduced an extended number format for terminfo files
	  which the terminfo parsing in Asterisk is not able to parse. This
	  results in some TERM values that do support color (screen-256color on
	  Ubuntu 20.04 for example) to not get a color console.

	  ASTERISK-29630 #close

	  Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3

2021-09-07 12:32 +0000 [347e9a7e4d]  Sean Bright <sean.bright@gmail.com>

	* app_voicemail.c: Ability to silence instructions if greeting is present.

	  There is an option to silence voicemail instructions but it does not
	  take into consideration if a recorded greeting exists or not. Add a
	  new 'S' option that does that.

	  ASTERISK-29632 #close

	  Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4

2021-09-03 00:30 +0000 [c1a575907b]  Jasper Hafkenscheid <jasper.hafkenscheid@wearespindle.com>

	* res_srtp: Disable parsing of not enabled cryptos

	  When compiled without extended srtp crypto suites also disable parsing
	  these from received SDP. This prevents using these, as some client
	  implementations are not stable.

	  ASTERISK-29625

	  Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a

2021-09-06 11:37 +0000 [689c703b2c]  Sean Bright <sean.bright@gmail.com>

	* dns.c: Load IPv6 DNS resolvers if configured.

	  IPv6 nameserver addresses are stored in different part of the
	  __res_state structure, so look there if we appear to have support for
	  it.

	  ASTERISK-28004 #close

	  Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d

2021-09-08 07:52 +0000 [de19836c24]  George Joseph <gjoseph@digium.com>

	* bridge_softmix: Suppress error on topology change failure

	  There are conditions under which a failure to change topology
	  is expected so there's no need to print an ERROR message.

	  ASTERISK-29618
	  Reported by: Alexander

	  Change-Id: Idc168b8588e018bf3a23769f08c4ad646086d481

2021-08-31 02:50 +0000 [479cc17f45]  sungtae kim <sungtae.kim@avoxi.com>

	* resource_channels.c: Fix wrong external media parameter parse

	  Fixed ARI external media handler to accept body parameters.

	  ASTERISK-29622

	  Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d

2021-08-25 10:21 +0000 [5c836c8e36]  Sean Bright <sean.bright@gmail.com>

	* config_options: Handle ACO arrays correctly in generated XML docs.

	  There are 3 separate changes here but they are all closely related:

	  * Only try to set matchfield attributes on 'field' nodes

	  * We need to adjust how we treat the category pointer based on the
	    value of the category_match, to avoid memory corruption. We now
	    generate a regex-like string when match types other than
	    ACO_WHITELIST and ACO_BLACKLIST are used.

	  * Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
	    ACO_BLACKLIST_EXACT since we only have one category we need to
	    ignore, not two.

	  ASTERISK-29614 #close

	  Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e

2021-08-18 14:44 +0000 [5a685249ce]  Naveen Albert <asterisk@phreaknet.org>

	* chan_iax2: Add ANI2/OLI information element

	  Adds an information element for ANI2 so that
	  Originating Line Information can be transmitted
	  over IAX2 channels.

	  ASTERISK-29605 #close

	  Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2

2021-08-31 15:03 +0000 [042ae05be7]  Mark Murawski <markm@intellasoft.net>

	* pbx_ael:  Fix crash and lockup issue regarding 'ael reload'

	  Currently pbx_ael does not check if a reload is currently pending
	  before proceeding with a reload. This can cause multiple threads to
	  operate at the same time on what should be mutex protected data. This
	  change adds protection to reloading to ensure only one ael reload is
	  executing at a time.

	  ASTERISK-29609 #close

	  Change-Id: I5ed392ad226f6e4e7696ad742076d3e45c57af35

2021-08-25 06:49 +0000 [dd980e00b4]  Naveen Albert <asterisk@phreaknet.org>

	* app_read: Allow reading # as a digit

	  Allows for the digit # to be read as a digit,
	  just like any other DTMF digit, as opposed to
	  forcing it to be used as an end of input
	  indicator. The default behavior remains
	  unchanged.

	  ASTERISK-18454 #close

	  Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b

2021-04-05 14:06 +0000 [ac492f2ff8]  Sebastien Duthil <sduthil@wazo.community>

	* res_rtp_asterisk: Automatically refresh stunaddr from DNS

	  This allows the STUN server to change its IP address without having to
	  reload the res_rtp_asterisk module.

	  The refresh of the name resolution occurs first when the module is
	  loaded, then recurringly, slightly after the previous DNS answer TTL
	  expires.

	  ASTERISK-29508 #close

	  Change-Id: I7955a046293f913ba121bbd82153b04439e3465f

2021-08-24 20:04 +0000 [e660a2c03b]  Naveen Albert <asterisk@phreaknet.org>

	* bridge_basic: Change warning to verbose if transfer cancelled

	  The attended transfer feature will emit a warning if the user
	  cancels the transfer or the attended transfer doesn't complete
	  for any reason. Changes the warning to a verbose message,
	  since nothing is actually wrong here.

	  ASTERISK-29612 #close

	  Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d

2021-08-20 15:35 +0000 [c7af46995e]  Naveen Albert <asterisk@phreaknet.org>

	* app_queue: Don't reset queue stats on reload

	  Prevents reloads of app_queue from also resetting
	  queue statistics.

	  Also preserves individual queue agent statistics
	  if we're just reloading members.

	  ASTERISK-28701

	  Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1

2021-08-25 09:23 +0000 [82d6bd7ec9]  Alexander Traud <pabstraud@compuserve.com>

	* res_rtp_asterisk: sqrt(.) requires the header math.h.

	  ASTERISK-29616

	  Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900

2021-08-25 09:29 +0000 [8410afc7ab]  Alexander Traud <pabstraud@compuserve.com>

	* dialplan: Add one static and fix two whitespace errors.

	  Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83

2021-06-19 23:36 +0000 [241686f860]  Sarah Autumn <sarah@connectionsmuseum.org>

	* sig_analog: Changes to improve electromechanical signalling compatibility

	  This changeset is intended to address compatibility issues encountered
	  when interfacing Asterisk to electromechanical telephone switches that
	  implement ANI-B, ANI-C, or ANI-D.

	  In particular the behaviours that this impacts include:

	   - FGC-CAMA did not work at all when using MF signaling. Modified the
	     switch case block to send calls to the correct part of the
	     signaling-handling state machine.

	   - For FGC-CAMA operation, the delay between called number ST and
	     second wink for ANI spill has been made configurable; previously
	     all calls were made to wait for one full second.

	   - After the ANI spill, previous behavior was to require a 'ST' tone
	     to advance the call.  This has been changed to allow 'STP' 'ST2P'
	     or 'ST3P' as well, for compatibility with ANI-D.

	   - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.

	   - For calls with an ANI failure, No. 1 Crossbar switches will send
	     forward a single-digit failure code, with no calling number digits
	     and no ST pulse to terminate the spill.  I've made the ANI timeout
	     configurable so to reduce dead air time on calls with ANI fail.

	   - ANI info digits configurable.  Modern digital switches will send 2
	     digits, but ANI-B sends only a single info digit.  This caused the
	     ANI reported by Asterisk to be misaligned.

	   - Changed a confusing log message to be more informative.

	  ASTERISK-29518

	  Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256

2021-08-05 11:55 +0000 [eb486db3af]  Andre Barbosa <andre.emanuel.barbosa@gmail.com>

	* media_cache: Don't lock when curl the remote file

	  When playing a remote sound file, which is not in cache, first we need
	  to download it with ast_bucket_file_retrieve.

	  This can take a while if the remote host is slow. The current CURL
	  timeout is 180secs, so in extreme situations, it can take 3 minutes to
	  return.

	  Because ast_media_cache_retrieve has a lock on all function, while we
	  are waiting for the delayed download, Asterisk is not able to play any
	  more files, even the files already cached locally.

	  ASTERISK-29544 #close

	  Change-Id: I8d4142b463ae4a1d4c41bff2bf63324821567408

2021-08-16 08:25 +0000 [b72425b1f0]  George Joseph <gjoseph@digium.com>

	* res_pjproject: Allow mapping to Asterisk TRACE level

	  Allow mapping pjproject log messages to the Asterisk TRACE
	  log level.  The defaults were also changes to log pjproject
	  levels 3,4 to DEBUG and 5,6 to TRACE.  Previously 3,4,5,6
	  all went to DEBUG.

	  ASTERISK-29582

	  Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d

2021-08-12 16:02 +0000 [dffc5e7f5c]  Naveen Albert <asterisk@phreaknet.org>

	* app_milliwatt: Timing fix

	  The Milliwatt application uses incorrect tone timings
	  that cause it to play the 1004 Hz tone constantly.

	  This adds an option to enable the correct timing
	  behavior, so that the Milliwatt application can
	  be used for milliwatt test lines. The default behavior
	  remains unchanged for compatability reasons, even
	  though it is incorrect.

	  ASTERISK-29575 #close

	  Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c

2021-06-28 09:25 +0000 [c52ef4ac79]  Naveen Albert <asterisk@phreaknet.org>

	* func_math: Return integer instead of float if possible

	  The MIN, MAX, and ABS functions all support float
	  arguments, but currently return floats even if the