Update CHANGES and UPGRADE.txt for 20.0.0

doc/CHANGES-staging/app_read.txt

-Subject: app_read
-
-A new option allows the digit '#' to be read literally,
-rather than used exclusively as the input terminator
-character.

doc/CHANGES-staging/app_sendtext.txt

-Subject: app_sendtext
-
-A ReceiveText application has been added that can be
-used in conjunction with the SendText application.

doc/CHANGES-staging/app_voicemail.txt

-Subject: app_voicemail
-
-Add a new 'S' option to VoiceMail which prevents the instructions
-(vm-intro) from being played if a busy/unavailable/temporary greeting
-from the voicemail user is played. This is similar to the existing 's'
-option except that instructions will still be played if no user
-greeting is available.

doc/CHANGES-staging/app_voicemail_danish_syntax.txt

-Subject: app_voicemail
-Subject: Applications
-
-added support for Danish syntax, playing the correct plural sound file
-dependen on where you have 1 or multipe messages
-based on the existing SE/NO code

doc/CHANGES-staging/app_voicemail_nodelete.txt

-Subject: app_voicemail
-
-The r option has been added, which prevents deletion
-of messages from VoiceMailMain, which can be
-useful for shared mailboxes.

doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt

-Subject: ari
-Subject: stasis_channels
-
-Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
-to ARI channel resources as 'protocol_id'.
-
-ASTERISK-30027

doc/CHANGES-staging/ast_coredumper.txt

-Subject: ast_coredumper
-
-New options:
- --pid=<asterisk_pid>
-   Allows specification of an Asterisk instance when trying to
-   and the script can't determine it itself.
- --libdir=<system library directory>
-   Allows specification of a non-standard installation directory
-   containing the Asterisk modules.
- --(no-)rename
-   Renames the coredump and the output files with readable
-   timestamps. This is the default.
-Removed unneeded or confusing options:
- --append-coredumps
- --conffile
- --no-default-search
- --tarball-uniqueid
-Changed Variables:
- COREDUMPS is now just "/tmp/core!(*.txt)"
- DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ'
-Changed behavior:
- If you use 'running' or 'RUNNING' you no longer need to specify
- '--no-default-search' to ignore existing coredumps.

doc/CHANGES-staging/bundled-pjproject-build.txt

-Subject: Core
-
-Bundled PJProject Build
-
-The build process has been updated to make pjproject troubleshooting
-and development easier. See third-party/pjproject/README-hacking.md or
-https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject
-for more info.

doc/CHANGES-staging/cdr_disable.txt

-Subject: cdr
-
-A new CDR option, channeldefaultenabled, allows controlling
-whether CDR is enabled or disabled by default on
-newly created channels. The default behavior remains
-unchanged from previous versions of Asterisk (new
-channels will have CDR enabled, as long as CDR is
-enabled globally).

doc/CHANGES-staging/chan_dahdi_cadences.txt

-Subject: chan_dahdi
-
-Previously, cadences were appended on dahdi restart,
-rather than reloaded. This prevented cadences from
-being updated and maxed out the available cadences
-if reloaded multiple times. This behavior is fixed
-so that reloading cadences is idempotent and cadences
-can actually be reloaded.

doc/CHANGES-staging/chan_dahdi_polarity.txt

-Subject: chan_dahdi
-
-A POLARITY function is now available that allows
-getting or setting the polarity on a channel
-from the dialplan.

doc/CHANGES-staging/chan_iax2_ani2.txt

-Subject: chan_iax2
-
-ANI2 (OLI) is now transmitted over IAX2 calls
-as an information element.

doc/CHANGES-staging/chan_iax2_dial.txt

-Subject: chan_iax2
-
-Both a secret and an outkey may be specified at dial time,
-since encryption is possible with RSA authentication.

doc/CHANGES-staging/chan_pjsip_180_sdp.txt

-Subject: chan_pjsip
-
-added global config option "allow_sending_180_after_183"
-
-Allow Asterisk to send 180 Ringing to an endpoint
-after 183 Session Progress has been send.
-If disabled Asterisk will instead send only a
-183 Session Progress to the endpoint.

doc/CHANGES-staging/chan_pjsip_flash.txt

-Subject: chan_pjsip
-
-Hook flash events can now be sent on a PJSIP channel
-if requested to do so.

doc/CHANGES-staging/chan_sip_pickup_AST_STATE_DOWN.txt

-Subject: chan_sip.c
-
-resolve issue with pickup on device that uses "183" and not "180"

doc/CHANGES-staging/chan_sip_session-timer_on_update.txt

-Subject: chan_sip
-
-Session timers get removed on UPDATE
-Fix if Asterisk receives a SIP REFER with Session-Timers UAC
-that Asterisk maintains Session-Timers when sending UPDATE request
-

doc/CHANGES-staging/channel_internal_api.txt

-Subject: channel_internal_api
-
-CHANNEL(lastcontext) and CHANNEL(lastexten)
-are now available for use in the dialplan.

doc/CHANGES-staging/cli_eval_function.txt

-Subject: cli
-
-A new CLI command 'dialplan eval function' has been
-added which allows users to test the behavior of
-dialplan function calls directly from the CLI.

doc/CHANGES-staging/cli_refresh.txt

-Subject: cli
-
-The "module refresh" command has been added,
-which allows unloading and then loading a
-module with a single command.