[asterisk-dev] asterisk release 21.0.1

[Asterisk Development Team]

The Asterisk Development Team would like to announce security release  
Asterisk 21.0.1.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/21.0.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files](https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ](https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)


Change Log for Release asterisk-21.0.1
========================================

Links:
----------------------------------------

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)  
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)  
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:
----------------------------------------

- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.

User Notes:
----------------------------------------

- ### http.c: Minor simplification to HTTP status output.
  For bound addresses, the HTTP status page now combines the bound
  address and bound port in a single line. Additionally, the SSL bind
  address has been renamed to TLS.


Upgrade Notes:
----------------------------------------

- ### chan_sip: Remove deprecated module.
  This module was deprecated in Asterisk 17
  and is now being removed in accordance with
  the Asterisk Module Deprecation policy.

- ### res_monitor: Remove deprecated module.
  This module was deprecated in Asterisk 16
  and is now being removed in accordance with
  the Asterisk Module Deprecation policy.
  This also removes the 'w' and 'W' options
  for app_queue.
  MixMonitor should be default and only option
  for all settings that previously used either
  Monitor or MixMonitor.

- ### app_osplookup: Remove deprecated module.
  This module was deprecated in Asterisk 19
  and is now being removed in accordance with
  the Asterisk Module Deprecation policy.

- ### app_cdr: Remove deprecated application and option.
  The previously deprecated NoCDR application has been removed.
  Additionally, the previously deprecated 'e' option to the ResetCDR
  application has been removed.

- ### chan_skinny: Remove deprecated module.
  This module was deprecated in Asterisk 19
  and is now being removed in accordance with
  the Asterisk Module Deprecation policy.

- ### chan_mgcp: Remove deprecated module.
  This module was deprecated in Asterisk 19
  and is now being removed in accordance with
  the Asterisk Module Deprecation policy.

- ### translate.c: Prefer better codecs upon translate ties.
  When setting up translation between two codecs the quality was not taken into account,
  resulting in suboptimal translation. The quality is now taken into account,
  which can reduce the number of translation steps required, and improve the resulting quality.

- ### app_macro: Remove deprecated module.
  This module was deprecated in Asterisk 16
  and is now being removed in accordance with
  the Asterisk Module Deprecation policy.
  For most modules that interacted with app_macro,
  this change is limited to no longer looking for
  the current context from the macrocontext when set.
  The following modules have additional impacts:
  app_dial - no longer supports M^ connected/redirecting macro
  app_minivm - samples written using macro will no longer work.
  The sample needs to be re-written
  app_queue - can no longer call a macro on the called party's
  channel.  Use gosub which is currently supported
  ccss - no callback macro, gosub only
  app_voicemail - no macro support
  channel  - remove macrocontext and priority, no connected
  line or redirection macro options
  options - stdexten is deprecated to gosub as the default
  and only options
  pbx - removed macrolock
  pbx_dundi - no longer look for macro
  snmp - removed macro context, exten, and priority

- ### chan_alsa: Remove deprecated module.
  This module was deprecated in Asterisk 19
  and is now being removed in accordance with
  the Asterisk Module Deprecation policy.

- ### pbx_builtins: Remove deprecated and defunct functionality.
  The previously deprecated ImportVar and SetAMAFlags
  applications have now been removed.


Closed Issues:
----------------------------------------

None