[asterisk-dev] Asterisk 16.19.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.19.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.19.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

New Features made in this release:
-----------------------------------
 * ASTERISK-29446 - app_confbridge: New ConfKick application
   
      (Reported by N A)
 * ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
      be suppressed
      (Reported by N A)
 * ASTERISK-29431 - Minimum and maximum dialplan functions
     
      (Reported by N A)
 * ASTERISK-29439 - func_volume: Volume function can't be read
 
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
      up during application execution
      (Reported by N A)
 * ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
      domain name
      (Reported by George Joseph)
 * ASTERISK-29441 - Core reload making TCP endpoints go offline

      (Reported by Luke Escude)
 * ASTERISK-29433 - res_rtp_asterisk: Server reflexive
      candidates use incorrect raddr for RTCP
      (Reported by
      Chris)
 * ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
      happens when unsubscribe an application from an event source
   
      (Reported by Lucas Tardioli Silveira)
 * ASTERISK-28393 - Multidomain support issue
      (Reported by
      Andrea Sannucci)
 * ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760
      UASs
      (Reported by George Joseph)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
      in PJSIP NOTIFY event: dialog  XML body
      (Reported by Marco
      Paland)
 * ASTERISK-29372 - file.c switch does not account for flash
      events
      (Reported by N A)
 * ASTERISK-29377 - cpool_release_pool "double free or
      corruption (out)"
      (Reported by Robert Sutton)
 * ASTERISK-29370 - chan_sip does not recognize
      application/hook-flash
      (Reported by N A)
 * ASTERISK-29358 - chan_pjsip: Trace message for progress is
      output even if frame is not queued
      (Reported by Michael
      Maier)
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
      wrong SSRC) gets inserted when switching from progress to
      established
      (Reported by Matthias Hensler)
 * ASTERISK-29407 - chan_local: Filtering audio formats should
      not occur on removed streams
      (Reported by Joshua C. Colp)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29450 - Allow setting channel variables using
      Originate application
      (Reported by N A)
 * ASTERISK-29460 - Recognize application/hook-flash in PJSIP
  
      (Reported by N A)
 * ASTERISK-29459 - Missing configuration from PJSIP to SIP
      conversion script
      (Reported by N A)
 * ASTERISK-29434 - Asterisk reveals pjproject version in STUN
      packets
      (Reported by Jeremy Lain��)
 * ASTERISK-29349 - Silent voicemail option is not completely
      silent
      (Reported by N A)
 * ASTERISK-29380 - Add Flash AMI event to handle flash events
 
      (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0-rc1

Thank you for your continued support of Asterisk!
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