[asterisk-users] Asterisk 16.19.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Jun 24 08:03:37 CDT 2021

The Asterisk Development Team would like to announce the release of Asterisk 16.19.0.
This release is available for immediate download at

The release of Asterisk 16.19.0 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:

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
 * 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
      (Reported by George Joseph)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
      in PJSIP NOTIFY event: dialog  XML body
      (Reported by Marco
 * ASTERISK-29372 - file.c switch does not account for flash
      (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
      (Reported by N A)
 * ASTERISK-29358 - chan_pjsip: Trace message for progress is
      output even if frame is not queued
      (Reported by Michael
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
      wrong SSRC) gets inserted when switching from progress to
      (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
      (Reported by Jeremy Lain��)
 * ASTERISK-29349 - Silent voicemail option is not completely
      (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, please see the ChangeLog:

Thank you for your continued support of Asterisk!
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