[asterisk-users] Asterisk 18.5.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 18.5.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.5.0 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-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-29433 - res_rtp_asterisk: Server reflexive
      candidates use incorrect raddr for RTCP
      (Reported by
      Chris)
 * 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-29370 - chan_sip does not recognize
      application/hook-flash
      (Reported by N A)
 * ASTERISK-29377 - cpool_release_pool "double free or
      corruption (out)"
      (Reported by Robert Sutton)
 * ASTERISK-29372 - file.c switch does not account for flash
      events
      (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-29407 - chan_local: Filtering audio formats should
      not occur on removed streams
      (Reported by Joshua C. Colp)
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
      wrong SSRC) gets inserted when switching from progress to
      established
      (Reported by Matthias Hensler)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29450 - Allow setting channel variables using
      Originate application
      (Reported by N A)
 * ASTERISK-29459 - Missing configuration from PJSIP to SIP
      conversion script
      (Reported by N A)
 * ASTERISK-29460 - Recognize application/hook-flash in PJSIP
  
      (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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0

Thank you for your continued support of Asterisk!
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