[asterisk-dev] Asterisk 14.6.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 14.6.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.6.0-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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
      events
      (Reported by Ove Aursand)
 * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP
      sockets.
      (Reported by Louis Jocelyn Paquet)
 * ASTERISK-27074 - core_local: local channel data not being
      properly unref'ed and unlocked
      (Reported by Kevin Harwell)
 * ASTERISK-27075 - bridge: stuck channel(s) after failed
      attended transfer
      (Reported by Kevin Harwell)
 * ASTERISK-27060 - Comment typo format_g729.c
      (Reported
      by Matthew Fredrickson)
 * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
      execution and application unregistration
      (Reported by
      Frederic LE FOLL)
 * ASTERISK-27026 - res_ari: Crash when no ari.conf
      configuration file exists
      (Reported by Ronald Raikes)
 * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
      sorcery.c
      (Reported by Ryan Smith)
 * ASTERISK-27024 - nat/external_media settings ignored in
      14.4.1
      (Reported by Christopher van de Sande)
 * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
      get_write_timeout
      (Reported by Jørgen H)
 * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
      RTCP component
      (Reported by Michael Walton)
 * ASTERISK-26923 - bridging: T.38 request is lost when channels
      are added to bridge
      (Reported by Torrey Searle)
 * ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
      during transfer
      (Reported by Kevin Harwell)
 * ASTERISK-27052 - Asterisk build process fails with flag
      --with-pjproject-bundled with curl download command and slow
      network
      (Reported by alex)
 * ASTERISK-27039 - chan_pjsip: Device state is idle when
      channel from endpoint is in early media
      (Reported by
      Joshua Colp)
 * ASTERISK-26996 - chan_pjsip: Flipping between codecs
     
      (Reported by Michael Maier)
 * ASTERISK-26281 - chan_pjsip would send INVITE to
      'Unreachable' endpoints
      (Reported by Jacek Konieczny)
 * ASTERISK-26973 - bridge: Crash when freeing frame and
      snooping
      (Reported by Michel R. Vaillancourt)
 * ASTERISK-19291 - Background in realtime
      (Reported by
      Andrew Nowrot)
 * ASTERISK-27025 - channel / meetme: Fix missing parentheses
  
      (Reported by Joshua Colp)
 * ASTERISK-27021 - GET /recordings/stored returns 500 Internal
      Server Error
      (Reported by Tim Morgan)
 * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets
      in wrong byte order on Intel platform when using slin codec
    
      (Reported by Frankie Chin)
 * ASTERISK-23951 -  Asterisk attempts and fails to build
      format_mp3 even if mp3lib was not downloaded
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-25294 - srtp's crypto_get_random deprecated
     
      (Reported by Tzafrir Cohen)
 * ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
      describe BEEP argument
      (Reported by Rusty Newton)
 * ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
      variable" command without args
      (Reported by Antoine
      Pitrou)
 * ASTERISK-25662 - Malformed AGI 520 Usage response
     
      (Reported by Tony Mountifield)
 * ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
      fmtp optional parameters have a space
      (Reported by John
      Harris)
 * ASTERISK-26399 - app_queue: Agent not called when caller is
      parked
      (Reported by wushumasters)
 * ASTERISK-26400 - app_queue: Queue member stops being called
      after AMI "Redirect" action for queues with wrapuptime
     
      (Reported by Etienne Lessard)
 * ASTERISK-26715 - app_queue: Member will not receive any new
      calls after doing a transfer if wrapuptime = greater than 0 and
      using Local channel
      (Reported by David Brillert)
 * ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
      agents not to receive queue calls after transfer queue call
    
      (Reported by Lorne Gaetz)
 * ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
      not play user name recording while leaving
      (Reported by
      Robert Mordec)
 * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
      authentication failure 10 or 110
      (Reported by Javier
      Riveros )
 * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
      completion failure/delay if client offers rtcp-mux as
      negotiable
      (Reported by Stefan Engström)
 * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
      when request and To URI differ
      (Reported by Yasin CANER)
 * ASTERISK-26789 - Audit manipulation of channel flags without
      locks
      (Reported by Joshua Colp)
 * ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
      6869i)
      (Reported by Matthias Binder)

Information Requests made in this release:
-----------------------------------
 * ASTERISK-26976 - libsrtp-2.x.x support
      (Reported by
      Alex)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
      block PJSIP taskprocessor on startup
      (Reported by Alexei
      Gradinari)
 * ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
      with LibreSSL
      (Reported by Guido Falsi)
 * ASTERISK-27042 - Unpatched asterisk sources fail to build on
      FreeBSD due to missing crypt.h file
      (Reported by Guido
      Falsi)
 * ASTERISK-26419 - audiohooks: Remove redundant codec
      translations when using audiohooks
      (Reported by Michael
      Walton)
 * ASTERISK-26124 - res_agi: Set audio format for EAGI audio
      stream
      (Reported by John Fawcett)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0-rc1

Thank you for your continued support of Asterisk!
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