[asterisk-announce] Asterisk 13.17.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 13.17.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.17.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27108 - Crash using 'data get' CLI command
     
      (Reported by Sean Bright)
 * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add
      only really different domain with TLS.
      (Reported by
      Alexander Traud)
 * ASTERISK-27100 - channel: ast_waitfordigit_full fails to
      clear flag in an error branch.
      (Reported by Corey Farrell)
 * ASTERISK-27090 - PJSIP: Deadlock using TCP transport
     
      (Reported by Richard Mudgett)
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
      events
      (Reported by Ove Aursand)
 * ASTERISK-27065 - call hangup after leaving app_queue
     
      (Reported by Marek Cervenka)
 * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()

      (Reported by Ross Beer)
 * 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-24052 - app_voicemail reloads result in leaked IMAP
      sockets.
      (Reported by Louis Jocelyn Paquet)
 * ASTERISK-27060 - Comment typo format_g729.c
      (Reported
      by Matthew Fredrickson)
 * ASTERISK-27026 - res_ari: Crash when no ari.conf
      configuration file exists
      (Reported by Ronald Raikes)
 * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
      execution and application unregistration
      (Reported by
      Frederic LE FOLL)
 * 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-25101 - DTLS configuration can not be specified in
      the general section - documentation
      (Reported by Ben
      Langfeld)
 * 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)

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-26976 - libsrtp-2.x.x support
      (Reported by
      Alex)
 * ASTERISK-26124 - res_agi: Set audio format for EAGI audio
      stream
      (Reported by John Fawcett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0

Thank you for your continued support of Asterisk!
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