[asterisk-announce] Asterisk 13.9.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.9.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.9.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25963 - func_odbc requires reconnect checks for stale
      connections (Reported by Ross Beer)
 * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
      Dmitriy Serov)
 * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
      LAST_INSERT_ID() always returns zero. (Reported by Edwin
      Vandamme)
 * ASTERISK-25927 - Removed option "registertrying" is still
      documented in sip.conf.sample (Reported by Etienne Lessard)
 * ASTERISK-25947 - Protocol transfers to stasis applications are
      missing the StasisStart with the replace_channel object.
      (Reported by Richard Mudgett)
 * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
      fails to get app name (Reported by John Bigelow)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
      ConnectedLine information (Reported by George Joseph)
 * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
      thread (Reported by Joshua Colp)
 * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
      not raised (Reported by Joshua Colp)
 * ASTERISK-25934 - chan_sip should not require sipregs or
      updateable sippeers table unless rt (Reported by Jaco Kroon)
 * ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
      of app_queue.c (Reported by Sébastien Couture)
 * ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
      exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
      Joseph)
 * ASTERISK-25707 - Long contact URIs or hostnames can crash
      pjproject/Asterisk under certain conditions (Reported by George
      Joseph)
 * ASTERISK-25123 - Bracketed IPv6 Contact header parameter
      unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
 * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
 * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
      without adding them to the local hangupcauses via
      ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
 * ASTERISK-25885 - res_pjsip: Race condition between adding
      contact and automatic expiration (Reported by Joshua Colp)
 * ASTERISK-25910 - pjproject:  Via headers are not parsed when
      "received" contains an IPv6 address (Reported by George Joseph)
 * ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
      (Reported by Harley Peters)
 * ASTERISK-25894 - [patch] webrtc video broken due to missing
      marker bits in RTP streams (Reported by Jacek Konieczny)
 * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
      a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
 * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
      cannot find -lasteriskpj (Reported by Hans van Eijsden)
 * ASTERISK-25882 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Part 2) (Reported by
      Richard Mudgett)
 * ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
      Jacek Konieczny)
 * ASTERISK-24605 - res_parking option parkeddynamic does not work
      with the core Features 'parkcall' (DTMF initiated parking)
      (Reported by Philip Correia)
 * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
      (Reported by Ross Beer)
 * ASTERISK-24596 - Unclear how to use Park application with
      res_parking 'parkeddynamic' enabled. Documentation? (Reported by
      Philip Correia)
 * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
      possible codecs configured for peer as opposed to intersection
      of configured codecs and offered codecs (Reported by Taylor
      Hawkes)
 * ASTERISK-25825 - Crashes during shutdown when running CLI
      commands (Reported by Mark Michelson)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25510 - [patch]Log to syslog failing (Reported by
      Michael Newton)
 * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
      data corruption (Reported by Gianluca Merlo)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25865 - Message-Account Missing From PJSIP MWI
      (Reported by Ross Beer)
 * ASTERISK-25444 - [patch]Music On Hold Warning misleading
      (Reported by Conrad de Wet)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0

Thank you for your continued support of Asterisk!