[asterisk-users] Asterisk 13.8.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.8.0 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:

New Features made in this release:
-----------------------------------
 * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
      contents to file (Reported by Ray Crumrine)
 * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
      Journo)
 * ASTERISK-25480 - [patch]Add field PauseReason on
      QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25849 - chan_pjsip: transfers with direct media
      sometimes drops audio (Reported by Kevin Harwell)
 * ASTERISK-25113 - install_prereq in Debian 8 without "standard
      system utilities" (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
      (Reported by Sergio Medina Toledo)
 * ASTERISK-25023 - Deadlock in chan_sip in
      update_provisional_keepalive (Reported by Arnd Schmitter)
 * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
      channel (Reported by Filip Frank)
 * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
      separating multiple AORs (Reported by Mateusz Kowalski)
 * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
      Stasis application. (Reported by Javier Riveros )
 * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
      Bright)
 * ASTERISK-25582 - Testsuite: Reactor timeout error in
      tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
      Jordan)
 * ASTERISK-25811 - Unable to delete object from sorcery cache
      (Reported by Ross Beer)
 * ASTERISK-25800 - [patch] Calculate talktime when is first call
      answered (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
      PJSIP requirement (Reported by Gergely Dömsödi)
 * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
      when calling from Gosub (Reported by Jacques Peacock)
 * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
      OutboundSubscriptionDetail ami action (Reported by Kevin
      Harwell)
 * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
      heap-use-after-free (Reported by Badalian Vyacheslav)
 * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
      returns garbage (Reported by Etienne Lessard)
 * ASTERISK-25751 - res_pjsip: Support
      pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
 * ASTERISK-25606 - Core dump when using transports in sorcery
      (Reported by Martin Moučka)
 * ASTERISK-20987 - non-admin users, who join muted conference are
      not being muted (Reported by hristo)
 * ASTERISK-25737 - res_pjsip_outbound_registration: line option
      not in Alembic (Reported by Joshua Colp)
 * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
      udptl_rx_packet cause ast_frdup crash (Reported by Walter
      Doekes)
 * ASTERISK-25742 - Secondary IFP Packets can result in accessing
      uninitialized pointers and a crash (Reported by Torrey Searle)
 * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
      Vulnerability - Investigate vulnerability of HTTP server
      (Reported by Alex A. Welzl)
 * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
      non-default timert1 (Reported by Alexander Traud)
 * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
      upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
      Nic Colledge)
 * ASTERISK-25730 - build:  make uninstall after make distclean
      tries to remove root (Reported by George Joseph)
 * ASTERISK-25725 - core: Incorrect XML documentation may result in
      weird behavior (Reported by Joshua Colp)
 * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
      sip_sipredirect (Reported by Badalian Vyacheslav)
 * ASTERISK-25709 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Reported by Mark
      Michelson)
 * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
      by Badalian Vyacheslav)
 * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
      script (Reported by Joshua Colp)
 * ASTERISK-25712 - Second call to already-on-call phone and
      Asterisk sends "Ready" (Reported by Richard Mudgett)
 * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
      incorrect values (Reported by Gianluca Merlo)
 * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
      test sporadically failing (Reported by Joshua Colp)
 * ASTERISK-24097 - Documentation - CHANNEL function help text
      missing 'linkedid' argument (Reported by Steven T. Wheeler)
 * ASTERISK-25700 - main/config: Clean config maps on shutdown.
      (Reported by Corey Farrell)
 * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
      a transfer (Reported by Kevin Harwell)
 * ASTERISK-25697 - bridge_basic: don't play an attended transfer
      fail sound after target hangs up (Reported by Kevin Harwell)
 * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
      with MALLOC_DEBUG  (Reported by yaron nahum)
 * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
      schema is an integer (Reported by Marcelo Terres)
 * ASTERISK-25690 - Hanging up when executing connected line sub
      does not cause hangup (Reported by Joshua Colp)
 * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
      reload' cause a crash (Reported by Sean Bright)
 * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
      address when multihomed (Reported by Olivier Krief)
 * ASTERISK-25637 - Multi homed server using wrong IP (Reported by
      Daniel Journo)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25640 - pbx: Deadlock on features reload and state
      change hint. (Reported by Krzysztof Trempala)
 * ASTERISK-25681 - devicestate: Engine thread is not shut down
      (Reported by Corey Farrell)
 * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
      shutdown (Reported by Corey Farrell)
 * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
      Corey Farrell)
 * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
      Daniel Journo)
 * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
      by Corey Farrell)
 * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
      Farrell)
 * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
      Mark Michelson)
 * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
      (Reported by Corey Farrell)
 * ASTERISK-25647 - bug of cel_radius.c: wrong point of
      ADD_VENDOR_CODE (Reported by Aaron An)
 * ASTERISK-25317 - asterisk sends too many stun requests (Reported
      by Stefan Engström)
 * ASTERISK-25137 - endpoint stasis messages are delivered twice
      (Reported by Vitezslav Novy)
 * ASTERISK-25116 - res_pjsip:  Two PeerStatus AMI messages are
      sent for every status change (Reported by George Joseph)
 * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
      transfer initiated channel (Reported by Dmitry Melekhov)
 * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
      Brandon)
 * ASTERISK-25442 - using realtime (mysql) queue members are never
      updated in wait_our_turn function (app_queue.c)  (Reported by
      Carlos Oliva)
 * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
      caching (Reported by Joshua Colp)
 * ASTERISK-25601 - json: Audit reference usage and thread safety
      (Reported by Joshua Colp)
 * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
      sungtae kim)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25495 - [patch] Prevent old-update packages on
      repository Debian systems (Reported by Rodrigo Ramirez
      Norambuena)
 * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
      (Reported by Andrew Nagy)
 * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
      Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
      Messina)
 * ASTERISK-24813 - asterisk.c: #if statement in listener()
      confuses code folding editors (Reported by Corey Farrell)
 * ASTERISK-25767 - [patch] Add check to configure for sanitizes 
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
      core set (Reported by Rusty Newton)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0

Thank you for your continued support of Asterisk!