and what about
https://www.asterisk-blog.com/2016/02/17/odbc_gutting/

Dne 30.3.2016 v 0:12 Asterisk Development Team napsal(a):
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@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!





--
---------------------------------------
Marek Cervenka
=======================================

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