[asterisk-users] Asterisk 13.8.0 Now Available

Wed Mar 30 06:54:53 CDT 2016

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 DÃ¶msÃ¶di)
>   * 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 MouÄka)
>   * 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 EngstrÃ¶m)
>   * 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!
>
>
>


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

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