Publicada la versión Asterisk 13.8.0
El equipo de desarrolladores de Asterisk acaba de publicar la nueva versión Asterisk 13.8.0
Puedes ver la lista de cambios de esta versión en el siguiente enlace:
Descargar Changelog
También puedes esta versión en el siguiente enlace:
Descargar Asterisk
El anuncio oficial:
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 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 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!
