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 – 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 – 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 – 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 – res_phoneprov: memory leak and

     heap-use-after-free (Reported by Badalian Vyacheslav)

* ASTERISK-25272 – 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 – 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 – 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 – 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 – 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!

Leave a Reply

Your email address will not be published. Required fields are marked *

two × 2 =