Asterisk 13.8.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 13.8.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.8.0-rc1 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 candidate: Release Notes – Asterisk – Version 13.8.0

Bug

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

Improvement

New Feature

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0-rc1

Thank you for your continued support of Asterisk!

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