Asterisk 13.8.0 Now Available

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

This release is available for immediate download at

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:


  • [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
  • [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


New Feature

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

Thank you for your continued support of Asterisk!

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