[asterisk-announce] Asterisk 18.0.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Tue Oct 20 07:00:12 CDT 2020

The Asterisk Development Team would like to announce the release of Asterisk 18.0.0.
This release is available for immediate download at

The release of Asterisk 18.0.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
 * ASTERISK-28589 - chan_sip: Depending on configuration an
      INVITE can alter Addr of a peer
      (Reported by Andrey  V.
 * ASTERISK-28580 - Bypass SYSTEM write permission in manager
      action allows system commands execution
      (Reported by Eliel
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
      declined stream causes crash
      (Reported by Alexei

New Features made in this release:
 * ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits
      as non-root on Linux
      (Reported by Matt Addison)
 * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
      / "maxredirs" doesn't do anything
      (Reported by candrews)
 * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
      ability to match on source port
      (Reported by Sean Bright)
 * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
      PlayDTMF instead of only "sending"
      (Reported by lvl)
 * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type
      (Reported by Martin Tomec)
 * ASTERISK-28533 - func_jitterbuffer: Add support for video
      (Reported by Joshua C. Colp)
 * ASTERISK-17808 - [patch] Unregister a realtime moh class
      (Reported by Byron Clark)
 * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
      chan_pjsip to setup From header URI domain
      (Reported by
      Stas Kobzar)

Bugs fixed in this release:
 * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not
      progress calls due to codec negotiation after upgrading from
      Asterisk 16
      (Reported by Ross Beer)
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
      (Reported by Ove Aursand)
 * ASTERISK-29043 - app_queue: Leave empty sometimes not
      recorded as abandoned
      (Reported by Kfir Itzhak)
 * ASTERISK-29042 - res_parking: Parker UUID is no longer
      (Reported by Misha Vodsedalek)
 * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
      asterisk 16
      (Reported by Joseph Ades)
 * ASTERISK-29046 - pbx: Deadlock when doing a reload, while
      simultaneously doing an ExtensionState on a pattern match hint
      that ends up adding an extension
      (Reported by Ramarajan)
 * ASTERISK-29040 - res_speech: Assertion on format
      (Reported by Nickolay V. Shmyrev)
 * ASTERISK-29001 - chan_pjsip does not process or forward 181
      (Reported by Torrey Searle)
 * ASTERISK-29034 - Lastpause of realtime members is reseting
      (Reported by Evandro C��sar Arruda)
 * ASTERISK-27273 - app_voicemail: When a voicemail is marked as
      "Urgent", it is not sent by email/processed by the mailcmd
      (Reported by Leandro Dardini)
 * ASTERISK-29033 - res_pjsip_session: Aggressively terminates
      session on failed re-INVITE
      (Reported by Joshua C. Colp)
 * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
      appended RTP string to each message block.
      (Reported by
      Thomas Johnson)
 * ASTERISK-29011 - chan_sip: ToHost property not cleared on
      (Reported by Dennis)
      certified versions
      (Reported by cmaj)
 * ASTERISK-28927 - Asterisk crash in music on hold
      (Reported by David Cunningham)
 * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer
      triggered INVITE when NAT is active (UDP transport with
      (Reported by Michael Neuhauser)
 * ASTERISK-28995 - res_pjsip_registrar: Expires on statically
      configured contacts is not correct
      (Reported by tootai)
 * ASTERISK-28987 - BridgeCreated ARI event shows wrong
      video_mode info
      (Reported by sungtae kim)
 * ASTERISK-28978 - acl: named_acl rule misconfiguration results
      in segfault on reading rule from realtime
      (Reported by
      Andrew Yager)
 * ASTERISK-28975 - res_http_websocket: Text payload data
      doesn't necessary include trailing zero
      (Reported by
      Nickolay V. Shmyrev)
 * ASTERISK-28951 - Inconsistent behaviour queues.conf when
      there is (not) a [general] section
      (Reported by Walter
 * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
      contacts on AOR
      (Reported by Joshua C. Colp)
 * ASTERISK-28930 - ./configure --without-ssl build failure
      (Reported by Jaco Kroon)
 * ASTERISK-28957 - chan_sip: chan_sip does not process 400
      response to an INVITE.
      (Reported by Frederic LE FOLL)
 * ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in
      pjproject 2.7.2
      (Reported by Jared Smith)
 * ASTERISK-28888 - res_corosync: causes asterisk crash in huge
      distributed environment.
      (Reported by Universit�� di
      Bologna - CESIA VoIP)
 * ASTERISK-28954 - StreamEcho() only returns 1 active stream
      (Reported by Bill Kervaski)
 * ASTERISK-28955 - "setvar" doesn't work properly in
      (Reported by Marin Odrljin)
 * ASTERISK-28953 - res_pjsip_session: Preserve stream label
      (Reported by Joshua C. Colp)
 * ASTERISK-28942 - res_sorcery_memory_cache: Individual object
      expiration behaves unexpectedly with full backend caching
      (Reported by Joshua C. Colp)
 * ASTERISK-28950 - Stale code in app_queue to check untouched
      (Reported by Walter Doekes)
 * ASTERISK-28644 - Stale comment in app_queue about ring_entry
      (Reported by Walter Doekes)
 * ASTERISK-28952 - Queue wrapuptime sometimes not respected
      (based on stale lastcall time)
      (Reported by Walter Doekes)
 * ASTERISK-28938 - core_unreal / core_local: Add support for
      multistream and re-negotiation
      (Reported by Joshua C.
 * ASTERISK-28948 - ARI channel create doesn't referencing the
      channel_id parameter
      (Reported by sungtae kim)
 * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive
      buffers on non-WebRTC
      (Reported by Joshua C. Colp)
 * ASTERISK-28944 - bridge_softmix: Transitioning a stream from
      inactive -> sendrecv/sendonly doesn't re-negotiation
      (Reported by Joshua C. Colp)
 * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
      (Reported by Yury Kirsanov)
 * ASTERISK-28940 - /channels/create doesn't get any parameters
      from the body
      (Reported by sungtae kim)
 * ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
      (Reported by Walter Doekes)
 * ASTERISK-28900 - res_fax: Double frame free when gateway in
      use with off-nominal format usage
      (Reported by Gregory
 * ASTERISK-28929 - pjproject_bundled: Honor
      (Reported by Alexander Traud)
 * ASTERISK-28932 - res_pjsip_logger writing too big packets
      (Reported by nappsoft)
 * ASTERISK-28920 - bridge show all causes crash
      by sungtae kim)
 * ASTERISK-28921 - Wrong return value check for fwrite when
      writing to pcap file
      (Reported by nappsoft)
 * ASTERISK-28794 - res_pjsip: Crash when escaping during URI
      (Reported by nappsoft)
 * ASTERISK-28884 - x-ast-orig-host not filtered out from
      request URI and To header
      (Reported by nappsoft)
 * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on
      call answer
      (Reported by Alexei Gradinari)
 * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be
      wrong in SDP/SDES.
      (Reported by Alexander Traud)
 * ASTERISK-28898 - bridge_softmix: Conference bridge not
      passing silent rtp packets
      (Reported by Jonathan Hunter)
 * ASTERISK-28892 - res_musiconhold: Module res_musiconhold
      throws false warning
      (Reported by Nicholas John Koch)
 * ASTERISK-28904 - RTP ICE leaks the memory
      (Reported by
      sungtae kim)
 * ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when
      (Reported by Peter Sokolov)
 * ASTERISK-28854 - SIGSEGV when pjsip show history encounters
      IPV6 address
      (Reported by Roger James)
 * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format
      (Reported by Alexander Traud)
 * ASTERISK-28776 - Non async-signal-safe syscalls used after
      fork before exec
      (Reported by nappsoft)
 * ASTERISK-28870 - streams: One memory leak and one issue
      cloning streams
      (Reported by George Joseph)
 * ASTERISK-28829 - app_queue: leaking stasis subscription when
      Redirecting call 
      (Reported by lvl)
 * ASTERISK-25844 - app_queue: Ghost channels in "core show
      channels" output
      (Reported by Etienne Lessard)
 * ASTERISK-28859 - pjsip: Increase maximum candidate count
      (Reported by Joshua C. Colp)
 * ASTERISK-22920 - Crash while Forwarding from TLS extension
      with CHANNEL args secure_bridge_media and
      (Reported by Shlomi Gutman)
 * ASTERISK-28852 - Unprotected access to nochecksums variable,
      causes build failures
      (Reported by Guido Falsi)
 * ASTERISK-28848 - app_fax: Compile.
      (Reported by
      Alexander Traud)
 * ASTERISK-28846 - stream: Enforce formats immutability
      (Reported by Joshua C. Colp)
 * ASTERISK-28847 - ARI channels cuts the endpoint string over
      80 characters
      (Reported by sungtae kim)
 * ASTERISK-28811 - Crash occurs when fax session switches from
      T.38 to audio
      (Reported by Alexey Vasilyev)
 * ASTERISK-28839 - Sporadic crashes with Segmentation fault
      (Reported by Joeran Vinzens)
 * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted

      (Reported by Daniel Heckl)
 * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port
      (Reported by Anton Satskiy)
 * ASTERISK-24428 - Document that Asterisk will use the default
      SIP ports (5060 for TCP, 5061 for TLS) if the extern option
      variants aren't used
      (Reported by sstream)
      not mention
      (Reported by Alexander Traud)
 * ASTERISK-28841 - app_confbridge: Add support for disabling
      text messaging for a user
      (Reported by Joshua C. Colp)
 * ASTERISK-28837 - pjproject_bundled: Honor
      (Reported by Alexander Traud)
 * ASTERISK-28827 - res_rtp_asterisk: Loop when receive buffer
      is flushed by a received packet that is also in receive buffer
      with NACK
      (Reported by nappsoft)
 * ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket,
      ignoring TCP and TLS sockets
      (Reported by Joshua Roys)
 * ASTERISK-28826 - res_rtp_asterisk: Duplicate seqnos being
      added to send buffer with NACK
      (Reported by nappsoft)
 * ASTERISK-28812 - First DTMF is not get
      (Reported by
      Bernard Merindol)
 * ASTERISK-28758 - pjsip startup errors when using "with-ssl"
      configure option
      (Reported by Patrick Wakano)
 * ASTERISK-28824 - BuildSystem: Search for Python/C API when
      possibly needed only.
      (Reported by Alexander Traud)
 * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python
      Programming Language is python-2.7.
      (Reported by Alexander
 * ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not
      setup yet
      (Reported by Kevin Harwell)
 * ASTERISK-28819 - [patch] bridge_softmix_binaural: Show state
      in menuselect.
      (Reported by Alexander Traud)
 * ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and
      doc/pdf leftovers.
      (Reported by Alexander Traud)
 * ASTERISK-28818 - [patch] BuildSystem: Allow space in path.
      (Reported by Alexander Traud)
 * ASTERISK-28809 - [patch] res_rtp_asterisk: Avoid absolute
      value on unsigned subtraction.
      (Reported by Alexander
 * ASTERISK-28796 - func_channel: cannot read fields exten,
      context, userfield, channame from dialplan
      (Reported by
      S��bastien Duthil)
 * ASTERISK-28803 - [patch] chan_unistim: Avoid tautological
      warnings with clang.
      (Reported by Alexander Traud)
 * ASTERISK-28808 - [patch] test_stasis: Avoid always true
      warning with clang.
      (Reported by Alexander Traud)
 * ASTERISK-28056 - res_pjsip: Incorrect endpoint status after
      endpoint synchronization for a specific AOR
      (Reported by
      Jason Hord)
 * ASTERISK-28795 - channel: write to a stream on multi-frame
      (Reported by Kevin Harwell)
 * ASTERISK-28789 - test_utils: incorrectly printing error
      'declined to load'
      (Reported by Alexander Traud)
 * ASTERISK-28788 - func_aes: incorrectly printing error
      'declined to load'
      (Reported by Alexander Traud)
 * ASTERISK-28790 - Crash during conference call using
      confbridge and video
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd
      unless asterisk is running as root
      (Reported by Jaco
 * ASTERISK-21205 - [patch] dundi_read_result crash due to
      negative number
      (Reported by Jaco Kroon)
 * ASTERISK-28784 - res_pjsip_sdp_rtp: Only do hold/unhold on
      first audio stream
      (Reported by Joshua C. Colp)
 * ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP

      (Reported by sungtae kim)
 * ASTERISK-28783 - res_pjsip_session: Allow default non-audio
      streams to have reflected state
      (Reported by Joshua C.
 * ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously
      triggered during direct-media (native_rtp) bridge
      (Reported by Michael Neuhauser)
 * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample
      are not consistent with examples. Missing examples.
      (Reported by Olivier Krief)
 * ASTERISK-28780 - app_mixmonitor: Memory leak due to race
      condition between AMI MixMonitor and hangup
      (Reported by
      Joshua C. Colp)
 * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp
      bridge is active
      (Reported by Torrey Searle)
 * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support
      is enabled but not used
      (Reported by Torrey Searle)
 * ASTERISK-28759 - A non negotiated rtp frame causes call
      disconnection when there is a SSRC change
      (Reported by
      Paulo Vicentini)
 * ASTERISK-26711 - func_enum: ENUM code wrong case
      (Reported by Vitold)
 * ASTERISK-23407 - Fix the FSF address in the headers of lots
      of pjproject files
      (Reported by Jared Smith)
 * ASTERISK-19460 - [patch] Function TXTCIDNAME never actually
      makes DNS calls and always returns an empty string
      (Reported by George Joseph)
 * ASTERISK-28766 - PJSIP blind transfer not completed after
      using Proceeding()
      (Reported by lvl)
 * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and
      seqno handling
      (Reported by Joshua C. Colp)
 * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in
      the "variables" field
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28685 - check_expr2: linking (when hardening) and
      cross-compiling troubles
      (Reported by Sebastian Kemper)
 * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After
      (Reported by Ross Beer)
 * ASTERISK-28697 - res_pjsip: Named ACL does not update on
      reload if changed
      (Reported by Timothy Vanderaerden)
 * ASTERISK-28746 - res_pjsip_outbound_registration keeps
      retrying the first entry in a SRV record set
      (Reported by
      George Joseph)
 * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to
      complete before allowing sending
      (Reported by Benjamin
      Keith Ford)
 * ASTERISK-28738 - Incorrect state machine used when
      MOH_PASSTHRU is used
      (Reported by Torrey Searle)
 * ASTERISK-28742 - res_rtp_asterisk: static for audio due to
      incomplete dtls/srtp setup
      (Reported by Kevin Harwell)
 * ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting
      to SLIN
      (Reported by Ross Beer)
 * ASTERISK-28730 - res_pjsip_session: Fix out of order session
      (Reported by Joshua C. Colp)
 * ASTERISK-26955 - pjsip: SIP Packets with Via "received="
      Containing IPv6 Address Delimited by "[]" Rejected
      (Reported by Peter Sokolov)
 * ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are
      depleted, should return 503
      (Reported by Walter Doekes)
 * ASTERISK-28713 - res_stasis_playback: Error building JSON
      (Reported by S��bastien Duthil)
 * ASTERISK-28714 - REGRESSION: Feature
      subscription_persistence_recreate (ASTERISK-27759) Causes
      (Reported by Ross Beer)
 * ASTERISK-26082 - res_pjsip_messaging: MessageSend
      Content-Type can't be changed
      (Reported by Alex)
 * ASTERISK-28423 - ARI causes STASIS Deadlock
      by Ross Beer)
 * ASTERISK-28679 - stasis application is destroyed after its
      (Reported by Francois Blackburn)
      spite of the error when sending
      (Reported by Dmitriy
 * ASTERISK-28686 - chan_sip strictrtp=yes fails when media
      source is changed: no audio
      (Reported by Walter Doekes)
 * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes
      Asterisk To Drop Calls
      (Reported by Paul Brooks)
 * ASTERISK-28677 - CDR billsec is always 0 for transferred
      (Reported by Maciej Michno)
 * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
      dialtone times out after 0:16:40
      (Reported by Andrew
 * ASTERISK-24484 - Update documentation for statsd module -
      usage requirements unclear
      (Reported by Dan Jenkins)
 * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
      translation' output
      (Reported by Sean Bright)
 * ASTERISK-28695 - core: minmemfree watermark uses free RAM,
      not available RAM
      (Reported by Kevin Flyn)
 * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
      whitespace appears empty in the dialplan
      (Reported by
      Frank Matano)
 * ASTERISK-23739 - [patch]Segfault forwarding voicemail with
      ODBC storage enabled and realtime voicemail_data is used
      (Reported by Stas Kobzar)
 * ASTERISK-27622 - empty voicemail.conf required for ARA
      (realtime) voicemail to leave message
      (Reported by Jim Van
 * ASTERISK-21794 - CLI command 'realtime update2' syntax
      failure when using according to usage help
      (Reported by
      Cedric BASSAGET)
 * ASTERISK-28349 - Pause reason not reported in QueueMember AMI
      (Reported by Niksa Baldun)
 * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
      support for hostnames
      (Reported by Joshua C. Colp)
 * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
      be present instead of just one
      (Reported by
 * ASTERISK-28682 - app_record: Lack of `beep` audio file causes
      application to return error and hangup
      (Reported by Corey
 * ASTERISK-28507 - Wiki docs missing for MessageWaiting
      (Reported by David M. Lee)
 * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
      does not preserve XML <dialog-info> version number
      (Reported by Bryan Nelson)
 * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
      with concurrent command pri show span X
      (Reported by Dirk
 * ASTERISK-28633 - stasis bridge topic leak
      (Reported by
      Joeran Vinzens)
 * ASTERISK-28492 - pjsip reload not reloading wizard
      endpoint/pickup_group endpoint/call_group
      (Reported by
      Jean-Denis Girard)
 * ASTERISK-28562 - SIP WSS message not processed until next
      frame arrives
      (Reported by Robert Sutton)
 * ASTERISK-28667 - Asterisk ignores parsing of config files if
      a Byte order mark is present
      (Reported by Robin Leffmann)
 * ASTERISK-28625 - Playback of local files impacted by large
      media cache
      (Reported by Kevin Reeves)
 * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
      it's supposed to due to invalid syntax
      (Reported by
      Richard Kenner)
 * ASTERISK-28664 - "trustrpid" is misspelled in
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
      fails to deactivate CDR.
      (Reported by Frederic LE FOLL)
 * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
      build on 17.0.0
      (Reported by George Joseph)
 * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
      non-existent media stream if codecs create additional streams
      and offer does not have them
      (Reported by nappsoft)
 * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
      with config option
      (Reported by Kevin Harwell)
 * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c
      (Reported by Ted G)
 * ASTERISK-28651 - chan_sip logs errors on tx to non-existent
      TCP connections
      (Reported by Jaco Kroon)
 * ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER
      200 Response Contact
      (Reported by Ross Beer)
 * ASTERISK-28641 - res_pjsip Segfaults when realtime
      configuration to an AOR points to a not existent AOR
      (Reported by Ross Beer)
 * ASTERISK-28647 - chan_sip: RTP frames not transmitted after
      emitting a COLP
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
      compatibility check failure when negociated ptime is not default
      (Reported by Frederic LE FOLL)
 * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
      UTF-8 string on hangup when TEST_FRAMEWORK enabled
      (Reported by Bernhard Schmidt)
 * ASTERISK-28631 - res_parking: Doesn't park when parkee and
      parker are the same
      (Reported by Ross Beer)
 * ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok
      (Reported by Salah Ahmed)
 * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV
      (Reported by Kevin Harwell)
 * ASTERISK-28608 - app_amd: Use time calculation to calculate
      (Reported by Michael Cargile)
 * ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down,
      Active" after a short alarm
      (Reported by Frederic LE FOLL)
 * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when
      sent packet length doesn't match
      (Reported by Joshua
 * ASTERISK-26481 - FILE function grabs garbage along with read
      data when target line has no newline
      (Reported by Jonathan
 * ASTERISK-28618 - bridge_softmix: hold not cleared when
      joining a softmix bridge
      (Reported by Kevin Harwell)
 * ASTERISK-28616 - parking: Deadlock when multi call parking
      (Reported by Joshua C. Colp)
 * ASTERISK-28572 - Memory leaks in res_calendar_exchange and
      (Reported by Yoooooo Ha)
 * ASTERISK-28585 - ari/resource_events: Crash in event session
      (Reported by Kevin Harwell)
 * ASTERISK-28590 - utils.c throws repeated warnings;
      "pthread_attr_setstacksize: Invalid argument"
      (Reported by
      Speed Dial Dave)
 * ASTERISK-28578 - race condition on pjsip channelstats
      (Reported by Salah Ahmed)
 * ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
      removed) column
      (Reported by Christoph Moench-Tegeder)
 * ASTERISK-28575 - MWI Send Notify Crash on 16.6
      (Reported by Joshua Elson)
 * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
      (Reported by Niklas Larsson)
 * ASTERISK-28561 - Asterisk Deadlocks
      (Reported by
 * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF
      over AMI
      (Reported by Jeremiah Gadd)
 * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
      unsolicited_mwi container
      (Reported by Kevin Harwell)
 * ASTERISK-28566 - CDR backend unload problem during active
      (Reported by Marian Piater)
 * ASTERISK-28553 - stasis.c: Crash during unload
      (Reported by Kevin Harwell)
 * ASTERISK-28544 - Wrong contact representation in ipv6 mode
      (Reported by J��rgen H)
 * ASTERISK-28534 - Segmentation fault when there is no priority
      for an extension
      (Reported by Timothy Vanderaerden)
 * ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
      is configured
      (Reported by Juan Martin)
 * ASTERISK-28521 - pjsip: Memory Leak
      (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
      by Cyril Rami��re)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
      (Reported by Guido Falsi)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
      (Reported by Joshua C. Colp)
 * ASTERISK-28497 - func_odbc: truncating Unicode string on
      (Reported by Boris P. Korzun)
 * ASTERISK-23756 - setvar directive when used in template and a
      child of said template, results in duplicate variable names
      (Reported by Michael Goryainov)
 * ASTERISK-28527 - ChanIsAvail() creates a CDR if
      unanswered=yes is set in cdr.conf
      (Reported by Frederic LE
 * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
      PRI channel hangs up
      (Reported by Frederic LE FOLL)
 * ASTERISK-28511 - codec_resample: Bad sound quality when up
      sampling from SLIN16 to SLIN32
      (Reported by Ruddy G)
 * ASTERISK-28499 - translate: Crash when frame does not have a
      "src" field set
      (Reported by Gregory Massel)
 * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
      type not at end of a struct
      (Reported by Alexander Traud)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
      (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
      characters, NEC only supports up to 32 characters
      (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
      leave_voicemail because not checking mailstream
      by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
      by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
      srtp->session to srtp_protect/unprotect causing SEGV
      (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect
      (Reported by Joshua C. Colp)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

      (Reported by Salah Ahmed)
 * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
      (Reported by Ian Jones)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
      (Reported by Torrey Searle)

Improvements made in this release:
 * ASTERISK-28959 - res_pjsip: Added option for disable rport
      parameter set
      (Reported by sungtae kim)
 * ASTERISK-28958 - Continue reading string when ping received
      by websocket
      (Reported by Nickolay V. Shmyrev)
 * ASTERISK-28945 - AMI SendText - add Content-Type parameter
      (Reported by Kevin Harwell)
 * ASTERISK-28949 - res_http_websocket: Add masking to websocket
      (Reported by Moises Silva)
 * ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
      (Reported by Kevin Harwell)
 * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality
      (Reported by Joshua C. Colp)
 * ASTERISK-28896 - ari: Add support for specifying variables on
      channel create
      (Reported by Joshua C. Colp)
 * ASTERISK-28879 - pjproject has race conditions in it's build
      (Reported by Guido Falsi)
 * ASTERISK-28866 - third-party/pjproject/configure.m4 contains
      (Reported by Guido Falsi)
 * ASTERISK-28853 - Missing include on FreeBSD
      by Guido Falsi)
 * ASTERISK-28832 - chan_mobile creates PCMA streams that make
      some VoIP clients crash or not render received audio
      (Reported by Peter Turczak)
 * ASTERISK-28813 - func_volume: Allow decimal numbers as
      parameter to improve granularity
      (Reported by Jean Aunis -
 * ASTERISK-28777 - Codec Negotiation: add
      outgoing_call_offer_prefs option
      (Reported by Kevin
 * ASTERISK-27946 - dial (API): Storage of dialed target uses
      AST_MAX_EXTENSION when it shouldn't
      (Reported by Joshua
 * ASTERISK-28782 - Add support for Content-Disposition header
      in multi-part INVITES
      (Reported by Torrey Searle)
 * ASTERISK-28787 - res_pjsip_session: Decide more intelligently
      when to add video
      (Reported by Joshua C. Colp)
 * ASTERISK-28756 - Codec Negotiation: add
      incoming_call_offer_pref option
      (Reported by Kevin
 * ASTERISK-28750 - TLS/SSL Key too small error
      by Martin Zeh)
 * ASTERISK-28733 - stream: Add support for adding/removing
      streams during SFU/calls
      (Reported by Joshua C. Colp)
 * ASTERISK-24798 - Documentation - Clarify That Format Is Set
      By File Name Extension In MixMonitor
      (Reported by xrobau)
 * ASTERISK-28726 - install_prereq script uses the interactive
      mode when installing aptitude
      (Reported by Sylvain
 * ASTERISK-28710 - Should be able to disable the /httpstatus
      URI in the built-in HTTP server
      (Reported by Sean Bright)
 * ASTERISK-28484 - Add AudioSocket support
      (Reported by
      Se��n C. McCord)
 * ASTERISK-28638 - Simplify dialplan for Dial, Page, and
      (Reported by cmaj)
 * ASTERISK-28673 - GET FULL VARIABLE documentation
      (Reported by Jonathan Harris)
 * ASTERISK-28629 - [patch] Add an "inhibitCOLP" flag to the
      bridges REST API
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28658 - app_confbridge: Add support for setting
      maximum sample rate
      (Reported by Joshua C. Colp)
 * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
      retries reached
      (Reported by Daniel)
 * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
      (Reported by Sam Banks)
 * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
      backend when format differs from attachfmt column
      (Reported by cmaj)
 * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should
      clear out any .lock files in the voice mail directory on
      (Reported by Michael)
 * ASTERISK-28542 - [patch] add the ability for asterisk to
      generate on-hold re-invites
      (Reported by Torrey Searle)
 * ASTERISK-28512 - Add pass-through support for H.265 (HEVC)
      (Reported by Florian Floimair)

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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