[asterisk-users] Certified Asterisk 16.8-cert1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Apr 30 09:02:28 CDT 2020

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1.
This release is available for immediate download at

The release of Certified Asterisk 16.8-cert1 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
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
      no body causes crash
      (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
      (Reported by Francesco Castellano)

New Features made in this release:
 * 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)
 * ASTERISK-28375 - res_pjsip: New configuration setting to
      allow disabling norefersub
      (Reported by Dan Cropp)
 * ASTERISK-28320 - Added ARI resource
      (Reported by
      sungtae kim)

Bugs fixed in this release:
 * 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-28826 - res_rtp_asterisk: Duplicate seqnos being
      added to send buffer with NACK
      (Reported by nappsoft)
 * ASTERISK-28795 - channel: write to a stream on multi-frame
      (Reported by Kevin Harwell)
 * ASTERISK-28790 - Crash during conference call using
      confbridge and video
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28783 - res_pjsip_session: Allow default non-audio
      streams to have reflected state
      (Reported by Joshua C.
 * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and
      seqno handling
      (Reported by Joshua C. Colp)
 * ASTERISK-28730 - res_pjsip_session: Fix out of order session
      (Reported by Joshua C. Colp)
 * ASTERISK-28746 - res_pjsip_outbound_registration keeps
      retrying the first entry in a SRV record set
      (Reported by
      George Joseph)
 * ASTERISK-28742 - res_rtp_asterisk: static for audio due to
      incomplete dtls/srtp setup
      (Reported by Kevin Harwell)
 * ASTERISK-28679 - stasis application is destroyed after its
      (Reported by Francois Blackburn)
 * ASTERISK-28423 - ARI causes STASIS Deadlock
      by Ross Beer)
 * ASTERISK-28714 - REGRESSION: Feature
      subscription_persistence_recreate (ASTERISK-27759) Causes
      (Reported by Ross Beer)
 * 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-28706 - silk 24hHz doesn't show up in 'core show
      translation' output
      (Reported by Sean Bright)
 * ASTERISK-24484 - Update documentation for statsd module -
      usage requirements unclear
      (Reported by Dan Jenkins)
 * 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-28349 - Pause reason not reported in QueueMember AMI
      (Reported by Niksa Baldun)
 * ASTERISK-21794 - CLI command 'realtime update2' syntax
      failure when using according to usage help
      (Reported by
      Cedric BASSAGET)
 * 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-27243 - contrib: valgrind.supp doesn't suppress what
      it's supposed to due to invalid syntax
      (Reported by
      Richard Kenner)
 * ASTERISK-28497 - func_odbc: truncating Unicode string on
      (Reported by Boris P. Korzun)
 * ASTERISK-28647 - chan_sip: RTP frames not transmitted after
      emitting a COLP
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28667 - Asterisk ignores parsing of config files if
      a Byte order mark is present
      (Reported by Robin Leffmann)
 * ASTERISK-28664 - "trustrpid" is misspelled in
      (Reported by Pascal Cadotte Michaud)
 * 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-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
      fails to deactivate CDR.
      (Reported by Frederic LE FOLL)
 * 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-28644 - Stale comment in app_queue about ring_entry
      (Reported by Walter Doekes)
 * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
      UTF-8 string on hangup when TEST_FRAMEWORK enabled
      (Reported by Bernhard Schmidt)
 * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
      compatibility check failure when negociated ptime is not default
      (Reported by Frederic LE FOLL)
 * 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-28625 - Playback of local files impacted by large
      media cache
      (Reported by Kevin Reeves)
 * 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-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-28086 - chan_pjsip: Crash when initiating PlayDTMF
      over AMI
      (Reported by Jeremiah Gadd)
 * 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-28538 - chan_pjsip: Deadlock on fax detection
      (Reported by Joshua C. Colp)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
      (Reported by Guido Falsi)
 * ASTERISK-23756 - setvar directive when used in template and a
      child of said template, results in duplicate variable names
      (Reported by Michael Goryainov)
 * ASTERISK-28511 - codec_resample: Bad sound quality when up
      sampling from SLIN16 to SLIN32
      (Reported by Ruddy G)
 * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
      PRI channel hangs up
      (Reported by Frederic LE FOLL)
 * ASTERISK-28527 - ChanIsAvail() creates a CDR if
      unanswered=yes is set in cdr.conf
      (Reported by Frederic LE
 * 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)
 * ASTERISK-28477 - Crash when not specifying "dbfile" in
      (Reported by Dennis)
 * ASTERISK-28478 - Crash performing "core reload" with modified
      (Reported by Dennis)
 * ASTERISK-26968 - chan_pjsip: Transfer() does not result in
      TRANSFERSTATUS reflecting SIP response to transfer
      (Reported by Dan Cropp)
 * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
      deadlocks (in chan_sip)
      (Reported by Walter Doekes)
 * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
      systems caused by ASTERISK-28317
      (Reported by abelbeck)
 * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable
      (Reported by Michael Maier)
 * ASTERISK-26006 - Show offending IP for TLS setup failures in
      (Reported by Oleksandr Natalenko)
 * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
      not logged
      (Reported by Bernhard Schmidt)
 * ASTERISK-28419 - app_amd: Does not work with silence
      (Reported by Nasir Iqbal)
 * ASTERISK-28018 - IP Fragmentation happening instead of DTLS
      fragmentation on handshake server hello certificate
      (Reported by vijay kumar)
 * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
      Asterisk attempts to generate hangup event
      (Reported by
      Abhay Gupta)
 * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
      (Reported by Dmitry Svyatogorov)
 * ASTERISK-27981 - res_fax: Fax session leak with fax
      (Reported by pasandev)
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
      source files, causes build failure
      (Reported by Guido
 * ASTERISK-28421 - Wrong type used for timestamp in
      (Reported by Morten Tryfoss)
 * ASTERISK-27994 - PJSIP: Early media ringback not indicated
      after Progress()
      (Reported by Gregory Massel)
 * ASTERISK-28412 - GCC 9 catches more string formatting issues

      (Reported by George Joseph)
 * ASTERISK-28379 - pjsip: show channelstats incorrect
      information output
      (Reported by Vyrva Igor)
 * ASTERISK-28399 - channel.c: Exceptionally long queue length
      (Reported by Abhay Gupta)
 * ASTERISK-28392 - The no-partial-inlining flag isn't passed to
      the bundled pjproject or jansson builds
      (Reported by
      George Joseph)
 * ASTERISK-28402 - res_pjsip_registrar: SEGV in
      (Reported by Ross Beer)
 * ASTERISK-27756 - bridge: Failure to impart a channel results
      in bad data causing crash
      (Reported by Abhay Gupta)
 * ASTERISK-26718 - ARI: Bridge destroying doesn't work as
      (Reported by Marin Odrljin)
 * ASTERISK-28143 - app_amd: Infinite loop on silent calls 
      (Reported by Abhay Gupta)
 * ASTERISK-28353 - stasis: Crash at shutdown when statistics
      (Reported by Joshua C. Colp)
 * ASTERISK-28374 - latest asterisk unconditionally launch gcc
      --version, even if the compiler is different
      (Reported by
      Guido Falsi)
 * ASTERISK-28391 - res_indications: Crash requesting
      autocomplete on indications cli command
      (Reported by Lucas
 * ASTERISK-27935 - app_voicemail: emailbody per user can't
      contain commas
      (Reported by S��bastien Duthil)
 * ASTERISK-17695 - extenpatternmatchnew=yes cannot find
      extensions with '-' in them
      (Reported by test011)
 * ASTERISK-17799 - AEL reload causes loss of control in a
      (Reported by Kirill Katsnelson)
 * ASTERISK-18593 - AEL for loops use Macro app and pipe
      (Reported by Luke-Jr)
 * ASTERISK-14939 - AEL parsers does not find existing label
      (Reported by klaus3000)
 * ASTERISK-20182 - Parsing a label beginning with a numeric
      character in all Goto/GotoIf/GotoIfTime application causes
      unexpected behavior
      (Reported by Janu)
 * ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323
      (Reported by Dmitry Shubin)
 * ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
      lead to both inband and info
      (Reported by Salah Ahmed)
 * ASTERISK-28319 - musl: Crash on startup when loading modules

      (Reported by Sebastian Kemper)
 * ASTERISK-28362 - strtok_r() makes gcc compile warning
      (Reported by sungtae kim)
 * ASTERISK-28255 - res_rtp_asterisk: REMB RTCP packet sending
      may be incorrect
      (Reported by Joshua C. Colp)

Improvements made in this release:
 * ASTERISK-28787 - res_pjsip_session: Decide more intelligently
      when to add video
      (Reported by Joshua C. Colp)
 * ASTERISK-28733 - stream: Add support for adding/removing
      streams during SFU/calls
      (Reported by Joshua C. Colp)
 * ASTERISK-28710 - Should be able to disable the /httpstatus
      URI in the built-in HTTP server
      (Reported by Sean Bright)
 * ASTERISK-28638 - Simplify dialplan for Dial, Page, and
      (Reported by cmaj)
 * ASTERISK-28673 - GET FULL VARIABLE documentation
      (Reported by Jonathan Harris)
 * 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)
 * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support
      for DUNDi
      (Reported by Kirsty Tyerman)
 * ASTERISK-28401 - app_confbridge: Add *_all remb behavior
      (Reported by Joshua C. Colp)
 * ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add
      support for transport-cc
      (Reported by Joshua C. Colp)
 * ASTERISK-28363 - Millisecond-resolution call stats including
      PDD in channel variables
      (Reported by Antoni Goldstein)
 * ASTERISK-20207 - Asterisk should clear out any .lock files in
      the voice mail directory on startup.
      (Reported by Steven
 * ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
      work with.
      (Reported by Corey Farrell)
 * ASTERISK-28343 - Added app_name, app_data to channel type
      (Reported by sungtae kim)
 * ASTERISK-28264 - Added topic_all container
      (Reported by
      sungtae kim)

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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