Asterisk 16.1.0 Now Available


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The Asterisk Development Team would like to announce the release of Asterisk 16.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.1.0 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-28127 - Buffer overflow for DNS SRV/NAPTR records
  
      (Reported by Jan Hoffmann)
 * ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
      Upgrade requests
      (Reported by Sean Bright)

New Features made in this release:
-----------------------------------
 * ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
      in Contact header in chan_pjsip
      (Reported by Torrey
      Searle)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28151 - app_voicemail: MWI fails with
      mailboxes=##@device instead of mailboxes=##@default
     
      (Reported by Ronald Raikes)
 * ASTERISK-28125 - app_queue: Revert broken queue channel
      reference patch
      (Reported by lvl)
 * ASTERISK-28162 - [patch] need to reset DTMF last sequence
      number and timestamp on voice packet with marker bit
     
      (Reported by Alexei Gradinari)
 * ASTERISK-28159 - SIGABRT caused by stack corruption in
      hashkeys_read when no matching keys present
      (Reported by
      Michael Walton)
 * ASTERISK-28140 - repeated segmentation faults 
     
      (Reported by Eyal Hasson)
 * ASTERISK-28169 - ARI /channels/create handler causes core
      dump
      (Reported by sungtae kim)
 * ASTERISK-28103 - stasis: Filter messages at publishing to
      reduce work done
      (Reported by Joshua C. Colp)
 * ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
      Re-Invite omits routset
      (Reported by Torrey Searle)
 * ASTERISK-28158 - Some conditions prevent running of el_end,
      break the terminal.
      (Reported by Corey Farrell)
 * ASTERISK-28110 - rtp: Incorrect Packetization
      (Reported
      by Robert Cripps)
 * ASTERISK-28146 - pbx_config: Only the first [globals] section
      is processed.
      (Reported by Corey Farrell)
 * ASTERISK-28150 - Formatting error in documentation
     
      (Reported by Scott Griepentrog)
 * ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
      report AST_CEL_PICKUP in handle_invite_replaces
      (Reported
      by Luit van Drongelen)
 * ASTERISK-28137 - res_pjsip_notify: improve realtime
      performance on CLI completion on the endpoint
      (Reported by
      Alexei Gradinari)
 * ASTERISK-27980 - Caller ID cannot be changed on Attended
      Transfer before dialing out
      (Reported by Alexei Gradinari)
 * ASTERISK-28107 - app_confbridge:  Participant info labels
      aren't being added to the SDPs
      (Reported by George Joseph)
 * ASTERISK-28089 - function ast_sendtext() create RTP realtime
      packets with a trailing null byte in the payload
      (Reported
      by Emmanuel BUU)
 * ASTERISK-28076 - bridging: Asterisk crashes when receiving an
      empty realtime text frame
      (Reported by Emmanuel BUU)
 * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding
      AMI
      (Reported by Andrej)
 * ASTERISK-28077 - res_pjsip: improve realtime performance on
      CLI 'pjsip show contacts'
      (Reported by Alexei Gradinari)
 * ASTERISK-27920 - app_queue: Queue member considered inuse
      after immediately hanging up during dialing.
      (Reported by
      Cao Minh Hiep)
 * ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does
      not work
      (Reported by Cameron)
 * ASTERISK-28065 - res_odbc: missing SQL error diagnostic
     
      (Reported by Alexei Gradinari)
 * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
      differently to CLI
      (Reported by Peter Katzmann)
 * ASTERISK-28045 - configure script does not enforce
      libunbound2 version
      (Reported by Samuel Galarneau)
 * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
      ports below 10000
      (Reported by Joshua C. Colp)
 * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP
      instance can't be set up
      (Reported by Lei Fu)
 * ASTERISK-28034 - chan_sip unstable with TLS after asterisk
      start or reloads
      (Reported by David Hajek)
 * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
      2.8
      (Reported by Joshua C. Colp)
 * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
     
      (Reported by Sergej Kasumovic)
 * ASTERISK-28047 - chan_pjsip: Declined video stream is added
      when no video codecs configured and session refresh with removed
      video stream occurs
      (Reported by Will)
 * ASTERISK-28033 - AMI event "NewExten" is set to the wrong
      class
      (Reported by lvl)
 * ASTERISK-28049 - res_pjproject build failure
      (Reported
      by Jaco Kroon)
 * ASTERISK-28029 - [patch] res_musiconhold : music on hold will
      not start if previous hold just reached end of file
     
      (Reported by Frederic LE FOLL)
 * ASTERISK-28005 - channel.c: ARI ring only once
     
      (Reported by Hajek Michal)
 * ASTERISK-28032 - Realtime queuemembers are not updated during
      retry phase
      (Reported by lvl)
 * ASTERISK-27988 - alembic: PJSIP
      "mwi_subscribe_replaces_unsolicited" field is integer not
      boolean
      (Reported by Joshua C. Colp)
 * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
      'received' for IPv6
      (Reported by Sean Bright)
 * ASTERISK-28002 - When T.140 realtime text is negociated, a
      lot of debug traces are generated
      (Reported by Emmanuel
      BUU)
 * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
      authentification error
      (Reported by Ian Gilmour)
 * ASTERISK-28022 - res_pjsip realtime: uri column in
      ps_contacts table can be too short
      (Reported by Florian
      Floimair)
 * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
      other than 100 before 200 for T.38 reINVITE
      (Reported by
      Joshua Elson)
 * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of
      offer
      (Reported by Torrey Searle)
 * ASTERISK-27398 - No joint capabilities with video and
      audio-only streams
      (Reported by Benjamin Keith Ford)
 * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
      LEAVEEMPTY
      (Reported by Valentin Safonov)
 * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
      Do not undef s_addr.
      (Reported by Alexander Traud)
 * ASTERISK-27999 - Wrong SRTP use status report
      (Reported
      by Salah Ahmed)
 * ASTERISK-28001 - res_pjsip_registrar: Improve performance of
      inbound handling
      (Reported by Joshua C. Colp)
 * ASTERISK-27966 - pjsip: Race condition in 183 re transmission
      can result in a deadlock
      (Reported by Torrey Searle)
 * ASTERISK-15331 - make menuselect fails due to undefined
      symbols (initscr32, w32addch) in menuselect_curses.o
     
      (Reported by Majdi Bsoul)
 * ASTERISK-14935 - [regression] menuselect compilation failure
      on Solaris 10
      (Reported by Samuel Owens)
 * ASTERISK-12382 - menuselect compilation failure on Solaris 10
      / gcc 3.4.3
      (Reported by rleasure)
 * ASTERISK-9107 - menuselect compilation failure on Solaris
      10/gcc-4.1.1
      (Reported by Bob Atkins)
 * ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
 
      (Reported by Alexander Traud)
 * ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
      matches against "generic string" headers
      (Reported by
      George Joseph)
 * ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in
      Developer Mode.
      (Reported by Alexander Traud)
 * ASTERISK-27591 - Frack errors in stasis.c and memory leakage

      (Reported by Siruja Maharjan)
 * ASTERISK-27978 - res_pjsip: Change default transport
      keepalive to preserve behavior
      (Reported by Joshua C.
      Colp)
 * ASTERISK-27968 - systemd: asterisk.service
      (Reported by
      seanchann.zhou)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
      parse an URI and return a specified part of the URI
     
      (Reported by Alexei Gradinari)
 * ASTERISK-28136 - Allow the sip_to_pjsip script to be used in
      a pipe
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28046 - Remove stale nonoptreq references
     
      (Reported by Walter Doekes)
 * ASTERISK-27164 - [patch] Add IPv6 Support for DUNDi
     
      (Reported by Adam Secombe)
 * ASTERISK-28006 - PJSIP: Missing
      "party=calling"/"party=called" in Remote-Party-ID
     
      (Reported by Eric Dantie)
 * ASTERISK-27995 - pjproject_bundled: Find shared libraries in
      root --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27993 - pjsip_wizard example gives wrong info about
      unsupported SRV records
      (Reported by Jonathan Harris)
 * ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
      backspace or end of line are merged with regular text and it
      causes some UA to break
      (Reported by Emmanuel BUU)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0

Thank you for your continued support of Asterisk!

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