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Asterisk 16.1.0-rc1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.1.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.1.0-rc1 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-rc1

    Thank you for your continued support of Asterisk!