[asterisk-dev] Asterisk 16.0.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Aug 8 16:17:32 CDT 2018


The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.0.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.0.0-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 candidate:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-27807 - iostreams: Potential DoS when client
      connection closed prematurely
      (Reported by Sean Bright)
 * ASTERISK-27818 - Username bruteforce is possible when using
      ACL with PJSIP
      (Reported by John)
 * ASTERISK-27658 - WebSocket frames with 0 sized payload causes
      DoS
      (Reported by Sean Bright)
 * ASTERISK-27583 - Segmentation fault occurs in asterisk with
      an invalid SDP fmtp attribute
      (Reported by Sandro Gauci)
 * ASTERISK-27582 - Segmentation fault occurs in Asterisk with
      an invalid SDP media format description
      (Reported by
      Sandro Gauci)
 * ASTERISK-27618 - Crash occurs when sending a repeated number
      of INVITE messages over TCP or TLS transport
      (Reported by
      Sandro Gauci)
 * ASTERISK-27640 - SUBSCRIBE message with a large Accept value
      causes stack corruption
      (Reported by Sandro Gauci)

New Features made in this release:
-----------------------------------
 * ASTERISK-27286 - Add the ability to read the media file type
      from HTTP header for playback
      (Reported by Gaurav Khurana)
 * ASTERISK-27704 - Add cache_pools debug option to
      pjproject.conf
      (Reported by Richard Mudgett)
 * ASTERISK-27581 - Add new AMI Action for PJSIPShowContacts
   
      (Reported by sungtae kim)
 * ASTERISK-27547 - res_pjsip: Add new AMI Action for
      PJSIPShowAuths
      (Reported by sungtae kim)
 * ASTERISK-27117 - core: Add support for timelen parsing to
      ast_parse_arg and ACO.
      (Reported by Corey Farrell)
 * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
      incoming INVITE Request-URI.
      (Reported by Richard Mudgett)
 * ASTERISK-27413 - Add cache_media_frames debugging option.
   
      (Reported by Richard Mudgett)
 * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
      endpoint identification to IP only
      (Reported by Ben
      Merrills)
 * ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
     
      (Reported by Thomas Sevestre)
 * ASTERISK-27322 - [New Feature] Add mute and DTMF passthrough
      to ARI add channel to bridge
      (Reported by Darren Sessions)
 * ASTERISK-27162 - [patch]chan_sip: Access incoming SIP REFER
      headers in the dialplan
      (Reported by Kirill Katsnelson)
 * ASTERISK-27163 - chan_sip: Dialplan function SIP_HEADERS() to
      complement SIP_HEADER().
      (Reported by Kirill Katsnelson)
 * ASTERISK-27063 - Add support for systemd socket activation
  
      (Reported by Corey Farrell)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27978 - res_pjsip: Change default transport
      keepalive to preserve behavior
      (Reported by Joshua Colp)
 * ASTERISK-27880 - [patch] pjproject_bundled: Repair
      ./configure --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27810 - BASIC-RETRANS: Implement receive
     
      (Reported by Benjamin Keith Ford)
 * ASTERISK-27972 - res_sorcery_config: Allow object name based
      matching
      (Reported by Joshua Colp)
 * ASTERISK-27965 - module: Remove old modules, update support
      levels
      (Reported by Joshua Colp)
 * ASTERISK-25548 - stasis: Improve message type "Use of before
      init/after destruction" error 
      (Reported by Joshua Colp)
 * ASTERISK-27967 - srtp: rejecting short sdes lifetimes
      incompatible with obihai ATAs
      (Reported by Nick French)
 * ASTERISK-27961 - res_pjsip: Spurious ERROR logging when
      printing headers in sip_msg
      (Reported by Nick French)
 * ASTERISK-27563 - pjsip modules always get -O2 even when
      DONT_OPTIMIZE is set
      (Reported by George Joseph)
 * ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS
      keep-alives.
      (Reported by Alexander Traud)
 * ASTERISK-27957 - PJSIP proposes ICE candidates on answer even
      if not in offer
      (Reported by Torrey Searle)
 * ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST'
      undeclared.
      (Reported by Alexander Traud)
 * ASTERISK-27955 - res_pjsip_session: sdp group:BUNDLE
      attribute truncated
      (Reported by Kevin Harwell)
 * ASTERISK-27956 -  res_pjsip_pubsub: segfault in function
      publish_expire
      (Reported by Alexei Gradinari)
 * ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not
      correctly handle two Reason headers
      (Reported by Ross
      Beer)
 * ASTERISK-27763 - res_pjsip_session: Initial INVITE with
      audio+fax results in 488 instead of declining stream
     
      (Reported by Thiago Coutinho)
 * ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause
      58(Bearer capability not available)
      (Reported by Jared
      Hull)
 * ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection
      if remote leg has T.38 disabled
      (Reported by Torrey
      Searle)
 * ASTERISK-26686 - res_pjsip: Lock inversion in transport
      management
      (Reported by Ross Beer)
 * ASTERISK-27939 - [patch] bridge_softmix_binaural: Enable
      FFTW3 in Solaris 11.
      (Reported by Alexander Traud)
 * ASTERISK-27783 - res_pjsip_pubsub: apparent crash on
      shutdown
      (Reported by Kevin Harwell)
 * ASTERISK-27870 - app_confbridge: Conference bridge and
      announcer channels are not removed if conference is ended as
      soon as it starts
      (Reported by Robert Mordec)
 * ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch
      and submit_unscheduled_batch
      (Reported by Denis Lebedev)
 * ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading
      module pbx_dundi.so with dundi peers
      (Reported by Kirsty
      Tyerman)
 * ASTERISK-27943 - AMI: Action SendText needs to use the
      correct thread.
      (Reported by Richard Mudgett)
 * ASTERISK-27942 - res_pjsip_messaging doesn't accept
      application/* content-types.
      (Reported by George Joseph)
 * ASTERISK-27936 - res_pjsip_session doesn't update media when
      a 200 comes in with a different port than a 183
      (Reported
      by George Joseph)
 * ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11.
   
      (Reported by Alexander Traud)
 * ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective
    
      (Reported by Corey Farrell)
 * ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in
      Solaris 11.
      (Reported by Alexander Traud)
 * ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not
      POSIX compatible.
      (Reported by Alexander Traud)
 * ASTERISK-27903 - menuselect: GCC 8: restrict-qualified
      parameter passed and aliased.
      (Reported by Alexander
      Traud)
 * ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure
      --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27705 - chan_iax2: Stops listening for traffic
     
      (Reported by Kirsty Tyerman)
 * ASTERISK-27848 - rtp: DTMF Breaks With telephony-event/16000

      (Reported by Dominic)
 * ASTERISK-27908 - [patch] crypto.h: Repair ./configure
      --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27905 - [patch] res_srtp: Repair ./configure
      --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27888 - SQL fetch error on query which return 0
      columns
      (Reported by Alexei Gradinari)
 * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
      responses
      (Reported by George Joseph)
 * ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated
      before terminating nul.
      (Reported by Alexander Traud)
 * ASTERISK-27872 - res_pjsip: Modified qualify_frequency
      doesn't effect until pjsip reload
      (Reported by Alexei
      Gradinari)
 * ASTERISK-27094 - res_fax: Deadlock when using Local channels
      and fax gateway
      (Reported by David Brillert)
 * ASTERISK-25261 - Manager events for MeetMe have incorrectly
      documented key name 'Usernum' - should be 'User'
      (Reported
      by Francois Blackburn)
 * ASTERISK-27878 - [patch] tcptls.h: Repair ./configure
      --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured
      with no-dh.
      (Reported by Alexander Traud)
 * ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x
      configured with enable-ssl3-method no-deprecated.
     
      (Reported by Alexander Traud)
 * ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause
      marker bit error
      (Reported by Torrey Searle)
 * ASTERISK-27831 - res_rtp_asterisk: Add support for
      abs-send-time RTP extension
      (Reported by Joshua Colp)
 * ASTERISK-27863 - config/ast_destroy_realtime_fields:
      successful DELETE is treated as failed
      (Reported by Alexei
      Gradinari)
 * ASTERISK-27865 - [patch]: tcptls: Repair ./configure
      --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails
      with recent MariaDB version.
      (Reported by Nic Colledge)
 * ASTERISK-27853 - Incorrect error reported when
      leaving/retrieving a ODBC voicemail
      (Reported by Nic
      Colledge)
 * ASTERISK-27726 - chan_mobile: presents incorrect inbound
      Caller-ID names
      (Reported by Brian)
 * ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip:
      Unregister the module for headers.
      (Reported by Alexander
      Traud)
 * ASTERISK-27852 - cli: "manager show settings" mislabels HTTP
      timeout as being minutes.
      (Reported by Corey Farrell)
 * ASTERISK-27824 - Fix issues exposed by GCC 8
      (Reported
      by George Joseph)
 * ASTERISK-27850 - [patch] rtp_engine: Allow Media Formats with
      add_static_payload(-1) on egress again.
      (Reported by
      Alexander Traud)
 * ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3
      compatibility.
      (Reported by Alexander Traud)
 * ASTERISK-27841 - digest over for manager (ami) over http
      fails on too long uris
      (Reported by Jaco Kroon)
 * ASTERISK-26570 - Macro allows an infinite loop of dialplan
      inclusion resulting in a crash
      (Reported by Tzafrir Cohen)
 * ASTERISK-27572 - cdr_mysql creates empty records if
      reconnects when mysql was not up on module load
      (Reported
      by Tzafrir Cohen)
 * ASTERISK-27801 - Asterisk got stuck while enabling "ari set
      debug all on"
      (Reported by shaurya jain)
 * ASTERISK-27795 - chan_sip: one way / no audio with srtp
     
      (Reported by Florian Kaiser)
 * ASTERISK-27800 - One way audio when calling from Asterisk(sip
      trunk) to another number where both are connected to a SBC using
      TLS+SRTP
      (Reported by Artur Pires)
 * ASTERISK-26806 - pjsip_options: rework to make more
      efficient
      (Reported by Kevin Harwell)
 * ASTERISK-27814 - translate: interpolated frames are not
      passed through
      (Reported by Kevin Harwell)
 * ASTERISK-27812 - When the  ooh323 debug is on there is no
      ringing signal to incoming calls via H323 trunk.
      (Reported
      by Dimos)
 * ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if
      debug is enabled only on the module
      (Reported by Marco
      Giordani)
 * ASTERISK-27804 - bridge_softmix / app_confbridge: Add support
      for combining REMB reports
      (Reported by Joshua Colp)
 * ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on
      FreeBSD and DragonFly BSD.
      (Reported by Alexander Traud)
 * ASTERISK-27418 - app_confbridge: "core show profile bridge"
      does not output "sfu" when video_mode is sfu
      (Reported by
      Carlos Chavez)
 * ASTERISK-27809 - [patch] utils/pval: Add -lBlocksRuntime for
      compiler clang conditionally.
      (Reported by Alexander
      Traud)
 * ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field
      designator extension.
      (Reported by Alexander Traud)
 * ASTERISK-27806 - BASIC-RETRANS: Implement send
     
      (Reported by Benjamin Keith Ford)
 * ASTERISK-27774 - res_musiconhold: Music on hold restarts
      after every announcement
      (Reported by lvl)
 * ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition
   
      (Reported by Evandro C��sar Arruda)
 * ASTERISK-27614 - res_pjsip_session: SDP origin does not use
      resolved address
      (Reported by John M.)
 * ASTERISK-27776 - res_rtp_asterisk: Add support for sending
      RTCP feedback messages
      (Reported by Joshua Colp)
 * ASTERISK-27740 - chan_sip: New Channel creation from new SIP
      dialog with Replaces failed to be properly tracked and
      destroyed
      (Reported by Shannon Price)
 * ASTERISK-27786 - app_confbridge: Add ability to enable and
      configure REMB support
      (Reported by Joshua Colp)
 * ASTERISK-27706 - PJSIP: Deadlock shutting down subscription
      TCP connection and sending subscription message.
      (Reported
      by Ross Beer)
 * ASTERISK-27688 - res_pjsip: Crash on TCP PJSIP Transport
      Disconnect
      (Reported by Ross Beer)
 * ASTERISK-27758 - res_rtp_asterisk: Add support for raising
      RTCP feedback messages
      (Reported by Joshua Colp)
 * ASTERISK-26366 - rtp: RTCP messages with REMB trigger fast
      picture update
      (Reported by Joshua Colp)
 * ASTERISK-27773 - Command line not being parsed correctly with
      getopt not from glibc
      (Reported by Guido Falsi)
 * ASTERISK-27435 - [patch] configure:
      pjsip_evsub_set_uas_timeout not found.
      (Reported by
      Alexander Traud)
 * ASTERISK-27761 - [patch] BuildSystem: With external editline,
      do not require libs for internal editline.
      (Reported by
      Alexander Traud)
 * ASTERISK-27755 - ConfBridge: raise ConfbridgeTalking when put
      on hold and clear talking status
      (Reported by Kevin
      Harwell)
 * ASTERISK-27743 - Generic PLC doesn't work if the 2 codecs on
      a channel are equal
      (Reported by George Joseph)
 * ASTERISK-27745 - [patch] BuildSystem: Remove unused
      dependency on libltdl.
      (Reported by Alexander Traud)
 * ASTERISK-12841 - [patch] Make format_ogg_vorbis work on
      OpenBSD
      (Reported by Michiel van Baak)
 * ASTERISK-27720 - [patch] BuildSystem: Enable Advanced Linux
      Sound Architecture (ALSA) in NetBSD.
      (Reported by
      Alexander Traud)
 * ASTERISK-27741 - res_pjsip_rfc3326.c
      rfc3326_use_reason_header doesn't account for more than one
      'Reason' header
      (Reported by Ross Beer)
 * ASTERISK-27734 - [patch] BuildSystem: Enable IMAP storage on
      openSUSE and Arch Linux.
      (Reported by Alexander Traud)
 * ASTERISK-27686 - [patch] install_prereq: Update FreeBSD
      libraries.
      (Reported by Alexander Traud)
 * ASTERISK-27733 - [patch] res_srtp: Add support for libsrtp2.x
      on openSUSE.
      (Reported by Alexander Traud)
 * ASTERISK-11015 - NetBSD Build Needs RPATH set in 1.2.25
     
      (Reported by Curt Sampson)
 * ASTERISK-27641 - BuildSystem: Enable Better Backtraces in
      FreeBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27671 - Deprecate legacy modules
      (Reported by
      Corey Farrell)
 * ASTERISK-25586 - uuid_generate_random detection failure
     
      (Reported by John Nemeth)
 * ASTERISK-27721 - [patch] BuildSystem: Enable PortAudio in
      NetBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27715 - [patch] BuildSystem: AC_PATH_PROG sets to
      colon character when not found.
      (Reported by Alexander
      Traud)
 * ASTERISK-27554 - res_pjsip_rfc3326: Order of 'Reason' headers
      break many endpoints
      (Reported by Ross Beer)
 * ASTERISK-27703 - AMI Action VoicemailUsersList returns 0
      MessageCount
      (Reported by S��bastien Duthil)
 * ASTERISK-27674 - chan_sip: RTP framing issues on outgoing
      calls
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-27441 - PJSIP: Forked INVITE SDP negotiation gets
      one way audio.
      (Reported by lvl)
 * ASTERISK-27718 - [patch] BuildSystem: Enable Lua in NetBSD.
 
      (Reported by Alexander Traud)
 * ASTERISK-27722 - [patch] BuildSystem: Depend not implicitly
      but explicitly on external libraries.
      (Reported by
      Alexander Traud)
 * ASTERISK-27719 - [patch] res_http_post: Enable GMime in
      NetBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27716 - [patch] BuildSystem: Enable autotools in
      NetBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27714 - [patch] chan_unistim: NetBSD has an
      incompatible struct in_pktinfo.
      (Reported by Alexander
      Traud)
 * ASTERISK-27713 - [patch] BuildSystem: Cast any intptr_t
      explicitly to its proposed type.
      (Reported by Alexander
      Traud)
 * ASTERISK-27712 - [patch] BuildSystem: Detect whether
      uselocale(.) is available.
      (Reported by Alexander Traud)
 * ASTERISK-27711 - [patch] BuildSystem: Avoid re-defining of
      pthread_* on NetBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27710 - [patch] BuildSystem: Install init scripts on
      openSUSE Tumbleweed.
      (Reported by Alexander Traud)
 * ASTERISK-27709 - [patch] BuildSystem: Avoid == for comparison
      in ./configure.
      (Reported by Alexander Traud)
 * ASTERISK-27610 - app_amd.so returning TOOLONG before reaching
      the timeout
      (Reported by Michael Cargile)
 * ASTERISK-26688 - Documentation: voicemail.conf.sample shows
      512 limit for emailbody field, however this is only true if
      compiled with LOW_MEMORY option
      (Reported by Fran Vicente)
 * ASTERISK-27568 - PJSIP: Crash during SIP attended transfer.
 
      (Reported by Bryan Walters)
 * ASTERISK-27659 - Output from rawman truncated if output is
      long enough
      (Reported by Bojan Nem��i��)
 * ASTERISK-27692 - bridging: Sometimes cloning the stream
      topology causes a crash
      (Reported by Richard Mudgett)
 * ASTERISK-27488 - core: If frame with unnegotiated format is
      read crash will occur
      (Reported by S��bastien Duthil)
 * ASTERISK-24488 - Wrong remote identity and target in dialog
      package XML in NOTIFY
      (Reported by Alejandro Padilla)
 * ASTERISK-24386 - Asterisk "doc/lang/language-criteria.txt"
      needs update or removal.
      (Reported by Rusty Newton)
 * ASTERISK-27646 - ICE fails with no candidate nominated
     
      (Reported by Thomas Guebels)
 * ASTERISK-27689 - [patch] rtp_engine: Load format name / mime
      type in uppercase again.
      (Reported by Alexander Traud)
 * ASTERISK-27679 - res_pjsip: Endpoint destruction does not
      free DTLS configuration
      (Reported by Mak Dee)
 * ASTERISK-27684 - [patch] install_prereq: Update OpenBSD
      libraries.
      (Reported by Alexander Traud)
 * ASTERISK-27680 - [patch] res_calendar: Specialized calendars
      depend on symbols of general calendar.
      (Reported by
      Alexander Traud)
 * ASTERISK-27681 - [patch] BuildSystem: Enable IMAP storage on
      OpenBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27677 - [patch] BuildSystem: Enable system provided
      libedit on OpenBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27670 - [patch] BuildSystem: Remove chan_h323
      leftovers.
      (Reported by Alexander Traud)
 * ASTERISK-27595 - [patch] BuildSystem: Invoke ldconfig with
      previous paths.
      (Reported by Alexander Traud)
 * ASTERISK-27631 - [patch] BuildSystem: Do not warn when bash
      is not installed.
      (Reported by Alexander Traud)
 * ASTERISK-27666 - chan_sip: Crash processing CANCEL request
  
      (Reported by Leandro Dardini)
 * ASTERISK-27584 - Internal pjproject build doesn't disable
      bcg729
      (Reported by Stuart Henderson)
 * ASTERISK-27669 - [patch] codecs: Add support for WebRTC iLBC
      2.0.
      (Reported by Alexander Traud)
 * ASTERISK-27634 - Determine if the internal editline and
      stdtime libraries are still relevant
      (Reported by George
      Joseph)
 * ASTERISK-27642 - [patch] backtrace: Avoid
      -Wlogical-not-parentheses.
      (Reported by Alexander Traud)
 * ASTERISK-27555 - [patch] install_prereq: Update Debian/Ubuntu
      libraries.
      (Reported by Alexander Traud)
 * ASTERISK-27656 - CDR: Leaking channel snapshots allocated by
      stasis_channel.c
      (Reported by Kristijan Vrban)
 * ASTERISK-27426 - chan_console: cannot read and write at the
      same time with alsa backend
      (Reported by Tzafrir Cohen)
 * ASTERISK-27621 - (null) string tailing after AsyncAGIEnd AMI
      event
      (Reported by sungtae kim)
 * ASTERISK-27652 - Null pointer Crash in PJSIP MWI
     
      (Reported by Joshua Elson)
 * ASTERISK-27571 - res_pjsip: If SIP response is received
      during shutdown a crash may occur
      (Reported by Joshua
      Colp)
 * ASTERISK-27619 - Build System: Require compiler to provide
      built-in support for atomic references.
      (Reported by Corey
      Farrell)
 * ASTERISK-27612 - Subscriptions Persist After Expiration and
      TCP/TLS Disconnect
      (Reported by Ross Beer)
 * ASTERISK-27637 - [patch] BuildSystem: Enable autotools in
      FreeBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27635 - [patch] app_voicemail: Avoid always true
      warnings with clang.
      (Reported by Alexander Traud)
 * ASTERISK-27599 - [patch] install_prereq: Update
      RHEL/CentOS/Fedora libraries.
      (Reported by Alexander
      Traud)
 * ASTERISK-26563 - core: macOS devmode build fails: variable
      'freeswap' set but not used
      (Reported by David M. Lee)
 * ASTERISK-27630 - [patch] editline: Avoid shifting a negative
      signed value.
      (Reported by Alexander Traud)
 * ASTERISK-16172 - Problems with siren14 codec; problems with
      siren7 sound files.
      (Reported by Steve Murphy)
 * ASTERISK-16951 - [patch] configure.ac in 1.4.37 broken with
      autoconf 2.60
      (Reported by St��phan Kochen)
 * ASTERISK-27603 - [patch] install_prereq: Download latest
      Jansson.
      (Reported by Alexander Traud)
 * ASTERISK-27620 - New module loader aborts startup if a
      required module declines load.
      (Reported by snuffy)
 * ASTERISK-27607 - [patch] res_config_mysql: Avoid the header
      mysql_version.h.
      (Reported by Alexander Traud)
 * ASTERISK-24598 - When running
      ./contrib/scripts/install_prereq install-unpackaged pjproject is
      installed in wrong place
      (Reported by PowerPBX)
 * ASTERISK-27602 - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs
      a directory.
      (Reported by Alexander Traud)
 * ASTERISK-27600 - [patch] BuildSystem: Allow make clean all
      again.
      (Reported by Alexander Traud)
 * ASTERISK-27598 - [patch] install_prereq: Support package
      manager DNF.
      (Reported by Alexander Traud)
 * ASTERISK-26596 - Placing call on hold temporarily locks up
      set
      (Reported by Igor Goncharovsky)
 * ASTERISK-27596 - [patch] BuildSystem: Use the detected name
      for MD5 everywhere.
      (Reported by Alexander Traud)
 * ASTERISK-27594 - [patch] BuildSystem: Invoke install not in
      GNU but POSIX style.
      (Reported by Alexander Traud)
 * ASTERISK-27593 - [patch] BuildSystem: In OpenBSD, xmlstarlet
      is xml.
      (Reported by Alexander Traud)
 * ASTERISK-27592 - [patch] BuildSystem: Detect external library
      Lua in version 5.3.
      (Reported by Alexander Traud)
 * ASTERISK-27491 - res_pjsip_endpoint_identifier_ip only
      matches against header if match by ip fails
      (Reported by
      George Joseph)
 * ASTERISK-26832 - res_pjsip: Segfault when calling
      pjsip_hdr_print_on in sip_msg.c:581
      (Reported by Ross
      Beer)
 * ASTERISK-27589 - [patch] BuildSystem: Avoid $EUID and use id
      -u instead.
      (Reported by Alexander Traud)
 * ASTERISK-27585 - [patch] BuildSystem: Resolve resolv.h not
      via Generic but Particular Header-Check.
      (Reported by
      Alexander Traud)
 * ASTERISK-27575 - menuselect : remove obsolete TRACE_FRAMES
      compiler flag
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-27576 - [patch] res_config_pgsql: Avoid typecasting
      an int to unsigned char.
      (Reported by Alexander Traud)
 * ASTERISK-27560 - [patch] clang 5 does not know
      -Wno-format-truncation
      (Reported by Alexander Traud)
 * ASTERISK-27578 - [patch] app_osplookup.c: Avoid a format
      truncation.
      (Reported by Alexander Traud)
 * ASTERISK-27577 - [patch] chan_ooh323: Avoid typecasting an
      int to unsigned short.
      (Reported by Alexander Traud)
 * ASTERISK-27534 - chan_sip: Assumes iostream is non-NULL when
      it may not be
      (Reported by Lubos Dolezel)
 * ASTERISK-27549 - [patch] translate: Avoid absolute value on
      unsigned substraction.
      (Reported by Alexander Traud)
 * ASTERISK-27566 - res_pjsip_session: Improve WebRTC interop
      with bundling during renegotiation
      (Reported by Joshua
      Colp)
 * ASTERISK-27553 - [patch] res_curl: Avoid error message on
      unload.
      (Reported by Alexander Traud)
 * ASTERISK-27557 - [patch] clang 5.0: implicit conversion to
      char changes value to negative.
      (Reported by Alexander
      Traud)
 * ASTERISK-27550 - [patch] bridge_softmix: Avoid warning about
      an uninitialized variable.
      (Reported by Alexander Traud)
 * ASTERISK-27559 - [patch] editline: Avoid comparison between
      pointer and zero character constant.
      (Reported by
      Alexander Traud)
 * ASTERISK-27558 - [patch] codec_gsm: Avoid shifting a negative
      signed value.
      (Reported by Alexander Traud)
 * ASTERISK-25329 - Asterisk configure fails on 'cannot find
      ptlib-config', despite ptlib-config existing
      (Reported by
      Rusty Newton)
 * ASTERISK-27552 - [patch] chan_ooh323: Limit outgoinglimit to
      positive values as intended.
      (Reported by Alexander Traud)
 * ASTERISK-27551 - [patch] ooh323cDriver: Fix typo in header
      guard.
      (Reported by Alexander Traud)
 * ASTERISK-26046 - [patch] Avoid obsolete warnings on
      autoconf.
      (Reported by Alexander Traud)
 * ASTERISK-20346 - Modules need to ensure that any functions,
      apps, AMI actions, etc. they register are unregistered if the
      module declines loading
      (Reported by Mark Michelson)
 * ASTERISK-27539 - 'cdr submit' fails: batch mode not enabled.

      (Reported by Tzafrir Cohen)
 * ASTERISK-27498 - ICE candidate parser - ICE foundation
      parsing too short
      (Reported by Michele Pr��)
 * ASTERISK-25128 - Datastore: Implement automatic module
      references.
      (Reported by Corey Farrell)
 * ASTERISK-27366 - Asterisk Turkish Language Set Problem
     
      (Reported by Halil ��brahim YILDIZ)
 * ASTERISK-23133 - Documentation fix - MASTER_CHANNEL
      Unexpected Behaviour
      (Reported by Shane Mitchell)
 * ASTERISK-27531 - Compiler optimizations can break module load
      sequence.
      (Reported by abelbeck)
 * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
      Contact crashes asterisk
      (Reported by Ross Beer)
 * ASTERISK-24198 - Typo's
      (Reported by Walter Doekes)
 * ASTERISK-27229 - bridge: Old channel video source not set to
      NULL after unref
      (Reported by Richard Kenner)
 * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
    
      (Reported by Corey Farrell)
 * ASTERISK-25079 - AMI bridge of channels results in MOH not
      destroyed and robotic audio on one channel
      (Reported by
      Zane Conkle)
 * ASTERISK-27490 - chan_console: 'set active' fails to work
   
      (Reported by Tzafrir Cohen)
 * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
      read()
      (Reported by Abhay Gupta)
 * ASTERISK-24756 - ConfBridge sound_muted does not work from
      CLI or AMI
      (Reported by Thomas Frederiksen)
 * ASTERISK-25649 - Transfer application does not work with
      Local channels - documentation misleading
      (Reported by
      Ivan Ullmann)
 * ASTERISK-25869 - chan_sip: "rejected because extension not
      found" should be logged as a security event
      (Reported by
      Brian J. Murrell)
 * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
      streams
      (Reported by Wim De Vlaminck)
 * ASTERISK-19657 - Coverity Report: Fix issues for error type
      CHAR_IO
      (Reported by Matt Jordan)
 * ASTERISK-27175 - iax.conf demo peer is invalid
     
      (Reported by Tzafrir Cohen)
 * ASTERISK-27430 - README refers to security documents that do
      not exist.
      (Reported by Corey Farrell)
 * ASTERISK-20281 - "core set verbose" behaves strangely, can't
      alias it, cli.conf example broken
      (Reported by Tim
      Ringenbach at Asteria Solutions Group)
 * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
      FXS gateway
      (Reported by Tzafrir Cohen)
 * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
      RTCP packet will write past where it should
      (Reported by
      Vitezslav Novy)
 * ASTERISK-27408 - Identify causes and fix
      pjsip/resolver/srv/failover/in_dialog/transport_tcp
     
      (Reported by Corey Farrell)
 * ASTERISK-18411 - Queue members with hints for state_interface
      get stuck in "In Use" state.
      (Reported by Steven T.
      Wheeler)
 * ASTERISK-26131 - chan_sip: Crash Asterisk (in
      sip_request_call at chan_sip.c) by making a call to a single
      character in a dot pattern match
      (Reported by Dwayne
      Hubbard)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
      not applied on reload
      (Reported by John Bigelow)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
      Variable CDR(amaflags)=...
      (Reported by Richard Mudgett)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
      results in one way audio.
      (Reported by Richard Mudgett)
 * ASTERISK-20643 - SIP ICE support - remove hardcoded
      limitation on SDP size, make ICE support disabled by default in
      SIP, maybe provide a better warning message
      (Reported by
      Roy)
 * ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS)
      if existing peer from same IP.
      (Reported by Alexander
      Traud)
 * ASTERISK-26980 - pjsip: Clean up WebRTC disables
     
      (Reported by abelbeck)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
      flooded with unauthenticated requests
      (Reported by George
      Joseph)
 * ASTERISK-27454 - res_http_post: Don't require
      GMIME_MAJOR_VERSION
      (Reported by Joshua Colp)
 * ASTERISK-23735 - Transcoding makes bad choice in high-rate
      translations
      (Reported by Richard Kenner)
 * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
      message.
      (Reported by Frank Durden)
 * ASTERISK-24662 - [patch] column and row headers for Signed
      Linear format variants in output of 'core show translation' are
      ambiguous
      (Reported by Rusty Newton)
 * ASTERISK-27353 - H323 audio starts with a delay of 2
      seconds.
      (Reported by Marco Giordani)
 * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
      media
      (Reported by Kevin Harwell)
 * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
      (srflx) with Dual-Stack.
      (Reported by Alexander Traud)
 * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
      IPv6 addresses.
      (Reported by Alexander Traud)
 * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra

      (Reported by Ivan Larionov)
 * ASTERISK-27431 - Asterisk fails to build when openssl headers
      are not installed.
      (Reported by Corey Farrell)
 * ASTERISK-27421 - RTP source learning not working with devices
      that have some clock issues
      (Reported by nappsoft)
 * ASTERISK-27361 - Attended transfer crashes in Asterisk
      13.17.2
      (Reported by Alessandro Pimenta)
 * ASTERISK-27238 - Bridging: Crash freeing a frame that's
      already been freed
      (Reported by Richard Kenner)
 * ASTERISK-27412 - core: Audiohook freeing interpolated frame
      when it shouldn't.
      (Reported by Mikhail)
 * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
      channel variable before closing the file
      (Reported by
      George Joseph)
 * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
      insert same ip address in "source ip address" and "destination
      ip address" fields in HEP packets
      (Reported by Max Norba)
 * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
      is equal to RemoteAddress)
      (Reported by Vasilii Rogin)
 * ASTERISK-27415 - asterisk.conf: Setting astctl without
      setting astrundir is ineffective.
      (Reported by Corey
      Farrell)
 * ASTERISK-27411 - pjsip: TCP connections may not be destroyed

      (Reported by Joshua Colp)
 * ASTERISK-27404 - DEBUG_FD_LEAKS does not record socketpair,
      timerfd_create or eventfd.
      (Reported by Corey Farrell)
 * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
      responses.
      (Reported by Corey Farrell)
 * ASTERISK-27337 - chan_sip: Security vulnerability with client
      code header (revisited)
      (Reported by Richard Mudgett)
 * ASTERISK-27319 - (Security) Function in PJSIP 2.7
      miscalculates the length of an unsigned long variable in 64bit
      machines
      (Reported by Kim youngsung)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)
 * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
      contact read from astdb or database
      (Reported by Aaron An)
 * ASTERISK-27290 - res_pjsip: PIDF contact field has
      malformed/invalid XML
      (Reported by basildane)
 * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
      values
      (Reported by seanchann.zhou)
 * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
      certificates
      (Reported by Sean Bright)
 * ASTERISK-26426 - format_ogg_opus: remove from source
     
      (Reported by Kevin Harwell)
 * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
    
      (Reported by Alexander Traud)
 * ASTERISK-27378 - Modules: Fix issues with CLI completion.
   
      (Reported by Corey Farrell)
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael
      Maier)
 * ASTERISK-27364 - channel: Crash when fax gateway is in use
      with PJSIP
      (Reported by Jared Hull)
 * ASTERISK-27390 - Audit menuselect module dependencies
     
      (Reported by Corey Farrell)
 * ASTERISK-27389 - Optional API modules should not allow
      unload.
      (Reported by Corey Farrell)
 * ASTERISK-27369 - Bridge() dialplan application fails without
      setting BRIDGERESULT channel variable
      (Reported by James
      Terhune)
 * ASTERISK-27067 - res_ari_channels: channel_state_invalid
      always leaks snapshot reference.
      (Reported by Marin
      Odrljin)
 * ASTERISK-27379 - stream: Allow streams on a topology to be
      put into groups
      (Reported by Joshua Colp)
 * ASTERISK-27374 - alembic: PJSIP scripts are missing column
      bundle in ps_endpoints table
      (Reported by Florian
      Floimair)
 * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
      documentation
      (Reported by Igor Goncharovsky)
 * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
      'imap_delete_old_greeting'
      (Reported by Anthony Messina)
 * ASTERISK-27194 - jitterbuffer: Does not handle case where
      translator returns null frame.
      (Reported by Joshua Elson)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * ASTERISK-26639 - core: Disabling xmldoc support does not
      work. Also results in abort during Asterisk startup.
     
      (Reported by Mr Dini)
 * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
      absence of the Expires header field with an unsubscribe action.

      (Reported by Jonathan Cloots)
 * ASTERISK-25960 - The config_hook unit test causes Asterisk to
      crash if run a second time
      (Reported by George Joseph)
 * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
      when rtp_ipv6 set to yes
      (Reported by Martin Cis��rik)
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
      but first on SDP media level.
      (Reported by Alexander
      Traud)
 * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
      Assertion on un/re-load: mod.id == -1
      (Reported by Tzafrir
      Cohen)
 * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
      enabling with conf file option - also 'sip set debug off'
      reports debugging disabled, when it really isn't
      (Reported
      by Rusty Newton)
 * ASTERISK-27350 - app_macro deprecation
      (Reported by
      Corey Farrell)
 * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
      any missing participant streams
      (Reported by Joshua Colp)
 * ASTERISK-27333 - sip_to_pjsip not correctly handling
      disallow=all directive
      (Reported by Torrey Searle)
 * ASTERISK-27343 - Fails to build in FreeBSD due to
      sys/sysmacros.h not existing there
      (Reported by Guido
      Falsi)
 * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
      (o=) contains local address.
      (Reported by Alexander Traud)
 * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
      configured codecs, but subset based on caller
      (Reported by
      lvl)
 * ASTERISK-27340 - backtrace.c: Crash due to double-free.
     
      (Reported by Corey Farrell)
 * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
      stopping.
      (Reported by Alexander Traud)
 * ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
      caller-id when it shouldn't be.
      (Reported by dtryba)
 * ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double
      user=phone parameters to URIs
      (Reported by dtryba)
 * ASTERISK-27301 - [patch] app_queue: Music On Hold for
      real-time queues is not reset to default
      (Reported by
      Nathan Bruning)
 * ASTERISK-25266 - Application Originate returns SUCCESS to
      ORIGINATE_STATUS upon failure to originate
      (Reported by
      Allen Ford)
 * ASTERISK-27270 - cdr_mysql: various crashes at second module
      reload if cdr_mysql.conf is configured
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-27328 - Missing openssl dependencies in
      res_rtp_asterisk and tcptls
      (Reported by Tzafrir Cohen)
 * ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
      unavailable endpoints
      (Reported by Richard Mudgett)
 * ASTERISK-27305 - res_ari: Memory leaks in ARI when using
      Content-Type: application/json
      (Reported by David Hajek)
 * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address

      (Reported by Ksenia)
 * ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
      IPv4 client via TCP/TLS
      (Reported by Alexander Traud)
 * ASTERISK-27317 - vector: multiple evaluation of elem in
      AST_VECTOR_ADD_SORTED.
      (Reported by Corey Farrell)
 * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
      ast_strings_match
      (Reported by Corey Farrell)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      (Reported
      by dtryba)
 * ASTERISK-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      (Reported
      by Beno��t Dereck-Tricot)
 * ASTERISK-27216 - app_queue: does its
      check-makeannouncement-logic twice each head-caller-loop
     
      (Reported by Stefan Engstr��m)
 * ASTERISK-27298 - Problem with expires on pjsip /
      outbound-publish
      (Reported by Cyrille Demaret)
 * ASTERISK-27295 - Contact is improperly translated after
      d178f497
      (Reported by Sean Bright)
 * ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
      (SSRC Changes)
      (Reported by Ross Beer)
 * ASTERISK-27289 - A codeblock that maintains a bug,but maybe
      the codeblock will never run
      (Reported by Huangyx)
 * ASTERISK-27277 - bridge: Renegotiate if source stream
      changes.
      (Reported by Joshua Colp)
 * ASTERISK-27264 - res_pjsip_session: Crashes after sending
      PRACK and receiving 200 OK
      (Reported by Daniel Heckl)
 * ASTERISK-27283 - Realtime config fail with PostgreSQL version
      before 9.1
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-27260 - [pjsip] chan_pjsip_indicate: Don't know how
      to indicate condition 36
      (Reported by Daniel Heckl)
 * ASTERISK-27257 - bridge_native_rtp: half-way direct media
      when using early bridging
      (Reported by Jean Aunis -
      Prescom)
 * ASTERISK-16898 - SRTP unprotect: authentication failure when
      RTP sequence number switches from 65535 -> 0
      (Reported by
      Marcello Ceschia)
 * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
      Possible PJSIP Vulnerability
      (Reported by Ross Beer)
 * ASTERISK-25524 - module reload res_calendar.so does not
      reload everything in calendar.conf
      (Reported by Jesper)
 * ASTERISK-27274 - RTCP needs better packet validation to
      resist port scans.
      (Reported by Richard Mudgett)
 * ASTERISK-27252 - RTP: One way audio with direct media and
      strictrtp=yes.
      (Reported by Richard Mudgett)
 * ASTERISK-24588 - res_calendar does not process CalDAV from
      Owncloud [fix included]
      (Reported by Stefan Gofferje)
 * ASTERISK-25523 - res_calendar: Warning about invalid channel
      value (for notification) occurs even when event has no
      notification configured.
      (Reported by Jesper)
 * ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
      wireshark disagree
      (Reported by Tzafrir Cohen)
 * ASTERISK-27248 - [patch]external_media_address and
      external_signaling_address don't always honor localnet
     
      (Reported by Walter Doekes)
 * ASTERISK-27165 - CDR: CDR(start,u) function won't work in
      cdr_custom config
      (Reported by Jacek Konieczny)
 * ASTERISK-24066 - res_smdi: convert to astobj2
      (Reported
      by Corey Farrell)
 * ASTERISK-27217 - chan_sip: Asterisk crashing when
      subscription doesn't get set
      (Reported by Bryan Walters)
 * ASTERISK-17540 - SDP origin attribute modified when issuing
      re-INVITE because of directmedia=yes
      (Reported by saghul)
 * ASTERISK-27254 - alembic: prune_on_boot fix erroneous
     
      (Reported by Florian Floimair)
 * ASTERISK-27232 - When in queue on g722 with interruptions,
      music on hold can get stuck and no longer play
      (Reported
      by Jens T.)
 * ASTERISK-27024 - nat/external_media settings ignored in
      14.4.1
      (Reported by Christopher van de Sande)
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
      local_net options are provided
      (Reported by Matt Jordan)
 * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
      channel_internal_api.c:478 during T.38 Fax Receive
     
      (Reported by Ross Beer)
 * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
     
      (Reported by Richard Kenner)
 * ASTERISK-27177 - ooh323c: misleading indentation in
      addons/ooh323c/src/ooSocket.c
      (Reported by Tzafrir Cohen)
 * ASTERISK-27241 - libc segfault upon entry into app_directory

      (Reported by David Moore)
 * ASTERISK-27152 - Sending a "tel" uri in a From or To header
      in an unauthenticated message causes asterisk to crash
     
      (Reported by Ross Beer)
 * ASTERISK-27103 - core: ast_safe_system command injection
      possible.
      (Reported by Corey Farrell)
 * ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
      with strict RTP enabled
      (Reported by Joshua Colp)
 * ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
      change due to renegotiation
      (Reported by Joshua Colp)
 * ASTERISK-26994 - Confbridge: CBAnn channels intermittently
      become stuck when caller hangs up before recording name
     
      (Reported by James Terhune)
 * ASTERISK-27222 - core: Don't queue up multiple video update
      frames.
      (Reported by Joshua Colp)
 * ASTERISK-20858 - app_minivm fails to clean up mkstemp files
 
      (Reported by Walter Doekes)
 * ASTERISK-16777 - several filename bugs in Record()
      application
      (Reported by klaus3000)
 * ASTERISK-27168 - alembic: PJSIP scripts are missing column
      dtls_fingerprint in ps_endpoints table
      (Reported by
      Florian Floimair)
 * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
      is used
      (Reported by Torrey Searle)
 * ASTERISK-19103 - When using realtime queues, function
      QUEUE_MEMBER_LIST() will return an error if no other
      app/function has loaded the queues first. This problem does not
      exist if queues.conf is used.
      (Reported by Jim Van
      Meggelen)
 * ASTERISK-21241 - When using voicemail as announce only
      (maxmsg=0), the star dtmf to enter the voicemail is not honored

      (Reported by Eelco Brolman)
 * ASTERISK-27212 - bridge_softmix: Quickly joining/leaving may
      cause video stream to remain in SFU
      (Reported by Richard
      Mudgett)
 * ASTERISK-27204 - [patch] app_queue: Wrong queue stat
      calculation
      (Reported by sungtae kim)
 * ASTERISK-27207 - XMPP OAuth not working due to inverted
      logic
      (Reported by Michael Kuron)
 * ASTERISK-27174 - res_calendar_icalendar: Recurring events not
      being loaded from Google calendar using ical
      (Reported by
      Mark Thompson)
 * ASTERISK-27202 - If wget is not installed and "or" is not
      available, external components (excluding pjsip) are not
      installed
      (Reported by Se��n C. McCord)
 * ASTERISK-27200 - manager: hook event is not being raised
    
      (Reported by Kevin Harwell)
 * ASTERISK-27147 - Either asterisk or pjproject isn't re-using
      tcp connections (again)
      (Reported by George Joseph)
 * ASTERISK-27193 - IPv6 receive address in message doesn't
      include brackets
      (Reported by Scott Griepentrog)
 * ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
      are not available when native bridge is used
      (Reported by
      Torrey Searle)
 * ASTERISK-26745 - Asymmetric codecs when
      asymmetric_rtp_codec=no
      (Reported by Jesse Ross)
 * ASTERISK-27189 - Make --with-pjproject-bundled the default
      for Asterisk 15
      (Reported by George Joseph)
 * ASTERISK-27110 - RTP session is not fully destroyed on
      channel hangup
      (Reported by Matt Jordan)
 * ASTERISK-27182 - bridge: Crash when mapping streams
     
      (Reported by Joshua Colp)
 * ASTERISK-27180 - channel: requester leaks joint_cap on
      success.
      (Reported by Corey Farrell)
 * ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
      incorrect
      (Reported by Kevin Harwell)
 * ASTERISK-27119 - res_pjsip: parse/add msid attribute when
      webrtc is enabled
      (Reported by Kevin Harwell)
 * ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
     
      (Reported by Ira Emus)
 * ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
      around the status element in XML
      (Reported by Abraham
      Liebsch)
 * ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
      devmode enabled.
      (Reported by Corey Farrell)
 * ASTERISK-27001 - res_pjsip: TLS connection not stable
     
      (Reported by Ian Gilmour)
 * ASTERISK-27130 - Applications ARI: Unsubscribe action for
      deviceStates does not remove old subscriptions properly
     
      (Reported by Sergej Kasumovic)
 * ASTERISK-25810 - say.c calls for sounds in the subdir
      "digits" that don't exist (in Core). SayUnixTime or other Say...
      apps will fail out when they call these sounds.
      (Reported
      by Nicolas Riendeau)
 * ASTERISK-27142 - sounds: Conflict between files in
      asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
     
      (Reported by Corey Farrell)
 * ASTERISK-27143 - bridge_softmix / res_rtp_asterisk: Fix
      packet loss and renegotiation issues.
      (Reported by Joshua
      Colp)

Improvements made in this release:
-----------------------------------
 * ASTERISK-22825 - Dialplan Function for Checking Parking Lot
      Slot
      (Reported by JoshE)
 * ASTERISK-27912 - [PATCH] Add predial handler to app_queue
   
      (Reported by Kristian H��gh)
 * ASTERISK-27929 - [patch] BuildSystem: Enable autotools in
      Solaris 11.
      (Reported by Alexander Traud)
 * ASTERISK-27752 - Ten seconds of silence after mp3 playback
  
      (Reported by Sam Wierema)
 * ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL
      configured with no-deprecated.
      (Reported by Alexander
      Traud)
 * ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured
      with no-deprecated.
      (Reported by Alexander Traud)
 * ASTERISK-27877 - app_confbridge: Add talking indicator for
      ConfBridgeList AMI response
      (Reported by William McCall)
 * ASTERISK-27873 - documentation: Error on wiki description of
      Asterisk 13 "MeetmeMute" event
      (Reported by Alessandro
      Polidori)
 * ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory
     
      (Reported by Ted G)
 * ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2
      configured with no-deprecated.
      (Reported by Alexander
      Traud)
 * ASTERISK-27796 - res_hep: Allow create_address to resolve a
      provided hostname
      (Reported by Sebastian Gutierrez)
 * ASTERISK-27820 - [patch] Add DragonFly BSD.
      (Reported
      by Alexander Traud)
 * ASTERISK-25129 - wrong automatic ras address assignment if
      multihomed
      (Reported by Dmitry Melekhov)
 * ASTERISK-27793 - cppcheck identifies redundant "if"
     
      (Reported by Ilya Shipitsin)
 * ASTERISK-27697 - Enable in-dialog NOTIFY on chan_pjsip
      channels
      (Reported by Nathan Bruning)
 * ASTERISK-27770 - [patch] install_prereq: Add Slackware
      (somehow).
      (Reported by Alexander Traud)
 * ASTERISK-27769 - [patch] install_prereq: Add Gentoo Linux.
  
      (Reported by Alexander Traud)
 * ASTERISK-27738 - [patch] install_prereq: Add Arch Linux.
    
      (Reported by Alexander Traud)
 * ASTERISK-27736 - [patch] install_prereq: Add SUSE.
     
      (Reported by Alexander Traud)
 * ASTERISK-27253 - [patch] libsrtp-2.1.x support
     
      (Reported by Alexander Traud)
 * ASTERISK-27728 - [patch] BuildSystem: Add NetBSD.
     
      (Reported by Alexander Traud)
 * ASTERISK-27730 - PJSIP: Update bundled PJPROJECT to version
      2.7.2
      (Reported by Richard Mudgett)
 * ASTERISK-27729 - [patch] install_prereq: Add NetBSD.
     
      (Reported by Alexander Traud)
 * ASTERISK-27683 - [patch] BuildSystem: Allow newer autotools
      on OpenBSD.
      (Reported by Alexander Traud)
 * ASTERISK-27348 - [patch]contrib/scripts: add a way to migrate
      from chan_sip to chan_pjsip realtime
      (Reported by Torrey
      Searle)
 * ASTERISK-27661 - Add new AMI Event for Load, Unload
     
      (Reported by sungtae kim)
 * ASTERISK-27651 - app_confbridge: Add Muted to ConfbridgeJoin
      and channel snapshot headers to ConfbridgeList AMI events
     
      (Reported by Richard Mudgett)
 * ASTERISK-27647 - app_confbridge/bridge_softmix: When channel
      muted report talking stopped if was talking.
      (Reported by
      Richard Mudgett)
 * ASTERISK-27084 - Reduce verbosity while loading PBX
      extensions.
      (Reported by Ludovic Gasc (Eyepea))
 * ASTERISK-24372 - [patch] Add config option to play a prompt
      to the "winner" in app_followme
      (Reported by Graham
      Mainwaring)
 * ASTERISK-27537 - res_pjsip: Add new AMI Action for
      PJSIPShowAors
      (Reported by sungtae kim)
 * ASTERISK-27483 - Allow wrapuptime to be set for each queue
      member
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24297 - cdr.c: Minor code optimizations.
     
      (Reported by Richard Mudgett)
 * ASTERISK-27470 - Add new object for VoicemailUserEntry
     
      (Reported by sungtae kim)
 * ASTERISK-27461 - 3PCC patch for AMI "SIPnotify"
     
      (Reported by Yasuhiko Kamata)
 * ASTERISK-27449 - [PATCH] When failing to acquire target
      during attended transfer, display wanted extension
     
      (Reported by Niklas Larsson)
 * ASTERISK-27456 - app_voicemail: Add new object for
      VoicemailUserEntry
      (Reported by sungtae kim)
 * ASTERISK-27380 - ast_coredumper: allow pointing out the
      asterisk binary explicitly
      (Reported by Tzafrir Cohen)
 * ASTERISK-23556 - Compilation warning for invert.c (array
      subscript is above array bounds)
      (Reported by Marcello
      Ceschia)
 * ASTERISK-27359 - pjproject bundled: Don't disable assertions
      when --enable-dev-mode is used.
      (Reported by Corey
      Farrell)
 * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
     
      (Reported by Richard Mudgett)
 * ASTERISK-27335 - CDR performance needs improvement.
     
      (Reported by Richard Mudgett)
 * ASTERISK-27278 - [patch] chan_sip: Provide access to read the
      full SIP Request-URI from INVITE
      (Reported by David J.
      Pryke)
 * ASTERISK-27255 - alembic: Add support for Microsoft SQL
      server
      (Reported by Florian Floimair)
 * ASTERISK-27220 - Enable CHANNEL function to get from and to
      tag from SIP Headers
      (Reported by Andre Nazario)
 * ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
  
      (Reported by Andrey)
 * ASTERISK-27173 - Support for GMIME 3.0
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
      chan_pjsip
      (Reported by Torrey Searle)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0-rc1

Thank you for your continued support of Asterisk!
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