[asterisk-dev] Asterisk 17.4.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Apr 30 10:41:06 CDT 2020 

The Asterisk Development Team would like to announce the release of Asterisk 17.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.4.0 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:

New Features made in this release:
-----------------------------------
 * ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits
      as non-root on Linux
      (Reported by Matt Addison)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28852 - Unprotected access to nochecksums variable,
      causes build failures
      (Reported by Guido Falsi)
 * ASTERISK-28846 - stream: Enforce formats immutability
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28847 - ARI channels cuts the endpoint string over
      80 characters
      (Reported by sungtae kim)
 * ASTERISK-28811 - Crash occurs when fax session switches from
      T.38 to audio
      (Reported by Alexey Vasilyev)
 * ASTERISK-28839 - Sporadic crashes with Segmentation fault
   
      (Reported by Joeran Vinzens)
 * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted

      (Reported by Daniel Heckl)
 * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port
      (TCP)
      (Reported by Anton Satskiy)
 * ASTERISK-24428 - Document that Asterisk will use the default
      SIP ports (5060 for TCP, 5061 for TLS) if the extern option
      variants aren't used
      (Reported by sstream)
 * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does
      not mention
      (Reported by Alexander Traud)
 * ASTERISK-28841 - app_confbridge: Add support for disabling
      text messaging for a user
      (Reported by Joshua C. Colp)
 * ASTERISK-28837 - pjproject_bundled: Honor
      --without-pjproject.
      (Reported by Alexander Traud)
 * ASTERISK-28827 - res_rtp_asterisk: Loop when receive buffer
      is flushed by a received packet that is also in receive buffer
      with NACK
      (Reported by nappsoft)
 * ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket,
      ignoring TCP and TLS sockets
      (Reported by Joshua Roys)
 * ASTERISK-28826 - res_rtp_asterisk: Duplicate seqnos being
      added to send buffer with NACK
      (Reported by nappsoft)
 * ASTERISK-28812 - First DTMF is not get
      (Reported by
      Bernard Merindol)
 * ASTERISK-28758 - pjsip startup errors when using "with-ssl"
      configure option
      (Reported by Patrick Wakano)
 * ASTERISK-28824 - BuildSystem: Search for Python/C API when
      possibly needed only.
      (Reported by Alexander Traud)
 * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python
      Programming Language is python-2.7.
      (Reported by Alexander
      Traud)
 * ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without
      server.
      (Reported by Alexander Traud)
 * ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not
      setup yet
      (Reported by Kevin Harwell)
 * ASTERISK-28819 - [patch] bridge_softmix_binaural: Show state
      in menuselect.
      (Reported by Alexander Traud)
 * ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and
      doc/pdf leftovers.
      (Reported by Alexander Traud)
 * ASTERISK-28818 - [patch] BuildSystem: Allow space in path.
  
      (Reported by Alexander Traud)
 * ASTERISK-28809 - [patch] res_rtp_asterisk: Avoid absolute
      value on unsigned subtraction.
      (Reported by Alexander
      Traud)
 * ASTERISK-28796 - func_channel: cannot read fields exten,
      context, userfield, channame from dialplan
      (Reported by
      S��bastien Duthil)
 * ASTERISK-28808 - [patch] test_stasis: Avoid always true
      warning with clang.
      (Reported by Alexander Traud)
 * ASTERISK-28803 - [patch] chan_unistim: Avoid tautological
      warnings with clang.
      (Reported by Alexander Traud)
 * ASTERISK-28056 - res_pjsip: Incorrect endpoint status after
      endpoint synchronization for a specific AOR
      (Reported by
      Jason Hord)
 * ASTERISK-28795 - channel: write to a stream on multi-frame
      writes
      (Reported by Kevin Harwell)
 * ASTERISK-28789 - test_utils: incorrectly printing error
      'declined to load'
      (Reported by Alexander Traud)
 * ASTERISK-28788 - func_aes: incorrectly printing error
      'declined to load'
      (Reported by Alexander Traud)
 * ASTERISK-28790 - Crash during conference call using
      confbridge and video
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd
      unless asterisk is running as root
      (Reported by Jaco
      Kroon)
 * ASTERISK-21205 - [patch] dundi_read_result crash due to
      negative number
      (Reported by Jaco Kroon)
 * ASTERISK-28784 - res_pjsip_sdp_rtp: Only do hold/unhold on
      first audio stream
      (Reported by Joshua C. Colp)
 * ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP

      (Reported by sungtae kim)
 * ASTERISK-28783 - res_pjsip_session: Allow default non-audio
      streams to have reflected state
      (Reported by Joshua C.
      Colp)
 * ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously
      triggered during direct-media (native_rtp) bridge
     
      (Reported by Michael Neuhauser)
 * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample
      are not consistent with examples. Missing examples.
     
      (Reported by Olivier Krief)
 * ASTERISK-28780 - app_mixmonitor: Memory leak due to race
      condition between AMI MixMonitor and hangup
      (Reported by
      Joshua C. Colp)
 * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp
      bridge is active
      (Reported by Torrey Searle)
 * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support
      is enabled but not used
      (Reported by Torrey Searle)
 * ASTERISK-28759 - A non negotiated rtp frame causes call
      disconnection when there is a SSRC change
      (Reported by
      Paulo Vicentini)
 * ASTERISK-26711 - func_enum: ENUM code wrong case
     
      (Reported by Vitold)
 * ASTERISK-23407 - Fix the FSF address in the headers of lots
      of pjproject files
      (Reported by Jared Smith)
 * ASTERISK-19460 - [patch] Function TXTCIDNAME never actually
      makes DNS calls and always returns an empty string
     
      (Reported by George Joseph)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28853 - Missing include on FreeBSD
      (Reported
      by Guido Falsi)
 * ASTERISK-28813 - func_volume: Allow decimal numbers as
      parameter to improve granularity
      (Reported by Jean Aunis -
      Prescom)
 * ASTERISK-27946 - dial (API): Storage of dialed target uses
      AST_MAX_EXTENSION when it shouldn't
      (Reported by Joshua
      Elson)
 * ASTERISK-28782 - Add support for Content-Disposition header
      in multi-part INVITES
      (Reported by Torrey Searle)
 * ASTERISK-28787 - res_pjsip_session: Decide more intelligently
      when to add video
      (Reported by Joshua C. Colp)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0

Thank you for your continued support of Asterisk!
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