[asterisk-dev] Asterisk 17.4.0-rc1 Now Available

[Asterisk Development Team]

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

The release of Asterisk 17.4.0-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:

Improvements made in this release:
-----------------------------------
 * 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)

Bugs fixed in this release:
-----------------------------------
 * 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)

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)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0-rc1

Thank you for your continued support of Asterisk!
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